irclog2html for #asterisk on 20060729

00:00.34Un1xanyone here can please help me with a simple dailplan for my extensions.conf
00:04.03*** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com)
00:05.02crochatHello !
00:05.17*** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net)
00:05.27SpaceBassUn1x, have you checked the handbook?
00:05.41Qwell[]Un1x: There are at LEAST 3 different ways to call Digium
00:05.51SpaceBassUn1x, not being dismissive at all...just that I read it once and went cross-eyed then read it again and it started to make sens
00:06.08crochatI have a little problem with Asterisk... actually, I think the problem isn't Asterisk, it's the network... but I don't understand :-( I need help !
00:06.23SpaceBasscrochat, tcpdump is your friend :)
00:06.24Qwell[]Un1x: http://www.digium.com/en/company/contact.php
00:06.34CrashHDcrochat: you will get further by asking your question than you will by talking about it
00:06.40crochatSpaceBass: Sure, it's already done
00:07.08hads|homeUn1x: The free after sales support from Digium doesn't include setting up a dialplan AFAIK
00:08.02crochatSo now, I have two computers with exactly the same Asterisk configuration, and those computers are connected directly on the Net
00:08.14*** join/#asterisk watchy2 (n=wiit@h236.176.255.206.cable.cmdn.cablelynx.com)
00:08.30watchy2should i beable to call through my tivo using a ATA to asterisk?
00:09.02crochatI tried the tcpdump command : "tcpdump -i eth0 dst host 66.234.138.73" (66.234.138.73 is my SIP provider) on both computers
00:09.04CrashHDwatchy2: data is same as fax...you will have unreliable results
00:09.15watchy2yea thats what i thought
00:09.23watchy2i wonder how the shit im suppose to update this tivo then
00:09.37CrashHDnetwork connection
00:09.38crochatThis tcpdump command shows nothing on my server, but shows that on the second computer :
00:09.45crochat01:42:13.070013 IP 84-74-153-239.dclient.hispeed.ch.sip > 66.234.138.73.sip: SIP, length: 399
00:09.45crochat01:42:13.241658 IP 84-74-153-239.dclient.hispeed.ch.sip > 66.234.138.73.sip: SIP, length: 614
00:09.48CrashHDunless it's a direct tv tivo then you need to have a phone line
00:09.57watchy2it is directv tivo homie
00:10.02CrashHDyour screwed
00:10.06watchy2but the bitch is complainin about a phone call
00:10.17CrashHDmine use to do that every 6 months
00:10.21CrashHDI would just take it to a friends
00:10.25Un1xanyone wanna help me build a simple dail plan?
00:10.29watchy2yea ill probably just take it to work
00:10.32CrashHDstart the connection, and leave it over night
00:10.41watchy2so data is that unreliable over freakin voip huh?
00:10.46watchy2i kinda imagined it was
00:10.52watchy2but never inquired up on it
00:10.57CrashHDthere are some standards to allow data
00:11.06CrashHDbut not fully supported here, from what I'm told
00:12.21Un1xanyone wanna help me build a simple dail plan? please?
00:13.12*** join/#asterisk watchy2 (n=wiit@h236.176.255.206.cable.cmdn.cablelynx.com)
00:13.21watchy2i guess i need a new freakin cable provider to
00:13.44watchy2so i guess im just screwed. ill take my tivo to work and do it there i guess
00:15.08*** join/#asterisk Dr-Linux (n=ubuntu@202.59.73.131)
00:15.15*** join/#asterisk prepaid (n=nikhil@ip68-229-76-11.ri.ri.cox.net)
00:15.34prepaidI'm trying to user the Record cmd, but was wondering if anyone knows a way to make it beep 10 seconds before the max recording time?
00:15.37Dr-Linuxhi guys
00:16.49Dr-Linuxprepaid: put some wait after the beep sound?
00:17.41prepaidDr-Linux I guess, but I'm a little confused... I execute Record with max time = 50 seconds, and want it to beep at 10 seconds before 50, what exactly are you proposing?, playback a 40 second blank file with a beep at the end?
00:19.01Un1xshuit man
00:19.36Dr-Linuxno, i mean after the beep   put Wait(10)
00:20.10Dr-Linuxaww
00:20.23Dr-Linuxbeep is a part of record application?
00:20.23prepaidhuh
00:21.01prepaidit can be for the beep on record startup
00:21.07prepaidor you can just play teh 'beep.gsm' file
00:21.34Dr-Linuxyeah
00:21.44prepaidbut i don't get how to play a beep 40 seconds into a record
00:22.55*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
00:25.19dlynes_laptopNo, there's an indication in indications.conf for the beep
00:25.27dlynes_laptopNot quite sure where it's used in asterisk, though
00:25.50prepaidis there any idea on how to accomplish what i'm trying to do? something that seems like a rather simple request?
00:28.39dlynes_laptopprepaid, i don't see anywhere in the code where it's used for anything you'd need it for, either
00:28.41dlynes_laptopprepaid, gimme a sec
00:29.05*** join/#asterisk vader-- (n=johndoe@204.183.88.101)
00:30.39caio1982tzafrir_laptop: ping
00:33.18dlynes_laptopprepaid, have a look in apps/app_dial.c
00:33.36dlynes_laptopprepaid, then do a search for ast_get_indication_tone
00:33.50dlynes_laptopprepaid, you'll find some code around there on how to utilitize an indication tone
00:34.31Un1xdlynes man i been waiting for you
00:34.34dlynes_laptopprepaid, you'd probably need to modify some code in the apps/app_monitor.c code to make use of it
00:34.45Un1xi need help bro, i got asterisk libpri and zaptel installed on slackware
00:34.54dlynes_laptopprepaid, i'm assuming you're using Monitor(), and not Record(), anyways
00:34.54Un1xbut problem comes here that i need some help with configuration
00:34.59dlynes_laptopUn1x, ?
00:35.09Un1xextensions.conf need a dailplan i think
00:35.18tzafrir_laptopcaio1982, here
00:36.01dlynes_laptopUn1x, [incoming] exten => s,1,Dial(SIP/100) ; [outgoing] exten => _X.,1,Dial(SIP/terminator/${EXTEN})
00:36.07caio1982tzafrir_laptop: are you working on pkg-voip these days or are you a little bit away as your coworker said? i'd ask you about my last email... that new unicall stuff
00:36.47tzafrir_laptopI'm a little bit away
00:36.47dlynes_laptopeh?  I thought it was coppice that was doing unicall?
00:37.08caio1982dlynes_laptop: debian packages, not upstream code
00:37.09tzafrir_laptopI'm now back for a while, till Monday or so
00:37.18dlynes_laptopcaio1982, ah
00:37.35caio1982tzafrir_laptop: ah, ok
00:38.08tzafrir_laptopI have a laptop there and ton of free time. But no internet connection
00:38.37*** join/#asterisk detien (i=detien@unaffiliated/detienn)
00:38.45dlynes_laptoptzafrir_laptop, sounds like me...except the ton of free time part
00:38.45tzafrir_laptopI end up playing Wesnoth too long
00:39.01*** part/#asterisk detien (i=detien@unaffiliated/detienn)
00:41.21caio1982there where?
00:41.27caio1982:P
00:43.18*** join/#asterisk engineeer (i=1001@adsl-68-94-5-68.dsl.rcsntx.swbell.net)
00:47.40prepaiddlynes_laptop: sorry about the delay in responding... i'm actually using record, not monitor... i just want the user to know we're recording a 50 second clip, then warn them when there are 10 seconds up
00:49.06dlynes_laptopprepaid, there's an easier way to do it, you know?
00:49.21engineeeris there g729 support for the lastest svn branch out there somewhere or am I missing it somewhere
00:49.41hads|homeengineeer: Not at this stage aparantly.
00:49.54prepaiddlynes_laptop: nope no idea, fill me in :)
00:50.08dlynes_laptopprepaid, and you don't even have to patch any C code :)
00:50.18prepaiddlynes_laptop: i'm liking where this is going...
00:50.19engineeerok thanks about every TA I have uses g729
00:50.22dlynes_laptopprepaid, Use the dial command to dial into a Local/extension
00:51.03dlynes_laptopprepaid, so when they call into the extension, they'll get Dialed() into a Record() application
00:51.07hads|homeengineeer: I think they are going to release it sometime around when 1.4 goes into beta
00:51.14Un1xdylnes_laptop, msg please
00:51.25engineeerok thanks for the info
00:51.25dlynes_laptopUn1x, hold your horses
00:51.28Un1xkk
00:51.34dlynes_laptopUn1x, you're not the only person i'm talking to
00:51.39Un1xsry
00:51.53prepaiddlynes_laptop: okay... but how does that help with the beep 10 seconds before end.. or is that what you're getting at now
00:52.19caio1982hads|home: do you have URL with more info about it or something?
00:52.24dlynes_laptopprepaid, If you read the documentation for the dial command, you'll see there's some options to warn the user x number of seconds before the end of their call
00:53.00dlynes_laptopprepaid, It's a hack, but it should work :)
00:53.11hads|homecaio1982: No sorry, I just picked it up somewhere, dev list or here or something.
00:53.38dlynes_laptopprepaid, linux/unix is the swiss army knife of computing; asterisk is the swiss army knife of telecommunications :)
00:53.44hads|homecaio1982: So I could well be off, but that's what I heard.
00:54.16prepaiddlynes_laptop: ahhh very true.. was not aware of that feature in Dial  L(x[:y][:z])  :)
00:54.33caio1982hads|home: no problem, i'll try google... although i'm skeptical about it
00:54.46dlynes_laptopprepaid, exactly...you found the exact option i was talking about
00:54.54dlynes_laptopprepaid, just couldn't remember what the option was
00:55.09prepaiddlynes_laptop: it's okay... but i appreciate it, i never knew Dial had that option...
00:55.13dlynes_laptopprepaid, it's in there originally i think for calling card applications
00:55.50SwKanyone know how to hard factory reset Hitachi WIP5000 ?
00:56.36prepaiddlynes_laptop: yah that'd make sense... well i do appreciate the insight and the new idea, thanks!
00:57.06dlynes_laptopprepaid, well, i'm a programmer, so I always look for the easy way out :)
00:57.15dlynes_laptopprepaid, no point reinventing the wheel
00:57.23prepaiddlyes_laptop: hehehe doesn't everyone but much thanks yet again
00:58.20prepaiddlyes_laptop: so in the AGI i dial them to a record extention, but is there any easy way to get them back to the original calling AGI?
00:58.28prepaiddlyes_laptop: yah i'm obviously missing something simple within my head
00:58.45*** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie)
00:59.08*** join/#asterisk trbldwine (i=troubled@71.194.161.170)
01:00.07BugKhamthe mpg123 processes occupy 100% of my cpu
01:00.19BugKhamanyone had this problem before?
01:03.56Zodiacalanyone know why parkandannounce() doesn't speak the parked #? heres the wiki for parkandannounce() http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ParkAndAnnounce
01:04.15Zodiacalit does park for me tho
01:05.12dlynes_laptopprepaid, hrm....good question
01:05.27engineeerSwK  http://www.abptech.com/mainpages/support/hitachi_downloads.html
01:05.42prepaiddlynes_laptop: heh yah i thought maybe i was just overlooking something stupid
01:05.45dlynes_laptopprepaid, write it as a macro?  and pass a value to the macro, where the value is the extension it was dropped into, from?
01:06.37dlynes_laptopI don't know if that's possible though...I've never actually written a macro, so i'm not terribly knowledgable as to what their capabilities are
01:07.03prepaiddlynes_laptop: well hmm maybe, it's originally going to get called from an AGI since it's interacting with a database to see what the max record time can be.
01:07.36dlynes_laptopprepaid, and that's something i know even less about
01:07.43dlynes_laptopprepaid, i'm going to have to learn agi soon
01:07.48dlynes_laptopprepaid, but not just yet :)
01:08.10prepaiddlynes_laptop: well it's really simple... i appreciate your hack idea, i'll see if i can figure out a way to incorporate it, or maybe i just have to hack the source of the Record command
01:08.27*** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com)
01:08.47dlynes_laptopprepaid, well, you've got access to the dial(L(::)) command from agi though, don't you?
01:09.09prepaiddlynes_laptop: yes i do
01:09.21xbmodder_lappyDoes Digium still have its "Digium certified Expert"
01:09.35ghentoHi folks.  I am having a bit of a problem with asterisk.  Basically when someone calls from a cell phone, the extensions.conf works perfectly.  HOwever, I just had someone call from a landline, and it wasn't working at all.  It appears that Read() wasn't reading their input.  Has anyone experienced this before?
01:09.51dlynes_laptopxbmodder_lappy, if you can read chan_sip.c and full understand it, I'll certify you :)
01:10.16xbmodder_lappylol
01:10.24xbmodder_lappydlynes_laptop, do you work for digium?
01:10.31dlynes_laptopnope :)
01:10.36xbmodder_lappybecause I've always wanted to kill someone who works at digium
01:10.43Un1xlol
01:10.59dlynes_laptopxbmodder_lappy, #asterisk-dev is dominated by digium employees
01:11.09dlynes_laptopxbmodder_lappy, and a good number of them hang out in this channel, too
01:11.27*** join/#asterisk andymul (n=iCallAnd@cpe-69-203-217-237.nyc.res.rr.com)
01:11.30dlynes_laptopright, russellb ?
01:12.07andymulAnyone interested in some PHP/Asteirsk programming work please PM me
01:25.11*** join/#asterisk anthm (n=anthm@CPE-69-76-83-52.wi.res.rr.com)
01:25.11*** mode/#asterisk [+o anthm] by ChanServ
01:28.57*** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)
01:30.16*** join/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au)
01:33.41JunK-Ysome1 has been able to send the correct event to a gxp-2000, which makes it rebooting via a sip notify?
01:36.39Un1xby the way anyone spoofed caller id with asterisk before i was told how to do it it was setcallerid command, but forgot where and how...
01:37.25engineeerthere is an agi script to do that on demand
01:37.27*** join/#asterisk watchy2 (n=wiit@h236.176.255.206.cable.cmdn.cablelynx.com)
01:37.32watchy2anyone ever update a tivo or voip
01:38.23Un1xengineeer, i dont wanna use a agi script better to enter it into a conf or lift up the phone press somrthing like 11 and it ask u to enter the number u want to appear on the id
01:38.56engineeerthe script does all that for U from an ext but U have to use a providerthat will support it like voipjet
01:39.09*** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)
01:39.15*** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)
01:39.15Un1xor nufone
01:39.43engineeereither or I use voipjet for my "stuff"
01:39.53Un1xengineeer where can i get the script?
01:40.09engineeerI have it on my server but I think its on the wiki somewhere
01:40.35dlynes_laptopUn1x, are you using all analog lines?
01:40.41dlynes_laptopUn1x, if so, you can't spoof the caller id
01:40.56dlynes_laptopUn1x, you can only do that if you're using a pri
01:41.13dlynes_laptopUn1x, or a voip line, and most voip providers don't support it
01:41.39Un1xdlynes analog line?
01:41.45Un1xno im using VOIP to make outgoing calls...
01:41.48dlynes_laptopUn1x, pots/pstn/...
01:41.54Un1xnopes im not using those lines...
01:42.00Un1xim using voip only on my asterisk :)
01:42.20Un1xnufone publicy supports it and i know some others ;)
01:42.38dlynes_laptopwell, you told me you were using a tdm400p, so i thought you might be using one or two analog lines
01:43.28Un1xno not at all
01:43.34watchy2data over voip is impossilbe isnt it
01:44.00engineeerthe file I have is called cidspoof.agi but I dont remember if I renamed it
01:44.09*** join/#asterisk ComputerWarm (n=donc@209.29.157.109)
01:44.12Un1xdlynes i started asterisk but i didn't get a tone...
01:44.25Un1xengineer wanna send?
01:44.27ComputerWarmHEllo all
01:45.04dlynes_laptopdidn't get a tone?
01:45.05engineeerI can do that give me a sec to find it in the dir
01:45.08dlynes_laptopwhat's a tone?
01:45.25dlynes_laptopwatchy2, moip?
01:45.41ComputerWarmany agi programmers in just a quick question if there is. can anyone see a reason this wouldn`t work write ("SET CONTEXT callback"); write ("EXEC GoTO s|100");
01:45.50dlynes_laptopwatchy2, not impossible...just unreliable
01:45.50Un1xdlynes dailtone...
01:46.05dlynes_laptopUn1x, you need to configure your tdm400p
01:46.07dlynes_laptopUn1x, that's why
01:46.10*** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1)
01:46.17dlynes_laptopUn1x, and i'm not the person to talk to about that...never used one
01:46.21Un1xdamn
01:46.31hads|homeWhy do you have a TDM400 if you aren't using analog lines?
01:46.43dlynes_laptophads|home, he's using analog handsets
01:46.48watchy2dlynes_laptop: i was trying to update my tivo over voip using a sipura ata it dont seem to work
01:47.02dlynes_laptopwatchy2, heh..good luck on that
01:47.03JunK-YComputerWarm: you should stop ur agi, then start ur stuff at callback|s|100
01:47.12dlynes_laptopwatchy2, but force it to use ulaw
01:47.14hads|homeAh.
01:47.21watchy2dl: hrm that may help?
01:47.25dlynes_laptopwatchy2, you can't use any kind of compressed codec
01:47.29Un1xhads|home can ya help?
01:47.34dlynes_laptopwatchy2, or you'll guarantee data loss
01:48.03watchy2dl: it says in the info its using g711u right now
01:48.06watchy2what should i pick
01:48.21dlynes_laptopwatchy2, you definitely need to use ulaw, and make sure echo cancel isn't enabled, and make sure silence suppression isn't enabled
01:48.44asterisk-dudI have a tdm405p card for fxo ports and channel banks for fxs ports and asterisk keeps hanging up calls after about ten minutes when the come in for the fxo port and are routed to a fxs channel, can anyone help me?
01:48.45watchy2FAX CED Detect Enable how about that
01:48.58watchy2i dunno what that is
01:48.59dlynes_laptopwatchy2, also are you trying to download this tivo over voip stuff from outside your lan?
01:49.02ComputerWarmJunK-Y:  ok but if i execute the exit command before sending the caller back to the extensions config would i not lose them?
01:49.05dlynes_laptopwatchy2, don't worry about that
01:49.19watchy2dl: haha yea im trying to do it over nufone
01:49.28watchy2i got a feeling im fighting a losing battle?
01:49.28dlynes_laptopwatchy2, hah...good luck with that
01:49.42watchy2id have better luck going through a tdm400p?
01:49.45*** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net)
01:49.54watchy2or something connected to the pstn?
01:49.55dlynes_laptopwatchy2, you might have better luck with a premium voip provider, not a budget voip provider
01:50.05dlynes_laptopwatchy2, but even then, good luck
01:50.06watchy2well i use it for testing not production
01:50.13watchy2would i have better luck with pstn?
01:50.18dlynes_laptopwatchy2, and you'll need to use a really low baud rate, too
01:50.36ComputerWarm?
01:50.45watchy2yea
01:50.54watchy2fuck it ill just quit being lazy and take it to my parents
01:50.56dlynes_laptopwatchy2, you shouldn't have any problem with pstn, but again, if you're going to go pstn->ulaw->ata->modem, you'll need to use 9600bps or lower
01:51.15watchy2yea i dont think i can force my tivo to go that low
01:51.18watchy2so its impossible
01:51.29watchy2how do folks do pstn to ata to fax?
01:51.39dlynes_laptop9600 baud or lower
01:51.46watchy2ah
01:51.46dlynes_laptopand it's not 100% reliable, either
01:51.54watchy2so most people had dedicated fax lines i take it/
01:52.04dlynes_laptopwatchy2, pretty much, yeah
01:52.20JunK-Yu can save it in db?
01:52.43watchy2dlynes_laptop: ok well ill jus take it to my parents when the upgrades out
01:52.46dlynes_laptopwatchy2, i've seen people have good luck using fax2email -> internet -> email2fax using hylafax for the fax servers
01:53.02watchy2ah
01:53.10watchy2but no one sends fax through asterisk
01:53.18watchy2not in a production enviroment
01:53.19dlynes_laptopwatchy2, yes they do
01:53.27dlynes_laptoplike i said..hylafax
01:53.29watchy2oh
01:53.29engineeerun1x or Unix my screen cleared
01:53.37engineeerUn1x?
01:53.46dlynes_laptopthey use iaxmodem's virtual modems in combination with hylafax
01:53.53dlynes_laptopthey seem to get pretty good reliability with that
01:53.59watchy2oh
01:54.16dlynes_laptopother people get good reliability with app_txfax/app_rxfax
01:54.24dlynes_laptophowever, i've had 0% success rate with that
01:54.35*** join/#asterisk kb3nnj (n=theblue@c-69-140-159-42.hsd1.md.comcast.net)
01:54.40watchy2so like this co i deployed a voip box to has been having major issues
01:54.47watchy2so i do some checking using that zttest shit
01:54.53watchy2there irq shit is REALLY bad %
01:54.55andymulAnyone interested in some PHP/Asterisk programming work please PM me
01:55.04dlynes_laptopwatchy2, yeah...i ran into that, too
01:55.06watchy2i ran fucking hdparm -t and got 12mb/s
01:55.13dlynes_laptopwatchy2, couldn't get a zttest score over 88%
01:55.16watchy2i ran it on another box and got 150 hahah
01:55.24watchy2dl: replacing the box fix alot of issues/
01:55.33dlynes_laptopwatchy2, so i said forget it, tossed the digium card, went sangoma, never had a problem since
01:55.39watchy2haha
01:55.43*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)
01:55.52watchy2im having major issues with a tdm400p with echos on 8 lines
01:56.00watchy2i want to stab digium in the fucking throat for this bs
01:56.11dlynes_laptopi've got a sangoma a200 with hwec, and i've got 0 echo
01:56.14ComputerWarmanyone here willing to give me a head with this issue with the agi script?
01:56.18watchy2but im gonna replace the emachine they gave me to be a server
01:56.20*** part/#asterisk kb3nnj (n=theblue@c-69-140-159-42.hsd1.md.comcast.net)
01:56.24*** join/#asterisk Jason99 (n=jason@jason.unitz.ca)
01:56.24dlynes_laptopit sounds better than a nortel system
01:56.26watchy2and replace it with another box
01:56.42CrashHDdlynes can you recommend a good premium voip provider ?
01:56.44watchy2if replacing it with another box dont work im gonna call digium and tlel them shove the cards up there ass
01:56.51dlynes_laptopCrashHD, ME!
01:56.53dlynes_laptopheh
01:56.58dlynes_laptopj/k
01:57.04Jason99hello, i'm just wondering if someone can explain to me how the MWI works... What triggers Asterisk to tell the phone there is a VM and how does Asterisk know which phones to tell?  I could not find any documentation about this..
01:57.14CrashHDOEM you're the one I need to talk to I've trying to remember who I talked to on IR see about Felipe service
01:57.15*** join/#asterisk Sponge_bob (n=None@cpe-66-27-162-13.socal.res.rr.com)
01:57.15dlynes_laptopwe don't do voip to residential customers
01:57.30dlynes_laptoponly commercial
01:57.38CrashHDand testing out speech to text so you'll need to excuse my language
01:57.45watchy2do you use myspace dlynes_laptop
01:57.50dlynes_laptopnope
01:57.54CrashHDMy space is the devil
01:58.01watchy2you should i want to add u to my friends list
01:58.04dlynes_laptopit's just a blogging site, isn't it?
01:58.05watchy2and we can be emo together
01:58.10dlynes_laptopemo?
01:58.13watchy2dont you want to be emo with me
01:58.14dlynes_laptopsounds kinky
01:58.19CrashHDMy space is like a virus
01:58.22watchy2we can wear makeup and lipstick and cut on each other
01:58.24watchy2itll be great
01:58.59Jason99anyone know anything about message waiting indicator ?
01:59.05dlynes_laptopCrashHD, I think watchy2 might be your neighbor on myspace
01:59.18dlynes_laptopJason99, just ask your questio
01:59.24dlynes_laptop~suggestions
01:59.26jbotrumour has it, suggestions is 1) Don't ask to ask. Just say your problem, 2) Don't repeat until 5 mins after, 3) Read and re-read the docs first, then admit it if you REALLY don't understand. You're wasting your time and ours if you haven't at least tried. 4) If your problem ain't solved, come back in 12 hrs or 24 hrs later. We're very international. 5) Be polite ...
01:59.27Jason99I did above
01:59.28CrashHDlol
01:59.35Jason99What triggers Asterisk to tell the phone there is a VM and how does Asterisk know which phones to tell?  I could not find any documentation about this..
01:59.35Zodiacalanyone know of a way to saydigits when a blind transfer gets initated to the user that initated the blind transfer?
01:59.43CrashHDmyspace is just myspace
01:59.44Zodiacalor is a blind transfer final and cuts off the connection
01:59.45CrashHDlame
01:59.47dlynes_laptopJason99, the phone subscribes to mwi indication
01:59.56watchy2hehe
02:00.05watchy2i want to get high and stab kittens
02:00.07*** part/#asterisk Amilcar_ (n=amilcar@201.34.202.17)
02:00.20dlynes_laptopJason99, each time it checks mwi status, asterisk will let it know how many messages there is
02:00.24rene-wtf watchy2
02:00.32watchy2rene: im emo dude
02:00.33*** join/#asterisk mog_home (n=mogorman@68.62.237.103)
02:00.33*** mode/#asterisk [+o mog_home] by ChanServ
02:00.38Jason99dlynes_laptop: ok, I must be doing something wrong then.. ok thanks, let me take a look
02:00.39dlynes_laptop~emo
02:00.52rene-ahhh
02:00.59dlynes_laptop~wiki emo
02:01.10Jason99dlynes_laptop: Asterisk knows by the mailbox field in the sip.conf?
02:01.21dlynes_laptopJason99, correct
02:01.29dlynes_laptopJason99, mailbox=101
02:01.33dlynes_laptopJason99, mailbox=101@default
02:01.44dlynes_laptopJason99, mailbox=101@mycontext
02:01.54dlynes_laptopdefault is the default context for voicemail
02:02.00*** join/#asterisk engineeer (i=1001@adsl-68-94-5-68.dsl.rcsntx.swbell.net)
02:02.04dlynes_laptopso if you don't specify @context, it'll assume default
02:02.21Jason99dlynes_laptop: but which context would I want to put?  I don't get that
02:02.42watchy2whatever context your set your mailboxes under homie
02:02.56dlynes_laptopJason99, are all of your mailboxes defined in the 'default' context in your voicemail.conf file?
02:03.01rene-watchy2 i'd tought that emo dudes would be very much against stabbing kittens
02:03.13Jason99dlynes_laptop: ah, yes I get it
02:03.19watchy2rene: well i dunno im not really emo, i just play a emo fag on irc?
02:04.04dlynes_laptopwtf is emo music?
02:04.09rene-somebody told me emo really meant gay
02:04.26dlynes_laptopand i thought a screeching weasel was some kinda alcoholic drink?
02:04.31watchy2apparently emo music is dead
02:04.37watchy2it died like 10 years ago
02:04.41watchy2seriously
02:04.44rene-AIDS?
02:04.54watchy2maybe rene, it did go out with the 80s i think
02:05.05engineeerUn1x did U get you file?
02:05.23*** join/#asterisk AJaymn (n=FreePBX8@156-77.dsl.scc.net)
02:07.10rene-off
02:08.52watchy2how the fuck do you copy in putty?
02:08.55watchy2just highlight right
02:10.47*** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca)
02:14.41*** join/#asterisk Assid (i=assid@203.115.83.215)
02:15.17watchy2i want to meet a hotchick who shows me how to use asterisk and marrys me
02:15.34Assidwatchy2: good luck
02:15.59watchy2i can barely find girls who arent fat
02:16.00watchy2:(
02:16.55*** join/#asterisk BugKham (i=CKGLOB@221.128.110.41) [NETSPLIT VICTIM]
02:16.55*** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) [NETSPLIT VICTIM]
02:30.31dlynes_laptopwatchy2, stay out of mcdonald's then
02:30.50dlynes_laptopwatchy2, and the bar scene
02:31.05dlynes_laptopwatchy2, take up felching
02:31.07watchy2haha
02:31.54rob0watchy2 meets these fat girls, and to be polite, says, "For a fat girl you sure don't sweat too much!"
02:32.06watchy2haha
02:32.09watchy2im fat to
02:32.25dlynes_laptopall girls are the same, fat or not
02:32.32dlynes_laptopjust roll them in flour and find the wet spot
02:32.35rob0(Gomer Pyle, USMC ... an obscure reference to an old sitcom.)
02:34.01ComputerWarmguys question or maybe point me to some reading if you will i am trying to figure out how to update a mysql table of how long the caller was online and delete that amount from his account
02:34.23ComputerWarmi need to do this after he hangs up. and everytime i try the script terminates when the carrier hangs up
02:34.52CrashHDI would do agi
02:35.34ComputerWarmCrashHD:  thats how i am attempting to do it but when the caller hangs up it doesn`t update the database i can`t figure out how to get the script to continue until its completed the update
02:38.00CrashHDto continue? or to not continue until the agi call is done?
02:38.38ComputerWarmcontinue sorry
02:39.25CrashHDthat was a question
02:39.54CrashHDyou do not want to continue until the AGI has completed the update correct?
02:40.01CrashHDohh
02:40.09CrashHDI see
02:40.10CrashHDhmm
02:40.19CrashHDsoo this is an incoming call?
02:40.30ComputerWarmactually out going call
02:40.32CrashHDor outgoing?
02:40.44CrashHDusing the continue after disconnect flag for the dial command?
02:40.57ComputerWarmdidn`t know there was one
02:41.06CrashHDg? I think
02:41.07CrashHDlet me look
02:41.31CrashHD"g: When the called party hangs up, exit to execute more commands in the current context. "
02:41.51CrashHDI think that will fix your problem
02:41.58ComputerWarmok thank you
02:42.36ComputerWarmonce this is fixed its just a matter of figuring out the math of the billing increments and then test this thing
02:43.00*** part/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com)
02:43.02CrashHDsounds like a whole ton of fun
02:43.05*** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r)
02:43.56ComputerWarmCrashHD:  ya not really.. it was until i got stock on this part... but now i think with what you recommended it should work
02:43.58*** join/#asterisk racter (n=racter@c-67-167-206-229.hsd1.il.comcast.net)
02:44.19CrashHDwalls are meant to be climbed
02:44.38racterhey folks -- i'm looking for a really simple IVR solution and asterisk just looks so complicated; do i really need to buy all this hardware just to set up a one-line touchtone-navigable system?
02:45.27tuck3ra few days ago i upgraded to kernel 2.6.17 and zaptel 1.2.7, now in /var/log/messages i'm getting lots of "rtc: lost some interrupts at 1024Hz." not sure what caused it, any ideas?
02:45.32*** join/#asterisk sp0n9e (n=sp0n9e@69.12.216.48)
02:45.40tuck3rracter: just get a cheap ATA
02:47.17ractertuck3r: if i buy an ata, asterisk will recognize it and i can use asterisk to program a simple IVR?
02:47.48tuck3r"recognize", no you have to set it up but yeah
02:48.01ractercool
02:48.13watchy2hey whats it take to configure an fxs port on a tdm400p? besides /etc/zaptel.conf and zapata.conf
02:48.23watchy2ive never confed a fxo port on a tdm
02:48.52watchy2i mean an fxs
02:49.25dlynes_laptopwatchy2, that's it
02:49.37dlynes_laptopwatchy2, maybe adjust your gains and that kinda thing, too
02:49.37watchy2and you can pick up the phone and dial?
02:49.51watchy2it doesnt need to authenticate or anything?
02:49.51hads|homeDepending on your dialplan, yes.
02:49.58dlynes_laptopwatchy2, well, obviously you need to define a context for it, and write that context in your dial plan
02:50.05watchy2yea
02:50.06dlynes_laptopwatchy2, and set up an extension for it
02:50.11racterthx for the tip, tuck3r
02:50.23watchy2but if its setup in /etc/zaptel.conf zapata.conf should it get a dialtone after that?
02:50.47hads|homeOnce you have loaded the module and run ztcfg, yes.
02:50.53watchy2after an asterisk restart that it
02:51.10hads|homeYes.
02:51.46watchy2hrn i steped un1x through it and hes got asterisk back but hes not getting a dialtone for some reason
02:51.57watchy2<Un1x>       1            default         en         default
02:52.08watchy2port 1 and 2 are fxs ports on a tdm400p
02:53.59watchy2http://pastebin.ca/104673
02:54.07watchy2does that look correct for a fxsport on a tdm400p?
02:55.03hads|homelooks alright at a glance
02:56.09ComputerWarmCrashHD:  at times i am thinking its just to much work but oh well :-)
02:56.17*** join/#asterisk oadaeh (n=jason@wsip-24-234-160-51.lv.lv.cox.net)
02:56.29sp0n9ehow hard would it be for an average sysadmin to create a pbx with asterisk that has 10 incoming lines and connects 16 phones and provides a queue and an autoattendant?
02:56.40watchy2you know linux?
02:56.43ComputerWarmnot hard at all
02:56.47sp0n9epretty well
02:56.47ComputerWarmtake a few minutes to read
02:56.55watchy2then a few hours time and shit
02:56.58sp0n9ei've been flipping through a draft manual
02:57.09watchy2its pretty cool. i did it with 40 phones sp0n9e
02:57.14watchy2and im a newbie to asterisk
02:57.23watchy2i got like automated directory and shit up its pretty cool
02:57.35sp0n9eokay, i feel comfortable editing conf files, etc...just new to a lot of the phone terminology
02:57.46watchy2yea get the asterisk book
02:57.51sp0n9eplace of employment is moving and everything is getting upgraded
02:57.53watchy2itll explain alot and its good to read at home
02:58.00sp0n9erecommendations? i'll get my boss to buy one
02:58.09watchy2its a bright yellow book i dunno what its called
02:58.47sp0n9ehow much proc/memory would i need for 10 incoming lines and 16ish phones?
02:58.53hads|home~thebook
02:58.55jbotwell, thebook is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
02:59.13sp0n9eoooooOOOOooooo
02:59.14watchy2sp0n9e: a decent box i wouldnt use some gay amd shit
02:59.34hads|homeriiiight.
02:59.36tuck3ramd shit?
02:59.43watchy2im not a fan of amd
02:59.54watchy2intels new proccessor has raped them
02:59.57tuck3rnobody cares really
02:59.59hads|homeObviously. Although that doesn't make them shit.
03:00.10sp0n9ethe budget for this is probably around $1500
03:00.25watchy2you can easily get a box to handle all that on $1500
03:00.32sp0n9ei would prefer for everything to fit in the rack :)
03:00.34watchy2now phones is a different story
03:01.31sp0n9ewhat am i looking at for phones?
03:01.34watchy2id recommend polycoms, ciscos are nice but i liked my polys better
03:01.50sp0n9ewhat kinds of protocols should i look for?
03:01.53watchy2sip
03:02.04watchy2you gonna have a secretary ?
03:02.12sp0n9enot i
03:02.12watchy2watching all lines and doing transfers and stuff?
03:02.14*** join/#asterisk EvilDeshi (i=evildesh@oxford-bb-occam3-ws-100.dsl.maqs.net)
03:02.21sp0n9ewe may...eventually
03:02.22watchy2is there gonna be 1 chick answering all the phones?
03:02.25watchy2ah
03:02.38watchy2then just get some 301s i think or 501s, look at em both and compare
03:02.38sp0n9ethe ebay guy will probably do it
03:02.52*** part/#asterisk racter (n=racter@c-67-167-206-229.hsd1.il.comcast.net)
03:04.01*** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net)
03:04.06watchy2well it shouldnbt be to hard to do
03:04.13watchy2plus folks in this channel are always willing to help
03:04.34sp0n9eyeah, the channel looks big enough that there's help here 24/7 :)
03:04.44tuck3rjust don't take processor advise from them
03:04.45watchy2yea
03:05.00watchy2bookmark voip-info.org to
03:05.04watchy2that websites great
03:05.41hads|home~docs
03:05.42jboti heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
03:05.51sp0n9ewow
03:05.54engineeerpolys have the best speaker phones as well
03:08.22sp0n9ei'm getting 10 lines from a cox "combined services" fiber line...it's going to run through some type of cisco device (forgot the name) what type of hardware will i need?
03:08.32watchy2hrm
03:08.39watchy2is it coming out to be normal pstn lines?
03:09.14watchy2im guessing cox is going from digital cable to standard analog
03:09.18sp0n9ei'm looking through some of the information i have access to
03:09.23sp0n9ewell, it's off of fiber
03:09.23watchy2youll need analog cards
03:09.34sp0n9ebut i think it may come out analog
03:09.54watchy2if its coming out analog get some of those sangora analog cards with built in hardware echo
03:10.07watchy2i wont ever buy a card without built in hw echo again
03:13.04sp0n9ei'll do the rest of the research on-the-clock :)
03:13.07sp0n9ethanks for the feedback
03:13.25*** part/#asterisk sp0n9e (n=sp0n9e@69.12.216.48)
03:15.25watchy2we got dewds fxs workin
03:15.35watchy2i think he just had it pluged into the wrong hole
03:17.11watchy2can something be in more then 1 context in *?
03:17.27*** part/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r)
03:18.47*** join/#asterisk wunderkin (n=wunderki@216-19-202-8.getnet.net)
03:19.12watchy2i need some more kidbop cds
03:28.10Assidwoohoo
03:28.11Assidwassup
03:29.28Jason99One I have set a variable, how would I clear it?
03:29.58*** join/#asterisk coppice (n=chatzill@127.166.17.210.dyn.pacific.net.hk)
03:30.24Un1xok
03:30.31Un1xanyone here know someone who uses splitfinity
03:35.15engineeerdid U get your file?
03:36.31engineeerthat was for Un1x
03:42.48*** join/#asterisk Clausian (i=reginald@203-206-65-20.dyn.iinet.net.au)
03:43.18Clausiananyone here got FWD working with *?
03:47.55*** join/#asterisk bmg505 (n=leon@dsl-165-156-57.telkomadsl.co.za)
03:52.09Un1xengineeer yes i got the agi script
03:52.12Un1xfor cidspoofing :)
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04:04.10*** join/#asterisk xanderp (n=pphillip@74.133.18.245)
04:05.21xanderpcould someone please give me the $.02 explanation as to the major difference between astlinux and trixbox?
04:06.10Un1xALOT!
04:06.20Un1xfeatures support flexibility
04:06.22Un1xmany things my freind
04:07.05*** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com)
04:09.14xanderpi've not built a pbx before (* or any other) but am fair at linux.  i want to tinker with setting up a few phones connected via sip to an  box.  my end goal is to setup extensions for each of the bedrooms/kitchen/basement/etc... and have voicemail, etc, and have them able to call out my x100p over my vonage account.  any suggestions as to a good  distro to start with to learn?
04:09.59xanderpoops.. the asterisk's changed my text to bold :)
04:10.20hads|homexanderp: Trixbox is based on CentOS, it's large and includes FreePBX, FOP and many other things. It is designed to be run on a standard PC. Astlinux is a custom built distro specifically for running Asterisk on embedded systems such as a Soekris or Wrap and such.
04:10.56hads|homexanderp: IMHO the best way to start is to take the distro that you feel most comfortable with and then download and  install Asterisk from source.
04:11.32CrashHDagree'd
04:11.39xanderpok.. i'm going to install on a dual p3 550mhz box with 512meg ram.  it's got a 9gig hd.  it's an older machine i don't use for anything, so i thought i'd give this a go on it.
04:12.07hads|homeThat box will be plenty for your purpose.
04:12.30xanderpgentoo is my distro of choice, but to compile everything on this old tank would take forever... ;)
04:12.40hads|homeYou will probably run into issues with the X100P though, most people do.
04:12.56xanderpecho?
04:13.06hads|homeYes.
04:13.12xanderp(grrrrr)
04:13.20xanderpi knew it was cheap for a reason
04:13.37benjkX100P or more precisely Ambient MD3200 softmodems are good for providing a Zaptel timing source
04:13.43EyeCueIt's been fun on freebsd, far less of a heacache than i expected :)
04:13.49benjkthey are not good for anything else
04:14.11mog_homex100p arent that bad
04:14.35benjkif they were manufactured a few years ago, they might be acceptable
04:14.46hads|homexanderp: I'm not in the US, but Vonage is a VoIP provider right? There's not much point going from VoIP -> analog -> VoIP
04:14.52benjkbut if they are manufactured more recently, they are garbage
04:14.52mog_homeyeah i have true ones purchased by digium ^_^
04:14.58mog_homei didnt know that
04:15.07mog_homei knew that intel stopped making the chip, or so i heard
04:15.15benjkthat's because the chipset is not being manufactured anymore
04:15.21symlinkmog_home: you just grab everything you can eh?
04:15.32mog_homewhats that file
04:15.36xanderphads|home: this is just to get my head around asterisk, and then possibly replace the vonage for long term.
04:15.37benjkfirst Intel acquired Ambient, then they stopped making the chip
04:15.52symlinkmog_home: x100ps... *cough*usb adapters*cough*
04:15.52hads|homexanderp: Cool.
04:15.55benjknow the Chinese manufacturers use refurbished chips
04:16.05mog_homes100us sucj
04:16.08mog_homeer suck
04:16.15symlinkthey don't exist.
04:16.20mog_hometricky chinese theives
04:16.21benjkand even worse, some use left over chips that didn't pass quality control
04:16.49benjkbut they are good as Zaptel timing sources
04:17.01benjkjust don't use them as FXO interfaces
04:17.04xanderpat that point i would replace the x100p with straight voip to a provider, and possibly hardware ip phones, but for now, it's just alpha testing
04:17.29benjkyou can always buy a Sipura3000 SIP-FXO adapter
04:17.39benjkits about 70 USD or so
04:17.58mog_homeor tdm400p
04:18.00benjkworks reasonably well, certainly much better than those junk modems
04:18.03mog_homei think thats 00 or so
04:18.04hads|homexanderp: Fair enough, there's nothing wrong with what you have for testing. You may find it works fine for what you need.
04:18.08mog_homeer 100
04:18.21benjkTDM400 is not that economical if all you need is a single port
04:18.33mog_hometrue
04:19.08benjkalso, Sipura has support for many more caller ID schemes than Zaptel drivers
04:19.10xanderpi was looking at the sipra 2002 linksys box that is locked to earthlink, but didn't know if i could unlock it easily.  they were only 60 bucks.
04:19.28hads|homebenjk: Yes but the TDM400 can do things like distinctive ring and such that the SPA3102 can't.
04:19.30benjklocked ones are probably not such a good idea
04:19.38*** join/#asterisk tempest1 (n=asf@adsl-144-60-181.chs.bellsouth.net)
04:19.45Un1xi need some help
04:19.48Un1xwhen somone calls me
04:19.51Un1xthey get, Congrats thing
04:19.58benjkonly makes sense if it does detect caller ID properly though
04:20.04Un1xfrom asterisk on how they successfuly installed asterisk
04:20.08mog_homeUn1x: its your default context stuff
04:20.12mog_homeits still active
04:20.19xanderpare there ANY inexpensive adapters that can do like 4 lines of analog phones fair enough for home use?
04:20.22mog_homeyou need to edit extensions.conf
04:20.32mog_hometdm400p
04:20.48xanderpmog_home: that for me?
04:20.51Un1xyea imm do that in a bit
04:20.53benjkso if you happen to be in a place which uses a caller ID scheme that Zaptel doesn't support, then your disctintive ring feature isn;t going to do you any good
04:20.53mog_homeyes
04:21.05mog_homelol true benjk
04:21.05xanderpthanks will check on it.  what should i expect to $?
04:21.16mog_homeyou want 4 lines in or out
04:21.25mog_homei think its around 300 for that
04:21.45xanderp4 lines in the house that can call each other, the outside world, or be called by the outside world.
04:21.53benjkits generally about 70-80 USD per port no matter where you go
04:22.01hads|homebenjk: Huh? callerid is seperate from distinctive ring.
04:22.08xanderpthanks, nice to have a good rule of thumb
04:22.08mog_hometill you add a lot of ports that is
04:22.22benjkit is separate, but you do distinctive ring to indicate who is calling
04:22.25mog_homehttp://forum.gbadev.org/viewforum.php?f=20
04:22.35benjkso if you don't know who is calling, then what good is distinctive ring?
04:22.36mog_homeoops other thing im playing with
04:23.15xanderpwould i be ok using 2 sipra 2002's with 2 ports each?  or is that overly complicated for what i am trying to do?
04:23.29benjksipura 2000 is only for telephones
04:23.35benjknot for telephone *lines*
04:23.38hads|homebenjk, maybe it's different over here. Distinctive ring is used on POTS lines as a way of adding an extra number.
04:23.57hads|homebenjk: So you can differentiate between faxes or whatever.
04:24.09xanderpbenjk: so they won't sip 2 phones to the pbx as 2 different extensions?
04:24.11benjkthat's inbound distinctive ring yes
04:24.18*** join/#asterisk s0lid (n=jlq@210.213.240.222)
04:24.20benjkbut that too, depends on which caller ID scheme is used
04:24.46benjkso if your Zaptel device or driver doesn't support the caller ID scheme your telco uses than that feature will not be available to you
04:25.38benjkxanderp, there are two kinds of analog ports
04:25.48benjkones for connecting to the telco, called FXO ports
04:26.07benjkand ones for connecting a telephone, called FXS ports
04:26.27[TK]D-Fenderbenjk : He's got his head on straight.... you need to follow....
04:26.31benjkif you want to hook up two analog phones, then the sipura 2000 is fine, cause it has 2 FXS ports
04:37.03*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
04:37.03*** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.10 and Zaptel 1.2.7 released! (July 14, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx
04:37.06xanderpso to tinker and learn would trixbox be a good place to get my feet wet and have it at least hold my hand a little in the installation
04:37.12Clausiancan you try ringing 789140?
04:37.20[TK]D-Fenderbenjk : Though I'd love to know what you class as "nice" at $95
04:37.21benjktrixbox isn't for tinkering
04:37.31benjkACT 104
04:37.47[TK]D-Fenderbenjk : .jp available model?
04:37.57xanderpisn't it the next rev of the @home distro?
04:38.01[TK]D-Fenderbenjk : never heard of it over here.
04:38.08benjkyes, they also have JP display capable firmware
04:38.16hads|homexanderp: IMHO the best way to start is to take the distro that you feel most comfortable with and then download and install Asterisk from source.
04:38.20[TK]D-Fenderxanderp : you won't learn anything with Trixbox.  and it is the new name for A@H
04:38.35benjkits a Taiwanese manufacturer, called Advanced Century Telecommunications (or Technology?)
04:38.37xanderpok, i'll roll my own...
04:38.40[TK]D-Fenderbenjk : Got a link... would love to see something cheap AND good...
04:38.57benjklet me try to find it
04:38.58quid2478I'm using TB... but to be honest, I'm doing most of my work with the config files using vi. :)
04:39.08xanderp(3 days worth of gentoo compiling is not something i look forward to, but i do so love gentoo!)
04:39.29Un1xi hate vi
04:39.32Un1xit's terrible
04:39.38[TK]D-Fenderfile not found!!!!!
04:39.43benjkhttp://www.act-tel.com.tw/_pg/products/productItemR.ASP?ContentsManageID=11&UnitsManageName=IP%20Phone
04:39.47quid2478well that and nano, though I admit... i'm used to vi
04:39.47symlinkI know, it's sad :(
04:39.56benjkand they do have IAX2 firmware
04:40.03xanderpnano is my savior... i never can remember vi commands
04:40.17[TK]D-Fenderbenjk : I've seen that exact frame remarked under several names including GNET
04:40.18benjkbut you need to bother them a lot to get it
04:40.23Clausiancan someone ring fwd 789145 please?
04:40.27quid2478xanderg:  the thing that is drawing more to nano, is that it's the same editor off of old PC BBS software I used to use
04:40.28benjkyes, they are more in the OEM business
04:40.40quid2478Clausian:   Use the "ring me" feature of FWD
04:40.41benjkmany companies buy them in bulk and sell them under their own brand
04:40.46Clausianit never works
04:40.50Clausian:X
04:41.04Clausianmy softphone is registered fine though
04:41.06[TK]D-Fenderbenjk : Believeable, but I strongly suspect the quality wouldn't fall under my idea of "good"
04:41.09benjkHere in Japan, Fujistu rebrand those with a nicer enclosure and sell them against NECs rebranded Cisco 79xx
04:41.23quid2478I trashed FWD... who needs it
04:41.31benjkthey are good phones
04:41.58benjkhave you ever used any of those?
04:42.26[TK]D-Fenderbenjk : No, afer seeing GNET sell the ATCOM PA1688 stuff it made me shudder....
04:42.28benjkthe display is a little small
04:42.44benjkits not an ATCOM PA1688 phone
04:42.46[TK]D-Fenderbenjk : Looks too 1980 Radio-Shack for em...
04:43.05benjkthe industrial design of the enclosure isn't all too hip, true
04:43.09[TK]D-Fenderbenjk : jsut iguring any company selling it wouldn't be selling anything worthwhile :)
04:43.16benjkbut the phone is solid quality
04:43.19xanderpwell, it's late here, thanks for all the info everyone!
04:43.35benjkalso, it looks better in reality than it does on the photo
04:43.40*** part/#asterisk xanderp (n=pphillip@74.133.18.245)
04:43.57[TK]D-Fenderbenjk : Yeah, its kinda... BLEH.  But if you its FUNCTIONAL, then it has a place.  Then again, what is your take on Aastra?  Very classic look, basic usability, good price IMO.
04:43.58coppicepeople only judge phones by the case. how it works is largely irrelevant :-)
04:44.03benjkthe buttons are better than anything I have seen in the range up to 200 USD
04:44.29benjkyeah Aastra is a nice phone too
04:44.36benjklooks more stylish
04:44.39coppicethe buttons on a $1 analogue phone are fine, but on $50 IP phones they are nasty. something is definitely wrong there
04:44.48benjkTaiwanese aren't that good at industrial design
04:45.03benjkcoppice, :D
04:45.51[TK]D-Fendercoppice : Glad I pay over $100 for mine :)
04:46.10[TK]D-Fendercoppice : So I can get that incredible 1$ analog phone button feel!
04:46.13benjkin my opinion, the ACT 104s is the most solid IP phone in the price range up to 200 USD
04:46.30hads|homeThat's fairly bold
04:46.32coppiceyou're glad to pay >$100 for something that costs $15? weird
04:47.04Un1xwhy is zaptel doing this
04:47.04Un1x<PROTECTED>
04:47.04Un1x<PROTECTED>
04:47.04Un1x<PROTECTED>
04:47.04Un1x<PROTECTED>
04:47.06benjkthe looks are of course a different thing
04:47.10[TK]D-Fenderbenjk : I cannot for a second believe that it competes with a Polycom IP 501.......
04:47.13Un1xi pick up the phone and after i press like 4t5h digit to dail
04:47.17Un1xi get busy signalish
04:47.24Un1xzaptel hanging up on me why is that
04:47.29Clausiani have asterisk registered on fwd, and a softphone registered on another account. whenever i dial asterisk's fwd number, the softphone tells me its busy. why is asterisk doing this?
04:47.39benjkwell, my customers clearly like the ACT better than the Polycoms
04:47.54[TK]D-Fendercoppice : Find me a $15 source for this kind of quality then :)
04:48.08benjknobody is exactly thrilled by its looks, but functionality wise its there
04:48.31coppicebenjk: the taiwanese are good at industrial design. the ODMs there just know the case will get reworked to suit a big customer, so the original case the make isn't that important. you see this with lots of products. The final case will also be a taiwan design, but will look good
04:48.44[TK]D-Fenderbenjk : if you say so.. I can't fathom it.  2 line LCD vs pixel, high end speakerphone, nice solid feel, PoE options, .
04:48.54benjkcoppice, fair enough
04:49.23Un1xComon someoen help please...
04:49.26benjkas I said, Fujitsu is selling the P104s here in Japan with their own (more stylish) enclosure and sell it against NECs rebranded Cisco 79xx
04:49.38benjkfor 400 USD
04:49.46benjk:)
04:50.36benjkthey do have customised firmware though
04:51.37Un1xanyone know why
04:51.38Un1xzaptel fires up the channel and kills it
04:52.30De_monUn1x you broke it
04:52.32benjkthe P123 doesn't look too bad
04:52.34benjkhttp://www.act-tel.com.tw/_pg/products/productItemR.ASP?ContentsManageID=69&UnitsManageName=IP%20Phone
04:52.46benjkbit of a Cisco knock off
04:52.50symlinkUn1x: does the number you're dialing exist in the context that it will be searching?
04:53.01benjkI will have to enquire about those
04:53.13Un1xwhat do you mean symlink...
04:53.23Un1xim dailing a regular number...
04:53.30[TK]D-Fenderbenjk : Only the color scheme...
04:53.48coppicebenjk: well the ciscos are made in taiwan too :-)
04:53.50benjkthe general look
04:53.50symlinkUn1x: you have to tell Asterisk how to handle that stuff, it's not psychic - you tell it "if I dial this number, do these things" "if I dial a number matching this, do these things"
04:53.53[TK]D-Fenderbenjk : then again, I've seen that combo around all over anyways.... not much of a strech
04:54.02benjkthe display is still kinda smallish though
04:54.10benjkcould be a little larger
04:54.14Un1xoh yes thats in my extensions.conf
04:54.21Un1xit was working a minute ago, but after that it stopped working...
04:54.39coppicea bigger display really pushes up the BOM. It could go as high as $20 :-)
04:55.43Un1xweird if i dail long distance works perfectly
04:55.52Un1xbut anything within North America and it hangs up
04:56.26symlinkcheck your dialplan logic... it's probably not matching any extension in the context it is searching so it gives up
04:57.04Un1xsymlink would you like me to paste my extensions.conf on pastebin?
04:57.19symlinkyou can, but it's 2AM here and I'm slightly tired
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04:59.24Un1xhttp://pastebin.ca/104733
04:59.28Un1xsymlink; http://pastebin.ca/104733
05:00.04Clausiani have asterisk registered on fwd, and a softphone registered on another account. whenever i dial asterisk's fwd number, the softphone tells me its busy. why is asterisk doing this?
05:00.06symlinkand what are you dialing? and that's really insecure... someone could make calls out your box right now
05:00.25Un1xsymlink how, so and how can i secure it?
05:00.35symlinkyou really need to learn how the dialplan works
05:00.42Un1xim new to this man
05:00.52Un1xive gotten pretty far knowing nothing less then 5 hours ago
05:00.54hads|home~thebook
05:00.57jbotextra, extra, read all about it, thebook is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
05:00.57symlinkokay, but there's lots of docs out there - and a book
05:00.57Un1xand getting this runnign and stuff
05:01.02Un1xyea i got the book with my card
05:01.08Un1xi read it all twice, it dont explain too much
05:01.13Un1xso now im going to read more docs
05:01.19Un1xtry to fix it if i can and then go sleep
05:01.37symlinkso on IAX2 there's a user entry called guest, with no password... it goes to the context default
05:01.49symlinkif someone were to use that entry on your box right now, they could call any long distance or international number
05:02.34Un1xi see
05:03.13Un1xok well symlink wouldn't it be better to tell me how to temporarioly disable it or secure it rather then tell everyone how to get into my box lmao
05:03.46symlinkget rid of the entry in iax.conf for the guest user, or split up your contexts, or rename the default context
05:03.52benjkjust kidding
05:04.20Un1xsymlink; just comment this
05:04.21Un1x[guest]
05:04.21Un1xtype=user
05:04.21Un1xcontext=default
05:04.21Un1xcallerid="Guest IAX User"
05:04.46symlinkI really just suggest learning about contexts, and splitting it up and getting it out of your default context
05:05.11symlinkhaving your outbound dialing in default is just... not good
05:05.20Un1xwell i will learn more but i doubt i'll be able to learn all and go sleep in the next hour it's already 1 am here, so please help me out, and just tell me can i just comment that out, and feel safe and go sleep :P?
05:05.34benjkdeafult context should be like ...
05:05.35benjk[default]
05:05.35benjk;
05:05.36benjk; do not accept any calls to s@default
05:05.36benjkexten => s,1,NoOp(incoming connection attempt from ${CALLERID} to s@default)
05:05.36benjkexten => s,2,SetVar(PRI_CAUSE=21) ; Call reject
05:05.37benjkexten => s,3,Hangup
05:05.45hads|homePull the plug and you will be safe
05:05.45symlinkjust don't have Asterisk running...
05:05.51Un1x:|
05:05.56Un1xwhat a great idea thanks
05:06.11symlinklearn this stuff in the morning, try to read about dialplans again and ask logical questions
05:06.26benjkPRI_CAUSE is only there if you use a PRI (or BRIstuff)
05:07.04benjkin any event, you want to hangup in default, but you also want a record of the attempt, thus the NoOp()
05:07.10Un1xheh i commented out iax.conf :p
05:07.14*** join/#asterisk oej (n=oej@65.197.203.67)
05:07.36symlinkSIP has default as the default context too...
05:08.02ClausianJan  1 13:03:17 NOTICE[5427]: chan_sip.c:9683 handle_response_invite: Failed to authenticate on INVITE to '"unknown" <sip:789140@fwd.pulver.com>;tag=as6b79c6fe
05:08.05Clausianwhy it do this?
05:08.07benjkand mgcp and skinny and h323
05:08.18[TK]D-Fenderok, I'm fried.... later all...
05:08.27symlinkfried chicken!
05:09.13benjkClausian, because you "Failed to authenticate on INVITE"
05:09.30benjkyour password is wrong, or not there
05:09.32*** join/#asterisk sumasuma (n=sumase@cm222.omega183.maxonline.com.sg)
05:09.34Clausianwhat does that mean though? all the passwords etc are right
05:09.43Clausianasterisk is registered with it succesfully :X
05:09.45benjkobviously not
05:09.51sumasumawhich software is best to use with text to voice conversion?
05:09.52benjkthat's INVITE
05:09.55benjknot REGISTER
05:09.57sumasumaalong with asterisk
05:10.03benjkyou may have the right password for REGISTER
05:10.07benjkbut not for INVITEs
05:16.17*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
05:16.17*** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.10 and Zaptel 1.2.7 released! (July 14, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx
05:20.02dlynes_laptopClausian, because your register => ... line is correct and your [sipcontext] username=... ; secret=... is not correct
05:21.14*** join/#asterisk MikeJ (n=vircuser@d14-69-8-30.try.wideopenwest.com)
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05:24.50Clausianyes i fixed it
05:25.58Clausianbut now when i call my FWD number that asterisk is registered under on my softphone, it says 'Call Rejected: 486 Busy here'
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05:42.34Clausianhow can i tell asterisk to use a nat proxy?
05:48.34*** join/#asterisk Kerry_G (n=Kerry_G@ip70-187-129-227.oc.oc.cox.net)
05:49.07Kerry_Gare there any tools available to monitor trunks and alert if one goes down?
05:49.26sevardTOOORRNANDDOOO
05:49.42sevardCOMEON FUCKAR I GOT A LIL CAPTIN IN ME
05:54.54*** part/#asterisk Kerry_G (n=Kerry_G@ip70-187-129-227.oc.oc.cox.net)
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06:33.37L|NUXany one used chan_jabber application ?
06:35.04*** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com)
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06:44.06RaHaiLhi there
06:44.17RaHaiLis there any one here can make me interface
06:45.47sevardthere is an interface
06:46.09RaHaiLfor billing purpose and usage
06:46.18sevardi can make you interface
06:46.22sevardwith a large rock
06:46.27RaHaiL:)
06:46.38sevardfor your issue
06:46.40sevardcheck this out
06:46.43sevardgoogle perl
06:46.48sevardthat'll give you the tool you need
06:46.50sevardplus
06:46.55RaHaiLI am not a coder
06:47.00RaHaiLdont know nothing
06:47.03sevardinvent a system to take alcohol out of one's body
06:47.14sevardyou suck at english, that's for sure
06:47.26RaHaiL:) nop suck at gramer
06:47.29RaHaiLyou got it wrong
06:47.31RaHaiL:)
06:47.32sevardand spelling.
06:47.35RaHaiLyeah
06:47.38RaHaiLyou got it right
06:47.39sevardand sentence syntax
06:47.40sevardthus
06:47.44RaHaiLyeap
06:47.44sevardenglish
06:48.04sevardi'm drunk and i can still tell, dude that's bad.
06:48.12sevarde
06:48.21RaHaiLhehe I know I am trying my best to get rid of this bad habbit
06:48.30sevardm-w.com
06:48.35sevardspellcheck.com
06:49.20RaHaiLeh
06:49.30RaHaiLdo you think you can make me one interface
06:50.55L|NUXcan some one help me with asterisk + jabber
06:50.57sevardfor
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06:51.14sevardjabber is homogay
06:51.56RaHaiLheheh
06:51.58L|NUXsevard: hummmm
06:52.35sevardRaHaiL: i'm not going to build you an 'interface' but i'll help you build one yourself
06:52.45sevardyou ought to check out prebuilt bulling platforms though
06:52.54sevardlots listed on voip-info.org
06:53.03L|NUXRaHaiL : check this out http://www.voiceone.it
06:53.55sevardhahahaha
06:53.56sevardhttp://www.photosbydolph.com/SouthernModels/NerdFinalWebDCLogo.jpg'
06:54.02sevardhttp://www.photosbydolph.com/SouthernModels/NerdFinalWebDCLogo.jpg
06:54.44cy3o3I have an inquery.  What is the best reccommended (as far as cheap and CID spoofing goes) voip provider?
06:56.21RaHaiLno seriously
06:57.07*** join/#asterisk tengulre (n=tengulre@219.144.140.10)
06:57.25tengulreHi,all
06:58.01RaHaiLL|NUX
06:58.39tengulreanybody know why not registry when I running commmand on CLI> iax2 show registry?
06:59.27tengulreI have dual asterisk and distruble two defferent place.
07:00.32sevardcy3o3: most voip providers will allow you to pass whatever CID you send down the line, not CNAM, or course, shellshark, teliax... whatever.
07:01.10cy3o3Just kind of looking for reccommendations for a good company to go with .. pricing, support, uptime, etc..
07:01.16cy3o3what do you use sevard?
07:01.18sevardi gave you two.
07:01.33cy3o3Okay, well I have a huge list.  That's why I am torn between them all :/
07:01.56cy3o3I'll peep those two.. thx sevard
07:01.57sevardshellshark is pretty good. teliax is okay but more expensive
07:02.04cy3o3coo
07:02.07tengulreanybody can help me? pls!
07:02.27RaHaiLtrt telasip
07:02.35RaHaiLtry*
07:03.16tengulreHow can i connect two asterisks with IAX2 protocol?
07:05.20Juggiertfm
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07:07.15RaHaiL<PROTECTED>
07:07.24RaHaiLAny php coder here
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07:16.11cy3o3aight went with shellshark I guess
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07:24.39tainted_core dump core dump core dump YAY!
07:25.04RaHaiLif i know how
07:25.16tainted_just make calls in asterisk
07:27.24tengulreanybody know how to connect two asteirsk?
07:28.36RaHaiLtainted_
07:28.45RaHaiLdo you want help me with that small project
07:31.48RaHaiLi need some one help to get a interface for user where they can login and see there usage
07:32.09docelmohehe
07:32.23docelmothats simple enoguh if your MySQL CDR Logging
07:32.47RaHaiLI am using * home
07:32.51docelmoACK!
07:32.54docelmonevermind
07:32.56docelmoyour own your own
07:33.02RaHaiLoh man
07:33.18docelmocheck #freepbx
07:33.52pnlarssontengulre: http://www.voip-info.org/wiki/view/Asterisk+Connect+2+servers
07:34.22tengulrepnlarsson, thanks for answer! :)
07:36.27RaHaiLeh no on
07:36.33RaHaiLanway i guess no luck for me
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07:51.52tengulrepnlarsson: I can not registriy the iax2 to remote server, do u know why?
07:53.51tengulreI got : IAX2/219.233.118.37:4569-3 is circuit-busy
07:54.28tengulreanybod know why?
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08:03.53L|NUXany one used chan_jabber application ?
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09:12.22L|NUXjbot : !!!
09:12.23jbot\"Multiple exclamation marks,\" he went on, shaking his head, \"are a sure sign of a diseased mind.\" - Terry Pratchett, Eric
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09:19.43L|NUXany one worked on asterisk + gtalk ?
09:20.02QwellL|NUX: mog
09:20.12Qwellhe is teh man, when it comes to jabber
09:20.25*** join/#asterisk hack1 (i=1076@203.199.110.93)
09:20.26L|NUXQwell : i have little issue
09:20.31*** part/#asterisk hack1 (i=1076@203.199.110.93)
09:20.33L|NUXmight be some one can help me out :)
09:21.19*** join/#asterisk SanketMedhi (n=sanket@202.63.175.78)
09:21.35L|NUXQwell : mog is at home :)
09:21.44Qwellsleeping, hopefully
09:21.45L|NUXQwell : might be he will not come on from home :)
09:21.47L|NUXyupz
09:22.00Qwellwell, what's the problem, specifically?
09:22.06docelmosay qwell got a sec?
09:22.22L|NUXwell i am using svn
09:22.31Qwelldocelmo: sure
09:22.50L|NUXi have provided user credentials for my gtalk user
09:22.59L|NUXbut when i try to call its not working
09:23.36QwellL|NUX: yeah, you'll wanna ask mog tomorrow
09:24.33docelmook.. this is a dev q.. When a call is setup in asterisk its in the dialplan. Whats the easiest way to find out the channel name? For instance.. I need to know the channel name for something I am doing inside of chan_sip. It keeps kicking pbx_helper_setvar to global when I need it set to the specific channel
09:24.37docelmoAny ideas?
09:25.02QwellIt's like 2:30am
09:25.11docelmoits 5:30 here.. I know
09:25.14docelmotell me bout it.
09:25.16Qwellthere is a var though for channel name
09:25.21Qwell${CHANNEL}?
09:25.24Strom_Cdocelmo: you wave the appropriate dead chicken
09:25.37Qwellno, CHANNEL is a var...umm
09:25.45Qwellchannelvariables.txt
09:25.54Qwellor doc/README.variables on 1.2
09:26.14QwellI'm just gonna give up and go to bed
09:27.02docelmowell no.. I mean for instance.. When I setup a struct ast_channel *chan; How when I set that can I manually push the current A leg of the channel into it?
09:27.28L|NUXQwell : will ask him for sure
09:27.32SanketMedhihow do I check if my Mysql Realtime is working? When I use asterisk -vvvvvvvvvvvvdddddddddddd I see that MySql is registered properly, but the SIP phone gets an authentication failure. I have used this page http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip for setting up SIP Realtime
09:27.37Qwelldocelmo: ...tomorrow
09:27.53L|NUXQwell : i have only find two people who have worked on it one is developer which is mog and one another person cparty
09:27.53docelmo:~(
09:27.54L|NUX:)
09:28.34*** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net)
09:29.19SanketMedhianyone?
09:29.58AssidSanketMedhi: have oyu added the sip credentials to the database?
09:30.13SanketMedhicredentials?
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09:30.42SanketMedhiI have added one record for the phone I have
09:31.36Assidokay and what does verbose show with respects to mysql
09:34.08Qwellbed
09:35.27Assidnini Qwell
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11:00.32Clausianhow would i go about being able to listen in on calls being made through my asterisk box?
11:00.55*** join/#asterisk svenna_ (n=svenna@p548D0452.dip0.t-ipconnect.de)
11:03.05svenna_hi all, im living in germany and am using chan_capi as my outgoing device. when someone rings me, for him it sounds like calling england - i cant remember where to configure the tone. does someone?
11:04.13benjk_that's amazing
11:05.09jhiverindications.conf
11:06.07benjk_you mean you are speaking German and he's getting English?
11:07.41Nuggetheh
11:10.01Clausianasterisk supports realtime babelfish now?
11:12.09benjk_yeah, sounds like it
11:12.45benjk_maybe its a new feature of capi though
11:17.58*** join/#asterisk adorah (n=Administ@84.94.208.224.cable.012.net.il)
11:18.40svenna_naaaaa you know what i mean :-)
11:23.11svenna_it was indications.conf:- thx
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12:05.19e-ddienå ja, eg gidde'sje sitta å nørda her
12:05.23e-ddieheheh
12:05.26e-ddiewrong channel :D
12:12.07*** join/#asterisk potsboy (n=chrisg@dsl-145-210-106.telkomadsl.co.za)
12:13.56potsboygoodday all, have started playing with *.. what is the alarmreciever.conf used for?? precisous little docs in this regard
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12:49.23vltHello. How can I activate logging to /var/log/asterisk/event-log? This file is empty now.
12:50.04potsboyvlt: waht do you want to log there?
12:50.34CyberMadsorry for OOT: how many cent is 1 euro?
12:50.42CyberMadis that 100 cents?
12:51.27hypnoxlol, yeah
12:51.30hypnoxhence cent
12:52.20CyberMad^^ thanks..
12:54.39vltpotsboy: INcoming/outgoing calls, numbers, everything ;-)
12:55.14potsboyrather have a look at /var/log/asterisk/cdr-csv/ ..is that not what you want?
12:56.27vltpotsboy: Ehem, ok. Yes, that's what I was looking for ... ;-)
12:57.18potsboyk, may want to check /etc/asterisk/cdr_custom to change the format :)
12:58.58vltThank you.
12:59.13*** part/#asterisk vlt (n=daniel@dslb-088-073-249-127.pools.arcor-ip.net)
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13:12.58croczHello everyone,
13:14.42croczI am having a little trouble with Asterisk. Well it all works, calling and recieving calls to and from  voip-voip  / voip-phone / phone-voip. But there is a little problem. when I am calling to another user (coip to voip) that person cannot hear me, whilst I can hear them very well.
13:15.12croczAt first I thaught it was my firewall, but when I call from voip to phone I can talk without any problems.
13:15.29pnlarssonsip?
13:16.06croczAnyone has a clue what the reason for this might be? PS: If other people are calling from other places(location / other networks ..) it simply works even for voip to voip
13:16.07croczYes
13:17.09pnlarssonthe other user is connected to your *?
13:17.09*** join/#asterisk jimmy_deanPB (n=jhodapp@cpe-24-166-23-17.indy.res.rr.com)
13:17.16pnlarssonis he behind nat?
13:17.39croczYes
13:18.13pnlarssonyes on both?
13:18.33pnlarssonAre your * behind nat?
13:19.37croczNo
13:19.58*** join/#asterisk abatista (n=Ariel@70-46-87-158.ftl.fdn.com)
13:20.19pnlarssonit's a nat prob, go and check the voip-info section for nat.
13:20.22croczThe only difference between me and the other users is that they are forwarding ports 5000 to 5100 to their machine
13:20.44potsboyrtp runns from 10000 -> 20000 by default
13:21.06potsboyif he has a ast box set rtp tp 500 -> 5100
13:21.51croczpotsboy, the documentation for "ekiga" said to forward  that range (5000 to 5100)
13:22.56pnlarssonekiga?
13:23.23pnlarssonpotsboy: Thats incomming to *
13:24.00potsboyekiga = ubuntu
13:24.01pnlarssonDo you have nat=yes for this user?
13:24.40potsboyi would run a tcpdump on the asterisk side you will proly notice their is no rtp comming from ekiga
13:24.58pnlarssonAha, so the other user is running * and he is forwarding 5000-5100, has he changed the setting in * to reflect that?
13:25.02potsboybrb
13:26.43croczpnlarsson, Hmmm let me check mate
13:27.09croczpnlarsson, I do.
13:29.52potsboycrocz: change the rtp range to 5000 -> 5100 on yourside they both have to match up as per sip invites, had the same problem with a nasty cisco
13:31.42croczpotsboy, The problem here is that I cannot change  this place's router's settings. But what's really troubling me is, when I call phones, the other party CAN hear me. Whilst when I call another SIP id, they cannot
13:32.54croczFurthermore, I have tried NOT to forward that range at a friend's, and not being a DMZ. He still was able to make calls, hear people  and others could hear him. It's not really making sense to me :(
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13:33.36pnlarssonif you only have one device behind that nat, and set nat=yes it's normally just works...
13:34.27potsboyyep .. natting issue..sip sucks iax rules :)
13:34.59pnlarssonBut if the other user is running *, why not use iax instead of sip?
13:35.52potsboycrocz: the best thing is to make a call and run either tethereal or tcpdump and send it to pastebin else we are just guessing
13:38.57croczpotsboy, hold on.
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14:05.19n9urkI was poking around the openpbx site.  Is there anything significant going on in openpbx?  Has anyone used it before?  (I do know it is/was a fork of * but that is about it)
14:05.46eKo1I've remained faithfull to *.
14:06.55croczOk, now THIS is weird! I found out that if I am the caller, the other person will be able to hear me. But if I am being called, that very same person won't.
14:07.08n9urkeKo1: I don't see any reason to switch, I am just curious as to if anything is going on.  I see about 5 os projects a week where it looks like they are going to save the world and end global warming but there is nothing going on in reality - Someone had/has an idea and knows how to put a site together and then has no follow through
14:07.43croczn9urk, I am using FreePBX for administrating Asterisk
14:09.27n9urkcrocz: cool, I am looking at the site now.  Is it an additional layer to asterisk or does it include some version of asterisk as well?
14:09.48croczin my case, additional.
14:10.31n9urkcrocz: Ok I see "launched the freePBX (formerly Asterisk Management Portal) "  I know of AMP
14:10.40n9urkI didnt know the name changed
14:11.44croczpnlarsson, potsboy, did you see the last thing I said about my problem?
14:14.40*** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net)
14:19.24potsboycrocz i stand by with the previous rtp range issue.. the fact that you dont have control over the router doesnt help, do a "tethereal -v port 5060 and host <ip of remote host>" and you can compare the "working" call against the other this should sort it out
14:20.10potsboyanother thought maybe that the remote router does not support udp stream's that do not origante from the internal lan on the other side
14:20.19potsboyl8r all
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14:48.42CherebrumAnyone do load balancing with Asterisk?
14:49.52CherebrumI have two geographicly located asterisk servers and a SRV record designed to allow the UAs to failover to ASterisk #2 in the case taht #1 fails.. however.. I need a way to make Asterisk #2 register to the sip proxy and start accepting calls on the trunk side when Asterisk #1 fails
14:51.28CherebrumI've also got to keep the asterisk configs synced up so I think I'm just going to have to run rsync in a cron job and wost case they loose some voicemail
14:53.06pnlarssonYou need to sense that *1 goes down... Is a ping enough?
14:53.43Cherebrumwell... I was thinking maybe I could use my nagios server and have it execute a script as an escillation procedure
14:53.59CherebrumI was hoping there was a more elegant way
14:54.00Cherebrum:p
14:54.37pnlarssonnagios does a good job :)
14:54.37croczHmmm
14:54.41croczmay be heartbeat
14:54.50Cherebrumover a wan?
14:55.03Cherebrumone is in Florida and one in Illinois
14:55.35*** join/#asterisk af_ (n=af@ip-164-6.sn2.eutelia.it)
14:56.38CherebrumSales guy sold it and just assumed we could make it work. ;)
14:57.56Cherebrumoh! I know what I can do...
14:58.12Cherebrumif my OpenSER proxy can do SRV lookup I can just point it to the SRV record
14:58.27Cherebrumand it will work just like the phones do
14:58.44Cherebrumit will send the call to the asterisk that ACKs the call
15:00.19eKo1that seems like a good idea
15:00.31CherebrumI just wont use registrations
15:01.25Cherebrumhttp://www.voipuser.org/forum_topic_3624.html
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15:51.31*** join/#asterisk mrtass111222 (n=Tassi@83.229.72.14)
15:51.52mrtass111222hi
15:52.21*** join/#asterisk [Airwolf] (n=airwolf@dsl54026FD7.pool.t-online.hu)
15:52.32mrtass111222what's the best calling card platform for use with asterisk
15:53.14eKo1asterisk is the calling card platform
15:53.18eKo1so that question makes no sense
15:53.48mrtass111222ok which billing is the best to use with asterisk
15:53.58croczHe probably wants * to function as a gateway for  some other voip provider
15:54.14croczoh... I was wrong :)
15:54.31mrtass111222i need to build a calling cards platform so far i have installed asterisk on fedora
15:54.37mrtass111222what's next
15:55.28croczhttp://www.voip-info.org/tiki-index.php?page=Asterisk+billing
15:55.32eKo1mrtass111222: search voip-info.org for calling cards and start reading/investigating
15:55.39croczfirst google result
15:56.21tzafrir_laptopmrtass111222, I would seriously consider the support lifetime of Fedora
15:57.19mrtass111222can anyone knows how these stuff works i'll pay for any help
15:58.47folderCan anyone see why this call failed? I tried again a few seconds later and it worked. http://pastebin.ca/105335
16:00.12*** join/#asterisk kristalino (n=kristali@84-50-84-146-dsl.trt.estpak.ee)
16:00.24folderhere's the sucessful call: http://pastebin.ca/105337
16:00.32eKo1mrtass111222: contact your nearest * consultant near you.
16:10.37mrtass111222can anyone help me configure my asterisk ?
16:11.53*** join/#asterisk crparr (n=crparr@213.129.243.185)
16:13.12crparrHi! I'm really new to asterisk and have one question: is it possible to access one phone and one fax using one sipura/linksys adapter with extensions?
16:13.27*** join/#asterisk EvilDeshi (i=evildesh@oxford-bb-occam3-ws-100.dsl.maqs.net)
16:13.39eKo1Define 'access'.
16:14.14crparreg. phone: ext 10, fax ext 20
16:14.37crparrbut both connectzed to one sipura adapter
16:15.15crparrwhat other hardware is needed when using the adapter?
16:15.18crparrIs it possible to run asterisk behind a firewall?
16:15.28mrtass111222u need configure sipura channels
16:15.48mrtass111222how many lines the sipura support
16:15.52mrtass1112222 ?
16:15.55crparr2
16:16.09crparrat least it hass 2 phone connectors
16:16.24mrtass111222go to the web interface for the sipura
16:16.35mrtass111222and configure dialpeers there
16:16.50mrtass11122210 for line 1
16:16.54mrtass11122220 for line 2
16:17.02crparrthanks.
16:17.18crparrbut - I'm only gathering information at the moment.
16:17.34mrtass111222i worked with sipura
16:17.37crparrwhat hardware is needed in the pc? one lan card?
16:18.10mrtass111222u need connect the sipura to the network
16:18.18crparrok.
16:18.24crparrthats no prob then.
16:18.55crparrone last question: Is it possible to run an asterisk pbx behind a firewall?
16:19.04mrtass111222there should be connectivity between the asterisk and sipura and u need register te sipura onthe asterisk
16:19.17mrtass111222i dont know buddy abt asterisk
16:19.22mrtass111222i'm trying get help here
16:19.27crparroh sory
16:20.58tzafrir_laptopmrtass111222, do you have sip users defined for those?
16:21.14mrtass111222xlite1/xlite1
16:21.46tzafrir_laptopCan you register to them with any other sip client?
16:21.48mrtass111222but i can't see the sip port 5060 opened when i scan the ASTERISK SErver
16:21.56*** join/#asterisk potsboy (n=chrisg@dsl-145-210-106.telkomadsl.co.za)
16:22.09tzafrir_laptopmrtass111222, port 5060 UDP, not TCP
16:22.29tzafrir_laptop'netstat -lnup'  on the Asterisk server
16:22.41tzafrir_laptopshould show you listening UDP ports
16:23.30mrtass111222not showing 5060
16:23.42tzafrir_laptopIs asterisk runinng? Can you connect to it with 'asterisk -r' from the asterisk host?
16:23.53mrtass111222in sip.conf i set bind address to 0.0.0.0 as they say
16:24.03mrtass111222and restarted the service still nothing
16:24.20mrtass111222yes i can't connect to it
16:24.27mrtass111222with asterisk -r
16:24.29potsboyhey all, what is the purpose of dnsmgr.conf?
16:24.30tzafrir_laptopcan or can't?
16:24.33*** join/#asterisk oej (n=oej@65.197.203.67)
16:24.44mrtass111222i can sorry
16:25.57tzafrir_laptopnext: let's see of the sip module is loaded. from the asterisk CLI (asterisk -r) write 'sip' and press TAB twice. Does it complete some commands?
16:27.22mrtass111222IS-0665*CLI> sip
16:27.23mrtass111222debug    history  no       notify   prune    reload   show
16:27.32mrtass111222it does
16:28.02tzafrir_laptopSo chan_sip.so is loaded.
16:28.20mrtass111222maybe it's firewall issue on fedora
16:28.27mrtass111222how can i turn off the firewall
16:28.42tzafrir_laptopThere is no process that listens on UDP port 5060? (from the output of netstat)
16:28.46*** join/#asterisk SanketMedhi (n=sanket@221.135.148.49)
16:28.57mrtass111222no
16:29.03tzafrir_laptopmrtass111222, netstat's output ignores the firewall settings.
16:29.07mrtass111222i also tried nmap localhost
16:29.21mrtass111222no 5060
16:29.37tzafrir_laptopnmap from localhost laso ignores most firewall settings as you normally won't filter anythin from localhost
16:29.54tzafrir_laptoptry these:
16:29.59tzafrir_laptopset verbose 5
16:30.06tzafrir_laptopsip reload
16:30.13*** join/#asterisk s0lid (n=jlq@203.177.12.98)
16:30.34tzafrir_laptop(should that attempt to re-bind the UDP port?)
16:31.01mrtass111222i can see 5060 with netstat
16:31.03mrtass111222but not with nmap
16:31.19potsboytelnet?
16:31.19Nuggettelnet is eeeeeeevil!
16:31.31potsboymmkay :(
16:31.37eKo1telnet telnet telnet!
16:31.56tzafrir_laptopmrtass111222, this is now more of a firewall issue...
16:32.08mrtass111222yes i beleive so
16:32.41tzafrir_laptopI'm not familiar with FEdora's firewall mechanism. Any Fedora user here? Which version of Fedora is it, BTW?
16:33.03mrtass111222core 5
16:33.10eKo1Just enter 'service iptables stop' in your shell
16:34.06mrtass111222i did
16:36.27potsboymrtass explain your setup, is the system remote /local router etc
16:37.56*** part/#asterisk SanketMedhi (n=sanket@221.135.148.49)
16:37.56eKo1mrtass111222: so the firewall is off then
16:38.10mrtass111222yes i can register on the asterisk now
16:38.15mrtass111222thx
16:39.12*** join/#asterisk salviadud (n=ralfalfa@201.123.130.161)
16:39.42mrtass111222'service iptables stop' will stop the firewall but will it load again on the next reboot ?
16:40.46potsboychkconfig --level 2345 iptables off.. will stop the firewall its best to edit /etc/sysconfig/iptables and add 5060 in the input chain
16:41.29*** join/#asterisk pengyong (n=lala@218.93.158.200)
16:42.02eKo1mrtass111222: probably
16:47.23*** part/#asterisk CANO-1982 (n=alejandr@190.48.74.28)
16:47.57tzafrir_laptop"5060" there refers to a UDP port or a TCP port?
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16:50.16ManxPowerUDP, I assume you would specify the protocol
16:50.24ManxPowerdon't forget to allow the RTP ports as well.
16:51.33*** join/#asterisk asterisk-dud (n=dwwollma@64-42-247-120.mb.skyweb.ca)
16:51.54asterisk-dudhow can i get asterisk to alert the user that they have a message in thier inbox?
16:52.14*** join/#asterisk apardo (n=apardo@eu85-87-2-82.clientes.euskaltel.es)
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16:53.31potsboydud, send a message to their email
16:53.48tamp4xis there away to use array variables?
16:53.51tamp4xa way
16:55.12tamp4xhmm i guees i can use concat then parse that
16:57.27JunK-Ycaster*CLI> show function ARRAY
16:59.20ManxPowerIs there actually an ARRAY function now?
16:59.39MikeJ*CLI> show function ARRAY
16:59.40MikeJ:P
17:00.29ManxPowerI would have to SSH over a connection with 900ms - 2000ms latency to connect to an Asterisk box.  Not worth the pain.
17:01.05clyrradHey MikeJ - did you see my solution I PM'ed you yesterday?
17:02.55MikeJclyrrad, no
17:03.17clyrradremember we were pondering over the Local or LOCAL
17:03.24clyrradto see if it was case sensitive?
17:04.07MikeJI remeber the discussion.. but you said it wasn't working..
17:04.15clyrradthe problem was that in queues.conf your member = Local/ext@context has to bee all one string - you have have a "-" or "_" in the context name
17:04.44clyrradI had Local/ext@context-likethis .... and you cant do that
17:04.57MikeJoh.. you had a space... hehe.. yeah.. that won't work
17:05.02clyrradyou can use "-" or "_" in extensions.conf - but not with local in queues.conf :)
17:05.28clyrradhahah yea.... that was driving me nuts
17:05.28clyrradLOL
17:05.35MikeJdon't use spaces in context names... it shouldn't even allow that.. not sure why it does.
17:06.15clyrradno not spaces..... I was using a "-"
17:06.19shido6[Space The_finalFront ier]
17:06.27clyrradlike nameed-context
17:06.31clyrradand thats does not work.....
17:06.37MikeJyou can't use them at all ?
17:06.41clyrradNope
17:06.42MikeJthat's even more broken
17:06.46clyrradin the dial plan you can
17:06.50clyrradbut not in queues.conf
17:06.54MikeJsigh.. oh well..
17:07.03clyrradhahaha took a bit of playing with to find that out
17:07.34clyrradDo you know of a Dial flag or option that will only allow Dial to make outgoing callsed based on what patterns are set in outbound?
17:07.36MikeJso, did you fix it and submit your patch to the bug monster
17:07.49clyrradNo - I was not sure if it was a bug....
17:07.53clyrradbut it seems that way...
17:08.00MikeJseems broken to me
17:08.11clyrradyea - its not consistant thats forsure
17:08.17MikeJpatterns set in outbound?
17:08.24clyrradyea....
17:08.43MikeJas in, ... what does that say.. that makes no sense to me
17:08.45clyrradbut I want to use a Dial statment in a macro - but only want to allow dial patterns as defined in outbound
17:08.58MikeJwhat is outbound
17:09.37clyrradwell in the dial plan I have a bunch of pattern match stamtnets that allow calls out over IAX.... and not others....
17:09.48clyrradto restrict international calls for example
17:10.06MikeJI know what dial patterns are.. I still don't know what outbound is
17:10.25clyrradim using "oubound" as a word to say what I am trying to do
17:10.29clyrradits not a context or antying
17:10.35clyrradits a file i made called outbound.inc
17:10.36MikeJummm
17:10.39clyrradand it has all those dial patterns in it
17:10.50MikeJso make it a context, and send those calls to that context.
17:10.52clyrradthen I just included that file in the phones context
17:10.57MikeJand it will only work if it's in there
17:11.16clyrradyea - thats how it works now when you dial out from a phone.....
17:11.25MikeJso what's the problem?
17:11.30clyrradbut if you dial out though the macro I am creating it lets you dial whatever you want
17:11.35clyrradincluding international...
17:11.51MikeJno... it's based on what is in the context
17:12.09MikeJin a macro, you are still in the calling context
17:12.12clyrradstrange.... becase it allows international to be made....
17:12.16clyrradyea thats what I thought
17:12.17MikeJset your includes right
17:12.38clyrradyep - it works everywhere just not in this macro lol
17:12.54MikeJummmmm
17:13.18MikeJI have no idea what your saying.. but the good news is... I am going to go get a bite to eat...
17:13.22MikeJYUM!
17:13.30clyrradhahahaha
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17:27.08NoName|Rhi all can some on give me a hand i can get zap installed i am using centos 4.3 and Linux pbx.cantella.com 2.6.9-39.0.2.EL #1 Thu Jul 13 04:53:11 CDT 2006 x86_64 x86_64 x86_64 GNU/Linux
17:28.02potsboywhats the prob noname
17:28.05NoName|Ri doesnt seem to be making /dev/zap and doest seem to be loading the zap module
17:28.10leejohnNoName|R: if you can't get zaptell install look for spinlock.h bug
17:28.17leejohnzaptel*
17:28.32*** part/#asterisk rata (n=rodrigo@princed/developer/rata)
17:28.44NoName|Rleejohn, i looked at that and doesnt seen to be an issue with this ver
17:29.09potsboynoname its definately in 4.3
17:29.11NoName|Rdo u have to rebuild the kern after installing the zaptel?
17:29.20leejohnNoName|R: no
17:30.38NoName|Rpotsboy, i looked and it didnt see the type in there
17:30.55potsboyoop only saw no your running x86_64.. my bad
17:31.04leejohnNoName|R: pastebin the last output
17:31.23NoName|Rbut i am willing to try anything as i am not getting anywhere
17:31.55NoName|Rleejohn, what output u want?
17:32.15leejohnNoName|R: the line where broke your installation
17:32.31NoName|Ri am only seeing 1 line in logs with zap in them
17:33.18leejohnNoName|R: could you please be specific? can you explain more?
17:33.23NoName|Rit seems to install fine doesnt through any errors just never makes /dev/zap or loads module
17:34.04NoName|Rsorry for spelling and slowness i broker my elbow so i am forced to type 1 handed
17:34.21rob0~centosbug
17:34.23jbotcentosbug is probably a problem with the latest Centos kernels (4.2 and 4.3).  To fix it, paste everything inside the quotes into a root shell:  "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h"
17:34.26rob0maybe?
17:34.52NoName|Rboot.log:Jul 29 12:07:33 pbx modprobe: FATAL: Module zaptel not found.
17:34.52NoName|Rboot.log:Jul 29 12:07:33 pbx zaptel: Loading zaptel framework:  failed
17:35.24leejohnNoName|R: no problem :) do you think it's a udev issue?
17:35.38leejohnNoName|R:  modprobe -l | grep zaptel what's d output?
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17:38.46NoName|Rleejohn, nothing :/
17:39.21NoName|Rhmm
17:39.22leejohnNoName|R: it means that the compilation of zaptel wasn't successful
17:39.27NoName|Rhold on'
17:39.30leejohnkk
17:39.32leejohnOk
17:40.11NoName|Rseems like kernel-2.6.9-39.0.2.EL and kernel-smp-devel-2.6.9-39.0.2.EL are installed
17:40.58leejohncould you try jbot suggestion?
17:40.59NoName|Rbut [root@pbx linux]$ uname -r
17:40.59NoName|R2.6.9-39.0.2.EL and /usr/srn/kern is smp
17:41.19leejohnouch
17:41.39NoName|Rsould i unistall non smp and reinstall snp dev
17:41.41leejohnkernel-devel package from your current running kernel
17:41.54*** join/#asterisk toerkeium (i=oo@201.216.206.221)
17:42.08leejohnthen try to redo the installation of zaptel
17:42.32NoName|Rif that is the prob i will kill my self so simple but i missed it
17:42.40leejohn:)
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17:43.40folderCan anyone see why this call failed? I tried again a few seconds later and it worked. http://pastebin.ca/105335
17:43.42folderhere's the sucessful call: http://pastebin.ca/105337
17:44.12leejohnfolder: wait let me check
17:44.19folderokie
17:46.34DaveHopeHello all. At present, My dialplan entry for dialing normal users goes something like this: _2XX,1,Dial(${EXTEN}) However, that only allows me 99 real people. Say I want user extensions to be 200-300, would I do the following: _[2,3]XX,1,Dial(${EXTEN}) ?
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17:52.15leejohnfolder: what do you try to accomplish? i can't imagine the logs i'm really tired =(
17:56.34folderleejohn: calls come in from sipgate.co.uk , asterisk calls my cellphone via the sip gsm gateway on 192.168.253.200, and connects the two. It seems that the final part - the gsm bit, isn't always working. It is sometimes but not others.
17:57.29leejohnfolder: are you using voiceblue? or something
17:57.41folderI have been reading the Asterisk TFOT dialplan chapters the last few nights, and I think I am ready to ditch Trixbox and D.I.M(yself) now though, which might help debugging.
17:58.37folderleejohn: similar. Portech MV-370 from http://www.portech.com.tw/eweb/MV370/mv370.htm
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18:07.58tamp4xif i have a variable of size 20 characters, and i do Dial(SIP/${variable:0:10}&SIP/${variable:10:20}&SIP/${variable:20:30})    will the 3rd sip chan not be dialed or will asterisk crash?
18:08.43tamp4xor will it complain
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18:12.41folderDaveHope: I beleive it should actually be _[23]XX,1,Dial(${EXTEN})
18:12.56folderDaveHope (no comma)
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18:13.35ctooleyis there a way to make "SayDigits" interruptible?
18:14.45folderDaveHope: That would match 200 - 399. Not the 200 - 300 that you asked for though. I presume that's what you're after.
18:15.00DaveHopeHrm. Thanks.
18:15.46folderDaveHope: I'm only going off what it says in "Asterisk - The Future of Telephony".
18:16.01folderDaveHope: perhaps /the/ book is wrong ? [shrug]
18:16.46folderDaveHope: actually, that link says the same. It doesn't say to use a comma there does it?
18:17.10DaveHopeNope :)
18:17.17folderah :)
18:17.31DaveHopeI was doing what the comment said, rarher than the example of what it'd achieve :)
18:17.32DaveHopeThanks.
18:17.37folderdid you confuse the comments bit? or am I barking up the wrong tree?
18:17.38folderahh.
18:17.42folderlol
18:18.54folderbrb, poo.
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18:23.00mrtass111222asterisk died with code 1
18:23.14mrtass111222what's wrong ???
18:24.20mrtass111222guys can anyone help have  a proper installation of asterisk from scratch
18:24.33mrtass111222i spent all night reading the manual and now it's dying with code 1
18:24.40hypnoxlook at the log
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18:26.35the-chaos5where can i find an instalation documentation for the open source asterisk ?
18:26.49mrtass111222asterisk.org
18:27.06mrtass111222support
18:27.37the-chaos5theres only one for the Asterisk Business Edition
18:28.56hypnoxhttp://www.google.com/search?q=installing+asterisk
18:30.35mrtass111222http://www.digium.com/en/supportcenter/documentation/
18:31.22the-chaos5jes but theres only one for the Business Edition
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18:33.12mrtass111222if u have ur system on public i'll help u set it up
18:33.12mrtass111222but not more then initial setup :S
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18:35.10tzafrir_laptopmrtass111222, where did you get the message that asterisk died with code 1?
18:35.43mrtass111222after i rebooted the server
18:35.53mrtass111222i was trying to lunkch the asterisk service
18:35.59tzafrir_laptopHow do you start asterisk?
18:36.01mrtass111222then i start getting that message
18:36.14mrtass111222from /etc/init.d
18:36.17mrtass111222./asterisk start
18:36.28tzafrir_laptopDo you run asterisk as a user or as root?
18:36.43mrtass111222root
18:37.07tzafrir_laptopTry running it manually to see if it can start: asterisk -cvvvvvvvvvv
18:37.21mrtass111222k
18:37.40tzafrir_laptopIf it gets to 'Asterisk Ready' it is probaly OK. Exit it.
18:38.19mrtass111222[root@IS-0665 ~]# asterisk -cvvvvvvvvvv
18:38.20mrtass111222<PROTECTED>
18:38.20mrtass111222<PROTECTED>
18:38.20mrtass111222<PROTECTED>
18:38.20mrtass111222<PROTECTED>
18:38.27tzafrir_laptop~pb
18:38.29jbotsomebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/
18:38.50tzafrir_laptoptoo late
18:38.53sevardthe internet is boring today.
18:40.10potsboymrtass: i gues asterisk died cuase zaptel was not loaded
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18:40.27potsboymrtass: i gues asterisk died cuase zaptel was not loaded
18:40.31the-chaos5wb
18:41.29mrtass111222after i installed asterisk i installed mysql
18:42.10*** join/#asterisk vlrk (n=vlrk@202.65.134.119)
18:43.10mrtass111222now it's calling for module "res_config_mysql.so"
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18:47.48ph|berim getting this on my incoming.. Subclass: REJECT   CAUSE           : No such context/extension.   my iax connections has context=from-pstn.. my [from-pstn] extention has  exten => _XXXX,1,Goto(Sip/2000,1)
18:47.53ph|berany ideas?
18:48.12[TK]D-Fenderph|ber : Your GOTO is bad.
18:48.27ph|bergoto is bad?
18:48.31mrtass111222case sensitive ?
18:48.36[TK]D-Fenderph|ber : Sip/200 is NOT a context
18:48.51[TK]D-Fenderph|ber : OR and exten.
18:48.51ph|berahh.. hrm... k
18:49.11ph|berexten => _XXXX,1,Goto(internal,${EXTEN},1)
18:50.05[TK]D-Fenderph|ber : Looks a lot better....
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19:05.29mrtass111222can anybody provide direct help
19:05.46mrtass111222i need some love...
19:06.32*** join/#asterisk Jason99 (n=jason@jason.unitz.ca)
19:06.49Jason99does anyone know why music on hold would play in slow motion?
19:07.10Jason99I'm using the default moh mp3 that come with asterisk
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19:09.13tzafrir_laptopJason99, there's plenty of classical music availble (that is in the public domain)
19:09.45Un1xshit man when someone dails my did, on incomng they get this bullshit that you have successfully installed asterisk
19:09.49tzafrir_laptopsineapps.com has some samples
19:09.50Un1xand my phone dont ring :S
19:10.07tzafrir_laptopmrtass111222, so did your asterisk start successfully?
19:10.16Jason99tzafrir_laptop: thanks
19:11.12tzafrir_laptopUn1x, time for some basic RTFM on asterisk dialplan? get to know the magic s extension?
19:11.53Un1xlmao yea imma look for some docs on dailplan later
19:11.56Un1xto lazy atm :p
19:12.21tzafrir_laptop~docs
19:12.23jboti heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
19:12.26*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
19:12.50tzafrir_laptopvoip-info.org has a useful page on dialplans setup
19:14.19sp0n9ecan i add spaces in exten statements to enhance clarity?
19:15.03sp0n9esomething like...exten => s,    1,  Answer()
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19:18.25tzafrir_laptopsp0n9e, generally: no. Only around the '=>'
19:18.44*** join/#asterisk ttymachine (i=w0rmzw3r@ool-18b8840d.dyn.optonline.net)
19:19.11tzafrir_laptopOtherwise it will complain that it can't find an application called ' Answer'
19:20.30sp0n9ewill it trim the whitespace around 1?
19:21.08ttymachineanyone know of a password identify addon for asterisk , a caller calls in if his passcode is correct he is accepted into the menu system , if 5 wrong pass codes and his number is blocked.
19:21.42*** part/#asterisk mrtass111222 (n=Tassi@83.229.72.14)
19:24.19clyrradttymachine you can code that in the dial plan
19:25.15ttymachinei'm a complete n00b to this , I want to set my own system up for learning exp.
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19:27.55ttymachineanyone have a code like thatt they can show me for example?
19:29.53clyrradreasearch the Read application its what you need
19:31.26Un1xshit i wish
19:31.34Un1xi could get my asterisk running normaly lol
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19:43.17Un1xanyone know why i get this
19:43.18Un1xhttp://pastebin.ca/105620
19:44.25[TK]D-FenderUn1x : Maybe you could apstebin the defective file, and not just the errors it creates.
19:44.41[TK]D-FenderUn1x : mind you its TELLING your what ares are bad and is should sorta stand out....
19:44.53[TK]D-FenderUn1x : I'm betting its apparent withing 2 seconds of viewing...
19:45.32tzafrir_laptopUn1x, you need 'type=peer' or 'type=friend' or 'type=user' for each section in sip.conf (except the general one)
19:45.49Un1xok well i fixed the last 2 error's in the terminator and 100
19:45.51Un1xbut now it's this
19:45.54Un1xone sec let me pastebin
19:46.05Un1xhttp://pastebin.ca/105623
19:46.08Un1xnow it's that...
19:46.22clyrradI have read a telephone number in using READ and have the value stored in a variable caled dialnumber - how do I get that variable on an outbound channel so that its pattern matched to be sure of a valid number before being set to the Dial application?
19:46.24tzafrir_laptopBTW: #asterisk-dev is not second-level support for #asterisk
19:46.47Un1xdidn't say it was tzafrir...
19:47.14tzafrir_laptopwell, I don't see anything suspcious there.
19:47.46tzafrir_laptopthe messages from chan_zap are because a 'reload' can't change the type of channels, only their configuration
19:48.02Un1xoh okays
19:48.09tzafrir_laptopAnd the others (indications.conf and CDR) seem rather harmless
19:49.08clyrradanyone know the answer to my question?
19:50.10tzafrir_laptopclyrrad, goto() a context with those patterns in its plan?
19:50.40clyrradI will expalin the seneario....
19:51.38clyrradBasically what I have is a bunch of pattern match exten's that are used on all internal phones to be sure the dialed OUT number is valid before calling the Dial() application - now that works fine on all internal phones that are dialing out.  The problem is if I have a macro somewhere in the dial plan (lets say for doing call forward) - it will not check that the number is an "allowed" or proper number - I just dials
19:52.18clyrradI need a way to avoid that and make the system use the dial rules that the phones do...
19:52.26clyrraddoes that make more sense now?
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19:52.43[TK]D-Fenderclyrrad : Use a Goto for the read.  if it doesn't match the dialplan will continue with the next priorit at which point you can deal with it as you choose.
19:53.38clyrradhrm.......
19:53.53clyrradso GoTo(dialnumber,1) like that?
19:59.16Un1xw00t
19:59.20Un1xfinaly i got this shit working lmao
19:59.39Un1xtoo bad dont have time for the cid spoof script :/
20:00.08Un1xhmm imma tell my freind to call me lol
20:00.18Un1xand put a auto attendent on
20:00.18Un1x:P
20:00.55*** part/#asterisk vlrk (n=vlrk@202.65.134.119)
20:03.20[TK]D-Fenderclyrrad : Something like that....
20:08.00clyrradthanks TDK :)
20:10.06*** join/#asterisk Assid (i=assid@203.115.83.215)
20:13.32*** join/#asterisk MACscr (n=MACScr@adsl-75-23-106-111.dsl.peoril.sbcglobal.net)
20:14.00MACscrhey, whats the function called for rewriting the caller id name
20:15.04clyrradSetCallerID
20:15.40Un1xheh
20:16.35MACscrsry guys, just trying to explain it to someone that supposedly knows how to manage an asterisk system
20:16.53MACscrim trying to explain that i need caller id rewriting so i know whatcompany teh user called for
20:18.28[TK]D-FenderMACscr : the CALLERID function.  SetCallerID is deprecated
20:19.27Assidhey tkd
20:19.31Assidwhats cookin
20:20.20MACscrUn1x, are you getting my messages?
20:21.20Un1xyea
20:22.32clyrradhrm.... TKD - the GoTo idea was neat but does not seem to work becase it looks for a context in the name of the telephone number that was entered - as opposed to trying to pattern match it so that it does the Dial application.... any ideas?
20:24.35[TK]D-Fenderclyrrad : Yeah... you're formatting it wrong...
20:24.48[TK]D-Fenderclyrrad : Pastebin what you've done so far.
20:24.54clyrradah... ok....
20:25.06clyrradhere is the pattern that needs to be matched _NXXNXXXXXX
20:25.09clyrradwill pastebin the rest
20:25.17MACscrlol, why to voip providers say unlimited incoming, then put an asterisk, then you see that its really only 1500 a month
20:25.19MACscri hate that
20:26.15clyrradhttp://pastebin.ca/105659
20:28.47clyrradthis is what the CLI Shows -> Executing Goto("SIP/2000-09650f58", "9059998888") in new stack
20:29.06clyrradwhich is wrong becase it thinks the number I dialed is a context....
20:30.19clyrradI need a way to get 'dialnumber' out onto the outgoing channel somehow so that its pattern matched just like a regular outgoing call made from a telephone...
20:31.44[TK]D-Fenderclyrrad : S,16 = bad.  read the instructions for Goto.....
20:32.06[TK]D-Fenderclyrrad : and no, it did not this it was a context.... read again
20:32.40clyrradGoto(context,extension,priority)
20:33.22clyrradwhat would I put as teh context?
20:34.07clyrradexten => _NXXNXXXXXX,6,Dial(IAX2/${ACCOUNTCODE}/${EXTEN},,W) thats the line that should do the dial...
20:34.36clyrradam I supposed to set  _NXXNXXXXXX as the context?  Becase if I do - what if they dont enter the number in that format?
20:36.24[TK]D-Fenderclyrrad : not wuite... read it GAIN.....
20:36.47clyrradare you talking about the GoTo sub?
20:36.54clyrradthere are a few ways it can be written
20:36.58clyrradim on the wiki right now
20:37.38clyrradshows 6 ways of writing it.... all do not show a pattern match example... im not exactly sure what you want me to see?
20:38.08clyrradThis is what I am reading http://www.voip-info.org/wiki-Asterisk+cmd+goto
20:42.51clyrradTKD?
20:43.38the-chaos5have anyone installed asterisk on suse?
20:46.21*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
20:47.31[TK]D-Fenderclyrrad : Please read the brackets on this line carefully : Goto([[context|]extension|]priority)
20:47.39*** join/#asterisk tempest1 (n=asf@adsl-144-58-142.chs.bellsouth.net)
20:47.52the-chaos5noone?
20:47.58[TK]D-Fenderthe-chaos5 : Seriously avoid.  SUSE makes it hell to try and compile kernel modules for it.
20:48.24[TK]D-Fenderclyrrad : Pattern matching is inhreent.  its your call format that isn't quite right yet.
20:49.00clyrradok i am taking another look at it
20:49.33the-chaos5ok think you i have a chance?
20:49.41clyrradbrackets messing me up a bit :p
20:50.11clyrradI would think dialnumber would need to go where [context] is....
20:50.13*** join/#asterisk jcollie (n=jcollie@161.210.6.204)
20:50.35jcollieafternoon all
20:50.57[TK]D-Fenderthe-chaos5 : Chance yes, but its going to be hell I'm sure...
20:51.28[TK]D-Fenderclyrrad : You going to create a context for evey possible number?  I don't think so... keep at it.. you're close.
20:52.10clyrradno - ideally it will just GoTo whatever it pattern matches thats already been defined...
20:52.14clyrradthats the plan anyway :)
20:54.33the-chaos5whitch system is the best for asterisk
20:54.59[TK]D-Fenderthe-chaos5 : Anything "standard" with normal libraries & kernels, etc....
20:55.21clyrraddo the '|' mean anyting - or they are just trying to say "OR" as in its optional?
20:55.25[TK]D-Fenderthe-chaos5 : Debian, Slackware, RHEL/CentOS.  FC can be a little problematic, but not "drastic"
20:55.52clyrradchaos5 - CentOS works great with asterisk
20:56.00[TK]D-Fenderclyrrad : You're starting to get it... you need 1-3 parms.  and WHICH ones they are depends on how many you specify.....
20:56.47clyrradTKD all I really have to specify are 2 things - the number that was entered.... and that it needs to start at priority 1 on the pattern match context....
20:57.14the-chaos5hmmm only one problem im very new with linux"ing" and i have already problems with suse
20:57.26clyrradtry CentOS
20:57.57[TK]D-Fenderclyrrad : You're almost there.. now ask yourself WHERE are these extens my Goto is looking for....
20:58.37[TK]D-Fenderthe-chaos5 : CentOS is indeed a great start, and you only need to take 2 very well known bug into account and you're set.
21:01.14clyrradSWEEEEEEEEEEEEEEEEEET :)
21:01.20clyrradthanks TKD
21:01.26clyrradI understand what you meant now
21:02.01clyrradI was not telling the GoTo to look for a pattern match under the context that defined all the outgoing pattern matches....
21:02.43clyrradtherefore here was the proper syntax exten => s,16,Goto(my_phones,${dialnumber},1)
21:03.18Assidhey would asterisk take advantage of a duo core AMD ?
21:03.39[TK]D-Fenderclyrrad : I knew you'd get it eventually, and I did want you to figure it out for yourself...
21:03.59[TK]D-Fenderclyrrad : Worth more that way.  Jut giving you the answer would have taken 2 seconds :)
21:04.28[TK]D-FenderAssid : no clue
21:04.54DaminAssid: Yes.. it would..
21:05.04clyrradTKD - Oh yea - I will remember that for life now :)  I dont mind experimenting and learning - just needed to be pointed in the right direction - thanks bro!
21:05.38AssidDamin: i read alot of people were having issues with x86-64 with amd,
21:06.12Assidthats why most just use the generic kernel and packages
21:06.22*** join/#asterisk swytch (n=ezcall@d83-179-145-193.cust.tele2.fr)
21:06.28Assidbut if you do that, i dont see how it would take advantage of the hardware
21:07.43clyrradTKD - do you know if there is an application that you can use to input a 'dialednumber' to test if it matches a given pattern?  That would be a neat thing....
21:08.23[TK]D-Fenderclyrrad : You can create a dummy context with the matching patterns to test it.
21:08.48[TK]D-Fenderclyrrad : where the patterns that match jump back to where you resume knowing the outcome.
21:09.21clyrradOh yea - I guess that would work too - and invalid ones you could just send to Hangup
21:10.33the-chaos5so i have instaled it sucessfully but i have problems with starting
21:10.48clyrradwhat distro?
21:12.46[TK]D-Fenderclyrrad : No, invalid ones continue from the non-functional goto so you know what doesn't match
21:13.40clyrradim going to implment that dummy extension to protect the call forward feature
21:15.07DaminAssid: Don't believe what you read. I run most of my production stuff on Dual Core Opteron and Intel boxes..
21:15.35DaminAssid: And compile your own kernel if you want the best optimizations for your hardware..
21:16.23AssidDamin: yeah always do.. but was just doing a quick study on it.. thats what i came across..
21:16.48Assidbut there is a benefit from running these high end machines even if they are 64bit right ?
21:17.31rene-does anybody has a copy of nvlinedetect lying around? i am in dire need of it...
21:17.37DaminAssid: Yep. Transcoding latency decreases substantially and the number of concurrent channels that you can transcode goes up..
21:18.02rene-i asked newmantelecom for a copy but since my dns died for a couple of days i am out of luck in that matter too
21:19.04Assidhow much do you push your dualcore opteron to? like how many simulanous channels and stuff befoee you feel the load
21:22.15*** part/#asterisk rene- (n=rene-@gea-gye-internet.telconet.net)
21:22.40*** part/#asterisk jcollie (n=jcollie@161.210.6.204)
21:26.02the-chaos5YEHAAA why do you think on suse its hell it works ;)
21:26.47*** join/#asterisk bjohnson (n=bjohnson@i216-58-92-233.cybersurf.com)
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21:29.29type0hey.. i realize this isnt much of a help channel or anything other than asterisk
21:29.38type0but does someone wanna look at my zone file real quick
21:30.52type0http://pastebin.ca/105718
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21:43.09DaminAsterisk is incapable of more than about 300 SIP channels before it shits itself on any platform, so I push to about 300. ;)
21:43.43Assidnice
21:46.22*** join/#asterisk CANO-1982 (i=alejandr@190.48.74.28)
21:56.08MACscrtype0, whats your questions about your zone
21:56.20MACscrand why are you using that type of address for your mx record
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22:01.03*** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com)
22:06.26SpaceBassanyone using Gizmo with asterisk?
22:08.08NivexI use sipphone with asterisk
22:08.11Nivexwhich is fairly close
22:21.08*** join/#asterisk Lukie (n=lantins@lucie.hex.lividpenguin.com)
22:22.07LukieAnyone here got a 7961 ? Or 7941 ? I could do with having a word if poss.
22:26.03*** join/#asterisk tarvid (n=tarvid@dpc6919101029.direcpc.com)
22:28.08tarvidI need to trunk a trixbox to trixbox. I can use SIP or IAX2, but how might I use a stun server in registration?
22:28.34*** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be)
22:32.30[TK]D-Fendertarvid : * does not support stun and #asterisk does not support Trixbox.
22:32.42tarvidOk
22:32.45[TK]D-Fendertarvid : As the channel topic pretty clearly states
22:32.46tarvidthx
22:33.06*** part/#asterisk tarvid (n=tarvid@dpc6919101029.direcpc.com)
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22:43.10*** join/#asterisk StewLG (i=user@216-99-218-126.dsl.aracnet.com)
22:44.21StewLGCan someone recommend the best online guide to a from-scratch install of Asterisk? I'm setting up a new install in parallel to my Trixbox install, and want to do things the long way this time.
22:44.29*** join/#asterisk jeebusmobile (n=jeebusmo@ip68-7-6-157.sd.sd.cox.net)
22:45.31Qwell~docs
22:45.33jbotsomebody said docs was probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
22:45.40Qwell~wikis
22:45.41jboti guess wikis is http://www.voip-info.org
22:45.48QwellStewLG: all of those are pretty good
22:45.53StewLGThank you.
22:45.57QwellI would personally recommend the oreilly book
22:46.36StewLGI have the oreilley book. Is it current enough to be relevant?
22:46.43Qwellsure
22:47.05StewLGOk then. I'm just used to Linux making prior how-tos obsolete rapidly.
22:48.56StewLGOff to page 33.
22:50.24*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org)
22:51.21sp0n9eokay, i just finished the oreilly book...and now i want to build a home asterisk server to play with...thanks :(
22:52.35StewLGIs 1.2.7.1 current enough?
22:52.57StewLGHmm, maybe not.
22:53.18StewLGBut this is a scratch install. So fine for now.
22:54.04*** join/#asterisk Amilcar_ (n=amilcar@201.34.202.17)
22:54.07[TK]D-FenderStewLG : Enough to be compatable with the docs you have sure... mostly bug fixes in later releases
22:55.29*** join/#asterisk ptblank (n=MURDER1@68-169-175-248.lmdaca.adelphia.net)
22:59.55sevard[TK]D-Fender: post this on voip-info.org http://popularitydialer.com
23:02.34*** join/#asterisk mog (i=ejabberd@68.62.237.103)
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23:16.06*** join/#asterisk mrh2 (n=chatzill@host-84-9-253-120.bulldogdsl.com)
23:16.46mrh2hi is it true agentcallbacklogin is being thrown out?
23:17.31Amilcar_mrh2: yeah. :-(
23:17.34Qwellmrh2: eventually
23:17.44*** join/#asterisk Iam8up (n=iam8up@156.63.171.206)
23:17.46Qwellbut it *WILL* be there in 1.2, just deprecated
23:18.00Amilcar_Qwell: 1.4, you meant.
23:18.05Qwellwhatever :p
23:18.08mrh2:( worried about what he can do for roaming extensions
23:18.09Iam8upis there any way to have the verbose text displayed from asterisk -r saved to some sort of a log?
23:18.09Amilcar_:-)
23:18.23Amilcar_mrh2: yeah, me too.
23:18.37mrh2it is pretty business critical
23:18.54sp0n9ei don't want to sound like an idiot, but sip phones connect to a normal ip network, right?
23:18.59QwellWhen it was announced that it would be deprecated, it was made clear that it is being removed because there are other better ways to do so
23:19.29mrh2are there any examples floating around?
23:19.49QwellIt was also made clear that examples would be included with 1.4
23:20.24mrh2for roaming though not the dynamic queue stuff?
23:20.41Qwellvery simple dialplan logic
23:21.11Amilcar_Qwell: like we talk another day, i'm pretty sure that the roaming capabilities of agentcallbacklogin could not be replaced by dynamic members.
23:21.56symlinkAmilcar_: setvar, astdb
23:22.07Amilcar_at least, without the use of hacks like storing things in astdb...
23:22.10Qwellsymlink: he was already told all of that
23:22.14Qwellastdb is HARDLY a hack
23:22.22symlinkastdb is there for this sort of stuff
23:22.38Amilcar_Qwell: no, please, i don't think astdb is a hack.
23:22.52mrh2how about notation of agent as the originating channel in the cdr?
23:23.22mrh2think that is not an option for changing with the dialplan
23:23.55Amilcar_mrh2: maybe using custom fields with setvar....
23:24.13mrh2you can't change all the fields
23:24.18mrh2just some
23:24.40Amilcar_Maybe we have to use this "some".... :-)
23:27.25mrh2ok so i guess it is wait and see - really hope there is an actual  workable solution for this going forward  otherwise i know at least one call centre that will be starting to look for an alternative
23:27.51*** join/#asterisk icyfire0573 (n=icyfire@u1016342.ul.warwick.net)
23:29.03icyfire0573When I call music on hold to test it, it says starting music on hold. It then immediatly says stopping music on hold.
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23:49.22*** join/#asterisk pbuckley (n=pbuckley@mail.amcat.co.uk)
23:49.48pbuckleyHey there, Ive got a problem with the license for the business edition of asterisk, and wondered if anyone can help me?
23:50.23mogwhats wrong pbuckley
23:51.20pbuckleyIve changed network card in the server, and upon restarting (and configuring the card) BE is no longer registered. I went to run registerbe but its asking for a code which I dont actually have. Someone else did this part and he is not available.
23:54.49pbuckleyIm in the UK btw
23:58.47*** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk)
23:59.05pbuckleyIm either wondering if I can get a code, or find out what it is. Or if I put the hardware back the way it was if it will come back up.
23:59.17mrh2not that i'm using the bus edition but might be based on mac address - i'd put the old card back until u can get hold of the person
23:59.54pbuckleyIll go and give it a go, Ive been sat here for 3 hours so far trying to sort it out.

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