irclog2html for #asterisk on 20070322

00:01.02aldoenviroI guess this is a good first question...  will asterisk work with a normal voice modem?
00:03.57mvanbaaknormally, no
00:04.25mvanbaakif it's a X100P clone model, it will work
00:04.37mvanbaakbut my experience is, most voice modems wont work
00:06.12aldoenvirowell that defeats the whole purpose.  I wanted to test it out using incoming calls on a voicemodem.  Run the calls through a menu system, voicemail etc...
00:08.19*** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net)
00:09.03*** join/#asterisk mroberto (n=dads@S0106001346face5f.ed.shawcable.net)
00:09.37mrobertoI need some help with ceptral i have the module install just not sure how to use it ? I am using version for 2.4
00:10.25Voice2Uniqueid has a dot.. think i can force it not with a dot ?
00:10.45Voice2that UNIXTIMESTAMP + random 0000-9999 ?
00:12.19aldoenviroanyone know any products that would allow incoming calls on a voice modem to be answered by an auto attendant with voicemail?
00:15.06aldoenviroSo an X100P FXO PCI Card
00:15.18aldoenviroThat would do the trick for asterisk?
00:16.35mvanbaakif you are talking analog lines, yeah
00:17.06*** join/#asterisk anthonyl (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net)
00:17.09JTif you didn't want a very good card, then maybe
00:17.48aldoenviroWhat is a decent "hobyist" card in that case?
00:18.31JTthere's no such thing as a card that is both cheap and decent
00:18.38JTan ATA might be a better plan
00:20.22aldoenviroCheap isn't necessarily a requirement...  just a desire
00:21.09aldoenviroATA?
00:21.13Voice2ok wahts the alternatve to uniqueid ? i dont want a dot
00:21.13Qwell~ata
00:21.15jbotata is probably Analog Telephone Adapter which is used to put a normal analog phone onto ethernet, see http://www.voip-info.org/tiki-index.php?page=Analog%20Telephone%20Adapters for more info
00:22.22`p4r14haldoenviro: get a tdm400p before you start buying x100p's
00:23.30`p4r14hthat is if you plan on using it for more than a hobby
00:24.15JTi prefer to do my "hobby" stuff properly too :)
00:24.36aldoenviroin all actuality, I would like it to answer my phone...  I might eventually like to have a wifi cordless...
00:24.49`p4r14hwell, if you just want to learn it and not use it heavily the x100p works just fine if you only need one PTSN interface
00:24.54JTwifi ip phones, don't go there
00:25.23aldoenviroI played with a cisco phone a few months ago...  it was pretty cool
00:25.33ChrisHardieJT/File/anyone: I'm still stuck with getting asterisk to build/use a chan_zap driver.
00:25.58*** join/#asterisk Meaty` (n=meaty3@office.abi.ca)
00:25.58ChrisHardieI installed the zaptel driver from SVN and got it to recognize the card
00:26.19ChrisHardiemake menuselect recognizes chan_zap as an option and has it selected
00:26.28ChrisHardiebut still no chan_zap.so being built.
00:28.02ChrisHardieIf anyone has any advice/pointers, that'd be great.
00:28.12Strom_Mwhich versions of everything are you running?
00:28.17Strom_MChrisHardie:
00:28.27aldoenviroWould this do the trick? http://www.newegg.com/Product/Product.aspx?Item=N82E16833203012
00:28.36ChrisHardieAsterisk 1.4.2, and I was using zaptel 1.4.0 but was told to use the version from SVN instead.
00:28.56*** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-97-91.ph.ph.cox.net)
00:28.57Strom_MChrisHardie: so you got zapte;-1.4 svn branch?
00:29.00ChrisHardieyes
00:29.04Qwellaldoenviro: very, very low rating.  avoid
00:29.07Strom_Mor did you get zaptel trunk?
00:29.18ChrisHardieI had problems with getting modprobe to find it so I had to copy the .ko files to the right kernel directory,
00:29.35ChrisHardieI used " svn co http://svn.digium.com/svn/zaptel/branches/1.4 zaptel"
00:29.58tzafrirChrisHardie, this may be a bad sign. It may mean you were building with an incorrect configuration
00:30.08christois it possible to pass a call through an AGI before it's answered?
00:30.08Strom_MChrisHardie: ok...if you'll wait for a few minutes, i'm building the latest 1.4 brahcn of everything right now
00:30.19Strom_Mer
00:30.21Strom_Mbranch
00:30.25Strom_Mgod, i'm going dyslexic :)
00:30.30tzafrirAnd thus the modules will refuse to load (if this is indeed the case)
00:30.33ChrisHardieI don't doubt that something is flubbed up.  I just didn't see how doing "make / make install" in each one was going to steer me wrong.
00:31.06cr4z3dhey i'm completely new to asterisk and i've been trying all day to get my VoIP system up and running.. I'm having some trouble figuring out what some of the debug stuff means
00:31.09cr4z3dcan anyone help me out with that?
00:31.22aldoenviroI am reading that the zaptel drivers will work with my modem...  any success on this?
00:31.35ChrisHardieStrom/tzafrirI got the modules to load once they were in the right directory.  ztcfg shows 4 channels configured for the 4 ports on the card.
00:31.41tzafrirwhich modem do you have?
00:31.55tzafrirChrisHardie, does ztcfg give an error?
00:32.12ChrisHardienope, and when run with -vvvv it shows 4 channels configured.
00:34.06tzafrirWhat is your kernel version?
00:34.20ChrisHardie2.6.12-9-386
00:34.52tzafrirAnyway, building asterisk actually has nothing to do with the loaded zaptel. It has to do with /usr/include/zaptel/zaptel.conf (in 1.4)
00:34.57aldoenviroCommunication controller: Intel Corporation 536EP Data Fax Modem    I thought it was a 537...
00:35.16tzafrirChrisHardie, Do you have /etc/asterisk/zaptel/zaptel.conf ? Is it from today?
00:35.23aldoenviroCommunication controller: Rockwell International HCF 56k Data/Fax/Voice/Spkp (w/Handset) Modem (rev 01)  one of these too
00:35.52ChrisHardietzafrir: it's /etc/zaptel.conf now, but yes it's from today
00:35.57aldoenviroel cheapo modems that have been hanging around in a box for ages
00:36.55tzafriroops, my typo: /usr/include/zaptel/zaptel.h
00:37.10ChrisHardietzafrir: yes, have that too
00:37.12tzafrirI managed to make a mess of three different files
00:37.38*** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il)
00:37.49tzafrirChrisHardie, your problem is with building asterisk, right? How exactly do you see that something is wrong?
00:37.49*** join/#asterisk MaartenB (n=Maarten@84-105-197-100.cable.quicknet.nl)
00:38.10MaartenBhello everyone, can somebody please help me, I can not access my asterisk anymore with -r
00:38.19ChrisHardietzafrir: When I try to make a call incoming or outgoing using the digium card, I get an error about the zap channel being unavailable
00:38.24DovidMaeertenB: please explain
00:38.27MaartenBI got asterisk running as user asterisk now, but it says it is not running, while it is :(
00:38.33ChrisHardieI looked in the asterisk build directory in /channels, and it's not building chan_zap
00:38.39tzafrirMaartenB, is there an asterisk process running (ps aux | grep asterisk  # or whatever)
00:38.45DovidMaeertenB: u may need to connect as that user.
00:38.53ChrisHardieIn menuselect, the chan_zap option is available and selected.
00:38.55tzafrirrun: make menuconfig
00:39.00MaartenBtzafrir, yes, there is, it is working too, only asterisk -r is failing
00:39.18MaartenBit says " Unable to connect to remote asterisk (does /var/run/asterisk.ctl exists?)
00:39.27DovidMaeertenB: I personally start asterisk in a screen session.
00:39.30tzafrirChrisHardie, look in channels . Is chan_zap X-ed out? deselected?
00:39.40ChrisHardiein menuconfig under "Channel Drivers", "chan_zap" is NOT X-ed out, and is selected.
00:39.40MaartenBit is started with the /etc/init.d/asterisk script
00:39.57DovidMaeertenB: I had that issue when I was trying to connect to it as a diffrent user than the one that it is running on
00:40.00Dovid(non root)
00:40.16MaartenBhow did you solve it?
00:40.23tzafrirMaartenB, Asterisk either failed writing to /var/run/asterisk.ctl or wrote to another file (/var/run/asterisk/asterisk.ctl ?)
00:40.23Dovidran asterisk as root
00:40.36Dovidtzafrir knows more - follow him
00:40.39tzafrirdo you have /var/run/asterisk ?
00:40.56MaartenBtzafrir, yes, it exists, is owned by asterisk
00:41.01MaartenBtzafrir, empty... :(
00:41.05ChrisHardietzafrir: I'm considering just trying to go back to the 1.2 series for now in hopes it would work better.
00:41.46tzafrirChrisHardie, I asked you a simple question. Please go back to the asterisk build dir, run 'make menuselect' and answer it
00:42.12ChrisHardieI did answer it: "in menuconfig under "Channel Drivers", "chan_zap" is NOT X-ed out, and is selected."
00:42.15tzafrirMaartenB, how exactly do you start asterisk?
00:42.28MaartenBtzafrir, service asterisk start
00:43.16tzafrirChrisHardie, have you tried re-running 'make' ?
00:43.27ChrisHardieYes, several times, after "make clean"
00:44.01tzafrirAre there any local modifications to channels/Makefile?
00:44.02christoHi all - I am dialling out and trying to run the call through an agi script as soon as it's placed (ie before it is answered). Is this possible? My agi isn't completing until I pick up.
00:44.16ChrisHardietzafrir: I haven't made any, no.
00:44.22Strom_MChrisHardie: does the word "zap" auto tab complete at the asterisk CLI?
00:44.54tzafrirwe're in the build phase now
00:45.53ChrisHardieIf someone's available for hire to help address this, I would be happy to negotiate something quickly.  3 hours of having our system down hasn't been pleasant. :)
00:46.52*** join/#asterisk bkuhn (n=bkuhn@fsf/member/bkuhn/bkuhn)
00:47.13*** join/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net)
00:51.09christoline 4 in this script implies that it's possible to do what I'm trying - is this correct? http://www.oldskoolphreak.com/tfiles/voip/mysql_call_logger.agi
00:51.55cr4z3dthe person you are trying to reach is currently unreachable, please try again later
00:52.09cr4z3dis that a message setup by default on asterisk?
00:54.46cr4z3di have iax2 debug enabled and i keep getting a bunch of tx-fram retry and rx-fram retry
00:54.55cr4z3ddoes that mean it's not autheticating properly?
00:56.05*** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com)
01:00.10christocr4z3d - I think it would be more obvious at the CLI if your auth was failing
01:00.27cr4z3di was at the CLI
01:01.05cr4z3dthis is the exact messages: http://forums.digium.com/viewtopic.php?t=14469
01:01.42cr4z3di'm using version 1.12 (ubuntu repository only has that one)
01:03.58spanglesontoastI don't think my sip is registering
01:04.02*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
01:04.48*** join/#asterisk Dovid[Laptop] (n=Dovid@l192-117-114-1.broadband.actcom.net.il)
01:12.24cr4z3d*CLI> iax2 shoMar 21 18:12:00 WARNING[15991]: res_musiconhold.c:421 spawn_mp3: Found no files in '/usr/share/asterisk/mohmp3'
01:12.24cr4z3dMar 21 18:12:00 WARNING[15991]: res_musiconhold.c:493 monmp3thread: Unable to spawn mp3player
01:12.28cr4z3dwhy does that randomly come up
01:16.10spanglesontoastmissing the music on hold files
01:16.45spanglesontoastthat's why
01:16.54cr4z3dyeah but why are they even trying to go
01:16.58cr4z3di don't have that setup
01:17.06spanglesontoastno idea
01:17.10spanglesontoastjust likes to miss them ;0
01:17.17cr4z3di can't even seem to get my asterisk to connect to nufone
01:17.31spanglesontoastI can't get asterisk to connect to anything lol
01:17.36cr4z3dyeah me either haha
01:18.33spanglesontoastand I've dmzed this sucker
01:20.36*** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl)
01:24.28cr4z3dyeah asterisk doesn't seem like the kind of program i can just jump right into and have it working in 5 minutes
01:24.35*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
01:26.18[TK]D-Fender~osmosis~sipnat
01:26.28[TK]D-Fender~sipnat
01:26.31jbotextra, extra, read all about it, sipnat is for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
01:26.31spanglesontoastyea been messing with nat
01:26.31spanglesontoast;)
01:26.31spanglesontoastaswell
01:26.54bullejust forward 5060 and rtp ports to the asterisk box behind nat, and enter the routers public ip in the sip.conf file, as the comments tell you to
01:27.09bulleand it will work, its not harder then that
01:27.21spanglesontoasthmm
01:27.43bulleyou most likely want to narrow the default rtp port ranges a bit, as they are from 10 000 to 20 000
01:27.44cr4z3dis it normal when i type reload chan_iax2.so in the CLI for it to just stop after Registered IAX2 to blah blah
01:28.51bullestart from say 16384 and add as many ports as you might need, if unsure, and for just personal use, add say 100 ports
01:29.03spanglesontoastbut the thing is dmzed :|
01:29.31spanglesontoasttheres no firewall between them
01:29.47spanglesontoastI was wondering where I could see if it's registering or not
01:29.47cr4z3dTx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX     Subclass: ACK
01:29.57cr4z3ddoes that mean i've autheticated with the iax2 server
01:30.04bullespanglesontoast: the asterisk box wont have your public ip, even if its dmzed
01:30.20bullespanglesontoast: so you still need to set the public ip of the router, in the sip.conf file, as per documentation
01:30.33bullespanglesontoast: ifconfig eth0 tells you what ip number ?
01:31.07spanglesontoastit's 192.168.1.3 I set them manually ;)
01:31.15bullesee
01:31.19bulleso it cant possibly work that way
01:31.45spanglesontoastbindaddr is bound to that and extern is to my outside ip
01:32.40bulleexternip=real ip
01:32.45bulleand then a localnet definition ?
01:33.11spanglesontoastsubnet you mean ?
01:33.12lokkju_wrkquestion: would anyone be interested in a script you could point at some text files (configurable) that would read off the per minute rate for the outbound number you are dialing, and then ask you to confirm that you would like to make the call - perhaps also allow you to set it so it only activates when the rate/min is over a specified amount? (or, does someone know of a script that already does this?)
01:33.25bullespanglesontoast: no, localnet most likely, in your sip.conf, if memory servers me right
01:33.37bullelocalnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
01:33.48spanglesontoastlocalnet = 192.168.1.0/255.255.255.255
01:34.15Qwell255.255.255.255?
01:34.29spanglesontoastyea
01:34.30JThmm yeah
01:34.34JTspanglesontoast: that's wrong
01:34.37Qwellumm, why?
01:34.40spanglesontoaster why ?
01:35.07JT<PROTECTED>
01:35.09bullespanglesontoast: because that netmask makes no sense ?, the it basicly says that all bits in the ip are for the net and no for the host
01:35.33spanglesontoastso I need to change it on the router :|
01:35.35bullespanglesontoast: also, dont use spaces between = etc, it tends to make asterisk go nuts from time to time
01:35.47bullespanglesontoast: on the router ?
01:35.50spanglesontoastyea
01:35.57spanglesontoastthat's what the router is assigning
01:36.12JT255.255.255.255 is the broadcast address
01:36.16[TK]D-Fenderqwell : what are you talking about?  thats a perfectly valid host entry! ;)
01:36.30QwellI never said it wasn't :P
01:36.38[TK]D-Fender;)
01:36.51*** join/#asterisk Weems (n=frodo@unaffiliated/weems)
01:37.41spanglesontoastwell should i change the subnet ?
01:38.05cr4z3d255.255.255.0
01:38.08cr4z3dshould be your subnet
01:38.09bulleyes, most likely to 255.255.255.0
01:38.23bulleif all your machines are on ips on 192.168.1.
01:38.26cr4z3dyeah
01:38.55spanglesontoasthang on that's screwed up
01:39.01spanglesontoastthe drop down says 255.0
01:39.18spanglesontoastbut on the other it says 255.255
01:39.36JTthe drop down... yes
01:39.39cr4z3dquick question, how do i find out if i'm connecting to my iax2 server correctly in the CLI?
01:39.56JT...we know what you're talking about..
01:40.03spanglesontoastooh I can switch off nat :)
01:40.13cr4z3duh
01:40.16cr4z3di wouldn't recommend it
01:40.19JT~wglwat
01:40.22jboti guess wglwat is well, good luck with all that
01:40.40lokkju_wrkcr4z3d, iax2 show peers, perhaps?
01:40.43marc\cbaspanglesontoast what are you trying to do?
01:40.44spanglesontoastah
01:40.46spanglesontoast192.168.1.0255.255.255.00.0.0.0LAN & Wireless
01:40.54spanglesontoastit is that your quite right
01:41.04spanglesontoastthat was the wan one
01:41.35cr4z3dlokkju_wrk, when i do that i get 0online 0 offline and 1 unmonitored
01:42.14spanglesontoasthmm still don't work grrr
01:42.24cr4z3dwoah Mar 21 18:41:09 WARNING[17408]: chan_iax2.c:9680 load_module: Unable to open IAX timing interface: No such file or directory
01:42.42cr4z3dthat could be a huge reason it's not working couldn't it
01:45.44spanglesontoastis there anything more verbose than -v
01:45.45spanglesontoast:)
01:46.23cr4z3dscrew this i'm removing asterisk and building it from source with the newest one..
01:46.30JTspanglesontoast: -vvvvvvvvvvvvvvvvvvvvvvvvvv
01:46.40JTreally, only 5 to 10 vs are needed
01:46.43cr4z3dset verbose 10
01:46.48mmartinn"set verbose 99999999999999999999999999"
01:46.49JTthere is no more verbosity after that
01:47.10mmartinnI guess it's "core set verbose 99999999999999999999" now
01:47.19cr4z3danyone here ever setup asterisk with NuFone.. i'm having such a hard time getting it working
01:47.19spanglesontoastlol
01:47.20JTyeah but who uses 1.4? :P
01:47.26Strom_Mcore set verbose 9999999999999999999999999999999999999999999999999999999999999999999999 :)
01:47.27cr4z3di'm using 1.12
01:47.34JT1.12
01:47.35cr4z3dsince that's what ubuntu has with apt-get
01:47.37spanglesontoasthmm it showed me something I didn't see
01:47.38JT?
01:47.53spanglesontoastyea I'm using ubuntu maybe why we have so many problems cr4z3d
01:48.07JTi doubt it
01:48.14JTdistro is usually irrelevant
01:48.21cr4z3dyeah i doubt it has any effect either
01:48.27spanglesontoastwhich version of ubuntu you using ?
01:48.30mmartinnThe * from ubuntu's multiverse is 1.12?
01:48.31cr4z3dedgy
01:48.34spanglesontoastah
01:48.37spanglesontoastfiesty here :)
01:48.42[TK]D-Fenderspanglesontoast, You have * running.  once you've gotten that far you need to learn how to USE it
01:48.45JTcr4z3d: i doubt it's 1.12
01:48.51QwellNo such version
01:48.52JTcr4z3d: show version
01:48.59cr4z3dhold on i just removed it
01:49.00Qwell1.2.12 maybe
01:49.07*** join/#asterisk oQPa (n=uawename@33.Red-83-34-60.dynamicIP.rima-tde.net)
01:49.19spanglesontoastwell it's just a mission for the thing to connect to this sip server
01:49.33cr4z3doh my bad
01:49.35cr4z3d1.2.12
01:49.38Qwelleww
01:49.43Qwelleww
01:49.51Qwellboth of you need to upgrade
01:49.58mmartinn1.2.12 is still better than 1.12, lol
01:50.02cr4z3dwill that help out at all though
01:50.08spanglesontoastAsterisk 1.2.16
01:50.10JTbetter than something that doesn't exist
01:50.11cr4z3di can't even get it to connect to nufone
01:50.12spanglesontoastoh
01:50.13spanglesontoastlol
01:50.16Qwellspanglesontoast: You too.
01:50.18QwellUpgrade
01:50.20spanglesontoastlol
01:50.21*** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id)
01:50.22*** join/#asterisk oQPa (n=uawename@33.Red-83-34-60.dynamicIP.rima-tde.net)
01:50.26spanglesontoastthis the reason then ?
01:50.29spanglesontoast:D
01:50.59[TK]D-Fenderspanglesontoast, pastebin your [general] section of sip.conf masking only passwords
01:51.01[TK]D-Fender~pb
01:51.05jbotsomebody said pb was a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste
01:51.55cr4z3di'm reinstalling and building from source.. this way i know it's nothing wrong with the one ubuntu gives you
01:51.55mihinomenestI have a serious problem with asterisk.
01:52.19bullemihinomenest: who doesnt
01:52.23mihinomenestit goes down everytime the power goes out because of the thunderstorm.
01:52.39sevardget a UPS
01:52.45[TK]D-Fendermihinomenest, Your ire is misplaced
01:52.46spanglesontoasthttp://pastebin.ca/405972
01:53.10[TK]D-Fendermihinomenest, And your lack of strategy apalling :)
01:53.44spanglesontoastI can't believe how much I've learn't over the past few hours
01:53.53spanglesontoastI know it's only the sip part
01:53.55orlockmihinomenest: you know that your phones will BLOW UP if a lightning bolt hits them?
01:54.01orlockin fact your whole house may catch on fire.
01:54.03lokkju_wrk[TK]D-Fender, did you see my idea for a script earlier?  have you seen anything that already does it?
01:54.03spanglesontoastbut with channels etc and conference rooms
01:54.05mihinomenestyou actually banged your head on the desk because of me, didn't you [TK]D-Fender?
01:54.08orlocki would go and bitch to somebody about that too.
01:54.15[TK]D-Fenderlokkju_wrk, No I didn't
01:54.29[TK]D-Fendermihinomenest, No.... you're not worth it :)
01:54.32sevardTK bangs his head on the table every 3 and a half minutes.
01:54.42lokkju_wrk[TK]D-Fender,  a script you could point at some text files (configurable) that would read off the per minute rate for the outbound number you are dialing, and then ask you to confirm that you would like to make the call - perhaps also allow you to set it so it only activates when the rate/min is over a specified amount? (or, does someone know of a script that already does this?)
01:54.48sevardhe's like old faithful
01:54.50mihinomenestgood idea.
01:55.01mihinomenestforgoe the frustration and go right for the pain.
01:55.21*** join/#asterisk billzybub (i=bill@68.82.147.79)
01:55.21[TK]D-Fenderlokkju_wrk, Sounds easy enough, but nothing "canned" out there that I've heard of off-hand
01:55.34billzybubgood evening fine people
01:55.48sevardbillzybub: eat smack
01:55.57[TK]D-Fenderharem*
01:56.00lokkju_wrkk, cool...  I want it so I get warned if I am about to make a 10/min call, so I'll just have to write it myself :)
01:56.04spanglesontoasthttp://pastebin.ca/405972 ....
01:56.05[TK]D-Fender*bangs*
01:56.06billzybubsmack?
01:56.10[TK]D-Fendergeez... can't type tonight
01:56.11sevardlokkju_wrk: I could make such a script
01:56.12billzybubas in heroin?
01:56.25lokkju_wrksevard, I'll have no problem writing it, thank you much :)
01:56.28*** part/#asterisk oQPa (n=uawename@33.Red-83-34-60.dynamicIP.rima-tde.net)
01:56.29sevardlokkju_wrk: ah.
01:57.18spanglesontoastdamn flies
01:57.20spanglesontoast:)
01:57.46mmartinnlokkju_wrk: I think I saw something about a rate list of exchanges and rates per carrier when searching about NANP once
01:59.09lokkju_wrkmmartinn, very likely - but I want a verbal "this call will be charged at XX per minute.  Please dial 1 to continue"
01:59.20lokkju_wrkmmartinn, course, I *would* like to see if anyone has reasonable rates for calling iridium...  most people are $8+/min - some are as high as $11
01:59.23billzybubhey, can somone tell me what the best front end for asterisk is?
01:59.38orlockvi
01:59.52mihinomenestnano.
02:00.01tzafrirvim
02:00.06billzybubi like vi, but i dont think my employees will
02:00.20cr4z3dMar 21 18:59:59 WARNING[18388]: chan_sip.c:12863 reload_config: Failed to bind to 0.0.0.0:5060: Address already in use
02:00.30mihinomenestemployees don't configure asterisk.
02:00.35lokkju_wrkwell, technically a soft or hard phone is a "front end" for asterisk...  so the Polycoms!
02:00.38[TK]D-Fenderspanglesontoast, http://pastebin.ca/405983
02:00.38billzybubmine do
02:00.47mihinomenestLord of the Network configures asterisk.
02:00.51tzafrirwhat do they need to configure?
02:00.56orlockbillzybub: freepbx
02:00.58[TK]D-Fenderspanglesontoast, What EXACTLY have you forwarded to your * server?
02:01.02cr4z3dhow come i get Mar 21 18:59:59 WARNING[18388]: chan_iax2.c:9680 load_module: Unable to open IAX timing interface: No such file or directory
02:01.12spanglesontoastno that's outbound
02:01.14cr4z3dwhen i start up asterisk after a fresh install
02:01.17lokkju_wrkno, no, the BOFH configures it for you...  and controls your life.  and the life of everyone on his network....
02:01.18tzafrirdestar has nice user config
02:01.25spanglesontoastbasically you dial the prefix and it dials a number on there
02:01.45orlocki am a bastard and i'm ok.. i work all night and i sleep all day!
02:02.07cr4z3doh man asterisk now sounds so easy
02:02.08spanglesontoastthe voipuser one
02:02.27[TK]D-Fendertzafrir : I keep reading that as Death Star ;)
02:04.54*** join/#asterisk Hansin321 (i=Eric@c-71-56-216-97.hsd1.co.comcast.net)
02:05.09mmartinnthe manager's manager_event is making me cry...
02:05.45billzybuborlock, thanks checking out the screenies now
02:06.21JT[TK]D-Fender: yeah, misreadings are funny like that
02:06.28JT[TK]D-Fender: i read deskstar as deathstar
02:07.21*** join/#asterisk AJaymn (n=Me@66-188-80-40.dhcp.mdsn.wi.charter.com)
02:07.26[TK]D-FenderJT : They WERE nick-named that :)
02:07.39[TK]D-FenderJT Hitachi/IBM's IDE drive series that is...
02:07.41*** part/#asterisk AJaymn (n=Me@66-188-80-40.dhcp.mdsn.wi.charter.com)
02:07.42spanglesontoastalso what you mean forwarding ?
02:07.57[TK]D-Fenderspanglesontoast,  What ports are your forwarding from your router to *?
02:08.04spanglesontoastit's dmzed
02:08.07JTi know :)
02:08.11JTi had one :/
02:08.26[TK]D-FenderJT, Worse still.. my frined had 2 ... in raid *0*
02:08.49[TK]D-FenderJT, "repetitive tragic failure" comes to mind...
02:09.29mihinomenestmy dad calls those drives "deathstar"
02:09.46[TK]D-Fenderspanglesontoast, Ok well * is set up for basic NAT work.  Now its up to you to make sure you are registring properly (as you appear to be attempting), and that those entries you made match the auth you need to send to place calls
02:11.15spanglesontoast-- Executing Dial("SIP/edd-08195f40", "SIP/8009178765@voipuser|60") in new stack
02:11.15spanglesontoast<PROTECTED>
02:11.20spanglesontoastthat's what it's doing
02:11.37spanglesontoastand just carries on
02:12.51bullei bet there is no host named voipuser on the internet
02:13.06spanglesontoastwww.voipuser.org
02:13.19spanglesontoastallows you to make free uk calls :)
02:13.49spanglesontoastand I know it works as a community demoed it on the radio with asterisk
02:14.09spanglesontoastI can use a normal soft client with it fine
02:14.37spanglesontoastand they have a setup aswell http://www.voipuser.org/forum_topic_330.html
02:14.38cr4z3dcan someone help me figureout why i keep getting these two errors when loading asterisk?
02:15.35bullespanglesontoast: so what is the name of the voipuser.org sip proxy then ?
02:15.37*** join/#asterisk ooor4 (i=blah@69-163-163-195.atlsfl.adelphia.net)
02:15.41sevardhttp://www.tfproject.org/tfp/showthread.php?t=114784
02:15.57spanglesontoastsip.voipuser.org
02:16.02*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM]
02:16.02*** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM]
02:16.02*** join/#asterisk kore (i=kore@mindwipe.org) [NETSPLIT VICTIM]
02:16.05bulleso place the call to that then
02:16.10spanglesontoasthmm
02:16.14sevarduh, wrong paste.
02:16.18sevardnsfw :\
02:17.32mmartinn<awkward_silence/>
02:17.35[TK]D-Fenderspanglesontoast, And why don't you have a peer entry for [voipuser] like that sample SHOWS you?
02:17.51spanglesontoastdoesn't make any difference
02:18.01spanglesontoastplus I've been messing with it
02:18.15sevardhehehe
02:18.37[TK]D-Fenderspanglesontoast, that dial statement is LOOKING FOR IT.  Damn right its IMPORTANT
02:19.43spanglesontoastexten => _81.,1,Dial(SIP/${EXTEN:3}@voipuser,60)
02:20.47spanglesontoastoh
02:20.54spanglesontoastthat did something error 503 on sjphone
02:21.14spanglesontoastsays the circuit is busy
02:21.19spanglesontoastcould that be true ?
02:22.01[TK]D-Fenderspanglesontoast, you don't have an entry for [voipuser].  it is also not a valid hostname.  that entire line is a DEAD END.
02:22.16[TK]D-Fenderspanglesontoast, Go follow what the sample handed to you
02:22.24spanglesontoastwell I changed it to [voipuser]
02:22.25[TK]D-Fenderspanglesontoast, And then expand on it
02:27.48*** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner)
02:28.57*** join/#asterisk ppyy (i=ppyy@218.93.70.78)
02:29.19Dovidspanglesontoas: u have ${EXTEN:3}
02:29.29Doviddont u want ${EXTEN:2} ?
02:29.50bullehe firstly needs a proper entry for the peer
02:30.02Dovidbulle: he dosent have the peer ?
02:30.10Dovidrofl
02:30.21bullenope
02:30.26Dovidhahahahahaha
02:30.50Dovidits like a client of mine who's wifi wasnt working. it wasnt plugged in (true story)
02:31.01Qwellgrr, I need a second tv tuner..  missing the last 10-20 seconds of a show so it can start recording another is annoying
02:31.33Dovidhehe
02:31.43DovidQwel: what r u using to record ?
02:31.46Qwellmythtv
02:31.51Dovidthere is an open source os...
02:31.53Dovidah. u got it..
02:31.55Dovidhow is it ?
02:31.58Qwelllove it
02:32.01Dovidhehe
02:32.06Dovidhow many gigs per hour ?
02:32.11Qwelldunno
02:32.28Dovidwhen i get back to the US ia m getting one
02:32.31Dovidwith multiple cards.
02:32.36Qwellprobably just about a gig
02:32.40Dovidcan u set it to remove commercials by default ?
02:32.49Qwellyeah
02:32.52Dovidlike the old tivo's ?
02:32.56Dovidnice !!!
02:32.57Qwellit's pretty good at it too
02:33.51carrarwoot
02:34.04carrarso tired of walking around all day!
02:34.37*** join/#asterisk Ac1dcrawl (n=cow@64.31.169.118)
02:34.46bullecarrar: get a segway
02:35.10carrarman that would be perfect here
02:35.16Qwell"here"?
02:35.18carrarput a seat on it
02:35.22Ac1dcrawlI'm having a problem, I have all incoming calls set to ring to a single extension.  When I call into the asterisk box, it rings once on the extension then goes busy
02:35.23carrarspringvon
02:35.27Qwellahh, figured
02:35.29Ac1dcrawlI see the following line in the logs: DIALSTATUS=CANCEL
02:35.31Ac1dcrawlany ideas?
02:35.45spanglesontoasthmm
02:35.48Qwellcarrar: except for those damn escalators :p
02:35.53carrarheh
02:35.53spanglesontoastwhy is the network congested :|
02:35.55carrarelevator!
02:35.58Qwellthose bugged the hell out of me, heh
02:36.45QwellI actually wonder how a segway would work on an escalator
02:37.23spanglesontoastI'm sleepy now i'm going to bed ty for all those put up with my dumbness ;)
02:40.34*** join/#asterisk anthonyl (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net)
02:42.38carrarmy obligratory photo with Mark  http://www.osburn.com/IMG_5659.jpg
02:42.41carrarheh
02:43.06Qwellmy picture with Mark rocks :D
02:43.10carrarheh
02:44.34cr4z3dhmm.. i keep getting these 3 error messages when starting up asterisk. can anyone help me? i posted the errors and config files on the forum: http://forums.digium.com/viewtopic.php?p=47177#47177
02:46.55*** join/#asterisk boch (n=fran@190.48.194.212)
02:49.19Qwellcarrar: see msg ;)
02:51.03*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
02:54.25billzybubcr4zed, why dont you grab the 1.4x source, rebuild and see how that works for you?
02:55.51[TK]D-Fendercr4z3d, First you're running Ekiga on the same box as * aren't you?
02:55.52billzybubcr4zed, also, since your using ubuntu make sure you have all the development tools installed
02:56.13[TK]D-Fenderbillzybub, * is INSTALLED and running.  that isn't his problem so far
03:01.00billzybubnice, a nuclear radioactive device has been stolen in philly
03:01.02cr4z3d[TK]D-Fender, yeah ekiga came with ubuntu default
03:01.09cr4z3dis that a problem?
03:01.28billzybubekiga is just a client, that should keep you from binding to 5060 on your interface
03:01.43cr4z3dooh alright so that's what's stopping that
03:01.46[TK]D-Fendercr4z3d, tahts the reason you have the "already bound" error.  Ekiga has claimed ownership of the SIP port before *.
03:02.02cr4z3dok so if i just turn of ekiga that won't happen anymore?
03:02.03billzybubcrazed, is ekiga running ?
03:02.05[TK]D-Fendercr4z3d, You need to run your soft-phone on a differen port and set taht in *
03:02.08cr4z3dyeah it's running
03:02.24billzybubi dont run my xlite on a different port...
03:02.44cr4z3di'm just going to run a windows soft phone instead
03:02.54cr4z3dto avoid any of that port stuff for now
03:03.06cr4z3di just want to get it up and connected to my NuFone
03:04.29*** join/#asterisk djs_2_6 (n=DJS@cpe-071-077-048-198.nc.res.rr.com)
03:04.51cr4z3dok so now i disabled ekiga and damn
03:04.58cr4z3di got like 3 pages of notices and warnings
03:05.16cr4z3dmost seem to do with musiconhold.c
03:06.05[TK]D-Fendercrazed, just make sure that "mode=files" in musiconhold.conf
03:06.41billzybubcrazed, get into a shell and type this in: netstat -tap
03:06.53[TK]D-Fendercr4z3d, And that post you made did not actualyl show a SINGLE error related to your nufone setup
03:06.58cr4z3ddefault, mode=quietmp3
03:07.01billzybublook for ekiga
03:07.10[TK]D-Fendercr4z3d, Change it and reload
03:07.14billzybubwhat port is it established tto
03:07.47[TK]D-Fenderbillzybub, 5060... thats th clrea reason for the error....
03:07.59*** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net)
03:08.16cr4z3dok now i get notices about res_odbc.c
03:08.41cr4z3dand this warning
03:08.42cr4z3dMar 21 20:07:50 WARNING[21264]: res_odbc.c:565 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified
03:09.31billzybubcrazed try this for your softphone http://www.xten.com/index.php?menu=download_xlite&platform=linux
03:09.33cr4z3d[TK]D-Fender, really? so how can i see if i'm connected to NuFone or not? iax2 show peers says 0 online
03:10.15cr4z3dalright billzybub i'll take a look at as soon as i can get these errors away
03:10.31[TK]D-Fendercr4z3d, You are not issuing a "qualify" to it so don't EXPECT to see anything.
03:10.47[TK]D-Fendercr4z3d, That is not in and of itself indicative of anything.
03:10.53[TK]D-Fendercr4z3d, lets move along...
03:11.06*** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net)
03:11.19[TK]D-Fendercr4z3d, in modules.conf add "noload => res_odbc.so"
03:11.25[TK]D-Fendercr4z3d, and start it up again
03:12.57cr4z3dhm this time i got a really bad error and the program crashed
03:13.10cr4z3dMar 21 20:12:31 WARNING[21446]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/func_odbc.so: undefined symbol: odbc_obj_disconnect
03:13.10cr4z3dMar 21 20:12:31 WARNING[21446]: loader.c:554 load_modules: Loading module func_odbc.so failed!
03:13.43[TK]D-Fendercr4z3d, in modules.conf add "noload => func_odbc.so"
03:13.51Qwellthere will be more
03:14.04Qwellcdr_odbc et al
03:14.05[TK]D-Fenderqwell : and they will in turn be stifled :D
03:14.18billzybubdamn, see that chicago cop beeting up that lady bartender for flaggin him?
03:14.39tenebillzybub: "flaggin"?
03:14.39[TK]D-Fendercr4z3d, How about you go into Synaptic and install unixODBC :)
03:15.04cr4z3done error
03:15.06cr4z3dMar 21 20:14:36 WARNING[21535]: chan_iax2.c:9680 load_module: Unable to open IAX timing interface: No such file or directory
03:15.22*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
03:15.29*** join/#asterisk xtr-II (n=94752345@S0106000c41ed11e1.vf.shawcable.net)
03:15.31cr4z3dunixODBC?
03:16.18Qwellif res_odbc got built, it is installed
03:16.24Qwellon 1.4 at least?
03:16.26*** join/#asterisk mroberto (n=dads@S0106001346face5f.ed.shawcable.net)
03:16.43mrobertoI need some help using cepstral with asterisk, is anybody using it
03:16.49cr4z3dyeah it's already installed
03:16.52cr4z3dand newest version too
03:17.57mrobertoi need some help using it
03:18.09[TK]D-Fenderqwell :Ubuntu packaged <0
03:18.29*** join/#asterisk dmux (n=dmux@c906497d.virtua.com.br)
03:18.41cr4z3dit's version 1.2.12 if that helps at all
03:18.42[TK]D-Fendercr4z3d, Make sure to install Zaptel as well
03:18.54billzybubtene, its a term used by bartenders that pretty much means no more serving a patron because they are obviously thoroughly inebriated or disorderly
03:19.10mroberto<[TK]D-Fender>: Do youhave experience with cepstral
03:19.11cr4z3dit's install zaptel right now
03:19.25cr4z3doh wait it ran into an error while installing
03:19.29[TK]D-Fendermroberto, Nope
03:19.41billzybubmroberto, thats that do?
03:19.49cr4z3dZaptel telephony kernel driver: FATAL: Module ztdummy not found.
03:20.12billzybubmodprobe ztdummy
03:20.17billzybubreload
03:20.18mrobertoI have the module setup and when i use it it i dont get any sound
03:20.47cr4z3dFATAL: Module ztdummy not found.
03:21.06billzybubcrazed type modprobe ztdummy
03:21.14cr4z3dyeah i did and i got that fatal message
03:21.33billzybubyou need to make install that then
03:21.37*** join/#asterisk kgx (n=kgx@60.234.20.178)
03:22.33cr4z3dhm how would i go about doing that and what exactly is ztdummy? it hink i read something about it during the setup guide
03:23.57JTyou only need zaptel if you are using zap hardware, or meetme conferences, or iax2 trunking
03:24.14JTztdummy if you have zaptel but no zap hardware
03:24.43billzybubztdummy is a timer that runs off of some imbeded usb hardware i think
03:25.07cr4z3dhm i dont plan on doing any meetme confrences or using zap hardware
03:25.16cr4z3dand i'm pretty sure i'm not doing any iax2 trunking
03:25.19JTyes it uses usb
03:25.20[TK]D-FenderOf either USB or the 2.6 kernel timer
03:25.33JTeither way
03:25.34billzybubare you just usinf a softfone or do you have any fxs setup?
03:25.41JTyou don't need it unless you do :)
03:25.53cr4z3dmy plan is to just use a softphone until i can get a sip enabled hardware phone
03:26.39billzybubso just build asterisk, its my understanding zaptel is a seperate package i think, just installed asterisk myself yesterday
03:27.16JTit is a seperate package
03:27.18cr4z3di have it installed ad the only error left thanks to [TK]D-Fender is Mar 21 20:26:47 WARNING[22830]: chan_iax2.c:9680 load_module: Unable to open IAX timing interface: No such device or address
03:27.38cr4z3doh timing that was the zaptel thing wasn't it
03:27.38Juggie'modprobe ztdummy'
03:27.43Qwellcr4z3d: it's a warning, NOT an error
03:27.54cr4z3doh so does it really mean anything for me?
03:28.02Juggieyou wont be able to use iax trunking without it.
03:28.05QwellNot if you aren't using iax trunking
03:28.18Qwell(and if you're connecting to nufone - you aren't)
03:28.29cr4z3dok cool so everything should be working then?
03:28.55[TK]D-Fendercr4z3d, Well.. lets say nothing "pre-broken" :)
03:29.16*** join/#asterisk BigCanOfTuna (n=arustad@dsl-mac-66-18-226-119-cgy.nucleus.com)
03:29.38cr4z3dwell yeah now it's just error from my configurations if it's not working right?
03:29.48billzybubcrazed why arent you using the 1.4 source?
03:29.54[TK]D-Fendercr4z3d, So far...
03:30.05cr4z3dbecause i didn't compile it from source i just apt-get install asterisk
03:30.10cr4z3dand used the one it came with
03:30.16billzybuboh no, dont do that
03:30.19[TK]D-Fenderbillzybub, Becuase he's on Ubuntu and working with lovely packages!
03:30.29billzybubim on ubuntu too
03:30.44cr4z3dpackages are awesome i wish windows had something like that
03:30.50billzybubheh
03:31.14cr4z3dbut anyway how do i check to see if i connected with nufone?
03:31.29billzybubis nufone a softfone?
03:31.40JTit's an itsp
03:31.41cr4z3dno it's my service provider
03:31.45billzybubsudo asterisk -r should put you in the console
03:31.51cr4z3di'm in the console
03:31.56cr4z3diax2 show peers
03:32.03cr4z3di see it but it says status is unmonitored
03:32.47billzybubthat may be a good thing :)
03:33.04cr4z3dreally?
03:33.04billzybubwhat do they charge you over there?
03:33.11cr4z3duh 2 cents a minute i think
03:33.16cr4z3dand $5 a month for a phone #
03:33.20billzybubon all calls?
03:33.28cr4z3d5 cents international i believe
03:33.32[TK]D-Fendercr4z3d, Ok, new concept for you.  you are not "connected" to anything.  SIP/IAX do not maintin constant "connections".
03:33.48cr4z3doh did not know that
03:33.53cr4z3dso how can i check if they can connect
03:34.12[TK]D-Fendercr4z3d, dial the # they gave you that you supposedly set up.  Try dialing OUT.
03:34.20[TK]D-Fendercr4z3d, You know... USE THEM.
03:34.34billzybubcrazed, how many out-bound connections does that give you?
03:34.38cr4z3di can't test dialing out yet since i had to turn off ekiga for now but i'll see if ican dial in
03:34.47*** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net)
03:34.47*** mode/#asterisk [+o mog] by ChanServ
03:35.17billzybubtry xlite when you get things set up, i think its neet-er :D
03:35.37billzybubkinda looks like a phone instead of a window
03:35.47cr4z3dok when dialing in 15 seconds later it goes "the person you are calling is currently unreachable"
03:36.00*** join/#asterisk vader-- (n=me@71.226.201.15)
03:36.02vader--hola
03:36.03JTare you registering?
03:36.05cr4z3dand the iax2 show peers status never changed
03:36.22cr4z3dnothing shows up in the CLI window
03:36.25[TK]D-Fendercr4z3d, you are not WATCHING the peer, nor do you need to
03:36.29JTyou need to register
03:36.33billzybubhey crazed, when you get things rolling and i peer to you for my outbound domestic?
03:36.34[TK]D-Fendercr4z3d, Go set up a phone now.
03:36.47[TK]D-FenderJT, No, he DOESN'T. (not to place calls anyways)
03:36.47vader--Do you guys know of any good voip providers that offer goodpricing that have the following, regular lines, fax and 800 with the ability for me to port a current fax line over and a 800 number over
03:37.01JT[TK]D-Fender: he said he's calling himself
03:37.07JTpay attention :)
03:37.20JThe "can't" call out because ekiga is off
03:37.21vader--i don't think vonage will allow me to port numbers over to fax line or port a 800 number over
03:37.24[TK]D-Fendercr4z3d, "iax2 show registry"
03:37.25cr4z3di have it set up with register => user:pass@switch-1.nufone.net if i remember the config correctly
03:37.26billzybubintervention is on A&E it a great show
03:37.40cr4z3dstate registered
03:37.41JTvader--: umm, fax?
03:37.42billzybuboxycontin addict this week :D
03:37.44vader--ya
03:37.48[TK]D-Fendervader--, FoIP = death
03:37.52JTvader--: very few people do fax properly yet
03:37.56QwellFoIP == good
03:37.57JTFoIP is alright
03:38.00QwellFoVoIP == dumb
03:38.02JTFoVoIP bad
03:38.04vader--i have a pots line that is used for faxing
03:38.15vader--vonage offers a fax line with a voice line for 50$ a month
03:38.33JTvader--: very few people do T.38 yet
03:38.34cr4z3d[TK]D-Fender, the state shows registered
03:38.39JTT.38 is what you need for faxing
03:38.53[TK]D-Fendercr4z3d,  Yay.  No on to the more than probable networking issues!
03:39.00vader--i need to figure out if i can port a 800 number over
03:39.08cr4z3dhaha alright
03:39.18billzybubhey dont you need a timer for IAX peering?
03:39.19[TK]D-Fendercr4z3d, So.. your * server behind NAT?
03:39.31[TK]D-Fenderbillzybub, No, only for TRUNKING.
03:39.35cr4z3dwell i thought iax tunneled around nat?
03:39.37cr4z3dbut yeah
03:39.48vader--i need to figure out what provider owns this 800 number
03:39.50[TK]D-Fendercr4z3d, You need to forward 4569 UDP to your * box
03:40.02cr4z3dhm alright no problem
03:40.15JT[TK]D-Fender: not if he registers he shouldn't
03:40.29JTas long as there's no interfering firewall
03:40.40cr4z3ddoes ubuntu have a firewall by default
03:40.46JTno idea
03:40.47cr4z3di know on my router i have the spi firewall disabled
03:42.17[TK]D-FenderJT, He's behind NAT.  Don't think for a second that Nufone will be wasting packets keeping a UDP port forwarding towards his *
03:42.32[TK]D-Fendercr4z3d, Yes, you have to forward it.
03:42.55JT[TK]D-Fender: umm, i'm behind NAT too
03:43.18[TK]D-FenderJT, if his NAT doesn't get any UDP keep-alive it'll close down the exterior port.
03:43.20JT[TK]D-Fender: and have NEVER needed to port forward to use SIP or IAX if i'm connected to a provider (ie. clients don't connect to my asterisk)
03:43.37[TK]D-FenderJT only way it'd survive is if YOUR system was nagging the outside world constantly.
03:43.49JT[TK]D-Fender: that's what registration in a period of time less than the NAT timeout
03:43.57[TK]D-Fenderso fer cryin' out loud, just forward the darned port!
03:43.59[TK]D-Fender:)
03:44.02JT[TK]D-Fender: that's how most VoIP nat-punching systems do it
03:44.15[TK]D-FenderJT you shouldn't be registering every other minute :)
03:44.31[TK]D-FenderJT thats not the way to run an * server for it...
03:44.32JTbut you should :) the providers build their servers to handle the load
03:44.49[TK]D-FenderJT : thats amongst the "last ploys of the desperate".
03:45.01[TK]D-FenderJT Oh.. thats a LOAD alright! ;)
03:45.07JTwell this is how softphones work by default
03:45.15JTyou only need a REGISTER every minute or two
03:45.30*** join/#asterisk michaelross (n=michael@203.59.123.167)
03:45.59[TK]D-FenderJT : thats one way, or the OPTIONS packet for QUALIFY.  But again, this is what you do for more intermittant CLIENTS, not your SERVER
03:46.23cr4z3dok ubuntu has no firewall by default but anyway i'll go ahead and forward that port ad see if that help
03:47.34JT[TK]D-Fender: if his IP is dynamic, he has to register anyway
03:48.38cr4z3dok port is now forwarded
03:48.47cr4z3dshould i just restart asterisk to be safe?
03:49.10JTit won't actually do anything
03:49.13JTrestarting it
03:49.21JTsince port forwarding is completely seperate
03:49.33cr4z3dtrue
03:49.48cr4z3ddamn even the port forward i get the same error
03:50.04cr4z3dwell not really an error but basically nufone saying nothing is connected
03:50.17billzybubhey is anyone fammiliar with TDMOE?
03:50.25JTbillzybub: midly
03:50.28JTmildly
03:50.48billzybubtime division multiplex over ethernet
03:51.44billzybubit kinda gives me the impression i could use an off the shell ethernet card as a t1 card and plug it straight from the dmarc?
03:51.48*** join/#asterisk djs_2_6 (n=DJS@cpe-071-077-048-198.nc.res.rr.com)
03:52.13JTNO
03:52.17JTmost definately not
03:52.18*** join/#asterisk bmg505 (n=leon@196.209.249.12)
03:52.27billzybubdoesnt it sound like it though?
03:52.29JTit allows TDM to be framed over ethernet
03:52.37billzybubi got real excited when i first read it
03:52.42JTit doesn't allow money to appear from nothing
03:52.43JT;)
03:52.50cr4z3dhaha
03:53.11cr4z3dso it does say registered
03:53.28cr4z3dwhat could other possible problems be that are stopping me from getting to my * server
03:53.48billzybubif you can frame doesnt it stand to reason that you should be able to send those frames without encapsulation?
03:54.13JTbillzybub: what?
03:54.27JTbillzybub: seriously, i have no idea what you're talking about now
03:55.05billzybubim kinda talking about an ethernet card taking the place of expensive t1 interfaces
03:55.14*** join/#asterisk ManxPower (n=manxpowe@72.sub-70-196-33.myvzw.com)
03:55.35billzybubbut i just smoked some crypler so maybe im not making much sense
03:55.41JTbillzybub: i'm kind of suggesting your crazy
03:55.44JTyes
03:55.53JTtotally different L1 and L2
03:56.03[TK]D-Fenderbillzybub, YOU'RE CRAZY.
03:56.24[TK]D-FenderJT, there.. I SAID IT.  You have real commitment iddues ;)
03:56.28[TK]D-Fenderissues*
03:56.50JThar har
03:57.09JThey, i have a winmodem.....
03:57.13[TK]D-Fendercr4z3d, ok, enable IAX2 debug and pastebun the failed inbound call attempt
03:57.15[TK]D-Fender~pb
03:57.27jbotsomebody said pb was a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste
03:57.27JTi need to terminate a t3 though
03:57.30billzybubi dare to dream
03:58.02JTthat hurt :(
03:58.04cr4z3dnothing comes up when i call in
03:58.09billzybubseriously though, is it truly that inconceivable?
03:58.16JTyes
03:58.18cr4z3doh wait i got some tx-frame retrys
03:58.32JTbillzybub: is there a reason why you don't plug your keyboard into the power point?
03:58.39JTsurely it's just a software issue
03:58.46JTkeyboard over electricity
03:59.01cr4z3d~pb
03:59.12jbotfrom memory, pb is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste
03:59.48billzybubwhy is that, no one ever thought for a second in 1980 that one day, not only would they be able to play pac-man on a home computer, but actually play the original game rom and every other game in their local arcade.
04:00.19*** join/#asterisk rrrobert (n=rrobert@58-65-160-140.nayatel.pk)
04:00.27JTyes, but they also built computers to play pacman
04:00.28cr4z3d[TK]D-Fender, http://pastebin.ca/406120 looks like the same pattern keeps repeating every 30 seconds or so since i enabled iax2 debug
04:00.50JTthey didn't just get the joystick of an arcade machine and hammer it together with a home tv set
04:01.14billzybubi could make a million analogies, mark my words, we will see it done
04:01.51billzybubi must get back to installing freepbx
04:01.54JTbillzybub: you are saying one day we will be able to plug a T1 into a 10/100base-T ethernet card?
04:02.07cr4z3dhaha that would be insane
04:02.14JT~freepbx
04:02.26jbot[freepbx] unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
04:02.29billzybubi am saying one day it will be possible, yes
04:02.51JTbillzybub: you must be smoking something good to think it's possible, they're TOTALLY DIFFERENT
04:02.57JTyou'd need a hybrid chipset
04:03.04JTit is not a software issue
04:04.01billzybubnot so different, you just need to figure out how to hand the media off to the cpu
04:04.10JTno, the voltage levels are different
04:04.13JTthe PINS are different
04:04.13JTthe framing is different
04:04.14*** join/#asterisk ManxPower (n=manxpowe@57.sub-70-196-73.myvzw.com)
04:04.14JTthe clock is different
04:04.18billzybubmost modern cards have auto detection for the pins, turn a cross over cable into straight through per see
04:04.21cr4z3doh crap i gotta go.. [TK]D-Fender, if you can figure out what's wrog with that just pm it to me either way thanks for the help man
04:04.30billzybubclock can be handled by an external application
04:04.41billzybubframing can be handled by an external app
04:04.50JTit would be possible to build a card to do it
04:04.55JTnot make a current one do it
04:05.08billzybubi bet you could hack the right ethernet card to do it
04:05.13JTand the reason it's unlikely anyone would built such a card is... WHERE ON EARTH IS THE DEMAND?
04:05.17JTno demand = no product
04:05.33JTi bet you've been smoking too  much
04:05.55billzybubyour right, as the world becomes more digital the demand would go down
04:06.04JTthe demand was never there
04:06.13billzybubwhats the point in voip really if your still using trunks
04:06.18JTpeople with T1s/E1s = not poor people
04:06.34JTwhat's wrong with trunks?
04:06.41JTand the point is obvious
04:07.09*** join/#asterisk Belize (n=vasya@80.237.99.222)
04:07.56billzybuboh i disagree, spent the last 6 years of my life as a director in a marketing organizations, we had call centers ive provisioned many a t1 for use in NEC NEAX and 2000 series phone switches
04:08.35billzybubwe used pri and t's (d4-ami)
04:08.54billzybubt1 cards would go bad on occasion
04:09.06JTsure, that's possible
04:09.25billzybubso why would you say the demand was never there ?
04:10.09JTit means the demand is for a replacement card or a failover solution, not a crackwhore solution of a mutant ethernet + pri card which solves no real problem
04:10.34billzybubits not a crackwhore solution, it would be fun to work on
04:10.59JTfun, that's about it, it doesn't solve a real issue
04:11.01billzybubdo you have any idea what big iron t1/e1/pri interface cards cost?!
04:11.10JTyes i bet they cost a bit
04:11.17JTso why would someone make you a cheap one?
04:11.20JTdefies logic
04:11.28*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
04:11.29JTthings are cheaper when mass produced too
04:11.41billzybubit doesnt , i think it would be such an increadible boost for asterisk
04:11.43JTthe demand is obviously not that high for pri cards
04:12.03JTonly if someone made one for cheap, and there is no reason for any company to do so
04:12.16billzybubiwhat if you can hack one together for cheap
04:12.27billzybubthats the point i trying to make, dont be such a nay sayer
04:12.28JTthen when it fails you get fired, good idea.
04:12.34JTi'm being realistic
04:12.40JTyou're being fanciful :)
04:12.41billzybubyour being boring
04:12.45ManxPowerUh, what are we talking about?
04:13.00JTT1s aren't meant to be a fun exciting party trick
04:13.08ManxPowersounds like billzybub is not an Asterisk True Believer
04:13.18JTManxPower: billzybub things someone should make a driver to make an ethernet card work as a T1/pri card
04:13.23JTs/things/thinks/
04:13.28*** join/#asterisk Strom_M (n=strom@12-189-87-2.att-inc.com)
04:13.30billzybubim and talking about cost effectively hacking your off the shelf eathernet card to replace expensive t1/e1 interfaces
04:13.35ManxPowerFun and exciting T-1s get you fired.
04:13.40JTexactly
04:13.47billzybubjobt they've already done half the work with tdmoe
04:14.01ManxPowerJT: Ah.  How fanciful
04:14.03JTno, i keep telling you it's totally different
04:14.20Strom_Mno, quite obviously it's exactly the same
04:14.27Qwellclearly
04:14.29JT:)
04:14.29Strom_M</throwing shit into the fire>
04:14.32ManxPowerJT: For one thing T-1 voltages would blow an ethernet card
04:14.32acidchildwhats a good asterisk howto?
04:14.41JTManxPower: that's what i was trying to say
04:14.47ManxPoweracidchild: The Book
04:14.48ManxPower~book
04:14.50jbotextra, extra, read all about it, book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
04:14.50JTapparently it's only a software issue
04:14.58acidchildthanks.
04:15.17billzybubmanpower, ive plugged t1 from the dmarc into and ithernet card with no bad side effects
04:15.27acidchildpdf's blow ;/
04:15.30ManxPowerJT: Ever had Polycom phones lose their call forwarding settings?
04:15.34JerJeracidchild: i am slightly biased but here's one:  http://tinyurl.com/2aal8h
04:15.37JTacidchild: free books don't though
04:15.43JTManxPower: nup
04:16.02QwellJerJer: You write docs now? :P
04:16.28[TK]D-Fenderbillzybub, Congratulations, no giant mushroom cloud.  But will it do anything USEFUL?
04:16.36ManxPowerProperty Services got REALLY PISSED that their forwarding to the answering service failed 2 nights in a row.  I had to write a damn dialplan based call forwarding for them
04:16.49[TK]D-Fenderbillzybub, If you think the CARDS are expensive now, who has a PRI at HOME!?
04:17.06JerJerQwell: guess so
04:17.11JTi have a T1 at home
04:17.12[TK]D-FenderJerJer, Get me some crack while you're at it!
04:17.16JTbut it's 3 metres long
04:17.30JerJercrack is a waste of good cocain yo
04:17.37billzybubim not talking about home use
04:17.39[TK]D-FenderJT and NO a channel bank does NOT count :p
04:17.41ManxPowerThe thing about DSP-less T-1/PRI cards these days is that they cost about about the same as 1 - 2 months of T-1/PRI service.
04:17.45JT[TK]D-Fender: :(
04:17.48ManxPowerAnyone that can get a PRI can afford a card
04:17.49billzybubsilly, who would want 24 lines in their house
04:17.49*** join/#asterisk HaMYaI (i=HaMYaI@202.8.86.162)
04:17.50[TK]D-Fenderbillzybub, So whats the big deal of 50)$ for a PRI card?
04:17.56[TK]D-Fender$500
04:18.28[TK]D-Fenderbillzybub, You're jsut here thinking that it should cost 25$ for the card.
04:18.47[TK]D-Fenderbillzybub, Like the dev time and sales volume would warrant such a price
04:18.58QwellAny of you guys ever done RAID0+10?  heh
04:18.58billzybubwell, it does stand to reason that should somone come up with the solution it wouldnt be very good for Digium would it?
04:19.11ManxPowerIf DSPless T-1/E-1 cards had the VOLUME of ethernet cards they would cost the same as ethernet cards
04:19.22billzybubyou think?
04:19.36ManxPowerbillzybub: MANY people have tried to make cards significantly less that Digium.  They have all failed.
04:19.38JTit's obvious
04:19.42JTsupply and demand
04:19.58JTQwell: O+10? :o
04:20.01Qwellsure
04:20.09ManxPowerSome people would say there are BETTER cards than Digiums -- but they still cost about the same
04:20.11JTdon't you mean raid 10?
04:20.21[TK]D-Fenderbillzybub, If someone came up with such a think it would KILL Digium.  Diguim is in the business of making cards.  if someone undercut them that much they'd SINK.
04:20.21Qwellno, I mean RAID 0+10
04:20.32JTQwell: define 0+10 :)
04:20.49Qwell0+1, with an extra 0 :p
04:21.10ManxPower[TK]D-Fender: that would suck for Asterisk. 8-|
04:21.16JTwell RAID10 is a 2 * RAID0
04:21.29JT0+10 would be that plus another raid 0?
04:21.44Qwellyes
04:21.53Qwellbut at the lowest level
04:22.05[TK]D-FenderManxPower, Yes it would.  Whicle the project could live on, the loss of paid full-time programmers would be a serious blow
04:22.06*** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)
04:22.06Qwellso, 8 drives
04:22.12Qwellstriped, redundant, and striped again
04:22.25JTseems pretty pointless
04:22.30Qwellnot really :D
04:22.39[TK]D-FenderRAID 3 - For that "plaid" feeling :)
04:22.42JTbut it is
04:23.01JTno real gain for money better spent on backup or clustering/failover
04:23.19Qwellbut I want uber-fast :p
04:23.30Qwell...except on writes
04:23.44billzybubdo you really think it would crush asterisk? I dont, asterisk got very far before digium started making money, if theyve actually made any money yet. You dont think the development community is strong enough to support asterisk on it own - without asterisk?
04:23.47DocHollidaytoo bad asterisk cant do failover without dropping calls :|
04:24.13JTnothing this side of $20k can
04:24.24[TK]D-Fenderbillzybub, It would seriously hamper growth, yes.
04:24.37billzybubman theres some crazy thing on discovery hd right now
04:24.49JTQwell: what you're refering to is RAID100 btw
04:24.58QwellJT: nope, different levels of strip
04:24.59Qwellstripe*
04:25.03[TK]D-FenderSounds more like HAL 9000 to me....
04:25.18Qwellraid100 is a striped striped set of redundant disks
04:25.19JTQwell: what actually supports this mode?
04:25.22billzybubid be interesting to see digiums numbers
04:25.23Qwellnothing
04:25.28JTnice! :D
04:25.50QwellIt's just software though :D
04:25.57[TK]D-FenderQwell : Roughly how many drones do you guys keep on staff these days?
04:26.02Qwell90+
04:26.07ManxPowerbillzybub: Digium employs almost all of the Asterisk core programming team
04:26.17[TK]D-Fenderbillzybub, and thats just the DRONES.  Digium is not a "tiny" company
04:26.26Qwelloh, "those" drones
04:26.26Qwellno
04:26.36Qwellsignificantly less than that
04:26.48[TK]D-Fenderqwell : No, I'm not referring to your pre-patch chan_skinny botnet ;)
04:26.50Qwellthat's the entire company
04:27.06Qwelldon't quote me on that though
04:27.17billzybubanyone use beryl?
04:27.19*** join/#asterisk viking78 (i=aherbert@66-168-102-94.dhcp.jcsn.tn.charter.com)
04:27.36[TK]D-Fenderbillzybub,  "<Qwell> that's the entire company <Qwell> don't quote me on that though"
04:27.57billzybubeh?
04:27.58[TK]D-Fenderbillzybub, You are just ALL "bling" aren't you?
04:28.08billzybubwhat do you mean?
04:28.10[TK]D-Fender</sarcasm>
04:28.11[TK]D-Fender:D
04:29.31[TK]D-Fenderqwell : c'mon, that was witty, and unexpected!  And this medium robs of my Keystone Cops-style of hunour..
04:30.42QwellNo, witty is r131 of the aadk repository :P
04:31.12QwellAdd a comment for clarification, to explain where this file is generated.
04:31.12QwellIt isn't where you'd expect.
04:31.12Qwell(perhaps unexpectedly, it is not the Spanish Inquisition that does it)
04:31.42*** join/#asterisk djs_2_6 (n=DJS@cpe-071-077-048-198.nc.res.rr.com)
04:32.03fileQwell is so mean
04:32.07Qwellindeed
04:32.24QwellI would try to push JerJer over too, but...yeah
04:32.31JerJeryeah
04:32.35QwellI think we all know how well *THAT* would work
04:32.43*** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr)
04:34.30[TK]D-FenderJerJer, Imposing, are we? ;)
04:34.56JerJermore like huge
04:35.01Qwell[TK]D-Fender: the ratio is...very low
04:35.30[TK]D-FenderQwell : ... I'm afraid to ask WHICH.....
04:35.35Qwellhe's probably > a foot taller than me :P
04:35.40[TK]D-Fenderqwell : Although a few come to mind....
04:36.52*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
04:36.54[TK]D-FenderI'm nearing the break-point before pumping serious iron again.... floating at 190lbs currently @ 6'2-3"
04:37.19JerJeri'm 375 and 6'5"
04:37.20DocHolliday[TK]D-Fender, i got a personal trainer to gain weight :P
04:37.36Qwell> 1 foot, and > 3x weight...
04:37.57[TK]D-FenderJerJer, ouch
04:38.18Qwell5'3, 115 - what now? :P
04:38.26[TK]D-FenderJerJer, You are at some serious risk there....
04:39.29JerJerI was almost 400
04:39.42[TK]D-FenderJerJer, I hit a point about 5 years ago when I said "enough" when I was just risen back to my high of 265.  In 5 months I hit 195.
04:40.24*** join/#asterisk kb1_kanobe (i=user@d207-216-143-5.bchsia.telus.net)
04:40.30[TK]D-Fenderthen 2 years later at the start of summer I freakish dropped from around 195-200 to 176 :|
04:40.52[TK]D-Fenderout of thin air... never figured quite what the trigger was...
04:41.19[TK]D-FenderI'm doing decent where I'm at and jsut need to get off my ass and actually WORK for my goals.
04:42.16*** join/#asterisk gammah (n=gammah@cpe-66-69-224-62.austin.res.rr.com)
04:43.43billzybubmy wireless network card is giving me headaches
04:44.47vader--tk thats kinda my goal
04:44.54vader--last year i dropped from 250 down to 219
04:45.00vader--and now im back up to around 235
04:45.05billzybubive got one of those linksys wireless g cards that you need to run the firmware cutter on to extract the driver and its flakey at best
04:45.08[TK]D-Fendercontrol*
04:45.09[TK]D-Fenderashdasasdf
04:45.20vader--i started drinking coffee
04:45.20[TK]D-Fenderaphasia setting in full.. its getting late...
04:45.28vader--and my coffee is 800 calories
04:45.30vader--it's killing me
04:46.06billzybubvader can you sleep after you drink coffee?
04:46.09[TK]D-Fendervader--, Mine is NEGATIVE.  over the last 2 months I've switched to blak and only 1 artificail sweetener, and have gone pure-black on occasion
04:46.10vader--ya
04:46.18billzybubwish i could
04:46.24vader--i drink it in the morning
04:46.25[TK]D-Fendervader--, the trick is repeatability and portions...
04:46.26billzybubi cant sleep for sh1t
04:46.28vader--one 24oz cup
04:46.29vader--thats is
04:46.35*** part/#asterisk kb1_kanobe (i=user@d207-216-143-5.bchsia.telus.net)
04:46.44vader--tkd ya i lost all that weight before by cutting all soda and eating right
04:46.45[TK]D-FenderYou need to knock off the EVIL shit though...
04:46.46billzybubsleep is a big part of weight loss though
04:47.00vader--well the problem was i always felt fatigued
04:47.08vader--with coffee i don't feel fatigued
04:47.12vader--i feel how i should feel
04:47.16vader--but im gaining soo much weight
04:47.35Juggiedont put sugar in it
04:47.42vader--ya i put 4-5 packets
04:47.46[TK]D-Fendervader : meanwhile the insulin spikes you're getting hit with plus the caffeine are wreakiong havoc
04:47.48vader--i have to learn to drink it black
04:47.49Juggiewell... there you go, jeeze.
04:47.55Juggieuse splenda.
04:47.57billzybubi got that virus thing from mono and never got rid of it the doctors tell me, causes all kinds of problems one of them being insomnia
04:48.06[TK]D-FenderJuggie, for 800 cal trust me its not the sugar HE's adding
04:48.13Juggiesplenda rox.
04:48.14vader--splenda gives me headaches
04:48.14CrashHDwhat makes asterisk iniate the voicemail box setup greeting?
04:48.26CrashHD*initiate
04:48.31Juggieyou could just go to the gym and get in shape
04:48.33[TK]D-Fendervader--, Actually... I'm beginning to think I'm experiencing the splenda bit myself...
04:48.35billzybubcrash the voicemail program
04:48.36vader--im drinking dunkin donuts french vanilla
04:48.37vader--XL
04:48.38Juggiethat helps having more energy.
04:48.43vader--extra cream, extra sugar
04:48.45[TK]D-FenderJuggie, I lost my 70lbs doing jack shit.
04:49.01[TK]D-FenderJuggie, Like the law of programming says.  GIGO <-
04:49.03Juggiewell, yah, if you are 70pounds overweight you can loose it by just not eating.
04:49.14CrashHDbillzybub: it asked to record the directory name on a box, I had never heard that. what is the indicator it uses to actually determine that?
04:49.15JerJernot if you like to eat
04:49.18vader--tkd i couldn't figure out why i was getting these headaches when all i was drinking was flavored water
04:49.24vader--here the water was flavored with splenda
04:49.28Juggiemyself, i go to the gym 4-5 times a week.
04:49.29JerJeri get crazy headaches if i don't eat
04:49.32[TK]D-FenderJerJer, You can.  You just need to change WHAT you eat and when.
04:49.36billzybubcrash, ask me that again but in a different way
04:49.37vader--i cut that stuff out and headaches went away
04:49.56CrashHDwell
04:50.00CrashHDI have a voicemail box
04:50.05JerJer[TK]D-Fender:  oh yeah - i eat 4-5 small meals a day
04:50.11[TK]D-Fendervader--, I started getting them more when I cut the creme from my coffee
04:50.15JerJerand 30-45 minutes carido every day
04:50.22CrashHDthat is constantly telling the user to setup the box, resetting password, setting directory recording
04:50.23JTlosing weight by not eating is STUPID
04:50.27[TK]D-FenderJerJer, Sounds like you're on the right track.
04:50.30JerJer5-6 glasses of water
04:50.34billzybubalot of people think thats the key to losing weight, many small meals so your never hungry and never binge
04:50.34JTyou lose muscle mass as well as fat
04:50.37Juggieyah drink lots of liquid.
04:50.43JTwhich is bad for you
04:50.45[TK]D-FenderJT I never advocated NOT eating.  just change what & when.
04:50.55CrashHDI'm trying to figure out why
04:51.01CrashHDand determine what I can do about it
04:51.22Juggieif water bores you, drink crystal light.
04:51.29[TK]D-FenderI lost 10 -15 since taking up eating 5 meals a day, bringing fruit to work, and having a smoothie at night.  keeps things even through the day
04:51.29JTJerJer: what's in the meals though?
04:51.57Juggieanother thing is to NEVER EVER EVER eat late
04:52.01Juggiedont ever eat later then like 6-7pm
04:52.02[TK]D-FenderJT : If you are in an unhealthy state, you can afford a little muscle loss for the larger gaint o be had.
04:52.06billzybubcrash, i followed the tuorial here: http://www.asteriskguru.com/tutorials/asterisk_voicemail.html , works for me
04:52.15JerJerJT:  chicken, fish some beef
04:52.30JT[TK]D-Fender: with the right diet, you can lose fat and gain muscle at the same time
04:52.36JTJerJer: good good
04:52.44JerJerminmial carbs, but i cannot cut them out all the way
04:52.51Juggieif you are doing exercise you have to loose 17 pounds to loose 15.
04:52.58Juggiethats what i've been told.
04:53.02JTJerJer: heard of carb cycling?
04:53.15Juggiebecause if you burn 17 pounds of fat, your going to add muscle.
04:53.17ManxPower[TK]D-Fender: I believe that eating many small meals helps.
04:53.20billzybubeveryone i know who did the atkins diet lost alot of weight. But they also gained it all back :(
04:53.28[TK]D-FenderJT You can, but that does require a fair amount of work.  Muscle mass takes a lot of work to gain.  Its a qustion of whre you are in the phase of development.  If you are simply coming DOWN a LOT then put your focus on the easy bit first
04:53.34JTJuggie: depends how you burn it
04:53.38Juggiebillzybub, i did that diet, lost 25 pounds and gained back about half.
04:53.46[TK]D-FenderManxPower, Indeed,  I've subscribed to that for some time.
04:54.01JT[TK]D-Fender: you can reliably lose a lot of weight with carb cycling
04:54.02Juggiei am 210 now (down from 225 @ christmas)
04:54.05JT2 days a week no carbs
04:54.11Juggieand my goal is 190. which is a long way off.
04:54.13[TK]D-FenderAtkins has a GREAT theory.. if it doesn't KILL you (like its founder)
04:54.15billzybubi think losing weight is a real lifestyle commitment and it takes an enormous amount of will power
04:54.15JT2 days low
04:54.26Juggie[TK]D-Fender, he slipped and fell on ice.
04:54.30JT2 days carbs ok
04:54.33Juggiebut i truly do believe less carbs are better.
04:54.36ManxPower[TK]D-Fender: I have known VERY few overweight people that eat many small meals
04:54.39Juggiebut you cant remove them all together.
04:54.58JTi'm talki8ng about for dieting
04:55.07[TK]D-FenderLets say "no refined sugar". no "crap" fats. no junk like chips, canies, etc.
04:55.13billzybubwhen i was a kid i was always fat, in my twenties i starved myself and became anorexic for 5 years, now im fat again
04:55.17[TK]D-FenderNobody ever got fat on fruits & veggies.
04:55.21vader--tk do you have a 6 pack?
04:55.28JTsome saturated fats are good for you
04:55.33JTjust not too much
04:55.40billzybubchocolate covered strawberries
04:55.44ManxPowerI suppose y'all will hate me if I mention that I eat mostly fat and sugar
04:55.55billzybubfat is good for you
04:55.58JTfat is not the enemy
04:55.59JuggieManxPower, i dispise people like you, screw you and your high metabolism :)
04:56.01[TK]D-FenderTurkey, Salmon, porc are all great meats these days.  Yogurt is a great add-on.  Donane Silhoette is only 40cal per .....
04:56.01JTcarbs are :P
04:56.01billzybub:D
04:56.08[TK]D-FenderDanone*
04:56.15ManxPowerJuggie: I eat many small meals.
04:56.19[TK]D-Fendervader--, Sure.. in the back of my fridge :)
04:56.29JTnobody ever got smart on fruits and vegies
04:56.35[TK]D-Fendervader--, Though technically I'm close to beginning the real work on it :)
04:56.53JunK-Y[TK]D-Fender: now get a gf to get good foods and for cooking ur meals :P
04:56.54vader--just curious because im the same height
04:57.00billzybubyou need fat, even small amounts in your stomach makes you feel satiated.
04:57.06vader--and i was wondering what would be showing if i got down to your weight
04:57.10JTyeah, that and protein
04:57.19[TK]D-FenderJT Both are the enemy for most people, especially TOGETHER.  Hece the origin of the De Montignac diet
04:57.37[TK]D-FenderJT and is a basis for Atkis as well.
04:57.37ManxPowerIf I don't eat I turn into a total asshole, get a headache and start to sweat.
04:57.55[TK]D-Fendervader--, a 6-pack is a LOT of hard work.  thats typically the last to go as an endomorph.
04:57.58QwellI subscribe to the one meal a day philosophy :D
04:58.08billzybubive done the south beach diet, it works very well, kinda like a light version of atkins but the food prep is insanely time consuming
04:58.12[TK]D-FenderManxPower, How would we tell? ;)
04:58.14JT[TK]D-Fender: fruits and vegies?
04:58.16[TK]D-Fender(J/K)
04:58.26JunK-Ybetter: eat an apple a day and keep the doctor away.
04:58.33ManxPower[TK]D-Fender: It is a good thing I like you. 8-)
04:58.33[TK]D-Fendersorry.... can't ...resist ... temptation ;)
04:58.34JTmost diets with stupid names are retarded :P
04:58.34vader--endomorph?
04:58.46JTnothing worse than chick diets
04:58.56Juggiethe worst thing you can do is snack on bad food.
04:58.58ManxPowervader--: The Wiki is your friend
04:59.06QwellJT: what about that nympho diet..?
04:59.06Juggieso one thing i do is keep good food accessible.
04:59.11[TK]D-FenderJT indeed.... thats why I apply a lot of common sense and the known bits around, and control the portions and time for the rest.
04:59.23Juggieeg, pieces of chicken pre cooked and cut up in the fridge.
04:59.29[TK]D-Fendervader--, those who put weight on around the middle.
04:59.39JTJuggie: freshly cooked is better for you
04:59.56JTwhen you cook food with fats and then leave it, the fats become more toxic to the body
04:59.57JuggieJT, how does a piece of cooked chicken change
04:59.58[TK]D-FenderJuggie, Core in my method is making sure that food is NEVER complicated or time consuming.
05:00.01JThearder to break down
05:00.14JTthe fats go "rancid" for want of a better word
05:00.19JuggieJT, but it hasnt changed.
05:00.23Juggiethere is 0 fat in chicken
05:00.24vader--see im always eating out im never home
05:00.27JTit does change
05:00.29JTlol 0 fat
05:00.34vader--i live between my parents and my gf's parents house
05:00.38[TK]D-FenderJuggie, thats what kills so many diet plans for people.  the effort.  My foods are all so damned easy to wrok with and you don't have to eat every meal well-rounded so long as it evens out.
05:00.49vader--so i need to find food that i can eat out that is healthy
05:00.56billzybubi like peanut chews and recees peanut butter cups at about 3 in the morning
05:00.58JTvader--: meat, lots of it :)
05:01.00Juggie[TK]D-Fender, my suppers in the evening, are usually wraps.
05:01.15Juggielike a wrap with some meat, lettuce, peppers, hot peppers, and mustard.
05:01.16[TK]D-FenderSo for dinner home at 5pm I'll just doup some steak or salmon.  when I come home from the gym I'll have a smoothie.
05:01.19vader--jt by eat out i don't mean resturants
05:01.29vader--i mean like fast food places or wawa, 7-11 joints
05:01.30JTvader--: i know, it could be anywhere
05:01.44DocHollidayi made steak and pasta tonight :)
05:01.50JTmeat is good,
05:01.59JTbread rice and pasta conspire to make you fat :)
05:02.15[TK]D-FenderJuggie, Wraps can be good or bad depending.  be careful about the wrap itself, but I have considered writing that off and super stuffing them with just ssalad greens, tomatoe, etc.  Minimal blasamic dressing.
05:02.32Juggieits a spinach high fiber wrap.
05:02.34JTthat's like a lump of carbs soaked in fats that are no longer fresh (most of the time)
05:02.36[TK]D-FenderJunK-Y, Tabarnac retourn-toi donc a ton maudite poutine!
05:02.40JTbut i still eat chips every so often :P
05:02.45billzybubomg
05:02.48Juggiethere are also some low carb wraps you can get, but last time i went to the supermarket they were out
05:02.58JunK-Ypoutine hummmm
05:03.06[TK]D-FenderJT I HAVE completely cut out bread from my home....
05:03.24flendersthere's also low carb beer
05:03.27[TK]D-FenderJuggie, Best is to make it yourself....
05:03.43Juggie[TK]D-Fender, the only bread i buy is low carb
05:03.55Juggieand even so i use it very infrequentally.
05:04.04JTisn't low carb bread like saying low-bread bread? :P
05:04.09[TK]D-Fender;)
05:04.27[TK]D-FenderJT, I got a pouch of dehydrated water... now what do I do with it ;)
05:04.33JT;)
05:04.41JTit's a very fine powder isn't it?
05:04.42billzybubthis cafe near me make a mean cob wrap with romain, avacado, bacon, chicken and other goodies that escape me at the moment
05:06.31[TK]D-FenderJT, You're right..... I should mail it to the gov't for analysis... unmarked to IRS right? :)
05:06.49JTright
05:07.08k-man_[TK]D-Fender, have you tried rehydrating it?
05:09.18[TK]D-FenderAnother GREAT guilt-free filler/dessert : Jell-o fat free pudding.  30 cal, VERY filling, and a super protein/carb ratio
05:10.00[TK]D-Fenderand I bought 7lbs of bananas today :)
05:10.23DocHolliday[TK]D-Fender, im just glad i am trying to increase my weight, the decreasing thing would kill me
05:10.39[TK]D-FenderCunningPike, Don't worry.. I'm still anal-retentive ;)
05:10.46CunningPike[TK]D-Fender: lol
05:10.59k-man_jt, so how was the meeting last night?
05:11.06[TK]D-Fenderload chan_selfdeprecation.so
05:11.12JThttp://www.mindandmuscle.net/articles/jonathon_fass/carb_cycling
05:11.14k-man_jt, also what is the mailing list i should join?
05:11.24JTk-man_: still have the url i sent?
05:11.31JTk-man_: that has a link to the list
05:11.32k-man_no
05:11.33[TK]D-FenderDocHolliday, Want to lose weight?  Eat 6 meals a day.  Want to gain weight?  six meals a day.
05:11.44[TK]D-FenderDocHolliday, Lessons learned from my personal trainers.
05:12.12ManxPowerWell, walked across it at least from the car is parked to where I'm staying.
05:12.19DocHolliday[TK]D-Fender, haha
05:12.38DocHolliday[TK]D-Fender, i eat a lot of stuff.. its just not the right kind of food
05:13.14[TK]D-FenderDocHolliday, If you're that low, have they put you on Boost yet?
05:13.32ManxPowerDocHolliday: No matter how many calories you consume, you still can't gain weight.  I can understand that.
05:13.39[TK]D-FenderDocHolliday, And weight training sounds like a good idea...  you'll need a base to build on...
05:13.50ManxPowerI'll sometimes eat 2,000 calories at one meal.
05:13.51JTk-man_: http://lists.openvoip.org.au/cgi-bin/mailman/listinfo/openvoip
05:14.06billzybubheh
05:14.15billzybubi can do that with one cheese steak
05:14.18toombaloombaman thats the first time I scroll up and theres pages of non-asterisk talk
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05:14.42k-man_jt, thanks
05:14.49JTi have never had a cheese steak
05:14.54JTnot something we see over here
05:14.56billzybubreally?
05:14.57k-man_jt, so how was it anyway?
05:15.00billzybubwhere ya from?
05:15.05JTaustralia
05:15.06Qwellit's a russellb
05:15.11k-man_jt, no... does cheese and steak go well together do you think?
05:15.20billzybubwelp, if you ever co to philadelphia, check out jims
05:15.22JTk-man_: not too bad, had a good chat at the pub afterwards
05:15.23DocHolliday[TK]D-Fender, yeah i see the trainer twice a week and i go once on my own
05:15.38JTk-man_: sounds like a good carb free source of protein ;)
05:15.44[TK]D-FenderDocHolliday, What are your weight & height currently?
05:15.51billzybubshit dide, you got them funnel web spiders near where you live?
05:15.56JTyes
05:16.03billzybubomg
05:16.06k-man_jt, was the talk any good?
05:16.10[TK]D-Fenderbillzybub, We're talking Philly-chese steak here, right?
05:16.10billzybubi couldnt sleep
05:16.12JTi burn them with blowtorches
05:16.17billzybubyeah
05:16.27JTbut they're not a big worry really
05:16.31[TK]D-Fenderbillzybub, Yeah... AWESOME.  EVIL, but still awesome
05:16.36JTgenerally you have to aggrevate them to get bitten
05:16.50billzybubnah, theyll only turn your lungs into bloody jelly in about 5 minutes
05:16.53JTk-man_: yeah, interesting to see where voicetronix was going
05:16.59JTit was the ceo who did the talk
05:17.04JTtalked to him at the pub too
05:17.06JTnice guy
05:17.35k-man_ah cool
05:17.36JTbillzybub: what turns your lungs into bloody jelly?
05:17.47billzybubthe venom from the funnel web
05:17.48k-man_so what are they planning to do with their products jt?
05:20.02russellbfile: how are you, sir?
05:20.25Qwellrussellb: got any good toys from VON yet? :D
05:20.38filerussellb: sleepy... and headache-like... but meh
05:20.41filerussellb: you?!?
05:20.55russellbQwell: haven't collected any schwag, no ...
05:20.58russellbfile: very tired
05:21.02russellbvery sore
05:21.19denonyou know, I think vendors should just load bags for attendees .. and give them to you at the door
05:21.26denonbags full of everyone's stuff
05:21.31russellbagreed
05:21.33denonso you dont have to do all the work
05:21.49denonthey could just deliver them to your car for you
05:22.05russellbexcept it would be a lot more expensive
05:22.05russellbbecause people generally only want to give their stuff to people that are interested
05:22.07denonor fedex the junk right to your office
05:22.21denonrussellb: yeah, I know -- but we all pretend to be interested at the vendors with truly cool stuff
05:22.23denonand it always works
05:22.32[TK]D-FenderJT : thanks for the link... good read
05:22.44denonIve got a handful of free PC hardware to prove it
05:22.47JTnp
05:23.06russellbyup
05:23.07JThttp://www.mindandmuscle.net/articles/anssi_manninen/low_carb_diets
05:23.07JThttp://www.mindandmuscle.net/articles/twin_peak/carbohydrate_cycling
05:23.11JTthere's a couple more
05:23.19denonrussellb: well, that, and the people with good food
05:23.24Qwelldenon: oh so true...
05:23.25denonice cream stuff always goes over well on a hot day
05:23.35Qwellthe pretending to be interested part :D
05:24.01denonQwell: and then you let em scan your badge, with the phony DID and PO box
05:24.07QwellI don't ;)
05:24.09denonwell, not a phony DID - just a DID that terminates to voicemail
05:24.11denonno?
05:24.20denonI usually set up an alias that I remove after words
05:24.21denonand a temp did
05:24.33denonand I hand out my biz card to anyone I really care about
05:24.41denon"they got my info wrong on the badge .. here. use my card.. "
05:25.01QwellI somehow always end up with cards...
05:25.16denonthey sneak em into the bags
05:25.19russellbi intentionally lose cards
05:25.27Qwellwell, by people handing them to me, I mean
05:25.30Qwellnot vendors either
05:25.30DocHollidayrussellb, ouch
05:25.31denonsalespeople always think that by giving you contact info, you're going to want to call them
05:25.35russellb:-p
05:25.43denonah
05:25.47russellbnot really.
05:25.54DocHollidaydenon, i have seen someone find their own business card on the ground once, wasnt pretty
05:26.02denonhaha
05:26.07Qwellrussellb: wait until after the show to admit that ;)
05:26.11russellboh, right
05:26.18russellbit was just a joke anyway :-p
05:26.19denonIve only given my card to vendors I actually plan to do stuff with
05:26.33filehave your people talk to my people
05:26.40denonfile has people?
05:26.43denonoh, at the bakery
05:26.44Qwellbrb
05:26.45filedenon: do I ever!
05:27.07DocHollidaydenon, last weekend someone asked me for a card and i didnt have any.. used infrared to send my mobile card :P
05:27.07denonfile: I do wish you'd quit handing out irc.freenode.net/#asterisk as your "management team"
05:27.08JTDocHolliday: they had a public hissy fit?
05:27.19filedenon: pfft
05:27.20DocHollidayJT, yeah
05:27.24JTwow
05:27.30denonDocHolliday: would have been more fun if you had used infrared to upload a virus, and wipe their pda ;)
05:27.34DocHollidayinfrared is best, then they cant lose it
05:27.35JTyou'd have thought they're pretend nothing happened
05:27.57russellbi'd like to code ... but i don't think my brain can handle it
05:28.05DocHollidayyeah i wasnt very concerned, i dont give out a lot of business cards, infrared is neat though
05:28.12denonmy favorite is the people who dial your number into their sell phone . hit dial, then hang up ..
05:28.16denonand say "good, now its saved"
05:28.21denonand refer to their dialed calls logs later
05:28.31denonthe sheer ignorance is comical
05:28.54JTwhat's wrong with that? :(
05:28.57denonthey're usually the ones who ask for my number again a couple weeks later
05:29.01denonsaying they werent sure which one I was
05:29.03JTwell there's that
05:29.18JTi only do it when i'm going to need a number as a once off or something
05:29.19denonyou occasionally get wrong number calls from those people
05:29.23denonwhich is funny, you know what they're doing
05:29.39denonJT: I spose.. but doesnt your phone let you hit save after you dialed the number?
05:29.59denonyou could put 2 letters in, in like 2 seconds, and probably remember it a lot better
05:30.00JTi can, but i don't want to save randoms i am going to call from my mobile once
05:30.13denonspose
05:30.20denondidn't mean to insult my friends down under :)
05:30.23filerussellb: go to sleep!
05:30.29JTyou're right though
05:30.31JTthere is a line
05:30.32DocHollidaydenon, im a big fan of infrared.. most people dont even know they have it
05:30.45DocHollidaybut once i give it to them, rarely have ever 'lost it'
05:30.48denonDocHolliday: unfortunately, lots of newer stuff doesnt
05:30.55JTif you plan to call a number more than once ever from your mobile, store it or memorise it
05:30.58denondarn bluetooth
05:31.02JTi memorise heaps of numbers
05:31.21denonI remember paying like $100 extra on my first tdma phone to get infrared
05:31.37DocHollidaydenon, i dont know why it never took off
05:31.39denonand being sorely disappointed to find out my laptop's infrared chipset was screwy, and didnt communicate well
05:31.45DocHollidayhaha
05:31.48`p4r14hwinders mobile with exchange is good for work :)
05:31.50denonDocHolliday: too slow for anything useful :(
05:32.02denonthat new linear optical deal is kinda neat
05:32.04DocHollidayi take the attachments i get on my smartphone and use infrared to upload them to my laptop on the go
05:32.09denoninfrared idea, but much faster
05:32.19`p4r14hDocHolliday: what kind of smartphone?
05:32.20denonand recognized people/etc
05:32.22*** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net)
05:32.23denonwithout knowing what you're doing :)
05:32.30denoner recognizes
05:32.43DocHolliday`p4r14h Treo 650
05:32.44denonDocHolliday: yeah, I do the same with bluetooth
05:32.48[TK]D-Fenderok, way late... I'm off... later all
05:32.52denonbuilt into laptop, and phone, of course
05:32.55`p4r14hDocHolliday: that is palm isn't it?
05:32.57denongoes pretty quick
05:33.59DocHolliday`p4r14h correct, and btw why not pick a nick someone can pronounce?
05:34.20`p4r14hDocHolliday: `p TAB is all u need
05:34.25denonor at least something in binary
05:34.29`p4r14h:D
05:34.41j3j3j3j3jit's random nick day
05:34.51j3j3j3j3jeep!
05:34.52russellbwow, what random luck
05:34.56DocHollidaydenon, yeah i didnt select the bluetooth option on my laptop, didnt think i'd need it
05:35.07denoni'm such a n00b
05:35.33russellbhook, line and sinker ..
05:35.39russellb:)
05:35.45`p4r14hwhere would i set the DTMF tone default? my phones numbers arn't exactly in sync with say the voicemail system
05:35.47JThaha
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05:36.10denonsorry :)
05:36.11denoncouldnt resist
05:36.15russellbhehe
05:36.31denonfunny that the random characters ended up spelling your name
05:36.36denonthe odds are astronomical
05:36.43russellbcrazy
05:37.41denonif I must use your stupid service, at LEAST leave my default funding source as credit card
05:47.16billzybubPOOP!
05:47.22russellbum.
05:47.27russellbno?
05:48.02billzybubi just spent an hour installing and configuring all the pre reqs for freePBX to have the installer script tell me it doesnt support asterisk 1.4 :(
05:48.21JTheh what did i say earlier :P
05:48.24JT~freepbx
05:48.36jboti guess freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
05:48.39billzybubi need a drink
05:48.55DocHollidaywhy use freepbx anyway?
05:49.42JunK-Ycause customers ask it?
05:51.31denonyou actually have users asking for freepbx?
05:51.40denonsounds like a customer education issue
05:52.00billzybubyeah try telling a customer to use vi.
05:52.10*** join/#asterisk wwq222 (n=chatzill@c-71-231-5-6.hsd1.or.comcast.net)
05:52.10billzybubthan try living on bread and water.
05:52.16denonI wouldn't equate asterisk to vi
05:52.24JTcustomers shouldn't be making PBXes from scratch
05:52.26denonit's not that complicated, even for a novice
05:52.33JTthat's what the telephony guy is for
05:52.38denonnod
05:52.54JTi agree an interface to do simple things would be useful
05:52.57denondont expect to run your business on a system you dont understand, and don't have anyone to manage
05:53.01billzybubi would use it as a selling point
05:53.02JTlike changing the name of a station
05:53.06JTand some voicemail stuff
05:53.14wwq222Hi i'm trying to implement multiple calls (one call right after another) in Asterisk - is there some way I can do that aside from having some outside app do the scheduling?  I tried the 'g' option in the Dial option (to try to call Dial multiple times), but it doesn't seem to get past the 1st Dial
05:53.35billzybubim wrecked
05:54.13russellbbillzybub: try the new asterisk gui
05:54.18russellbit takes like 2 seconds to install
05:54.23denonyou could use one of the few thousand guis out there
05:54.31denonyeah. the official one even
05:54.34billzybubrussellb, whats it called?
05:54.41billzybubdidnt know there was one
05:54.53russellbbillzybub: yeah, you can see screenshots at asterisknow.org
05:55.08russellbAsteriskNOW is a full linux-distro that includes it, but you can just install it on a regular 1.4 install
05:55.52russellb<PROTECTED>
05:55.52russellb<PROTECTED>
05:55.52billzybubdo i have to reinstall my OS for it or is it just a layover?
05:55.53russellbsorry, copied that from a topic in another channel ...
05:55.53russellbeither one
05:55.53CrashHDanyone know how to do hints on parked calls when using app_valetparking?
05:55.59`p4r14hi use asterisk now, its alot better than trixbox+freepbx
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05:56.08russellb`p4r14h: i am very glad to hear that
05:56.23russellbit's basically brand new
05:56.31russellbbut we've got some folks working very hard on it
05:56.42`p4r14hits still just a complex dialplan, but its all developed by one company
05:56.51JerJerAsteriskNOW kicks ass
05:57.07russellb`p4r14h: the dialplan that the asterisknow gui generates is *nothing* compared to what freepbx does
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05:57.17Qwell^^ and that's a good thin
05:57.18Qwellg
05:57.25JerJermore than a good thing
05:57.35`p4r14hrussellb: you know anything about the voicemail sending out plain text emails, rather than one with an attachment?
05:57.37denon"what freepbx does" == generate more spaghetti than all of italy
05:57.39billzybubrussellb, that is extremely helpful, i wish you were here 2 hours ago when i asked and got nothing but a bunch of smart ass answers
05:57.55Qwell`p4r14h: That should be fixed if you update asterisk from conary or whatever
05:57.58russellb`p4r14h: i fixed that a while ago, it will be fixed in the next beta
05:58.05russellbbillzybub: i'm sorry :(
05:58.11`p4r14hhow can i fix it right now?
05:58.13billzybubnot your fault
05:58.34russellb`p4r14h: figure out how to update asterisk using conary ...
05:58.38`p4r14hplease inform me :D
05:58.48russellbi don't know how to use the rpath tools :(
05:59.06JerJerrussellb:  yeah they are pretty crazy
05:59.09`p4r14hi updated using rpath....
06:00.56russellbJerJer: they are supposed to come in and train all of us at some point ...
06:00.56russellbi just haven't spent the time to learn it on my own
06:00.56JerJerReal Soon Now(tm)
06:00.56russellbright
06:02.56russellbwe have some "AsteriskNOW for dummies" books here to give out at the show, heh
06:02.58russellbpretty cool
06:03.25*** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net)
06:03.50Juggierussellb, i know you have an iaxy or x100p in your office you want to send me.
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06:04.03JTthey're actually For Dummies books?
06:04.09russellbyeah
06:04.20russellbit's not very big, though
06:04.35denonpage 1, paragrph 1... download this file, quit your whinin'
06:05.04Juggieruss, whats the linux util to list the functions within a binary and the librarys they are linked against.
06:05.07Qwellrussellb: who is that author?  never seen the name
06:05.19Juggieor qwell, feel free to answer as well.
06:05.19QwellJuggie: ldd?  nm?
06:05.22russellbQwell: i have absolutely no idea.
06:06.56JuggieQwell, ldd shows the libs, but i'm looking for functions
06:07.02Qwellnm?
06:07.11Juggiesomething in this code is linking aginst GLIBC_PRIVATE
06:07.15Juggieand i'm trying to see what it is
06:07.37wwq222Does anyone know how I can make a call, have it hang up after x seconds, and then automatically call someone else?
06:08.19denonhmm, that reminds me .. asterisk needs an user-exposed threadpool :)
06:08.30russellbdenon: what?
06:08.30denoner a
06:08.55denonwell .. think of spool items .. but launched from extensions.conf :)
06:09.05denonwith the management properties of a thread pool
06:09.09denonpriorities, delays, etc
06:09.30russellbSystem(echo "asdfadfads" > /var/spool/asterisk/outgoing/foo.call)
06:09.37russellb:-p
06:09.43denonyou're so not getting this :)
06:09.48russellbnope
06:09.50russellbi am very tired
06:09.54denonditto
06:10.01Juggienmmm... Qwell, i see i see... now how to i know which one of those is GLIBC_PRIVATE hrmf.
06:10.01denonbut it's still a good idea :)
06:10.04denonanywho, I'm outta here
06:10.06russellbtry me again another day :-p
06:10.08Qwellbed
06:10.08russellbg'night
06:10.10denonsure
06:10.10denoncatch ya later
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06:16.49`p4r14hsudo conray update asterisk almost worked, minus /var/lib/asterisk/sound/silence was owned by another package or something.....
06:16.57`p4r14h*conary
06:19.12russellbmaybe an asterisk-sounds package?
06:19.42CrashHDanyone know about the app_valetparking module?
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06:20.27billzybubrussel, after the make do i need to do a reload from the cli to bring the httpd up?
06:20.32`p4r14hit is, im trying conary erase asterisk-sounds, then ill update asterisk, then i will reinstall sounds
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06:23.21russellb`p4r14h: ah, cool
06:23.34russellbbillzybub: yeah, assuming you made that necessary config changes
06:23.50russellbyou can run "make checkconfig" in the gui directory to make sure you set up the config correctly
06:24.08russellbbut I think "make install" of the gui does that automatically
06:25.56billzybubyeah, ran checkconfig 6 times to get it right ;)
06:26.03billzybubthis is awesome
06:26.23billzybubumma put this on my laptop and go hit the strip
06:26.35russellbcool
06:26.40codestr0mcan someone point me to a best practices for setting up a pbx.. I'm mostly interested in 1) which audio format that * supports will give the highest quality sound and small things like should I turn my sound files into a bigger file or just programmically string them together.. thanks
06:26.48JThit the strip?
06:27.03JTcodestr0m: g.711
06:27.28JTstringing the files together.. that depends on usage scenarios really
06:27.33billzybubjt: umma sell the shit outta this
06:27.45JTi see
06:28.31`p4r14hman conary has to d/l the file everytime =\
06:28.32`p4r14hvmplayer -> winxp -> pptp tunnell -> asterisk box, pretty lame i wish pptp was working on my box :(
06:28.37billzybubcan anyone recomend a wifi sip fone?
06:29.00JTno
06:29.06JTthey all suck to be honest
06:29.16billzybub:(
06:29.35JTuse a normal digital cordless phone with an FXS port
06:29.45billzybubjust like all the asterisk gui's i suppose?
06:30.50JTwhat?
06:31.16russellbbillzybub: hey now, be nice :-p
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06:33.02billzybubthis is some slick webwork
06:33.02JerJerbillzybub:  find one that supports WMM
06:33.04`p4r14hfor future reference, 'conary update asterisk' is the command to update, but if there are packages that it will conflict with uninstall using 'conary erase PACKAGE' then run 'conary update asterisk' again. after asterisk updates, reinstall the packages with 'conary emerge PACKAGE'
06:33.30russellb`p4r14h: cool, thanks
06:33.36JerJerput that on jbot
06:33.38JerJerperhaps
06:34.19billzybubhey russ, should the users iver defined in my extensions.conf show up on the web thingy?
06:34.30russellbprobably not ....
06:34.33`p4r14hthat was pretty straight forward, im impressed :D
06:34.43CrashHDanyone recommend a better parking solution for asterisk?
06:34.49CrashHDmulti parking lots?
06:35.13billzybubhrm, found the advanced tab but its grey'd out
06:35.23billzybubi only have sip users defined
06:35.37billzybubbut i cant click on the sip option
06:36.48`p4r14hrepeat question: anyone know what config file to change the default DTMF tones? my hardphones number key tones aren
06:36.57`p4r14h't working with voicemail system
06:41.37`p4r14hholy hell asterisk-sounds has a butt load of sound archives.....
06:42.09JTyep
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06:46.14`p4r14hSTOP IT NOW!! I WANT TO SLEEP!!! =\
06:49.06tzafrirtt-monkeys
06:49.16tzafrirtt-monkeys
06:49.17JTtt-weasels
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06:59.55billzybubanyone play with zfone?
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07:09.36tutt9876hi, anyone has an experiecne with 1.4.1: my peers are not answering
07:09.52billzybubi installed it yesterday :D
07:10.06tutt9876did you have any problem?
07:10.06billzybubas my first asterisk experiance
07:10.25billzybuba little bit but i worked wthrough it with some web tute
07:10.49billzybubim pure sip over here no fxs/fxo
07:11.00tutt9876do you know if there are special config to use ?
07:11.34tutt9876my peers are not answering ?
07:12.11billzybubwhat did you upgrade from?
07:12.27tutt9876from 1.2
07:12.41tutt9876I just compile over the 1.2
07:12.50billzybubi defined my sip users in sip.conf then defined them in extensions.conf
07:13.25tutt9876I just copy my config files from 1.2 that are sworking with it
07:14.13billzybubi would assume that your peers are no longer working do to some changes in syntax
07:14.27*** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl)
07:14.32tutt9876changes?
07:15.33billzybubare your peers sip?
07:15.57tutt9876yes
07:16.29billzybubare they showing up in the CLI
07:17.46tutt9876I found unmonitored: don't know is it's ok in fact
07:17.53tutt9876if
07:18.43tutt9876the command sip show peers doesn't not really intersting information
07:18.54tutt9876not really give
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07:37.49uwegood morning :)
07:40.22uweim trying to chage the queue ring strategy, and apparently anything other than ring all causes trouble with delays, the caller stays in the queue and only one agent is ringing at a time, what can be done? or do i have to hack app_queue.c ?
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07:58.11tengulrehi,all
08:08.04SwKuwe: what version of asterisk? there is a option you can set in later versions for that
08:13.58*** join/#asterisk Powerkill (n=Powerkil@84.205.154.179)
08:14.01Powerkillhi
08:14.06rwardhi
08:14.56PowerkillI upgrade to 1.2.17 and now I've plenty of error like this Mar 22 09:14:11 WARNING[17016]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission5898a4-c2010151-13c4-4a40ca-220d1704-3be3@sip.annatel.net                                                                                              for seqno 2 (Critical Response)
08:14.56PowerkillMar 22 09:14:11 WARNING[17016]: chan_sip.c:1245 retrans_pkt: Hanging up call 5898a4-c2010151-13c4-4a40ca-220d1704-3be3@sip.annatel.net - no reply to our critica                                                                                             l packet.
08:15.05Powerkillany idea ?
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08:19.43uweSwK: its 1.2.16 , i noticed there is autofill option is 1.4 , but i wish i can do it on 1.2.x
08:20.32PowerkillI found that this is the problem http://bugs.digium.com/view.php?id=5215
08:21.39SwKuwe: the original patch for that was 1.2 and should be on the tracker from 'twisted'
08:22.23sashiondoes autofill actually work and what is it's purpose ?
08:23.06SwKhttp://bugs.digium.com/view.php?id=5577
08:23.25SwKautofill is an app_queue patch that speeds up app_queue in delivering calls
08:23.48SwKbasically if there are 3 calls in queue w/out autofill it only handles delivering 1 call at a time to agents
08:24.22SwK(so in something like roundrobbin it would only ring 1 call from the queue at any time... the other calls would have to wait until the first call is answered
08:25.38SwKautofill w/ combines w/ like round robbin will send calls to agents as long as there are avail agents w/out calls loosing their place in queue if an agent doesnt pick up the call
08:26.00Powerkillplease someone know about these error ? Mar 22 09:14:11 WARNING[17016]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission 5898a4-c2010151-13c4-4a40ca-220d1704-3be3@sip.annatel.net for seqno 2 (Critical Response)
08:26.00PowerkillMar 22 09:14:11 WARNING[17016]: chan_sip.c:1245 retrans_pkt: Hanging up call 5898a4-c2010151-13c4-4a40ca-220d1704-3be3@sip.annatel.net - no reply to our critical packet.
08:26.39SwK(ie: 5 calls in queue, 3 avail agents, round robbin ring strategy, calls 1,2,3 ring the agents, calls 4,5 just wait, the agent that got call 2 didnt answer it so it stays at the head of the line and goes to the next avail agent
08:27.23SwKpowerkill the far end isnt answering thats what that means
08:28.01SwKand if you want to show people messages like that use the pastebin instead of spamming the channel
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08:28.06SwK~pastebin
08:28.17jboti guess pastebin is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well
08:28.17SwKpastebin
08:28.17SwK!pastebin
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08:29.07Powerkillok
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08:30.17*** join/#asterisk sashion (n=djbdsf@dsl-244-58-123.telkomadsl.co.za)
08:32.14sashionSwK: basically at a load point (where I have about 8 - 15) calls in queue, the system starts seeing logged in and available agents (agent show online) as busy (by doing a queue show queuename) and qpp_queue never passes a call until I do a reload on app_queue
08:33.51codestr0mI downloaded the ulaw sounds from (http://www.astlinux.org) and I'm getting this perm denied message (WARNING[6263]: file.c:804 ast_streamfile: Unable to open demo-echotest (format 0x4 (ulaw)): Permission denied) on the ulaw files, but not the sln files.. I double checked the perms.. and set modules autoload yes.. how can I trace this further.. I see ulaw under show file formats
08:34.18gfraysse<PROTECTED>
08:34.52tenecodestr0m: check the permissions on the files
08:35.10codestr0mtene: yeah. I did that first and it's asterisk:asterisk.. it's not that
08:35.22teneHuh.
08:35.54rwardcodestr0m: try restart?
08:35.55sashioncodestr0m: does ls /var/lib/asterisk/sounds/demo-echotest.* show anything ?
08:36.17codestr0m<PROTECTED>
08:36.17codestr0m<PROTECTED>
08:36.40Powerkillif I back revert to 1.2.13 everything is working again
08:36.55codestr0m644 perms and asterisk:asterisk.. yeah. I checked all this.
08:38.04codestr0mcould it be incorrect file extension? core show file formats shows == ulaw       au         au
08:38.26sashioncodestr0m: rename that file to .wav
08:40.13codestr0msame stupid message. if I change it to .au or .wav.. it's not recognizing ulaw.. (wth?)
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08:43.21Turt|eHi, i just made an queue, and the youarenext is only played on timeouts, i use 1.4.1. What might i overlook ?
08:45.46codestr0msashion: I started * as root and it worked.. what other file perms could be involved?
08:48.48JTumm
08:48.52JTcodestr0m:
08:49.05JTls -la <filename>
08:49.11JTeither message me output
08:49.14JTor use pastebin
08:49.33JTi will tell you why it's not working :)
08:49.52codestr0mls -lash demo-echotest.ulaw
08:49.53codestr0m156K -rw-r--r-- 1 asterisk asterisk 153K Mar 22 10:25 demo-echotest.ulaw
08:50.43codestr0mdoes it have to be +x ?
08:50.47sashioncodestr0m:  easiest way is to log in as asterisk under a shell and then traverse /var/lib/asterisk/sounds and see where asteirsk can't go any furthuer
08:50.47tzafrirno
08:50.54sashionif you understand what I mean :)
08:51.23tzafrircan you play any other sound file?
08:51.40codestr0msashion: I was one step ahead and yes. when I copied the dir over.. didn't put 755 perms on the dir
08:51.48JTcodestr0m: now show ls -la on a file that works fine with asterisk running as user asterisk
08:52.03JTseems quite weird
08:52.26*** join/#asterisk Avochelm (n=damien@144.136.166.42)
08:52.33codestr0mJT: yeah. my umask as root when I copied the dir over.. I'm not thinking. it's too early :P
08:54.17JTcodestr0m: wait, you've fixed the prob?
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08:55.24codestr0mJT: yeah.. sounds dir wasn't 0755. thanks.. now I just need to figure out the best way to convert my .wav to sln or ulaw with sox :)
08:55.37JTnice
08:55.50JTdid you sample the wav at 16bits?
08:56.28codestr0mJT: I had it done in a professional studio.. I have no idea, but I think it's 16bit 44100 wav files..
08:57.02JTah ok
08:57.11JTyou need 16bits at least for best quality
08:57.27JTas ulaw or alaw gives you more dynamic range than 8 bits
08:57.29codestr0mI may be wrong. (AUDIO: 48000 Hz, 2 ch, s16le, 768.0 kbit/50.00% (ratio: 96000->192000))
08:57.45JTyeah that spec doesn't specify
08:57.47JTgood enough!
08:57.51JT48kHz, heh
08:58.13*** join/#asterisk hermuli (n=Eladamri@a88-112-255-26.elisa-laajakaista.fi)
08:58.19JTthey should've recorded in mono though
08:58.20codestr0mthat good enough to start with you think?
08:58.23JTstereo is useless
08:59.02codestr0mI can always just drop a channel/track I suppose? I'm still trying to figure out audicity
08:59.28hermulihello everyone. can anyone help with this: i got debian etch and get this from trying to make asterisk-addons configure: error: no acceptable C compiler found in $PATH
08:59.31JThopefully the tracks are the same
08:59.36hermuliam not that good with linux anyway
08:59.40JThermuli: install gcc
09:00.04hermuliaww... too obvious
09:00.08JTcodestr0m: what country are you in?
09:00.09codestr0mJT: I was wrong. that was a format I exported as.. the orignal wav was (AUDIO: 44100 Hz, 2 ch, s16le, 1411.2 kbit/100.00% (ratio: 176400->176400))
09:00.19JThmm ok
09:00.24*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
09:00.35codestr0mJT: I'm currently in eastern europe, but was born in the US. family is swedish and I travel far too much
09:00.54JTok, better question, what country will the asterisk box be serving?
09:01.09hermulithanks JT
09:01.16codestr0mJT: I have 3 pops. 1 SJC 1 SEA 1 AMS
09:01.16JTnp
09:01.31JTdude those don't mean much to me :)
09:01.34JTi can guess
09:01.46JTsan jose california?
09:01.58JTamsterdam something something
09:01.59codestr0msan jose ca, seattle, wa and amsterdam
09:02.23JTso the first two are ulaw, the last one is alaw
09:02.39codestr0mnope. I'm not doing any zap to sip
09:02.47*** join/#asterisk sashion (i=synergy@41.208.192.24)
09:02.56JTit will never ever touch the pstn?
09:03.25codestr0mJT: not unless I start putting a box in each country and use SS7
09:03.49puzzledmorning
09:03.50JTprivate voice network?
09:05.13codestr0mJT: sorta.. zap, sip is high quality, but there are providers (*mci*) which can deliver quality over IP
09:05.45JTheh
09:05.56sashioncodestr0m: good luck with ss7 and asterisk :)
09:06.16codestr0msashion: lol. been there. it's not ready imho.
09:06.57sashioncodestr0m: likewise. Had an interlink to a Nokia switch, but had to drop to PRI as libss7 kept crashing...
09:07.18*** join/#asterisk jm|laptop (n=jm@sentry.flags.co.uk)
09:07.36codestr0msashion: well. it can work, but you have to patch it and really overbuild the machine which it runs on
09:07.59codestr0mIt's been about a year since I fought with it at all
09:08.41*** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl)
09:09.48sashioncodestr0m: yeah I noticed :) had to do a couple patches for feature like remote retreival, as well as additional optional params...
09:10.59codestr0mwell. it may be my lay impression, but the fact SS7 always requires there to be communication over the line seemed like it created much more load than pri.. not so say I know either protocol very well, but..
09:11.09codestr0m1000000000 little packets all the time == bad
09:12.17sashioncodestr0m: very true... guess ss7 is just a little paranoid about loosing it's peer :)
09:12.50sashionstarting to get irritated with asterisk's queuing....
09:12.50JTcant imagine why
09:12.54codestr0mwell. I wish the people who engineered bgp would have done ss7 :P
09:13.25JTthe Internet is less reliable than most tdm networks, i know which i'd rather
09:14.04codestr0mthe internet is only less reliable because the threshold set by ARIN and friends lets asshats route their own traffic
09:14.28JTi think there's a few more reasons than that...
09:15.44codestr0mJT: ok.. + the fact that most bandwidth is being used for porn and media.. :P
09:16.05JTtelcos have just sliiightly better hardware, staff and processes
09:17.05codestr0mgotta run guys.. thank a lot!
09:19.27*** join/#asterisk dngcy2k (n=dngcy2k@mail.netregy.com)
09:19.52dngcy2kanyone can help?
09:20.49dngcy2kcan Windows AD as user directory services for asterisk
09:23.10*** join/#asterisk RoyK (n=roy@217-175-152.100710.adsl.tele2.no)
09:23.11stimpiedngcy2k, asterisk can get the users from a ldap server
09:23.33dngcy2kwhat bout Windows Active directory?
09:23.57sbingnerWindows AD == LDAP
09:24.05dngcy2kic..
09:24.33stimpiedngcy2k, google for 'asterisk realtime' and 'asterisk realtime ldap'
09:25.04dngcy2kcan we use a single extension/user sign on on a distributed asterisk system
09:25.10dngcy2kthanks stimpie
09:25.11*** part/#asterisk codestr0m (n=asura@ns1.netsyncro.com)
09:26.02stimpiedngcy2k,  single sign on isnt that hard but with a distributed asterisk several other problems arise.
09:27.07dngcy2kmeaning it's not recommended to have single signon on a distributed system
09:28.32stimpieall depends on what you try too accomplish
09:29.30dngcy2kbasically we have 13 branches nationwide and we intend to implement asterisk solution over the branches
09:29.57dngcy2kwe're in the midst of designing the network topology and toying with the idea whether to centralised or decentralised the system
09:31.11stimpiewell one of the issues with single sing on with 13 branches, how do you know where an extension is?
09:32.41dngcy2kthat's what bugging us now
09:34.19*** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com)
09:34.53DrukenLPYhow many users per office?
09:36.11dngcy2kapprx. 200 users per branch
09:36.28DrukenLPYshit, your talking one huge fucken system...
09:36.38DrukenLPYdecentralized is the way to go
09:36.39dngcy2kyes
09:38.11stimpieyou could use sip phones with a centralized sip server for mobile users
09:39.57dngcy2kmmmmm
09:42.21dngcy2kany idea on asterisk HA solution
09:42.38DrukenLPYHA ?
09:42.44dngcy2kHigh availability
09:42.58rwardmaybe a SER server infront of the asterisk?
09:43.47DrukenLPYwhen your talking 2000+ seats, and 13+ sites.. you really gotta do your homework
09:44.10dngcy2kyes that's what we're doing now
09:44.22dngcy2kinformation and solution gathering
09:44.58DrukenLPYdo you require extensions or will be just incoming or outgoing calls only ie call center status
09:46.39dngcy2kFull functional PBX with incoming/outgoing and extension
09:47.02DrukenLPYsounds like loads of fun
09:47.35DrukenLPY5 digit extensions and server exchanges all over the place...
09:49.42*** join/#asterisk AlienPenguin (n=Miranda@ip-145-151.sn2.eutelia.it)
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09:50.57Turt|ehi, does someone know a way to make an dialplan so a user could pull an single call from an queue? ex dialing *99 and the first caller in the queue would be pulled from the queue ?
09:52.02uweSwK[Work]: thank  you very very much
09:53.00JTDrukenLPY: i can't see why you'll need more than 4 digit extensions
09:53.08JTunless the growth rate is massive
09:53.54AlienPenguincan anyone give me a hint on why if i put in features.conf the blindtrasfer => #1 does not work while if i just put # it does? (it seems not to wait for the following digits)
09:53.54*** join/#asterisk badcfe (n=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr)
09:54.31X-RobAlienPenguin, probably because you're using asterisk 1.2.12 or 1.2.13? I think that bug was fixed in .14
09:55.17AlienPenguinno i am on 1.4 right now
09:55.25AlienPenguinand i think it worked on 1.2.15
09:55.57*** join/#asterisk SwK (n=Silik0nJ@12-214-191-109.client.mchsi.com)
09:57.47dngcy2kanyone know what is flexible extension logic
09:58.46DrukenLPYJT: i was figuring 5 digit extensions... XX site and XXX extension
09:59.49JTi guess that's one way to do it
09:59.52dngcy2kfor interbranch prefix?
10:00.08JTyou could fit it in 4 digits with more complicated pattern matching
10:01.07DrukenLPYJT: yeah, but your users have to be able to figure out the pattern matching :) and well, i'm lazy hehehe
10:03.33JTwell you could have little sheets howing that 2500-2700 is in City A, etc
10:03.37JTshowing
10:05.24*** join/#asterisk [shodan] (n=shodan@ip211.96-113-216.pppoe1.joliette.intermonde.net)
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10:11.14DrukenLPYJT: you could...
10:11.25*** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org)
10:12.04JTif you had some CTI, they could select the person they're dialling from an address book on screen :)
10:13.05Mavviefunny logging:
10:13.11Mavvie<PROTECTED>
10:13.23Mavviethat should have been SIP/ccm-subscriber:
10:13.31Mavviehan_sip.c: Peer 'ccm-subscriber' is now UNREACHABLE.
10:16.54DrukenLPYJT: yeah... that would be nice eh?
10:20.23JTyep
10:22.42*** join/#asterisk pif (n=ldm@zenon.apartia.fr)
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10:32.14pifhi, does 1.4.x include misdn ?
10:33.26tzafriryes. And latest 1.2 as well
10:33.40sashionhi... can anyone tell me why there is a difference between agent show online and queue show? Attached pastebin: http://pastebin.ca/406343
10:33.44ParaNoirHey, Anybody succeeded with connecting Swyx to Asterisk via SIP?
10:34.03AlienPenguinpif: using 1.4.1 with misdn right now
10:34.16piftzafrir: but I don't see it in the debian package
10:34.48tzafrirI'm having problems getting the misdn lib in place
10:34.55ParaNoirpif: don't you have asterisk-chan-misdn?
10:35.08tzafrirThat package is currently rather broken
10:35.10pifthat's an outdated package
10:35.15ParaNoirok
10:35.25ParaNoirmy server crashed with that one ;)
10:35.39tzafrirI never really needed misdn...
10:35.51piftzafrir: in any case I appreciate your work with the * debs
10:36.57JTthere's always bristuff :D
10:37.05pifyuck!
10:37.13JTerr, excuse me?
10:37.18JTit's far better than misdn
10:37.23pifno way
10:37.24ParaNoir;0
10:37.29JTyes, it is actually
10:37.37pifI ran away from bristuff screaming
10:37.41ParaNoirlol
10:37.42*** join/#asterisk zotz (n=zotz@24.244.163.157)
10:37.45pifafter 1 year of various bugs
10:37.48ParaNoiri was trying 5 days to get it working....
10:37.54JTand went to misdn? you're crazy
10:37.57JTmisdn has way more bugs
10:37.59ParaNoiruntill i noticed they cut the line ;)
10:38.04pifmisdn has been a dream for me since then
10:38.13JTmaybe you have a very basic setup
10:39.30pifpretty basic: making and receiving phone calls
10:41.42JTyou've got to be more specific that, instead of being a smartarse
10:41.47JTlike what modes are you using
10:41.50JTwhat cards
10:41.54JThow many ports
10:41.59JTetc
10:42.21pifokay, 5 different sites, using 4BRI junghanns cards
10:42.55JTconnecting to only telco?
10:43.00pifvarious setups from 1 to 4 ports, some with idsn-dect phones attached
10:43.32JTmisdn's NT mode support is a joke
10:43.38JTand current bristuff versions are very stable
10:44.10pifI haven't stressed NT mode too much
10:44.24pifbasic use works fine
10:44.29JTmisdn doesn't even support group dial on nt ports
10:45.19pifNT ports are meant to ring a specific phone, not a group
10:45.30pifuse app_queue
10:45.39JTno, that's the stupid one-tracked thinking that misdn authors use
10:45.49JTno, i'd rather do it properly
10:46.24JTbristuff supports all zap grouping methods irrespective of TE or NT mode
10:46.27pifwell, say hi to kapejod (if you can find him)
10:46.38JTare you a misdn author?
10:46.41JTyou seem bitter
10:46.44pifnope
10:47.02JTi'm talking about what the situation is like these days, not years ago
10:47.11JTat the moment, bristuff is leading
10:47.19JTespecially if you use NT mode in any real capacity
10:47.26pifI just barely lost my job thanks to bristuff
10:48.15JTthat may be why you are bitter
10:48.31JTnever rely on free software unless it's been tested ;)
10:48.58*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)
10:49.57ParaNoirhey, is absolutely nobody familiar with Swyx?
10:50.06uweis there any good reason why on ringall sometimes the queue simply doesnt take calls frequently, it acts as if its another strategy?
10:53.51*** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com)
10:54.12piftzafrir: what kind of problems you had with the misdn libs?
10:54.16pifwhat version you using?
10:55.50JTas far as i'm concerned, all BRI solutions are fairly beta
10:56.00JTbut current bristuff is least beta
10:56.06JTespecially when doing anything complicated
10:57.00*** join/#asterisk k31th (n=keith@87.117.194.66)
10:57.02k31thIs there a way of having 1 phone book db with asterisk? so they is apears on the phones? (I have Linksys SPA942's)
10:57.43*** join/#asterisk basilisk (n=jerry@192.18.43.225)
11:00.11zeeeshi made 2 peers ... like 100 and 200 registered at server ... 100 can make calls to 200 but 200 is not able to call at 100 .. both extensions are same ... like 100 extensions is . exten => 200,1,Dial(SIP200@200) AND .. 200 extensions is exten => 100,1,Dial(SIP100@100) ... then y  200 is not able to make call ????
11:00.39*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
11:01.53JTzeeesh: i'm not terribly sure, but that dial command is wrong
11:02.08JTDial(SIP/200/200)
11:04.17zeeeshis it also works .. somewhere its works .. i checked somewhere its not .. exten => 200,1,Dial(SIP/200) ?
11:04.49*** join/#asterisk MaartenB_ (n=Maarten@h8441243087.dsl.speedlinq.nl)
11:05.10JTyes, that would work to dial sip entry 200 without specifying a number at that entry
11:05.45*** join/#asterisk nutcase (n=nutcase@i59F556B5.versanet.de)
11:05.48blitzrageare you looking for the format Dial(SIP/200@proxy)   ?
11:06.20*** join/#asterisk donkilla (n=rogers@196.200.26.174)
11:06.34donkillaHi
11:06.59blitzragey0
11:07.28donkillaI bought 2 Digum TDM04B and am having problem with the configuration
11:08.29donkillaI got an error "No channel type registered for 'zap' whenever i try to use a Zap Channel
11:08.42sashiondonkilla: did you load the modules ?
11:09.01donkillamodprobe wctdm ?
11:09.23sashiondonkilla: yep... hope you did a modprobe zaptel first ?
11:09.44donkillaYes
11:09.53blitzragelsmod | grep zaptel
11:09.56sashiondid you load the chan_zap.so module in asterisk ?
11:10.14donkillaHow do i load chan_zap.so?
11:10.25sashionload chan_zap.so in asterisk CLI
11:10.31blitzrage*CLI> load chan_zap.so
11:10.41sashionor if you using 1.4.x... simply type module load chan_zap.so
11:11.02donkillaI'm using 1.4
11:11.11blitzrageyou compiled and installed in this order right?    libpri, zaptel, asterisk
11:11.26blitzragechan_zap.so won't build without zaptel being installed before you compile Asterisk
11:11.33donkillayes
11:12.04sashiondonkilla: pastebin a ztcfg -vvvv
11:12.10donkillaNo i did Zaptel Libpri then Asterisk
11:12.58sashionyou don't really need libpri unless you have a TE4XXP card...
11:13.02uwehmm ... ok ... is this the way ringall works ? scenario : callers A,B,C ... extensioins X,Y,Z , caller A calls, XYZ ring, B in que and so is C, A . X answers A, B rings on Y and Z, Y picks up ... C stays waiting and Z doesnt ring
11:13.05sashionor planning to use a PRI interface
11:13.25JTsashion: don't forget the TE110P and others
11:13.30donkillaYes am planning to. I got a T1 card
11:13.44JTdonkilla: libpri, then zaptel, then asterisk
11:13.52tzafrirpif, basically: not sure what to use
11:14.07donkillaLooks like i will have to follow that order
11:14.20JTdonkilla: it's logically really ;_
11:14.22JT;)
11:14.25tzafrirpif, if you have any useful comments regarding misdn in the debs, I'd appreciate them
11:14.28JTzaptel depends on libpri
11:14.33JTasterisk on zaptel
11:14.54sashionand voip depends on asterisk :)
11:15.03sashionwell a portion of it anyway :)
11:15.08*** join/#asterisk cweiske (n=cweiske@dslb-088-074-128-101.pools.arcor-ip.net)
11:16.21donkillaWhen i do module load chan_zap.so i get "Error loading module 'chan_zap.so:usr/lib/asterisk/modules/chan_zap.so: cannot open shared object file: No such file or directory
11:16.36cweiskeHello. An misdn question: is someone using a sitecom dc-104? it seems hfcsusb does not recognize it when I plug it in
11:17.03donkillaI wanna reinstall and see if i'll be lucky
11:17.06*** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu)
11:17.25JTdonkilla: chan_zap did not build
11:18.25donkillaThanks JT
11:18.59blitzragedonkilla: with 1.4, after compliing and installing zaptel, run in the asterisk dir: make distclean ; ./configure ; make menuselect
11:19.08blitzragedonkilla: then in menuselect you should see chan_zap in the channel drivers
11:19.18blitzrage[*] and not XXX
11:20.38*** join/#asterisk MACscr (n=MACscr@adsl-75-23-64-115.dsl.peoril.sbcglobal.net)
11:22.00donkillalet me try it out blitzrage
11:24.57donkillai've gonne to menuselect and i can't see chan_zap
11:25.25MACscrhmm, how do agents login to a call queue
11:26.06donkillaSorry i've seen it
11:26.40uweMACscr: heh, same problem here :) but basically using AddQueueMember
11:27.02tzafrirzaptel does not depend on libpri
11:27.25MACscri had the system originally setup so that all users were always logged in, but after i switched my conf files to a new server, its not working anymore
11:27.27uweMACscr: if you get further , please tell me
11:27.56donkillaI can'nt select the chan_zap
11:28.05JTtzafrir: it does if you intend to use it.
11:28.12*** join/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br)
11:28.21uweMACscr: do you have them as static members in the queue ?
11:28.37donkillaIf i press 'y' nothing happens
11:28.38tzafrirzaptel does not use libpri . Only chan_zap does
11:28.47donkillaIt has XXX
11:28.59__freedom__loverhi all, good morning
11:29.09JTtzafrir: fine, you win :P
11:29.24donkillaGood morning there and good evening here!
11:29.29__freedom__loveri have a trouble with sound in asterisk, can anyone help me about this?
11:29.52tzafrir__freedom__lover, hopefully
11:29.58__freedom__loverdonkilla: where are you from? i'm in brazil
11:30.06donkillaKenya
11:30.17__freedom__loverhum.. africa..
11:30.23tzafrir__freedom__lover, what device do you use to hear that audio? What's on the other side?
11:30.23MACscrstupid linux question, but how do i simply copy all contents of a file to the clipboard so i can paste it in pastebin
11:30.37donkillaYes
11:30.59tzafrirMACscr, which clipboard? normally you just mark it with the mouse
11:31.00__freedom__loveri have a ac97
11:31.12__freedom__lovermy OS is freebsd..
11:31.21MACscrtzafrir: but what if i cant view the entire file at one time
11:31.29MACscrim talking about in terminal
11:31.38MACscrusing something like nano to view a file
11:31.52tzafrirpressing the right mouse button should extend the selection. At least on xterm
11:32.58__freedom__lovertzafrir_laptop: when my dialplan plays a sound, some message or just playback(tt-monkeys), it stay mute
11:33.33uweMACscr: mybe you can do "cat filename" then select the whole thing with mouse if the terminal has scroll bar
11:34.03MACscrhmm, thats an idea
11:34.10__freedom__lovertzafrir_laptop: in terminal, i've played a music, unsing mpg123, and i heard the sound, but in asterisk don't
11:34.40MACscruwe: when you asked if they were static members, did you mean persistent? as in all the time?
11:35.54uweyes, i ment that maybe since you had them all logged in by default, you might have had them added to the queue by default, and they didnt have to log into/added the queues
11:36.18uwei have to run
11:36.31tzafrir__freedom__lover, as a last resort, you can run things in script(1) . But before that, send them to the logs and grab from there...
11:38.05*** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net)
11:38.16donkillaI need little help. I need to select chan_zap module in menuselect but pressing 'y' or 'F8' does nothing. It's marked XXX
11:38.21__freedom__lovertzafrir: i1ll try this
11:38.50tzafrirIs there any script to paste text to a pastebin from a file?
11:42.31__freedom__lovertzafrir: are you talking to me?
11:44.16tzafrirno. Just asking the general channel population
11:44.21blitzragedonkilla: if it's XXX'd out then zaptel might not have compiled / installed properly
11:44.40blitzragetzafrir: pastebin.ca lets you upload a post
11:44.56*** part/#asterisk cweiske (n=cweiske@dslb-088-074-128-101.pools.arcor-ip.net)
11:45.31*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)
11:46.19donkillaLet me try it out again
11:47.05blitzragecd /usr/src/zaptel ; make distclean ; ./configure ; make menuselect ; make install ; cd /usr/src/asterisk ; make distclean ; ./configure ; make menuselect ; make install
11:47.46blitzrageyou might also check out the config.log file for zaptel stuff to see any errors and such
11:48.23NgIs there a way to check if a sip extension exists? I'm passing ddis to sip extensions with a pattern, but not all numbers will be used initially. Catching the unused ones and redirecting them would be ideal :-)
11:48.55MACscrwhat does:Timeout, but no rule 't' in context 'default' mean? what is rule 't'?
11:50.18blitzrageNg: ChanIsAvail()?
11:50.20*** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk)
11:50.41blitzrageMACscr: 't' is a built in extension (like 's') that you can use in that situation
11:50.54blitzragei.e. exten => t,1,Verbose(1|Timeout rule was hit)
11:51.05NgSounds interesting :-)
11:51.18blitzrageNg: show application ChanIsAvail
11:51.25blitzragemight do what you want
11:51.37MACscrblitzrage : i have no idea what s even is
11:51.44MACscrwhy use something so not descrip
11:51.59blitzrageMACscr: because I'd assumed you'd read some documentation like the O'Reilly book
11:52.11blitzrageall of this would be much more clear if you had done so
11:52.17donkillablitzrage - what about zttool
11:52.39donkillaIt has XXX and i see Depends on: libnewt
11:52.43blitzragedonkilla: what about it? it's just a utility -- you don't have ncurses-devel installed probably if it won't compile
11:52.52MACscrblitzrage : nah, im just playing with asterisknow. I had someone setup my other asterisk system
11:53.06blitzrageoh yah -- libnewt-devel -- you're still missing a package
11:53.21*** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg)
11:53.53skirmishaguys if i have sip trunk in asterisk and i have IAX traffic, can asterisk do conversion from iax to sip auto
11:54.09blitzrageskirmisha: yes
11:54.29skirmishaok great
11:54.50blitzragewhen you drop the IAX channel into the dialplan, then you just do Dial(SIP/my_trunk) and Asterisk will take care of the rest
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11:59.17donkillaWhen i compile libnewt i get error "/usr/lib/gcc/i486-slackware-linux/3.4.6/../../../../i486-slackware-linux/bin/ld: cannot find -slang
11:59.42*** join/#asterisk champster (n=asterisk@AH.tescogroup.com)
12:00.13donkillaI mean -lslang
12:01.40blitzragedonkilla: means you need the 'slang' package (dependency)
12:02.21*** join/#asterisk zmef420 (n=zmef420@metarb3-pool2-40.mtco.com)
12:02.37skirmishablitzrage what do u mean drop iax in dialplan
12:02.46skirmishathey just send iax traffic to me
12:02.54blitzrageskirmisha: right...... and you handle it via the dialplan
12:02.57skirmishaand asterisk should do routing transparently
12:03.04blitzragei.e. context=incoming_iax_traffic
12:03.19skirmishawell i don;t have iax users on my asterisk
12:03.28skirmishai just want to forward that traffic as sip
12:03.29blitzrageyou have an iax trunk
12:03.42skirmishai use trunk only as outgoing
12:04.05blitzragethen you're taking SIP users and sending them via IAX trunk>?
12:04.11skirmishanope
12:04.25skirmishaiax user from another asterisk will send iax traffic to my asterisk
12:04.31blitzrageright
12:04.32skirmishai just convert it and send it as sip
12:05.02*** join/#asterisk msetim (n=marcos@200.195.161.164)
12:05.03skirmishaso for asterisk is just transparent traffic
12:05.20skirmishaonly thing is to do conversion to sip
12:05.22blitzragethen you need an iax type=user in iax.conf, which has a context=incoming_iax_traffic, where you handle the call
12:05.53blitzrageWhen the call comes in, that context [incoming_iax_traffic] just basically would hit a Dial(SIP/my_trunk/${EXTEN}) or somethign like that
12:06.14orlockbwbwba, /. is down
12:06.18skirmishayes i have this
12:06.19blitzragegood
12:06.24blitzrageskirmisha: well there you go
12:06.29skirmishabut i am sending it to sip context
12:06.32skirmishanot iax
12:07.02skirmishabasically i have set in sip.conf all unknown calls to go to that context
12:07.06blitzragewell whatever context doesn't matter... you get the call via IAX, put it into the dialplan to be handled, then you send the call back out using the Dial() application
12:07.07skirmishaand from there to sip trunk
12:07.24skirmishaok hope it will wokr
12:07.26skirmishawork
12:07.58blitzrageiax user --> auth via iax.conf --> handled via context=foo directive --> dialplan --> application Dial() --> SIP trunk --> outgoing call
12:08.08blitzrageI can't make it much clearer than that...
12:13.02giasai68how can i redirect a call if these fail? e.g. exten => 1265,1,Dial(Phone/phone0) if phone0 is unavailbale how can i redirect on phone1?
12:14.56blitzrageGotoIf($[${DIALSTATUS} = CHANUNAVAIL]?failover,1)
12:15.05blitzrageor some other method of parsing on ${DIALSTATUS}
12:15.07*** join/#asterisk antoniobrandao (n=yytrttry@201-43-56-71.dsl.telesp.net.br)
12:15.15giasai68ok thank you very much
12:15.34blitzrageshow application dial will show the values DIALSTATUS may contain
12:15.51antoniobrandaohello, good morning. Does anyone knows wath means  "Planning to masquerade channel"
12:16.15antoniobrandaoMar 21 10:25:53 DEBUG[15218] channel.c: Planning to masquerade channel SIP/voiplink-1ecf8cd0 into the structure of Local/554733637471@callingcard-6501,1
12:19.43blitzrageantoniobrandao: it's a debug message that just tells you what Asterisk is doing -- turn off debugging
12:19.46blitzragethat's normal
12:22.36zoablitz!!!
12:22.39zoahow did you like the phone
12:22.40zoa?
12:22.41*** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com)
12:23.31MACscrhmm, im having an issue with my main menu playing for a trunk, even though its setup to automatically be sent to directly to an extension
12:24.08MACscrexten = _X.,1,Goto(default|644|1)
12:24.41MACscrwhere should i be looking besides extensions.conf for this issue
12:28.03*** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net)
12:29.52rwardMACscr - do you have a specific entry for (default|644|1)
12:30.00*** join/#asterisk Narkov- (n=Narkov@c58-108-246-199.kelvn1.qld.optusnet.com.au)
12:30.19rwardand don't you maybe mean to do a (Dial,644) (check that syntax :p )
12:30.26*** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198)
12:30.34Dr-Linuxexten => 32345,1,DeadAGI,pager2sms.agi|03339615969\@ufone.com
12:30.35Narkov-when on an analog ZAP channel the blind transfer function will only accept one digit before immediately saying invalid extension...any ideas?
12:30.46Dr-Linuxanything wrong with this dialplan line?
12:31.00MACscrdefault is the context name
12:31.07Narkov-my time out is set to 3000ms but it immediately says invalid extension
12:31.27rwardMACscr - yes, but do you have an entry in that context for 644?
12:32.05rwardNarkov - what function are you using?
12:32.17Narkov-blindxtfr
12:32.22MACscrrward: should that be in sip.conf or users.conf
12:32.41rwardtheres a timeout for the amout of time it will wait for the digits to be pressed and then theres one for the pause period between digit presses ..
12:32.51Narkov-i get the voice anouncement correctly "what extension" but as soon as I hit one number (which registers fine) it immediately "Accepts" the number as invalid
12:33.13rwardMACscr - are you using a GUI at all?
12:33.15blitzrageDr-Linux: yah -- you're calling DeadAGI() not from 'h'
12:33.24Narkov-i only see "featuredigittimeout" which i have set to 3000
12:33.32MACscrrward: for the most part, but im having to fix things in the conf files
12:33.39blitzrageDr-Linux: DeadAGI() is for being executed after the channel has been destroyed
12:33.55Narkov-my bad..."transferdigittimeout" is set to the default of 3 seconds
12:34.21Dr-Linuxblitzrage: ok i understand
12:34.37rwardMACscr - then in your extensions.conf , in the [default] section - add a line "exten => 644,1,Dial(SIP/AIX2, 644)"   (check that dial command)
12:34.42Dr-Linuxblitzrage: what about last part of the line?
12:35.01rwardMACscr - cos I think its not finding an entry in [default] for 644 - so its dropping to you default..
12:35.30blitzrageDr-Linux: well, I prefer the DeadAGI() syntax (not using commas), but not really anything I see wrong -- what is the error?
12:35.31rwardor change the "goto(default, 644,1)" to "dial(SIP/whatever, 644)
12:35.51Narkov-the config documentation says transferdigittimeout is in seconds but is it actually meant to be miliseconds?
12:36.04*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-185-4.buckeyecom.net)
12:36.12blitzragerward: that dial line would dial for 644 seconds
12:36.32rwardsorry - I know - can't remember the DIAL syntax - it needs to be fixed :p
12:36.42blitzragewhat needs to be fixed? It works fine
12:36.42rwardbut basically - dial extension 644 ..
12:36.52blitzrageDial(SIP/proxy/644)
12:36.54rwardsoryr - replying to MACScr
12:37.10Dr-Linuxblitzrage: basically i'm trying to use an SMS application in perl AGI, so i need to put number@email.com in dialplan
12:37.36Dr-Linuxblitzrage: but i'm not sure how can i type email address in dialplan bcoz of "@"
12:37.41blitzrageDr-Linux: ok -- I don't see any error in the syntax -- it must be a script issue
12:38.06blitzrageyou're escaping the @ -- what does the script see?
12:38.42*** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br)
12:39.44Dr-Linuxblitzrage: maybe you can see here the script: http://portal.mmasson.com/asterisk/files/pager2sms-0.2.agi
12:40.30blitzrageDr-Linux: I don't have time right now
12:40.46Dr-Linuxblitzrage: No Problem. Thanks
12:40.53blitzrageyou not showing any of the output -- like what asterisk is seeing or doing, or what the script is seeing or doing
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12:53.16donkillablitzrage: I sorted out slang package and now when i go to chan_zap in menuselect it still has XXX. Depends on:zaptel_vldtmf(E),zaptel(E),tonezone(E)
12:53.20*** part/#asterisk Narkov- (n=Narkov@c58-108-246-199.kelvn1.qld.optusnet.com.au)
12:53.45blitzragedonkilla: you need to checkout zaptel from SVN
12:53.52blitzragedonkilla: asterisk-1.4.2 right?
12:54.01donkillayes
12:54.13blitzragesvn co http://svn.digium.com/svn/zaptel/branches/1.4 zaptel-1.4
12:54.36blitzrageyah... that was a bit of a bug on Digiums part. I expect to see a Zaptel-1.4.1.1 come out to address that...
12:55.41donkillathanks.Let me try it out
13:01.06MACscrrward : here is my extensions.conf
13:01.06MACscrhttp://pastebin.ca/406431
13:01.18MACscri still havent been able to resolve the issue
13:01.25MACscri tried your different ideas
13:01.44MACscrwhats odd is that it used to work on one trunk, but not the other, then both dont work
13:01.49*** join/#asterisk friedrich| (n=friedric@e177249102.adsl.alicedsl.de)
13:02.15MACscrthe only things i have changed for incoming calls was from a  call queue to directly to an extension. Figured it was less complicated for testing purposes
13:03.11*** join/#asterisk lilwookie (n=lilwooki@30-82-252-216-static.enter-net.com)
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13:15.46MACscrwow, things died in here
13:15.53galerasHello. Which tcp ports do i need to open to place a sip call through a firewall?
13:16.33galerasinit 3
13:16.58sashiongaleras: UDP: 5060 and then range UDP 10000-20000 for RDP
13:18.22galerasthanks sashion, is 5060  used for signalling and 10000-20000 for voice?
13:18.31zoayes
13:19.12blitzragegaleras: note -- he said UDP, not TCP (asterisk doesn't use TCP)
13:19.20blitzragemight solve you some pains later :)
13:19.36galerasthanks guys
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13:34.41*** mode/#asterisk [+o mog] by ChanServ
13:34.56heisonanyone here using SPA-3102 ?
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13:39.11GaryHHi, is anyone here familiar with chan_misdn?
13:39.45EtienneGHi folks ... looking for an IRC that would interface with * to print incoming call number and CID in an IRC channel
13:39.52EtienneGdoes anybody know of such a thing ?
13:40.15EtienneG(an IRC bot, even)
13:40.30nasls_lsa??
13:40.48*** part/#asterisk galeras (n=root@200.118.211.115)
13:40.49nasls_lsaan idea I can give you
13:41.04nasls_lsathat find an IRC bot writen in PHP
13:41.10zeedoEtienneG: http://www.telephreak.org/code/astbot-1.0
13:41.14zeedoyou could use that as a basis
13:41.14nasls_lsaand then connect  with AGI ( I think )
13:41.29zeedonot sure exactly what functions it has, but it appears to interface with *
13:41.39zeedonever used it, just come across it
13:43.44*** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
13:49.11EtienneGthanks guy, I will look into astbot
13:50.12*** join/#asterisk bkuhn (n=bkuhn@fsf/member/bkuhn/bkuhn)
13:50.27bkuhnsterday
13:54.08JTeww php
13:54.14JTirssi does perl scripting
13:54.21JTthat'd be easy to interface with AGI
13:54.27marc\cbachaps
13:54.33marc\cbaif my SIP trunk provider
13:54.50marc\cbaspecifies the DID number, what was dialed to reach my pbx
13:55.01marc\cbain the second packet, the ACK (as opposed to the INVITE)
13:55.16marc\cbahow could i go about routing based on that in my dialplan?
13:56.03marc\cbai.e
13:56.33marc\cbainstead of this: INVITE sip:08450000010@000.000.228.32 SIP/2.0
13:56.36marc\cbathe trunk passes
13:56.48marc\cbaINVITE sip:000.000.228.32 SIP/2.0
13:56.56marc\cbaand it is not until the second packet
13:56.57marc\cbathe ACK
13:57.09marc\cbathat it mentions the DID
13:57.23marc\cbaTo: <sip:448450000010@n.e164.org.uk>
13:59.22rwarduuuummm i give up :/
14:05.35*** part/#asterisk EtienneG (n=etienne@modemcable178.77-70-69.static.videotron.ca)
14:05.35*** join/#asterisk stony (n=steinche@p57b38ac1.dip0.t-ipconnect.de)
14:05.49stonyhi, did the Dial() function change in asterisk 1.4 ?
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14:24.33giasai68how can i accept call only from some ip on asterisk? i explain better now i can call trought asterisk using all ip (sip,h323) can i accept call only from  some ip or can i set some users/IPs?
14:24.53b11duse a firewall?
14:24.56*** join/#asterisk Ac1dcrawl (n=cow@64.31.169.118)
14:25.10antoniobrandaohellou, is there any kind of media-proxy for asterisk? i would like to relay rtp traffic out of asterisk machine, but reinvite is not always possible due to nat issues
14:25.10b11dor, dont you specify the IP of the host in sip.conf with the "host = x.x.x.x" line?
14:25.24giasai68this is only mode (firewall)
14:25.32Ac1dcrawlI am running asterisk 1.2.13 and for the some reason when I am using the CLI when doing command-line completion the CLI locks up
14:25.39Ac1dcrawlany ideas what could be wrong?
14:25.41b11dthen upgrade
14:25.46b11d1.2.16 is out of the 1.2 branch
14:28.15*** join/#asterisk jm|laptop (n=jm@sentry.flags.co.uk)
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14:29.02*** join/#asterisk penguinFunk (n=penguin@87.224.86.46)
14:29.31b11d.
14:29.54*** join/#asterisk hijacked (i=qib1@cerebus.clandestineresearch.com)
14:31.53hphinchark, how is everyone?
14:32.16hphincis anyone an expert in presence / parking?
14:32.23*** join/#asterisk Mercestes (n=Merceste@cpe-24-175-82-3.houston.res.rr.com)
14:32.35sashiongiasai68: are you using static hosts for your sip hosts, or are you using host=dynamic ?
14:33.13b11dmercestes
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14:33.43Mercestesb11d
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14:36.25b11dgod i hate you
14:36.49*** join/#asterisk CreepyCrawly (n=creepy@62.135.86.11)
14:36.57cyanid3Is it possible with any phone / asterisk combination to make it so that when someone (a receptionist) presses a button on their phone for the ivr that's currently in use to be changed?
14:37.02CreepyCrawlygood afternoon people, is there a port of freepbx available ? from freebsd ?
14:37.16b11dask in #freepbx
14:37.20b11di use asterisk on FreeBSD just fine though
14:37.30MercestesI love you too, b11d.
14:37.32b11d:)
14:37.39b11di retract my statement of hatred
14:38.03Mercestesyay
14:38.09Mercestesyou are always off topic
14:38.10CreepyCrawlyb11d, i got it running and all :) but i wanna try out freepbx
14:38.16b11di know Mercestes..  it's what i do. :)
14:38.27b11doh.. well then go to #freepbx and talk to them about it :)
14:38.44CreepyCrawlyi am hehe :)
14:38.45MercestesCreepyCrawly, You want to try out retardation?   will it make you feel smarter when you give it up or are you seeking addiction to the ludicrous?
14:39.02CreepyCrawlyMercestes, none of those ;)
14:39.04b11dwtf is up with you today Mercestes.. what wrong?
14:39.15CreepyCrawlyheh
14:39.39MercestesCreepyCrawly, ;)  Freepbx is a royal mess and a complete clusterfsck.  I'd avoid it...unless you like pain.  Then I suggest #bondage as an alternative to #freepbx if no one is answering there
14:39.44stonymy asterisk installation isn't dialing .... it says all channels are busy, but they aren't ... any ideas ?
14:39.54CreepyCrawlyhaha
14:40.29MercestesCreepyCrawly, lol.  Just make backups and don't take it too seriously.  It should prove educational tho.  Good luck.
14:40.39CreepyCrawlyhah fsck it
14:40.57CreepyCrawlyi aint going through all that havoc ill just stay comfy with the cmd lime kthx
14:41.07Mercestesgood man. :)   If you need scripts or pointers or magick just ask in here, there is lots of help in here.
14:41.18CreepyCrawly:)
14:41.21Mercestesif you even have freepbx files on your install they give you the #freepbx boot.  :(
14:41.22hphincis there anyone who knows presence / parking in here?
14:41.27Mercesteswe're kinda elitest liek that.
14:41.36hphincI am looking to park a call and have the light light up on my Snom 360.
14:41.40hphincneed some guidance....
14:41.45Mercestesb11d:  I'm in a ...."mercestes mood"  :D
14:41.46b11dthere's a right place for the right things..
14:41.53b11di dont know what that means yet..
14:41.56CreepyCrawlyhah
14:42.30Mercestesb11d:  Both halves of your sentance are subject to opinion independently tho. =/
14:42.39Mercesteshphinc:  Ok.  What exactly is wrong then??
14:42.53*** join/#asterisk Opperior (n=chatzill@24.61.165.73)
14:42.59hphincMercestes: I don't understand how the function keys communicate with Asterisk.
14:43.01*** join/#asterisk antlers (n=antlers@ip70-173-90-39.lv.lv.cox.net)
14:43.01b11daye, all things are interpreted subjectively though.
14:43.20hphincPark() and Parkand Announce() I can read docs on...but how does the phone tell asterisk to execute those commands?
14:44.11Mercestesvia an extension.
14:44.15Mercestesor via features.conf
14:44.24*** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br)
14:44.25*** join/#asterisk antlers (n=antlers@ip70-173-90-39.lv.lv.cox.net)
14:44.29flujanhi guys...
14:44.29hphincok, let me re-read about features.conf, and I will get back to you. Thanks.
14:47.02hphincMercestes: does the phone use applicationmap to talk to asterisk in reference to parking calls?
14:48.28*** part/#asterisk GaryH (n=chatzill@2001:618:42d:101:213:72ff:fecf:8262)
14:48.58b11di wish i had an ipv6 address
14:54.12*** join/#asterisk step_quasar (n=step_qua@250-171-114-200.fibertel.com.ar)
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14:54.53chefrsAny idea when I dial an outbound trunk, about 50% of the time it rings once and then just sits there?
14:55.08hphincMercestes: I have parking working, but I want to use the function keys to automatically get the parked call. (and light up like a key system to show that someone is on that extension, parked). Any thoughts?
14:56.23Mercesteshphinc:  not that I am aware of.
14:56.35hphincbusted.....
14:56.36MercestesHphinc:  Yea, use a key system.
14:56.37hphinc:-(
14:56.50hphincI have seen it done, but I can't figure out how to do it.
14:56.57Mercesteshphinc:  You could map a "function key" to a "key map" or you couild just program in *8 as call park, with park and announce...
14:57.04hphincsomething about the metermaid patch, which I have done.
14:57.08Mercesteshphinc:  But you would still have to dial the extension to pick up the call.
14:57.20hphinchow do I map a function key to a key map?
14:57.25hphincin what conf file?
14:57.33JTantoniobrandao: canreinvite=no
14:57.45Mercesteshpinc:  As far as "lighting up a light" like on a key system.....I'm not even sure that would work as the lighting of the lights is a buddy watch, not a "line watch"  There are no lines in Asterisk, so therefore, there is no key system.
14:57.47hphincMercestes: I am able to park with the function key. Just not pickup or show the light.
14:58.10chefrsAny idea when I dial an outbound trunk, about 50% of the time it rings once and then just sits there?
14:58.15hphincMercestes, when I figure it out, you want me to post it back here?
14:58.25Mercesteshphinc:  When you park a call the call belongs to asterisk, not the snom.  The snom becomes oblivious.
14:58.56hphincright. But if I set the snom function buttons to 701,702,703....etc... I can have those buttons light up when a call is placed on park.
14:58.57antoniobrandaobut canreinvite=no can't "relay rtp traffic out of asterisk machine"
14:59.12Mercesteshphinc:  Let me use an analogy...it's like a girlfriend....  On a key system, she shows up at a party, and wants you.  On a key system, she comes to me and I bring her to you and I go "I have your girlfriend, here.."
14:59.25MercestesIn asterisk,  you get a sticky note that says "I have your girlfriend, come get her when yoru ready."
14:59.33chefrslol
14:59.43JTantoniobrandao: what are you quoting? yes, it can.
14:59.46hphincWhat if I have two girlfriends?
14:59.51hphinc(were I so lucky).
14:59.54hphinc:-)
15:00.10Mercesteshphinc:  in a key system, your girlfriends would conflict because you could only play with one at a time....
15:00.21Mercesteshphinc:  In asterisk you can play with 2 or even 100 girlfriends simultaneously.
15:00.21hphincMercestes: That's no fun.
15:00.22kremoo<PROTECTED>
15:00.39hphincMercestes: Asterisk: more chicks that you can handle.
15:00.46Mercesteshphinc:  exactly.
15:01.15Mercesteshphinc:  The point is....once you park the call, the snom is out of the picture entirely.  It cannot monitor the state of that call.  The call is released to the PBX and handled there and the snom is free to....set fire or something.
15:01.39Mercestesthere is no line 1,2,3,4   Everything is dynamic, free flowing, and assigned as needed.
15:02.07MercestesFlash Operator Panel can monitor the parked slots of 700, 701, 702, 703 and via the manager interface you can give an indicator, and maybe even HACK the Snom to do what you want.
15:02.13hphincso, then, how do we tell the Snom to light up that light?
15:02.17Mercestesbut manager API is about the only thing yo uhave to attempt what you want.
15:02.18b11d.
15:02.22hphincwhile it is setting fire to something....
15:02.27hphincextension hints?
15:02.46antoniobrandaoJT, quotting myself.  With canreinvite=no all rtp traffic will go in asterisk machine. I would like to not use asterisk machine bandwith for rtp traffic. Like openser does with rtp_rpxy and media_proxy.
15:03.07*** join/#asterisk tkowal (n=nospamto@74.93.82.14)
15:03.27JTantoniobrandao: sorry, make up your mind
15:03.34hphincMercestes: http://www.voip-info.org/wiki/view/Asterisk+phone+snom
15:03.48JTantoniobrandao: you either want rtp do go through the box or not?
15:04.29tzafrirkremoo, I figure you saw http://www.xorcom.com/pdfs/AB001_OpenDoors.pdf .
15:04.55chefrsAnyone have any ideas?
15:05.22hphincI had an idea once. It was brief, fleeting, and I eventually lost it.
15:05.29chefrsAny idea when I dial an outbound trunk, about 50% of the time it rings once and then just sits there?
15:05.50antoniobrandaoJT, i always made my mind long time ago. No rtp traffic in asterisk. Canreinvite=yes is great, but with nat doesn't work. So, what to do?
15:06.21JTantoniobrandao: nothing you can do except give them all public ips
15:06.41JTantoniobrandao: otherwise you can't reinvite
15:07.07*** join/#asterisk frenzy_ (n=frenzy@unaffiliated/frenzy)
15:07.30*** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn)
15:07.55antoniobrandaoJT, ok, tks. That what i tought. Will have to put some ser in this scenario
15:08.11JTantoniobrandao: SER won't help.
15:08.18JTit's a protocol issue
15:08.22JTthat SER does not solve
15:10.30*** join/#asterisk heh_v_water (n=heh_v_wa@71-210-51-58.hlna.qwest.net)
15:10.37antoniobrandaoJT, yes, it does. ser/openser with media_proxy solves by putting an external daemon. The natted clientes send rtp media to that daemon which relays to the another end.
15:11.15JTwhich is the same as running asterisk with canreinvite=no
15:11.23JTdifferent implementation
15:11.27JTbut same effect
15:12.31antoniobrandaoJT, no, its not. With canreinvite=no one machine is responsible to handle all rtp traffic
15:12.55*** join/#asterisk tkowal (n=nospamto@74.93.82.14)
15:13.01antoniobrandaowith media_proxy you can use many machines
15:13.21JTantoniobrandao: that's up to you
15:13.25JTthe implementation
15:13.34JTyou can use multiple asterisk machines too
15:13.37JTso moot point
15:13.52JTi'm not saying asterisk is necessarily better
15:13.55*** part/#asterisk frenzy_ (n=frenzy@unaffiliated/frenzy)
15:15.24*** join/#asterisk step_quasar (n=step_qua@250-171-114-200.fibertel.com.ar)
15:18.37*** join/#asterisk ctooley (n=ctooley@rrcs-71-42-115-242.sw.biz.rr.com)
15:20.27*** join/#asterisk vlt (n=dm@port-87-234-124-63.dynamic.qsc.de)
15:20.47*** join/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com)
15:20.59ctooleyI'm having an issue with ExternalIVR.  When a call comes in and is in the dialplan, they can press a button, the DTMF is received and action is taken, when ExternalIVR is called, no more audio, either way.
15:21.13ctooleyIt's like the sound conduits go away altogether.
15:22.32*** join/#asterisk nasls_lsa (n=chatzill@athedsl-179959.otenet.gr)
15:22.44vltHello. I have defined two SIP phones A and B each in "callgroup=1". But when I try to pickup a ringing call on A from B I get "chan_sip.c:10458 handle_request_invite: Nothing to pick up". What id I miss?
15:24.50*** join/#asterisk drachenfels (n=king@85.189.8.11)
15:27.05m4rkl4ri'm using the manager api to originate calls.  If I say: Channel: SIP/9043067733@joinuneta.com, then the CLI says:
15:27.05m4rkl4rchan_sip.c:1980 create_addr: No such host: joinuneta.com
15:27.05m4rkl4rchannel.c:2432 __ast_request_and_dial: Unable to request channel SIP/9043067733@joinuneta.com
15:27.05m4rkl4rHowever, joinuneta.com has an srv record, and using dig on the asterisk server shows it is properly defined.
15:27.05m4rkl4rFurthermore, I get precisely the same error if I use the direct hostname: SIP/9043067733@sip.joinuneta.com
15:27.13pifvlt: you forgot pickupgroup
15:27.34vltpif: Do I need both?
15:27.43pifyep
15:28.10JunK-Ym4rkl4r: quick hack, uses its ip.
15:28.11m4rkl4rfurthermore if I use the direct IP address, I get the same error.
15:28.49Gido-Em4rkl4r default gw problems?
15:30.22*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
15:30.30JTm4rkl4r: why don't you do it properly with a sip.conf entry for that host?
15:30.31m4rkl4rGido-E: no.. i am able to ping yahoo.com, for example
15:31.31drachenfelsdoes anyone know how to trobleshoot an fxo module? whenever we hook it up to a line, the line suddenly goes engaged - and we can't call in.. We've tried the same line, connected to a different port on the card and it works fine, which makes me think it's the module on the card, but I've no idea how to troubleshoot it..
15:31.49*** join/#asterisk nasls_lsa (n=chatzill@athedsl-179959.otenet.gr)
15:32.04*** join/#asterisk pif (n=ldm@zenon.apartia.fr)
15:32.16m4rkl4rJT: That doesn't work either.
15:32.32m4rkl4rDoing sip reload, I see something interesting:
15:32.58m4rkl4rar 22 11:32:39 WARNING[530]: chan_sip.c:12708 reload_config: Empty context specified at line 5 for domain 'uneta.com'
15:32.58m4rkl4rMar 22 11:32:39 WARNING[530]: chan_sip.c:12708 reload_config: Empty context specified at line 6 for domain 'youneta.com'
15:32.58m4rkl4rMar 22 11:32:39 WARNING[530]: chan_sip.c:12708 reload_config: Empty context specified at line 7 for domain 'sip.uneta.com'
15:32.58m4rkl4rMar 22 11:32:39 WARNING[530]: chan_sip.c:12708 reload_config: Empty context specified at line 8 for domain '66.129.95.19'
15:33.17puzzled~pb
15:33.18jbotpb is probably a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste
15:33.23JTmaybe you should specify a valid context
15:33.33m4rkl4ri am specifying a valid context.
15:33.41JTwith nothing in it?
15:33.43b11dlol
15:33.46m4rkl4rhaha
15:33.47m4rkl4rno.
15:33.58JTthat's exactly what the error suggests
15:34.01puzzleddoes anyone know if the sangoma patches for zaptel can be applied without interfering with the proper functioning of digium cards?
15:34.12JTsend the whole of your extensions.conf and sip.conf to pastebin.ca
15:34.20m4rkl4ri built a new asterisk box, copied over the config files, with extensions based off of odbc.
15:34.24m4rkl4rok.
15:34.30coppicepuzzled: I mix cards, and never had a problem
15:34.33JTodbc
15:34.37puzzledcoppice: thanks
15:34.42JTis the odbc connection working?
15:35.41drachenfelsdoes anyone know how to trobleshoot an fxo module? whenever we hook it up to a line, the line suddenly goes engaged - and we can't call in.. We've tried the same line, connected to a different port on the card and it works fine, which makes me think it's the module on the card, but I've no idea how to troubleshoot it..
15:35.52*** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com)
15:36.10JTdrachenfels: i swear i saw that exact same question not a couple of minutes ago..
15:36.25b11dweird, me too.
15:36.30m4rkl4rjt: extensions.conf in http://pastebin.ca/406616.
15:36.30m4rkl4rand odbc is working.
15:36.38puzzledyou guys need glasses. see everything double :)
15:36.46b11d")
15:36.48b11ddoh
15:36.48b11d:)
15:37.03*** join/#asterisk nasls_lsa (n=chatzill@athedsl-179959.otenet.gr)
15:37.27JTm4rkl4r: ah like i thought after you mentioned realtime, can't really diagnose that bit since i don't know what is in your db
15:37.54m4rkl4rjt: sip.conf: http://pastebin.ca/406618
15:37.56m4rkl4ryes.
15:38.04m4rkl4rthe thing that is odd, jt,
15:38.09b11dyou just need more marklar..
15:38.24b11dmarklar the marklar over the third marklar and then you'll have marklar working.
15:38.34b11dsee?
15:38.37m4rkl4rif I take the command that I'm sending to the manager api, change all the Variable directives to Set, and put it in the spool, it works.
15:38.41m4rkl4ryes i see, marklar
15:38.57JTm4rkl4r: do not use spaces with = symbols in sip.conf
15:40.09m4rkl4rjt: ok.. i didn't know that.. but the error message still is there after fixing that
15:40.20MrWupAastra 9133i VoIP Phone
15:40.22MrWupwhat do you think?
15:40.27MrWupive just ordered 20 of em
15:40.46pifyou shoulda asked before ordering
15:40.49JTa bit late to ask ;)
15:40.53MrWupi can always cancel
15:40.59MrWupi did ask beforehand too
15:41.01pifcancel and get polycoms
15:41.07L|NUXis there any bug in asterisk 1.2.17
15:41.07coppicei guess this is a masochism thing :-)
15:41.13m4rkl4rThey don't have stun, or encrypted channels for provisioning
15:41.19m4rkl4rthe astraphones, that is
15:41.21MrWuppolycoms are too expensive
15:41.35pif430 aren't
15:41.36*** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ma.comcast.net)
15:41.38m4rkl4rI've been pretty happy with the linsys spa 941 so far
15:41.56JTm4rkl4r: most people use sip phones on a lan, don't need stun, actually, you don't need stun if your sip server doesn't suck, even if you're behind nat
15:42.00L|NUXi am sending calls to my * no rule found for that number so it should go to s but its not working
15:42.08L|NUXany one have similar issue ?
15:42.14JTm4rkl4r: asterisk doesn't even support encrypted SRTP
15:42.14drachenfelssurely posting sip.conf with usernames and passwords is a bad idea?
15:42.27xpotanyone know of existing issues with automon in 1.4?
15:42.29m4rkl4ryes, I suppose it is, drachenfels
15:42.33MrWupi thought aastra was good quality
15:42.41OpperiorI've had to use stun with * when both the server and phone are behind a nat
15:42.58JTaastra is decent, but they're fairly low end phones
15:43.03*** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl)
15:43.04JThaven't used them personally
15:43.11JTsurely 1000X better than grandstream
15:43.24m4rkl4rdrachenfels: but those are old, invalid paswords
15:43.43Strom_Mhaving a white hot railroad spike driven six feet into your skull is better than grandstream :)
15:43.57L|NUXany one have similar issue like i have
15:44.00JTOpperior: server behind nat :(, need to port forward then
15:44.03L|NUXcan some one help me with that :)
15:45.15JunK-Ylinux: you are sending it s@context1?
15:45.29*** join/#asterisk marv[work] (n=timr@24.214.206.254)
15:45.34OpperiorJT: I do.  But with the phone behind a nat as well, RTP still won't work without stun.  At least, not that I've found
15:46.26pifMrWup : how much do they cost?
15:46.28Gido-EL|NUX do you have s,1 ?
15:46.40L|NUXyes
15:46.47JTOpperior: must've been setup wrong
15:46.51JTSTUN is really a hack
15:46.55JTfor bad sip servers
15:46.59L|NUXJunK-Y : nope
15:46.59JTthat are not nat smart
15:47.08JunK-Ylinux: what does: dialplan show s@context1 says?
15:47.20Opperiorwell, I won't deny the possibility exists
15:47.21JunK-Ylinux: ya just said yes to Gido-E.
15:48.17L|NUXwait
15:48.38L|NUXshow dialplan ?
15:48.43JunK-Yif 1.2, yes
15:48.53JunK-Yshow dialplan s@ur_context
15:49.36L|NUXok
15:49.46L|NUX<PROTECTED>
15:49.53*** join/#asterisk spanglesontoast (n=edd@eddland.plus.com)
15:50.04spanglesontoasthmm
15:50.10L|NUX<PROTECTED>
15:50.21L|NUX<PROTECTED>
15:50.59JunK-Yand when yo're calling that s,1, whats the output in the CLI?
15:51.47L|NUXwait
15:52.06JunK-Yand why are ya passing all these args? they are automatiquely passed to the agi anyways, ya can directly grab them in ur agi.
15:52.16L|NUXwell because AGI need them :)
15:52.17L|NUXhehe
15:52.30L|NUXJunK-Y : AGI has been written in that way
15:52.30JunK-Ythey are all passed by default to agi
15:52.35L|NUXyes
15:52.42JunK-Yso you dont need to pass them.
15:52.56JunK-Ydo agi debug, ya will see rxing from *
15:53.03L|NUXone thing i want to know
15:53.13L|NUXhwo can i get log of sepcific number
15:53.20L|NUXlike there are too many calls coming
15:53.49JunK-Yactually, ya cant directly in *. ya have to handle it urself.
15:53.57L|NUXokies
15:53.59L|NUXlogs
15:53.59L|NUX:)
15:54.02L|NUXfull will help me
15:54.03L|NUX:)
15:54.04JunK-Ybingo.
15:54.08*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)
15:54.24JunK-Ybut in the future, ya will be able to filter CLI output.
15:54.40*** join/#asterisk ping2921 (n=marc3234@206-248-178-169.dsl.teksavvy.com)
15:55.49L|NUXhummm
15:55.49L|NUXok
15:56.09xpotanyone know how to get automon to work properly?
15:56.53JunK-Ylinux: so ur cli output?
15:58.34*** join/#asterisk lukketto (n=lukketto@82.59.103.134)
15:59.27L|NUXwait
16:00.27*** join/#asterisk stony (n=steinche@p57b38ac1.dip0.t-ipconnect.de)
16:02.11*** join/#asterisk Hmmhesays (n=Neg@24-117-131-41.cpe.cableone.net)
16:02.40Hmmhesaysis it possible in linux to check what init state was used when you booted it?
16:03.18*** join/#asterisk astersip (i=53f08b07@gateway/web/cgi-irc/ircatwork.com/x-9390171b59d5b8a7)
16:03.33Qwell[]Hmmhesays: not really
16:03.47Qwell[]I mean, if you check inittab you can see the default, but that won't tell you if you used single or not
16:03.49astersiphi
16:04.13astersipif someone can help me....i have a E1 width 30 numbers
16:04.16Hmmhesaysyeah thats what I thought
16:05.38astersipi would like to make that one extension when make a outbound call
16:05.38astersipthat they got out by one of thouse 30 numbers
16:06.18*** join/#asterisk zmef420 (n=zmef420@metarb3-pool2-40.mtco.com)
16:06.51*** join/#asterisk kanelbullar (n=kanelbul@83.240.200.92)
16:07.16*** join/#asterisk spanglesontoast (n=edd@eddland.plus.com)
16:07.20spanglesontoastanyone played with voipuser ?
16:07.26[TK]D-Fenderastersip: Set(CALLERID(num)=1234567) , Set(CALLERID(name)=Just Me)
16:07.39[TK]D-Fenderspanglesontoast: Added that peer entry like the sample said yet? :)
16:08.02spanglesontoastwhat the one that calls throug to the correct extension ?
16:08.18[TK]D-Fenderspanglesontoast: the one that lets you dial out...
16:08.24spanglesontoastyea it dials up
16:08.31spanglesontoastbut says congestion
16:09.03spanglesontoastalso upgraded to 1.4.2
16:09.04[TK]D-Fenderspanglesontoast: We shouldn't even have to ASK for you to pastebin up the CLI output of that failed call and your matching configs w/o passwords.  Just FYI....
16:09.27astersip[TK]D-Fender: where do i put that ? i'm using trixbox and freepbx
16:09.38[TK]D-Fender~trixbox
16:09.46jbottrixbox is probably junk - avoid.  It is also unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/
16:09.47[TK]D-Fender~freepbx
16:09.50jboti heard freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
16:10.05[TK]D-FenderQwell[]: You've been... buys ;)
16:10.09[TK]D-Fenderbusy*
16:10.13Qwell[]I don't know what you're talking about.
16:10.31[TK]D-FenderQwell[]: Denial.... its not jsut a river in Egypt ;)
16:11.09Mercestesmorning, society
16:11.14gambolputtyNo, its a state of the union.
16:11.19gambolputtyonion
16:11.20[TK]D-FenderQwell[]: Fonality has a direct business relationship with Fonality do they not?
16:11.30Qwell[]I would hope so?
16:11.59Qwell[][TK]D-Fender: re-read and re-ask your question :)
16:12.06SwK[Work]hah
16:12.06[TK]D-FenderQwell[]: Just wondering, because looking at their new Trixbox "appliance (read : full server in day-glo case), they seem to be using Sangoma cards...
16:12.20Qwell[]it made no sense at all, heh
16:12.28SwK[Work]qwell[] hey qwell has a direct business relationship with qwell right?
16:12.31[TK]D-FenderQwell[]: : Yeah... cross-wire .  Clearly meant to say Digium....
16:12.36Qwell[][TK]D-Fender: no
16:12.42Qwell[]I mean
16:12.45Qwell[]no
16:12.49[TK]D-Fender:)
16:12.53Mercesteslol
16:12.59[TK]D-Fenderok, explains a bit...
16:13.02Qwell[]I don't know :P
16:13.05Mercestesyay, we got english and spainish translations.
16:13.13[TK]D-FenderFor some reason I just sort of figured they did...
16:13.29*** join/#asterisk uwe (n=uwe1@dogbert.palnet.com)
16:18.06spanglesontoasthmm what version of the sip protocol does 1.4.2 use ?
16:20.37[TK]D-Fenderspanglesontoast: Same as before with a few more RFC's for SLA
16:20.49spanglesontoastok
16:21.05SwK[Work]any telemarketers around? PM me have a question for you
16:22.55spanglesontoastthe annoying thing is theres lots of posts about the issue fender
16:22.58MercestesOf course, all telemarketers hang out in freenode: #asterisk    all of us being geeks and all.
16:23.04spanglesontoastbut no one seems to resolve it
16:23.10*** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290)
16:23.16Mercestesif you want telemarketers just post your phone # and I'll hook you up with all you can handle.
16:23.21[TK]D-Fenderspanglesontoast: What annoying is you keep complaining and don't "show us the money" :)
16:23.35[TK]D-Fenderspanglesontoast: Stop whining and start pastebin-ing!
16:24.00spanglesontoastpaste what lol I reckon it's voipusers end
16:24.32SwK[Work]mercestes you would be suprised how many hang out in here and use asterisk based dialers
16:24.46SwK[Work]mercestes I sell dialers to them all time
16:24.46MercestesSwK[Work],   Only in secret.
16:24.55SwK[Work]thats why I said PM me :P
16:24.58Mercestesasterisk is not pro+telemarketing bastards.
16:25.08Mercestesbut thanks for identifying yourself.
16:25.23SwK[Work]maybe true, but they are a large market segment that spends a lotta money
16:25.53MercestesON annoying people in futile and ineffective marketing techniques that preys upon the retarded and senile.
16:26.01SwK[Work]sure
16:26.02SwK[Work]why not
16:26.06[TK]D-Fenderspanglesontoast: Last we saw you didn't do it right, and you still are not showing us that you've corrected this or have any valid claim for it to be their fault.
16:26.20[TK]D-Fenderspanglesontoast: You are wasting time with this approach.
16:26.22SwK[Work]only a moron would assume it doesnt work... cause obviously it does work... or they wouldnt be pouring the kind of money into it that they are
16:26.29*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
16:26.55MercestesSwK[Work], Of course it works, there are lots of retarded and confused old people out there.  I said it was immoral and ineffectual.
16:26.59SwK[Work]its like spam... why do people keep spamming? because idiots read that crap then buy that product
16:27.09Mercestes....
16:27.19MercestesPpl spam just to cause internet problems.
16:27.41Mercestesbut I guess if you hit a billion ppl and find 1000 retards with a dollar....you make a pretty good profit on yoru open source software.
16:27.52SwK[Work]well you yourself just admitted its not ineffectual... as far as morality, who are you to dictate your morality on anyone else... you ahve the DNC
16:28.03Mercestesbut to me your the kind of person who'd sell abused raped females on the streets for cash if you could get away with it.
16:28.28MercestesPM me, I have products to make it more effective to prey on idiots .
16:28.32[TK]D-FenderSwK[Work]: DNC ... yeah... in the words of Dr. Phil "How's that workin' out for ya?"
16:28.33SwK[Work]mercestes: i did work in a strip club for 2 years does that count?
16:28.50SwK[Work]heh
16:28.53*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
16:28.54MercestesSwK[Work], It just makes me hpapy that I won't be dealing with you in the next life.  Is your marketing campaign done?
16:29.18[TK]D-FenderMercestes: Next life?  You don't even have one NOW! ;)
16:29.23SwK[Work]actually I'm not marketing anything here... I was wanting a marketers oppinion on something
16:29.27[TK]D-FenderMercestes: Get a life!
16:29.41[TK]D-Fender(and a haircut... and a real job... and ....
16:29.47Mercestes[TK]D-Fender, ...true.  pwned...:(  *cries*
16:30.14[TK]D-FenderMercestes: You seem to be able to do that nearly on cue... you should consider day-time dramas ;)
16:30.22*** part/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com)
16:30.32Mercestes[TK]D-Fender, LOL.  maybe you've seen me on the Spainish soaps, amigo.
16:30.42*** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk)
16:30.47errrIm not sure what to call this feature Im looking for but what I want to do is be able to put someone on hold in my office, then head to our server room and pick put the phone, dial a number and be able to transfer the call I had on hold in my office to the phone Im on in the server room. Can this be done, and if so what is this called?
16:31.14puzzledparking
16:31.18[TK]D-FenderMercestes: Nope.... I don't have cable/satelite/etc.  I also haven't even plugged in the 'ol rabbit ears (to my *52" HDTV*)
16:31.38errrpuzzled: ok thanks
16:31.44[TK]D-Fendererrr: Call Parking.  Look it up on the WIKI
16:31.51puzzlederrr: configure it in one of the conf files
16:31.53b11dTK, same here man..
16:32.09Mercestes[TK]D-Fender, Just for IRC huh?  Nice.  D-Fender, wireless keyboard and a 52" plasma mounted on the wall, on his ergonomic recliner...in his speedos...giving sound technical advice.
16:32.13errr[TK]D-Fender, puzzled thanks, having a name makes it eaiser to search for :)
16:32.42[TK]D-FenderMercestes: No, I use a 22" for my main PC.  the HDTV is for my * server :)
16:33.01puzzled[TK]D-Fender: for some reason ppl from North-America often mention the size of their TV. Why is that?
16:33.18puzzledI would not even know the size of my TV
16:33.24Mercesteslol.  52" of * cli baby!
16:33.25[TK]D-Fenderpuzzled: Dunno, I did mine for cdramatic contrast to the fact I don't watch TV per-se.
16:33.51[TK]D-Fenderpuzzled: Not knowing what you own just makes you an uninformed consumer :)
16:33.59coppiceour TV is only 42", but its what you do with it that counts :-)
16:34.49puzzledfunny because I have yet to see a decent quality American TV and the content sucks soo much that I'd rather have a tiny one hidden in the closet :)
16:34.54[TK]D-Fendercoppice: But is is..... Plug & play? ;)
16:35.56puzzledah, coppice must be off doing that special thing with his tv
16:36.45puzzled[TK]D-Fender: no deal. I'll never set a foot in Nevada again
16:37.06piftzafrir: I just rebuilt your packages with misdn support without problems
16:37.28puzzledpif: but did it pass a call?
16:37.57pifnot yet, next thing
16:37.59*** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com)
16:39.20[TK]D-Fenderok, heading out for a bit, back later
16:40.09Mercestestake care fender.
16:41.11*** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net)
16:48.15spanglesontoastwtf
16:48.19spanglesontoastI just rang japan
16:48.27spanglesontoastwell I'm getting somewhere :)
16:48.54pifmushi mushi?
16:48.55*** join/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br)
16:49.11Mercestesspanglesontoast, Are you ever going to post yoru error/sip.conf or are you just going to trol forever??
16:49.19__freedom__loverhi all, good afternoon
16:49.28aydiosmiocan someone point me to the Remote-Party-Identity configuration lines for *?
16:49.28MercestesAfternoon, freedom.
16:49.37b11danyone know of a good 4+ port KVM that supports both DVI & VGA ?
16:49.38Mercestesaydiosmio, Is that anything like callerID?
16:49.41__freedom__loversomeone here uses asterisk on freebsd?
16:49.45b11di do
16:49.52Mercestesb11d, netgear makes one.  Really good
16:49.56b11doh yeah?
16:49.59aydiosmioMercestes: kind of, it's a proxy identifier like Asserted-Identity
16:50.12Mercestesaydiosmio, Is that an ss7 thing?
16:50.15*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
16:50.18__freedom__loverb11d: you have a asterisk running on freebsd?
16:50.22aydiosmiobut I know asterisk doesn't support Asserted Identity natively
16:50.29aydiosmiohave to use SIPAddHeader
16:50.39b11di dont see any netgear kvm's
16:50.46aydiosmioMercestes: no, sip thing
16:50.48b11d__freedom__lover.. yes..
16:50.57Mercestesaydiosmio, I wouldn't know then.  I would put in a feature request because I dont' think that's a feature.
16:51.16aydiosmioI know I've seen it somehwere
16:51.19aydiosmioah! here it is
16:51.28Mercestesb11d, I'm pretty sure it's netgear.  I'll look it up tonight.  They make good 2 port and 4ports.  You can find them at bestbuyand stuff, pretty cheap
16:51.30aydiosmiohttp://voip-info.org/wiki/view/P-Asserted-Identity+and+Remote-Party-ID+header
16:51.42aydiosmiosendrpid = yes
16:51.48b11dboth DVI and VGA though eh
16:52.07MercestesAh cool, thanks.  I added it to my favorites so I can read it later.
16:52.20MercestesSee folks??  Good example!  he finds the answer...then posts it so I can educate myself.  :)
16:52.42Mercestesb11d:  Oh.  try Telepathy.  I honestly have no clue.  ;)
16:52.54MercestesI don't play with DVI KVMs.
16:52.56*** join/#asterisk jm|laptop (n=jm@sentry.flags.co.uk)
16:53.05b11dTelepathy? Is that a VoIP company?
16:53.13aydiosmioMercestes: Verizon requires either RPID or Asserted ID for wholesale termination
16:53.22aydiosmiothat's why I'm setting it up
16:53.56Mercestesb11d:  yea.
16:54.08Mercestesaydiosmio, Cool.  Let us know how Verizon is.  I'm watching that company.
16:54.15b11dneat.
16:54.27MercestesThey're bringing fiber to my area soon.  They have some cool residential stuff.
16:54.33*** join/#asterisk webman (n=adamg@52.87.233.220.exetel.com.au)
16:55.07aydiosmioyeah FIOS is leaking into NYC slowly
16:55.20aydiosmiohopefully Queens will get hooked up bfore I get tired of this city
16:55.54*** join/#asterisk deeperror (n=deeperro@mail.banctel.com)
16:55.54spanglesontoastcome on ;)
16:56.18webmananyone know how to get hylafax to route incoming faxes when received from iaxmodem?? I've seen to use CALLID4 and also tried to use CIDNAME but both are always blank??
16:56.39*** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-117-14.ph.ph.cox.net)
16:57.02*** join/#asterisk elpropagandista (n=elpropag@208-106-57-5.dsl.dynamic.sonic.net)
17:00.32cr4z3dcan anyone help me get my VoIP up and running with NuFone? i've been trying for about a day now. iax show registry shows i'm registered and when i turn on iax2 debug all i see is tx-frame retry and rx-frame retry over and over
17:01.09webmancr4z3d: are you trying to get inbound or outbound working ?
17:01.13cr4z3dinbound
17:01.27*** join/#asterisk dps (n=dps@129.64.30.213.rev.vodafone.pt)
17:01.32dpsHello
17:01.37webmancr4z3d: so when you call your number do you see any rx traffic ???
17:01.48cr4z3dnop
17:02.04*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net)
17:02.28*** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca)
17:02.38dpsCan anyone give me a clue how do i make the rtp pass throu both phones without asterisk staying in the middle?
17:03.07webmancr4z3d: then either you are not registered correctly, or else your firewall/something is blocking the traffic
17:03.13cr4z3dwebman, does that mean nufone is screwed up?
17:03.20cr4z3dor that they can't get it to me
17:03.55webmancr4z3d: I'd usually blame firewall/something.... check what IP asterisk says it has registered with??
17:04.21cr4z3dlast night someone told me to forward udp port 4569 on my router
17:04.25cr4z3di disabled the router spi firewall
17:04.34cr4z3dand ubuntu has no default installed firewall/i never installed one
17:05.04dpscr4z3d: type iptables -L
17:05.06webmantry and use "tcpdump port 4569" to see if you can see the incoming traffic
17:05.21dpscr4z3d: and see if there's any rule
17:05.42cr4z3diptables -L has nothing
17:05.57*** join/#asterisk darylvoip (n=darylvoi@c-71-224-42-97.hsd1.pa.comcast.net)
17:06.08cr4z3dwebman, so call the number and do tcpdump?
17:06.29cr4z3dtcpdump gave me no suitable device found
17:06.33webmanalso try to get asterisk to register, and then ASAP call the number (maybe your firewall has a short timeout)
17:06.50dpsFor the luve of god and madonna and my cousin, i'm about to go get drunk and pissed with everything that as eletricity, i need to make an rtp stream pass by 2 phone that are registered on asterisk and the bastard stays in the middle doing rtp proxy
17:06.56cr4z3doh would QoS settings mess any of this up?
17:06.59darylvoipHey...anyone around that can help me figure out why asterisk is giving me a "Looking for s in default" followed by a 404 in my new config (replaced some openser boxes with a sansay sbc)?
17:07.13dpsNerf asterisk rtp proxy!
17:07.46darylvoipdps: sounds like you have one of my unsolved problems too
17:07.57webmandps: ensure no nat between the phones or asterisk (ie all public IP or all same LAN)
17:08.05dpsSame switch
17:08.09dpssame network
17:08.13dpssame... same
17:08.20dpssame codec
17:08.23webmandps: check sip.conf then
17:08.35dpsyes.... keep going?
17:08.51webmandps: canreinvite=yes
17:08.59webmanfor each of the phones
17:09.02dpsyes
17:09.09Qwell[]nat=no?
17:09.10dpsit's on canreinvite=yes
17:09.14dpsnat is never
17:09.16bulleperhaps the phones themselves doesnt reinvite ?
17:09.32cr4z3dhm.. still can't get any results
17:09.32webmandps: also check what you set for the localnet settings
17:09.49cr4z3dshould i try setting the asterisk as a DMZ
17:09.59cr4z3djust to rule out any nat/routing issues?
17:10.02*** join/#asterisk ownerge (n=sergo@217.147.230.35)
17:10.09dpswebman: localnet in sip.conf is 10.0.0.0/24
17:10.09ownergehi guys
17:10.10Qwell[]cr4z3d: Making it DMZ doesn't give it a public IP
17:10.12ownergeneed help
17:10.17webmancr4z3d: well, could be worth a try ...
17:10.23ownergehave problem connection SIP phone to my ASterisk server
17:10.29cr4z3doh good point Qwell
17:10.36dpsyes ownerge what's up?
17:10.38ownergecan any1 help me?
17:10.47cr4z3dbut as far as QoS goes
17:10.48webmancr4z3d: basically your first step is to try and receive *something* ....
17:10.59cr4z3dyeah i know i can't get anything
17:11.02cr4z3dbut it says i'm registered
17:11.23ownergelook
17:11.32ownergeI have Cisco 7906 phone
17:11.37webmancr4z3d: if it says you are registered, then you must have had *SOME* two way traffic....
17:11.39Qwell[]7906?
17:11.41ownergei've created an XML file
17:11.46ownergeuploaded to phone
17:11.51ownergetrying to dial
17:11.55ownergenothing happens
17:12.03ownergephone  says that it tries to register
17:12.08dpsasterisk .-rvvvvvvvvvv
17:12.12webmananyone using iaxmodem and know how to set the CID info for faxdispatch to route properly??
17:12.15dpssip show peers
17:12.16cr4z3dyeah definitely says registered under iax2 show registry
17:12.17phearlesshow can I do a (shell?) script that get the list of the "sip show channels" ?
17:12.20dpssip show register
17:12.21ownerge1 sec
17:12.24ownergewill check now
17:12.28dpsgood
17:12.28phearlessis there a command to launch CLI commands?
17:12.35phearlessthat I can use in a script
17:12.36*** join/#asterisk s1gny|wrk (n=s1gny@p54916365.dip.t-dialin.net)
17:12.40dpsasterisk -r
17:12.42*** part/#asterisk s1gny|wrk (n=s1gny@p54916365.dip.t-dialin.net)
17:13.01webmanphearless: asterisk -rx 'some command'
17:13.38dpsgod! oh GOD! doesn't the sip idea is to make the rtp port of each user agent pass in the sip header and let the damnn phone talk to each other?
17:13.41cr4z3di do have QoS enabled on my router though you think that would do anything webman?
17:13.48ownerge911/911                    (Unspecified)    D   N      0        UNKNOWN
17:13.52ownergethat's what I got
17:14.02ownergeit's a my test extention :D
17:14.02phearlessthanks webman
17:14.03ownerge911
17:14.17dpsuse qualify=yes
17:14.30dpsto see if it's reachable
17:14.39*** join/#asterisk Assid (n=assid@59.183.60.24)
17:14.39webmancr4z3d: I have no idea in relation to your config.... but at some point you need to look at the traffic / network level before trying to go further. I like tcpdump for that....
17:15.15cr4z3dtcpdump gave me nothing when i tried it on 4569
17:15.21ownergeit is set to yes
17:15.26ownergealso
17:15.29ownergeforgot to tell
17:15.31bullecr4z3d: is surely must atleast show outgoing traffic
17:15.51*** join/#asterisk xo8ox (n=pride_32@wsip-66-210-250-2.ph.ph.cox.net)
17:15.57ownergeI've tried this extentiuon from VIOP pc client
17:15.58ownergeit works
17:16.12ownergecan be there any problem regarding dialplans or something like that
17:16.13ownerge?
17:16.38cr4z3dbulle, tcpdump 4569 gives me no suitable device found
17:16.53bulleuh ?
17:17.38cr4z3dsame with tcpdump port 4569
17:17.46webmancr4z3d: are you using linux?? what network devices do you have ??
17:17.55*** join/#asterisk phillipk (n=pkey@216.248.143.77)
17:17.58cr4z3dooh shit
17:18.05cr4z3dwasn't running it with root
17:18.06Assiderr.. can someone help me with something
17:18.07cr4z3di thought i was
17:18.15Assidall of a sudden my polycom doesnt want to register anymore
17:18.28cr4z3dalright it's listening on it
17:18.33cr4z3dand i'm trying to call in
17:18.43cr4z3dnothing shows up
17:18.46webmando a iax2 reload
17:18.57ownergeI'm mad
17:19.03ownergethis thing still not working
17:19.03cr4z3dnothing
17:19.03ownerge:@
17:19.17webmanyou should at least see the outbound register request
17:19.20cr4z3d-- Registered IAX2 to '66.225.202.72', who sees us as 70.162.117.14:4569 with no messages waiting
17:19.25cr4z3dis what i got in cli
17:19.28webmanhow many network devices do you have ??
17:19.31cr4z3dand nothing from tcpdump
17:19.34xo8oxguys I setup a que in *now but when I call the que I only hear the music on hold !
17:19.40cr4z3da wireless and an ethernet
17:19.41*** join/#asterisk l2cache (n=ghansen@64.128.254.98)
17:19.44cr4z3dit's on wireless right now
17:19.52cr4z3dwhich is eth1
17:19.54*** part/#asterisk elpropagandista (n=elpropag@208-106-57-5.dsl.dynamic.sonic.net)
17:20.05webmanso you need to do tcpdump -i eth1 port 4569
17:20.08cr4z3doh of course tcpdump is on eth0
17:20.32Assidi keep getting back SIP/2.0 401 Unauthorized
17:20.54cr4z3dok now it captured some packets
17:21.10webmanso now try to call in and you should get more packets
17:21.29*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)
17:21.43cr4z3dyep i got some packets when i called in
17:21.53webmanBTW: in my experience, some firewalls will drop the translation rules in under 30 seconds.... so you gotta be quick :)
17:22.01ownergelook
17:22.06ownergei have only 1 question
17:22.09webmancr4z3d: ok, did you see anything from asterisk ?
17:22.13ownergeif phone is not registered on Asterisk
17:22.20ownergehe can't create package
17:22.24ownergeright?
17:22.38cr4z3dasterisk cli gave me nothing
17:22.42webmanownerge: no, not registered means can't receive calls, but the phone could make calls
17:22.49webmancr4z3d: iax2 debug
17:22.51cr4z3dand the tcpdump is coming up with stuff every few seconds
17:23.03ownergemine can't do anything :D
17:23.07ownergecan't call
17:23.09webmancr4z3d: and make sure you started asterisk with asterisk -vvvvdc
17:23.13ownergecan't receive'em
17:23.21ownergethis is what i got in output
17:23.22cr4z3dwebman, with that on i get a bunch of tx-frame retry results every couple of minutes
17:23.24*** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com)
17:23.31webmanownerge: then you have a config error
17:23.38cr4z3dwebman, i have asterisk -vvvc and then did set debug 10 and set verbose 10
17:23.39ownerge911/911                    (Unspecified)    D   N      0        UNKNOWN
17:23.52ownergeother phones work
17:23.55ownergewith same cfg
17:24.12webmancr4z3d: I don't think that is always enough, I think you also need to start with d.... also check /etc/asterisk/logger.conf
17:25.12cr4z3dok i restarted it with -vvvvdc
17:25.18cr4z3dstarts up fine
17:25.39Dr-Linuxhi
17:25.43Dr-Linuxwhat's wrong with this:
17:25.44Dr-Linuxexten => 32345,1,Answer
17:25.44Dr-Linuxexten => h,1,DeadAGI,pager2sms.agi|03339615969\@ufone.com
17:26.04Assiderr.. can someone help me with this polycom registration?
17:26.06*** join/#asterisk thinwires (n=thinwire@24-49-196-96.kntnny.adelphia.net)
17:26.07b11dshoudlnt that be like, 32345,2, DeadAGi ?
17:26.13Assidit just doesnt want to work anymore
17:26.31cr4z3dwebman, i'm in logger.conf should i change anything from the default?
17:26.32__freedom__loverdoes someone use asterisk on freebsd
17:26.35__freedom__lover?
17:26.45Dr-Linuxb11d: DeadAGi only works with h
17:26.47thinwireshi, does anyone here use Cisco 7941G IP Phones?
17:26.50webmancr4z3d: just make sure the console => line has debug as well
17:27.09b11doh
17:27.17webmanDr-Linux: looks ok to me.... why should there be something wrong with it??
17:27.23cr4z3dwebman, nope just says console => notice,warning,error
17:27.31cr4z3dso i should through debug a t the end?
17:27.34webmanassid: try reboot the phone :)
17:27.45Dr-Linuxwebman: bcoz it's not working for me
17:27.47webmancr4z3d: yep add debug at the end
17:27.49Assidwebman: tried that.. even rebooted the modem
17:28.01Assidwebman: http://pastebin.ca/406719
17:28.07webmanDr-Linux: in what way... you haven't provided sufficient information
17:28.13Dr-Linuxi'm calling this extension and on CLI i can only see Answer execution
17:28.31Dr-Linuxwebman:
17:28.32Dr-Linuxnmmm
17:28.35*** join/#asterisk ToyMan (n=Stuart@12.23.30.130)
17:28.39cr4z3dwebman, ok reloaded it with debug in the consol line
17:29.00*** join/#asterisk Mnabil (n=Mnabil@196.202.44.224)
17:29.15webmancr4z3d: now try the iax2 reload and then the inbound call
17:29.55Mnabilhello, i'm installed asterisk , but i can get asterisk-core-sound working with me, ass their ext. is alaw ? who can i got them work or install them
17:30.03Dr-Linuxwebman: why the doc suggested DeadAGI?
17:30.33thinwiresOk, quick question, Cisco 7941? or PolycomIP601?
17:30.35cr4z3dwebman, still nothing but now i get debug messages like Mar 22 10:29:09 DEBUG[16482]: chan_iax2.c:7752 iax2_do_register: Registration created on call 2
17:30.51ownergeI found solution
17:30.55webmanDr-Linux: dunno, to me it looks like you need to call from one channel to another, and when one channel hangs up the other continues to the deadagi
17:30.56*** join/#asterisk Assid (n=assid@203.212.204.107)
17:30.56b11dpolycom 601
17:30.57ownergevery gooooooood solution
17:30.58b11dno questiojn :)
17:31.05ownergewill user analog lines
17:31.06ownerge:D
17:31.10ownergeno VOIP
17:31.14ownergeNO VOIP PHONES
17:31.20ownergeNO PAIN IN ****
17:31.21ownerge:D
17:31.24b11dyeah voip's on it's way out anyways..
17:31.29Dr-Linuxhhm.. :S
17:31.32cr4z3dis it really?
17:31.33b11d:)
17:31.34bulleno pain in 4-letter-word
17:31.36thinwiresb11d: do you know if the polycom IP601 supports XML?
17:31.40b11dyeah it does
17:31.44bulleownerge: what four letter word were you thinking of ?
17:31.44ownerge:D
17:31.45webmancr4z3d: do you see any packets/messages when you call your did ??
17:31.51b11dDICK ?
17:31.55b11dBUTT?
17:32.01b11dASSS ?
17:32.02b11d:P
17:32.04ownergeactualy i wanted to type 3 letters
17:32.05ownerge:D
17:32.12cr4z3dwebman, no i keep getting the same kind of packets it looks like
17:32.15ownergebut in my conditions
17:32.18webmanPSTN is a four letter work :)
17:32.25cr4z3dwebman, 10:32:10.119359 IP unknown.ord.scnet.net.iax > 192.168.1.109.iax: UDP, length 48
17:32.25cr4z3d10:32:10.119694 IP 192.168.1.109.iax > unknown.ord.scnet.net.iax: UDP, length 12
17:32.27webmans/work/word
17:32.33ownergeafter all this pain in a *** with this Cisco 7906
17:32.33b11dwell thinwires..  XML as in how?
17:32.34*** join/#asterisk robin01 (n=robin@010.152.dsl.concepts.nl)
17:32.38b11dwhere do you want to "use" xml?
17:32.39ownergemistakes can be forgiven :D
17:32.40b11din relation to the 601
17:32.51robin01hello all
17:32.56l2cachehi
17:33.01b11dholland for life!
17:33.01cr4z3dwebman, also iax2 debug keeps saying tx-frame retry and rx-frame retry messages
17:33.03Assidwebman rebooted for the 6th time.. lets see
17:33.10robin01I'm having trouble whit installing the D410P
17:33.15webmancr4z3d: ok, at least you know you have inbound traffic to the box....
17:33.31robin01can anybody help me
17:33.43l2cachewhat is your question robin01?
17:34.02robin01well I installed the card and it is detected
17:34.03Assidfinally worked.. yeay!
17:34.19webmancr4z3d: get a log of what happens, as I said, start with iax2 reload and then attempt the call. If you can, edit the log to show where you made the call, and post it to pastebin.ca or something
17:34.21cr4z3dwebman, this wouldn't have anything to do with my iax.conf, sip.conf, or extensions.conf at this point correct?
17:34.26thinwiresb11d: does it support XML phone directories?
17:34.33b11dyeah, it does.
17:34.41b11dall the polycom's use xml phone directories
17:34.44robin01at compiliung asterisk 1.4.2 the misdn driver is not detected
17:34.57cr4z3dwebman, alright i'll get that log for you
17:34.59robin01but I can see that the card is operational
17:35.11webmancr4z3d: well, that is what we should see in the debug output, if it is auth error, then problem is in iax.conf, or it might be no extension / context error
17:35.51robin01does anybody have any experiance with the B410P card
17:35.52robin01?
17:35.56*** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk)
17:35.58cr4z3dwebman, ok cuz yeah i'm not sure about my extension.conf setup/sip setup but at this point i'm just trying to get it connected correctly
17:36.22thinwiresb11d: awesome, thanks, I have a cisco 7941 on my desk now and I want to throw it out the frackin window. Nothing but trouble from starting it up to getting tech support
17:36.30zoarobin01: yes
17:36.39robin01ok great zoa
17:37.11b11dyeah
17:37.15b11dthe cisco phones pretty much suck ass
17:37.17robin01zoa: I followed the installation instruction in the cards manual, but no positive result
17:37.18b11dthey look nice though :)
17:37.24robin01i'm using opensuse 10.2
17:37.34*** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk)
17:37.40thinwiresoh yeah, it looks sexy... but it's like dating a hot girl with a horrible lough
17:38.28l2cachepolys are the best
17:38.34bullenokias are the best
17:38.51l2cachei have a 501 on my desk right now...beautiful
17:38.55webmancr4z3d: hows that debug going?? I gotta go soon, it's almost 5am  :(
17:39.04b11dyeah im all about my 501 too
17:39.04robin01can anybody help ?
17:39.09b11di
17:39.19cr4z3dwebman, i'm pasting the stuff in a pastebin right now
17:39.25b11di want an adjustable backrest, a backlit display, and a built-in ATA and i'd be in paradise
17:39.28*** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca)
17:39.31l2cachewe run call centers with over 150 poly 301's...they are awesome
17:39.33b11dyeah the 601's are a little to pretentious for me
17:39.46b11di dislike the 301s personally.. the 430's are nice.
17:40.07webmanb11d: yeah, but they are great when you don't have to pay for them yourself :)
17:40.11b11daye :)
17:40.44webmanI read there is a new phone 650 or something.... meant to have a better quality codec....
17:40.49b11dyeah the HD audio
17:40.53b11dit's pretty nice
17:41.02cr4z3dwebman, http://pastebin.ca/406746 and thanks for helping me out so much. i'm guessing you're in the UK since it's 5am
17:41.12b11dthe soundstation ip 4000 conference room phone is nice too, just overpriced.
17:41.38Assidbulle: i got a e61
17:41.38thinwireswell for me price isn't a big deal, one time expense... any suggestions on which phone to look at?
17:41.38b11di used to live up near Darwin for a short time.. AU rocks.
17:43.26*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
17:43.29ownergeppl
17:43.32ownergeanother problem
17:43.36webmancr4z3d: at what time did you attempt to call your number though??
17:43.42ownergeI just deleted old extention
17:43.45ownergecreated another one
17:43.47cr4z3dwell the end of 10:37
17:43.49ownergebut
17:43.51b11dthinwires.. get one of each then.  get a 430/501/650
17:43.58ownergewher i use sip show user
17:44.08ownergei can c that old one still exists
17:44.09*** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk)
17:44.15ownergebut new one doesn't show up
17:44.36thinwireslol, I just need one bloody phone to pick here, i'm looking into this HD Voice thing now, looks sexy
17:44.40ownergetyped extentions reload
17:44.47drakozaptel is independent of asterisk right
17:44.49cr4z3dwebman, called at 10:37 then it turned 10:38 so around 10:37:5something
17:45.09drakoi can have zaptel 1.2.16 with asterisk 1.2.14
17:45.10drako?
17:45.23webmancr4z3d: you are calling your number from a PSTN or mobile or something ??
17:45.30cr4z3dwebman, from a mobile
17:45.53webmancr4z3d: do you get any signal/tone/rva/something on the mobile ?
17:46.14cr4z3dwebman, nothing just silence followed by "the caller is currently unreachable"
17:46.22xo8oxguys when I do 'asterisk -r' i get this: Unable to connect to remote asterisk
17:46.29webmancr4z3d: is the number assigned by nufone or is it a ported number?
17:46.40xo8oxhow can I run asterisk ?
17:46.44webmanxo8ox: try asterisk -c instead :)
17:46.50cr4z3dwebman, also if i login to nufone.net i can see all the calls, and it was assigned by nufone
17:47.19*** join/#asterisk Vec (n=Vec@dsl-244-215-172.telkomadsl.co.za)
17:47.33xo8oxI did and I got bunch of warnings
17:47.37webmancr4z3d: well, there is no traffic at all generated in relation to the inbound call (that I can see).... so something is screwed
17:48.00cr4z3dwebman, should i contact nufone? or would it be my router blocking
17:48.09webmanxo8ox: warnings are not errors... fix the warnings when you are ready to learn more
17:48.24webmancr4z3d: what kind of router is it ?
17:48.53cr4z3dwebman, linksys wrt54g w/ DD-WRT firmware
17:49.03webmancr4z3d: can you make outbound calls through nufone ?
17:50.38cr4z3dwebman, i have no idea never tried.. i haven't setup any of the sip stuff yet
17:50.38webmancr4z3d: ok, simpler option... download and install diax and use that to register to nufone, and test..... that will rule out asterisk config
17:50.38aydiosmioso is Asseted-Identity supported in * yet or are we still adding out own headers?
17:50.38aydiosmioAsserted
17:50.38aydiosmiothis is a pretty important feature now
17:50.39webmanaydiosmio: important for who.... :)
17:50.39*** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net)
17:50.44cr4z3dwebman, ok so if i use diax that means my asterisk config is good ?
17:50.45aydiosmiofor people connecting to wholesale voip providers
17:50.55Mercestesfor people connecting to Verizon you mean.
17:51.06aydiosmioVerizon Biz support Remote-Party-ID but it's deprecated
17:51.27MercestesUse Telepathy.
17:51.28cr4z3dwebman, ah it's for windows i'll have to get my laptop out
17:51.32webmancr4z3d: no, but it means you are testing the router/internet without all the complex asterisk/linux config :) just a simple windows box with diax which has very few settings
17:51.36aydiosmionah there's a few wholesale voip providers requiring it now
17:51.46Mercestesaydiosmio, I would put in a feature request.
17:51.50aydiosmiome too
17:52.02aydiosmiowhere the heck do I do that?
17:52.03cr4z3dwebman, oh alright so basically at that point i know it's not the router if that works
17:52.24webmancr4z3d: yep, then you know if it also not nufone problem :)
17:52.55*** join/#asterisk [shodan] (n=shodan@ip213.99-113-216.pppoe4.joliette.intermonde.net)
17:53.06webmancr4z3d: I gotta get going now.... but good luck with it! hopefully someone else can help you out soon...
17:53.20Mercestesaydiosmio, bugs.something.com or something.  google asterisk bugs
17:53.31b11dbugs.digium.com
17:53.36*** join/#asterisk shinux__ (n=shinux@86.62.8.178)
17:53.46chefrsAny idea when I dial an outbound trunk, about 50% of the time it rings once and then just sits there?
17:53.48cr4z3dwebman, thanks, enjoy your sleep haha
17:54.03b11dthat's strange eh chefrs.. what does the console report?
17:54.30Mercesteschefrs:  Because you have a random number generator that 50% of the time does a Dial and the other 50% of the time it does a Ringing(6) and then does an endless loop routine??
17:54.50chefrsMight be it.
17:54.55MercestesYES!
17:54.57Mercestesman I'm good
17:55.00*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
17:55.06xo8oxasterisk wont run
17:55.08chefrsb11d: Reports a dial out, says a zap channel has answered it, but then the phone just... sits.
17:55.11MercestesAll without logs of a failed call and a copy of the configuration.
17:55.20chefrsb11d: Looks just like a normal, successful outbound call.
17:55.27b11dand the phone rings once, and then stops.
17:55.31chefrsYeah.
17:55.32Mercestesxo8ox:  we know.  we're sorry.  We hope to have a usable product some day.
17:55.36b11dis this possibly a reinvite issue?
17:55.43b11ddo you have canreinvite=yes or =no in sip.conf?
17:55.50chefrsHmm, haven't touched on anything like that. Lemme check.
17:56.07xo8oxlol
17:56.17b11di dont know enough about how that works to say for certain if it is or not.
17:56.25chefrsreinvite is no
17:56.40chefrsBut it's in extensions, not sip
17:56.43Mercestesxo8ox  Could yyou maybe type out an asterisk -cvvvvvvvvvvvvvvvvv and pastebin what that output says maybe>???
17:56.46xo8oxguys seriously I wish I was linux guro but the person who set up our asterisk line quit and now we are stock
17:56.56Mercestesit's guru.
17:57.02xo8oxguru
17:57.03xo8oxhehe
17:57.08xo8oxwhats the difference
17:57.10b11doh
17:57.11Mercestesand stuck
17:57.21Mercestesand I have a resume.  but I'm expensive.
17:57.25l2cacheso your place is hiring?
17:57.28Mercestesand I need the output of asterisk -cvvvvvvvvvvvvvvvvvvv
17:57.33l2cachewhere
17:57.36Mercestesfrom your command line, please.
17:57.38Mercestesin pastebin.
17:57.41Mercestes~pastebin
17:57.51jbothmm... pastebin is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well
17:57.51b11dumm..  reinvite stuff should be in sip.conf according to what I have..
17:58.13chefrsb11d: What is your reinvite set to?
17:58.16b11dno
17:58.18chefrsb11d: Also, what does it do?
17:58.31wunderkinxo8ox, i'm local if you need someone to come out
17:58.32b11dit has to do with handling traffic from phone to phone, or from phone to asterisk, to phone.
17:58.35b11dif I remember correctly.
17:59.25chefrsAlright well I'll set it in there and see if it starts to work a bit better.
17:59.35b11dyeah im not even sure it's related to the problem..
17:59.44chefrsWorth a shot.
17:59.47b11dsure
17:59.54b11dread voip-info.org's page on sip.conf too
18:00.03spanglesontoasthmm to use meetme do I need zaptel ?
18:00.24chefrsAlso any idea why when I dial out, the phone company says to add a 1 to the #, but Asterisk does add a 1?
18:00.27chefrsHeh.
18:00.39b11dlook at your extensions.conf
18:00.48b11dsee if it's adding a 1 or not.
18:00.52chefrsIt is.
18:00.54chefrsI can see it in CLI
18:00.55b11dok.. so whats the prob
18:01.04b11dohhh
18:01.04chefrs*shrugs*
18:01.07chefrs<PROTECTED>
18:01.13b11dit still asks for the 1 even though 1 is being passed to the telco?
18:01.17chefrsYeah.
18:01.29Mercestesspanglesontoast, Yes, we covered this two days ago
18:01.30b11dweird, dunno on that one.
18:01.33Mercestesspanglesontoast, Are you a bot?
18:01.37spanglesontoastwas yestaday P
18:01.38spanglesontoast;)
18:01.38chefrsRighto
18:01.55aydiosmiothe bug tracker no longer accepts feature requests
18:01.59spanglesontoast~zaptel
18:02.07jbotwell, zaptel is zapata telephony interface. A low level interface designed to abstract hardware access to a variety of devices for BRI, PRI or analogue access.
18:02.07aydiosmioHEY DEVS
18:02.08spanglesontoast:|
18:02.12spanglesontoastwhere can I get it
18:02.13aydiosmioadd P-Asserted-Identity to the SIP confifuration
18:02.15spanglesontoast:(
18:02.19b11daydiosmio.. #asterisk-dev :P
18:02.21aydiosmioyou mother farkers.
18:02.52Mercestesspanglesontoast,  uhh....from ftp.digium.com   or emerge zaptel.  or google zaptel downloads.
18:03.29Mercestesspanglesontoast, really.  you should give up.  You shouldn't be touching modern technology.  Your brain is an empty void of worthlessness that simply sin't absorbing any of this or registering any level of higher conciousnesss.
18:03.40spanglesontoastlol
18:04.07spanglesontoastwhy not just give me a gun
18:04.07spanglesontoast;)
18:04.07Mercestesspanglesontoast, You are an audible amoeba, asking for stimulus and blindly responding to stimulus with no abosorption or processing...nothing more.
18:04.09Mercestesspanglesontoast, Either evolve...or go away.
18:04.14b11dyou're a dick Mercestes :)
18:04.20b11di heart you
18:04.27spanglesontoastbah
18:04.32Mercestesspanglesontoast, Google is at http://www.google.com   It's a very useful tool.  reading is also a useful tool.  The world is at your fingertips...stimulate yourself a little.
18:04.45*** join/#asterisk friedrich| (n=friedric@e177242105.adsl.alicedsl.de)
18:04.48Mercestesspanglesontoast, If you get *STUCK* ...please....ask for help.  I will be glad to help you in your evolution...
18:04.59Mercestesspanglesontoast, Don't ask me to just stimulate you for random reactions.  I have pets for that.
18:05.05*** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br)
18:05.08spanglesontoastlol your sick hehe
18:05.09spanglesontoast;)
18:05.10Mercestesb11d, I agree.  :)  I'm a dick.
18:05.13b11dlol
18:05.19Mercestesspanglesontoast, human pets.   Sicker than you believe.
18:05.30spanglesontoasthehe
18:05.34spanglesontoastgit it done
18:05.35spanglesontoast;)
18:06.12b11dtrite.
18:07.12Strom_Mpossibly moronic question: is there a way to get a phone connected to an fxs port on a zaptel card to reset to dialtone rather than reorder?
18:08.07spanglesontoastyea Mercestes some of the stuff that asterisk has is kinda blowing my mind
18:08.30spanglesontoastI mean being able to dial to a phone line and make a script execute by a prefix that's awesome
18:08.36MercestesStrom_M,   by reset....you mean if it fails a call or you dial an invalid number??
18:08.59Mercestesspanglesontoast, I can format a harddrive with an extension.
18:09.05Mercesteswhat are you talking about?
18:09.08Strom_MMercestes: when the called party unsupervises
18:09.14Ac1dcrawlIn my dialplan, how would I analyze only 2 digits in the incoming number to match an extension?
18:09.49MercestesStrom_M, Man you telco ppl.  Should give blankness for about 60 seconds or so then reorder.  Kind of "the bells" standard.  This in asterisk??
18:10.26Strom_Myes, in asterisk
18:10.29Strom_Mit's quite simple
18:10.41MercestesStrom_M, Kind of a hack but you might be able to exten => h,1, it and Hangup() then Answer().
18:10.53Strom_Mwhen the called party unsupervises, have a battery drop on the local circuit followed by dial tone instead of reorder
18:10.56MercestesBut your looking at overriding a driver level function I dareasay
18:11.01Strom_Mwell, duh
18:11.14Strom_Mis there a way to set that? :)
18:11.20spanglesontoastbummer I forgot to uncommet ztdummy
18:11.28Mercestescan't do a h,1,Hangup h,2,Answer?
18:11.30b11dtotally bummer
18:11.38Mercestesyea, a h,1,1hangup would be retarded but.....you know what I mean.  in a new context.
18:11.41Strom_MMercestes: no, i want to do it channel-level
18:11.45Mercestesh1,1Goto(Do my freakystuff,1,1)
18:11.48MercestesOh.
18:11.51Strom_Mdoing it in extensions.conf screws the CDRs
18:11.58MercestesStrom_M,   Can't you just pactch zaptel?
18:12.01b11disnt there some NoCDR command?
18:12.20spanglesontoastah a make all will do it ;)
18:12.22xo8oxhow do you stop and start asterisk ..
18:12.25Mercestesspanglesontoast, I covered that too like 2 days ago
18:12.30b11dhahaha
18:12.36Mercestesxo8ox:  stop now.
18:12.46xo8oxin asterisk -r mode ?
18:13.02spanglesontoaststop now ;)
18:13.04xo8oxhow do you do it from linux command line
18:13.10spanglesontoastplease stop putting that thing in me :P
18:13.24xo8oxI can't get into the CLI
18:13.25b11dnice.. that's getting pulled out of context.
18:14.04Mercestesxo8ox, I would prefer in the channel but yes in the asterisk -r mod...
18:14.37*** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-97-91.ph.ph.cox.net)
18:14.50Mercestesxo8ox   um....killall -9 asterisk
18:14.57Mercestesor /etc/init.d/asterisk stop for me.
18:15.03spanglesontoastoh crud sticks
18:15.11Mercestesbut I embraced the gentoo tastyness and left the Centos behind me.
18:15.16b11dcrud sticks.. that rocks.
18:15.30spanglesontoastlol
18:15.41b11dgo the way of the BSDs.. they taste GREAT.
18:15.55MercestesBSD tastes like nutsack.
18:16.00spanglesontoasthate bsd so sucky
18:16.02b11dyou know what that tastes like?
18:16.03Mercestesyea, some ppl like it....but nobody normal.
18:16.11b11deatith me!
18:16.11aydiosmioBSD is amazing
18:16.15aydiosmioshuddap
18:16.19b11dyes.. yes it is
18:16.21spanglesontoastbsd doesn't have support for flash :)
18:16.24aydiosmioit runs linux binaries better than linux does
18:16.26MercestesBSD is .....yes...amazing is a good word for it.
18:16.27b11dyep.. lets turn this into an OS war :)
18:16.28b11dagain..
18:16.31spanglesontoastkeep it for servers :)
18:16.39Mercestesit's amazing that it ever made it into public.
18:16.39b11dyeah I do run asterisk on a server..
18:16.57wunderkinxo8ox, if you can't asterisk -r, then asterisk isn't running... what happens if you asterisk -vvvcng
18:17.02b11dMercestes.. im coming over to beat you for that one.
18:17.10Mercestesb11d, promises, promises
18:17.15spanglesontoastyea mines running on ubuntu best desktop ever ;)
18:17.17b11dlol
18:17.22*** join/#asterisk RoyK (n=roy@ti211310a080-5748.bb.online.no)
18:17.32spanglesontoastwell my lappy which is downstairs lol
18:17.35Ifaistosgreetings to all
18:17.36MercestesSaybayon is the best desktop ever.
18:17.44spanglesontoastsomeone say fried eggs ?
18:17.46spanglesontoast:P
18:17.53b11dAIX has the best desktop..  maybe QNX
18:17.54b11d:P
18:17.56MercestesUbuntu is the best............whatever it is.
18:18.01Ifaistoswhat caching dns server would you propose for use with an asterisk system ?
18:18.04Mercestesb11d,   .....  your a freak.
18:18.08b11dI use djbdns as an external cache
18:18.08Strom_Mubuntu is the best ubuntu
18:18.09b11dmyself
18:18.17b11dMercestes, thats why you like me so much.
18:18.19*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
18:21.17*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
18:21.40spanglesontoasthmm zaptel hates me
18:21.51b11dyeah they sure do
18:22.06spanglesontoastsomething is stopping it from compiling :|
18:22.12b11di just got off the phone with zaptel and they are pissed because you never return the CD's you borrow
18:22.14blitzrageafternoon all!  Anyone able to enlighten me as to why when you use one-touch recording with the 'W' flag the file is mixed BEFORE the 'h' extension executes, and if you use the 'w' flag, why it is mixed AFTER the 'h' extension executes?
18:23.47*** join/#asterisk HaDAk (n=hans@152.160.16.90)
18:24.06HaDAkHas anyone had experience installing Asterisk on a WRT54G?
18:24.18aydiosmiothe difference between W and w is who hangs up first
18:24.22spanglesontoastlol on a router lol give that up lol
18:24.32HaDAkspanglesontoast: don't be so cocky.
18:24.34HaDAkhttp://www.voip-info.org/wiki-Asterisk+Linksys+WRT54G
18:24.58HaDAknow. has anyone had experience installing asterisk on a wrt54g?
18:25.15*** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)
18:25.23*** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)
18:25.56b11dspanglesontoast.. you've exceeded the maximum number of permitted "
18:26.03b11d"lol"'s in a single sentance.
18:26.04spanglesontoasthuh
18:26.04b11ddoh.
18:26.10*** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
18:26.42IfaistosHaDAk : Well major problem is limited memory
18:26.58spanglesontoastbrb food
18:27.54IfaistosHaDAk : also almost one of the compressing codecs can work (gsm is the exception)
18:28.16HaDAkIfaistos: i have no problem using NAS
18:28.41IfaistosHaDAk : memory=ram
18:28.48HaDAkright.
18:28.53HaDAkit has, what? 15 meg?
18:29.05IfaistosHaDAk : 8
18:29.22HaDAkhmm
18:29.28HaDAkand what's the overhead on asterisk?
18:29.38IfaistosHaDAk : but depends on the model
18:29.50HaDAklemme check and see what i've got.
18:29.52MercestesHaDAk,   no.....I only have experience installing Asterisk on a WRT54GL   not a WRT54G.
18:29.52*** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252)
18:30.55Assidi guess if you use no transcoding.. there should be no issues at all
18:30.56cr4z3dsoftphone for windows, what's a good one
18:30.57Mercestesspanglesontoast, And yes, on a router, baby.  We are uber-133t.  You can't even compile zaptel.
18:30.59Assidulaw-ulaw calls
18:31.05Mercestescr4z3d   Xlite.
18:31.10cr4z3dthanks Mercestes
18:31.29HaDAkIfaistos: this is a WRT54G v2
18:31.30aydiosmiox-lite is good?
18:31.34HaDAkback when they made em nice.
18:31.38aydiosmioI think sjphone is easier to work with
18:31.49*** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br)
18:31.51zoacr4z3d: idefisk!
18:32.05zoaversion 2 prerelease available when you privmsg me
18:32.06zoa:)
18:32.11Mercestescr4z3d:  NP.  You can get the C# source for it too.  I really liked it for that.
18:32.17b11dno wayu
18:32.33cr4z3di'm just looking for a simple one that i can connect to my asterisk server with
18:32.39zoaidefisk!
18:32.55Strom_Mzoa: what does the name mean, anyway
18:32.57zoaMercestes: xlite comes with sources ?
18:33.00zoagood question
18:33.03zoanothing really
18:33.04cr4z3ddo i need to configure sip for idefisk? i'm thinking of just using iax2 to connect if that's even possible
18:33.13zoaidefisk does that
18:33.14Mercesteszoa:  yea, I got sources somewhere.
18:33.14Strom_Mit sounds like....you're making a shish kebab out of an IDE drive
18:33.30zoayeah i know
18:33.35zoaMercestes: WTF
18:33.36zoa:)
18:33.37aydiosmiohow can I do a SIPAddHeader from a .call file?
18:33.42zoafor xlite ???
18:33.50aydiosmioneed to put headers in the INVITE
18:33.58IfaistosHaDAk : v2 has 4MB flash 16Ram
18:34.24Assidgimme gimme
18:34.45Mercesteszoa:  uh..yea...
18:34.55IfaistosHaDAk : http://wiki.openwrt.org/OpenWrtDocs/Hardware/Linksys/WRT54G?highlight=%28OpenWrtDocs/Hardware%29
18:34.57cr4z3dHaDAk, try openwrt white russian
18:35.11MercestesHaDAk, I agree with cr4z3d
18:35.21cr4z3di was going to try that.. but i have a v5
18:35.24MercestesI hear freewrt is good too.  Discussed that over beer last night
18:35.26cr4z3dnot support by openwrt
18:35.38Assidcr4z3d: i think it does
18:35.39cr4z3di'm using dd-wrt instead
18:35.41*** join/#asterisk xtr-II (n=94752345@S0106000c41ed11e1.vf.shawcable.net)
18:35.54Ifaistosmost of the wrt distro's are pretty stable
18:36.06Assidanyone got anything for a BEFW11S4 ? to make it repeater mode?
18:36.11cr4z3dyeah i've been happy with dd-wrt for now
18:36.12HaDAkMercestes: there shouldn't be a huge difference between the wrt54g and the L
18:36.22cr4z3dthere is
18:36.24cr4z3dthe l
18:36.27cr4z3dis the same as v4
18:36.31cr4z3dthe last one to use linux
18:36.35cr4z3dnatively
18:37.22HaDAkIfaistos: sorry, plugging in this router lagged me a bit. white russian you recommend?
18:37.25*** join/#asterisk qdk (n=qdk@80.243.125.204)
18:37.34MercestesHaDAk, Well, there is one very important difference.  There *IS* a wrt54g that you cannot flash with a new image.  The "L" means "linux" which means you can image it.  Otherwise, they are identical.
18:38.00*** join/#asterisk Hmmhesays (n=Neg@24-117-131-41.cpe.cableone.net)
18:38.05__freedom__lover:D i've got it!!
18:38.25IfaistosHaDAk : white-russian or dd-wrt both are good
18:38.25Hmmhesaysanyone ever have any trouble with a telco not passing the name with number for callerid over a pri?
18:38.27Assidi gotta find a way to make my BEFW11s4 go into repeater mode
18:38.43cr4z3dMercestes, as far as i've read a few months ago v5 and v6 can only use the micro versions of dd-wrt
18:38.50b11dHmmhesays..  yeah and it was the telco's end.
18:39.03b11dit took me two weeks to convince them of that when I had the issue here with CP Telecom
18:39.18Ifaistosyeah recent versions are 2MB flash.... there isn't much you can fit in there
18:39.26b11dand it was like "yeah, the NAME is why i needed the PRI"
18:39.37MercestesHmmhesays, It's a telco thing.  Some do.  Some don't.  some charge extra.
18:39.44*** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290)
18:40.47IfaistosHaDAk : what ever you use make a cron script to restart asterisk at night
18:40.47Hmmhesaysyeah telco's are a pain in the @$$
18:40.51b11dyeah they can be..
18:42.14cr4z3dMercestes, when setting up xlite where it asks for authorization user name, what is that in the sip.conf?
18:42.35cr4z3dbecause i already entered a user name
18:42.41Mercestescr4z3d, Pretty sure it's the username in sip.conf and the username in xlite is the [name] in sip.conf
18:42.48Mercestescr4z3d, I make it easy and make them all the same thing.
18:43.28cr4z3dok so [xlite] user=xlite secret=1234
18:43.31cr4z3dshould be enough
18:43.37JunK-Ysome1 here is using asterisk-gui? where do we set to which ami user will it uses?
18:44.30cr4z3doh yeah Mercestes what about the dtmfmode? rfc2833?
18:44.36Mercestescr4z3d, yea.
18:44.43Mercestescr4z3d, Dtmfmode=auto
18:44.48cr4z3doh they have auto?
18:44.49Mercestesalways dtmfmode=auto
18:45.04Mercestesyea, it does what dtmfmode=rfc2833 is *SUPPOSED* to do.
18:45.32*** join/#asterisk RoyK (n=roy@ti211310a080-5748.bb.online.no)
18:47.02cr4z3dMercestes, [xlite] type=friend user=xlite host=dynamic defaultip=ipofcompusingxlite dtmfmode=auto secret=1234, that looks good right?
18:48.46MercestesLooks good to me.
18:49.02*** join/#asterisk Meaty` (n=meaty3@office.abi.ca)
18:49.03[TK]D-Fendercr4z3d: context=thecontextthatsayswhatyoucandial <_
18:49.06MercestesHaDAk, Turn your away messages and your auto away stuff off.
18:49.16[TK]D-Fendercr4z3d: defaultip= worthless idea, remove...
18:49.17Mercestesthatnks, D.  WB
18:49.46[TK]D-FenderMercestes: y0
18:50.56cr4z3dMercestes, wow good call completely forgot the context
18:51.09MercestesI didnt' call that, D-fender did.
18:51.16Mercestesthank him.  :)
18:51.43cr4z3doh wow
18:51.48cr4z3di didn't even read the name
18:51.52cr4z3dthanks [TK]D-Fender
18:52.08*** part/#asterisk deeperror (n=deeperro@mail.banctel.com)
18:52.31*** join/#asterisk sysreq (n=sysreq@86-198-0-72-ppp.3menatwork.com)
18:53.07[TK]D-Fendercr4z3d: np, good to hear we didn't lose you from yesterday :)
18:53.42cr4z3d[TK]D-Fender, haha i'm still not being able to make calls in so i decided to see if i could call out
18:54.21[TK]D-Fendercr4z3d: .... and?
18:55.18cr4z3d[TK]D-Fender, still in the process of setting it up, exten => _1NXXNXXXXXX,1,Dial,IAX2/cr4z3d@NuFone/${EXTEN} does that look like a good ext? that's what nufone gave me
18:56.24[TK]D-Fendercr4z3d: Doesn' look quite right
18:56.42[TK]D-Fendercr4z3d: Pastebin your sip.conf for them minus PW's
18:57.11cr4z3d~pb
18:57.18jbotit has been said that pb is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste
18:57.23aydiosmio[TK]D-Fender: any idea how to get SIPAddHeader to insert some headers inot the INVTE for an auto-dial (call file)
18:57.57[TK]D-Fenderaydiosmio: You shoveit in your dialplan like anything else.  Show us how YOU'RE trying it right now.
18:59.34cr4z3d[TK]D-Fender, http://pastebin.ca/406846 pasted my extensions and sip conf
18:59.56aydiosmioI have no idea where auto-dial runs from in the dialplan
19:00.19*** join/#asterisk topping (n=topping@204.152.96.238)
19:00.21aydiosmiotried it in the call file context, but of course that won't work
19:00.24[TK]D-Fendercr4z3d: You need to set up a SIP peer for NuFone.  You do not want that in yourdialplan the way you hve it.
19:00.33[TK]D-Fenderaydiosmio: SHOW
19:01.29JerJercr4z3d:  Take a look at the tutorial in the Members Portal
19:02.27cr4z3d[TK]D-Fender, i have nufone peer in the iax.conf i need one in sip too?
19:02.48[TK]D-Fendercr4z3d: oops... yeah... iax2.conf would be kinda nice :)
19:02.49cr4z3dJerJer, that's where i got the extension from and how i setup the iax.conf
19:02.57fbcitIs IAXTel working OK?
19:03.47fbcitRegardless of allow=all in iax.conf I still receive " Call rejected by 69.73.19.178: Unable to negotiate codec"
19:04.54[TK]D-Fenderfbcit: Odds are because it finds a match that includes a codec you CAN'T really do, yet * knows about like G.729 or G.723
19:04.57cr4z3d[TK]D-Fender, here's iax.conf http://pastebin.ca/406854
19:05.41JerJercr4z3d: pastebin the CLI
19:05.54*** join/#asterisk SomethingISODD (n=dan@NTL208H101-91-124.nt.net)
19:05.55[TK]D-Fendercr4z3d: exten => _1NXXNXXXXXX,1,Dial(IAX2/NuFone/${EXTEN})
19:05.55JerJerie show what happens when you actually make a call
19:06.03[TK]D-Fendercr4z3d: exten => _NXXNXXXXXX,1,Dial(IAX2/NuFone/1${EXTEN})
19:06.20[TK]D-Fendercr4z3d: exten => _NXXXXXX,1,Dial(IAX2/NuFone/1555${EXTEN}) <- change accordingly
19:06.20JerJerand you don't really want/need type=friend for [NuFone]
19:06.21SomethingISODDhello all question do i need any special hardware to beable to send sms messages through asterisk?
19:06.33Assid[TK]D-Fender: hjow much do you like polycom ?
19:06.43JerJerand you do not register
19:06.52JerJerPLEASE read the tutorial
19:06.57cr4z3dJerJer, i had them seperate before as the tutorial said
19:07.06Assidmy 301 just managed to piss me off TOTALLY.. it all off a sudden decides not to register anymore
19:07.15cr4z3dJerJer, i just put htem as one cuz i saw it somewhere online and it was less lines
19:07.18[TK]D-Fenderfbcit: Disallow=all , allow=ulaw , allow=alaw, allow=gsm
19:08.01JerJerdon't follow other examples - follow the one that NuFone provides
19:08.24cr4z3dJerJer, so the register => line is not needed?
19:08.38[TK]D-FenderAssid: Polycom is great.  Aastra is very decent as well and I hope to have a better idea shortly.
19:08.39JerJersimply to place outbound calls, no
19:09.10fbcit[TK]D-Fender:I get the same error after making those changes.
19:09.52*** join/#asterisk Assid (n=assid@203.212.204.107)
19:10.06cr4z3dJerJer, so if i remove register => from [general] it won't make any difference at all? and i'll just copy the rest from the tutorial?
19:10.15[TK]D-Fenderfbcit: And did you apply them?  perhaps you should enable debug for the protocol you are working with so as to see what they are offereing...
19:10.28Assid[TK]D-Fender: for some strange reason i cant get my polycom every now and then
19:10.31Assidsoftphone works perfect
19:11.01*** join/#asterisk joe (n=nnjsauer@ip66-107-33-195.z33-107-66.customer.algx.net)
19:11.04zoawhich one ?
19:11.09zoaidefisk ?
19:11.22JerJercr4z3d:  correct
19:11.29fbcit[TK]D-Fender:I did a 'iax2 reload' from the console. I'll enable debug and try it again.
19:11.42cr4z3dJerJer, ah wait nvm it says to have that for the incoming call config
19:11.50Assiddidnt get a chance to use that yet.. but xten for now
19:11.51JerJernot to switch-1
19:11.59cr4z3doh wow
19:12.06cr4z3di didn't see that tutorial the first time
19:12.21cr4z3dmaybe this is why i'm having so many difficulties haha
19:12.22*** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net)
19:15.04fbcit[TK]D-Fender:OK. I have some debug output. Shows perfered codec as ulaw. It does an AUTHREQ, an AUTHREP, and then a REJECT with the reason being the cause mentioned previously.
19:15.14cr4z3dJerJer, exten => 4022160528,2,Dial,<Something valid> <-- something valid meaning an extension setup to one of my sip phones?
19:15.27fbcit[TK]D-Fender:I can paste it somewhere if you want to see it.
19:15.33*** part/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br)
19:16.25fbcit[TK]D-Fender:I am registered OK with the IAXTel server.
19:16.49JerJercr4z3d: sure
19:17.05*** join/#asterisk RoyK (n=roy@ti211310a080-5748.bb.online.no)
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19:19.39*** join/#asterisk ClydeGoffe (n=ClydeGof@base/student/clydegoffe)
19:20.44[TK]D-Fenderfbcit: Yes, please pastebin that call + your iax2.conf minus passwords
19:20.54anonymouz666Can I use a function inside Another ?
19:21.38*** join/#asterisk mmgtrans (n=mmg@einstein.transtelco.net)
19:21.59*** join/#asterisk oQPa (n=uawename@33.Red-83-34-60.dynamicIP.rima-tde.net)
19:22.28mmgtransSomebody experiencinf DTMF problems with cisco phones, getting error New DTMF and can not find a free spot in dtmf array on console. Version 1.2.16
19:22.34*** part/#asterisk oQPa (n=uawename@33.Red-83-34-60.dynamicIP.rima-tde.net)
19:22.48cr4z3dholy shit i made a call out
19:24.27[TK]D-Fenderwhee!
19:25.10[TK]D-Fendercr4z3d: http://www.albinoblacksheep.com/flash/weeee.php
19:25.18cr4z3dhaha
19:25.26cr4z3dbut still getting nothing for callng in
19:26.37cr4z3d[TK]D-Fender, http://pastebin.ca/406873 is the cli output i get when trying to make a call in
19:27.21[TK]D-Fendercr4z3d: -- Executing Dial("IAX2/66.225.202.80:4569-6", "6000") in new stack
19:27.22fbcit[TK]D-Fender:Well, I am using the sample iax.conf and missed a later 'disallow all' which apparently negated my earlier allows. Seems to work now. Thanks for the help.
19:27.34[TK]D-Fendercr4z3d: That is not a valid way to dial anything
19:27.44cr4z3doh right
19:27.54cr4z3dhow would i make it dial that extension
19:28.03cr4z3di'm so bad with these extension things
19:28.33*** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-194-128.tx.res.rr.com)
19:28.39*** join/#asterisk toerkeium (i=oo@201.216.206.221)
19:28.42[TK]D-Fendercr4z3d: Mor appropriate might be : exten => 4022160528,2,Dial(SIP/xlite,20)
19:29.10cr4z3dyeah that's true but eventually i plan on having more than on extension that i can dial
19:29.29cr4z3dbasically i want to dial in and enter some options
19:29.31cr4z3deventually
19:29.37cr4z3dfirst things first haha
19:29.52spanglesontoastyea the question is why can't I comple zaptel
19:30.52[TK]D-Fendercr4z3d: Correct.  For now, let it ring 1 phone and do "whatever".  Then you can move on and learn how to make IVR's, etc.
19:31.23cr4z3d[TK]D-Fender, true i'll work on that stuff later then
19:31.24[TK]D-Fenderspanglesontoast: Sorry, this channel hasn't gotten its copy of res_psychic.so .....
19:32.15IOscanneris there a way to use precense from a cisco phone
19:32.54[TK]D-FenderIOscanner: Only is used with SCCP
19:33.06[TK]D-FenderIOscanner: one reason I don't suggest them
19:33.32cr4z3doh man i'm so happy i can call and recieve calls
19:33.53IOscannerThey have DND can't this not be used to set presence in the flash OP Panel
19:35.16spanglesontoastbah
19:35.41*** join/#asterisk Evil_Lyra (n=Evil_Lyr@viper.pop-pr.rnp.br)
19:35.48*** join/#asterisk kratzers (n=kratzers@martha.pa.net)
19:36.12cr4z3ddoes anyone know if all the sounds are installed by default using apt-get install asterisk on ubuntu?
19:36.16[TK]D-FenderIOscanner: * can't know about DND.  It isn't a constant state.  its the phone being told to reject calls and only comes up PER CALL.
19:36.32HaDAki'm working on getting this wrt54g working with asterisk. i've got white russian on it, but what i need to know is what packages i need to put on it from: http://downloads.openwrt.org/whiterussian/newest/packages/
19:37.04Evil_Lyrahi there
19:37.21kratzersis there a trick to getting ChanIsAvail to set ${AVAILSTATUS} to something other than 0 for a SIP peer?
19:37.23spanglesontoastFender you love to talk down to me :P
19:37.35spanglesontoastyou into bondage ;)
19:37.41b11dyou have been asking the same questions over and over spanglesontoast..
19:37.45b11dfor days
19:38.03[TK]D-Fenderspanglesontoast: I'd love to, but I'm all tied up right now....
19:38.18Evil_LyraI´m having a little troube trying to make asterisk transfer a call using "flash"
19:38.44[TK]D-Fenderspanglesontoast: And that wasn't meant to be taken as "condescending", so much as "sarcastic and inspiring".
19:38.54Evil_Lyramy * is connected to a pstn extension of a traditional PBX
19:38.57spanglesontoasthmm
19:39.08spanglesontoastso you help with zaptel ?
19:39.23tzafrir_laptopwe try
19:39.43tzafrir_laptopEvil_Lyra, and?
19:39.57Evil_LyraI wan to transfer a call using Flash(), SendDTMF(<extension>), HangUp()
19:40.00*** join/#asterisk RoyK (n=roy@ti211310a080-5748.bb.online.no)
19:40.06[TK]D-Fenderspanglesontoast: Maybe once you actually show us whats WRONG :)  You see you only said "it doesn't work, why?!?!".  That does not help us to help you.
19:40.12Evil_Lyrabut what´s happening is
19:40.20spanglesontoasthttp://www.pastebin.ca/406893
19:40.23spanglesontoast;)
19:40.28Evil_Lyrathe extensions ring one time only
19:40.53Evil_Lyraand the asterisk hangup the connection with the callee
19:41.01Evil_Lyraops, I mean the caller
19:41.25*** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com)
19:42.26Evil_Lyrawithout asterisk, we only do pick up the phone, press flash, dial extension and put the phone on the hook
19:42.37tzafrir_laptopwhere exactly do you press Flash? On which phone? connected where?
19:42.46[TK]D-Fenderspanglesontoast: Ok, now at least we have something to see.  I unfotuantely don't have any advice for your current state.
19:42.56Evil_Lyraok, from the beginning:
19:43.02Evil_Lyraforget asterisk
19:43.14spanglesontoastwell I got the kernel headers can't think what's missing
19:43.26[TK]D-FenderCan someone spare a sec to see if the reason for spanglesontoast's problem compiling zaptel 1.4.0 here : http://www.pastebin.ca/406893
19:43.27Evil_Lyrasomeone calls, pick the phone, press flash, dial extension, put the phone on the hook
19:43.37Evil_Lyramy pbx then trasnfer the call
19:43.44Evil_Lyraas it was supposed to do
19:44.04Evil_Lyranow, instead of having a phone, lets put a asterisk with a fxo card
19:44.37[TK]D-FenderEvil_Lyra: "show application flash"
19:44.46Evil_Lyraasterisk should Awnser, flash(), SendDTMF, and HangUp rigth?
19:44.58[TK]D-FenderEvil_Lyra: Sounds about right
19:45.04tzafrir_laptopspanglesontoast, why do you need zttranscode?
19:45.06Evil_Lyrabut doesnt
19:45.12tzafrir_laptopDo you have a transcoder card?
19:45.22Evil_Lyrano
19:45.23spanglesontoastnope
19:45.30spanglesontoastI just want the ztdummy for meetme
19:45.40tzafrir_laptopconsider just disable building zttranscode
19:45.57tzafrir_laptopgrep for transcode in the Makefile and disable it
19:46.24cr4z3dhow come i get a permission denied when trying to launch an AGI script?
19:46.27*** join/#asterisk Idle (n=brian@S010600a024969312.ed.shawcable.net)
19:46.32*** part/#asterisk acidchild (i=ash@unaffiliated/acidchild)
19:46.37JunK-Ycr4z3d: chmod +x script.agi
19:46.41Evil_Lyra[TK]D-Fender:
19:46.41SomethingISODDanyone know if i can SMS messages out through a VOIP line and recevie them also through a VOIP did?
19:46.41Evil_Lyra[Synopsis]
19:46.42Evil_LyraFlashes a Zap Trunk
19:46.45JunK-Yor a chown.
19:46.45Idleis there a nice list of IAX providors? I am trying to find one in canada..
19:46.47Nganyone about who's on a UK ISDN line? I can dial UK numbers, but not special ones like 123 or 118118, and I can't dial internationally.... any suggestions? :)
19:47.08zoawww.voipcharges.com
19:47.20zoathe page is a bit ugly though
19:47.35anonymouz666i already put saynumber() in a loop to say some numbers....100,200,300,400 - what do you suggest for me to do something like: press 1 for 100, press 2 for 200 - how can I mix the apps to make this possible?
19:47.42spanglesontoastok ty tza
19:47.48zoa11 there
19:47.53zoabut you could take one from any country
19:47.54Evil_Lyra[TK]D-Fender: flash seems to work, because I get one ring on the extensions that I´m transfering to
19:47.54zoaanywhere
19:48.25Evil_Lyraif I put a "Wait(30)" before hangup, I can hear the extension ringing across the room
19:49.10Evil_Lyrabut as soon as asterisk hangup it drop the connection with the original caller
19:50.24Idlehm, a best of 3 / 5 rating... thats not good
19:50.50Powerkillsomeone find a solution for this bug : http://bugs.digium.com/view.php?id=8923 ?
19:51.29zoaidle, no need to take that rating into account
19:51.36zoamostly competitors giving bad credits probably
19:51.37zoa:)
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19:52.06b11dPowerkill.. i've only seen that error come up when I've had a SIP peer "registred" but then isnt actually available.
19:52.50Idle:P
19:52.53tclarkdoes anyone know of another co that makes a impedance matching device like this
19:52.54Powerkillb11d I have this error lot of time since I upgrade to 1.2.14
19:52.55tclarkhehe that looks interesting http://sandman.com/echo.html LINE IMPEDANCE MATCHER
19:52.58IdleI mainly need a recomendation
19:53.01CunningPikeHmm - trying to upgrade a 1.4.1 install to 1.4.2 and 'make menuselect' won't let me select chan_zap
19:53.02Powerkillso I've back revert to 1.2.13
19:53.20*** join/#asterisk chefrs (n=joe@c-24-8-226-145.hsd1.co.comcast.net)
19:53.22Idlesomeone stable.. dont specificly need anything more then number portability (is that possible in canda?)
19:53.29PowerkillBut I need to get 1.2.17 for all the bug fix and I don't know how to correct the problem
19:53.47chefrsAny idea why my voicemail doesn't use the recorded message and instead uses the prerecorded gal that came with the system?
19:54.02Idlechefrs: did you set your flags correctly?
19:54.10chefrsWhat flags? Inbound?
19:54.50Idlewell
19:54.53Idlein your extensions
19:55.48Idlehttp://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+VoiceMail
19:56.42spanglesontoasthmm what would the build line be if I only wanted to compile zaptel with ztdummy support
19:58.57Mercestesspanglesontoast, cat makefile | sed s/#ztdummy/ztdummy/g > makefile && make clean && make && make install
19:59.22spanglesontoastnice
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19:59.29wunderkins/makefile/Makefile/
19:59.32wunderking
19:59.35spanglesontoastyea
19:59.37spanglesontoastnoticed ;)
20:00.18*** join/#asterisk steve___ (n=steve@kit-dhcp1.porchlight.ca)
20:00.18b11dsure
20:00.58Mercesteswunderkin, Thanks.  :)
20:00.59*** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net)
20:04.03*** join/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com)
20:04.18Idlehmmm
20:04.22spanglesontoasthmm that didn't work
20:04.32Idledoes Vonage let you connect asterisk, or linksys desk phones, to their service?
20:05.06MercestesIdle:  I think Vonage only wants you to use their stuff.
20:05.56brad_msswprobably want the 'business plus' plans if you plan on using your own equipment with vonage
20:06.01IdleMercestes: thats what I though
20:06.08m4rkl4rAre there techniques for making realtime extensions.conf faster?  When the dial plan goes from, say context a, exten 1 to context a, exten 2, I'm seeing delays measured in seconds.
20:06.14IdleI just had a guy call and wanted to know that, cause he wants to use a linksys phone
20:06.31Idlesadly, theres no good IAX/SIP providors in canada here... maybe one of these will work tho
20:06.32spanglesontoastdidn't understand what wunderkin did
20:06.50m4rkl4rthe database is postgres in a machine on the same gigabit switch
20:07.07brad_msswIdle: https://subscribe.vonage.com/smb-subscribe/index.htm?smb_id=acns  .. you can use your own PBX with those plans
20:07.43Idlejesus
20:08.00MercestesIdle:  Teliax?
20:08.08IdleMercestes: canada
20:08.22wunderkins/spanglesontoast/spoogesontoast/
20:08.23Mercestesspanglesontoast, Use a captial M instead of a lowercase M
20:08.23wunderkinwhat about that?
20:08.28Idleits kinda the trump card, sadly
20:08.35Idleanyhow, I've gotta get to class.....
20:08.36MercestesIdle, they can't do Canada?
20:08.37Idlebbl
20:08.54Mercestess/spanglesontoast/incurable_troll
20:08.58spanglesontoasts/#ztdummy/ztdummy/ was it mean't to be part of this
20:09.09Mercestesyes it was
20:09.16wunderkins/Mercestes/mercestes/ :P
20:09.24Mercestesahh, no!
20:09.43spanglesontoastyea but now make says no targets stop
20:09.59spanglesontoastcat Makefile | sed s/#ztdummy/ztdummy/g > Makefile && make clean && make && make install
20:10.21cr4z3dhow do i make it wait a few seconds after answering?
20:10.28cr4z3dwait(x seconds)?
20:10.29Mercestesspanglesontoast, Are you even in /usr/src/zaptel<tab>/ ?
20:10.38Mercestescr4z3d, Yes.
20:10.46Mercestescr4z3d, Just use one to fix the voice cutoff.
20:11.02spanglesontoasterm I'm in /home/edd/zaptel-1.4.0
20:11.04spanglesontoast;)
20:11.04cr4z3d1 is good for that?
20:11.05cr4z3dalright
20:11.43*** part/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com)
20:11.53cr4z3doh and is there an easier way to reload the extensions.conf file
20:11.58cr4z3dbesides stoping/starting asterisk
20:12.03*** join/#asterisk TechJournalist (n=tk421@d141-129-92.home.cgocable.net)
20:12.03b11djust "reload"
20:12.13aydiosmiodoes asterisk dial out form a context when a call file is used?
20:12.16aydiosmiofrom
20:12.16b11dit wont interrupt anything, and isnt a shutdown&restart
20:12.38aydiosmio(not the Context=, but whereever Dial() is executed for the Channel=)
20:13.01Mercestesspanglesontoast, .....
20:13.10spanglesontoastwell that's the src
20:13.36spanglesontoastdidn't know everyone moved it /usr/src
20:13.37Mercestesspanglesontoast, http://www.voip-info.org/tiki-index.php?page=Asterisk+consultants+USA
20:14.09spanglesontoast?
20:14.24b11dlol
20:14.33Mercestesspanglesontoast, The only page you'll ever need.
20:14.41Mercestesthe answer to all your problems.
20:14.53spanglesontoastdon't have cash...
20:15.09wunderkingoogle and visa, the rest ;D
20:15.10b11dsell your body
20:15.16MercestesSorry, your screweed.
20:15.26spanglesontoastalready sold myself...
20:15.27MercestesI offer an exchange rate for virgins.
20:15.31spanglesontoastnormally for ciggys
20:15.47aydiosmioSo I guess I can't do SIPAddHeader for a call file dial-out eh?
20:15.50aydiosmiothat sucks
20:16.02MercestesI guess that disqualifies you as a virgin so....that means you had better start recruiting.
20:16.31spanglesontoastbut using the command that you did just toasts the makefile
20:16.43Mercestes...
20:17.10spanglesontoastany alternatives to ztdummy
20:17.18Mercestesspanglesontoast, a consultant.
20:17.57spanglesontoasthmm
20:17.58[TK]D-Fenderaydiosmio: Sure you can...
20:18.14[TK]D-Fenderaydiosmio: Show what you've done to dat
20:18.35Mercestesspanglesontoast, What distro are you on, first of all?
20:18.37*** part/#asterisk MarkWD (n=MarkWD@rrcs-67-78-88-186.sw.biz.rr.com)
20:18.42spanglesontoastubuntu
20:19.39Mercestesspanglesontoast, Pastebin your build error
20:20.14spanglesontoaster basically I can't change the makefile to the command you have sent
20:20.38MercestesDude, just replace #ztdummy with ztdummy   zOMG
20:20.51cr4z3doh man ztdummy.. had problems with that yesteray
20:21.01aydiosmio[TK]D-Fender: I'm sorry, I have no idea what to do, I have a stock trixbox install and I have no idea what context calls that originate from a call file use. I can't use the Context= specification because that happens after the INVITE
20:21.02cr4z3ddude i can't believe i finally got it working
20:21.09spanglesontoastoh is it commented out ?
20:21.10Mercestesand pastebin your error
20:21.11*** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner)
20:22.03aydiosmiois there a debug for what path the dialplan takes?
20:22.18b11dyeah "set debug"
20:22.21b11dand watch the console
20:22.21b11d:P
20:22.29b11dand "set verbose"
20:22.42aydiosmioI set them both to 30 and I don't see anything
20:22.57b11dnothing at all?  nothing comes across your console?
20:23.04aydiosmiothe sip debug
20:23.08aydiosmiowhich I have enabled
20:23.14b11dturn it off for a few mins
20:23.22b11dmaybe your other debug info is being lost in that sea of information
20:24.00*** join/#asterisk lukketto (n=lukketto@82.59.103.134)
20:24.18aydiosmio<PROTECTED>
20:24.43aydiosmio> Channel SIP/vs-outbound-08eb0218 was answered.
20:24.58b11dthats all you see?
20:25.04Mercestesspanglesontoast, YES.   we covered that about 50 times.
20:25.14b11d/ignore spanglesontoast
20:25.16b11d:P
20:25.18spanglesontoastnope it ain't there
20:25.23spanglesontoastas commented
20:25.32aydiosmiob11d: that's all there is before the Context is connected
20:25.54Mercestesis it there uncommented?
20:26.00b11dand whats the problem then?
20:26.10Mercesteswould that maybe be because I told you to uncomment it like...2 days ago???
20:26.21aydiosmioI'm trying to figure out where asterisk Dial()s for call files
20:26.28aydiosmioI need to run SIPAddHeader for the INVITE
20:26.33spanglesontoasthttp://www.pastebin.ca/406951
20:26.35spanglesontoastthere we go
20:28.00Mercestesspanglesontoast,  Did you even check the md5 of this src?
20:28.57spanglesontoastnah normally go by size
20:28.57*** join/#asterisk sjaak_yen (n=chatzill@d5c53145.dsl.concepts.nl)
20:28.57*** join/#asterisk zotz (n=zotz@24.244.163.157)
20:28.57Mercestesyes, I know you like the big ones.  Cna you md5 your src please?
20:32.26[TK]D-Fenderaydiosmio: Ok you clearly haven't figured out what you SHOULD be using as your originateing channel.  You are now clearly direct-dialing an outside tech.  You need to dial a LOCAL CHANNEL.  and in there do your SIPAddHeader, then the actualy Dial to call the guy.
20:32.34Mercestesand I suggest you rdl regardless and start over.
20:32.43MercestesThat looks like a src error to me.
20:32.54[TK]D-Fenderaydiosmio: And what you're describing for this is basically an automated page (wakeup call, etc...)
20:33.19aydiosmiobasically
20:33.40aydiosmioah!
20:33.42aydiosmioI see
20:33.48SomethingISODDanyone know if i can SMS messages out through a VOIP line and recevie them also through a VOIP did?
20:33.49[TK]D-Fenderaydiosmio: so your originating channel should look like "Local/12345@myscriptcontext"
20:34.09aydiosmioSo if I use SIP as the originating channel I have no control over it
20:34.10spanglesontoasterm theres no md5 checker
20:34.11spanglesontoast:|
20:34.12aydiosmioI see
20:34.13[TK]D-Fenderaydiosmio: And use 1 entry for the guy to call, and 1 for the actions to take after answer
20:34.24*** join/#asterisk [shodan] (n=shodan@ip181.99-113-216.pppoe4.joliette.intermonde.net)
20:34.51*** join/#asterisk Waverly360 (n=irc@adsl-070-148-122-203.sip.bna.bellsouth.net)
20:36.54*** join/#asterisk keescook (n=kees@ubuntu/member/keescook)
20:37.05*** part/#asterisk Evil_Lyra (n=Evil_Lyr@viper.pop-pr.rnp.br)
20:37.58keescookcan anyone point to the svn commit that fixed the vulnerability mentioned in the 1.2.17 release notes?
20:39.12*** join/#asterisk docelmo (n=vircuser@c-68-45-140-42.hsd1.de.comcast.net)
20:39.28docelmoanyone know how to fix this: frame.c:214 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end
20:39.32docelmoIm using Asterisk 1.4
20:40.11spanglesontoasthang on a sec
20:40.27spanglesontoastwhy does it say 1.2.16 for zaptel and I'm using 1.4.0 :O
20:40.40spanglesontoastin the topic :|
20:41.22spanglesontoastoh
20:41.24spanglesontoast:|
20:42.12[TK]D-Fenderspanglesontoast: Using a mismatched version are we? :)
20:42.43spanglesontoastwell wondered where no one noticed it
20:42.43spanglesontoastlol
20:43.29docelmo[TK]D-Fender do you have any experience with that error?   I have disabled VAD on my switch and its still doing it.
20:43.45[TK]D-Fenderdocelmo: No clue, never touched 1.4 (well ONCE..)
20:44.23docelmoI would prefer not to but not my end decision.
20:44.57spanglesontoastbrb going for a smoke
20:46.22Ifaistosdocelmo : what g729 is the other end using ? g729, g729a g729b ?
20:46.23docelmoI need something stronger..
20:46.49docelmoyes..  I have g729, 729a and 729ab on one end and just asterisk's g729 on the pbx
20:47.07docelmoWell I have all 3 listed for use
20:47.19Ifaistosdocelmo : try g729a
20:47.21Strom_Mdocelmo: turn off VAD on the other end
20:47.27docelmoI killed vad already
20:47.31docelmoI will do that..
20:47.45Mercestesspanglesontoast, http://www.sing365.com/music/Lyric.nsf/The-Headless-Waltz-lyrics-Voltaire/554E5ADB2197F70D48256DAA00253BF2
20:50.52Waverly360I'm attempting to use a custom script to pipe audio from an external sound device for music on hold.  The script is executable, and when I run it, garbage gets piped to my screen from the audio.  So the script should be working fine.  However, when asterisk attempts to start music on hold, it immediately says that music on hold has been stopped.
20:51.25Waverly360I'm not getting any error messages or anything.
20:51.33Waverly360I have verbosity and debug set to 99
20:51.47Waverly360Is there another place where errors might end up?
20:51.56NgBT ISDN30e users - any hints for settings to make dialling international numbers possible? :)
20:53.19*** join/#asterisk dj-fu (n=ajc@unaffiliated/dj-fu)
20:54.21florzNg: Have you tried changing the pridialplan setting in the zapata.conf?
20:54.38docelmothe g729ab was causing the issue..  Thanks Ifaistos
20:54.51Ngflorz: I tried setting it to local, but I'm wondering if unknown would have been a better setting
20:56.11Ifaistosdocelmo : not all g729 are created equal :)
20:58.43Ngflorz: do you know of any other things I can/should test? I have no net access at the site in question so I'm trying to stock up a bunch of things to try
20:59.08florzNg: that ISDN line is not connected to the PSTN?
20:59.36Ngflorz: no it's on the PSTN, it's a regular BT ISDN30e
21:00.01florzNg: I mean, how come that you don't have net access then?
21:00.03anonymouz666anyone know in what package i have the word "for" recorded
21:00.04anonymouz666?
21:00.11Mercestesspanglesontoast, Even better.  http://www.tsrocks.com/g/gerhard_schoene_texts/spar_deinen_wein_nicht_auf_fur_morgen.html
21:01.04Ngflorz: well I suppose I could get a dialup account and then figure out how to get pppd to talk to the E1 card, but that seems like a lot of faffing about and liable to interfere with taking the asterisk server up/down a lot to test various things
21:01.45NgI have been promised the 100Mb LES would be live every day this week, so I was holding out my hopes for that ;)
21:01.59florzNg: IC :-)
21:02.39Ngis getting a ppp connection over a zaptel device documented somewhere?
21:03.11florzNg: Well, troubleshooting is a bit difficult without even some exact error message. But generally a wrong pridialplan setting would be a good candidate ...
21:04.35Ngflorz: ok. If changing that doesn't fix it (and I have my doubts, pridialplan=local broke everything) I'll get some pri debugging logs
21:06.55Waverly360Anyone ever had a MOH issue like mine before?
21:09.15*** join/#asterisk sashion (n=djbdsf@dsl-241-213-43.telkomadsl.co.za)
21:18.13*** join/#asterisk kgx (n=kgx@60.234.20.178)
21:21.47*** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net)
21:23.58*** part/#asterisk keescook (n=kees@ubuntu/member/keescook)
21:27.26*** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net)
21:28.32errris it possible to make a call from the asterisk cli?
21:28.38b11d|bblyes
21:28.40b11d|bbluse "Dial"
21:28.45errrah a big D
21:28.51errrI was using a little one
21:28.53b11d|bblno,  its not case sensitive
21:28.53errrty
21:28.57errroh?
21:29.01errrmaybe I did it wrong
21:29.01sashionerrr: core dial tech/number
21:29.16sashionor core dial tech/num@context
21:29.27*** join/#asterisk remmo (n=chatzill@smack.isp.net.au)
21:29.41errrhmm core is causing an error
21:29.56dlynes_laptoperrr: core is only a valid keyword if you're using asterisk 1.4 i think
21:30.00b11d|bblare you 1.4 or 1.2?
21:30.03errr1.2
21:30.13sashionthen its dial tech/num
21:30.15b11d|bbldial works fine for me on 1.2
21:30.18*** join/#asterisk friedrich| (n=friedric@e177243084.adsl.alicedsl.de)
21:30.19b11d|bbli dunno what your issue is.
21:30.34errrNo such command 'dial' (type 'help' for help)
21:30.39dlynes_laptopb11d|bbl: he probably doesn't have chan_alsa or chan_oss loaded
21:30.40b11d|bblweird, i've got it.
21:30.48b11d|bblyeah.. probably not.
21:30.51errrah
21:30.57dlynes_laptoperrr: do you have chan_alsa or chan_oss loaded?
21:31.02errrI doubt it.
21:31.09dlynes_laptoperrr: if not, that would be your conundrum
21:31.23dlynes_laptoperrr: that cli app isn't available unless you have one of those two channels loaded
21:31.32dlynes_laptoperrr: it pipes chan_alsa/chan_oss through chan_local
21:31.37errrdo I have to have a sound card to have them?
21:31.53dlynes_laptoperrr: well, to have a sound card driver, it usually  helps to have a soundcard, yes
21:32.08errrthis is a rack mount box, no card
21:32.31dlynes_laptoperrr: then how were you expecting to talk to the other end, or hear the other end?
21:32.33*** join/#asterisk twisted[asteria] (n=twisted@pdpc/supporter/active/twisted)
21:32.34*** mode/#asterisk [+o twisted[asteria]] by ChanServ
21:32.39sashionerrr: best other way is to use a .call file
21:32.40zoathen why would you like to use chan_Alsa ?
21:32.46errrI dont want to hear or talk, just prove a point to my boss
21:32.55dlynes_laptoperrr: use a sip phone, then
21:33.16dlynes_laptoperrr: or a softphone
21:33.24*** join/#asterisk tdi (n=tdi@reykin.pozman.pl)
21:33.44tdihi. does anybody know a page maybe where all asterisk sounds are explained in text form ?
21:34.06dlynes_laptoptdi: you can't tell that by the filenames?
21:34.14*** join/#asterisk pirast (n=martin@p508b2424.dip0.t-ipconnect.de)
21:34.18tdinot exactly
21:34.19errrthanks for the help everyone
21:34.40tdii want to give ex. all vm sounds to record them by once more
21:35.16pirasthi, i have a question to http://bugs.digium.com/view.php?id=9203. in the last comment, it says "there were no *security* fixes in 1.2.17". is a dos vulnerability not considered as security fix? compare: http://www.asterisk.org/node/48339 ("his release incorporates a fix for the SIP DoS vulnerability recently discovered by INRIA")
21:35.27dlynes_laptoptdi: that sounds like a good use for an ex
21:35.38tdiex?
21:35.40dlynes_laptoptdi: kinda like payback for them being your ex?
21:35.52Mercestespirast:  shh... we dont' talk about security issues.
21:36.01tdidlynes_laptop: i do not understand
21:36.04dlynes_laptoptdi i want to give ex. all vm sounds to record them by once more
21:36.10Mercestesyes, it's fixed.  K, thanks, bye.
21:36.15MercestesShhhh.
21:36.19tdidlynes_laptop: yes and ?
21:36.39dlynes_laptopyou want your ex to record all your voicemail sounds all over again?
21:36.43*** join/#asterisk keescook (n=kees@ubuntu/member/keescook)
21:36.50tdidlynes_laptop: aha nope
21:36.58Mercesteshmm...
21:36.59tdiex. - example osrry
21:37.07Mercesteslol
21:37.10dlynes_laptoptdi: still makes no sense :)
21:37.11tdii am giving it to the company that records such things
21:37.13Mercestesmy pet ex.
21:37.21dlynes_laptopyeah...i like to pet my ex, too
21:37.22*** join/#asterisk friedrich| (n=friedric@e177243084.adsl.alicedsl.de)
21:37.31Mercesteswoohoo
21:37.39*** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net)
21:37.45Mercesteswait....wrong channel.  >.>
21:37.52tdilook client want to record that the way he wants
21:38.04tdiwhy should i think about that :)
21:38.08Mercestestdi:  client no like comedian mail voice prompts?
21:38.16tdiyep
21:38.21dlynes_laptoptdi: write a script to automatically play them all back to you, so that you can write down what they are, then :)
21:38.27Mercestesclient make new sound voicemail he does, yes?
21:38.28dlynes_laptophehehehe
21:38.30tdidlynes_laptop: lol
21:38.42Mercesteslike the existing voicemail prompts he does not, no?
21:38.53tdiyes he wants new
21:38.54*** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com)
21:39.15tdithe same voice the menu is
21:39.15Mercestesmonitor you must do, with playback of sound, yes.
21:39.17Mercestesor with record would work, no?
21:39.34Mercestesasterisk cmd monitor, or asterisk cmd record, google you must.
21:39.45Mercestesmay the force be wtih you.
21:39.49tdiMercestes: why should i?
21:40.00Mercestesfor recording of voicemail prompts.
21:40.10spanglesontoastbah
21:40.13Mercestesdamnit!  Now I can't stop talking like that.
21:40.14tdiMercestes: i do not want to record them lol
21:40.24tdii want the text transcripts
21:40.24Mercestestdi:  would you like to soxmix them maybe?
21:40.24spanglesontoastwheres that usb digital tv turner gone
21:40.36Mercestestdi:  Then I suggset you take a speed writing class.
21:41.18Mercestesspanglesontoast, You can barely work a keyboard and your linux skills came from "hooked on phonics."  you don't have a USB digital TV tuner and even if you did you wouldn't know how to work it.
21:41.23dlynes_laptopspanglesontoast: they found out it was being used on Linux so Microsoft bought it, and hid it deep somewhere on their website
21:41.24Mercestesspanglesontoast, quit trying to pretend your a techie.
21:41.42[TK]D-FenderTalks does Mercestes often funny,  hmmmmMMM!!!@!?
21:41.43spanglesontoastnah it's transcoded to my xbox 360 ;)
21:42.01Mercestes[TK]D-Fender, ROFL   Multilingual, mercestes is, yes?
21:42.11[TK]D-FenderMercestes: s/your/you're/ ;)
21:42.41[TK]D-FenderMercestes: Clearly well versed in GIBBERISH.... and not always a comprehensible sub-dialect of which...
21:42.54Mercestes[TK]D-Fender,   indeed.
21:43.24dlynes_laptop[TK]D-Fender: what gibberish doth thou speaketh of?
21:43.25*** part/#asterisk keescook (n=kees@ubuntu/member/keescook)
21:43.53*** join/#asterisk ToyMan (n=Stuart@ool-45784fde.dyn.optonline.net)
21:44.02*** part/#asterisk ClydeGoffe (n=ClydeGof@base/student/clydegoffe)
21:44.36[TK]D-FenderFranglais :)
21:44.47*** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg)
21:45.01Waverly360I'm an idiot.
21:45.12Waverly360:P
21:45.23Waverly360stupid mispelled word caused me hours of grief
21:45.28Waverly360*sigh*
21:45.31dlynes_laptopWaverly360: for moh?
21:45.41Waverly360dylnes_laptop: Yeah.
21:45.50dlynes_laptopthere's your problem
21:45.55dlynes_laptopyou're even misspelling my name
21:45.56dlynes_laptopsheesh
21:46.00Waverly360lmao
21:46.01dlynes_laptopi think you need more timmy's
21:46.01Waverly360damnit
21:46.08Waverly360what I need is a vacation
21:46.15Waverly360and glasses that actually let me see better
21:46.30Waverly360You stare at the screen long enough, all the letters just run together.
21:46.46dlynes_laptopthat's the beer
21:46.50dlynes_laptopdon't blame it on the glasses
21:46.51tdihttp://www.voip-info.org/wiki/view/Asterisk+sound+files
21:46.57[TK]D-FenderWaverly360: Walleye-vision... I've seen it claim many a firend of mine...
21:47.09[TK]D-FenderWaverly360: When in doubt rely on auto-complete ;)
21:47.32Waverly360[TK]D-Fender: Unfortunately, auto-complete doesn't work within a vim session.
21:47.46Waverly360[TK]D-Fender: otherwise, that's what I use.
21:47.47[TK]D-FenderWaverly360: I meant in HERE....
21:48.14Waverly360[TK]D-Fender: ...I just now realized that auto-complete worked with my IRC client...
21:48.22dlynes_laptop[TK]D-Fender: obviously
21:48.39[TK]D-Fender"warning : people in mirror are dumber than they appear"
21:48.52JTeveryone hello
21:48.54Waverly360[TK]D-Fender: hah.  *sigh*
21:48.59dlynes_laptop[TK]D-Fender: you've been living in montreal too long :)
21:49.17[TK]D-Fenderdlynes_laptop: How so... besides "all of my life"? :)
21:49.22*** join/#asterisk santiago (n=santiago@debian/developer/santiago)
21:49.46[TK]D-Fenderdlynes_laptop: I exist perfectly between worlds.  This is a GREAT place for perspective if you aren't overly activist about anything...
21:50.13dlynes_laptop[TK]D-Fender: i just remember when i was there, there seemed to be a hell of a lot of stupid french people in the bilingual area
21:50.33dlynes_laptop[TK]D-Fender: well, i love it there
21:50.45dlynes_laptop[TK]D-Fender: don't know if i'd wanna live there though, with the politics
21:52.27[TK]D-Fenderdlynes_laptop: that falls under the category of "people are stupid.  Inidiviuals are smart." and "beware the power of stupid people in large groups"
21:52.27dlynes_laptopheh
21:52.27dlynes_laptopsounds like an advertisement for a government office
21:52.27[TK]D-Fenderdlynes_laptop: I healthily avoid politics.  Anybody who tries to shove a "party line" my way gets their ass handed to them :)
21:52.50dlynes_laptopyeah...politics in bc is just as screwed up
21:53.05dlynes_laptopnobody here knows what the hell they're going to vote for half hte time
21:53.05mcabalways has been
21:53.29mcabBC Politics is usually a complete gongshow
21:53.35dlynes_laptopand even then, half the time when they vote, it's a retaliation vote
21:53.41dlynes_laptopyeah, no doubt
21:53.52dlynes_laptopprovincial politics here is absolutely stupid
21:53.56JTi have to vote tomorrow
21:54.14dlynes_laptopvote Rhinoceros Party!
21:54.25JTnew south wales state election
21:54.29*** part/#asterisk l2cache (n=ghansen@64.128.254.98)
21:54.53[TK]D-Fenderdlynes_laptop: Not terribly different here.  Typically is PQ, and when they screw up BAD (ie referrendum, etc) we follow the LEADER we like the most which is why the Liberals got smashed alst time and the PC's made it back in offic for the first time in ages.
21:55.03mcabdlynes_laptop: not the Natural Law Party? :-)
21:55.10[TK]D-Fenderdlynes_laptop: Pot party!
21:56.00dlynes_laptopmcab: puuuuhleeaze...voting for a party that thinks they can levitate all the problems of the government away?
21:56.08JTcome to Australia, where the conservative party is called the Liberaly Party
21:56.20JTLiberal Party
21:56.29mcabJT: welcome to BC, we have the same thing :-)
21:56.30dlynes_laptop[TK]D-Fender: yeah...Mark Emery :)
21:56.55dlynes_laptopmcab: yeah...the liberal party here is a bunch of doofuses...they're only liberal because the socreds got destroyed
21:57.00*** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net)
21:57.05dlynes_laptopand the pc party in bc is completely non-existent
21:57.24[TK]D-Fenderdlynes_laptop: You guys actually HAVE NDP out there... damned socialists!
21:57.32dlynes_laptopwhat i wouldn't give for the pc's to get in, in bc
21:57.35mcabdlynes_laptop: c'mon, Natural Law's platform is as realistic as most of the main-stream platforms :-)
21:57.38dlynes_laptopi hate socialist parties
21:57.42dlynes_laptopespecially ndps
21:58.24dlynes_laptopndp is extremely bad for busines
21:58.31dlynes_laptops/busines/business/
21:59.21[TK]D-Fenderok, I'm off... back tonight...
21:59.44*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)
21:59.46PakiPenguinhi
22:01.19dlynes_laptopcessia galee
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22:06.18*** join/#asterisk RoyK (n=roy@ti211310a080-5748.bb.online.no)
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22:09.01spanglesontoasthmm
22:11.45Sweeperfeelings on astlinux?
22:13.46flendersJT: that's one of the good things about not being a citizen here.... I dont need to vote tomorrow
22:14.05JTif you think that's a good thing :)
22:15.36flendersaren't you annoyed with the ads on the radio?
22:15.36*** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca)
22:16.24JTyeah they're a bit annoying, they've stopped now as a ban on electronic election advertising is in force
22:16.45JTwhether or not you vote won't have any impact upon if those ads will be on
22:16.52flendersgood I'll start listening to radio again
22:16.53flenders:D
22:17.25JTand i think voluntary voting for able-bodied persons is stupid
22:17.48JTcompulsory is where it's at
22:18.08JTpeople will still put invalid votes in anyway if they really don't want to do it
22:19.54flendersJT: but if it's voluntary, sometimes you would be on a bbq, or at the beach, and even though you would vote right, you would just "nah... can't be bothered now"
22:21.03JTyeah
22:21.14JTlook at how corrupt the US's voluntary voting system is, also
22:21.26*** join/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net)
22:23.21spanglesontoastwhy doesn't meetme work even though I've got zaptel installed :|
22:23.37Strom_Mspanglesontoast: what do you mean by 'it doesnt work'?
22:23.52spanglesontoastload_resource: Module 'MeetMe' could not be loaded.
22:24.10Strom_Mdid you build and install zaptel before building asterisk?
22:24.24orlockJT: could you reccomend anywhere to buy some server-grade 2RU systems?
22:24.37JTsystems... like whole systems?
22:24.42orlockyeah
22:24.47*** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net)
22:24.49orlockcurrently looking at supermicro
22:25.00spanglesontoastno I did it after so I'm guessing I've gotta recompile it ?
22:25.12orlockfrom Digicor its something like $6k for a dual socket, dual core system
22:25.21orlockhot swap PSU, etc
22:26.11Strom_Mspanglesontoast: yes, you must compile zaptel FIRST
22:26.16orlockmight look at Sun and IBM as well
22:27.14*** join/#asterisk steveaj (n=steve@82-71-61-44.dsl.in-addr.zen.co.uk)
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22:29.30orlockfark.
22:29.47orlockdual cpu sun netra, $13k!
22:30.16*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
22:30.16*** mode/#asterisk [+o russellb] by ChanServ
22:30.35JTorlock: is that a niagra core?
22:30.51*** join/#asterisk Voice2 (n=Voice2@145-27.mc.cite.net)
22:30.59Voice2where is url to see patches in svn web ?
22:32.58orlockJT: yeah, think so
22:33.25JTorlock: well no wonder
22:33.29JTthey're awesome units
22:34.05JTorlock: you know each cpu has 8 cores?
22:35.19JToh wait, it might be 4
22:35.22JTnot sure
22:39.28JTorlock: i just checked
22:39.47JThelped jog my memory as to why the Niagra core is so good
22:39.52JT8 cores
22:39.59JT4 threads each
22:40.01JT32 threads
22:40.17spanglesontoastcan I recompile asterisk and it should work ?
22:40.37hijackedJT: you mean "pipelines" not threads, right?
22:40.54JThijacked: each core can run 4 threads at once
22:41.29mihinomenestyeah, pipelines.
22:42.12Qwell[]I own a T2000 :D
22:42.34Qwell[]it's slick
22:42.44fileyes... own...
22:42.46russellbVoice2: svn.digium.com/view
22:42.47Qwell[]indeed
22:42.48JTwhat do you use it for?
22:42.54Qwell[]JT: it sits in a rack taking up space
22:43.00JTsend!
22:43.05Qwell[]nah :p
22:43.08*** join/#asterisk MACscr (n=MACscr@adsl-75-23-64-115.dsl.peoril.sbcglobal.net)
22:43.11Qwell[]doesn't Sun do the try and buy in Oz?
22:43.37JTtry before you buy?
22:43.46Qwell[]yeah, they'll ship you one for free for like 60 days
22:43.57Qwell[]and they'll pay for return shipping - at least in the US
22:44.00JTi'm guessing you can't just be anyone ;)
22:44.07Qwell[]I'm just anyone
22:44.18Qwell[]hell, I even told them that I didn't have a credit card or a company
22:44.19JTdidn't get it under a company name?
22:44.20*** join/#asterisk af_ (n=getsmart@ip-156-32.sn2.eutelia.it)
22:44.22Qwell[]I did
22:44.24Qwell[]...
22:44.28JTheh
22:44.34Qwell[]and I was *very* clear that it was a fake company name
22:44.41JThmm ok
22:45.00JTnot sure if they're that liberal here but i'm sure they have a similar program
22:45.09MACscrcan someone point me to the updated asterisk manual. I keep stumbling upon depreciated things
22:45.26Qwell[]MACscr: deprecated
22:46.14MACscrQwell[]: correction noted
22:46.17mmartinnBoth :P
22:51.12orlockJT: ahh, these are all dual core, reason it was $13k is its a carrier grade system.. DC ower, etc
22:51.24data23tum te tum, 70 mins till ps3 is finally launched over ere, might try installing * on it at the w/e if i'm bored, you can get USB fxs adapters right?
22:52.08*** join/#asterisk jm|home (n=jm@zen.jamiem.com)
22:54.34JTorlock: ah, not a T2000?
22:54.52JTusb = :(
22:55.14Mercestesany PRI geniuses wanna help me troubleshot a HDLC abort issue?
22:55.17data23hmm i meant FXO anyway, always get them two mixed up :)
22:56.49fetcherIs there a way to specify *preferred* codec(s) in sip.conf, rather than just allowed/disallowed ?
22:57.07Mercestesfetcher, It goes in order of your speceification.
22:57.09JTdata23: just use an ethernet connected ATA
22:57.26Mercestesfetcher:  So if you allow ulaw, g729, gsm it will prefer ulaw
22:57.48data23aye but not quite as fun :), You were right btw JT, i never did get to the bottom of my ECT/2BCT issues and the developer of the patch has disappeared :)
22:58.03fetcherMercestes: or, allow=g729, then allow=ulaw on the next line would use g.729 preferentially, then fall through to ulaw if all licenses were in use?
22:58.06*** join/#asterisk Cyon (n=cyon@216.179.31.170)
22:58.18tuan_modulishere's a business idea that was pitched to me... .a myspace.... on a phone network
22:58.32tuan_modulisdo u think it might work at all?
22:58.33Qwell[]tuan_modulis: *gasp*, you could call it a "party line"
22:58.36Mercestesno...allow = g729,ulaw would .....probably give you an issue if all licenses were in use.
22:58.36CyonHey, anyone have a rough count of the number of asterisk installs in North America?
22:58.48JTdata23: heh, it's not really well supported at all in asterisk, ECT
22:59.00Qwell[]:(
22:59.00fetchertuan_modulis: the myspace that already exists is bad enough :)
22:59.14Mercestesfetcher, But theorhetically, you wuld use allow=g729,ulaw   not seperate allow statements...but..I think both are valid syntaxes by my recollection
22:59.32tuan_modulisthe only thing that has going for it is voicemail... and saydigits
22:59.38Mercestesfetcher, however, I think ifyou run out of licenses it gives you a hard "out of licenses" error and returns congestion without trying other codecs.  I would try it and see.
22:59.57*** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au)
23:00.01fetcherMercestes: ah... will have to do some testing then.
23:00.28JTdata23: usb is not fun anyway...
23:00.29Mercestesfetcher, Yea, give it a test.  if you think it's hard erroring without trying any other codecs then bug report it.  it *should* roll back to ulaw if g729 fails for whatever reason tho.
23:01.29*** join/#asterisk `p4r14h`work (n=josh@24-119-48-78.cpe.cableone.net)
23:02.40Mercestespretty please digum /pri expert support?
23:03.00JTMercestes: what's the problem?
23:03.39MercestesJT:  HLDC aborts
23:04.04JTis your zttest scoring ok?
23:04.04orlockwow
23:04.04JTno interrupt sharing?
23:04.14orlockJT: you know those shitty USB phones?
23:04.20JTyes
23:04.33MercestesJT:  Zttest is awesome.  I have 3 PRIs in a 4port card.  PRI 1 works great.  PRI2 works great...
23:04.38MercestesI plug in PRI 3 and PRI 2 goes to hell
23:04.49orlockJT: i have one plugged into my desktop, and have been playing with vmware.. vmware is using it for audio!
23:04.54JTMercestes: haven't we been through this before?
23:04.59MercestesIn fact...after I reset configs back to orig..PRI 2 was still giving me hell, I had to warm recycle zaptel while asterisk was running to stop the errors
23:05.06JTorlock: okay, and? :)
23:05.07MercestesJT:  yes
23:05.16orlockJT: nothing, its just cool :)
23:05.28JTorlock: they're just a sound card basically
23:05.28orlocki have forgotten how nice commercial software can be compared to open source stuff
23:05.46JTnice... highly debatable
23:06.04fetcherMercestes: that sounds like a possibly a timing source problem
23:06.09JTMercestes: speak to twtc at all?
23:06.32MercestesJT: Yea.  They test ok on their end.  In fact.  I unpluged PRI1 adn plugged TWTC into it without changing anything and everything was ok if I did that.
23:07.09Mercesteswhat gets me is PRI2 was giving errors under the original configs and the only way I could stop it was to recycle zaptel with asterisk humming
23:07.09JTokay
23:07.15Mercestes*THAT* bothers me.
23:07.27JTyeah sometimes zaptel does that
23:07.50Mercestesthat makes troubleshooting very difficult.
23:08.04Mercestesstarting zaptel after astierisk also gernates errors
23:08.45mostyis there any way to find out why a channel died? the logs show "spawn extension (context, number etc) exited non-zero on SIP/foo)
23:12.16tuan_modulisthe wiki explains a method.... but... it's time consuming
23:14.21Strom_Mthat was fun
23:14.25Strom_MI just ordered ISDN
23:14.46JTStrom_M: is this what the silly buggers with a b410p is about? :)
23:14.52Strom_Myes
23:15.06JTguinnea pig
23:15.10Strom_M:)
23:15.15Strom_Mhacktacular
23:15.24JTdo you think it will actually work?
23:15.26russellbStrom_M: um.  chan_misdn does not support bri in the us .......
23:15.43Strom_Mrussellb: yeah, but the question is how different NI1 is from Euro
23:15.47JTwho cares about chan_misdn anyway ;)
23:15.56russellbJT: people using the b410p.
23:16.06russellbor the various other bri cards
23:16.09JThaven't used one myself
23:16.12JTumm bristuff
23:16.15mostytuan_modulis: a method for what?
23:16.20JTeverything else definately works with it
23:16.32Strom_Mrussellb: because I've got a PRI circuit here which is set to NI2 on the network side and Euro on the CPE side, and the damn thing works just fine
23:16.34russellbumm not everyone wants some random unsupported patchset :-p
23:16.35tuan_modulismosty: lemme find it...
23:16.54JTrussellb: err, no more random than the unsuported patchset of misdn
23:17.01JTmisdn is alpha software
23:17.11*** join/#asterisk P4C0 (n=ash@200.124.22.34)
23:17.13JTand has really *bad* NT mode support
23:17.14russellbmisdn is included in asterisk, so we accept bug reports on it
23:17.33JTit's also nice to be able to use all zap features
23:18.12Strom_Mrussellb: so yes, i know full well going into this that it may not work at all
23:18.13P4C0hello guys, one question... I'm planning to buy a license for codec g.729, however in the page says that you need one license per channel (or that the license (10.00 usd) is only valid for one channel... is that the same as one call at a time?
23:18.22Strom_MP4C0: yes
23:18.30*** join/#asterisk JT_ (n=jon@unaffiliated/jt)
23:18.41JTmisdn debugging is horrible as well
23:19.15tuan_modulismosty: actually, my bad, it's to debug a deadlock
23:19.22tuan_modulishttp://www.voip-info.org/wiki/view/Asterisk+debugging
23:20.07P4C0Strom_M, humm right now , I'm using alaw between my server provider and my asterisk box, and I can have multiple calls at the same time, if I buy this license and change that link to 729, I'll only be able to have one call? so if I'm calling from an internal terminal and the voip provider pass me a call what will happend?
23:20.10mostythanks anyway
23:20.28Strom_MP4C0: you need the license to transcode
23:20.29mostyP4C0: only one g729 call
23:20.40Strom_MP4C0: if you do g729 passthrough, you dont need said license
23:21.17JT_I run BRI on production systems in TE and NT mode, and i can't stand misdn
23:21.18P4C0Strom_M, my internal phones doesn't support g729... so I need to transcode...
23:21.24JT_too buggy and lacking in features
23:22.10P4C0btw, since in the local net bandwidth is not a problem, what's the best codec where i can transcode the g729? (less cpu usage in the transcode?)
23:22.20JT_g.711
23:23.03spanglesontoastanyone know why asterisk doesn't compile the chan_zap.so module
23:23.06JT_what sort of phones don't support g.729 anyway?
23:23.19Strom_MP4C0: if you don't have bandwidth problems, don't bother with g729
23:24.01P4C0Strom_M, I don't have internal (local net bandwith problems, but the link between my voip provider is of 700 kbps)
23:24.14Strom_M700kbps is gobs
23:24.40Strom_Mive only got a 768kbps uplink at home, and it's ulaw all the way for me
23:25.03*** join/#asterisk quidpro (n=quid@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com)
23:25.33russellbspanglesontoast: get zaptel from svn
23:29.20JThrm, asterisk using 98% cpu
23:29.28JTseems non-optimal
23:29.32JT1.2.16
23:29.46JT<PROTECTED>
23:30.06data23:}
23:30.24JTthe thing is
23:30.37JTit's been doing absolutely nothing for over 12 hours, that instance
23:31.45P4C0steve___, 768 is my down... up is like 128 or so
23:31.55P4C0steve___, ups sorry that was for Strom_M ...
23:32.08mostyi have a wctdm card which isnt plugged into my phone line yet, when i try to dial over the card, the card answers the call and just waits. is there a way to make it realise that it cant make the call and return CHANUNAVAIL or something?
23:32.12P4C0Strom_M, actually let me check my real badnwitdth
23:32.49JT_why do people say "ups" when they mean "oops", i don't get that
23:34.28spanglesontoastwhy svn I just wnat the module
23:34.43JT_spanglesontoast: what version of asterisk?
23:34.51spanglesontoast1.4.2
23:35.25JT_spanglesontoast: you need to download zaptel from 1.4 SVN
23:35.33P4C0Strom_M, 733 kbps down, 249 kbps up... i think g729 will improve things... what do you think?
23:35.42spanglesontoastI've got the current zaptel
23:35.43spanglesontoast..............
23:35.45JT_as the release version of zaptel that works for 1.4.2 is not out i believe
23:35.55Strom_MP4C0: yeah, it'll help, but it sounds like ass
23:36.00JT_spanglesontoast: the latest RELEASE of zaptel does NOT work with 1.4.2
23:36.08JT_you must download zaptel from 1.4 SVN
23:36.14spanglesontoastwhere that ?
23:36.20JT_until the next release comes out
23:36.35JT_use a svn client or svn.digium.com
23:36.39*** join/#asterisk kink0 (n=k@161.pool62-37-205.static.orange.es)
23:36.41kink0hello
23:37.00spanglesontoastalso how do I remove the existing compiled one ?
23:37.19kink0a stupid question, when you get over 99.98% with zttest, the zttest is running while many calls are in progress ?
23:37.43JT_kink0: depends on the machine, but on some yes
23:38.16fetcherP4C0: are you talking to another Asterisk on the far end, or directly to another SIP phone?
23:38.37kink0JT_ is because I get 100% worst if I run while no calls or just fews calls, but while 40 calls, i get a worst value of 99.65%
23:38.50P4C0fetcher, another asterisk... my voip service provider asterisk (it may not be asterisk... thro)
23:39.27JT_kink0: 99.65 is unacceptable usually, do you notice problems?
23:39.47kink0JT_ yes, noise whith 99.65
23:39.48fetcherP4C0: some providers support iLBC, which is similar in bandwidth & quality to G.729 but doesn't require licenses
23:40.05JT_kink0: yeah that's no good
23:40.08spanglesontoastis there no other way of getting conferencing to work without timing
23:40.09JT_kink0: what are the specs?
23:40.10P4C0fetcher, mine only alaw and g729
23:40.21fetcherP4C0: also, using IAX instead of SIP, if you aren't already, will save a bit of bandwidth
23:40.24kink0JT_ but I tried to compile kernel again and again, dedicate one cpu to TE405, enable and dissable ec, enable and dissable HT, no way
23:40.57kink0JT_ I try two boxes, one is Dual Xeon 2.8, Supermicro, the other is Dual Xeon 3.2/2MCache, Supermicro
23:41.01fetcherP4C0: may be worth throwing a few test calls at them using various codecs anyway, just to check for undocumented features ;)
23:41.02P4C0fetcher, my voip provider doesn't support iax
23:41.13kink0JT_ no other user software is running ( except sshd )
23:41.16P4C0fetcher, already did :p
23:41.34fetcheroh, well...
23:41.38JT_kink0: hrm, how many PRIs?
23:41.45kink04 PRI
23:41.50JT_hmm
23:42.08JT_not sure what to suggest
23:42.15kink0JT_ I use g729 for most, but CPU idle still over 50%
23:42.28JT_kink0: any particular options you set in the kernel?
23:42.36JT_kink0: g.729 kills cpu
23:42.50kink0JT_ no, nothing. I also try some boot options, but same.
23:43.08P4C0I'll go now, fetcher Strom_M thanks for your tips, c u
23:43.11kink0yes, but as I get a high idle CPU, I think g729 is not the cause of lose IRQ
23:43.15JT_you should set kernel HZ to 1000
23:43.22JT_not sure if it will have a major impact
23:43.23kink0yes, is setted to 1000
23:43.38kink0no impact, changing from 250 to 1000
23:43.44JT_is the bus shared with anything else?
23:45.06spanglesontoastJT_ which version are you using
23:45.09kink0JT_, I think is not shared, as well as is ussing IRQ 48, and there no other device with that IRQ, but... If I do lspci -vb I see there others ussing IRQ 5
23:45.10*** join/#asterisk JT (n=jon@unaffiliated/jt)
23:45.41JT_spanglesontoast: 1.2. branch
23:45.44kink0I have try boot option append=noapi but that did nothing
23:45.55JT_1.4 is not stable enough for production without lots of testing
23:46.07*** join/#asterisk nasls_lsa (n=chatzill@athedsl-148609.otenet.gr)
23:46.07kink0I still having my IRQ APIC again
23:46.25*** join/#asterisk Ryanw (n=cableguy@ge0-0-15-lns0.207alg.qx21.net)
23:46.56rudholmStrom_M: why did you order ISDN when DSL is so much faster?
23:47.02spanglesontoastwhich outta these http://ftp.digium.com/pub/zaptel/releases/
23:47.08JT_rudholm: for voice :P
23:47.21Strom_Mrudholm: because ISDN makes my internets go TWICE AS FAST
23:47.28rudholmthat's right!
23:47.34rudholm'cause it's 2
23:47.35rudholm!
23:47.36kink0JT_ do you know if FSB has any influence here ?
23:47.50JTkink0: not sure
23:47.50rudholmFront-Side Bus?
23:48.05kink0I try 400 and 533Mhz FSB, no difference, but I have not try 800 or 1066
23:48.15JTkink0: anyway, i asked if the bus was shared
23:48.35rudholmyou people who live in !california really need to get on your PUCs.  BRI is 25$/month here.
23:48.51rudholmand that includes 200 b-channel hours, iirc.
23:48.56Ryanwdoes the goto command do pattern matching lookups in the destination context or must the extension you go to exist exactly ?
23:48.57kink0JT: APIC gives a unique IRQ, but I see if I do lspci -vb that there more devices ussing IRQ 5
23:49.03Strom_Mrudholm: 200!?  really??
23:49.10JT_kink0: i did not ask about IRQs
23:49.14JT_kink0: i asked about the bus
23:49.18Strom_Mrudholm: is that for voice and data calls?
23:49.19rudholmStrom_M: yeah, on Residential BRI, not Business.
23:49.26rudholmit's for data
23:49.36Strom_Mah ok...what about for voice?
23:49.59rudholmvoice is billed the same as POTS
23:50.11rudholmit's really quite a good deal.
23:50.12kink0JT_, yes the PCI bus 2 is also SCSI and Ethernet
23:50.23Strom_Mrudholm: ah ok, so the same $3 allowance?
23:50.35rudholmyeah, or possibly "unmeasured"
23:50.36rudholmeither way
23:50.37JT_kink0: try and give it a dedicated bus, some servers have 2 PCI buses
23:50.39rudholmit's good
23:50.45mostyi have a problem with my zaptel device, when it's not plugged into the line and i try to dial over it, it answers the call but just stops and waits there, when (i think) it should exit immediately and set DIALSTATUS to CHANUNAVAIL or something. what could be wrong?
23:51.00rudholmGTE was bad, all "data" calls were billed per minute and there was no allowance.
23:51.06spanglesontoastguess no one has a working conference module
23:51.18JT_spanglesontoast: conferences work fine here.
23:51.21kink0JT_ I see, but I have only one PCI slot available here, there just 2 PCI slots , one is for low profile card, and the other is where I place the TE405 card
23:51.24rudholmso I did a lot of data calls sent as voice calls.  if you weren't calling very far away, it'd almost always work.
23:51.26orlockmosty: its not plugged in to the line maybe?
23:51.35spanglesontoastthen why doesn't it compile the module with asterisk
23:51.48JT_kink0: where is there room for scsi and ethernet then?
23:52.19kink0JT_ are integrated on motherboard
23:52.23orlockrudholm: theres a similar thing here for ISDN.. making data calls flagged as voice
23:52.25kink0sorry, SCSI is in bus 3
23:52.27mostyorlock: if it's not plugged in, i dont want zap calls to just hang (i'm trying to setup a failover to dialling over iax)
23:52.29JT_kink0: are you sure they're on the pci bus?
23:52.33kink0only Eth and TE are both in bus 2
23:52.44JT_kink0: read the motherboard system block diagram
23:53.03kink0JT_ I think so, at least I get listed with lspci
23:53.04*** join/#asterisk shodan- (n=shodan@ip062.96-113-216.pppoe1.joliette.intermonde.net)
23:53.06rudholmorlock: yeah, I found with GTE's pricing for BRI, my ISDN bill went from 400$/month to 50$/month when I switched to data as voice.
23:53.12JT_kink0: hrm ok, would be nice if they were on different buses, but i don't know if that's causing your problem or not
23:53.15rudholmorlock: this was back in the late 90s
23:53.23*** join/#asterisk coppice (n=chatzill@90.203.17.210.dyn.pacific.net.hk)
23:53.28orlockrudholm: pci isdn cards?
23:53.49kink0JT_ as I know, the bus is not selectable by software and is fixed at hardware, right ?
23:54.06rudholmorlock: I was using a Netopia "ISDN Router" at the time.
23:54.09JT_kink0: yes
23:55.01mostyorlock: do you understand my problem?
23:55.11orlockmosty: yeah, but i dont know sorry
23:55.24mostythanks anyway
23:55.38JT_kink0: what are all your versions again?
23:58.08*** join/#asterisk X-Gen (n=X-Gen@dsl-242-28-178.telkomadsl.co.za)
23:58.47kink0JT_ I tried with 1.4.1 and 1.2.12
23:59.26kink0zaptel 1.4.0 and zaptel 1.2.9.1
23:59.41JT_hmm ok

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