IRC log for #asterisk on 20111108

00:01.14citywokHello
00:01.23r0m|uhola
00:08.48hardwirehelloha
00:10.07*** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek)
00:11.30Naikrovekholy moly it is hard to convince mgmt that used cubes aren't a ripoff.  "$800 per desk?!  Are they made of gold?!"  me: "New they're $5k per seat."
00:11.52hardwireyeh
00:12.38hardwireNaikrovek: this is where you request a brand new water cutter machine and a home depot card.
00:12.43hardwireand a sewing machine
00:12.45hardwireand profit
00:13.17NaikrovekI set down $41k of material requests on my CFOs desk today.  cubes were $24k of that.
00:13.35Naikrovekhe didn't even blink at anything else. the cubes made his head pop off his shoulders.
00:13.49Naikroveki should have asked him what he thought they cost.
00:14.05Naikrovekeven though I didn't, I know what his answer would be: "why can't we go to walmart?"
00:15.36p3nguinWould they prefer $99 desks and $49 chairs from Walmart instead of cubes?
00:19.03Naikroveki'm beginning to thinkso
00:19.13Naikrovekeverything is too expensive
00:19.30Naikrovekit's not that we don't have the money, it's that the CEO gets involved in every little decision and says "that's too much."
00:19.53F2Knightwhat are these cubes?
00:20.00p3nguinI can't understand how spending 4 times as much is a money saving plan.
00:20.05p3nguinBut that's what they do.
00:20.45NaikrovekSteelCase Series 9000 is what I want to get.  They are inexpensive used, and common.
00:20.54[TK]D-FenderF2Knight: They're like rectangles... only equally long on all sides ;)
00:21.24F2Knight[TK]D-Fender, so not like a TARDIS then
00:21.33Naikrovekreplacement parts (if ever needed) will be easy to obtain, and if we grow our office again, more of that make/model will be easy to obtain.
00:21.51hardwirejust get some wood and cement board
00:21.53hardwireand nails!
00:21.57hardwirethen!
00:22.01hardwireget some appliance epoxy
00:22.04hardwireand spray it on
00:22.06p3nguinScrew planning for the future, you're just spending money on something you don't even yet need!
00:22.09hardwireand make indestructible cubes
00:22.10p3nguin:/
00:22.20hardwirebetter yet
00:22.21hardwiretruck bed liner.
00:22.26Naikroveklol
00:22.27hardwirecheap.. effective.
00:22.40Naikrovekwell i have requirements driving this, i'm not spending for spending's sake
00:22.55Naikrovekit doesn't matter to me so much, but the stakeholders will shit themselves if this doesn't happen
00:23.16hardwireI can't wait until you can just put a bunch of used banana peels (don't ask) in the middle of a room and have nanobots turn it into a cube farm.
00:23.26Naikrovekhardwire: also, it must meet fire code.
00:23.37hardwireI'm pretty sure indestructble meets fire code.
00:23.44Igneousp3nguin: sorry, mind if I bug you with another dumb question?
00:23.56*** join/#asterisk tmrhmdv (4575afcd@gateway/web/freenode/ip.69.117.175.205)
00:24.13p3nguinigneous: If you already asked it and I didn't answer, don't expect me to answer if you ask me directly.
00:24.21Naikroveklol
00:24.22IgneousI know you're regarded as the dialplan guru in here.
00:24.23Naikrovekjust ask
00:24.24Naikrovekmaybe i know, hell
00:24.49Naikrovekp3nguin? dialplan guru? really?
00:24.52F2Knightor maybe I do too.
00:24.59*** part/#asterisk pietro (~pietro@88-149-224-154.dynamic.ngi.it)
00:25.08p3nguin~p3nguin
00:25.08infobotp3nguin strives to maintain his elitist reputation by refusing to use GUIs where they aren't beneficial.  See: FreePBX.
00:25.13F2Knighthands p3nguin a fish
00:25.19p3nguinhmm
00:25.22IgneousI'm just trying to figure out why ${QEHOLDTIME} isn't set unless it's called by membermacro (or after the channel is established).
00:25.33p3nguinInteresting... but no mention of dialplan guru.
00:26.14F2Knight~F2Knight
00:26.35F2Knightyep insignificant
00:26.50tmrhmdvHi folks! I am trying to achieve the best install. I've installed Asterisk over 16x just today and it... sucks. I'm a linux noob, but I know that it's possible to automate the install.process. So, can someone point me in the right direction, please?
00:26.55hardwireheheh
00:27.14p3nguintmrhmdv: What distro do you use?
00:27.24tmrhmdvp3nguin: Amazon Linux
00:27.38p3nguinIs that derived from something more... mainline?
00:27.54Naikrovekhe's on EC2 sounds like
00:27.59Naikrovek...maybe
00:27.59tmrhmdvYes, it's like CentOS/RedHat/Fedora etc. uses yum
00:28.04tmrhmdvYes Iam
00:28.08Naikrovekknew it.
00:28.17F2KnightSource install... :  cd asterisk ; ./configure ; make ; make install ; make samples
00:28.22p3nguinOkay, so install the digium/asterisk repo, and install asterisk and friends with yum.
00:28.29Naikrovekasterisk will work, but you should hone your craft on a virtual machine before you pay to learn it on EC2
00:28.34citywokdo what F2Knight said. wget it, tar it, cd in to it, ./config; make; make install
00:28.48citywokNaikrovek: you can learn for free on ec2 w/ a tiny instance
00:29.02Naikrovekcitywok: ah yeah forgot about that.
00:29.17citywokalthough i've never tried * in EC2 i only use it for web hosting
00:29.20Naikrovekstill - you have to select the proper kernel or you'll have timing issues
00:29.31Naikrovektiming issues = audio quality issues
00:29.36F2KnightI actually keep an SVN repo upto date locally, and run a bash script every night to rebuild and then create a deb package that I put on machines.
00:29.42Naikrovekaudio quality issues = preception of crap phone system.
00:29.46Naikrovekso, try on local hardware
00:29.52citywokF2Knight: i hope that is a dev environment you are testing in
00:30.03F2Knightcitywok,  yeppers.
00:30.29tmrhmdvLet me correct my question. I'm not having any issue with installing. It's just that I'm installing it over and over. Therefore just wanted to know if I could automate it with some (bash?) script or whatever it's called. + I'll have to install it on 4 server
00:30.53Naikrovektmrhmdv: with packages you can do it really quickly and easily
00:30.54p3nguinWhy are you installing it over and over and over?
00:30.54F2KnightSource install... :  cd asterisk ; ./configure ; make ; make install ; make samples
00:30.59Naikrovekp3nguin: he's experimenting
00:31.01*** join/#asterisk troyt (~troyt@PariahZero.broker.freenet6.net)
00:31.07Naikroveklocal virtual machines would solve this. (snapshots)
00:31.08p3nguinI think THAT is more of the problem than the fact that it takes a while to do.
00:31.19tmrhmdvYes
00:31.25tmrhmdvExactly, local VMs don't work
00:31.30p3nguinUnless you are changing the acutal code, there is no reason to keep installing again and again.
00:31.31tmrhmdvyou've to install the OS and blah blah
00:31.49citywokwhat p3nguin, you shouldn't need to reinstall unless you are doing dev in the source
00:32.02citywoks/p3nguin/what p3nguin/
00:32.03p3nguinChanging the configuration does not require reinstallation of the software.
00:32.04Naikroveklocal VM will work.  You can use them for testing.  You're a linux newb and you want to deploy asterisk?  I assume for production?  Practice locally
00:32.10F2Knighttmrhmdv, your choices are easy.. yum install asteirsk
00:32.15Naikrovekyep
00:32.17F2Knightor if from source.
00:32.19F2KnightSource install... :  cd asterisk ; ./configure ; make ; make install ; make samples
00:32.31Naikrovekyum install asterisk18 asterisk18-addons asterisk18-config
00:32.31tmrhmdvI'm comiling it myself
00:32.32Naikrovekthen wait
00:32.32F2Knightand you run that after you install the OS
00:32.33Naikrovekthen done
00:32.35tmrhmdvcompil*
00:32.41tmrhmdvcompiling* dammit
00:32.57p3nguinI still don't see an actual problem.
00:33.02tmrhmdvNo, no guys :D
00:33.02F2Knightoh a better way would be this...
00:33.36Naikrovektmrhmdv: okay take your time and explain because we're missing something
00:33.40F2Knightcd asterisk ; contrib/scripts/install_req install ; ./configure ; make ; make install ; make samples
00:33.42autofsckkp3nguin: hello, can you help me with my incoming calls from the trunk? it is already registering, i can make calls, but when receiving my ITSP says that it looks like busy signal
00:33.42p3nguinIf you explain an actual problem, maybe you can get an actual answer.
00:33.50tmrhmdvThere's no problem with installing, this is more of a Linux/CLI question. You know you can install many software with *.sh script that include all of your 'yum install yada yada'
00:33.54F2Knightthat would get all the 'extra' files and dependency you need
00:34.07Naikrovektmrhmdv: yes
00:34.14tmrhmdvI know all the packages/dependencies I need
00:34.17Naikrovekokay
00:34.29F2Knighttmrhmdv, do you know what a .sh file is?
00:34.32Naikrovekhe knows
00:34.47p3nguinautofsckk: Show me some configuration and I'll tell you what I think is wrong with it.
00:34.56Naikroveki dont' think english is his first language; give him a break
00:34.58Naikroveklet him say it
00:35.06F2KnightNaikrovek, I don't know that he does.. if he did he would now how it works.
00:35.24p3nguinautofsckk: I see the pastebin.
00:35.35tmrhmdv*facepalm* english isn't my 1st lang. but i'm fluent, i just can't explain what i'm trying to do xD
00:35.41p3nguinautofsckk: I need to also see your register statement.
00:36.14tmrhmdvF2Knight: I don't. I am a HS student, I was just playing around with asterisk and install it on 4 servers.
00:36.34tmrhmdvBut I don't want to install it from yum packages
00:36.39tmrhmdvI want to install it from source
00:36.54citywoktmrhmdv: then ./configure; make; make install in a shell / bash script
00:36.57p3nguinautofsckk: I want to see your entire sip.conf, too.  Hide only your passwords.  If there is anything else changed or hidden, I'm not going to waste my time.
00:37.41F2Knighttmrhmdv, okay you want me to send you my script?
00:38.01tmrhmdvF2Knight: can you please, I just want to see how it's done
00:38.08p3nguinautofsckk: Change your register statement.
00:38.21p3nguinautofsckk: Append /phonenumber
00:38.32p3nguinautofsckk: user:pass@host/phonenumber
00:40.01p3nguinautofsckk: Your ITSP is probably not behind nat, so change nat=yes to nat=no
00:40.23autofsckkok done
00:40.27p3nguinautofsckk: insecure=very shouldn't even work.  If you need to use it, it would be insecure=port,invite
00:40.39*** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net)
00:41.14p3nguinautofsckk: Then define an extension in your incoming_calls context for your phone number.  Make it do something useful.
00:42.01Naikrovektmrhmdv: ah so you want to install from source, is all?  that's not too hard.
00:42.08tmrhmdv"If you need to install Asterisk onto several machines, you may wish to build a set of scripts to help automate this process." That "build a set of scripts" is what I am interested in
00:42.11tmrhmdv:D
00:42.11Naikroveki had a good link for this, let me see if i can find it.
00:43.00F2KnightNaikrovek, I just sent him a install script
00:43.22Naikrovekah
00:43.23Naikrovektyvm
00:43.57tmrhmdvthank you! ooh, finally :D
00:44.05tmrhmdvwas able to explain
00:44.21F2Knighthttp://pastebin.com/x9yuBAHR
00:44.41F2Knightas you will notice tmrhmdv, it is the exact same thing you would type at the command line
00:44.45F2Knightnothing special at all
00:45.02p3nguinautofsckk: All I see in paste 18 is a failed attempt at an extension pattern.
00:45.14autofsckkp3nguin: i tried what you told me, but it still doesnt work, i cant see any incoming call al CLI :S
00:45.27p3nguinautofsckk: sip set debug on
00:45.47p3nguinautofsckk: Make a call to your system from your mobile or something.
00:45.52tmrhmdvOh, ok. I just found typing same lines of command on 4 machines would be time consuming, anyways, thank you F2Knight, Naikrovek, citywok and p3nguin
00:46.02p3nguinautofsckk: Pastebin the entire thing.  It may be very long.
00:46.02F2KnightIf you wanted to get a litlte more fancy you could get the latest name from the website and automaticly extract it and such using variables, but that will require a litle more learning on your part.
00:46.40autofsckkp3nguin: the only way i can call myself right now is with the same voip phone :S
00:47.01F2Knight4 machines one single line of commands, not a waste... 400 machines ... well either way you have to get the script on to the box. so thats why most people dont bother with it.
00:47.21autofsckkfrom the sip debug i get may info about destroying SIP dialog and diferent ips
00:48.01p3nguinautofsckk: Without seeing what's going on, I can't help further.  You're limiting me by not giving me entire configs from the start, and you're limiting me further by not being able to make test calls to troubleshoot.
00:48.42*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
00:48.54p3nguinI've seen the extensions.conf, but I haven't seen the sip.conf.
00:50.51tmrhmdvF2Knight: I never knew what a Linux was  :P I got started only a few days ago...excuse my noob-ness
00:51.22Naikrovekwe were all newbs once
00:51.29Naikrovekmany still are
00:52.08tmrhmdv:)
00:53.02p3nguinautofsckk: There is no autofallthrough setting in sip.conf.  Remove that line.
00:53.48p3nguinautofsckk: You've defined a localnet, which tells me your asterisk is behind NAT, but you have not configured nat=yes nor any externaddr/externhost values.
00:53.49Naikroveklinux is pretty awesome as a learning tool
00:53.59Naikroveki hate it as a desktop, but as a server it's tolerable
00:54.13tmrhmdvI agree, I 'm already falling in love with it
00:54.41zyphlar"everything's a file" is a wonderful concept. i'm still struggling with automating configuration of my windows servers and i've been doing windows for years
00:55.00tmrhmdvOoh, btw, if any onf these: Leif Madsen  Jim Van Meggelen  Russell Bryant gentlemen are here, I wanna thank them for their book, Asterisk™: The Definitive Guide, it taught me a lot!
00:55.08autofsckkp3nguin: well i think that for now i can get rid of that line and just make asterisk work internally right? after that i can make it receive connections from the outside?
00:55.17Naikrovekzyphlar: that's no fault of yours; windows doesn't really facilitate this well
00:55.19p3nguinautofsckk: I would also name my peer for my ITSP something a bit more descriptive, such as the name of the provider.
00:55.45p3nguinautofsckk: If your asterisk is behind nat, configure it correctly for working behind nat or don't bother configuring it at all.
00:55.54p3nguinYour problem was that calls are not coming in from outside.
00:56.00p3nguinThat involves nat.
00:56.42p3nguinMake sure you forward the necessary ports at the firewall: UDP 5060, and whatever UDP port range is defined in rtp.conf (usually 10000-20000).
00:57.31Naikrovekon recent Windows OSs, powershell goes a LONG way to reaching that goal, though, zyphlar
00:57.38Naikrovekespecially on the server
01:02.42*** join/#asterisk ketas (~ketas@ketas6-sixxs.si.pri.ee)
01:03.16*** join/#asterisk Hanumaan (~Hanumaan@132.230.17.77)
01:03.53autofsckkp3nguin: i think its working now :D
01:04.20autofsckkthanks a lot
01:04.30p3nguinI guess that's a good thing.  Confirm for a fact that it works.
01:06.54autofsckkp3nguin: well the thing is that i dont have credit on my cell, so the only way i can test it is by auto calling me with the same voip account, so i did it and i now receive the call on my pap and twinkle here on this computer, calling from my netbook running asterisk and twinkle too, so yes, it works now, thanks a lot :D
01:07.12p3nguinGreat.
01:09.54*** part/#asterisk osas (~osas@nslu2-linux/osas)
01:10.45*** join/#asterisk LiuYan (~liu.yan@211.154.128.135)
01:11.55autofsckkp3nguin: is something very strange, remember 2 days ago that i asked for help too? i was trying to configure an asterisk running from a vm on centos, that couldnt register, nor the asterisk running here on this computer, using the same info that im running on the netbook, what could it be wrong on those boxes that couldnt register with mi ITSP ? any ideas?
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01:13.04p3nguinautofsckk: Without seeing the evidence, I can't even begin to guess.
01:13.23p3nguinIf you would have given me the sip debug, maybe I would have been able to say what's wrong.
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01:15.01radenNaikrovek, Yo !!! you around bro ?
01:15.08Naikrovekyo bro
01:15.10Naikrovekgi joe
01:15.11autofsckki will try now with this files that are working too see if it still doesnt work, so for sure ill will be asking for help maybe later or tomorrow
01:15.17tmrhmdvautofsckk: I experienced your problem few times and it was because of NAT for me, but it may be something else that's causing your problem
01:15.20Naikrovekknowing is half the battle-o
01:15.34Naikrovekmy uncle was the voice of lion-o (not joking there)
01:15.42hardwirenuhuh
01:15.45autofsckktmrhmdv: i disabled iptables on both boxes, and it still wasnt working
01:15.49radenNaikrovek, for wisp I want to offer FTP backup service or something , 10 GB limit per user , how would I set this up in ftp ?
01:15.51Naikrovekyeah-huh
01:15.56p3nguinNAT usually doesn't prevent registration to your ITSP.
01:15.58hardwirenuhuh
01:16.21Naikrovekraden: ftp server software may offer this. i don't know how quotas work in linux but that woudl be the other way
01:16.26Naikrovekhardwire: yeah-huh
01:16.28autofsckki know, you can register, but you have audio problems, or not receiving calls right?
01:16.32hardwirecool
01:16.41radencould I make each user directory a partition ?
01:16.48raden( seems a lil extreme lol )
01:17.18WIMPyraden: Quota
01:17.18p3nguinautofsckk: Usually that's what happens with misconfigured NAT settings.
01:17.19Naikrovekhardwire: well, it's my wife's uncle.  Her cousin is trudy weigel on reno 911
01:17.29Naikrovekraden: linux kernel supports user quotas
01:17.41tmrhmdvp3nguin: yeah, as  I said there were many factors, like enabling 5060-5061, 10000-20000 and etc ports
01:18.06tmrhmdvautofsckk: When I tried installing and setting up Asterisk on local VMs, all failed and I couldn't get anything to work. Now, I'm just using EC2 and everything works with no problem + it doesn't cost too much
01:18.09p3nguinI have no idea why you would "enable" 5061.
01:18.23hardwireNaikrovek: claimin to faimin.. I like it
01:18.23p3nguinsince SIP is 5060, and all.
01:18.25tmrhmdvMe neither, but I had to on EC2
01:18.35hardwiremy uncle was on taxi
01:18.39p3nguinI doubt you had to.
01:18.44p3nguinsince SIP is 5060, and all.
01:19.00Naikrovekhardwire: who did he play
01:19.02tmrhmdvI read a 'guide' that said I 'had' to :)
01:19.04F2Knightraden, disk quota's... however... you have to run a script to determine the disk usage for each user. It is cpu intensive action and you must run it when ever you want to check. The end result is that you could have a user use more then 10GB of data up until the point that you check
01:19.15p3nguintmrhmdv: Post a comment that informs them of their mistake.
01:19.16hardwirejohn Burns (Randall Carver)
01:19.22Naikrovekhardwire: nice
01:19.34F2Knight5061 is used for sip over TCP i believe
01:20.03Naikrovekhardwire: he was in There Will Be Blood
01:20.06NaikrovekI love that scene
01:20.07p3nguinI guess forwarding UDP 5061 wouldn't do much good for that, then.
01:20.14F2Knightnope
01:20.30Naikrovek"Now, I'm not going to waste your time, Mr. Bankside; I'd appreciate it if you didn't waste mine."
01:20.34NaikrovekLOVE that movie
01:20.39hardwirethat's all I got for fame
01:20.48hardwireoh.. and I ended up in the kernel source more than once..
01:20.51hardwireso that's cool.
01:21.03hardwirebut that's not mainstream :)
01:21.36Naikrovekif you got in there early enough you would have been part of valinux ipo.  that would have netted you a lot of dougth
01:21.38Naikrovekdough&
01:21.40Naikrovek*
01:21.46hardwireyeh.. no.
01:21.50hardwireI'm a late bloomer
01:22.13Naikrovekthey used the linux kernel source to gather part of their list of pre-ipo offers.  many millionaires were made that day
01:22.41Naikrovekwatched it open at $100 and close at $400 (if memory serves)
01:23.27Naikrovekeh wikipedia says i'm full of shit
01:23.30Naikrovekdoesn't disagree.
01:23.37hardwirehaha
01:23.53tmrhmdv:D
01:24.13tmrhmdvsometimes, memory fails
01:24.29Naikroveksometimes becomes usually eventually
01:24.43tmrhmdvtrue
01:25.54Naikrovekhmm.  how do i convince my mgmt that these silly cubes are worth the money
01:25.55hardwiremy buddy Gareth worked for va.. thats aall I remember.
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01:26.14hardwireNaikrovek: piss on the other offers.. while they sit there watching.
01:26.22Naikrovekheh
01:26.25hardwireto make it more dramatic.. drink a lot of 151 something
01:26.27hardwirethen light it.
01:26.56Naikrovekwell they were the least expensive by a little bit, and i've compared the price of these cubes to new.  dramatic difference there
01:27.02Naikroveknot sure how else i can lay it out honestly
01:27.07*** join/#asterisk coppice (~chatzilla@m121-202-79-203.smartone-vodafone.com)
01:27.38Naikrovekmiddle management needs 16 more cubes in the space provided.  we have 12, but to fit 28 i have to replace everything.  this is what that costs
01:27.49Naikrovekup to the middle managers who need the seats to do the selling at this point
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01:39.25*** join/#asterisk bluregard (~mattbrei@c-98-228-3-34.hsd1.il.comcast.net)
01:39.38bluregardgood evening all
01:40.36bulletrtrHello
01:40.42bluregarddoes anyone know how the AMI Uniqueid: is calculated?
01:40.59bluregardI'm wondering if it's suitable as a DB primary key
01:50.18F2Knightbluregard, the UniqueID if I recall is not really all that unique. What is the application that you are requiring a unique ID for?
01:51.48F2Knightbut it is composed of epoch time of when a call starts, plus a monotonically incrementing interger.
01:52.56F2Knightthey will only be unique for calls on that box... this is using the ${UNIQUEID}
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01:58.07bluregardf2knight: I'd be using it as a primary key for a mysql table containing a list of calls
01:58.55bluregardits not that big of a deal if it's not unique enough, I can always just use an auto_increment if I have to.  I'd rather be safe than risk a duplicate key
02:07.11F2Knightbluregard, I would just create your own key. You are assured that will work. If you want to store the unique ID from asterisk for later searches, you can set an index on it to make looks eaiser. but I would not use it personally.
02:12.51bluregardf2knight: yeah, I'll probably just end up using it to relate events with individual calls rather than as a DB key
02:16.20bluregarddid google voice break something again?  I can get calls to go out from asterisk to my cell phone, but when I answer my cell it doesn't acknowledge I answered it, it just times out.
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03:24.43andyoutsideremind me how do you have asterisk read text please. More the url for that info
03:24.52dijibfestival
03:25.10dijibusing text2speech in a system command or the festival command
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03:45.32bulletrtrCan anyone recommend a good quality non HP/Dell/Gateway rackmount server that can support RAID, redundant power supplies and have capability to hold a Sagnoma or Digium card?
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04:29.23andyoutsidebulletrtr: why not any of those price?
04:38.57SwKbulletrtr: what not just get a dell off the secondary market theres tons of them cheap out there
04:39.33SwKlike a dell 1950, complete with raid, remote lights out management redundant PSUs and hold various PCI/PCIe cards
04:47.34bulletrtrPrice is an issue, but I am more concerned about redundancy and long term service to customers.
04:48.03bulletrtrI have a 2950 now which I like, but they don't seem to make them anymore.
04:48.24bulletrtrHas anyone heard of supermicro?
04:48.39SwKbulletrtr: theres a ton of that stuff on the secondary market
04:48.43SwKthats the great part about dells
04:48.45bulletrtrDell's service has been weak.
04:49.12bulletrtrI looked on Ebay for new stuff.  The selection was a bit weak.
04:50.26andyoutsideI use 2950 from ebay
04:51.00andyoutsidenormally we will call up a company that has a lot of them and tell them what we want in it and they give us a price
04:52.05SwKtheres better places then ebay heh
04:52.22SwKbulletrtr: are you in the states?
04:52.34bulletrtrYes.
04:52.42SwKcheck out stikc.com
04:53.23SwKlet me refer you to my sales guy there if you see something you like (disclaimer; the do a referal credit which I will use heh)
04:53.47SwKever server I have purchased in the last 4 years i have purchased from therem
04:53.50SwKerrr them
04:54.13bulletrtrThey look well put together and have been high reliability?
04:54.48andyoutsidenice
04:55.39bulletrtrIt looks like they have some reconditioned stuff too.  Thay might be an option.
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05:00.24andyoutsideand is this for asterisk?
05:01.00andyoutsidewho is your tel provider
05:01.55bulletrtrNo, it is for a email server/DNS/website, but I want it as a backup in the future for my present Asterisk server.  Centurytel.
05:03.24bulletrtrI am looking at starting a WISP and selling VOIP through my Asterisk server.
05:08.47andyoutsidein that case look at maybe two servers if you really want it to stay up
05:08.59irrootbulletrtr not always the easiest running voip over wifi ... but can be done
05:09.26andyoutsideI do it all the time
05:09.45andyoutsideplus across country to the server
05:10.02irrootbulletrtr what wifi kit you want to use we have some experiance doing it arround town
05:10.12SwKbulletrtr: i dont nothing but ITSP and CallCenter stuff
05:10.53irrootandyoutside bulletrtr it works no doubt about it Johannesburg and Durban have plenty sites on wifi
05:11.31SwKbulletrtr: and these guys sell refurb stuff... very little new stuff... the con is its not brand new so you wont get the lastest thing dell just launched yesterday... but the stuff is refurb'd/reconditioned before they ship it out
05:11.37andyoutsideThe biggest thing that you have control of is your isp
05:11.54andyoutsidewe use level3
05:12.04SwKandyoutside: how was your internet this morning? heh
05:12.09bulletrtrWe are planning on using the new UBNT AirMax and installing many sites close to the populations to reduce interference and offer better throughput (low latency back to Asterisk)
05:12.37andyoutsideinternet was working but another network messed up their routing tables
05:12.51andyoutsideso only part of the internet was avalable
05:13.16SwKlatest I heard they botched a juniper MX upgrade again
05:13.31SwKwhich isnt hte first time i have heard of such things from L3
05:13.59andyoutsideheck I will go see what the log says
05:16.25dijibhttp://ontario.kijiji.ca/c-buy-and-sell-computers-Dell-PowerEdge-2950-II-Server-2x-DC-3-0GHz-8GB-4x-73GB-15k-W0QQAdIdZ327676121
05:17.06irrootandyoutside mmm how another network can mess up there tables mmmm sounds like they did not set up BGP filtering properly
05:17.28andyoutsideThe IP NOC has advised that multiple links network wide are bouncing, which is affecting IP traffic.
05:18.15SwK<PROTECTED>
05:18.28bulletrtrdijib:  Thanks!
05:18.32SwKirroot: Level3 admitted to doing a router code update this morning
05:18.39dijibnp
05:19.07irrootah ok ouch that has to hurt thx Swk
05:20.11SwKirroot: yeah... the problem is once that started happening it started cascading... then you had route fapping and triggering flap hold downs heh
05:20.16SwKissues all over the place
05:20.19andyoutsidewhat ISP do you like better than level3?
05:20.31SwKandyoutside: use more then 1
05:20.41SwKwe use Cogent, L3, and XO right now
05:20.42andyoutsidenods
05:21.03andyoutsideif we were to use a second one it would be cox
05:21.33SwKdepends on where you colo or if you are trying to backhaul the connection to your site
05:21.51SwKif you are colo'd the choices are better and usually cheaper due to lack of backhaul costs
05:22.06andyoutsidethey are alwired and ready. We lease the land of one of their biggest if not biggest users
05:22.24SwKnice
05:22.37SwKyeah I colo in NYC and MIA
05:22.41irrootSwK we have only 2 International carriers and at most 4 providers so not much choice
05:23.10andyoutsidewhere are you swk
05:23.34SwKandyoutside: I'm in MS ... used to live in Huntsville (down the street from digium almost quite litterally)
05:24.22andyoutsideSome of the bigger ones you just have to call and ask how much for them to do it.
05:24.31SwKyeah
05:27.40andyoutsideso who here uses snort while we are off topic
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05:37.39SeRiquite night...
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06:30.52ChannelZfarts loudly
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06:39.14irrootChannelZ for a minute i thought it was the dog
06:44.58SeRilol
06:45.35SeRiis reading The Book... Silence I KILL YOU!
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06:59.22ChannelZhttp://failblog.org/2011/11/01/epic-fail-photos-family-fun-fail-3/
07:00.36SeRilol
07:09.40autofsckknight, anybody have used spa3102? a friend of mine told me that he have a lot of delay, jitter
07:10.09autofsckki have read somethings about that spa3102 and it says it is not so a good idea to use it as an F
07:10.18autofsckkFXO  sorries
07:12.02ChannelZI have one
07:12.59ChannelZI always had a problem with echo
07:13.02autofsckkChannelZ: is it good? i have read bad things about it, is it really that bad=
07:13.36ChannelZIt has a LOT of settings
07:13.48autofsckki've read that ther are some things you can try to fix it, but it cant be solved completly
07:14.00ChannelZI'm using it as an FXS now, works great for that
07:14.40autofsckki think its like a pap2 as a FXS, i have a pap2 and it works well
07:15.07autofsckkChannelZ: what do you use now as FXO?
07:16.02ChannelZwell nothing.. I got rid of my home phone line and do it VOIP
07:16.46ChannelZI have a TDM800 at work for FXO but a bit expensive for a home system :)
07:17.05SeRiguys is this worth the money? http://www.voiplink.com/Polycom_550_OB_p/polycom-550-ob.htm
07:17.45andyoutsidehm
07:17.50autofsckki have VOIP on my home too, im new to asterisk, today is the first day i could receive and make calls through my * box :D
07:17.58andyoutsidelets see how much is the 650
07:18.46ChannelZWhat do you need/want FXO for then?
07:19.21SeRiautofsckk, cool.
07:19.23autofsckknot for me
07:19.28ChannelZah
07:19.30autofsckkfor a friend of mine
07:19.44autofsckkSeRi: thanks
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07:19.53SeRiautofsckk, I have a PAPt2-NA and works grate. no echos no issues.
07:20.54ChannelZBut that's FXS
07:20.56autofsckki have a pap2 too, and i used to connect that to my itsp, used to have a little delay, but now its connected to my asterisk box and it sounds better
07:21.31andyoutsideI forget off hand what the difference is between the 550 and the 650 but I would go with the 650 for 20 more dollars
07:21.35SeRiwell it seems like a good deal.  cant find it for no less than 250 and i like the fact that it has a backlit lcd
07:21.51SeRiandyoutside, you think so?
07:22.07autofsckkChannelZ: i need to connect to pstn with fxo because where i live, there are no available numbers, i mean, my itsp doesnt have tel numbers from here
07:22.33ChannelZwow
07:22.55SeRiandyoutside, where do you se it for 20 more dollars?
07:22.56andyoutsideSeRi: for the 20 if you are buying a few I would go for the 650 I want to say it has a few better things in it
07:23.23autofsckki would like to buy a little usb sangoma FXO to connect it to my box
07:23.36andyoutsidehttp://compare.ebay.com/like/120807144401?var=lv&ltyp=AllFixedPriceItemTypes&var=sbar&_lwgsi=y
07:23.44andyoutsidehttp://www.google.com/products/catalog?q=polycom+650&hl=en&prmd=imvns&biw=1280&bih=655&um=1&ie=UTF-8&tbm=shop&cid=1354935552959657688&sa=X&ei=5Ni4TvHjDMG1tweeqpnGBw&ved=0CHwQ8wIwADg8
07:24.10ChannelZI don't really know of any other FXO that work well in the same price range
07:25.01SeRiandyoutside, thanks
07:25.06andyoutsidenp
07:25.51autofsckki saw digium clones at an excellent price, having the same issues as digium cards heheheheh, i forgot the name of those cards
07:26.31autofsckkthey also have the same irq problems as the original ones
07:26.41ChannelZmy TDM works great, once you tune it.. I'm not even using hardware echo cancellation
07:26.55ChannelZyou might be talking about the old single-port cards
07:27.52autofsckkthats what i was going to ask, what about the echo with those cards, is it really needed the echo cancellation card? or it depends on the land lines of your country?
07:28.48autofsckkChannelZ: no im talking about the openvox cards, have you tried them?
07:29.04ChannelZI suppose, and/or if you have not a very powerful computer to do software cancel
07:29.15ChannelZno sorry
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07:31.03autofsckkfor soft echo cancel what do you need the most? CPU? or RAM? or both? ha
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07:33.21ChannelZCPU.  It's not really a huge burden for small numbers of channels.
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07:34.44ChannelZIf you've got a big analog installation with one or more of those 24-port cards and high utilization, it's more worth it to get the HWEC module.
07:34.48autofsckkdo atom CPU work well?
07:35.22olliiautofsckk: how many extensions?
07:35.50andyoutsidefor server or destop or embeded
07:36.46autofsckkollii: 5 maybe, or how much extensions can handle with good quality?
07:37.25SeRiautofsckk, my alix handles up to 6 calls simultaneously with no issues... I have not tried more.
07:37.35SeRiAlix 2D3
07:37.46olliidepends on your needs...5 extensions should be fine...transcoding could be not good enoug
07:38.36olliiwith a generic atom we handle about 20 extensions with ~ 10 calls simultaneously and offering a php/mysql gui
07:39.20SeRig/n all!
07:39.36autofsckkollii: how much ram?
07:39.40autofsckkSeRi: good night
07:40.06ollii1024
07:40.16olliiram should not be a problem
07:40.51olliiif you do some transcoding (switching codecs...maybe from g729 to g711) could bring your cpu in trouble
07:40.55autofsckki think so, i have read about the capability of some old processors with so little ram doing good jobs
07:42.15autofsckkive been reading for the last month or so about asterisk, i can understand a lot more now, and it rocks
07:43.36olliibut please secure it...
07:43.47olliiotherwise it will be a very expensive try ;)
07:43.57autofsckkhehehe yes i know
07:44.54autofsckkmy voip is limited, i just get the amount of credit i put it, so isnt really very dangerous
07:46.13autofsckkbut i have read about security too, i think is not so insecure
07:47.30autofsckkollii: what linux distro do you use to put * on?
07:47.49olliiautofsckk: ubuntu server / centos
07:48.01olliiwith asterisk from source and own patches
07:49.26autofsckki installed centos on a vm and built ast from source too, but i dont use centos, so i had to look for some things i didnt know, it took me a lot of time
07:50.15autofsckkand i didnt like centos, it consumes a lot of ram i think
07:50.50autofsckkbut i see that centos is like the distro often used
07:51.14olliiits very near to redhat
07:51.20olliiso it might be a good choice
07:51.42olliibut if you use * from source you could almost use what you want
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07:52.22autofsckkdo you run ast as non-root user?
07:53.11olliiunfortunately no
07:53.36olliiwe ran into some issues dont remember what they were
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07:58.23autofsckkwell thanks for the help again everybody, good night
07:58.33autofsckksee you tomorrow
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08:30.13ik_5hello
08:31.01andyoutsidehello
08:31.15andyoutsideplease enter more data
08:31.36ik_5when a bridged call is hangup, I'm loosing all of the set variables of both channels. Does anyone know on a way to keep this information and access it after hangup ?
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08:35.27bulkorokdid you trie to send it to the h-extension?
08:37.22bulkorokik_5: description can be found here: http://www.the-asterisk-book.com/unstable/besondere-extensions.html#h-extension
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08:39.07_N1xhi guys , anybody help me about asterisk stress test?
08:40.08ik_5bulkorok, i did try the h extension, but it DumpChan does not display my variables
08:40.35ik_5bulkorok, I think this is because of the bridge cmd
08:42.40bulkorokik_5 maybe you have to set the variables after bridge again!? just a guess....
08:43.19ik_5bulkorok, I'll try, thanks
08:43.32bulkorok:-)
08:44.16singleryou could try using __variable while setting, this way variables will propagate to subchannells
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08:45.14olliibut is there any subchannel?
08:45.31bulkorok_N1x you can make stress-tests with sipp and a good dialplan. more info is here: http://www.voip-info.org/wiki/view/Asterisk+dimensioning
08:45.37singlerhe is using bridge, so I guess yes
08:45.56ik_5singler, I'm already using __variable
08:46.20singleroh
08:46.41bulkorokyeah... the underscore is the correct option
08:47.28_N1xbulletrtr: yep i making right now , but i need advices and recommendations at advance level :)
08:47.53_N1xbulletrtr: sorry .
08:48.02_N1xbulkorok:
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08:50.37krotoshi :)
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08:58.40schmidtsgood morning
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08:59.15th0mzhi
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09:01.53krotosi'm using ami for making a controll on active channel, and its duration
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09:04.07krotosif i need a list of active channel and then i want to hangup someone of this
09:04.45krotoswhich command i need to use for listing channels (using ami)? Sip show channels give me an incomplete name of channels for use channel request hangup SIP/
09:06.06kaldemarkrotos: CoreShowChannels
09:06.56kaldemarkrotos: if that doesn't show complete channel names, use Command to execute "core show channels concise".
09:09.59krotoskaldemar: thankyou, concise is that i need
09:10.00bulkorokkrotos you can use "core show channels verbose" too... you get some more information with this...
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09:34.32dymHey - im using a Primux 2S2M ISDN Controller - which is the best channel driver to use for faxing these days - chan_capi doesnt seem to be available in 1.8.X
09:35.06WIMPyDo you have any other driver than capi for that thing?
09:35.18dymGood question
09:35.34dymchan_capi compile seems to fail anyways
09:35.40WIMPyguesses no
09:35.54WIMPyDid you get the git version?
09:36.11WIMPyOr svn rather
09:36.19irrootdym i had it work on mISDN before had to hack the hfcmulti driver to recognise the id and i cant remember if i got the LED's right but it worked
09:36.52WIMPyIt's a HFC thing?
09:37.15irrootif its the same chip WIMPy had a primux 2S it was a HFC chip indeed
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09:37.25irrootbut will need to see the PCID
09:38.03WIMPyPatching in the ID is probably easier than to use capi.
09:38.16dymirroot: fuck
09:38.16irroot<PROTECTED>
09:38.18irroot<PROTECTED>
09:38.25dymhack the hfcmulti driver? :/
09:38.30dymdoesnt sound convenient
09:38.45irrootdym lol check if its the id above
09:38.49irrootlspci
09:38.52dymsec
09:39.02WIMPyOr check if it's a HFC chip at all.
09:39.32irrootWIMPy i have the primux card in my tree seen it only once but still in there
09:39.49dymdoesnt show up - yet to install drivers.
09:39.59dymirroot: did you use the primux drivers from the website?
09:40.02irrootshould see the device
09:40.11irrootnope dym
09:40.24irrootjust had the card it was hfc so i hacked it
09:41.49WIMPyFrom the pictures, there's no HFC on that card.
09:43.05dymlspci: 01:00.0 ISDN controller: Lattice Semiconductor Corporation Device e236
09:43.40irrootdym nope dym not same id
09:43.47dymwell
09:43.50dymthis is the "big" card
09:44.09dym60 channels
09:44.20dymfuck - doing my head in..
09:44.23WIMPysees TI logos
09:44.53irrootdym that is 2XE1 ?? the card i had was 2XBRI
09:45.45dymIts the 2402
09:45.49dymerrr
09:46.12WIMPyWell, the name 2s2m is reather obvious.
09:46.14dym2 ports. Whats XE Exactly?
09:46.26WIMPyXE?
09:46.31dymE1*
09:46.50WIMPyThe line that carries the PRI.
09:47.09dymwell, yes then its 2XE1
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09:47.38_N1xguys how i can increase simultaneosly calls  amount?
09:47.42WIMPyS2M is what youget behing the NT.
09:48.30WIMPythinks keyboards are getting too complicated for him :-(
09:48.43dymthe faq says install asterisk 1.4 - install chan_capi, install card driver
09:48.48dymoutdatedmuch :(
09:49.47dymdahdi?
09:50.06dymi doubt chan_misdn would apply here
09:50.09WIMPyWhat do you dream about at night?
09:50.13dym:D
09:50.21dymso i actually have to use 1.4?
09:50.23WIMPyMost probably not.
09:50.24dym:(
09:50.33kaldemar_N1x: do something about the bottleneck that limits them.
09:51.38WIMPyNo, it does not look like Linux has drivers for that card.
09:51.50WIMPyYou will have to stick with what Gerdes supplies.
09:52.30dymWell, there is capi drivers for linux
09:52.35dymhttp://www.primuxisdn.de/primux/inhalt/download.htm
09:52.42_N1xkaldemar: what you mean?...
09:53.19dymWIMPy: if i use that driver with the card - what would be the equivalent in asterisk? chan_capi only?
09:53.25dymand therefore => 1.4 ?
09:53.42WIMPyYes, chan_capi.
09:53.53dymokay. back to 1.4 then,.
09:53.59WIMPyBut I told you yesterday that it works with 1.8, just not 10.
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09:54.37dymchan_capi works with 1.8?
09:54.39kaldemar_N1x: that you need to provide more information. that kind of a question has no answer.
09:54.40dymi cant seem to compile it
09:55.04WIMPyDid you use the svn version?
09:55.38dymthe current website one
09:55.54WIMPyThat's for 1.6.
09:56.23dymhttp://pastebin.com/zRF1GJu1
09:56.30_N1xkaldemar: i making sipp stress test , and i cant making more than 200 sip channels
09:56.35WIMPysvn co svn://svn.chan-capi.org/chan-capi/trunk chan-capi-trunk
09:56.52_N1xand need to increase
09:57.30kaldemar_N1x: what prevents you from making more than 200? what happens when you try to make more?
09:57.48_N1xkaldemar: i 'll show you , wait a sec. (pastebin)
09:58.08dymWIMPy: ill try.
09:58.43WIMPyAnd I will ad a not to that effect.
10:01.46_N1xkaldemar: http://pastie.org/2830030
10:02.31kaldemar_N1x: did you notice the "Try increasing max file descriptors with ulimit -n" part in you pastebin?
10:02.35WIMPy_N1x: "too many open files"
10:03.07_N1xkaldemar: i see this debug but , how to do this command? or where?
10:03.11_N1xin asterisk cli?
10:03.11WIMPyI know it's mean, but sometimes it helps to read.
10:03.27kaldemar_N1x: ulimit is a system command. "man ulimit"
10:03.48_N1xkaldemar: not in my debian
10:03.54wdoekes2(man ulimit will get you the lib call, you want man bash)
10:04.08_N1xroot@debian:/etc/asterisk# unlimit
10:04.08_N1x-bash: unlimit: command not found
10:04.21kaldemar_N1x: who said unlimit? ulimit.
10:04.49_N1xroot@debian:/etc/asterisk# ulimit -n
10:04.49_N1x1024
10:04.58*** join/#asterisk markusl (~markus@carbon.gonicus.de)
10:05.07wdoekes2_N1x: adjust your init script to set ulimit -n 8192 just before starting the asterisk daemon
10:05.18kaldemarwdoekes2 was right, man bash or man sh will get you the help that has ulimit.
10:07.34_N1xhm go to test.
10:08.40WIMPyOuch. Gerdes still advertise that they support 1TR6 even on the leaflet telling you about their new PCI-e cards.
10:09.51_N1xkaldemar: wdoekes2 working thanks guys :)
10:10.41beccaraIs anyone able to tell me what the difference between local and remote bridging on RTP in calls is and is this now whats known as packet2packet bridging?
10:11.41*** join/#asterisk mathi (~Matthew@78.129.48.220)
10:16.26mathihi
10:20.08olliibeccara: mediastream over asterisk and mediastream directly between to peers
10:20.54bulkorokcan anyone help with compiling errors of ptlib for t38modem!?
10:21.04olliihttp://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite
10:21.33beccaraollii, cheers, so local bridge = rtp going via the asterisk server, remote bridge = rtp doing directly to the peer? If so does the local bridge mean the RTP is going via the asterisk core? I'm trying to get to a state where it's doing p2p bridging to reduce the load
10:29.23*** join/#asterisk happylife (~happylife@212.92.145.7)
10:38.02dymWIMPy: still aroundß
10:38.39dym== Parsing '/etc/asterisk/capi.conf': == Found [Nov 8 11:37:25] WARNING[11944]: chan_capi.c:8273 cc_init_capi: CAPI not installed, chan_capi disabled!
10:38.42dym:/
10:39.10WIMPyDosn't look bad.
10:39.15WIMPySo get the capi going.
10:39.24dymi installed the cards capi driver
10:39.43WIMPyDid you start it?
10:39.56dym(:
10:40.12WIMPyCan't remember the sames. Look at what the capi utils package offfers.
10:40.21WIMPycapiinit or something.
10:40.41WIMPyIt needs a conf file defining your interfaces.
10:43.55dymhttp://pastebin.com/d3uTBhvJ
10:44.52WIMPyHmm. rmmod *capi* and try capiinit again?
10:45.28dymapparently doesnt exist.
10:45.35dymwell - this is the oddish primux capi driver probably
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10:46.06WIMPyBut it moans about the main capi.ko, not the cards driver.
10:46.31dymthis is the systems capi driver?
10:46.41WIMPyThe framework.
10:46.56WIMPyWhat does modprobe capi give?
10:47.08WIMPyAnything useful in dmesg?
10:48.11dymi havent installed any
10:48.12dymFATAL: Error inserting capi (/lib/modules/2.6.32-5-amd64/kernel/drivers/isdn/capi/capi.ko): Device or resource busy
10:48.27WIMPyany what?
10:49.11dymdisregard that
10:49.28dymhttp://pastebin.com/ayXmwzB9
10:49.51WIMPyLooks like you're missing the devices.
10:50.01_N1xkaldemar: wdoekes2 guys http://pastie.org/2830204
10:50.06dymas in /dev/CAPI ?
10:50.10_N1xwhat is this?
10:50.20WIMPyOr the device numbers are used by something else.
10:50.29WIMPyyes
10:50.55dyminexistant
10:50.58dymno such devices in /dev
10:51.12WIMPygrep 68 /proc/devices
10:52.18dymurgh
10:52.20dymhttp://pastebin.com/6kW1TmLJ
10:52.22dymthat cant be good
10:52.36kaldemar_N1x: did you do what the output suggested?
10:53.05WIMPyDepends on the minors, but as it does mona, that looks rather unpleasant.
10:53.20WIMPymoan
10:53.33_N1xkaldemar: sorry?
10:53.53dymWIMPy: any idea on correcting this issue?
10:54.20kaldemar_N1x: read what it says to read?
10:55.06WIMPyGet rid of whatever claims 68 for sd or patch one of them to use another major.
10:55.15WIMPyPretty uncool.
10:55.33dymsd must be a hd
10:55.46_N1xkaldemar: how i can show current calls?
10:55.49_N1xsip show channels?
10:55.50dymthis means i'd have to hardcode another major into the driver?
10:56.29WIMPyIf I look in to /proc/devices here, I've got tons of sd. All but "8" are unused.
10:56.32kaldemar_N1x: "core show channels" and "core show calls"
10:57.33dymthis is really fucked up
10:57.40WIMPyyes
10:58.08dymWIMPy: also capiinit is not necessarily needed
11:00.33irrootdym www.unfuckitup.com :P
11:00.44WIMPyLooks like we have seriousely run out of device numbers.
11:01.08dymOkay, any idea how i could fix this issue?
11:01.23dymi r equipped with too basic linux knowledge for this.
11:01.26irrootdym not sure but enjoy that site
11:01.49_N1xkaldemar: when i making more than 2000 requests to asterisk from sipp , output is http://pastebin.com/mzuKQF1v
11:01.56WIMPyYou will have to either modify sd or capi and recompile your kernel.
11:02.04_N1xwhat is this?
11:02.21dymomfg
11:02.22WIMPyUnless there's soe magic parameter for sd that tells it hwich/how many majors to claim.
11:02.36_N1xi see my cpu is not full loaded
11:02.48dymWIMPy: what about modding the major within the capi drivers code and recompiling that?
11:02.48irrootdym the problem with any system FOSS in particular if the manufacturer does not supply a solution that is taken up or does not supply the drivers then its hard to get anywhere
11:03.03dymirroot: so i have to install suse? :(
11:03.08WIMPyThat is part of the kernel.
11:03.11dymdont say yes - ill cry
11:03.22dymit seems suse uses different major ranges then
11:03.39irrootdym no not at all unless suse has the bits you after but then you could get them from suse either way
11:03.54dymfuck
11:04.07dymso either kernel recompile with some modifications i have no idea about
11:04.09irrootuse of udev ??
11:04.09dymor changing to suse?
11:04.16dymirroot: care to explain?
11:04.19dymi know udev
11:04.40irrootif its using udev then this should be taken care of
11:04.50irrootnot all systems use it properly
11:05.06dymwell
11:05.09dymudev is installed
11:05.13dymcan i enforce a major change?
11:05.18WIMPyLoad capi before sd *eg*
11:06.15WIMPyIt would be possible.
11:06.51dymmhhh
11:07.01dymfrom K01lcapiinit:
11:07.03dym<PROTECTED>
11:08.25dymWIMPy: couldnt i just change it there?
11:09.03WIMPyYes, but your capi wouldn't know to dind to it.
11:09.06WIMPybind
11:09.40dymexcuse me? if i changed it there, things would stop working?
11:10.16WIMPyCONFIG_SD_EXTRA_DEVS
11:10.44WIMPySoory. That's obsolete.
11:11.35*** join/#asterisk freeman_u (~freeman@gw.quart-soft.net)
11:12.46dymWIMPy: Save my day!
11:12.47dym:(
11:13.08WIMPyAre you using an initrd?
11:15.16dymThis is Debian Squeeze Disk Install with grub2
11:15.23dymbut i guess so.
11:15.50WIMPyThen try to load capi before you load sd.
11:15.55kaldemar_N1x: http://svn.digium.com/svn/asterisk/tags/1.8.0/doc/sip-retransmit.txt
11:16.11WIMPyMaybe that's finally a use for an initrd :-)
11:17.02dymWIMPy: where would this be done?
11:17.25WIMPySomewhere in your init scripts.
11:17.25*** join/#asterisk JuanCri (~JuanCri@pc-205-210-86-200.cm.vtr.net)
11:17.37WIMPyBut I have NFI what they look like on Debian.
11:19.35_N1xkaldemar: i already see it , but how to resolve?
11:20.57dymWIMPy: http://pastebin.com/yxvfwSsX
11:21.08dymim not sure where the sd deviced are initialized.
11:21.37WIMPyThere will be a lot of modprobing somewhere.
11:22.49WIMPyI whould think there must be a parameter to sd, but I cant find any.
11:24.56kaldemar_N1x: maybe your HW hit its limits. buy a beefier box.
11:25.37dymmhhh
11:26.25*** join/#asterisk mathi (~Matthew@78.129.48.220)
11:26.30mathihi
11:27.11carrarHARRO
11:27.28dymWIMPy: if i changed the major in the capi init, would that disturb some other software cause of it addressing capi by its major?
11:28.13carrarGOODBYE Belgium
11:28.38WIMPyThe device inodes need to match the ones the kernel drivers bind to.
11:29.15dymand on compile of the capi driver, it selected whatever the kernel needs
11:29.42WIMPyThe kernel capi, yes.
11:29.47*** join/#asterisk mathi (~Matthew@78.129.48.220)
11:30.00mathihi
11:30.11dymfudge
11:30.15carrarHARRO
11:30.27dymcarrar: shut up.
11:30.30dymwe've seen you
11:30.30mathiwhy does Asterisk output WAV files and not for ex. MP3 files which takes less space on disk ?
11:30.32WIMPymathi: Jumping for joy today? *eg*
11:30.40carrarIt's dark here
11:30.43dymmathi: cause you told it to.
11:30.43carrarYou can't see me
11:30.49mathiWIMPy, hi, it's "dokg". this is my other nick
11:31.11mathidym, I told it to ?
11:32.40carrarmathi, which output are you talking about?
11:32.45carrarvoicemail?
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11:34.20dymmathi: course you did.
11:34.35dymWIMPy: any idea on how to proceed now?
11:34.59mathino, actually in my IVR he could leave a message if he press some menu (I don't need to record the whole call). Then I record this message in an audio file, and I have to send it to a remote server (I am restricted to FTP). That's why I would like the best quality/size compromise. WAV files are quite huge it seems to me, thus sending it by FTP may take some time, plus huge disk space on the remote server.
11:35.07WIMPyNo better idea that the module loading order or patching so far.
11:35.25carrardefaulted to that
11:35.28carrarchange it
11:35.30carrarread the docs
11:35.44carrarcore show application Record
11:35.46mathicarrar, me?
11:35.59carrarSee anyone else asking that question?
11:36.12mathicarrar, but what format do you suggest in my case ?
11:36.24carrarI'd just go with what you have personally
11:36.33carrarbut you don't seem to like it
11:36.50mathiI have nothing now, I am just studying the case)
11:36.52carrarYou sox to change it to a mp3
11:36.58mathisox ?
11:36.59carraror use ogg or flac
11:37.04carrarman sox
11:37.12mathiI can only use mp3, wav, or ogg
11:37.13carraryou=use
11:37.24carrarSounds great then
11:37.33carrarconvert it to one of those
11:38.06mathicarrar, but look I'm confused that in the reference guide we talk so much about WAV, as I could use MP3 and use less space seems to me
11:39.07WIMPymathi: If it's really about size, save as wav and convert to mp3 or ogg with a fine-tuned set of filters. They make quite some impact for voice.
11:39.36mathiWIMPy, like which converter ?
11:39.56WIMPyThe one you like.
11:40.41mathiWIMPy, why would I need a "fine-tuned set of filters" ?
11:40.53WIMPyWhat does the following line mean?
11:40.55WIMPyERROR[32019]: lock.c:258 __ast_pthread_mutex_lock: chan_sip.c line 14833 (register_verify): 'peer' really deep reentrancy!
11:41.16dymWIMPy: kinky ;)
11:41.36WIMPymathi: Because you can get much better size/quality ratio.
11:41.53mathiWIMPy, are you using one ?
11:42.06carrarmathi
11:42.09carrarMAN SOX
11:42.17WIMPyI experimentd with it many years ago.
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11:42.32carrarrecord it as a wav, then on the next line use sox t convert it to mp3
11:42.42carrarboom
11:42.45carraryour done
11:42.46WIMPymp3enc, lame, oggenc
11:43.11carraror one of those too
11:43.14carrarYOU PICK
11:43.17carrar:)
11:43.32mathicarrar, omg I thought you were talking about a man and his socks, I had no idea what ou were talking about
11:43.32dymcarrar: require chill pills?
11:43.36carrarYou ahve a PLETHORA of options!
11:43.38WIMPyAnd keep in mind, that you need a licence for mp3.
11:43.50carrarUNIX command line "man sox"
11:43.56carrarwihtout the quotes :)
11:44.10carrarif you have sox installed
11:44.16carrarwhich you probably don't
11:44.17carrarheh
11:44.35carraryes you also need a license to be on the INTERNET
11:44.42carrarplease quire
11:44.44carraraquire
11:44.44mathiuhm?
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11:44.59dymcarrar: do you always photograph yourself in shorts?
11:45.02mathiWIMPy,  licence to convert or to use MP3 ?
11:45.04carrarALWAYS
11:45.10carrarcause thats all I wear
11:45.15WIMPymathi: Both.
11:45.23carrarin fact
11:45.27carrarI have em on right now
11:45.30dymnice
11:45.42mathiWIMPy, and if I don't buy the licence ?
11:45.44carrarI just had real sushi with my shorts on
11:45.51WIMPycarrar: How many? 2 or 3? ecnr
11:45.53WIMPyscnr
11:46.04carrarscnr?
11:46.07carrarplates?
11:46.17carrarI had 13 plates
11:46.22WIMPymathi: You're an evil thief.
11:46.39carrarmathi, probably nothing
11:46.49carrarso use OGG
11:46.59mathiWIMPy, where do I buy it?
11:47.08carrarheh
11:47.31WIMPyFraunhofer Institure
11:47.42carrarThat is a good question by the way
11:47.43WIMPyOr use ogg/vorbis
11:48.27mathiI wonder how many users use MP3 without licence ? millions ?
11:48.40carrar7 billion
11:48.46WIMPyPlayback is free for private use.
11:49.22carraruse FLAC
11:49.27carrarit's free
11:49.55mathicarrar, I can't, I need <audio> in HTML5 for playback, I am limited to ogg, mp3 and wav for chrome
11:50.11mathiit's a private web app, users use chrome
11:50.13carrarogg
11:50.14WIMPyThen use ogg.
11:50.18mathiok:)
11:52.03carrarAny other questions?
11:54.29WIMPyYes, what are next weeks lottery numbers?
11:54.49carrar23 44 18 12 88 44 -- 42
11:55.07WIMPyOut of range
11:55.29WIMPyAnd a dupe
11:55.40carrarYou chances of winning could be lower
11:55.41WIMPyNow I can't trust you any more.
11:55.59carrarprobably as a result of never playing lottery
11:57.10WIMPyI usually try every few months, but this time it must have been more than half a year ago.
12:00.28mathicarrar, yes about speech recognition in french, in my IVR I would like people to pronounce a date, without typing on phone. typing a date is not that easy
12:01.19carrarSIRI
12:03.29cuscohello folks
12:04.31cuscoquestion: can I make cdr log an extra variable without the need to use Set(CDR(myVar)=${myVar}); in every dialplan that it is on?
12:04.47carrarmathi, try google? "speech recognition asterisk french"
12:04.54cuscocan I somehow bind a var to cdr everytimt that it is set?
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12:21.23patrickod_I'm tying to use SayAlpha in a test extension yet when I dial that extension I get no audio
12:21.28mathicarrar, there are many alternatives, anything to suggest ?
12:21.35patrickod_the asterisk CLI shows it playing the required letters without error
12:21.43patrickod_any ideas as to what might be causing the silence ?
12:22.37singlerdid you answer the call?
12:23.08patrickod_yep I just figured that out. thanks :)
12:23.15singlernp :)
12:23.53carrarmathi, I don't use speech recognition
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12:29.19puzzledhi
12:29.40leifmadsenohai
12:29.55carrarkonnbanwa!
12:31.05ixyd_hi guys, is it possible, to configure CEL in a way that it works in some kind of a batch mode like CDR?
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12:39.59leifmadsenixyd_: I don't think that's the point of CEL though
12:50.22ixyd_i dont see why using a batchmode with CEL would be more strange than it is with CDR?
12:53.38dymgreets leifmadsen
12:55.12leifmadsendym: ohai
12:55.30leifmadsenixyd_: oh wait, I think I misunderstood what you meant
12:55.48ixyd_ah :)
12:56.45leifmadsenixyd_: and yes I understand what you mean now, I just did a search in the conf files, and it doesn't appear to exist (batch mode)
12:57.12leifmadsenand I searched all the CEL .c code for batch, and no hits there either
12:57.38ixyd_ah ok thank you very much! :)
13:01.01leifmadsennp!
13:01.15leifmadsenixyd_: would be a good feature I agree, I wonder how hard it would be to implement...
13:02.02ixyd_would be very nice to have definitely!
13:02.23leifmadsenjust for fun I'm going to see if I can find the batch code in CDR and see how much there is of it
13:02.32leifmadsenif you wanted to implement it that's where I'd start
13:02.35leifmadsen(I'm also not a programmer)
13:03.43ixyd_iam not a real programmer to, i have to wait for the request of the customer...maybe you'll get a patch from us later ;)
13:06.05leifmadsenixyd_: ya, so you want to look in main/cdr.c and add the code to main/cel.c for batch mode. There is a lot of code, but none of it looks that complicated. Most of cdr.c seems to actually be batch code. I'd start there and see if you can make cel.c contain batch mode as well
13:06.18leifmadsenixyd_: good luck! ping me if you end up filing an issue with a patch
13:06.33ixyd_i will do so! thank you leif! :)
13:06.40leifmadsennp
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13:26.47olliiis their a list where all soundfiles for * 1.8 are named with its content?
13:28.17p3nguinYes.  Look at the .txt files in the sounds directory.
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13:29.12olliithat is to easy... :X
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13:29.26olliithank you.. ;-)
13:29.32p3nguinYou should have the core-sounds and the extra-sounds text files.
13:30.15ppeejjaayyHi everyone, has anyone tried DYNAMIC_FEATURES in Asterisk 1.4.36 ? i'm trying to use the application Goto as DYNAMIC_FEATURE
13:30.29olliip3nguin: yeah found it...that was way to easy
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13:33.39ppeejjaayywhile in a call i need to redirect the customer to an IVR, so i'm trying the DYNAMIC_FEATURES as a solution. I was able to launch a macro. In the macro I can't do waitExten, so i need to use the Goto app directly from the feature
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13:34.48ppeejjaayyfeatureTest => *77, Goto,myContext|myExtension|1
13:35.46mandlairroot, can you help me identify a line in my extension.conf where should add ,,t
13:35.57*** join/#asterisk mathi (~Matthew@78.129.48.220)
13:36.28irrootmandla its the line with a "dial" to the extension you looking for
13:36.44irrootthe problem with your extensions.conf is there many comments
13:37.10kaldemarppeejjaayy: looks like your feature is missing the <ActivateOn>[/<ActivatedBy>] part entirely.
13:37.13mandlairroot, man there are many lines with dial.
13:37.26irrootyes the one with the extension number
13:37.32mandlairroot, ok, let me look it up.
13:37.34ppeejjaayyfeatureTest => *77, peer,Goto,myContext|myExtension|1
13:37.38leifmadsenppeejjaayy: try GoSub
13:37.44ppeejjaayysorry i forgot the peer
13:38.00leifmadsenppeejjaayy: nevermind, I just read you're using 1.4
13:38.18ppeejjaayyin fact the the feature is working but the goto isn't
13:38.34ppeejjaayyi can do launch playbacks, macros, but no goto
13:38.37p3nguinI think I would have just used my transfer key.
13:38.50ppeejjaayywhat's the problem with * 1.4 ?
13:38.55p3nguinOr if I didn't have one, I'd configure DTMF transfers.
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13:39.31timeshellwhat's the command to kill a sip channel in CLI?
13:39.51leifmadsenchannel request hangup
13:40.26kaldemarppeejjaayy: your problem is most likely with more than one argument. try featureTest => *77,peer,Goto,"myContext|myExtension|1" or featureTest => *77,peer,Goto(myContext|myExtension|1), that's how its documented in newer versions.
13:40.57p3nguinIt seems like if you want to transfer a call to some arbitrary extension, the transfer key would be a good way to accomplish it.
13:41.03timeshellty
13:41.46leifmadsenkaldemar: in 1.4 I'm pretty sure arguments are separated with commas like he is doing
13:41.52leifmadsennewer versions are more sane
13:42.05ppeejjaayykaldemar: i'll try it right away, i thought the syntax is : AppName,Argument1|Argument2
13:42.08leifmadsenalthough I haven't used 1.4 in years
13:42.15mandlairroot, the only line with the extension is exten = _X.,1,Goto(default,917,1)
13:42.17leifmadsenppeejjaayy: look at the sample file, and follow that
13:42.36mathiwhy are software for asterisk usually sold per port/channel ?
13:42.53[TK]D-Fender<ppeejjaayy> while in a call i need to redirect the customer to an IVR, so i'm trying the DYNAMIC_FEATURES as a solution. I was able to launch a macro. In the macro I can't do waitExten, so i need to use the Goto app directly from the feature <--- wrong idea.
13:42.58irrootcool mandla now look in default for a priority 1 that matches 917
13:43.00kaldemarleifmadsen: 1.4 has the MOH class option also. depends on how the parser handles pipes.
13:43.01[TK]D-Fenderppeejjaayy, Just transfer the call.
13:43.15irrootit could be exten => _XXX,1,......
13:43.18[TK]D-Fenderppeejjaayy, Dynamic features are for in-line little bits not "transfer the call"
13:43.40leifmadsenoh ya, if the purpose is to transfer the call... just do that. There is built-in transfers or you can use the transfers from the phone.
13:45.12leifmadsenppeejjaayy: oh, and this:  http://pastebin.com/4NqZi8pR
13:45.21[TK]D-Fenderppeejjaayy, Multiple failures : The call stays bridged with the dynamic feature.  Also your IVR will end up using waitexten which will fail regardless.
13:45.43ppeejjaayyduring a call i want to launch an IVR to get Credit Card numbers by DTMF, so i thought using the dynamic_feature to a context where i'll put playbacks and waitExten
13:46.05mandlairroot, all entries in [default] are commented out, lol.
13:46.20irrootmandla is there an include ??
13:46.29irrootlook in the included section
13:46.37ppeejjaayyleifmadsen, thx for the link
13:46.53leifmadsenppeejjaayy: thats from features.conf.sample fyi
13:47.02mandlairroot, in my dialplan??
13:47.23mandlairroot, or in default?/
13:47.37irrootyes in extensions.conf either a file is included or a context is included in [default]
13:48.51mandlairroot, true, it includes demo, but its commented by asterisk gui.
13:48.57leifmadsenyuck [default]
13:49.34ppeejjaayyleifmadsen: indeed, i haven't read that part. so in my case what should i do ? transfer the call ?
13:49.43[TK]D-Fenderppeejjaayy, Yes
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13:50.18ppeejjaayyand when the customer finishes filling the data, retransfer to the original channel
13:50.33timeshellIs there a way in CLI to tell asterisk to stop trying to register a trunk?
13:50.34[TK]D-Fenderppeejjaayy, Sure
13:50.38timeshellWithout removing the trunk?
13:50.56[TK]D-Fendertimeshell, nope
13:51.09timeshellWell, there's a feature request.
13:51.11irrootmandla is it a file include or a context include ?
13:51.14ppeejjaayyD-Fender, thx
13:51.35irrootmandla looking at the calltrace it ends up in default so should be there
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13:54.03mandlairroot, its a context include, include=demo
13:54.20irrootmandla now find it in there ....
13:54.27irrootlook for [demo]
13:54.38mandlairroot, in demo, right
13:54.43mandlairroot, thanx
13:55.54mandlairroot, t should be the 3rd entry right??
13:56.57irrootmandla what you mean by 3 ? its Dial(<CHAN>,<TIMEOUT>,<OPTIONS>)  so the 3 opt yes
13:57.34ppeejjaayyD-Fender, how do i trigger the transfer ?
13:57.40mandlairroot, thats what i meant.
13:58.19ppeejjaayycan it be done from the dynamic features ?
13:58.28irrootmandla when you done do "dialplan reload"
13:58.50irrootand then maybe "dialplan show <NUM>@default"
13:59.00[TK]D-Fenderppeejjaayy, No... transfer is one of the most boring basic features of your phone
13:59.15[TK]D-Fenderppeejjaayy, What are yuo using?
13:59.24irrootsorry no @
14:00.06irrootwith it works i typo here
14:00.11ppeejjaayyas softphone ? a homemade
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14:01.35[TK]D-Fenderppeejjaayy, If you did it yourself.. you didn't make a SIP transfer feature?
14:01.48[TK]D-Fenderppeejjaayy, If not use the dial options for transfers via DTMF
14:01.54[TK]D-Fenderppeejjaayy, Thsi is not "dynamic features"
14:06.54WIMPyWhat does this mean or what should I watch out for? - ERROR[32019]: lock.c:258 __ast_pthread_mutex_lock: chan_sip.c line 14833 (register_verify): 'peer' really deep reentrancy!
14:09.42[TK]D-FenderWIMPy, Dunno ... sounds kinky ;)
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14:10.03WIMPyIt says ERROR. That's usually not a good thing.
14:10.56irrootWIMPy lol its a lock on the same resource in a loop without unlock ??
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14:12.14WIMPyHmm. I have 3 locks on sip_send_mwi_to_peer. But I got quite some of the above messages.
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14:20.07hudonyHi : sorry to ask it here but cant find a final answer : using cisco spa-303 and a poe switch.  The phone doesn't use poe from ethernet, I still have to use ac adapter.  When googling, I found that an adapter seems to be required to opereate POE on the phone.  IS that right?
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14:22.33[TK]D-Fenderhudony, What switch?
14:23.00hudonycisco sf300 24 ports
14:23.12hudonymanaged switch and I have enable poe
14:23.32hudonyall ports have it enabled but I can see 0 consumed mW
14:23.38mandlairroot, still not working.
14:23.51irrootmandla is it set
14:24.01mandlairroot, still doing that native bridging thing.
14:24.08irrootand when you dial you see it in the output
14:24.17mandlairroot, yah i think its set.
14:24.24kaldemarhudony: why did you think that the spa-303 would use PoE in the first place?
14:25.05hudonyah
14:25.08hudonydman
14:25.12[TK]D-Fenderhttp://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps10998/data_sheet_c78-601648.html
14:25.15hudonyjust noticed that not
14:25.31[TK]D-FenderPower over Ethernet (PoE) Support SPA 303G No
14:25.33hudonyguess ive been confused by the possibility to add it through the optional adapter
14:25.36hudonymy bad
14:27.08hudonythx anyway guys
14:27.16hudonyThx god only ordered 2 of these
14:27.21hudony:S
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14:30.52[TK]D-Fenderhudony, You really really should read the specs on products you order...
14:31.03hudonyI can tell :(
14:31.36[TK]D-Fenderhudony, God-aweful expensive switch ... to power cheap-ass phones with no PoE.  Prioritize man!
14:31.50leifmadsenI really like the D-Link switches with PoE in them
14:32.19hudonyWell.. the guy here only wanted cisco switch so we ordered that one for POE
14:32.31leifmadsenpowers Polycom phones quite well. The problem is that (and this seems to be an issue with all PoE switches) is that if you order a 24 port switch, don't expect to have enough power load in the switch to power 24 PoE phones
14:32.36hudonyBut the phone choice is a total fuck up
14:32.38hudonyhehe
14:33.02leifmadsenwhen you do the math, you can do about 8-10 phones at full power draw (I assume full draw happens at boot up)
14:33.30hudonyoh
14:33.32hudonyok
14:33.35hudonywe have 16 phones
14:33.51hudonyso I guess some will have to use ac
14:34.22Naikrovektest it before you make that assumption
14:34.36Naikrovekthe phones actually draw very little once they're booted up.
14:34.57[TK]D-FenderNaikrovek, The spec sheet says "don't bother"
14:34.57leifmadsenright
14:35.10[TK]D-Fender"<[TK]D-Fender> Power over Ethernet (PoE) Support SPA 303G No"
14:35.22leifmadsenthe only issue would be if you had all 24 phones boot up at the same time (like after a power outage)
14:35.37hudonyoh ok
14:35.46hudonyFound 504g and was looking for his power consumption
14:35.48leifmadsenwonder if Polycoms have a staggered boot sequence you could enable, then that'd not be a problem
14:36.20Naikrovekthe switches stagger by a few milliseconds in my experience
14:36.26WIMPyI guess you'd need that feature in the switch.
14:36.28Naikrovekthey don't all offer full power at once
14:36.40Naikrovekas the ports are brought online they are powered
14:36.55coppicePoE switches have considerable complexity in them to avoid trouble like that
14:36.57leifmadsenNaikrovek: right, but what happens when all the phones request power at the same time
14:37.06Naikrovekthey have to be powered on to request more power
14:37.20leifmadsenhuh, learn something new every day
14:37.21Naikrovekthe switch doesn't supply power to all ports at once when the power comes back on
14:37.37leifmadsenI've never actually tested all that to see what happens, so that'd be an interesting experiment
14:37.46Naikrovekyeah
14:38.11leifmadsenI applied theory to the solution, and played it safe in a couple of installs. Now I want to try plugging all the phones in and power them at once and see what happens :D
14:38.13coppiceits all laid down in the PoE spec
14:38.13[TK]D-FenderMy D-Link DES-1526's stagger the power-up.
14:38.14Naikrovekthey don't actually "request" power, it's a simple circuit that measures resistance that determines the amount of power to send down the wire
14:38.28leifmadsencoppice: I've never read the PoE spec
14:38.40Naikrovekonly on a 300ft run will a polycom phone need the full 15.4w or whatever it is.
14:38.44leifmadsen[TK]D-Fender: cool, I think those are the same ones I'm using
14:38.49Naikrovekand most of that will be due to losses in the cable
14:38.57coppiceit defines an elaborate procedure to managing the loads
14:39.07Naikrovekput in 22gauge instead of 24 and you'll use less power.
14:39.17Naikrovek(ethernet cable, i mean)
14:39.35leifmadsenNaikrovek: I only use audio cable for my ethernet deployments
14:39.41Naikroveklol
14:39.56leifmadsen16 gauge bwaaaaaa!
14:40.19Naikroveki've found that almost everything that people communally assume is true, is actually false.  i test everything i can anymore, just because i almost always learn something and i've always been a serial contrarian.
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14:44.04Naikrovekit comes with age, i think.  I used to believe just about anything that sounded reasonable.  now i realize that a whole lot of that was 100% fiction.
14:44.42Naikrovekso now i don't believe anything, and whenever someone says something with conviction, I always just presume they're wrong.  Turns out that's the approach that most reflects reality.
14:45.14WIMPyI only believe what I see.
14:45.28WIMPySince the invention of TV I believe everything.
14:46.29[TK]D-Fender"believe half of what you see and nothing that you hear"
14:46.32irrootNaikrovek WIMPy according to the dictionary a pessimist is a experianced optomist
14:46.55[TK]D-Fenderirroot, No, a realist is an experienced pessimist :)
14:48.12WIMPyThe pessimist calls a realist an optimist, while the optimist calls a realis a pessimist.
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14:51.44Naikrovekeveryone seems to call me dorkwad lately.
14:51.53Naikroveks/dorkwad/"dorkwad"/
14:52.08Naikrovekthey hate it when i'm right all the time and resort to calling me names
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14:52.54Naikrovek"install that phone, dorkwad," and "I'm late for pottery class, dorkwad," etc.
14:53.21leifmadsenNaikrovek: "Poor planning on your part does not constitute and emergency on mine"  <-- best cubicle sign ever.
14:53.33leifmadsenefff.... I continually type 'and' when I mean 'an'
14:53.42leifmadsenstupid auto pilot typing
14:54.07Naikrovekyeah that is an awesome weapon against stupidity
14:54.07p3nguinCisco PoE actually does request more or less power, and it is done via cdp.
14:54.41WIMPyYou mean "inline power"?
14:54.55[TK]D-Fender<WIMPy> The pessimist calls a realist an optimist, while the optimist calls a realis a pessimist. <- Sounds about right :)
14:55.46garymcI get the error : retrieve_conf failed, config not applied
14:55.52garymcasterisk wont reload
14:56.04garymcI must have broken something and i cant get it to work
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14:56.09[TK]D-Fendergarymc, Not an Asterisk problem....
14:56.24treborsuxlinux problem?
14:56.28[TK]D-FenderFREEPBX
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15:00.14p3nguinNo, I do not mean inline power.  I mean power.  Cisco CDP allows a device to say, "Hey, I don't need this much power; please turn down your available power."
15:00.38Naikrovekexcept it doesn't actually turn down power, it just changes the power reported as being used
15:01.06Naikrovekcurrent is drawn, not forced.  volts are forced, but poe is a standard voltage always
15:01.38Naikrovekthat CDP negotiation tells the switch how much is actually being used, which tells the switch how much capacity it has left
15:02.18Naikrovekand it uses that capacity to continue to power phones until capacity gets to zero
15:02.29Naikrovekat which point the next phone won't turn on.  the port will appear dead.
15:03.58Naikrovekbut yes, your general point is correct.  cdp does negotiate
15:04.06p3nguinYes it does.
15:04.35p3nguinAnd it can negotiate a lower or a higher available power amount.
15:04.40p3nguinWhich is what I said before.
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15:21.48SeRip3nguin, do you create xml files in your nix box?
15:22.06p3nguinUh, sure, I guess.
15:22.18SeRido you use any specific tool?
15:22.22p3nguinvim
15:22.38SeRivim sets the encoding for xml?
15:22.49p3nguinEncoding?  It's just text.
15:22.53SeRiI thought xml had some type of encoding
15:22.56SeRioo ok
15:23.01SeRiwell burn me :)
15:23.04SeRilol
15:23.05SeRiThanks
15:26.01*** join/#asterisk endemic (~endemic@2001:49f0:400e::5)
15:26.54[TK]D-FenderSeRi, "Build a fire for a man and he's warm for a day.  Light a man on fire and he's warm for the rest of his life." - Terry Pratchett
15:27.28SeRi[TK]D-Fender, very nice. thank you :)
15:34.35SeRi[TK]D-Fender, would this be a valid polycom softkey assignment? http://pastebin.com/raw.php?i=9CrUje1z
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15:36.58lal00is it possible to load a codec without restarting asterisk?
15:37.07p3nguinYes.
15:37.08[TK]D-FenderSeRi, Haven't touched the EFK stuff yet, though I definitely should start
15:37.20p3nguinmodule load codec_ulaw.so   for example.
15:38.28SeRi[TK]D-Fender, I am just going to try and see what heppens :P Do you know how I can disable the default "Forward" label in the screen?
15:39.08[TK]D-FenderSeRi, Not off-hand...
15:39.24SeRiThanks. Back to experimenting :)
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15:40.05lal00p3nguin: thanks
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16:19.21hardwiredislike nickserv sometimes
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16:23.06SeRiI give up
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16:23.08SeRilol
16:23.10SeRiConfiguration file "softkey.cfg" is from template Unknown, revision Unknown
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16:24.11LemensTSanyone here use phpagi swift function
16:24.17in0culahi, i'm on fedora, wich softphone i need to use for testing asterisknow?
16:28.43SeRiin0cula, you could use any softphone
16:28.51SeRia popular one is ekiga
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16:32.45[TK]D-FenderSeRi, Sounds ike you are missing some XML headers
16:33.11SeRiyea thats what I just figured
16:33.19SeRibut its looking for a revision
16:34.50[TK]D-FenderSeRi, Copy the heard tags off a sample
16:35.00SeRijust did. trying now
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16:37.45SeRiit took the file but I dont see the softkey.... not sure why. I am wondering if I am overlapping it or something
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16:57.06LemensTS[Nov  8 04:46:08] NOTICE[20856]: app_swift.c:429 app_exec: DTMF = 1
16:57.07LemensTSDoes this mean it is setting DTMF =1 as a channel variable?
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17:23.57hudonyHi : Been trying for 2 days to get it work by myself but it seems like I can't and I need advice : here is my config : asterisk 1.8 on machine with 2 nic (public and lan) with shorewall on it allowing everything outbound but only 10000-20000 udp and 5060 tcp/udp inbound.  Within the lan, it all goes well but when calling form the outside, the call is initated with no error in the console but...
17:23.59hudony...i got no audio
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17:24.11Qwell~sipnat
17:24.11infobot[~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions .  Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet.
17:24.24hudonyYesterday, D-fender helped me by telling me about nat=yes for my phones and careinvite=no
17:24.26hudonyBut still..
17:24.36Qwelland externip
17:24.39hudonyI read it
17:24.43hudonyexternip and local...
17:24.45hudonyforgot it
17:24.51hudonylocalnet
17:26.07hudonyAnd I have qualify = yes
17:26.18hudonyTried to follow various tutorials
17:29.05hudonyHere is my sip.conf configuration file : http://pastebin.com/28wSLqk4
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17:41.23autofsckkhello, iax2 needs natting too?
17:42.28Qwellhudony: yeah you might want to change your primus password.
17:42.57Qwelland I don't see nat=yes and canreinvite=no in the same peers
17:43.59hudonythis is not the good password
17:44.35hudonyoh my bad, I removed careinvite=no a moment ago for testing purpose
17:45.08hudonyhe's back in but I still have the same proble
17:45.16Qwellwhat ports are set in rtp.conf?
17:45.17hudonyI'm trying to see with sip set debug on
17:45.21hudony10000:20000
17:48.30hudonybut since its a rtp problem...and I don't think I'' find anything interesting
17:52.49hudonyHere is what I get : http://pastebin.com/Wew2vW8B
17:53.25[TK]D-Fenderhudony, add nat=yes to [general] , nat=no to [5142252220]
17:53.46[TK]D-Fenderhudony, Also what ver of * are you using?
17:54.39hudony1.8.7.0
17:54.59hudonyand i saw <--- SIP read from UDP:209.183.11.198:5060 --->
17:55.00hudonySIP/2.0 401 Unauthorized
17:55.06hudonyligne 88 of the pastebin
17:56.01hudonymodemcable082*CLI> sip show registry
17:56.03hudonyHost                                    dnsmgr Username       Refresh State                Reg.Time
17:56.04hudonyst-01.bvoice.primus.ca:5060             N      5142252220@b        74 Registered           Tue, 08 Nov 2011 12:36:55
17:56.06hudony1 SIP registrations.
17:56.11[TK]D-Fenderhudony, should be "directmedia=no" for all sections
17:56.15hudonyok
17:56.19hudonyI'll try that
17:56.37*** join/#asterisk michael-i (~anonymous@204.11.230.58.static.etheric.net)
17:56.46hudonyIt removes the careinvite=no from what i have read so do i remove the instruction
17:56.53hudonyor do I keep both ?
17:57.36hudonyoh nice
17:57.47hudonyI have progress! : I can now hear incoming audio
17:59.24hudonyThe only part still down is that the caller doesn't hear the person who answered the call
17:59.59hudonyI did a tcpdump on the asterisk box and udp packet are sent back from my external interface to primus when a call is occuring
18:00.00hudony:S
18:00.12hudonySo I don't see why this isn't working
18:01.13[TK]D-FenderPB new configs
18:01.27hudonySorry?
18:02.29[TK]D-FenderShow us what you've got now
18:02.51eppigydate with neuroscience girl
18:02.55eppigylife is interesting
18:03.38hudonyyou mean a sip log or my sip.conf file?
18:05.17michael-iIs there a magic flag somewhere to automatically exclude the calling channel from a multiple-party Dial()? E.g, SIP/201, SIP/202 and SIP/203 are in group 200. When 201 dials group 200, I want it to not call 201.
18:07.02[TK]D-FenderPB new configs
18:07.22[TK]D-Fendermichael-i, Don't put it in your dial.
18:07.39[TK]D-Fendermichael-i, You exclude it by not putting it there
18:08.08michael-i[TK]D-Fender: obviously…but I didn't want to dial n permutations of a single group of n extensions
18:08.37[TK]D-Fendermichael-i, Whatever you put in Dial() is what you're going to get.
18:08.55michael-is/dial/generate
18:09.16michael-i[TK]D-Fender: i seemed like a common problem and I thought there could be a magic flag in the dial app somwhere
18:09.32*** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn)
18:09.42[TK]D-Fendermichael-i, Of course not.  It does what you tell it.
18:09.55[TK]D-Fendermichael-i, Just like ever other app
18:10.36michael-i[TK]D-Fender: yes, and one of the things you could tell it to do is to ignore the calling channel. But this is not the case
18:11.07[TK]D-Fendermichael-i, What app gives you an option to ignore what you filled into another parm?
18:11.29[TK]D-FenderDoSomething(yes,IMeanNo)
18:11.50hudonyNew conf : http://pastebin.com/xrNhR2eZ
18:12.29[TK]D-Fender12:53 <[TK]D-Fender> hudony, add nat=yes to [general] , nat=no to [5142252220] <-------------
18:12.34michael-i[TK]D-Fender: can't think of one, just seemed a common problem. Other solutions as to how "call all other extensions in a call group" would be welcome
18:12.58[TK]D-Fendermichael-i, I don't see how its a problem.  Don't put it in there.  Sounds pretty easy.
18:13.24hudonyoh
18:13.27hudonymy bad..
18:14.02michael-i[TK]D-Fender: I'm generating each group as an extension in the context "groups". So, instead I must generate a specific version of the group for each phone in the system. It's ugly, just that
18:14.07hudonyWow, it works!
18:14.18[TK]D-Fender\o/
18:14.19hudonyDunno why but it does
18:14.33hudonySeriously...you guys are god...
18:14.42[TK]D-Fenderhudony, Because * didn't know it was behind NAT because those other settings only come in when you actually tell it they matter
18:14.58hudonyCan you tell me the nat=no directive to a phone that is behind a nat?
18:15.07hudonyoh
18:15.34[TK]D-Fender<hudony> Can you tell me the nat=no directive to a phone that is behind a nat? <- ... huh?
18:15.36hudonyCause i have the asterisk 1.6 book and it doesn't says you can put nat=yes in general
18:15.39hudonyonly in peer :(
18:16.03p3nguinThe sample config would have told you.
18:16.04[TK]D-FenderJust because it doesn't say you can doesn't mean you can't
18:16.14hudonyI can tell :(
18:16.42hudonyBut what about putting nat=no to my peer (sip phone).  The phone is nat.
18:16.52hudonyI don't understand this part
18:17.10[TK]D-Fenderhudony, * will tell your provider the wrong address if you don't put it in [general]
18:17.16hudonyoh
18:17.24p3nguinIf the phone is behind NAT, but it is the same NAT that Asterisk is behind, use nat=no for that phone.
18:17.26[TK]D-Fender"Where am I", vs "where are they"
18:17.29[TK]D-FenderBoth matter.
18:17.34hudonyok I see
18:17.39hudonyThanks again to all of you!
18:17.47hudonyYou guys are really helpfull
18:17.58[TK]D-Fendermichael-i, 2 lines of dialplan.  Hardly "messy"
18:17.59hudonyHave a good day!
18:21.40*** join/#asterisk nix8n82-phone (~AndChat@71-32-137-67.chyn.qwest.net)
18:34.09*** join/#asterisk ilj (~ilj@sourcemage/grimoire/apprentice/ilj)
18:41.39*** join/#asterisk umay (~chris@67-6-158-37.hlrn.qwest.net)
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18:56.55NaikrovekAnyone here have access to Polycom UCS SIP 4.0?
18:59.05*** part/#asterisk libryder (~david@209.33.214.243)
19:02.19tm1000Naikrovek:  yes
19:02.52Naikrovekhave you use it yet?  I want to try it but I don't want to wait for it to be released publicly
19:03.09NaikrovekIt could be some time before that happens.
19:03.30tm1000Yes. I've used it
19:03.33tm1000its on my 550
19:03.39Naikrovekyour thoughts?
19:03.45tm1000it just looked nicer
19:03.50tm1000everything else is functionally the same
19:04.28tm1000configuration files from 3.3.x work on 4.0 just fine
19:05.19Naikrovekyes, but SIP REASON header is supposed to be supported.  Have you tried that?  Also corp directory no longer requires the productivity license.
19:05.57Naikroveknew Web UI, etc.  tons of stuff added
19:06.04Naikrovekit's that stuff I'm curious about
19:07.49Naikrovekalso, the .tgz links on provisioner.net all go to non-existant topics.
19:07.55Naikrovekfyi.
19:08.24*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
19:08.49tm1000yes I know. considering the tgz are mainly for freepbx anyways and are hosted in a directory for direct access the links are kinda irrelevant
19:09.01tm1000not sure why youd want them on the #asterisk channel
19:09.05tm1000unless you are using freepbx?
19:09.08tm1000Naikrovek:  ^^
19:09.29Naikrovekwell i'm looking for a way to provision phones more intelligently, without using freepbx.
19:09.39Naikrovekguess your solution isn't what i'm looking for.  no biggie.
19:09.40tm1000well you should look at the repo
19:09.44Naikrovekalright.
19:09.47tm1000uhhh?
19:10.03tm1000it's used in freepbx, blue.box, whistle, and independently
19:10.06Naikroveki have perl scripts that i use now but there are limitations and it needs rearchitected anyway.
19:10.09tm1000it surely IS what you are looking for
19:10.18Naikrovek<-- not using freepbx
19:10.30Naikrovekor blue.box or whistle.
19:10.35tm1000Naikrovek:  did you skip over what I am talking about?
19:10.43tm1000It's distro independent
19:10.47tm1000it doesn't need any gui
19:10.56tm1000eg "independently"
19:10.58tm1000:-)
19:11.03p3nguin(1308.47) <tm1000> yes I know. considering the tgz are mainly for freepbx    <--------
19:11.05Naikrovekno, we're circling the drain of this conversation.  we're responding to .. nevermind i'll chekc the repo done
19:11.31tm1000p3nguin: Naikrovek  the tarballs, tgz are for freepbx, the actual project is not
19:11.43p3nguintgz IS a tarball.
19:11.48tm1000omg
19:11.50tm1000yes
19:11.53tm1000read what I am saying
19:11.58tm1000"the actual project is not"
19:11.59*** join/#asterisk xpot-mobile (~james@70-91-210-237-BusName-Utah.hfc.comcastbusiness.net)
19:12.04tm1000" the tarballs, tgz are for freepbx"
19:12.06tm1000p3nguin:  ^^
19:12.08irrootfurball or hairball
19:12.12Naikrovekyou're lagging or something.  lol  you're responding to 2 lines back all the time.
19:12.16p3nguinfurries!
19:12.34Naikrovekanyway you're caught up now.  everyone is on the same page
19:12.39Naikrovekgeepers
19:12.42tm1000p3nguin:  Naikrovek  https://github.com/tm1000/Provisioner
19:12.57*** join/#asterisk Hanumaan (~Hanumaan@dslb-092-074-233-014.pools.arcor-ip.net)
19:13.19Naikrovekyup already there
19:13.59tm1000Naikrovek:  i sent you a pm, check
19:17.42tm1000note to self. fix links on provisioner.net
19:17.49tm1000re-do site
19:18.16*** join/#asterisk beccara (~beccara@mail.ubergroup.co.nz)
19:19.00*** join/#asterisk Pio (~pio@reyes.longstair.com)
19:26.21*** join/#asterisk mjordan (~mjordan@nat/digium/x-azvmcmkpsmrinukc)
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19:28.20*** join/#asterisk nix8n82-phone (~AndChat@71-32-137-67.chyn.qwest.net)
19:29.11*** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176139514.dsl.bell.ca)
19:31.22WIMPyWhat does this mean or what should I watch out for? - ERROR[32019]: lock.c:258 __ast_pthread_mutex_lock: chan_sip.c line 14833 (register_verify): 'peer' really deep reentrancy!
19:31.50WIMPyI have no idea when it happens verbose and debug both at 9 don't show anything.
19:32.02p3nguinOH NOSE!  YOU'VE ENTERED THE ABYSS!
19:32.54*** part/#asterisk mjordan (~mjordan@nat/digium/x-azvmcmkpsmrinukc)
19:33.03Piowhen i use a certain softphone (qutecom) and asterisk transfers a call from hold music to the sip phone, i get a lot of distortion and choppy audio for the first second or two, then it goes away..
19:33.41*** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc)
19:34.04irrootPio this could be jitterbufers and or some form of echo can the phone may support
19:34.37irrootpio or even pc load when a call is made
19:34.37Piowhat i dont understand is, if i have directmedia=no in my sip.conf.. to prevent the softphone from ever talking to anyone but asterisk.. shouldnt that mean that, despite transferring or any other activity on asterisk, its one unbroken audio stream?
19:35.11p3nguinIt's not necessarily unbroken.
19:36.17Pioits weird, ekiga seems to be the only softphone which i have no issues with.. but ekiga's build on ubuntu oneiric is crappy, it segfaults half the time when you try to start it, other issues
19:36.42Piotwinkle and qute both have audio problems, but not the SAME audio problems.. its very odd
19:38.47*** join/#asterisk mjordan (~mjordan@nat/digium/x-azvmcmkpsmrinukc)
19:39.28irrootPio i use blink its quite usfull
19:39.53Piogoogles.. no linux client :(
19:40.06Piooh wit
19:40.09irrooti use it on ubuntu
19:40.16Pioyeah its just not on the front page here.. lets see
19:41.30SeRianybody experience with polycom phones? I am trying to disable a softkey and add a new one but no matter what I do the "Forward" softkey is all ways there
19:41.35Pioi'll try these natty packages
19:41.56SeRibetter said remove one
19:43.01irrootSeRi been a while all the magic in sip.conf is pain to edit
19:43.38SeRiirroot, my fingers are bleeding!
19:43.52SeRiall morning at it and nothing.
19:46.34irrootSeRi <sip><softkeys .....
19:46.48LemensTSSeRi: you are editing the cfg files right
19:48.06SeRiguys here is what I have
19:48.08SeRihttp://pastebin.com/raw.php?i=XigK7V9A
19:48.23SeRisoftkey.cfg gets called from mac.cfg
19:49.02SeRii remove all <softekeys> entries from sip_317.cfg
19:49.15SeRiThis is a polycom 501
19:49.33*** join/#asterisk _Corey_ (~chatzilla@64.121.4.75)
19:49.35SeRiLemensTS, Yes.
19:49.49irrootSeri looks like some spaces got mixed up
19:50.54SeRiirroot, Like what?
19:51.09irrootsoftkey.1.use.active="1"softkey.feature.newcal
19:51.20irrootthey args run into each other
19:51.36irrootthis may be a problem
19:51.37LemensTSSeRi: http://pastebin.com/yXi3RPMV
19:51.55Pioirroot, blink is working very well so far, thanks
19:52.20LemensTSSeRi: that is the config I run on some of mine, should give you some tips
19:52.34[TK]D-FenderSeRi, I think I see it : If you attempted to "export" that section to a separate file you need to recreate the XML tree up to that tage, IE : <sip/> .... etc
19:52.34irrootPio its not perhaps ideal but it seems stable
19:52.45SeRiLemensTS, Thanks. irroot Thanks for the point out.
19:52.50[TK]D-FenderSeRi, Because EFK isn't a base-level element
19:53.24Pioirroot, yeah, no audio problems, less buggy than ekiga, its a net improvement for me :)
19:54.43Piono tray icon? thats kinda weird
19:55.25SeRi[TK]D-Fender, I am not following.... :/ sorry. so I have to move my tag?
19:55.39[TK]D-FenderSeRi, You have to add the oter ones.
19:55.42[TK]D-Fenderouter*
19:57.34SeRi[TK]D-Fender, here is what I was following... some what
19:57.35SeRihttp://wiki.sipfoundry.org/display/sipXecs/Polycom+Phone+Customization
19:58.21[TK]D-FenderSeRi, "This XML snippet must be between the <sip></sip> tags which are at the top and bottom of the configuration file"
19:58.23[TK]D-Fender^^^
19:58.43[TK]D-Fenderadd those around your file
19:58.44SeRiops
19:59.13SeRiThanks for the point out :)
20:00.01irrootSeRi grab a xml tree viewer from the net should help you
20:00.25SeRiI was looking for one for nix and couldnt find one :(
20:00.29SeRislackware here
20:01.23*** join/#asterisk JuanCri (~JuanCri@200.72.190.92)
20:01.49SeRihere is what I got now
20:01.50SeRihttp://pastebin.com/raw.php?i=TuiAgL4B
20:02.32SeRifires up a vm
20:03.38irrootSeRi <sip> must come after <?xml header
20:04.17p3nguinconglomerate, editix, kxmleditor, qxmledit, serna, tx, xmlcopyeditor, xxe
20:04.24p3nguinall XML editors
20:04.28*** join/#asterisk kpettit (~kpettit@99-116-144-138.lightspeed.hstntx.sbcglobal.net)
20:05.47leifmadsenlikes xmlmind
20:05.57leifmadsenwrote multiple Asterisk books using that editor
20:05.58p3nguinaka xxe
20:06.07leifmadsenp3nguin: oh I totally missed that on the end
20:06.57p3nguinLast, but not necessarily least.
20:07.10leifmadsenor first!
20:07.16leifmadsenor best!
20:07.34leifmadsendecisions?! we don't need no stinkin' decisions!
20:07.46SeRilol
20:07.52Pioirroot, with blink, does it not support history when you use your own sip server? the 'history' section is just always grayed out
20:11.41Kattypeeks in
20:14.52SeRiI give up. I am tried.
20:16.08irrootPio no clue :P i just use it on the run and as a test phone
20:16.25irrootchecks Katty out
20:17.46cuscohi
20:18.05cuscoin cdr, how can I have a unique identifier for the whole call? as uniqueid keeps changing... ?
20:18.46p3nguinThere should only be one unique identifier for the channel.
20:19.54cuscowell the channel changes
20:20.04*** join/#asterisk simNIX (~simNIX@156-60.bbned.dsl.internl.net)
20:20.15cuscoit is Local/bla and SIP/bla etc
20:20.17cuscofor the same call
20:20.19p3nguinAs a call comes into asterisk, that channel remains the same until that call ends.
20:20.29cuscohu? no!
20:20.58cuscoas it enter the queue, queue dials SIP/1 then SIP/2 tehn SIP/3
20:21.03cuscoso it creates a new channel
20:21.22cuscoin the same call !
20:21.22p3nguinIf I pick up my phone and dial some extension, my phone brings up a channel and makes the call.  The channel that is created by my phone does not change until I hang up.
20:21.40*** join/#asterisk Netgeeks (~chris@gw1.netgeeks.net)
20:21.49cuscounless that ext I dial, is a local dialplan that goes to a queue first
20:21.57cuscoand it tries several dials
20:22.01cuscountil one is answered
20:22.07p3nguinAny call you make begins on some extension.
20:22.08cuscoso it created and destroyed several channels
20:22.17cuscoright...
20:22.24cuscowell..
20:22.38p3nguinSo if you're talking about someone calling inbound from the PSTN to asterisk, that's just one channel.
20:22.44p3nguinThere is a uniqueid for it.
20:22.52p3nguinIt remains until the caller hangs up.
20:22.56cuscothe uniqueid changes too
20:23.08*** join/#asterisk scubes13 (~scubes13@24.168.196.0)
20:23.22p3nguinThe channel created does not change until that leg of the call is disconnected.
20:23.23cuscofor every dial to the several queuemembers a new id
20:23.52p3nguinI'm tired of repeating myself, so I'll leave you to figure out what I'm saying.
20:24.10*** join/#asterisk simNIX (~simNIX@156-60.bbned.dsl.internl.net)
20:24.31SeRiis building xmlcopyeditor
20:24.39cuscoI (think I ) understand you, but what I'm trying to say is: there are several channels created and destroyed in a whole call
20:25.34p3nguinSet a variable on the inbound leg of the call and see how far you can carry it.
20:25.41*** join/#asterisk grandpapadot (~grandpapa@99.175.248.81)
20:25.41Netgeeksin asterisk 1.6 you could set a variable in a subroutine (gosub) like __channel_wide_variable = monkeys and you could read that variable from outside the subroutine, in asterisk 1.8, this appears to have changed, and you are not able to read the variable as set in the sub from outside the sub.  Is there a way to set a variable inside a sub in ast 1.8 that makes it accessable outside the sub?
20:25.56p3nguinSet(__myUniqueID=12345)
20:26.14grandpapadotHey guys, in 1.8.x, is the "h" extension no longer valid in a Macro?  I'm noticing my calling context's "h" extension being fired ... can't find anything discussing ...
20:26.31cuscop3nguin: im dong that, im taking the first uniqueid and keeping it
20:26.42cuscoproblem is i cannot set __CDR(UID)=bla
20:26.44p3nguinh shouldn't normally be run in a macro.
20:26.59p3nguinWhy can't you set it?
20:27.08leifmadsen'h' has never been valid in a Macro()
20:27.11[TK]D-FenderNetgeeks, * dialplan has no sense of scop from Gosub <-
20:27.11cusco__CDR(UID) CDR(UID) cannot be read after
20:27.17cuscoonly CDR(UID)
20:27.25cuscoI tried, but CDR is a function, right=
20:27.28cusconot a var?
20:27.29[TK]D-FenderNetgeeks, This isn't a high-level language.  all vars are global to the channel;
20:27.30grandpapadot@leifmadsen - tnx
20:27.40p3nguinCorrect, the CDR function is a function.
20:28.02cuscoso I cannot set __CDR() only __VAR
20:28.09leifmadsenNetgeeks: ya, GoSub() allows you to read it anywhere -- there is nothing stopping you from assigning a variable outside a GoSub() and reading from the GoSub(), and vice-versa (unless you use LOCAL())
20:28.15QwellCDR() is a function..
20:28.17leifmadsencusco: right, that is not valid
20:28.31leifmadsencusco: set a channel variable with the data, then set it on the other channel with the CDR() function
20:28.33cuscobut I would like CDR(UID) to be inherithed too
20:28.34p3nguinBut you can set a __variable and then set CDR() to the value of the variable.
20:28.59leifmadsencusco: you may need to look at using something like U() or M() flags to Dial()
20:29.08Netgeeks[TK]D-Fender:  well, i just upgraded an asterisk with a diaplan that is doing what I said.  in 1.6 it works just fine, I set a variable __dialed_exten=1234566 and the next line in the code after the return is a verbose(1,${dialed_exten}) .   when I load the same diaplan in 1.8, the variable comes up empty in the verbose statement
20:29.09leifmadsenit'll execute dialplan on the called channel
20:29.32[TK]D-Fender<PROTECTED>
20:29.36leifmadsenNetgeeks: there is no reason that wouldn't work as nothing should be changed
20:29.39[TK]D-Fenderthat's part of the problem.
20:29.44[TK]D-FenderSet like normal, use like normal
20:29.49leifmadsen^^ that
20:29.50cuscook thanks, I'll look that up
20:29.57Netgeeksroger
20:30.13p3nguinWhen the call comes in, Set(__myUniqueID=${UNIQUEID}) ... and then on other channels, Set(CDR(UID)=${myUniqueID}).
20:30.48cuscoI actually tried that... lol I need to review it
20:30.51cuscothanks
20:31.51Netgeeks[TK]D-Fender so you are saying don't use Set(__dialed_exten=${<some number string>}), use Set(dialed_exten=${....})?
20:32.05[TK]D-Fenderyes
20:32.26Netgeeksor in reality, in my case the real code is an ARRAY() but I assume the same for it
20:32.27[TK]D-FenderNetgeeks, In a call vars aren't local to a context
20:32.49Netgeeksbut if I don't use __ before the var, it isn't passed to child channels, correct?
20:33.09[TK]D-Fenderchannel inheretence is another matter
20:33.30p3nguinIf you use one underscore, it will be inherited by one newly created channel off the main channel.
20:34.10*** join/#asterisk michael-i (~anonymous@204.11.230.58.static.etheric.net)
20:34.23Netgeeksso, in 1.8 if I want to set a variable in a gosub, and have that variable read anywhere in the dialplan by that channel, and also have it inherited by subsequent channels, what form of set statement would I use?
20:34.36leifmadsenthe same as any other place
20:34.40leifmadsenGoSub() is not magical
20:34.43Netgeeksbut it's not working leif
20:34.45leifmadsenit is just a Goto() with memory
20:34.47NetgeeksI just tested
20:34.55leifmadsenthen provide enough information showing the error and ability to reproduce
20:35.38Netgeekskk, I'll be back with a simplifed dialplan that reflects this, or I'll be back with an embarrasing oops, I see what I did wrong
20:37.08*** join/#asterisk brdude (~brdude@12.155.183.30)
20:39.01leifmadsenNetgeeks: I suspect a typo
20:39.28p3nguinor failure to "dialplan reload" after making minor changes.
20:39.56Netgeeksleif, I wish, the thing that bothers me is that 1.6 the dialplan works just fine, 1.8 it doesn't and I can at this point state that the dialplans are identical (copied & diffed for extra sureness)
20:40.13p3nguinShow us.
20:40.20leifmadsenNetgeeks: works fine here:   http://pastebin.com/DhzaBzVs
20:40.24p3nguinI moved my 1.4 dial plan to my 1.8 system, and it didn't break.
20:40.28leifmadsenNetgeeks: I just tested on Asterisk 10
20:40.41p3nguinI had a few warnings about syntax change, but I don't remember anything breaking.
20:43.17p3nguinIf I have a complaint about choppy sounding audio on one side of a call, should I concentrate on bandwidth/network usage or is there any chance that an underpowered asterisk computer could cause it?  CPU usage remains very low, and there is no transcoding (using ulaw on both legs of the call).
20:43.35*** join/#asterisk lcat (~lcat@187.45.254.21)
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20:43.56*** part/#asterisk jplank (~G_Bove@208-104-67-27.dyn.fttp.comporium.net)
20:44.55leifmadsenp3nguin: sounds like a network issue to me if CPU isn't the problem, especially with no transcoding. What about I/O for recording calls? Typically though my experience with choppy audio nowadays stems from either a provider problem, or a network issue or some sort (i.e. I had a problem where my ITSP was getting DDOS'd -- that caused audio issues :))
20:45.02*** join/#asterisk jplank (~G_Bove@208-104-67-27.dyn.fttp.comporium.net)
20:46.31p3nguinI'll have to look into the I/O while MixMonitor() is recording the call.  When I review the call, I hear crystal clear audio on both sides, which makes me think the loss is between my upstream side of my CPE and my ITSP, rather than between me and the phone.
20:47.12*** join/#asterisk lhfnet (~lhfnet@net-cdd-fw01.cddlasmercedes.com)
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20:48.13p3nguinMy initial thought was that it could have been due to some torrent traffic... since it was during the weekend not during business hours.
20:48.17lhfnetHi, I am having this message continuously in the CLI chan_sip.c:6343 sip_write: Asked to transmit frame type alaw, while native formats is 0x4 (ulaw) read/write = 0x4 (ulaw)/0x4 (ulaw), I set disallow = all allow=ulaw in all my users in users.conf and in the sip.conf and iax.conf. Using Asterisk 1.8.7.1
20:48.43*** part/#asterisk jplank (~G_Bove@208-104-67-27.dyn.fttp.comporium.net)
20:48.49p3nguinI obviously can't make that determination after the fact, but I can check it next time when I'm told as it happens.
20:49.08vader--hmmm, i need to come up with a recommendation for a guy
20:49.26p3nguincan recommend a guy for vader--
20:49.43vader--he has 3 pots lines for his little store... right now he has a samsung prostar 816 plus pbx that isn't hooked up
20:49.48vader--he has the pbx and all the phones
20:50.43Netgeeksaha!  I reproduced it in a simplified dialplan
20:50.56Netgeeksit's broken when using ARRAY() to set the vars
20:51.14vader--so im not sure if i should recommend he stay with the prostar systema nd ill try and hook it up for him
20:51.27vader--or build some sort of small freepbx system, or myabe a talkswitch system or something
20:54.34Netgeekshttp://pastebin.com/uRHYTsQv
20:57.08Netgeekshrm, ignore that pastebin, it's author is braindead at the moment
21:00.06Netgeeks[TK]D-Fender  so what I was doing wrong is that the original author of this dialplan wrote an array assignment like Set(ARRAY(var1,var2)=value1\,value2)
21:00.36[TK]D-FenderNetgeeks, Well, I can't speak for hte use of ARRAY as I've never touched it... but OK :)
21:00.47Netgeekswell, in 1.6 that works just fine, in 1.8, the / escapes the comma, and .... all the values get put in var1
21:00.52Netgeeksand I was looking at var2
21:01.09SeRiLemensTS, I use your code with an xmleditor and the phone still does not change.
21:01.10Netgeekswell, var2 is now empty, because array thought it only had one long string
21:01.54Netgeeksso I just need to go remove all the / from in front of commas in his array statements
21:07.33*** join/#asterisk LittleFool (~LittleFoo@95.129.212.120)
21:10.52leifmadsenNetgeeks: oh ya, you don't need to escape things anymore
21:11.27NetgeeksI'm quite happy right now, this was nearly the easiest fix I've ever had to do
21:11.27leifmadsenthe dialplan is smarter about that now
21:12.07leifmadsenSet() is not longer a multi-Set() application (which is why commas needed to be escaped before). Even easier might have been s/Set/MSet/g :)
21:12.32leifmadsenthen you probably could have left the escape character there
21:12.42leifmadsen(what you did is better though)
21:12.58Netgeekssearching and replacing \, with , was simple enough
21:13.01michael-i[TK]D-Fender: does this look elegant enough to filter group members from group calls? Set(FILTERED_DIAL_CHANNELS=${STRREPLACE(${DIAL_CHANNELS},SIP/${CONTEXT})})
21:14.07michael-iIt keeps on returning an empty string…so I'm wondering if I can even use a variable like that in STRREPLACE
21:15.00[TK]D-Fendermichael-i, you just issues a replace with 2 parms.  Doesn't look kosher.  Perhaps you should reread each function's instructions.
21:15.48michael-i[TK]D-Fender: replace-with is optional
21:16.11[TK]D-Fendermichael-i, Well I don't ee what those vars start with ....
21:17.22*** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com)
21:17.49[TK]D-Fendermichael-i, I don't see that function in 1.6.2 ... that 1.8 new?
21:17.59michael-i[TK]D-Fender: 10
21:18.05*** join/#asterisk navaismo (~navaismo@187.170.0.233)
21:19.52[TK]D-Fendermichael-i, PB the function instructions and your complete code
21:20.44*** join/#asterisk garymc (~chatzilla@host86-176-88-100.range86-176.btcentralplus.com)
21:21.35vader--any recommendations on a 8-16 port cheap switch with poe?
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21:24.10leifmadsenmichael-i: you don't use ${DIAL_CHANNELS}  -- just use DIAL_CHANNELS
21:24.23leifmadsenmichael-i: you give it the name of the variable to work with, not the value of the variable
21:24.27[TK]D-Fenderleifmadsen, that's what I was suspecting from similar functions....
21:25.24leifmadsenmichael-i: Set(thisResult=${STRREPLACE(DIAL_CHANNELS,SIP/${CONTEXT},abc,CBA)})
21:27.36michael-ileifmadsen: thanks!
21:28.14michael-iI get a bit lost in escaping, etc since I'm writing code which generates this dialplan logic
21:30.44leifmadsenmichael-i: yuch :)
21:31.41garymcin asterisk cli what command shows me the sip phones trying to connect and failing?
21:31.58garymcive tried "sip set debug" but not working
21:32.05garymcis that command out dated now?
21:37.05michael-ileifmadsen: moving the logic into a one-time generation is the only way I can cut down on resource requirements :) it's an art unto itself sometimes
21:37.27leifmadsenmichael-i: I've gone the route of writing code to write code, and it always ended up in misery for me
21:37.36leifmadsengarymc: sip set debug on
21:37.45leifmadsengarymc: tab completion ftw
21:37.59garymcthanks
21:38.29garymcsip set debug on = no such command
21:41.02garymcleifmadsen: what is tab comletion ftw?
21:41.10garymc*completion
21:41.16leifmadsensip set<tab>
21:41.26leifmadsengarymc: what version of asterisk? is chan_sip.so loaded?
21:41.38garymcnot sure 1.6
21:41.41leifmadsensip<tab
21:41.46leifmadsenwill show you optoins
21:41.53leifmadsencore show version
21:42.09garymc1.6.2.11
21:42.50garymcno joy with any of those commands
21:45.08navaismotry with module reload chan_sip.so
21:45.31*** part/#asterisk sysreq (~sysreq@unaffiliated/sysreq)
21:45.44garymchow do i do that?
21:45.52navaismomodule reload chan_sip.so
21:46.46garymci did that nothing happened
21:47.11navaismoerror warning something?
21:47.13irrootgarymc it could be its snarled up
21:47.20irrootcore show locks ??
21:47.33irrootit may need a reastart
21:47.39leifmadsenya sounds like a deadlock
21:48.38garymccore show locks = nothing
21:49.06leifmadsenmay not be enabled in menuselect
21:49.09navaismostop asterisk, then asterisk -vvvvvcg and check if the module load
21:51.59garymcyeah its working now
21:52.04navaismoleifmadsen: which option in menuselect?
21:52.42leifmadsennavaismo: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
21:52.52navaismothx
21:53.02leifmadsennavaismo: see section "Getting Information for a Deadlock"
21:53.27irrootDHCP server gone bad <- horror movie for next halloween i haz one ATM
21:53.38garymcanyone know why my softphone is failing to register http://pastebin.com/Uk4UHqrz
21:53.43navaismothx leifmadsen
21:56.47navaismogarymic i see some Got SIP response 405 "Method Not Allowed" and SIP/2.0 403 Forbidden (Bad auth)
21:56.48navaismobrb
21:58.21garymcdo i have to create an rtp.conf file or is there already one
22:00.48garymcgetting this in asterisk cli now:
22:00.50garymcGot SIP response 405 "Method Not Allowed" back from 81.1
22:11.56*** join/#asterisk ruied (~ruied@po-217-129-154-119.netvisao.pt)
22:12.54ruiedhello, how can I do a "case" like function in asterirsk?
22:13.48ruiedis there any  "case"  function?
22:14.27Qwellseveral if checks
22:14.32navaismogotoif or agi
22:14.36navaismotoo late
22:14.39Qwellor gotoif, yeah
22:14.45Qwellor just goto, for that matter
22:16.26ruiedok, thanks :)
22:18.38*** join/#asterisk tzafrir (~tzafrir@212.179.75.202)
22:20.09*** join/#asterisk irroot (~irroot@197.172.248.230)
22:21.08*** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek)
22:22.41*** join/#asterisk saisoma (~saisoma@client72.jdcc.edu)
22:22.43ruiedI think goto is best for what I need, I'm  making a macro with blf custom lamps. Press once to redirect incoming calls to my store (green light buton), press again, incoming calls to my house (red button), press again to redirect to my mobile (red blinking)...
22:23.40ruiedand when I press the button it will say a voice of what it is doing ex: "Retirecting calls to Mobile"...
22:23.43*** join/#asterisk timeshell (~chatzilla@gw.lusi.on.ca)
22:23.47timeshellleifmadsen ping
22:23.52leifmadseno/
22:23.55timeshellOla
22:23.59timeshellWhere do I find LOG_DEBUG
22:24.05leifmadsenwhat do you mean?
22:24.18timeshellPer dumphistory = yes|no : Enable support for dumping of SIP conversation's transaction history to LOG_DEBUG. Default no. (New in v1.2.x)
22:24.26leifmadsenlogger.conf
22:24.27*** join/#asterisk Greenlight (~Wullie@cpc2-dund11-2-0-cust994.sgyl.cable.virginmedia.com)
22:24.53leifmadsenlog to a file with debug level logging, and 'core set debug 10'
22:26.20GreenlightEvening folks. I've an issue with my voip provider (voiptalk). At certain perioids during the day their servers stop replying to my SIP Invites in a timely fashion (sometimes no replies at all, sometimes takes 6 retries). This is causing delays of 20+ seconds for calls to start ringing, or sometimes causing complete timeout. Has anyone else ever had issues like this, and if so what was the
22:26.21Greenlightsolution?
22:26.52Qwellsolution: fire them
22:27.03QwellYou've already called and complained, right?
22:27.07wdoekes2ruied: Goto(s-${value),1) and then have exten s-CASE1..., s-CASE2...
22:27.16GreenlightIndeed - they've had 3 days to look over the traces now
22:27.40GreenlightThey asked me to increase the T1 timeout up to 1500ms (the SIP RFC states it should be 500ms)
22:27.45GreenlightIt didn't help ;/
22:28.03*** join/#asterisk KavanS (~KavanS@LINBIT/KavanS)
22:28.20GreenlightI'm leaning towards binning them, only thing is they have a really nice relseller system which we've got a bunch of our customers using
22:29.08ruiedwdoekes2, that seems to be the best structure
22:30.02GreenlightAnyone recommend decent SIP provider in the UK, who ideally offer reseller/wholesale options. Only a 3 or so million minutes a month at present.
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22:30.56*** mode/#asterisk [+o mnicholson] by ChanServ
22:31.08saisomahey guys, having an issue with voicemail and a branch office: http://pastebin.com/1zZfMmZ9  any assistance is greatly appreciated
22:31.59*** join/#asterisk mnicholson (~mnicholso@nat/digium/x-jlhmqhwkgixbgnss)
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22:32.24timeshellleifmadsen Should I strip out personal IP's and such from the file?  Or can I submit it somewhere where it will remain confidential?
22:33.23*** join/#asterisk nunne (~nunne@static-213-115-116-75.sme.bredbandsbolaget.se)
22:34.42GreenlightWhile I'm in here - Where would be the best place to go to find Asterisk consultants?
22:39.00leifmadsentimeshell: strip it because making it confidential greatly reduces the number of eyes who can look at it
22:39.10leifmadsenGreenlight: asterisk-biz mailing list I guess
22:39.14timeshellk
22:39.27GreenlightOkay, thanks
22:39.45_Corey_Greenlight: You can search online or put in a request with Digium's website for a referral too
22:42.15GreenlightI've had a brief look online, but wasn't too sure. I might try the Digium website. Am looking for a person/company who we can have investigate any issues with our Asterisk systems that we dont have the time or resources to deal with internally. We were thinking of recruiting someone for the post but I think our budget might be better spent on a reliable consultant who we can use as needed
22:42.53_Corey_There are a number of dCAPs that can be found on LinkedIn
22:43.03_Corey_(search for the dCAP group)
22:43.04SeRiGreenlight, talk to p3nguin
22:43.26Greenlight_Corey_: Cool will check out LinkedIn, ty
22:43.38GreenlightSeRi: Is he UK based?
22:43.45SeRiops. no sorry
22:43.58GreenlightUS ?
22:44.03SeRiYes
22:44.21*** join/#asterisk jrose_atDigium (~jon@nat/digium/x-wnprsiohpjmqjzfs)
22:44.29GreenlightMight still work actually - guess it may even be helpful if "out of hours" here is during working day there
22:45.01_Corey_Greenlight: My recommendation in the UK would be David Duffet, just google him
22:45.23GreenlightOk, that's perfect, Thanks again, I shall
22:46.40_Corey_No problem
22:46.57Greenlight_Corey_: Just wondering if you've worked with him, or just know of him?
22:49.58_Corey_We've both spoken at Astricon for the last few years, so I've known him at least that long.  He runs on the of the best Digium Resellers in Europe
22:50.25_Corey_Really good guy though, if I were in the UK and needed someone he'd be my first call
22:50.45GreenlightSounds like a solid endorsement! :)
22:51.07GreenlightWill be getting in touch with him I think, thanks again, it's appriciated
22:51.27_Corey_I'm not sure what sort of engagement terms his shop offers...  we mostly do maintenance contracts here
22:51.33_Corey_Best of luck though
22:51.54GreenlightHopefully he can offer something that is a fit with out needs
22:51.59Greenlight*our
22:53.08leifmadsenDavid Duffett is awesome, if you need someone in Europe, ya, call him
22:53.38GreenlightHe certainly sounds like the right chap
22:54.36*** join/#asterisk tzafrir (~tzafrir@bzq-218-155-145.cablep.bezeqint.net)
22:57.52GreenlightAnyone experience using AQL for voip termination in the UK ?
23:01.31vader--Whats a good phone for a small business? Cost effective but not crippled?
23:01.55GreenlightAs in hard phone?
23:01.59vader--ya
23:02.03vader--IP phone
23:02.09GreenlightDepends what features you're needing
23:02.28*** join/#asterisk mjordan (~mjordan@nat/digium/x-catefqwfgqdqvpfj)
23:02.36vader--hold mainly
23:02.36vader--hehe
23:02.41GreenlightWe use snom300's and never had any problems
23:02.51GreenlightI'd guess they're somewhat midrange
23:03.20vader--PoE?
23:03.40GreenlightYea - they also come with AC adaptors though
23:04.34vader--can you intercom with them?
23:05.05GreenlightNever tried to be honest, but I'd imagine so. Most hard phones generally support the required sip header
23:05.51vader--i guess the biggest thing this guy would want to be able to do is put a call on hold and pick up the call from another phone
23:06.17GreenlightWell that's more parking than hold
23:06.34GreenlightSo you'd just use a featurecode on Asterisk to handle that
23:07.40Greenlighthttp://downloads.snom.net/documentation/data_snom300_en.pdf
23:08.07GreenlightWorks great, and doesn't feel "cheap"
23:10.11vader--how are the cisco SPA phones?
23:10.42*** join/#asterisk neurosys_ (~neurosys@c-174-48-142-160.hsd1.fl.comcast.net)
23:16.57*** part/#asterisk mjordan (~mjordan@nat/digium/x-catefqwfgqdqvpfj)
23:22.57SeRi[TK]D-Fender LemensTS I got it!!!!!!!!!!!!!!! :D
23:23.08SeRiso much blood spilled....
23:23.13SeRibut I got it :D
23:26.23paulcLooking for a recommendation on SIP DECT phones.. Panasonic vs Gigaset S675 IP.. leaning towards the latter, perhaps.. anyone got any thumbs up or down for either?
23:45.15*** join/#asterisk ketas-ts (~ketas@82.131.20.5.cable.starman.ee)
23:45.16ruiedwdoekes2,  done!   :)
23:54.27*** join/#asterisk michael-i (~anonymous@204.11.230.58.static.etheric.net)
23:54.59michael-iIs there a way to jump directly into VoiceMailMain() to record a new greeting?
23:57.08ChannelZwonders if there is a shortcut as part of MiniVM
23:57.41michael-iDo they use the same directory structure?
23:59.30ChannelZNot positive.
23:59.59ChannelZYou could probably "roll your own" easily enough by constructing the path to the greeting and just using Record

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