IRC log for #asterisk on 20100211

00:03.13*** join/#asterisk jksM (jks@193.189.93.254)
00:15.27jayteewow, it's really jumping in here
00:17.30*** join/#asterisk SaiSoma (~SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net)
00:18.18beekHi jaytee
00:18.26jayteehi beek
00:19.41beekjaytee: I'm just sitting here watching the snow pile up.
00:20.22jayteesnowmageddon
00:20.53beekIt's a PITA, for sure.
00:21.25jayteewhere are you at?
00:22.07beekSouth of State College, PA
00:22.21jayteeoh, yeah! you're getting dumped on royally
00:22.59beekIt's a nice addition to our 14" from last Saturday.   The southern end of our county got 30"
00:24.14*** part/#asterisk etfonhomey (~etfonhome@74-131-159-160.dhcp.insightbb.com)
00:29.14*** join/#asterisk ChUbB (~IceChat7@cpc2-aztw12-0-0-cust229.aztw.cable.virginmedia.com)
00:32.41*** join/#asterisk staffmember (~singulari@static-71-183-79-24.nycmny.fios.verizon.net)
00:32.53staffmemberis there any reason why i wouldnt be able to do "sip show peers" from CLI
00:33.06staffmembersips not even an option when i do ?
00:36.12*** join/#asterisk ariel_ (~ariel_@c-24-127-196-248.hsd1.fl.comcast.net)
00:37.18*** join/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com)
00:38.10*** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7)
00:39.56p3nguinstaffmember: Sounds like chan_sip isn't loaded.
00:42.15*** part/#asterisk Cresl1n (~matt@asterisk/libpri-and-libss7-expert/Cresl1n)
00:45.42*** join/#asterisk bobnormal (~bobnormal@94-195-193-13.zone9.bethere.co.uk)
00:45.48bobnormalhow to show registered channel types?
00:46.47*** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58)
00:50.14*** join/#asterisk spenguin[work] (~penguin@122.182.0.38)
00:50.30*** join/#asterisk hluesea (~hulusikah@88.247.127.66)
00:52.28jaskewbobnormal: try core show channeltypes
00:52.41bobnormal<PROTECTED>
00:52.48jaskewnot sure if that's what u are looking for
00:56.03staffmemberp3nguin: whats not loading if i cant do "reload" from cli ?
00:57.09*** part/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com)
01:04.51*** join/#asterisk MatBoy (~MatBoy@wiljewelwetenhe.xs4all.nl)
01:12.42bobnormalafter upgrade from 1.4 to 1.6 on 'module load chan_dahdi.so' i get 'chan_dahdi.c:2121 dahdi_open: Unable to specify channel 1: No such device or address'.  cause?  wanrouter status confirms connected.
01:27.12*** join/#asterisk Faithful (~Faithful@ns.linuxterminal.com)
01:40.11*** join/#asterisk pentanol (~Unknown@91.195.60.231)
01:46.29*** join/#asterisk nix8n82 (~AndChat@63.162.27.14)
01:49.06*** join/#asterisk titter (~titter@c-76-101-240-142.hsd1.fl.comcast.net)
01:54.07spenguin[work]anyone here got any comments on ipkall?
02:05.16*** join/#asterisk voipmonk (~shido6@67.111.52.130.ptr.us.xo.net)
02:06.15*** join/#asterisk ticoit (~ticoit@201.191.190.112)
02:07.22*** join/#asterisk sebbl (~Momofu@HSI-KBW-109-192-163-191.hsi6.kabel-badenwuerttemberg.de)
02:07.32*** join/#asterisk iq (~chatzilla@unaffiliated/iq)
02:07.33iqHi
02:07.36sebblmoin
02:08.01sebblwhat isdn card can i use for 2 isdn lines?
02:15.19*** join/#asterisk Wgg (~hyena@75-119-229-164.dsl.teksavvy.com)
02:16.05WggHi folks, I'm trying to get PlayDTMF to work and having trouble
02:19.34p3nguinWere you planning to provide any details and get some help, or were you just letting everyone know that you're having a problem?
02:19.59WggI was waiting to see if anyone here was alive ;)
02:20.10bobnormalmmm
02:20.23WggOk, I get back the response "DTMF successfully queued" but never hear the tone
02:20.48WggI found a few other places where people mention this problem, but no solutions
02:20.49*** part/#asterisk sebbl (~Momofu@HSI-KBW-109-192-163-191.hsi6.kabel-badenwuerttemberg.de)
02:22.08WggAre there any other details I should provide?
02:23.30ChannelZYour credit card number
02:23.56WggCertainly!  It's 88
02:24.16*** join/#asterisk jmcdowell (~nooe@70-12-90-97.pools.spcsdns.net)
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02:41.35*** join/#asterisk thansen (~thansen@c-76-27-110-194.hsd1.ut.comcast.net)
02:43.54thansenaccording to this.. http://www.voip-info.org/wiki/view/say+digits  I can define escape digits.  It seems to be escaping no matter the digit pressed.  Is there any way to force all the digits be heard?
02:44.11*** join/#asterisk nix8n82 (~AndChat@63.162.27.14)
02:45.36jmcdowellWhat do you mean escaping?
02:45.48jmcdowellSorry, I just got here.
02:47.32thansenjmcdowell: that's all I've said so you haven't missed anything :)  It's not completing the readback of all digits
02:47.39jmcdowellAnyone know how to prevent an extension from dialing it's self?
02:48.04thansenfun..infinite dialplan loop
02:48.04jmcdowellSo... Where are you seeing this?
02:48.21thansenhttp://www.voip-info.org/wiki/view/say+digits
02:48.25jmcdowellin the asterisk debug output?
02:48.51thansenI'm calling and hearing that not all the digits are read and it just continues with the script
02:49.03thansen(when I hit a number)
02:49.22jmcdowellhuh
02:49.47thansenwhat don't you understand?
02:50.04jmcdowellI have never used "say" digits
02:50.09jmcdowellwasn't aware it was supported
02:50.32thansenwhat do you use?
02:50.37jmcdowellThe keys
02:50.56thansenhow do the keys talk to you?
02:51.08jmcdowellThey keys dont talk to me
02:51.12thansenlol
02:51.13jmcdowellohhh
02:51.19thansen:D
02:51.22jmcdowellthis is "playback" from asterisk ?
02:51.34thansenyes
02:52.04thansenit's playing back a set of digits...I want to force the person listening to hear them *all*
02:52.15*** join/#asterisk KC9NVQ (trelane@funtoo/staff/trelane)
02:52.20jmcdowellAhhh..
02:52.26jmcdowellI know it can be done, but don't know how
02:52.30thansenright now any dtmf is 'escaping' from playback
02:52.46jmcdowellright
02:54.53KC9NVQlooking for something similiar to asteriskstat that can rate calls and produce phone bills.  A2billing is ridiculously complicated, and seems to have to run on the server.  I'd prefer to just hook up to cdr_mysql's database
02:59.03*** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com)
02:59.33*** join/#asterisk Rapier1024 (~Rapier102@d-69-161-89-227.cpe.metrocast.net)
02:59.53Rapier1024Hi folks
03:01.46Rapier1024can anyone tell me if they have seen something like this before? I'm having intermittent dropout of audio on 480i's over a VPN connection, but only on the incoming stream to the server
03:02.18Rapier1024It's random, and can last from 2 to 10 seconds
03:03.21jmcdowellSetup QoS
03:04.01*** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id)
03:04.40Rapier1024The LAN on both sides is dedicated to the phones with no other ip devices on either of them. This is only from the remote office to the home office
03:05.16Rapier1024Everything going out on the port at the remote is "real time" QoS.
03:05.41jmcdowellWhy are you pushing them through a VPN?
03:06.01jmcdowellIs this a real VPN or some bullshit MS vpn server ?
03:06.24Rapier1024We had an issue a while back when I forwarded a port to my server. No this is a sonicwall hardware firewall with point ot point VPN
03:06.40jmcdowellHmmm..
03:06.48jmcdowellI would look at how you are doing VPN on the one side
03:06.57jmcdowellit too could cause issues since it's not balanced on the other end.
03:06.58Rapier1024Have a TZ150 on both sides
03:07.09Rapier1024same config on both ends
03:07.20jmcdowellRun IPTRAF
03:07.24jmcdowelliptraf
03:07.28jmcdowelland watch the traffic
03:07.40jmcdowellsee if there is something overwhelming your vpn.
03:08.00Rapier1024Yeah, the problem is this only happens once a day and it's pissing off my customer
03:08.09jmcdowellWhat time of day?
03:08.13jmcdowellLet me guess, LUNCH
03:08.35Rapier1024It changes. LOL sometimes in the moring, and other in the afternoon
03:08.41*** join/#asterisk styelz (~yoohoo@m0o0.mooo.com)
03:08.52Rapier1024nobody is talking at lunch so I don't know about that.
03:09.32Rapier1024This one is driving me nuts though. it's only on the incoming stream. The remote office can hear fine
03:09.40jmcdowellWhat time of day?
03:10.09Rapier1024it's not a specific time. sometimes early, sometimes in the afternoon.
03:10.29jmcdowellSo, I would run iptraf on both sides.. or at least the server side
03:10.57Rapier1024the phones are registering fine, and the problem shows up whether it's a call to the outside, or to one of the internal phones
03:11.28jmcdowellWhat kind of phones?
03:11.39Rapier1024aastra 480i's
03:12.04jmcdowellI don't know how to monitor your link, as it could be that too.
03:12.44*** join/#asterisk DarkFibre (~dmelouk@127.159.119.70.cfl.res.rr.com)
03:12.44Rapier1024iptraf rolls over doesn't it? in other words, I can leave it running for some time and have them let me know the second it happens, yes?
03:12.55jmcdowellYou can log it
03:13.01jmcdowellI don't if you can have it notify you
03:13.12jmcdowellbut I am sure they would let you know and then you could look at the logs.
03:13.24Rapier1024I'll have them user let me know the minute it happens.
03:13.39jmcdowellIs it ONE or all lines affected?
03:13.45Rapier1024all phones
03:13.55jmcdowellMy polycoms were doing that and I had to adjust their codecs and everthing was fine.
03:14.08Rapier1024hmmm
03:14.19jmcdowellI would almost bet that something is saturating the network link though because it doesn't like the VPN.
03:14.41jmcdowellMake sure you are watching ALL type of traffic in IP traf. IP, NON-IP and unknown.. EVERYTHING
03:14.57Rapier1024okay that makes sense
03:15.06jmcdowellIf you are experiencing some sort of an arp problem that is storming things, you want to be able to see it.
03:15.22jmcdowellI would also crank up all the debug values in asterisk
03:15.36jmcdowellso you can look there if IPTRAF doesn't show anything.
03:15.48Rapier1024We have a 5mb ethernet over copper at the customer's main site, and comcast business at the remote.
03:16.02jmcdowellI would also use "sar" or HTOP to watch the CPU and memory usage on your server.
03:16.03Rapier1024I don't have this problem with any of the phones at the main site.
03:16.53jmcdowellRight, but if you have a device that doesn't understand the fact that you have a VPN and starts spitting out information that causes confusion. It can cause problems.
03:17.28jmcdowellI would bet that if you setup QoS, you would be surprised.
03:17.57Rapier1024The computer and the phone lans share the connection at the remote, and I have limited the computer port on the main switch to 512kb leaving  1.5mb for thephones but it didn't solve it
03:18.17Rapier1024I need to make sure QOS is in place on the main side
03:18.24jmcdowellagain crank up the logging
03:18.36jmcdowelluse iptraf and watch the cpu memory utilization
03:18.57jmcdowellit could be that on the main site side, that QoS isn't allowing them what they need.
03:19.53Rapier1024can you tell me the best way to lock down the asterisk server? When I opened theport the last time, someone got in and was screwing with it.
03:20.19jmcdowellWell, you just threw a WHOLE different set of metrics in there.
03:20.27jmcdowellDid you re-format and re-install ?
03:20.43jmcdowellBecause without doing so, someone could still be screwing with it.
03:21.10Rapier1024It's actually run from a module, not a disk. so if they did anything, resetting the thing eliminates it.
03:21.25Rapier1024its set to run from a ram disk
03:21.36jmcdowellHmmm..
03:21.41jmcdowellLocking it down is fairly easy
03:21.54jmcdowellDont set stupid passwords and lockout the root account
03:22.02jmcdowellchroot jail ftp
03:22.15jmcdowelland I can't see how they could get in.
03:22.38Rapier1024Can you require certs in order for the phones to access the server?
03:22.51jmcdowellI think there is something like that, I have never done it
03:22.54jmcdowellI just use FTP
03:23.02jmcdowellwith the password, they aren't getting in
03:23.12jmcdowellno root ssh logins are allowed
03:23.32Rapier1024I thought so too, but haven't tried it
03:23.40jmcdowellNo they are by default
03:23.46jmcdowellI am saying they shouldn't be
03:24.00jmcdowellonly sudo su - access and even that is ridiculous hard to acheive
03:24.01p3nguin"sudo su" is retarrrrrrrded.
03:24.13Rapier1024I meant I thought certs wew supposed to work
03:24.15jmcdowellBetter than root logins allowed
03:24.30Rapier1024no, root logins aren't allowed
03:24.54jmcdowellMy default install allowed them, I disabled them yours is obviously different.
03:25.12jmcdowellI tried to "hack" my box once, even knowing everything but the password, I was never able to get in.
03:26.03Rapier1024I don't know how they got in, and it pissed me off. The saving grace was in order to write the config back to the card, you have to know it's there
03:28.14Rapier1024Well, I'll try what you suggested here. Thanks a great deal for your help, I really appreciate it.
03:28.47jmcdowellI don't know that I helped all that much..
03:29.09Rapier1024well you gave me places to look, and that's a start :)
03:29.47Rapier1024I manage this system, and have some experience with Asterisk and linux, but I'm certainly no guru
03:30.46jmcdowellI would find a way to look at Link saturation
03:31.06jmcdowelland you should be able to do that throught the modem or router provided by your link provider.
03:31.55Rapier1024I can see that on the router, and the logs will corellate the timing so that should tell me something
03:32.35Rapier1024Asterisk is a challenge for me. Ask me about XenServer or Virtual servers and I'm fine
03:33.09jmcdowellI love asterisk
03:33.14jmcdowellI have only been using it for a month.
03:33.14jmcdowell:D
03:33.26leifmadsenit's good stuff
03:33.36jmcdowellindeed it is
03:33.51jmcdowellI am working on a "Provisioning center" plugin module for freepbx./
03:34.04Rapier1024I haven't had too many issues, but it's the wierd stuff that makes me want to pull my hair out
03:34.20jmcdowellyep, just follow the KISS theory and it usually works out
03:34.26Rapier1024usually if you don't have audio, it's on bot directions
03:34.37Rapier1024in both rather
03:35.22Rapier1024Never had this issue with the SNOM phones, but I don't think there are any in the remote office
03:36.30*** join/#asterisk maxagaz (~maxagaz@222.128.36.151)
03:36.32jmcdowellOnce you collect all that info, I think you will be able to really zero in on something
03:36.38jmcdowellor at least say it's none of these issue.
03:36.51Rapier1024yeah...that makes sense.
03:37.07jmcdowellat which point I would really look at the codec tweaks on the phones
03:38.21Rapier1024I fought for some time with one phone that woul drop registration periodically. Discovered with the SonicWall TZ150 as the gateway, it was maxing out the connections. Didn't realize I could set the asterisk box as the gateway on the phones at that time
03:38.42Rapier102410 user limit on the sonicwall
03:39.05Rapier1024They don't have limits anymore thank goodness
03:40.15jmcdowellI would have just built linux based routers..
03:40.16jmcdowell:D
03:40.35Rapier1024hmmm....something just occurred to me. there are five phones in the remote office. there is one of them that is doing the same thing just recently. This could be the issue
03:41.00Rapier1024something else must be plugged into that lan up there.
03:41.17Rapier1024five phones wouldn't do that.
03:41.59jmcdowellWhat do you mean it's doing the same thing?
03:42.11Rapier1024intermittently showing NR
03:42.19Rapier1024losing it's regitration
03:42.25Rapier1024registration rather
03:43.05Rapier1024it happened before because the sonicwall would send an error instead of forwarding the packets
03:44.01Rapier1024retturn an error I meant
03:44.13Rapier1024boy I can't type tonight
03:44.19jmcdowellCould the sonic wall be "Storming" the link with bs packets?
03:45.48Rapier1024no, I've looked at the data stream before. It would more likely be that someone has plugged a computer or two into that lan. That would cause the error. Several connections from multiple devices could exceed the 10 user limit
03:46.15Rapier1024The thing is, it wouldn't always cause the NR
03:46.21jmcdowelllock it down by mac or install an IPCOP firewall and throw that sonicwall in the trash.
03:46.54*** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110)
03:46.58Rapier1024Trying to manage a lan that is 4 hours away is starting to get on my nerves!
03:47.07jmcdowelllol
03:47.28Rapier1024they have 10 times as many issues than the main office does
03:48.45Rapier1024I have no idea what they plug in where, and it kills a whole day to go up there. GRRRRRR
03:49.10Rapier1024I could tell these folks all day long what should go where, but it just doesn't sink in
03:50.20Rapier1024the sonicwall is actually a pretty good device, unless you start opening ports on it
03:51.59Rapier1024with that VPN in place, we have had no issue with folks getting into the lan
03:53.16Rapier1024well thanks again, I think it's time for bed
03:53.23jaskewraiper1024
03:53.36Rapier1024??
03:53.43jaskewJust catching up on the thread -  I had similar issues at my office
03:54.02Rapier1024what solved it? or is it solved?
03:54.02jaskewI was getting interruptions in the RTP stream
03:54.09Rapier1024from where?
03:54.19jaskewIt is - QoS is the answer, but for me was only part of the answer
03:54.44Rapier1024I need to make sure that's enabled on the main office end
03:54.57jaskewI'm not familiar with the SonicWall - I'm using a linux router here
03:55.38Rapier1024Rather than setting each device, I set everything going out over that one port to be "real time" on the main switch
03:55.45jaskewTHe problem was that my DSL modem has a pretty large queue.  That effectively un-did any QoS stuff I had in my router
03:56.08Rapier1024hmmm
03:56.14Rapier1024how did you solve that?
03:56.23jaskewI solved it by shaping my traffic to my DSL speed at the router.  That kept the queu from building up in my modem
03:56.24Rapier1024I was reading on buffer size
03:56.55jaskewThen I was able to control the prioritization properly.
03:57.29jaskewOtherwise, I was prioritizing everything at my router and then just sticking all of the traffic into the (dumb - fifo) queu in the DSL modem
03:57.43Rapier1024I had planned on doing something similar tomorrow. limiting the total bandwidth at the switch to 2mb
03:58.01jmcdowellI don't know if that's the answer
03:58.01Rapier1024and then throttling the computer lan to only use half that
03:58.18jaskewSo the VPN complicates it a little bit - do you know if the VPN passes through the QoS?
03:58.20jmcdowellI think I would do some serious logging, again Iptraf can be very very revealing of rouge devices
03:59.04Rapier1024I think so, it just wraps the packets in UDP and then delivers them on the other end exactly as they originated
03:59.26jaskewso does the "envelope" UDP packet preserve the QoS (on the outside where it can be seen)?
03:59.31Rapier1024I plan on doing the logging and iptraf to see that.
04:00.05Rapier1024I can set the QoS both inside the packet, and then outside. Inside at the firewall, and then tag the UDP at the switch
04:01.01jaskewThat's good.  Here's a discussion of what I was talking about
04:01.02jaskewhttp://mailman.ds9a.nl/pipermail/lartc/2007q3/021607.html
04:01.14jaskewRead that & follow the link to lartc.org.
04:01.45jaskewIf you can do something similar, it might help celar things up.  Of course, you'll have to fact ro the VPN stuiff in
04:01.59Rapier1024I think throtling the bandwidth (outbound) to the total for the line, and then setting the QoS on inside and outside should help this.
04:02.12jaskewworked for me :)
04:02.43Rapier1024also making sure that the sonicwall isn't kicking the packets because of device user limit (ie...some idiot plugged in a computer to the lan)
04:03.07jaskewyou probably will need to tune both ends to make sure that the RTP packets get put on the line ASAP when they come up
04:03.48Rapier1024that's the frustrating part. If the sonicwall is kicking the packets, it would do EXACTLY what it's doing. It wouldn't be long enough to kill the call, but to just block outgoing traffic for 2 to 10 seconds
04:03.56jaskewputting them ASAP into a queu that is 1000ms deep doesn;t help ;)
04:04.04Rapier1024I agree
04:04.40jaskewGood luck w/ it - hope that helps!
04:04.41Rapier1024I just found out that there is a new router in place that belongs to comcast between our switch and the cable modem
04:04.58Rapier1024have no idea who put that in place, or why
04:05.16Rapier1024Thanks to both of you for your help! I'm off to bed
04:05.19jaskewwell, if you limit the sonic wall output to the upstream speed, it should still prevent a queue from forming.
04:05.31Rapier1024that's what I thought
04:05.35jaskewupstream speed - say 5%
04:05.44jaskewI men  "minus" 5%
04:05.47Rapier1024it could never overcome the queu
04:06.51Rapier1024the sonicwall allows me to type a specific value in. The switch I have options I can pick, 2mb is one of them
04:06.55jaskewoh - you need to be in control of ALL traffic.  That's a prerequisite!
04:07.41Rapier1024yeah, there are two sonicwalls on this line out. one for computer lan, and one for phone so I have a good amount of control over what goes in or out
04:08.06jaskewthat could present a challenge - best of luck with it
04:08.34Rapier1024fortunately they allow seperate limits for both in and out so I won't be killing download speed
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04:08.50Rapier1024thanks again, good night all
04:08.55jaskewyou bet
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04:31.55CyberCodanyone familiar with the SpoTel TDM410's? are they comparable to the digium cards?
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04:56.28sevvi can't figure this out - are all of the pap2 firmwares compatible with the spa-2002 hardware?
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05:12.59aruntomari'm getting this error " Unable to forward voice or dtmf", i googled a lot, but can't find the solution
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05:48.10kukuI get You do not appear to have the sources for the 2.6.18-164.6.1.el5.centos.plusPAE kernel installed - but kernel-PAE-devel is installed.
05:55.36voipmonkwhat version, kuku? did you make symlinks?
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07:55.13benngard~phones
07:55.14infobotWhile personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, Cisco 79XX, everything else.  Do not consider Grandstream phones.  Ever.
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07:58.54bastyHi
07:59.04*** join/#asterisk ktwilight[m] (~ktwilight@27.170-65-87.adsl-dyn.isp.belgacom.be)
08:00.02bastyI have a strange problem. I just switched from Asterisk 1.2 to 1.4.29. I am using AgentCallbackLogin(${CALLERID(NUM)}||${CALLERID(NUM)}@intern) for my agents to login. As soon as the queue gets called - the Agent is ringing. But as soon as the Agent tries to answer - nothing happens - it keeps ringing...
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08:02.12TommyBottenbasty: Try to do a 'core set verbose 6' and paste the information to pastebin.ca and provide the link
08:02.37bastyTommyBotten: the strange thing is, that in the cli debug everything seems normal...but lemme try again
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08:07.55TommyBottenin that case, enable sip debugging, and see what the phone is doing at pickup time
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08:10.43bastyTommyBotten: ah okay...it seems that I have esthablished the call - but I cant hear anything
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08:15.05bastyTommyBotten: the state of the agent is still
08:15.09basty"ringing" i mean
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08:15.38TommyBottenHmm.. Did you look at the SIP traffic?
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08:22.25bastyTommyBotten: nope...i dont know how...i am kinda newbie ;)
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08:24.47bastyTommyBotten: but it seems that the local channel is making the problem
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08:26.31TommyBottenDo you get any error messages on that?
08:26.45TommyBottenfor sip debugging, do a 'core set debug' and then TAB to see your options
08:29.09bastyTommyBotten: http://pastebin.com/m336c06bc
08:29.17bastythats all i see...with full debug on asterisk cli
08:29.36bastyTommyBotten: the agent is trying to answer the channel..but I keep listen to the "music on hold".
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08:31.11bastyTommyBotten: If I set static agent with "SIP/87" into the queue.conf - it works...so it has to do something with the local channels
08:36.12TommyBottenAha.. so the agent is not static, I take it?
08:36.25TommyBottenHow is it configured?
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08:40.43bastyTommyBotten: the agent is using the agentcallback like AgentCallbackLogin(300${CALLERID(NUM)}||${CALLERID(NUM)}@intern) in the agents.conf I have the agent => 300100 for example
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08:53.53mbpWhat do people prefer for queue statistics? Asternic, QueueMetrics, queue_stats or something else?
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08:55.24kaldemarTommyBotten: SIP debug is enabled with sip set debug, not with core set debug.
09:06.52TommyBottengood point
09:06.55TommyBottenSorry about that
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09:12.40bastyTommyBotten: if i change the stuff to "add queue member" everything works
09:12.48bastyso it has to be something weirdo with the "local" channel it seems
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09:14.15TommyBottenSounds reasonable
09:14.25bastybut i would like to fix that
09:14.25basty;)
09:14.31TommyBottenI haven't used the config with AgentCallBack
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09:33.24_omerhello
09:34.04_omerIs there a way to get AGENT or PEER logged in time ? for example:  3 minutes 24 seconds since logged in ... or something like that?
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09:46.11krionhi guys
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09:55.00MrM4xXxhello i'm european and i configured a patton as isdn-voip gateway.., when i call the signal results the typical american signal, not the european..; any idea?
09:55.21MrM4xXxis it an asterisk conf?
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10:29.32kaldemarMrM4xXx: depends on the whole scenario. in asterisk, you need to set your country in indications.conf. but your phone may also give you wrong tones if you're using a voip one.
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10:34.35MrM4xXxkaldemar: thx, I try
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10:47.36MrM4xXxkaldemar: thx, it's ok, now it's working properly. ;)
10:56.28krionanyone with a png who explain what's expected when 407 happened ?
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11:12.50kaldemarkrion: what 407?
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11:46.49Zap-Wdoes asterisk behave like a RTPProxy by default?
11:47.57kaldemarZap-W: no
11:48.52kaldemarZap-W: asterisk doesn't force itself on the media path by default.
11:49.31Zap-Whow do I turn it on, then it will modify some SIP headers and some arguments in SDP payload?
11:49.39kaldemarwhat version are you using?
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11:50.10Zap-W1.4.26.1
11:50.46kaldemarput canreinvite=no into sip.conf
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12:03.15Zap-Winteresting http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite
12:07.32arj__is there more to enabling "blind transfer" than placing it into features.conf? I enabled it, but #1 (or ## or #) don't seem to be doing anything
12:09.44kaldemarZap-W: what is?
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12:19.15lbarthhello
12:20.12lbarthCan anyone here help me with this ao2_unlock issue? (15915)
12:21.11lbarthi have no clue how to get asterisk 1.6 stable on debian
12:21.35lbarthit can't be true that i am the only one running asterisk 1.6 in a heavy load envireonmen
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12:23.51kaldemarlbarth: can you use another timing module?
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12:24.37Dovidoej: ping
12:26.29lbarthkaldemar: currently i have no interface cards in the maschine so i can currently only se pthread
12:27.12lbarthbu i just disovered that there is a packege timerfd
12:27.15kaldemarlbarth: there's timerfd in 1.6.2 and always dahdi_dummy
12:27.55lbarthis dadi_dummy a good solution for production?
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12:28.19kaldemarpeople do use it, but you better test it for yourself first.
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12:31.49lbarthgreat compiling dahdi fails... i hate those days
12:33.05Dovidlbarth: You will live
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12:39.36lftsyHello everybody, since my server is currently in a CPU 100% mode, and flooding SIP OPTIONS, I was able to provide many logs for ticket https://issues.asterisk.org/view.php?id=16382 Is there a developper that can say if another dump would be useful? Thanks
12:41.16Dovidlftsy: Try asterisk-dev in a few hours when they wake up in Huntsville
12:42.34lftsyDovid: Thanks, it's just that I cannot keep this server blacklisted many time since all calls are now running on only one server... I will try it anyway. Thank you
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12:43.33proutehi everybody
12:43.34kaldemarlftsy: how about changing rtautoclear and ignoreregexpire?
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12:43.56mbrevdalooking for waywas to covert a pdf to tif for FFA - without using GhostScript
12:44.06prouteDoes Dahdi works fine with B410P and asterisk 1.6.0.x? Or is it better to works with asterisk 1.6.1.x?
12:44.21mbrevdaI thought imagemagick could do it, but I cant get the correct commands
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12:46.19bastyHi
12:46.24lftsykaldemar: I have tried to play with rtautoclear and ignoreregexpire but there is always peers with Expire -1 that apperas
12:47.07bastyI just updated my asterisk pbx from 1.2 to 1.4.29. I have a lot of snom phones, that uses the blf function. It worked fine in 1.2.X but after the upgrade - the blf doesnt work anymore. Anyone knows why ?
12:47.26kaldemarmbrevda: is this chennel a phreaking google proxy to you?
12:47.42kaldemarmbrevda: imagemagick can do it, the command is convert.
12:48.36mbrevdakaldemar: convert itself doesnt work without the right commands...
12:48.44mbrevdahmm s/commands/arguments ?
12:48.52kaldemarmbrevda: convert your.pdf yout.tif
12:49.06mbrevdadid that - ffa wont send the fax then
12:49.15*** join/#asterisk razu (~razu@razu.data.ee)
12:49.16mbrevda'cannot queue file' or somethign
12:49.29kaldemarask digium about the format then.
12:50.01mbrevdakaldemar: why? are they a chennel for phreaking google proxy to you?
12:50.22mbrevda(sic)
12:50.57kaldemarmbrevda: fax for asterisk is a commercial product that they provide support for...
12:55.43ManxPower-workbasty: the UPGRADE*.txt files should mention the call-limit changes that effect BLF.  Always read the UPGRADE*.txt files when upgrading.
12:56.32bastyManxPower-work: mh - so that means, i will have to set a call-limit on every sip account ? When I try to check "sip show subscription" everything is on "idle".
12:57.04ManxPower-workbasty: It means you should read the UPGRADE*.txt file.
12:57.17ManxPower-workI guess I could go read it for you, but I'm not that nice.
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12:58.18bastyManxPower-work:  hehe - but maybe you know it without reading it.. ;-)
12:58.25ManxPower-workbasty: I don't.
12:58.34bastymhh well
12:58.37bastythanks!
12:58.45ManxPower-workThe call limit stuff is constantly changing in Asterisk.
12:59.25ManxPower-workbasty: You should know that it does not matter WHAT the limit option is set to, just as long as it it set.
12:59.34ManxPower-workas it is set.
12:59.43bastyokay thanks
13:00.20bastyManxPower-work:  one more question...do I need to set the call-limit for every single sip account, or could I put something into the global section of sip.conf ?
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13:06.49ManxPower-workbasty: I don't know, but sip.conf.sample should indicate where you can put the option.
13:07.03bastyokay
13:07.20bastywell..i just set the call-limit for every peer..but it is still not working..strange thing that.
13:09.20ManxPower-workbasty: did you read the ;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ---------------------------- section of sip.conf.sample?
13:09.28bastyyep
13:09.43ManxPower-workso you have limitonpeers as the other settings set?
13:09.48bastyyup
13:10.17Naikrovekyawns. omg tired
13:10.54ManxPower-workbasty: put your sip.conf on pastebin.ca masking only passwords
13:11.20bastyokay sec
13:12.41bastyAH...I guess I found out, what the problem is
13:13.00bastyi had to restart the phone..
13:13.09bastyold firmware....6.2.2 installed there..
13:13.48leifmadsenbasty: what version of asterisk?
13:14.41leifmadsenah, 1.4.29
13:14.47*** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163)
13:14.58leifmadsenand I didn't read far enough in to see you fixed it
13:15.00leifmadsenmoves along now
13:19.03basty;-)
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13:28.46bastybye
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13:33.43Zap-Whi, in SIP RFC it says, server uses the port in the SIP header message for  responding to requests "SIP operates in this manner so that a server   can listen for all messages, both requests and responses, on a single   IP address and port."   I don't understand why it wouldn't be able to use a single IP address and port if it didn't look the port in the header message and instead used the UDP port the request came from
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13:43.20ManxPower-workZap-W: nat=yes tells asterisk to look at the packet headers, not the data portion of the SIP packet.
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13:44.39florzZap-W: which section would that be? I can't find any such text ...
13:44.54Zap-Wflorz, http://www.ietf.org/rfc/rfc3581.txt
13:45.39ManxPower-workZap-W: By default all SIP devices look at the contents of the packet to find the address/port.  This is the way SIP works.   nat=yes tells Asterisk to stop doing that and try to infer the port/address from the packet header, not the SIP header.
13:45.49florzZap-W: that's not exactly the SIP RFC =:-)
13:45.59ManxPower-workZap-W: that's not the SIP RFC.
13:47.10smooth_penguinIAX is sooo nice
13:47.22smooth_penguin1 port and all
13:47.24Zap-Wan extension
13:47.32Zap-WI tell a white lie
13:48.02Zap-Wthanks
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13:52.53rhpHi all. I would like to setup a system where a number of users can talk to eachother using head-sets. I was looking whether asterisk with SIP is an answer for this. Is it possible to talk with more than one person at a time with asterisk?
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13:55.00[TK]D-Fenderrhp: Yes.  MeetMe conferencing.
13:56.04rhpthanks. I've tried looking for some tutorials for setting this up, but I could not find much. Could you direct me to a tutorial, if it exists?
13:57.05[TK]D-Fenderrhp: Install *.  Read the book.  make a basic dialplan to call MeetMe
13:57.08[TK]D-Fender~book
13:57.08infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
13:57.42rhp[TK]D-Fender: thanks a lot.
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14:02.41ManxPower-workrhp: Asterisk is a toolkit that allows you to build a PBX.  It's not designed to be functional "out of the box"
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14:07.14Kattyhi
14:07.39[TK]D-Fenderrhp: think of it like a box of artist's pencils.  They may have instructions on how to properly open and close the box and how to use the sharpener it comes with, but don't expect to get instrucion on how to draw some specific thing that comes to mind.
14:08.03sun28moin \o/
14:08.15[TK]D-Fenderrhp: Everyone does things their own way.  Dialplan is programming.  You say what you want * to do with every step
14:08.22Kattygood morning.
14:08.43smooth_penguinhi Katty :>
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14:10.07bsdmailcan someone help-me with this issue: http://pastebin.com/d49e55da3 ?
14:11.47Kattywonders where all the critters are
14:12.02ManxPower-workbsdmail: you have a DTMF mode issue with your provider
14:12.19ManxPower-workbsdmail: you do not have one-way audio issues?
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14:14.19bsdmailManxPower-work no, i can hear and he can hear.
14:15.12bsdmaili don't unterstand why this is happening, because i have to type the number of my friend, so before he answer my dtmf is being recognized
14:16.53ManxPower-workbsdmail: it's being recognized on the INBOUND CALL.  There is also the OUTBOUND call.
14:18.51Kattyhugs smooth_penguin
14:19.21Kattysmooth_penguin: what did you make for dinner?
14:19.50smooth_penguinKatty, well nothing as yet just had some tea :>
14:21.05*** join/#asterisk Akiraa (~Akiraaaa@79.112.39.45)
14:22.13*** join/#asterisk jaytee (~jforde@unaffiliated/jaytee)
14:22.59bsdmailManxPower-work both calls are from same trunk/provider
14:23.12ManxPower-workbsdmail: That does not matter.
14:23.45ManxPower-workDTMF issues are some of the most complex and hardest things to fix.
14:24.43bsdmailok i think i'm gonna give up
14:24.51bsdmailthanks
14:25.35smooth_penguinKatty, whats cooking ?
14:25.54[TK]D-Fenderbsdmail: What mode do you have set?
14:26.37bsdmaili dont know
14:27.24[TK]D-Fenderbsdmail: This is YOUR configuration.  How do you NOT know what you set?
14:28.38krionkaldemar: i wasn't calling the right sip URI...
14:28.48bsdmailbecause i'm new and i didn't set this kind of config, so i'm using the default?
14:29.35bsdmailDTMF is when you press a key and the pbx recognize the sound and assume a number to this sound right?
14:29.49krionyes
14:30.19krionyou got the cmd Read who's fun with that
14:30.35krionRead(digito||1)
14:30.38krionSayDigits(${digito})
14:30.53krions/Read/SayDigits
14:30.58[TK]D-Fenderbsdmail: Go look at your configs.
14:31.18krionhi [TK]D-Fender
14:31.18bsdmail[TK]D-Fender can u tell where i should look?
14:31.26bsdmailsection....
14:31.33[TK]D-Fenderbsdmail: SIP.CONF.
14:32.03kukuI get You do not appear to have the sources for the 2.6.18-164.6.1.el5.centos.plusPAE kernel installed - but kernel-PAE-devel is installed.
14:33.22[TK]D-Fenderkuku: Also need the headers and a bunch of other stuff.  Read the insttructions in the tarball
14:36.33*** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil)
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14:38.47kuku[TK]D-Fender: ok
14:42.16*** join/#asterisk mrprozac (~mrprozac@132-82-ftth.onsneteindhoven.nl)
14:46.29*** join/#asterisk drako (~luisjose@nelug/coreteam/luisjose)
14:46.51drakoWhat's a good TextToSpeech ?
14:46.57*** join/#asterisk jakent (~john@soleil.johnkent.mooo.com)
14:47.20*** join/#asterisk crochat (~crochat@80.83.52.178)
14:47.39jayteeCepstral
14:48.08Faustovisn't that more the other way, speech recognition?
14:48.28jayteeFaustov, no, you're probably thinking of Lumenvox
14:48.48kuku[TK]D-Fender: I read the README's for dahdi-tools and dahdi-linux, and I don't anything on headers
14:51.20Kattyherroes.
14:51.21Kattyagain
14:51.51drakojaytee: looks pretty nice
14:52.26leifmadsenhuh, well recording video prompts with H263 looks ok, but not so much with H264 :)
14:53.54ManxPower-workkuku: do "rpm -qa | grep kernel" and pastebin the result.
14:56.47kukuhttp://pastebin.com/d4ed9b79
14:57.43*** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil)
14:57.56ManxPower-workkernel-headers-2.6.18-164.11.1.el5.centos.plus
14:58.08ManxPower-workthat does not match your running kernel.
14:58.20kukuremove ?
14:58.34ManxPower-workI would remove the old RPMs and make sure you have the same version of kernel, kernel-dev and kernel-headers
14:58.53ManxPower-workthen reboot to make sure your running kernel is the same as your current kernel
14:59.32ManxPower-workyou should be able to leave the actual kernel RPMs, but remove any of the kernel-dev and kernel-headers and reinstall the same version of those RPMs as your kernel.  If they don't all match DAHDI won't compile.
14:59.46kukumakes sense
15:00.11kukuyum wants to remove gcc as part of a dependency of kernel-headers-2.6.18-164.11.1.el5.c                            entos.plus
15:01.34kukuI'm looking for the flag so it doesnt remove gcc
15:02.29leifmadsenif you install the newer versions first, then it might not do that
15:03.34leifmadsenkuku: rpm -e --nodeps packagename  -or-
15:03.34leifmadsenyum remove --nodeps packagename
15:03.57kukuhttp://pastebin.com/d37172bb8
15:04.04kukuok
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15:13.40*** part/#asterisk benngard (~benngard@213.88.138.230)
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15:14.55kukuleifmadsen: thank you sir ! I removed the old devel rpm's and installed an older PAE devel ( since I had the latest kernel, but since I didn't reboot, it wasn't loaded )
15:15.13kukuleifmadsen: dahdi is now compiling - thank you once again.
15:15.28mnick86does somebody know if US-analog-phones use the same standard than german analog phones ?
15:15.41leifmadsenkuku: thank ManxPower-work, he did most of the suggesting :)
15:16.55*** join/#asterisk Zambezi (Zulu@unaffiliated/zambezi)
15:16.58kukuspoke too soon: http://pastebin.com/d5006db0f
15:19.52mnick86some german asterisk user online ?!
15:20.29lbarthyes
15:21.07*** join/#asterisk mike8 (~mike@c14.audioline.ba.cust.gts.sk)
15:21.20lbarthmnick86 how can i hel you
15:21.56*** join/#asterisk styelz (~yoohoo@m0o0.mooo.com)
15:22.09*** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq)
15:24.51mnick86lbarth: have you ever tried a german analog phone on a Digium analog card ?
15:26.36jayteebe back later, going home early
15:27.35*** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage)
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15:35.55angryuserHave someone found the origins of :  WARNING[22868] channel.c: Exceptionally long voice queue length queuing to Local/1010@from-internal-2277,2   (asterisk 1.4.27) ?
15:35.58tzafrir_laptopmnick86, is there such a thing as a german analog phone?
15:36.04*** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net)
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15:38.32*** join/#asterisk ELSEGO (~Juankar@unaffiliated/elsego)
15:38.56ELSEGOhi people
15:39.00mnick86I can plug an anlog phone into "FXO Kewlstart" right ?!
15:39.08*** part/#asterisk ELSEGO (~Juankar@unaffiliated/elsego)
15:39.53Chainsawmnick86: An FXS port running FXO kewlstart signalling is suitable for that, yes.
15:40.15*** join/#asterisk szasz (~szasz.sza@89.238.223.70)
15:41.54szaszhi all
15:41.55Naikrovekkewlstart?
15:42.18szaszin some calls I found that my Asterisk doesn't answer the BYE message
15:42.22mnick86ok thanks ... damned phones
15:43.09tzafrir_laptopNaikrovek, though the phone couldn't care less if it is Kelstart or Loopstart
15:43.16ChainsawNaikrovek: It's a fancy word for disconnect supervision.
15:43.19Naikroveki was asking wtf kewlstart was
15:43.27Naikrovekkewlstart doesn't seem like a real name is all
15:43.32Naikrovekbut if it is that's kewl
15:43.40szaszhow can I solve this issue?
15:44.03Naikrovekszasz: you'll need to provide some SIP debugs of the behavior
15:44.26*** join/#asterisk Skeeter- (~Skeeter@190-141.cgocable.ca)
15:44.27Naikrovekso we can see the reason (or at least what Asterisk sees as the reason) for ignoring the BYE
15:45.07JonaYHi, I'm having a problem with asterisk - as soon as I connect our ISDN line I can no longer make internal calls, I've only got one outbound route (9|.) anyone got any ideas?
15:45.36ChainsawJonaY: It depends on where you're connecting this ISDN line really.
15:46.25mnick86how many wires does an american analog phone has in their connection cable ?
15:46.38JonaYIt's into a Fritz PCI card, using the mISDN driver
15:47.05szaszit's hard to reproduce, but I will try
15:47.11[TK]D-FenderJonaY: FreePBX is NOT supported here...
15:47.12[TK]D-Fender~freepbx
15:47.13infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx
15:47.16[TK]D-Fender^^^^
15:47.30*** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net)
15:49.14JonaYok thanks I'll try there
15:50.03szaszNaikrovek : are you talking about sip debugs from Asterisk or is enough the sip trace from my wireshark?
15:51.52krionhow do you get the jitter from different call ? and what's an acceptable average jitter
15:52.59*** join/#asterisk Tagor (~none@s55928c6d.adsl.wanadoo.nl)
15:54.09TagorIt seems my Asterisk is using random ports. I have set port 5060 as SIP port in sip.conf and RTP 10000_20000. But in my debug I see this:
15:54.13Tagorast_rtcp_write_sr: RTCP SR transmission error to xxx.xxx.xxx.xxx:5005, rtcp halted Operation not permitted
15:54.49TagorAnyone who knows how to force Asterisk to use port 5060?
15:54.56[TK]D-FenderTagor: TRANSMISSION.  the port you set are what you RECEIVE ON
15:54.56TagorOr does it need to use more ports?
15:55.37Tagor[TK]D-Fender: Ah I see, 5005 is the port of the other party?
15:55.47[TK]D-FenderTagor: Clearly
15:58.06Naikrovekszasz: sip debugs from asterisk.
15:58.27TagorAnother thing, in my debug output I see this: Peer audio RTP is at port 194.120.0.166:41880. This is also the other party right?
15:59.51*** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler)
15:59.55ManxPower-workTagor: that looks NATish
16:00.17*** join/#asterisk cweagans (~cweagans@71-33-110-201.bois.qwest.net)
16:00.20ManxPower-workTagor: we don't know since you masked all your IP addresses.
16:00.35TagorManxPower-work: that's the ip of my sip provider
16:01.11*** join/#asterisk Defraz (~Defraz@corp.fuzecore.com)
16:01.18ManxPower-workremember you can only control the ASTERISK side of the connection.
16:01.33TagorManxPower-work: I'm trying to find out why I don't hear anything when my firewall is on. I opened 5060 and 10000_20000. As far as I can see in the debug (Audio is at 85.92.137.169 port 14758) it should work
16:01.53TagorManxPower-work: does that mean that if the provider uses other ports I need to open them too?
16:02.50ManxPower-workTagor: do you have a FIREWALL or just a NAT ROUTER?
16:03.19*** join/#asterisk myrthful (~emondpd@mercury3.Physics.McMaster.CA)
16:03.24TagorManxPower-work: just a firewall (APF), the server is connected to the internet directly
16:03.42ManxPower-workthen you should see which packets are being blocked in your firewall logs.
16:04.04TagorManxPower-work: any idea how I can do this with APF?
16:04.08ManxPower-workI suepect you set your firewall to expect the far end to use the same ports as you configured in Asterisk.  That is not the case.
16:04.19ManxPower-workTagor: I can't help support your firewall.
16:04.25myrthfulWould someone mind helping me figure out why my SendDTMF calls don't seem to be working?
16:04.44ManxPower-workmyrthful: You don't hear the DTMF when calling that extension?
16:04.59myrthfulExactly
16:05.13TagorManxPower-work: does that mean it uses other ports besides the ones I configured in Asterisk (5060 / 10000_20000)?
16:05.14myrthfulI've read about other people having this problem, but can't find a solution
16:05.21ManxPower-workmyrthful: pastebin the output of a failed call including the SendDTMF.
16:05.32ManxPower-workTagor: the FAR END will, yes.
16:05.52ManxPower-work~pb
16:05.53infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org ,  http://bin.cakephp.org/ , or apt-get install pastebinit
16:05.56myrthfulManxPower-work: Ok, one minute
16:06.02myrthfulManxPower-work: and thanks for your help
16:06.48TagorManxPower-work: sorry, could be my English, but what do you mean with 'the far end will'?
16:06.58ManxPower-workTagor: This is networking 101, you should know that the source port of a connection is usually chosen by the OS.
16:07.25ManxPower-workTagor: each connection has two endpoints.  In your case, one endpoint is Asterisk, the other endpoint is your provider.
16:08.12ManxPower-workYou can only control the ports Asterisk uses.  Your firewall rules should not specify the ports or the IP the provider will use.
16:09.06*** join/#asterisk GGD (~GGD@ip72-196-241-104.dc.dc.cox.net)
16:09.16ManxPower-workThe provider could easily send audio from a totally different set of IP addresses and ports.
16:09.38*** join/#asterisk styelz (~yoohoo@m0o0.mooo.com)
16:10.04TagorManxPower-work: but if Asterisk is only listening on the ports I configured then it wouldn't work either if I had turned my firewall off?
16:10.20ManxPower-workTagor: That is wrong.
16:10.34ManxPower-workThe port numbers are NEGOTIATED for each call.
16:10.37*** join/#asterisk Deeewayne (~dwayne@75.76.254.162)
16:10.38*** mode/#asterisk [+o Deeewayne] by ChanServ
16:10.56bmoraca_workTagor: the RTP ports you specify are the ports that asterisk will use to initiate a connection with an endpoint.  the actual ports used will be, as ManxPower-work states, are negotiated as part of the call setup (INVITE)
16:11.42ManxPower-workTagor: So it doesn't matter what ports the far end uses, the far end tells Asterisk what ports it will use during the call setup.
16:11.46TagorManxPower-work: ok so they are assigned by my Asterisk box? So if I say the traffic should go over 10000_20000, it WILL go over these ports?
16:12.21ManxPower-workTagor: Do you understand that all IP connections have a source IP/source port and destination IP/destination port?
16:12.33TagorManxPower-work: yes I do understand that part
16:12.54*** join/#asterisk Skeeter- (~Skeeter@190-141.cgocable.ca)
16:12.56ManxPower-workTagor: then your question makes no sense.
16:13.03ManxPower-work(11:11:46 AM) Tagor: ManxPower-work: ok so they are assigned by my Asterisk box? So if I say the traffic should go over 10000_20000, it WILL go over these ports?
16:13.14ManxPower-workThat makes no sense since you do not talk about the other end of the connection.
16:13.45*** join/#asterisk jakent (~john@68-247-204-36.pools.spcsdns.net)
16:13.45ManxPower-workAsterisk will SEND AUDIO FROM the ports you configure in rtp.con.  the DESTINATION PORT could be ANY PORT.
16:13.58TagorManxPower-work: I thought it negotiates over port 5060 (standard sip port) and then it will establish a connection on one of the RTP ports I allow?
16:14.23ManxPower-workTagor: Each call has two audio connections.  Asterisk -> provider and provider-> asterisk.
16:15.09TagorManxPower-work: Ok, so the provider's ports do matter?
16:15.21TagorManxPower-work: so I should find out which ports my provider uses?
16:15.22ManxPower-workwhen the call gets set up, asterisk says, this the range of audio ports on my side that I will talk on.  In your case that is 10,000 - 20,000.  The provider then says "these are the ports I will accept audio on".
16:15.33*** join/#asterisk Heretic (~fallen@dsl-246-127-184.telkomadsl.co.za)
16:15.40ManxPower-workTagor: no!  You should not limit what ports you allow in your firewall for packets coming from or going to your provider.
16:15.54Hereticlo all
16:15.54*** join/#asterisk atis_work (~atis_work@193.238.212.171)
16:16.00ManxPower-workYou are doing that.  Stop doing that.
16:17.12TagorManxPower-work: but what if my provider uses several ips? They use a domain and the ips are different every time
16:17.25ManxPower-workTagor: Nobody ever said this was going to be easy.
16:18.07*** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk)
16:18.12ManxPower-workI wish you the BEST of luck and cannot help you further.
16:18.21*** join/#asterisk darkskiez_ (~dz@62-50-207-156.client.stsn.net)
16:18.49TagorManxPower-work: well the thing is this worked with my previous provider. I just openend 5060 and 10000_20000 and it worked. Besides that if I want to connect to my Asterisk box from another location I will never be able to if I need to enter the ips first
16:19.36*** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844)
16:19.57ManxPower-workIt's not that difficult to set your firewall to ignore the far end address/port
16:20.18p3nguinWhy would you want to ignore your RTP stream, anyway?
16:20.39p3nguinForward 5060 and 10000:20000.  Done.  Enjoy.
16:20.42*** join/#asterisk cobolfoo (~cobolfoo@bas7-quebec14-1096763909.dsl.bell.ca)
16:20.57cobolfooHello, I need some support with DAHDI developement in C, there is the good channel to ask questions?
16:20.58fenrusallow established tcp sessions <3
16:21.07fenruscoffe.
16:21.17p3nguinRTP is UDP, not TCP.
16:21.24ManxPower-workfenrus: too bad SIP and RTP is UDP, not TCP.
16:21.41ManxPower-workcobolfoo: #asterisk-dev
16:21.45cobolfoothank you
16:21.46p3nguinSIP _can_ be TCP, but not many people use it there.
16:21.46fenrusManxPower-work, indeed.
16:21.52Tagorp3nguin: I have allowed 5060 10000_20000 UDP in my firewall. But I still have no audio
16:22.16*** join/#asterisk joesuffceren (~chatzilla@ip68-104-167-226.ph.ph.cox.net)
16:22.20p3nguintagor: Are there other problems that I don't know about?  Keep in mind I just got here.
16:22.22joesuffcerenanyone know if it's possible to setup a call reject/ignore button or softkey on the cisco 7940s with SIP firmware?
16:22.38ManxPower-workp3nguin: his problem is that he neither understands SIP nor his firewall.
16:23.34ManxPower-workMaybe he has SIP AGL enabled on his firewall.  It's a firewall issue, not an Asterisk issue.
16:24.39*** join/#asterisk moy (~moy@74.12.129.100)
16:25.03Tagorp3nguin: the problem is that my Asterisk box doesn't work (no audio) when my firewall is enabled. I just switched my outgoing sip provider. It worked fine with the same firewall rules with my previous provider
16:25.31*** join/#asterisk CunningPike (~CunningPi@204.239.8.157)
16:26.01TagorManxPower-work: I know it's a firewall issue. But I don't understand what's wrong. And it looks like I don't understand you, at least it's confusing me
16:27.24Skeeter-which OS fits best Asterisk?
16:27.40TagorManxPower-work: I don't understand why you can't tell me what is wrong or how I can solve it. I mean I don't have to know how to make a car to be able to drive it.
16:27.52ManxPower-workSkeeter-: Linux
16:28.01Skeeter-i meant distro, sorry
16:28.08ManxPower-workSkeeter-: the one you prefer.
16:28.09QwellSkeeter-: doesn't matter
16:28.21Skeeter-Debian is giving me a hard time
16:28.21*** join/#asterisk defswork (~andy@mx2.3gcomms.co.uk)
16:28.32ManxPower-workTagor: you are not trying to drive a car.  You are trying to rebuild part of your car.
16:28.39Skeeter-I would like to switch to another one, and i need suggestion
16:29.09*** join/#asterisk jmacz (~jmacz@190.144.75.22)
16:29.19TagorManxPower-work: what's the problem with telling me what I'm doing wrong?
16:29.43*** join/#asterisk eppigy (~Dave@216-139-241-102.aus.us.siteprotect.com)
16:29.51eppigyhello i am dave
16:31.05TagorManxPower-work: you're trying to force me to make a study of this instead of telling me what I'm doing wrong. I first need to do a knowledge test before you tell me what I'm doing wrong... I don't see the point of this
16:31.06Skeeter-ManxPower-work, Qwell : what distro are you guys using
16:31.06ManxPower-workTagor: because I am not interested in spending 6 hours working with you to learn your firewall, your network, your setup, and your provider.  Specifically I'm not interested in learning, and then teaching you how to debug your firewall.
16:31.13ManxPower-workSkeeter-: I use CentOS
16:31.40*** join/#asterisk hluesea (~hulusikah@88.247.127.66)
16:32.00ManxPower-workSkeeter-: You'll find people here that run Asterisk just peachy on Debian
16:32.02TagorManxPower-work: probably this is some small thing I did wrong. For example I allowed the wrong ports
16:32.44TagorManxPower-work: I have never asked you to teach me how a firewall works etc. I just asked what could cause that there's no audio
16:33.11*** join/#asterisk kruemeltee (~Maddin@port-92-198-62-82.static.qsc.de)
16:33.18kruemelteehello again ;-)
16:33.23ManxPower-workAnd I told you what could cause no audio.
16:33.49*** join/#asterisk JT (~j@unaffiliated/jt)
16:34.04ManxPower-workLog what the firewall is blocking, then go from there.  Exactly like anyone else would debug the same issue.
16:34.06TagorManxPower-work: I already knew, my firewall is blocking traffic. But the question is WHY is it blocking the traffic? And don't tell me know 'cause I configured it like this', cause I already know that
16:34.16Skeeter-ManxPower-work, asterisk part is very fine, i cant get those Sangoma A200 to work properly
16:34.22ManxPower-workTagor: you are missing the "WHAT is being blocked".
16:34.43ManxPower-workSkeeter-: did you follow the Asterisk install instructions from Sangoma's Wiki site?
16:35.13ManxPower-workTagor: when you can put a pastebin of your firewall blocking packets then someone may be able to help your further.
16:35.23ManxPower-workBut *I* have a job and cannot spend any more time with you.
16:37.01Skeeter-ManxPower-work, yes i did, somehow it try to go out via ZAP/g0 instead of DAHDI/g0
16:37.55ManxPower-workSkeeter-: We can't support a GUI Asterisk here.
16:37.57Skeeter-Ive done that kinda of setup before and got everything working just fine.. I would like to fix it instead of formatting and reinstalling everything
16:38.46[TK]D-FenderSkeeter-: If you don't like the channel type its dialing, then fix your config, fix your GUI, or ditch both and do it yourself
16:39.49Skeeter-ManxPower-work, the GUI is not installed yet, i always try to make the Sangoma work then i install the GUI
16:40.23*** join/#asterisk ttl- (~patrick@d5153A420.access.telenet.be)
16:40.27[TK]D-FenderSkeeter-: well "somehow it try to go out via ZAP/g0 instead of DAHDI/g0" = dialplan.  Go fix it
16:40.37ManxPower-workSkeeter-: Dude, YOU CONFIGURED it.  Just fix the damn channel name.
16:41.22Skeeter-[TK]D-Fender, ManxPower-work : making a call
16:41.45Skeeter-aight, instead of CHANUNAVAILABLE, i get CONGESTION
16:41.57p3nguintagor: "iptables -L -nv" and put the output into a pastebin.  If you're NATing, show "iptables -t nat -L PREROUTING -nv" too.
16:43.12myrthfulManxPower-work: I've got a pastebin for the PlayDTMF problem I mentioned earlier: http://pastebin.com/d633b1478
16:43.15Skeeter-i used ./Setup dahdi, with that command, there is not much more during the configuration process that you can do
16:43.49p3nguinskeeter-: Distro choice is like choosing what to have for supper.  Some people use Debian, some CentOS, I use ArchLinux, I've seen people using Gentoo...  Pick one, use it, like it.  If if you don't like it, change it.
16:43.56Chainsawleifmadsen: Ready for testing now, I've backported for 1.6.1 & 1.6.2: https://issues.asterisk.org/view.php?id=16470
16:44.41Skeeter-p3nguin, i like Debian, i got it for my home server and works just as espected for what I wanna do, i just think that it aint the best choice for Asterisk
16:44.48leifmadsenChainsaw: okie, will update status
16:45.13leifmadsenSkeeter-: works fine -- just don't use the built in packages which tend to be woefully out of date
16:45.57Skeeter-leifmadsen, what do you mean by built in packages??
16:46.04p3nguinskeeter-: I agree... I wouldn't use Debian for Asterisk, but I also wouldn't use it for anything.
16:46.09leifmadsenSkeeter-: I mean don't install asterisk via apt-get
16:46.15leifmadsenp3nguin: amen :)
16:46.19Skeeter-leifmadsen, i dont use apt-get for asterisk
16:46.31leifmadsenSkeeter-: I never said you did -- I just said don't use the packages via apt-get
16:46.33Skeeter-leifmadsen, i download the files from asterisk and Compile
16:46.44leifmadsenI like Ubuntu and CentOS
16:46.50Skeeter-leifmadsen, no problem, just wanted to specified
16:46.55leifmadsenbut whatever works is what you should use
16:47.00p3nguinUbuntu falls into the Debian category.
16:47.20Skeeter-Ubuntu has no restriction for non-free package
16:49.05Skeeter-the Linux-guy here always uses Gentoo
16:49.27TagorManxPower-work: here's the answer you asked for: PROTO=UDP SPT=19842 DPT=24872
16:49.57Skeeter-btw, is there any Timecard addon/app for Asterisk?
16:50.06p3nguintagor: Is that on an outbound packet?
16:50.15Tagorp3nguin: yes
16:50.29Tagorp3nguin: ** OUT_UDP DROP ** IN= OUT=eth0 SRC=85.92.137.169 DST=77.72.168.144 LEN=200 TOS=0x00 PREC=0x00 TTL=64 ID=0 DF PROTO=UDP SPT=19842 DPT=24872 LEN=180
16:50.51theharAnyone in Utah needing a job.  We're hiring a Director of NetOps @ XMission http://saltlakecity.craigslist.org/sad/1596128717.html
16:50.54*** join/#asterisk JT (~j@unaffiliated/jt)
16:51.05bmoraca_workSkeeter-: you could build one.  there are so many ways that you could do it that no one else's method is probably going to work.
16:51.06Skeeter-2nd, Cisco guy came to my work this week to show me how Cisco works, Cisco got a little "Wizard" when a user use the Voicemail for the 1st time, that would be great to be included in Asterisk
16:51.20p3nguintagor: It's dropping the audio packets?  That seems like a problem.
16:51.46Tagorp3nguin: thanks I will look into that
16:51.48Qwellthehar: How does it feel to host the best website on the planet?
16:52.15theharQwell: haha..
16:52.20Qwellerr, universe
16:52.24theharQwell: he is the only customer that has a subdomain to us as well
16:52.29Qwellseriously?
16:52.31theharyup
16:52.38Qwellhow'd he manage that?
16:52.45theharbecause maddox is awesome
16:52.51QwellThat he is..
16:53.19*** join/#asterisk dandate2 (~dan@112.206.130.149)
16:53.35theharnods
16:54.02p3nguintagor: The iptables output I asked for would most likely reveal the problem.
16:54.19dandate2when using asterisk 1.4: queue set to auto-fill: when there are a lot of people waiting on hold the cpu use of trying to ring multiple agents becomes so astronomical it damages the sound quality. <> Is this fixed in asterisk 1.6 or do I just need to get a faster machine?
16:54.53p3nguindandate2: What speed is your CPU?
16:55.11dandate22.2 ghz dual core intel
16:55.40dandate2with 10 people waiting on hold the cpu use jumps to 90%, if i disable auto-fill it dallies between 2% and 25%
16:58.16dandate2but sucks without auto-fill because there are 30 people ready to take calls, and it only rings one at a time
16:58.43*** join/#asterisk Zambezi (Zulu@unaffiliated/zambezi)
16:58.46p3nguineven using ringall strategy?
16:58.56dandate2yes ringall strategy takes a lot of cpu also
16:59.25bmoraca_workdandate2: just use roundrobin with a lower timeout.
16:59.41Tagorp3nguin: according the the output of iptables -L I think it's set correctly:
16:59.44TagorChain OUTPUT (policy ACCEPT)
16:59.44TagorACCEPT     udp  --  anywhere             anywhere            udp dpts:10000:20000
16:59.57p3nguinrrmemory, that is... since roundrobin is deprecated
17:00.12bmoraca_workwell, that's a given.  the concept is the same, though
17:00.14*** join/#asterisk paulc (~paulc@unaffiliated/paulc)
17:00.22fenrusanyone have a rrmemory configuration guide?, cant seem to get it working quite right
17:00.29dandate2i just need to get a quad-core u-server huh
17:00.47bmoraca_workno...
17:00.55p3nguinfenrus: It's simple.  Have strategy set to rrmemory and have multiple agents waiting for calls.
17:01.05sun28strategy = rrmemory
17:01.08sun28servicelevel = 10
17:01.13sun28timeout = 3
17:01.17sun28retry = 2
17:01.21sun28weight = 10
17:01.24fenrusp3nguin, well - asterisk wont send a call past the first phone..
17:01.25sun28ringinuse = no
17:01.25p3nguinfenrus: What doesn't work?  There's really nothing more to it.
17:01.29sun28timeoutrestart = yes
17:01.31sun28-_-
17:01.39ManxPower-workfenrus: IT SHOULD
17:01.43*** join/#asterisk timeshell (~chatzilla@gw.lusi.on.ca)
17:01.57fenrusManxPower-work, no matter how i'ts configured..?
17:02.16ManxPower-workfenrus: Obviously you screwed up your config somewhere.
17:02.27fenrusManxPower-work, that's why i asked for some input ;)
17:02.39ManxPower-workFor example maybe you don't have a call-limit set, or maybe yo don't have ringinuse=no
17:03.02*** join/#asterisk came0 (~came0@rrcs-71-42-53-182.se.biz.rr.com)
17:03.36bmoraca_workindeed...queues don't work right without call-limit set...even if it's set to an absurdly high number
17:03.44*** join/#asterisk ecrane (~ecrane@o1-69-19-166-10.static.o1.com)
17:04.11*** join/#asterisk etnos (~chatzilla@190.98.20.221)
17:04.38etnoshi you all.
17:04.42fenrushm, interesting..
17:06.24fenrussetting the call-limit of the member of the group?
17:09.56ManxPower-workbmoraca_work: you'd think that info would be in queues.conf.sample or sip.conf.sample or UPGRADE*.txt
17:10.34p3nguinIf it isn't, maybe someone will include it soon.
17:12.22ManxPower-worknot if he's running 1.4
17:13.48*** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net)
17:15.51*** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net)
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17:21.54bmoraca_workManxPower-work: it's been an issue since I started using Asterisk...i'd be surprised if it isn't in almost every single config file associated with queues and endpoints
17:24.24bmoraca_workhrm...at&t's increasing the cost of my measured rate line...$15/mo instead of $12/mo
17:24.34p3nguinThere's nothing about call-limit in queue.conf.
17:25.46bmoraca_workp3nguin: all i know is that it's one of the first issues I ran up against.  anything that requires the status of a peer doesn't work if call-limit isn't set.  i suspect that hints require it as well.
17:26.08p3nguinHints were the only thing I knew about.
17:26.23*** join/#asterisk albertoandrade (~albertoan@187.59.25.126)
17:33.36*** join/#asterisk xperia (~chatzilla@zux182-249.adsl.green.ch)
17:35.03xperiahello to all. i have a quick small question about asterisk. is it possible to provide telebilling with asterisk. i need the possibility to offer telebilling and search for information how to do this from the hardware and software side.
17:36.19*** join/#asterisk xperia (~chatzilla@zux182-249.adsl.green.ch)
17:37.15xperiaooops connection problem. was loged off. my question again. is it possible to make telebilling with asterisk ? if not what for possibility does exist ?
17:39.07*** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir)
17:39.28*** join/#asterisk avb (~avb@94stb14.codetel.net.do)
17:39.34avbhe all
17:39.35avbhey
17:39.58avbguys, is there is any easy way to initiate outgoing calls and bridge them after?
17:40.19avbcalls will be done via the sip
17:40.20p3nguinavb: originate from CLI
17:40.29avbfrom the extensions
17:40.50p3nguinokay
17:41.22avbwhat i figured out is only to create a meetme conference, and invite both of them
17:41.49avbbut that doesnt seems clean and easy solution
17:42.06avbas sip can bridge 2 channels by itself
17:42.25p3nguinoriginate SIP/yourphone extension 5551212@outgoing
17:42.29bmoraca_workxperia: it is.  you can use AGI or FUNC_ODBC depending on what other software you need to communicate with.
17:43.31avbp3nguin: sorry, havent got your, say i need to bridge 123123@myvoipprov and 242324@myvoipprov
17:43.37avbits kinda a callback
17:43.58crochatHi everybody ! Got a big problem here with Dahdi 2.2.1 and multiple BRI in DDI (EuroISDN)
17:44.17crochatMy hardware interface is a Junghanns QuadBRI (PCI)
17:44.43xperiabmoraca_work: woooww that is very impressive ! having the possibility to provide telebilling with asterisk is exactly what i need ! if it is really possible then big compliments to the programmers !
17:45.12crochatOnly the first BRI is "Provisioned, Up, Active". All the others are "Provisioned, In Alarm, Down, Active"
17:45.15bmoraca_workxperia: asterisk is simply a platform.  what you implement with it is your own deal.
17:46.32xperiabmorca_work: do you have any experience with telebilling ? i am asking how exactly i can get the money from the caller after he called my number normally. i am totally new to this stuff. need a litlle more info about this.
17:46.45crochatAll the BRI are connected ok. I tried to swap BRI (e. g. put BRI 2 on span 1 and it worked. Span1 was then up)
17:47.04KavanSthoughts....dahdi vs. zaptel...real reasons not just because "omg dahdi is new"
17:47.13KavanSfor t400p hardware
17:47.23KavanSright now, I use asterisk 1.4.x and zaptel and it "just works! TM"
17:47.30KavanSI'm apprehensive about switching...
17:48.05bmoraca_workKavanS: if it ain't broke, don't fix it.  that said, dahdi is very easy (easier than zaptel) to set up
17:48.10p3nguinIf you're using an older 1.4 version and zaptel works, why are we even discussing it?
17:48.28KavanSwell, I just wanted to see if there was any reasons to upgrade to zaptel I was unaware
17:48.29KavanSof...
17:48.41KavanSI suppose reading the changelog would be a step in the right direction, but sometimes just asking works...
17:48.41bmoraca_workthere probably is not
17:48.48KavanSok, roger that, thanks for the pro tips
17:49.07bmoraca_workon a new install, though, go dahdi
17:49.13drakoim about to install a new system with a AEX800 and im wondering if i should go zaptel or dahdi
17:50.01KavanSbmoraca_work, yep it's new hardware
17:50.19crochatKavanS: We had sooooo many crashes with BRIstuff... so Dahdi was the only solution for us
17:50.20KavanSI also have hylafax using the t400p for fax, that also "just works"
17:50.31bmoraca_workdrako & KavanS:  new install, go dahdi
17:50.38KavanSroger that
17:51.01bmoraca_workKavanS: nothing wrong with hylafax...i use it, too.  though i've heard that the fax functions in 1.6 work pretty well, I have not used them.
17:51.35KavanSahh right on....well my EU counterparts are using 1.6, I'm just not ready to take the dive yet I suppose
17:51.36crochatBut we had 3 BRI working with BRIstuff... with kernel crashes twice a month. And now, we have only one BRI working with Dahdi !!??
17:51.45drakobmoraca_work: should i go 1.4 or 1.6 ?
17:51.50KavanS1.4 works, and I have some "custom" dialplans....I just don't know how it'll play out in 1.6/1.8
17:51.58KavanSshould read some more I suppose
17:52.07bmoraca_workdrako: brand new install, no existing configs, go 1.6
17:52.18bmoraca_workKavanS: UPGRADE-*.txt will tell you any incompatibilities
17:52.21drakook
17:52.22p3nguin1.4 is considered long-time support, where 1.6 won't always be around.
17:52.50drakowe wont be always around here
17:54.49KavanSdrako, heh what?
17:57.08*** join/#asterisk Badrobot- (~badrobot@cpe-76-173-229-89.socal.res.rr.com)
17:59.46p3nguinAnyone here use TurboTax Home and Business?  I'm wondering if I need to purchase the 2009 version, or can I update the 2008 version and continue to use it this year.  Anyone know?
17:59.57*** part/#asterisk xperia (~chatzilla@zux182-249.adsl.green.ch)
18:00.22*** join/#asterisk QubeZ (~qube@64.128.254.34)
18:00.26QubeZhello all
18:01.13Qwellp3nguin: you buy the new version..  if you haven't tried the online version, you might consider doing so.  it's pretty great
18:02.54p3nguinI'll take a look at it.  If the online one is as easy as the installed one, there won't be any trouble with it.
18:03.07Qwellit is.  and it's online, so...heh
18:03.34p3nguinYeah, that'll keep me from having to use someone's Windows computer to do my taxes.
18:03.38Qwellnot having to worry about storing the data files is great
18:03.58Qwellthey store it all for you, and you can import previous years returns and such.  pretty useful
18:05.13tzafrir_laptopcrochat, basically Asterisk and libpri need to know to ignore that alarm and actively start the channel
18:05.47*** join/#asterisk drako (~luisjose@nelug/coreteam/luisjose)
18:06.14*** join/#asterisk Maximo (~maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com)
18:08.15spenguin[work]test
18:08.36bmoraca_workspenguin[work]: fail.
18:09.42spenguin[work]ohnoes
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18:12.25myrthfulManxPower-work: I've got a pastebin for the PlayDTMF problem I mentioned earlier: http://pastebin.com/d633b1478
18:13.07myrthfulOr, if anyone else can help me, I'm having trouble with PlayDTMF, it sends ok but I never hear the tones
18:13.18myrthfulAny help is appreciated
18:17.12abatistahello folks
18:18.03[TK]D-Fendermyrthful: What is that device?
18:20.02myrthful[TK]D-Fender: I'm not sure what you mean (warning, I'm an Asterisk newb)
18:20.25drakoI dont get why are 2 branch 1.6.1 and 1.6.2 , whats the diference?
18:20.29[TK]D-Fendermyrthful: Channel: SIP/emondpd-081bfd90 <------ what is this device?
18:20.41[TK]D-Fenderdrako: 0.0.1
18:21.17myrthful[TK]D-Fender: I'm connecting from linphone, is that what you mean?
18:21.18drakoerm
18:21.38[TK]D-Fendermyrthful: Could be that the phone doesn't generate a tone so you won't hear it.
18:22.38myrthful[TK]D-Fender: Hmm, I can send tones from the linphone command line.  I don't suppose that matters?
18:22.58[TK]D-Fendermyrthful: Correct.  it doesn't
18:23.20myrthful[TK]D-Fender: Ok, thanks
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18:25.23avbok
18:25.28avbmeetme seems working fine
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18:26.32*** join/#asterisk slima (slima@unaffiliated/slima)
18:29.36benngardshouldnt it?
18:30.07avbbenngard: u missed my question :)
18:30.19avbi was looking a way to make a callback
18:30.50avbthought that meetme will not be a good solution, but in the end its not that bad :)
18:31.43benngardto like originate from meetme to setup a call?
18:31.58benngardor rather 2 calls, it will ofc work
18:32.10Kattystretches
18:32.25Kattygoes in search of lunch
18:32.32avbbenngard: i made it easy. im generating to originate calls can sending them into an extension which invites them into the conference
18:32.44*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw)
18:32.51benngardgoes and search fro a beer!
18:32.51avbgenerating spool files*
18:33.10benngardeasy solutions are often the best ones
18:33.40avbyeh :)
18:33.40avbnow i need to figure out whats wrong with a signaling on the freaking nokia internet calls app
18:33.47avbasterisk ignoring hangups from it
18:35.26*** join/#asterisk Pimmetje (~Pimmetje@83.119.156.35)
18:35.36voipmonkavb whats the sip debug say?
18:35.54avbvoipmonk: havent worked that out yet
18:36.28*** join/#asterisk sier (~sier@unaffiliated/sier)
18:36.37avbpreparing a manual for a client :)
18:37.48Kattyreturns with shrimp
18:38.05*** join/#asterisk uqlev (~yuriy@91.184.221.31)
18:38.25voipmonkooh technical writing.... fun
18:38.27voipmonksnorez
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18:41.48spenguin[work]shrimp!
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18:44.55voipmonkwakes up....
18:44.57voipmonkwhere?
18:46.20spenguin[work]voipmonk: I smell it
18:48.36carrarpours alfredo sauce all over Katty
18:50.55Naikrovekvoipmonk: technical writing is interesting for me because, as i write, i think of what the reader would ask and often i wind up learning a ton about things i thought i knew
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18:58.31Kattycarrar: ^_-
19:00.16benngard"Lanab Design AB (fax)" <038075855> <- is that a correct clid, i am thinking about the "(" and the ")"?
19:00.30[TK]D-FenderNaikrovek: If you end up learning a lot of stuff.. then you clearly don't know what you're writing about :p  Go find someone qualified!
19:00.38Naikroveki say feh
19:00.50Naikroveki write documentation for software I write and I wind up learning things
19:01.06NaikrovekI wind up finding features I didn't know existed, etc
19:01.19Naikrovekand lots of bugs
19:01.45Naikrovekespecially when i write APIs
19:03.07*** join/#asterisk Wildy (~simba@178.176.93.29)
19:04.16Skeeter-anyone got access to work with Asterisk CRD, CSV files??
19:04.47Naikrovekaccess? as in MS Access?
19:05.02Skeeter-ya
19:05.09Naikrovekcan't say i have
19:05.22Skeeter-im open to any other program
19:05.47*** join/#asterisk oej (~olle@eduroam-193-157-113-13.uio.no)
19:05.47Skeeter-i just wanna print out some stats
19:06.27*** join/#asterisk oej_ (~olle@1x-193-157-196-250.uio.no)
19:06.44Naikroveki've always put that kind of thing into mysql but that probably isn't the best solution for everyone
19:07.16Skeeter-Naikrovek, I would love to do that but i dont know how mysql works
19:07.23Naikrovekyeah that's fine
19:07.24*** join/#asterisk Godfather_ (~Godfather@184.Red-83-58-87.dynamicIP.rima-tde.net)
19:07.34Naikrovekyou wanna do total call time or something like that
19:07.39Godfather_o/
19:07.43Naikrovekaccess may be the best tool
19:07.47ManxPower-workSkeeter-: MS Access should support importing CSV files.  I would recommend a spreadsheet instead.
19:07.50Naikrovekbut i don't have access so i can't really help
19:08.08Skeeter-ManxPower-work, i tried excel but Access is much faster for reports
19:08.25Skeeter-ManxPower-work, i get an import error while importing it
19:08.27Naikrovekif he wants to query that data access would be better, i'd think
19:08.31ManxPower-workSkeeter-: I avoid using MS products when I can so I really can't help you.
19:08.43Skeeter-ManxPower-work, its ok
19:08.54ManxPower-workI would not even us Windows if I didn't have work applications that require it.
19:08.55Naikrovekyou may ask in ##windows
19:09.09NaikrovekManxPower-work: that's the whole basis of Windows' adoption
19:09.15Skeeter-I just want to get how many calls where made to XXX destination
19:09.33NaikrovekSkeeter-: access is probably best
19:09.50Skeeter-I dont use Windows a lot, its on my Virtualbox, some dude knows access well, but it doesnt import anything
19:09.52ManxPower-workgrep "XXX" /var/log/asterisk/cdr-csv/Master.csv | wc
19:10.17Naikrovekthat's not a bad idea
19:10.29*** join/#asterisk lbarth (~lbarth@62.216.165.71)
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19:12.24Skeeter-<PROTECTED>
19:12.38Naikrovekwc -l will show you the number of lines (results)
19:13.03Skeeter-ok
19:13.14Skeeter-so that person received 2k calls right?
19:13.17Naikrovekfirst, just grep "XXX" Master.csv | less , and eyeball the output to make sure your grep is getting only what you want
19:13.39Naikrovekif it looks good, pipe it to wc -l rather than less
19:13.56*** join/#asterisk rare1980_ (~sniper_ja@203.175.76.219)
19:14.10rare1980_hello
19:14.19Skeeter-Naikrovek, works pretty well i guess
19:14.31rare1980_any one expert in asterisk dialplan
19:14.34Naikrovekit's not ideal probably, but for quick things grep is nice
19:14.34rare1980_?
19:14.41Naikrovekrare1980_: just ask your question, the experts will pipe up
19:14.49rare1980_sure
19:17.40hlueseahow i can voip termination in asterisk (installation packets and extensions)? and are anyone prefer a free billing solution for sip termination ? I probably want to make comming calls forward another provider and make billings on my side.
19:20.11rare1980_Naikrovek---- i need some help in asterisk dial plan... i hve miltple sip account has 3 channls and each account has limit to 300 mins call per week. i want 1st astiersk will use 3 sip chanls from one account and as soon as these 3 sip chanls are bz on call. then on 4th call astierks will use 2nd sip account for dialing. and it will use its 3 chanls and it will go like this..
19:20.50Qwellrare1980_: are these 300 minute per week accounts free outgoing "test" accounts, or something?
19:20.56*** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163)
19:21.05rare1980_no
19:21.07ManxPower-workrare1980_: you will end up spending weeks writing something that will do what you want.
19:21.36[TK]D-FenderManxPower-work: How bad?
19:21.48Qwell[TK]D-Fender: sip accounts limited to 300 minutes per week
19:21.50rare1980_is it diffcult?
19:22.09[TK]D-FenderQwell: Not THAt bad....
19:22.23Qwell...switching between the accounts once the limit is reached
19:22.44[TK]D-FenderQwell: You mean outbound load balancing?
19:22.45rare1980_2nd requirment= as soon as asterisk will use 300 mins from any account . then asterisk will not use that account for rest of the week
19:22.59Qwell[TK]D-Fender: if by "load" you mean "limited number of outbound minutes per week"
19:23.18[TK]D-FenderQwell: Yeah I mean rotating trough carriers not to get screwed on overages.
19:23.32ManxPower-work[TK]D-Fender: keeping track of how mins have been used per peer along with rollover between the peers when either the number of calls exceeds 3 for that peer or if the mins used that month exceed 300 mins for that peer.
19:24.05ManxPower-worksounds practically impossible for a newbie to setup in even a day.
19:24.10[TK]D-FenderManxPower-work: Yeah, a few hours to tractk the calls, etc
19:24.37ManxPower-workSomeone like you or me or anyone else good with dialplan would not find it too hard to do.
19:24.42*** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek)
19:24.59rare1980_soo
19:25.00ManxPower-workBut someone like you or me does not count every single penny spent on calls.
19:25.10rare1980_Manxpower_ can u do this?
19:25.40Naikrovekwhy not just get a provider that lets you make unlimited calls
19:25.44ManxPower-workrare1980_: Assuming you set up billing with the billing department and convince my boss to loan me out to you, then yes, I could do it.
19:26.02rare1980_ok
19:26.09ManxPower-workI don't know what they would set my hourly rate at.
19:26.11rare1980_and how much that billing would be?
19:26.22ManxPower-workrare1980_: I have no idea.  We are a CLEC not a consulting company.
19:26.42ManxPower-workI doubt you'd get my boss to agree to loaning me out.
19:26.44Naikrovektypical rate is $125/hr for this kind of thing, I think
19:26.50NaikrovekManxPower-work: moonlight
19:27.05ManxPower-workNaikrovek: I moonlight at my current job.
19:27.07rare1980_and how many hours would it take?
19:27.12rare1980_to do this work?
19:27.16Qwellwonders how many tenths of a cent would be saved by doing this
19:27.18Naikrovekwow you really need this
19:27.19rare1980_?
19:27.26ManxPower-workrare1980_: I do not know.  I suggest you look at other options.
19:27.33Naikrovekrare1980_: contact voipmonk
19:27.37Naikrovekhe can help you
19:27.48Naikrovekmaybe it's "they" now
19:27.55rare1980_ok
19:28.00rare1980_thanks guys
19:28.00Naikroveki've worked with voipmonk, he's good
19:28.03Naikrovekwelcome
19:28.07rare1980_for ur help
19:28.24Naikrovekthank voipmonk when he solves this for ya
19:28.33rare1980_:)
19:30.38Naikrovekrare1980_: he'll be back online soon; he just signed off half an hour ago
19:30.41Naikrovekhe'll be back
19:30.45carrarrare1980_, get writting, thats a simple agi script
19:31.06carrarYou will learn good stuff in the process
19:31.32[TK]D-Fendercarrar: Depending on how he intends on hoandling concurrency, timeouts, etc
19:31.54carrarwell a simple roll over based on time used with that link is pretty simple
19:31.55[TK]D-Fendercarrar: otherwise that brings in a live monitoring script and AMI all over the place
19:33.10carrarI have somthing similure for load balancing multiple sip peers based on usage
19:33.34[TK]D-Fendercarrar: concurrency adds a lot of mess
19:33.57carrarMessy is where the fun begins :)
19:35.45*** join/#asterisk atis_work (~atis_work@193.238.212.171)
19:36.58Skeeter-ManxPower-work, just found that Access give you a SQL table, but in the Master.csv there is no field
19:37.16Naikrovekthe fields are comman separated
19:37.21Naikroveks/comman/comma/
19:37.50carrarWhat is Access
19:37.56NaikrovekMS Access
19:37.59carrarWhats MS?
19:38.01Naikrovekeh
19:38.04Skeeter-rofl
19:38.08Skeeter-i luv haters
19:38.11carrar:)
19:38.18Skeeter-M$
19:38.34carrarI live about 3 bocks from the MS Bing building
19:38.34Naikrovek"M$" is the trademark of a douche
19:38.47Naikroveks/douche/blowhard/
19:39.23Skeeter-But Its Not Google
19:39.42Skeeter-wish i could make a graffiti of that on the building
19:39.51[TK]D-FenderSkeeter-: Either way, its a simple CSV
19:40.06benngardgot a feelingt that u guys dont like ms ;)
19:40.15Skeeter-[TK]D-Fender, im aware of that, and it is very accurate
19:40.23benngardjoins the party!
19:42.15Naikroveki like MS software, the company is getting better but still evil
19:42.51carrarMS Software has gone downhill
19:42.58carrarI don't miss it
19:43.00Naikrovekwrong
19:43.01Naikrovekwell
19:43.03Naikrovekyou may miss it
19:43.07Naikrovekbut it's gotten WAY better
19:43.15NaikrovekWAY WAY better
19:43.18fenrusWin7 semms rather nice
19:43.26Naikroveki'm talking server stuff
19:43.29Naikrovekbut yes win7 is nice
19:44.01ManxPower-workI guess that depends on what you call "way better".   XP does not run nearly as fast as Win2k on my laptop.
19:44.02*** join/#asterisk Poincare (~jefffnode@213.219.184.23)
19:44.10ManxPower-workI would call that "not better"
19:44.16NaikrovekXP doesn't run as fast as windows 7 on my laptop
19:44.27Naikrovekthat's what I would call better
19:44.36Naikrovekobviously YMMV but overall they're far, far better
19:44.38ManxPower-workJust remember "better" does not mean "changed the tool bar so it looks different"
19:44.55Naikrovekthat's not what i'm talking about
19:45.07ManxPower-workSkeeter-: Why don't you try OpenOffice.
19:45.15NaikrovekFAR easier to administrate now, I can script just about everything
19:45.23Naikrovekopenoffice is great on paper, useless in practice
19:45.39ManxPower-workI find each version of Microsoft products to be even more confusing than the previous version.
19:45.59[TK]D-FenderMS Office 2007 burned a lot of users
19:46.17Naikrovekwho can't learn new things, yes
19:46.20ManxPower-workI use Office 2003 on my work machine.
19:46.36ManxPower-workNaikrovek: you mean RElearn.
19:46.43Naikroveki can do more with O2007 than I ever did with O2003
19:46.53Naikrovekand I can do it all without the mouse
19:47.04ManxPower-workOther then Outlook randomly wanting me to reauthenticate, it doesn't seem to bog down my machine too much.
19:47.40bmoraca_worki freakin love Office 2007.  it's so much faster and easier to use than 2003.
19:47.46Naikrovekagreed
19:48.06Naikrovekthe keyboard shortcut mechanism introduced in 2007 is freaking stellar
19:48.06carrarIf you can't do it with VI you shouldn't be doing it
19:48.29ManxPower-workI wonder how well 2007 would run on my 1.3 Ghz laptop (single proc, no HT) in my work VMWare virtual machine.
19:48.40ManxPower-workcarrar: finally some sense around here.
19:48.47bmoraca_workNaikrovek: i like the ribbons, tbh
19:48.48*** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at)
19:48.56ManxPower-workbmoraca_work: pervert
19:49.04Naikrovekhah
19:49.10bmoraca_workManxPower-work: my wife runs it on a 1ghz Athlon XP with 768mb RAM without issue
19:49.23bmoraca_workManxPower-work: don't be a hater.  we all have our own tastes.  :P
19:49.54carrarand some have poor taste
19:49.55Naikrovekthat's his whole schtick
19:49.56ManxPower-workMS has done more than any other person or company to lower end user expectations of reliability on a PC.
19:50.07NaikrovekManxPower-work: then you haven't used anything recent
19:50.23ManxPower-workNaikrovek: why would I.  If they can't get right after 10 years.....
19:50.49ManxPower-workMuch like Grandstream, they simply have too much bad history.
19:50.59carraryup
19:51.14carrarThey have too much cruft
19:51.27Naikrovekno they don't
19:51.28Naikroveklisten
19:51.32Naikroveki'm no fan of microsoft
19:51.37Naikrovekbut their recent software is really good
19:51.46Naikrovekand i say that because it genuinely is
19:51.50Naikroveknot because i want to argue
19:51.54Naikrovekor because i know a guy who works there
19:52.04Naikroveki know because i use their stuff extensively
19:52.07carrar<Naikrovek> listen
19:52.07carrar<Naikrovek> i'm no fan of microsoft
19:52.08Naikrovekand have for 15 years
19:52.09carrarheh
19:52.10ManxPower-workStarting with Win2k, I consider their OS to be "reasonably stable"
19:52.36NaikrovekManxPower-work: well win7 is worth a look for you i think
19:52.50ManxPower-workNaikrovek: How much is it going to cost me?
19:52.57Naikrovekin fact, the only bluescreens i've had in XP, Vista, or 7 have been related to hardware
19:53.03NaikrovekManxPower-work: nothing to try it for 120 days
19:53.15ManxPower-workno, how much is it going to COST me?
19:53.23Naikrovekwell i'm talking money
19:53.28Naikrovekwhat are you talking about
19:53.28ManxPower-workSo am I.
19:53.34Naikrovek$0 for 120 days
19:53.41Naikrovekpast that you have to reinstall or buy
19:53.45ManxPower-workThe 120 days is just a hook.  That is not the cost of the OS.
19:53.46bmoraca_workless dollars out right than man hours trying to make linux behave like windows in an active directory scheme
19:53.58Naikroveklinux is cheaper only if your time is worth $0
19:54.04carrarManxPower-wor, you will also need to buy 4GB more memory and bigger/faster processors
19:54.14carrarthere is some real costs
19:54.21carrarand of course the license
19:54.22Naikrovekyou won't need 4gb of ram
19:54.24ManxPower-workNaikrovek: *nod*  So with Linux, I have to learn the OS.  With MS I have to learn the OS AND pay money.  I don't see the advantage to me.
19:54.35NaikrovekManxPower-work: okay
19:54.36carrarand all the other licenses if you want to have remote users
19:54.39ManxPower-workIs it true you can't disable all the Win7 eye candy?
19:54.43Naikrovekyou stay there while i experience progress.
19:54.49ManxPower-workNaikrovek: so how much does Win7 COST>
19:54.52NaikrovekManxPower-work: not true
19:54.54carrarI expirence process with UNIX
19:54.57carrarit's faster
19:54.57NaikrovekManxPower-work: for me: very little in time or money
19:55.02carrarprogress
19:55.13bmoraca_workManxPower-work: i paid $29 for my Windows 7 Professional upgrade license :P
19:55.21ManxPower-workI'm not spending $200 in a fscking OS.
19:55.28ManxPower-workbmoraca_work: upgrade from what?
19:55.31NaikrovekManxPower-work: i spend less administrating 200 years on windows server 2008 with active directory and exchange, than I did on 10 users on linux
19:55.35bmoraca_workXP
19:55.39carrarHow much was XP?
19:55.46bmoraca_workfree
19:55.50ManxPower-workNaikrovek: and in my experience it is exactly the opposite.
19:55.55carrarXP is not free
19:55.59bmoraca_workit was for me
19:56.04bmoraca_workacademic license ftw
19:56.10Naikrovekonly fools pay MSRP
19:56.25carrars/MSRP/MicroSfot/
19:56.29ManxPower-workbmoraca_work: you must be an MSDN subscriber or you are not counting the cost of the OS being built into the cost of the PC.
19:56.37bmoraca_workManxPower-work: there are a LOT of applications that do not work on linux that a lot of businesses need.
19:56.45bmoraca_workManxPower-work: nope.  both were academic discounts
19:57.13Naikrovekmy employer would be out of business with linux.  how much would linux cost us?  everything, that's how much.
19:57.15carrarlike?
19:57.24ManxPower-workbmoraca_work: That happens when you are a convicted monopolist.
19:57.32bmoraca_workManxPower-work: for example, dental offices uses practice management and digital radiography packages that do not work and cannot work on Linux
19:58.03bmoraca_workManxPower-work: i'm not talking about potential alternatives like openoffice (which is a piece of garbage).  i'm talking about specialized software that DOES NOT RUN on anything but Windows.
19:58.06ManxPower-workbmoraca_work: you mean because they accepted the MS lockin when they went with their software.
19:58.15carrarhe
19:58.16carrarh
19:58.18bmoraca_workManxPower-work: no, because there is no alternative
19:58.34ManxPower-workbmoraca_work: and in those cases the choice to move away from MS is more expensive than the end user is willing to pay.
19:58.37Naikrovekand it's not because MS has a lock, it's because there's no one else who can make anything better
19:58.40carrarCould have been just as easy to write it on unix
19:58.45ManxPower-workIt sucks, but they made their bed, they can sleep in it.
19:58.46Naikrovekcarrar: lol
19:58.47carrarif not easier
19:58.51Naikrovekeasier? wtf
19:58.52Naikrovekhahaah
19:58.57carraropen standards
19:59.01bmoraca_workManxPower-work: no, the choice to move away from MS does not exist because there are no alternative software packages.
19:59.21Naikrovekopen standards don't make it easier.
19:59.24Naikrovekor better
19:59.27ManxPower-workbmoraca_work: they can hire someone to write one.  As I said, it would be too expensive for the customer.
19:59.42ManxPower-workBut there is ALWAYS a choice.
19:59.55Naikrovekyes always a choice
19:59.59Naikrovekgo out of business or use windows
20:00.03bmoraca_worknot a rational or logical choice, but I suppose.
20:00.32ManxPower-workI have a choice of quitting my job and going to work for a company that does not use Windows.   That choice is too much problem and effort for me to choose that option.  I still have a choice.
20:00.39Naikrovekand let's not even get started on how shitty linux desktops are
20:00.45Naikrovekomg what pile those things
20:00.47Naikrovekare
20:00.51bmoraca_workpoint being:  linux is absolutely NOT an option in every case.  particularly at the workstation level.  my god, I'd hate to have to support an office full of morons using linux workstations.  i can't even begin to imagine that bullshit.
20:01.03Naikrovekany system that has to read from disk when i start typing in a notepad application is FLAWED
20:01.18ManxPower-workbmoraca_work: we are looking at moving some of our sales people to Linux workstations to cut down on support issues.
20:01.34ManxPower-workIf you don't have permission to install crap support issues drop.
20:01.37bmoraca_workManxPower-work: that's ass-backwards logic if i've ever heard of it.
20:01.51carrarnope
20:01.59Naikrovekyup
20:02.02carrarMS is ass-backwards logic
20:02.03ManxPower-workOnce we get over the INITIAL training costs, of course.
20:02.30ManxPower-workNo more removing spyware, no more removing viruses no more messing with the registry
20:02.31Naikrovek"howcome my fonts are unreadable"
20:02.35bmoraca_workManxPower-work: get system administrators worth their salt, and you won't have to worry about permissions issues and people installing bullshit apps on their systems
20:02.52Naikrovek"howcome my tab key doesn't switch input fields?"
20:02.53ManxPower-workbmoraca_work: it is cheaper to retrain the existing admin.
20:03.03ManxPower-workNaikrovek: I've never had either of those issues.
20:03.23NaikrovekNEVER experienced a font issue?
20:03.34Naikroveki'm almost convinced that you don't use a computer :)
20:03.39ManxPower-workOh!  Yes, I did have an issue with crappy fonts for about a day until I ran yum update
20:03.42bmoraca_workManxPower-work: i'm not going to argue that linux has no uses, because I know it does.  it just doesn't have very MANY uses where the majority of computer users are concerned.
20:04.02carrarhaha
20:04.07ManxPower-workbmoraca_work: Of course Linux has issues.  Some pretty significant ones on the desktop.  So does Microsoft.
20:04.09carrarignorance is bliss
20:04.27[TK]D-Fenderbmoraca_work: As a home user who doesn't need special tax software, doesn't do PC gaming, Linux suits me jsut fine.
20:04.32Naikrovekjust take a look at Germany's governmental rollout of linux for exery reason in the world and some new ones you've never heard of, of why linux on the desktop is a terrible idea
20:04.44Naikrovekand that's not anecdotal, that's emprical evidence
20:05.15bmoraca_workcarrar: i'm still waiting for your linux practice management software.  until you can point me to it, you're absolutely full of shit, because i have litterally thousands of customers who ABSOLUTELY MUST use this software which ONLY exists for Windows.
20:05.42Naikrovektechnically, linux could be awesome
20:05.46*** join/#asterisk ELSEGO (~Juankar@unaffiliated/elsego)
20:05.51Naikroveki used to run several linux desktops
20:05.51ecraneNaikrovek: That's not fair. I know nothing about the German government, but deciding something is bad because a government tried to use it and failed is not a fair comparison.. governments fail at a lot of things they try, sometimes even good things...
20:05.54Naikrovekbut i got tired of compiling
20:06.06carrarWhat is practice management software
20:06.07bmoraca_workanyway, i'm going to lunch.  ideological arguments set my teeth on edge.
20:06.08Naikrovekecrane: they didn't fail, they're still rolling out
20:06.16bmoraca_workcarrar: now who's ignorant?
20:06.18Naikrovekecrane: just read up on all the trouble they're having with user training
20:06.21carrarplease explain
20:06.33carrarI'm not familure with "practice management software"
20:06.36Naikrovekecrane: and system deployment issues.
20:06.44Naikrovekcarrar: medical practice management software.
20:06.48bmoraca_workcarrar: you have no idea about what you are talking, so you cannot begin to claim that "linux works for everyone".
20:07.11carrarYou're right, I've wasted the last 20 years of my life making money with Linux
20:07.15ManxPower-workcarrar: claimed that :linux works for everyone: ?
20:07.22carrarstupid me
20:07.32carrarhahah
20:07.35carrarYou kids are funny
20:07.49ManxPower-workcarrar: about as stupid as saying the same thing about Microsoft.
20:07.50Naikrovekit's always the old guys that are anti microsoft i'm noticing
20:08.15carrarwith age comes wisdom my child
20:08.21benngardis pretty old
20:08.29Naikrovekwell youth has the ability to see one's errors
20:08.33Naikrovekold = stodgy
20:08.57Naikrovekand it's provably easier to learn new stuff as a youth
20:08.58carrarand recklessness comes with youth
20:09.11Naikrovekwisdom comes with youth.  it's just ignored until you can't rely on anything else
20:09.17carrarhahah
20:09.29*** join/#asterisk ruben23 (~AGENT@122.55.48.243)
20:10.49benngardbut, can people stop sending fax through MY *, i wanna restart it :(
20:10.54carrarbut alas I digress, I'm not a hater, I just have some time to kill because all my Linux boxes run themselves
20:11.05Naikrovekwell that's fine
20:11.13Naikrovekmy windows boxes run themselves, too
20:11.27Naikrovekit was a pointless discussion anyway
20:11.27carrarSECURITY UPDATE TODAY
20:11.38Naikroveknot a single one of us had our opinion changed
20:11.40spenguin[w0rk]?
20:11.47carrarmust reboot all windows boxes
20:11.53carrarheh
20:11.57ecraneNaikrovek: I searched google and bing but I am having trouble finding info about the problems the german government has. Do you have any links?
20:12.03Naikrovekcarrar: tuesdays.  and they reboot themselves.  and we don't have to recompile a damned kernel to implement the fix
20:12.08Naikroveknor any kernel modules
20:12.27carrarcause you couldn't if you wanted too
20:12.40Naikrovekcarrar: don't *ever* need to
20:12.43spenguin[w0rk]recompiling kernel is fun and 3l33t
20:12.47spenguin[w0rk]unlink
20:12.54spenguin[w0rk]unlike*, click click click
20:12.56spenguin[w0rk]:p
20:13.00*** part/#asterisk cobolfoo (~cobolfoo@bas7-quebec14-1096763909.dsl.bell.ca)
20:13.18Naikrovekecrane: nothing that isn't in German and provided by a german citizen friend.  let me dig something up though
20:13.33*** join/#asterisk fibres (~no@cpc2-nfds1-0-0-cust1021.lei3.cable.ntl.com)
20:14.24ecraneNaikrovek: Thanks! I have no doubt you are correct; IT projects rarely go smoothly no matter what OS is involved.. I'm just interested in the details.
20:17.28*** join/#asterisk gelpg (~chatzilla@qf.invitel.hu)
20:19.26Naikrovekyou linux guys should read the Linux Hater's blog for ideas about things to fix in Linux.  he makes some points that no educated linux user can honestly deny
20:20.04ManxPower-workNaikrovek: And us linux users don't have to reboot every tuesday.  I finally stopped automatically installing updates when I finally got tired that.
20:20.17NaikrovekManxPower-work: windows users don't HAVE to either
20:20.26Naikrovekmost of the fixes are ignorable
20:20.49Naikrovekand windows sites of any size have a local windows update services server so we can control the updates as we please
20:20.52Kobazhow hard is it to replace a piston in a 2 cycle short block engine
20:21.03ManxPower-workI have it notify me when new updates are available.
20:21.33NaikrovekKobaz: that's a random question.  i did it when i was a kid, don't remember it being too hard
20:22.14KobazNaikrovek: k
20:22.31Kobazmy snow blower sucked in a screw from the throttle plate in the carbutator\
20:22.38Naikrovekooh
20:22.48Kobazi just finished rebuilding the carb and it was running great for about 5 minutes
20:22.53Naikrovekwhatever it was I worked on had removable sleeves
20:22.55Kobazthe cylinder looks okay
20:23.04Kobazthe piston has several nicks in it
20:23.19Naikrovekso i pulled the head off, pulled the cylinder off the piston and replaced ... something.  it's been a long time
20:23.48Kobazit's like 160 for a new engine.... 30 bucks for a new piston
20:24.08Kobazalthough 160 is pretty cheap compared to the 300 i spent on a new table saw motor
20:24.33Kobazcompression is at 60psi
20:25.44Kobazi wonder if a new piston will put it back at 90
20:26.32Naikrovekother than dents in the surface is the piston okay?
20:26.43Kobazit seems to be
20:26.54Naikroveki would imagine there are probably some grooves in the wall, yes?
20:27.00Kobazi didn't see any
20:27.08Naikrovekwhy you replacing the piston then
20:27.30Kobazwell the piston looks guaged in some spots
20:27.45Naikrovekgouged ?
20:27.53Kobazyeap
20:27.54Kobazthat too
20:27.55Naikroveki suppose it might burn through any thin spots
20:27.59Naikrovekthat would be bad
20:28.09Naikrovekpiston only or connecting rod too
20:28.17Kobazhaven't looked at the rod
20:28.27Naikrovekwell if you replace only the piston you gotta remove it from the rod
20:28.30Kobazi think the screw bounced around in the cylinder and got thrown out the exhaust
20:28.31Naikrovekthose bearings are press-fit
20:28.38Naikrovekooh
20:28.44Naikrovekyou may have a bent valve
20:28.53Naikrovekbut if you can hold 60psi probably not
20:29.31ecraneThis linux hater blog guy has a lot of hate in him.
20:29.36Kobazgood thing for a small intake... it tried to suck in the throttle plate too
20:29.43Naikrovekit's not difficult to take those small engines apart
20:29.47Kobazyeah
20:29.49Naikrovekecrane: and a lot of really good points
20:29.52Kobaz6 screws and i had the head off
20:30.35Kobazi'll have to do more inspection to see if there's any other damange
20:34.58*** join/#asterisk fibres (~no@cpc2-nfds1-0-0-cust1021.lei3.cable.ntl.com)
20:39.21Naikrovekecrane: the guy clearly uses linux daily, and has been holding back criticism for some time
20:40.11Naikrovekdoes linux have any Windows Deployement Services-like functionality?
20:40.20NaikrovekWe had it on FreeBSD at Yahoo!
20:40.39Naikrovekplug a machine in, netboot, enter a command and 5 minutes later the server is ready to be deployed, all required software installed
20:40.44ManxPower-workMost distros support zeroconf as well as automated installation
20:40.49Naikroveknice
20:41.11ManxPower-workAll our installs are from a custom CentOS DVD.
20:41.26ManxPower-work(actually CD, not DVD)
20:41.28*** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr)
20:41.43ManxPower-workNaikrovek: Redhat based distros call it "kickstart"
20:42.08Naikrovekmy windows installs are all custom, done from a distribution server.  netboot, install, shutdown, carry to users desk, turn on, done.
20:42.17Naikrovekit's very handy; glad to see that it's in linux as well
20:42.22Naikrovekit was the bees knees when i used it at yahoo
20:42.43Naikrovekmachine downtime due to hardware failure was less than 10 minutes even if i had to open a new server from the box it was shipped in
20:44.08*** join/#asterisk Circlefusion (~circlefus@74-132-116-73.dhcp.insightbb.com)
20:45.19ecraneHow is licensing for windows managed? Can/do all the images made from the distribution server have the same product keys?
20:46.39*** join/#asterisk Tech_Travis (~tech_trav@208.179.137.131)
20:48.08bmoraca_workecrane: depends.  windows volume licensing is almost all honor-system.
20:50.27*** join/#asterisk brezular (~brezular@adsl-d237.84-47-127.t-com.sk)
20:53.35ecraneI guess that's one argument against windows.... I'm making assumptions here, but I suspect volume licensing is out-of-reach for the average person... and I suspect that it is a hurdle in computer imaging. I remember at one job the admin had to do some kind of windows registry change on each machine afterwards, making up a unique id...maybe it's gotten better since then...
20:54.53Naikrovekecrane: anyone can get a VLK, even if they buy a single license
20:55.08Naikrovekbut yes, the images distributed are predefined with their key
20:55.13Naikrovekand preactivated
20:55.22Naikrovek... maybe not preactivated
20:55.23Naikrovekactually
20:55.29Naikrovekbeen a while since i deployed a desktop
20:55.33Naikrovekoh that's right
20:55.39bmoraca_workecrane: he was doing it wrong, then.
20:55.40Naikroveknot preactivated, but they activate on boot
20:55.53Naikrovekhave a local activation "server" here
20:55.59Naikrovekwhat's it called ... mlk?
20:56.03Naikrovekforget
20:56.04Naikrovekdang
20:56.30Naikrovekecrane: he was using ghost or something if he had to change crap in the registry
20:56.32bmoraca_workecrane: a desktop volume license key can be used on as many systems as you want.  however, each system will need its own system ID, etc.  the easiest way to do that is to "sysprep" the image before you take it.  it's not hard and it works really well.
20:57.23bmoraca_workecrane: volume licensing is more expensive than regular licensing because the restrictions on it are far less.  volume licensing really only makes sense on a large scale, as do most of these deployment and update services.
20:57.26Naikrovekoh man what is that server process called that activates your machines for you
20:57.42Naikrovekbmoraca_work: yes
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20:57.58bmoraca_workNaikrovek: that's new in 2008.  i don't remember what it's called, but 2003/XP VLK did not need to be activated
20:58.11Naikrovekbut we're a small office, just a few servers and maybe 25 desktops, but i have WSUS, WDS and whatever that activation thing is now
20:58.33Naikroveki don't remember because i didn't have to install it.  if you activate a single windows server with that key, it BECOMES a local activation server
20:58.53Naikrovekbut from there on you don't have to know what your product keys are, they're all activated whether you're licensed or not
20:58.59Naikroveki'm licensed, of course
20:59.28hardwireok.. using asterisk 1.2 I have 4 DID registered with another provider and 4 SIP accounts per DID however whenever a call comes in they always appear to be the last mentioned DID as the SRC channel.
20:59.30bmoraca_workNaikrovek: yes, that's one of the ways they're doing VLK for 2008.  It's KMS, I believe.  you can also convert your KMS keys into MAK keys that allow you to activate to Microsoft instead of a local activation server
20:59.38Naikrovekthat'
20:59.39hardwireeven though the pcap says differently.
20:59.41Naikrovekthat's it, KMS
20:59.42hardwireany idea how to get around that?
21:01.56funtoo_nbuanyone know why there might be delays when you transfer a call?
21:02.50bmoraca_workfuntoo_nbu: almost exclusively an issue with the phone's configuration.
21:03.27funtoo_nbuthe phone's config itself or its conf inside extensions.conf or sip.conf ?
21:04.01bmoraca_workthe phone's configuration itself.  if i'd meant extensions.conf, I'd have said "dialplan" or sip.conf would ahve been "peer"
21:04.09funtoo_nburoger that
21:04.12funtoo_nbuany hints?
21:04.17funtoo_nbuthey are polycom 450s
21:04.33Naikrovekwhat kind of delay are you talking about
21:04.39Naikrovek3 seconds?
21:04.43bmoraca_worksip debugs and dialplan logs that corroborate your delay would be useful
21:04.45funtoo_nbuit seems random
21:04.54funtoo_nbusometimes none sometimes > 20s
21:04.57Naikrovekwow
21:05.03Naikroveknot a digitmap timeout issue then
21:05.08funtoo_nbulike, caller calls in
21:05.33funtoo_nbuphone a answers, says ok ill transfer, hits transfer button (hold music plays)
21:05.54funtoo_nbuphone a is calling phone b, phone b answers, hold music stops playin for a bit then phone b answers
21:06.08spenguin[w0rk]whats in the console?
21:06.08Naikrovekwhere is the lag in that scenario
21:06.47funtoo_nbunot on console yet, im gona call in and try to replicate
21:08.57*** join/#asterisk Tobarja (~chatzilla@user-0c8h5rb.cable.mindspring.com)
21:09.06Naikrovekmy receptionist had a similar problem when she transferred
21:09.30Naikrovekshe'd be on a call, press transfer, then dial the extension, and then the digitmap timeout would make her wait 5sec before it would dial
21:09.52Naikroveki taught her to push # after the extension number and it dials when she pushes it now
21:09.59Naikrovekit would have before if she'd pushed it
21:10.09Naikrovekbut yea that was the problem i had here
21:10.12Naikrovekwas user problem
21:10.15Naikroveknot system problem
21:11.06Naikrovekif the extensions started with something other than a 1 i would be able to configure the digitmap entry to not timeout but it does, so i have to allow someone to dial 1 area code etc
21:11.38funtoo_nbumy delay is after the other transferring phone picks up
21:11.51Naikrovekreally
21:11.52funtoo_nbuim trying to get them to pick up my call so i can get some log on it :D
21:12.04funtoo_nbuo shit, its lunch :/
21:12.14Naikrovekeeeeeat
21:12.20funtoo_nbumaybe ill eat too
21:15.40ManxPower-workNaikrovek: can you have them never dial a 1 for outside line?  That way first digit = 1 = extension, first digit != 1 = outside call.  Seems backwards but it might work.
21:16.04*** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-170.cablep.bezeqint.net)
21:16.08Naikrovekwell my intention is to switch to 4 digit extensions that don't start with a 1 soon so the problem will go away then
21:16.32funtoo_nbuhey thats a good idea
21:16.34KavanStrying to compile 1.4.29 from source and it complains about iLBC codec....1.4.28 compiles fine
21:17.50bmoraca_workKavanS: iLBC source was removed from SVN.  you need to provide it separately or deselect iLBC from menuselect
21:19.17funtoo_nbuif the patten matching is done right can you make it so if you dial 101 it knows its an extension and dial it right away (without having to click send)?
21:19.34Naikrovekyes
21:20.08Naikroveknot all my extensions start with 1
21:20.25KavanSbmoraca_work, roger that, thanks for the tip
21:21.11ManxPower-workCan anyone think of a reason that 'asterisk -rvvv' works, but 'asterisk -rx "sip show peers' gives me: Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
21:22.14kukuspoke too soon: http://pastebin.com/d5006db0f
21:23.57funtoo_nbumm my patten for outgoing is _1NXXNXXXXXX
21:26.11Naikrovek_?
21:27.08funtoo_nbuan extension name is a pattern if it starts with the underscore symbol (_).
21:29.03carrarManxPower-work, Might i suggest you install win7 to fix that issue
21:29.35[TK]D-Fendercheckout time, bbiab
21:29.50*** join/#asterisk oej (~olle@1x-193-157-196-250.uio.no)
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21:34.01*** join/#asterisk etfonhomey (~etfonhome@74-143-192-74.static.insightbb.com)
21:34.36etfonhomeyIs this the error message you get if you don't have the g729 transcoding license?      [Feb 11 16:32:12] WARNING[3139]: chan_sip.c:3724 sip_write: Asked to transmit frame type 4, while native formats is 0x100 (g729)(256) read/write = 0x100 (g729)(256)/0x100 (g729)(256)
21:34.42etfonhomeyFrame type 4 is g711u
21:36.15*** join/#asterisk nny (~scott@64.203.239.83)
21:36.23nnyhmmph
21:36.35KobazNaikrovek: okay so... upon further inspection. the cylinder is perfect as far as i can tell
21:36.48nnyso these cisco 504g phones have the ability for programmable softkeys! Hooray.. ermm wait
21:36.56KobazNaikrovek: there is a 1mm gouge on the side of the piston where it hit the screw
21:37.10KobazNaikrovek: and i found the screw.. it was in the crank arm housing
21:37.17NaikrovekKobaz: think it leaks compression?
21:37.18Naikrovekwhoa what
21:37.21Naikrovekin the crank case?
21:37.29Kobazit got below the piston somehow
21:37.30Naikrovekhow the F did it get in there
21:37.40nnyyou can set the PSK for say, connected to be different (in this case, I am setting it up as a park button, *9). Sadly the stupid phone tries to dial *9 as a fresh dial, instead of overlayed onto the existing call :\
21:38.00KobazNaikrovek: i think it most certainly leaks compression... it's quite a bit of a nick
21:38.02nnyer rather you can set the PSKs based on different call states
21:38.46nnygonna ping the cisco engineers (you'd think it's be trivial) wonder if I can fashion a workaround through * though
21:38.51Kobazthat's the only thing i can find that would affect the compression... i think replacing the piston would do it...but there's a very very strange bolt holding the arm to the drive shaft.. it's like inverse star-drive
21:38.52Naikrovekwell disconnect the connecting rod from the crankshaft, and take the connecting rod and piston to a local small engine place.  they'll pull off the piston and order you a new one
21:38.52nnyit'd be*
21:40.14Kobazi think i found the piston i need online for like 30 bucks
21:40.24nnyhmm a 1mm gouge shouldn't leak compression though, the oil should fill that gap no?
21:40.32Kobazthe problem is getting the piston out.. i'll take a picture of the bolt
21:40.45nnyguess the only way to know is to put the head on and test the compression
21:40.50Kobazit's at 60psi
21:40.54nnyahh
21:41.00nnyyeah, that's low lol
21:41.04Kobaza bit
21:41.08nny:D
21:41.33Kobazmaybe it's bigger than 1mm. it's quite a gouge, it's not big, but it's not small
21:41.51nnyyeah I imagine so, someone drop a bit of some love down the intake
21:41.52nny?
21:41.59Kobazthat didn't work
21:42.20nnyer rather, my question was did someone drop something in the intake to cause the damage?
21:42.26Kobazoh
21:42.27Kobazyeah
21:42.38Kobazthe screw from the throttle plate got sucked in while running
21:42.57Kobazand the throttle plate has a big chunk missing from it
21:43.01nnybeen there before, but it was the wingnut for the aftermarket air filter on my merc
21:43.22Kobaza friend of mine got a brand new nissan sentra, and it sucked in a screw
21:43.36nnyouch
21:43.39Kobazcompletely wrecked the engine.. so they warrentyd it
21:43.45nnynice
21:44.06Kobazthe guy at the shop was like... did you change the oil?  and my friend said uhh... it's got 1500 miles on it.. no... that's not why the engine died
21:44.30*** part/#asterisk Tobarja (~chatzilla@user-0c8h5rb.cable.mindspring.com)
21:44.50Kobazanyways... so i mean like... with a smoother than a baby cylinder, and nicks in the piston
21:45.18*** join/#asterisk momelod (~smelo@CPE00a065c98ce6-CM0012c91df0bc.cpe.net.cable.rogers.com)
21:45.22momelodgreetings channel
21:45.24Kobazwell, the missing chunk of the throttle plate would probably affect compression too
21:45.29nnyrephrasing my issue/ quandry here. These phones have the ability to dial *something* during a call. I tried some feature codes, but the dumb things dial the extension as a fresh line, or rather don't seem to try and treat it like you would a DTMF style entry. Any clever advice?
21:45.38Kobazthat's like a 3mm missing chunk
21:45.49nnythe phones have the ability to speed dial something during a call*
21:45.54nnyif I use the handset it works fine
21:45.56momelodI don't know a better place to ask so sorry if this is the wrong place.  What is the best sip conference phone for use w/ asterisk?
21:46.01nnygah handset||keypad
21:46.14nnymomelod had a lot of luck with the polycomms
21:46.30nnyip9000 or something rather one sec
21:46.35momelodnny any particular model u liked?
21:46.36Kobazmomelod: sip phone? huh?  this is #engine_repair
21:46.40nnylol
21:46.48momelod:P
21:47.16nnymomelod: http://www.polycom.com/products/voice/conferencing_solutions/conference_phones/soundstation/soundstation_ip7000.html
21:47.36Naikrovekyeah ip7000 is good
21:47.46nnymomelod: seen that phone rebranded and sold by other vendors (mitel for example)
21:48.31Naikrovekcisco rebrands a couple of polycom phones
21:48.38momelodawesome thanks
21:48.41Naikrovekand yes i confirmed that they are rebranded
21:49.12nnyyeah heh
21:53.47momelodouch, not cheap
21:56.39*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194)
21:57.13bmoraca_workIP6000 is good, too...no real reason to go to IP7000 if you don't need all the extra mics and stuff
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22:05.03p3nguinnaikrovek: Which models?  I know they used to rebrand some devices, but it was not something hidden or even attempted to be kept secret.
22:07.39*** join/#asterisk mediaprodigy (~chatzilla@72.20.157.179)
22:07.43mediaprodigyhello
22:07.56*** join/#asterisk TSM2 (~the_softw@87-194-32-212.bethere.co.uk)
22:08.44bmoraca_workp3nguin: Cisco 79x0s and pretty much everything older than that were rebranded from another company (don't remember the name right now)
22:09.45jaskewCan anyone recommend a 'favorite' appliance-style box for a small (say 5 SIP phones and two analog trunks) install.  I've heard of PIKA Warp.  Any thoughts?
22:09.50mediaprodigyQuestion: Is there an open source project already available that is similar to OCS cisco cups.
22:09.59mediaprodigyQuestion that has these features
22:10.03mediaprodigy• Complete desktop call control
22:10.05mediaprodigy• Incoming and outbound interactive call windows with name and number
22:10.06mediaprodigy• Multiple call notification
22:10.08mediaprodigy• Database directory for system and personal numbers
22:10.10mediaprodigy• Dynamic redial list with most frequently called or received numbers
22:10.11mediaprodigy• Speed dial for one click dialing
22:10.13mediaprodigy• Busy lamp field indicators for internal users
22:10.14mediaprodigy• Peek-a-Boo
22:10.16mediaprodigy• Silent Message
22:10.17mediaprodigy• Absence Message
22:10.19mediaprodigy• Call Reminders
22:10.20mediaprodigy• Email Page
22:10.22mediaprodigy• Call Log with dial capability
22:10.23mediaprodigy• PIM integration to popular databases with optional Power Dialer module
22:10.25mediaprodigysorry
22:10.31mediaprodigythought i had put in comma's
22:10.43ManxPower-workYou must now do 300 hours of community service.
22:10.54nnywouldn't fop2 handle some of that?
22:10.57jaskewor Peek-a-Boo with inner city youth
22:11.05etfonhomeyjaskew, I used this with a Sangoma B600DE:  http://www.newegg.com/Product/Product.aspx?Item=N82E16816101262&cm_re=SuperMicro_atom-_-16-101-262-_-Product
22:11.20mediaprodigyi do not know what peeka-boo is
22:11.23mediaprodigybut you get the idea..
22:11.33mediaprodigyis there a project out there that is already doing this.
22:12.10bmoraca_workmediaprodigy: i don't know of an opensource one, but iSymphony is very good for providing those types of features, and if you have 5 or fewer users, it's free.
22:12.12carrarmediaprodigy, are you on a windows box?
22:12.28bmoraca_workmediaprodigy: additionally, iSymphony does not require windows.
22:12.44*** join/#asterisk andres833 (~andres833@190.144.75.22)
22:13.16jaskewetphonehomey:  Thanks.  any others I can/should look at.  I assume Flash based stuff is out if it has Voicemail.
22:13.37etfonhomeyManxPower-work, Is this the message you get if you don't have the g729 license?      [Feb 11 16:32:12] WARNING[3139]: chan_sip.c:3724 sip_write: Asked to transmit frame type 4, while native formats is 0x100 (g729)(256) read/write = 0x100 (g729)(256)/0x100 (g729)(256)
22:13.58[TK]D-Fendermediaprodigy: http://pastebin.com/m5e107a20
22:14.40carrarheh
22:14.51etfonhomeyjaskew, that's the only small install one I've tried other than the circa 2002 ThinkPad I run my person Asterisk install on.  Builtin battery backup!  If the battery is still good...
22:18.14bmoraca_workjaskew: http://www.newegg.com/Product/Product.aspx?Item=N82E16816101262&cm_re=intel_atom-_-16-101-262-_-Product
22:18.24*** join/#asterisk mediaprodigy (~chatzilla@72.20.157.179)
22:18.38mediaprodigyFirebox crashed
22:18.49mediaprodigyso someone mentioned iSymphony
22:18.49carrarthat red thing?
22:19.02bmoraca_workmediaprodigy: i don't know of an opensource one, but iSymphony is very good for providing those types of features, and if you have 5 or fewer users, it's free.
22:19.13mediaprodigythe reason why i ask
22:19.37mediaprodigywell first off would this be something that the telephony asterisk group would even want
22:19.42mediaprodigyor find interst in
22:19.51KavanShrm, can asterisk 1.4.29 use zaptel?
22:19.56KavanSlooks like it's only dahdi...
22:20.23bmoraca_workmediaprodigy: i'm not sure what you're asking.  if you have a specific question, let's hear it.
22:20.27mediaprodigyi am trying to convince someone to release their company project to open source..
22:20.28[TK]D-Fendermediaprodigy: "the telephony asterisk group" <- Pardon?  Who/what is this?
22:20.31bmoraca_workKavanS: i believe that's correct, yes.
22:20.37carrarTAG
22:20.39carraryour it
22:20.47mediaprodigythe people in this channel
22:20.48KavanSok roger that, thank you
22:21.07carrar[TK]D-Fender, you don't have the jacket?
22:21.09[TK]D-Fendermediaprodigy: You mean would WE want a 3rd party program that offers this functionality to * released as OSS?
22:21.13bmoraca_workmediaprodigy: i doubt that someone is going to take a useful, profitable app and give it away for free just because you asked
22:21.29carrarI've got the "Telephony Asterisk Group" patch
22:21.55[TK]D-Fendercarrar: I tried that patch... but the cravings returned
22:21.59carrarhaha
22:22.47mediaprodigybmoraca_work: there are several major companies releasing software that will over take this software.. instead of letting it die out over the next few years.. what if that project was taken the open source route
22:23.31mediaprodigyI understand your perspective.. i am just here to get a feel for the waters.
22:23.34jaskewbmoraca_work & etfonhomey:  thank yo u - two votes for that one - must be the ticket!
22:24.07carrarmediaprodigy, when do you start working on it/
22:24.08carrar?
22:24.13*** join/#asterisk brettnem (~brett@user-0vvd88f.cable.mindspring.com)
22:24.16bmoraca_workmediaprodigy: it's not up to us whether a developer takes their product opensource or closed source...it's up to the developer.  what we want is completely and totally irrelevant
22:24.22brettnemhello all
22:24.45mediaprodigybmoraca_work: i do not know you but you are obviously missing my point
22:24.50*** join/#asterisk fofware (~chatzilla@190.7.25.160)
22:24.57nnypersonally I am still trying to find *soemthing* to offer my clients that compares to HUD or similar. I use FOP2 right now + openfire and spark (rebranded yay!) however I have found some aspects of the system (spark asterisk plugin) aren't maintained. Short answer: yes it would be good to find a solution, i am going to ttry isymphony again
22:25.01bmoraca_workmediaprodigy: if you want a developer to open the source of a successful product, you'd better be prepared to pay for it
22:25.01brettnemhey, I have a fresh (today) 1.4 svn. Polycom attended transfers fail.. if I roll back to the latest 1.4 stable, they work.. anyone seen this?
22:25.55bmoraca_workmediaprodigy: perhaps.  then again, perhaps i'm not.  you're not clearly expressing what it is you're looking for.  what it sounds like is you want an application to be released for free.  i'm simply providing you with the pragmatic viewpoint that it's not going to happen.
22:25.55brettnemon the attended transfer, the attended part of it works fine. .however when you hit transfer the second time to bridge the parties together, the call drops
22:26.17mediaprodigycarrar: i have just brought it up a few times.. i think there is potential there..
22:26.31mediaprodigybmoraca_work: pessimism will get you nowhere.
22:26.46*** join/#asterisk rossand (~aross@dhcp-233-179.tb-classrooms.carleton.ca)
22:26.46bmoraca_workmediaprodigy: pragmatism != pessimism.
22:27.02etfonhomeyjaskew, my only beefs with that device are that you have to buy the PCI express riser board separate.
22:27.03mediaprodigybmoraca_work: is there a need for call control like microsoft ocs
22:27.09nnyso back to my quandry earlier, anyone use the SPA504g cisco phones? They offer PSk for in call use, but the dumb thing tries to start a new channel to perform the dial, I am just looking to add feature codes from features.conf to the phone interface
22:27.15mediaprodigyor ciscos version with Asterisk
22:27.33*** join/#asterisk cvnet (~cvnet@dsl-69-172-67-161.acanac.net)
22:27.36cvnethi all
22:27.42etfonhomeyjaskew, and if you want RAID 1 or 0 with 2 x 2.5" drives, you have to buy the drive cage separately as well.
22:27.44bmoraca_workmediaprodigy: of course there is.  and there exist a number of applications which provide it.  they are, as I understand it, fairly successful.
22:27.55carrarcringes at the s ound of smeone calling a SPA phone a cisco :(
22:28.02mediaprodigywhere you have software control, with instant messaging and potentially webcam support, active directory
22:28.06carrarI call it like it is, LinkSys
22:28.16nnycarrar: hmm?
22:28.28nnycarrar: they are cisco small business now, as branded
22:28.47carrarI know
22:28.57nnyer rather small business pro*
22:28.59ManxPower-workThey are still Linksys design, not Cisco design
22:29.04nnythey work better with asterisk than the cisco phones
22:29.12mediaprodigybmoraca_work: if there is already something out there then great.. I am just asking
22:29.28cvnetI answer a call and then dial plan looks like this exten => _X.,n,VoiceMailMain(6000,u)  <== instead of saying please leave a voice message, it asks for password, what am doing wrong here?
22:29.39nnyI dunno, cisco, the company and cisco's engineers are the people I deal with when I have an issue, so i call them cisco
22:29.40bmoraca_workmediaprodigy: iSymphony.  to a lesser extent, hudlite.  to an even lesser extent, flash operator panel.
22:29.56nnythe cisco support office is in the same state as me
22:30.00ManxPower-workcvnet: do a "core show VoiceMail" and "core show VoiceMailMain" and notice the diffences
22:30.05nnyand the support techs aren't outsourced
22:30.27mediaprodigybmoraca_work: thanks for your help
22:30.27cvnetthanks
22:30.42p3nguinetfonhomey: As far as I know, lacking the license will not produce any failure whatsoever.
22:30.44jaskewetfonhomey:  good - a few options.  It's a charity project for a non-profit, so I get to do as musch as I can for free :)
22:31.18nnymatter of fact the guy I am dealing with on a now fixed sidetone issue is listed as Cisco SBSC Network Engineer.
22:31.38etfonhomeyp3nguin, other than no transcoding
22:31.50p3nguinetfonhomey: No.  No failure of any sort.
22:32.11p3nguinetfonhomey: The license is a LEGAL issue, not a functional issue ... to my knowledge.
22:32.12etfonhomeyp3nguin, before I just added the license it would not transcode from g711 to g729
22:32.32mediaprodigybmoraca_work: it is a shame.. but thanks for the information
22:33.06nnycvnet: Voicemail vs Voicemailmain
22:33.06p3nguinetfonhomey: I have known some people to not have license but also do have the codec and it does transcode both to and from g.729.
22:34.12p3nguinetfonhomey: If the lack of a license prevented the codec from working, their systems would not have been transcoding.
22:34.12etfonhomeyp3nguin, Actually, I just didn't have the codec.  Had to load it.
22:34.12bmoraca_workmediaprodigy: i'm not sure why it's a shame.  i understand the want for an opensource software of this type, but as long as there is demand willing to pay for it, it's unlikely that existing products will be released as open source.  if you want to start an opensource version, you're more than welcome, and i would embrace it...but it's a big, big undertaking
22:34.13p3nguinetfonhomey: precisely my point.
22:34.14nnyhonestly I'd rather pay for it
22:34.26nnyin regards to bmoraca_work and mediaprodigy discussion
22:34.30etfonhomeyp3nguin, but, I'm legal now anyway.  It's only $8 and only needed one channel
22:34.42p3nguinetfonhomey: Sounds good to me.
22:34.48nnyall the other attempts that are free always seem to get abandoned etc
22:35.14nnyi'd pay 200 bucks or so on top just for a full featured non trixbox/freebx/fonality call center heads up solution
22:35.46bmoraca_worknny: i built one myself a while back...i could drag&drop transfer calls and had BLF capabilities as well as see who they were talking to...but i felt like I was reinventing the wheel and just ended up using iSymphony
22:36.11etfonhomeyp3nguin, I only noticed the issue because my ITSP negotiated g711 with * while my phones negotiated g729 with *.  So, when * bridged the calls, it couldn't transcode.  I had g729 first in both lists, but my ITSP must have left out g729 on its side that time.
22:36.39nnybmoraca_work: yeah gonna look at that again, fop2 is pretty damn sweet and the guy only wants 50 bucks for a 15+ extension license, just missing anything other than the panel aspects.
22:36.39bmoraca_worknny: iSymphony is ~$650 for the server and ~$40 for each client...but it's well worth it if you need that functionality
22:36.53p3nguinetfonhomey: If you have g.729 on all your devices and g.729 on your "trunk," I don't think you even need to have the codec at all.
22:37.09mediaprodigybmoraca_work: I worked on this software, microsoft and cisco have released OCS and whatever cisco's version is called.. to me i think that the software this company makes will not live many years longer because of the budgets allocated by these bigger entities.. i had already approached and asked once, shut down.. the second approach seemed to generate some interest.. but it is a big...
22:37.11mediaprodigy...undertaking.. my idea is that if we release the code, who knows what developers might create.. the software already works..
22:37.15nnybmoraca_work: price all depends on funcitonality. i would also like to rebrand it if possible
22:37.42etfonhomeyp3nguin, exactly, but my ITSP did not negotiate g729 for some reason like it has been doing for weeks.
22:37.46bmoraca_worknny: the guys at i9 Technology will allow you to rebrand it.  for instance, Intuitive Voice rebrands it as "iView"
22:37.55nnybmoraca_work: that's good
22:38.14*** join/#asterisk nightrid3r (kvirc@41.214.236.77)
22:38.25carrarbut then Apple will sue you
22:38.28mediaprodigybmoraca_work: well i did not mean to get all into a discussion.. just had a question and it has been answered.
22:38.30mediaprodigythank you to all
22:38.35p3nguinetfonhomey: disallow=all  allow=g729
22:38.40p3nguinetfonhomey: Then it wouldn't have a choice.
22:38.44*** join/#asterisk nightrid3r (kvirc@41.214.236.77)
22:38.46etfonhomeyp3nguin, I had never had to deal with the codec.  So, it was good to see the process through once in case I had a client who needed it.
22:38.59nnybmoraca_work: pricing is a little wonky, per queue, per client, etc. Feel like it's nortel lol
22:39.02etfonhomeyp3nguin, it wouldn't have choice but to drop the call if it for whatever reason couldn't negotiate g729.
22:39.14nnywish they would stop trying to dip into larger profits and just offer flat rates
22:39.17p3nguinetfonhomey: Yeah, I guess that's true.
22:39.24etfonhomeyp3nguin, which is what is why I left g711u as the 2nd option after disallow=all,
22:39.31nnynot like it requires more effort on their end to manage larger setups
22:39.47p3nguinetfonhomey: Sounds reasonable.
22:39.48nnyunless they are gonna install it for me, train the client, and handle support lol
22:40.40etfonhomeyp3nguin, of course, it must have been temporary because after I bought the codec and installed it, I did a sip show channels on a test call and, of course, both legs were g729...
22:41.27p3nguinetfonhomey: For the $8, it's worth it to know how to configure it.
22:41.45p3nguinetfonhomey: Even if you never use it.
22:42.28*** part/#asterisk mediaprodigy (~chatzilla@72.20.157.179)
22:44.55etfonhomeyp3nguin, this is true
22:47.44bmoraca_worknny: meh.  i've found that, honestly, there's very little reason to provide that to anyone but receptionists and managers.  it's easy enough to make a queue status application using AMI.
22:49.08nnybmoraca_work: true although we pride our setups on not having per user licensing
22:49.15nnyer rather take pride
22:49.19nnyman i suck at typing what I am thinking
22:49.55bmoraca_worknny: we do as well...but the fact remains that there's always going to be a "per-seat" examination of total cost.  and I treat this type of software as "extra"
22:52.38nnybmoraca_work: indeed, i am going to evaluate it
22:53.52[TK]D-Fendernny: I've been evaluating WinZip for about a decade now apparently ;)
22:56.07carrarhahah [TK]D-Fender, how do you like it?
22:56.30carrarNot sure if it's really what you're looking for?
22:56.32Kattydrags in
22:56.54Kattyplops down
22:56.54carrarharro katty
22:56.59Kattystares blankly
22:57.15Kattyharro carrar
22:57.36nny[TK]D-Fender: lol
22:59.49Kattyugah, what a day
23:02.55Kattyso quiet.
23:03.04carrarKatty, tell us about your day!
23:03.13Kattyit was boring. that's all there is to it.
23:03.13*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw)
23:03.19Kattyhow was your day?
23:03.42carrarSo far not bad, wife made chocolate truffle cupcakes
23:03.50Kattyyum.
23:04.01carrarand said she made them because I use UNIX and not windows
23:04.06Katty:>
23:04.09Katty<3
23:04.41carrarAbout to go for a walk and attempt to burn that off
23:04.47KavanSok, so if I'm switching from zaptel to dahdi, what would I have to replace in my extensions.conf ?  I'm googling around but I'm finding mixed results on what to do for an upgrade
23:04.50Kattyk
23:05.07[TK]D-FenderKavanS: s/zap/dahdi/
23:05.20KavanSk, roger that
23:10.21Kattyomnomnomnoms strawberry and banana slices
23:10.29raden_work:)
23:10.44Kattyhugs raden
23:11.10Kattyraden_work: haven't noticed any changes yet, but i've only been taking the omega 3 for about a week now...maybe less
23:11.37raden_workhugs Katty
23:12.14raden_workKatty, you will notice changes over a longer time period things will balance out
23:13.36Kattyraden_work: lucky for me strawberries produce an immediate change
23:13.41Kattyraden_work: smile inducing!
23:14.16*** join/#asterisk beta2k (1000@d24-36-68-97.home1.cgocable.net)
23:14.30beta2kAnyone know if asterisk can handle distinctive ring on a FXO port?
23:14.42beta2kOr will all rings be answered the same?
23:15.34nnybeta2k: try http://lists.digium.com/pipermail/asterisk-dev/2003-November/002196.html
23:16.02nnyor http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels#DetectingDistinctiveRingonIncomingCalls
23:17.09raden_workKatty, :) lol
23:20.02pfnhmm, what's wrong with teliax
23:20.04pfndid they shut down
23:23.52raden_workpfn, why u say that ?
23:24.10pfntheir website is busted, err500 and no response from their proxies
23:24.17pfnand damnit, my voipjet account is busted, too
23:25.15raden_workno problem here
23:25.18raden_workcheck your provider
23:25.32raden_workjon@Server200:~> ping teliax.com
23:25.32raden_workPING teliax.com (63.211.239.26) 56(84) bytes of data.
23:25.32raden_work64 bytes from www.teliax.com (63.211.239.26): icmp_seq=1 ttl=48 time=57.2 ms
23:25.32raden_work64 bytes from www.teliax.com (63.211.239.26): icmp_seq=2 ttl=48 time=55.1 ms
23:25.40pfnit was an error 500 page
23:25.45pfnbut that appears to have been resolved
23:25.48*** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl)
23:25.50pfnnow waiting for their proxy to come back
23:25.50*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194)
23:25.59*** join/#asterisk styelz (~yoohoo@m0o0.mooo.com)
23:26.00*** join/#asterisk mnt_real (~sinan@bas1-montreal43-2925257034.dsl.bell.ca)
23:26.00*** join/#asterisk anthm (~anthm@freeswitch/developer/anthm)
23:26.00*** join/#asterisk ttwhy (~tekkno@p4FECFC04.dip.t-dialin.net)
23:26.00*** join/#asterisk e4 (~e4@rrcs-76-79-59-194.west.biz.rr.com)
23:26.06pfn[Feb 11 15:21:36] NOTICE[486]: chan_sip.c:13048 handle_response_peerpoke: Peer 'teliax' is now Reachable. (916ms / 2000ms)
23:26.08pfnah, finally
23:26.23*** join/#asterisk slima (slima@unaffiliated/slima)
23:26.30raden_workthere website tells you nothing about your registration
23:26.40*** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage)
23:26.40*** mode/#asterisk [+o leifmadsen] by ChanServ
23:26.42pfnraden_work, both were down
23:26.44raden_workI would ponder to guess it was your ISP
23:26.50raden_workpfn, if u say so
23:27.01pfnno, it wasn't
23:27.08pfnhttp://status.teliax.com/
23:27.30*** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee)
23:27.36raden_workwow get a real provider
23:28.13pfnyeah, I should switch away to someone else instead of teliax
23:28.15pfnshrugs
23:28.42raden_workVitelity
23:28.51raden_work185 days no connection interuption
23:30.17pfnI tried signing up for vitelity a while back, signing up with a credit card was a pita
23:30.47*** join/#asterisk etfonhomey (~etfonhome@74-131-159-160.dhcp.insightbb.com)
23:32.22raden_workpfn, cause of the verification ?
23:32.51pfnyeah, the verification was funky or something other, I just remember it being a pain and didn't bother following through
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23:36.53raden_workpfn, yeah its a pain was for me as well
23:37.11pfnbut it's a good point, I should add another provider to my list
23:37.31pfnI have voipjet also, but I don't really use them much and it kinda got unconfigured
23:38.32raden_workpfn well you ever need serive I have $3/mo wi 1.55 cents per min
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23:41.59*** join/#asterisk `Sauron (sauron@dsl001-130-033.aus1.dsl.speakeasy.net)
23:45.32raden_workdadhi Install >>>    You do not appear to have the sources for the 2.6.31.8-0.1-default kernel installed.
23:45.33raden_workmake[1]: *** [modules] Error 1
23:45.41raden_workinstalled kernel-source
23:45.42raden_worksame thing
23:45.54spenguin[w0rk]is it the same kernel source?
23:46.04*** join/#asterisk tzafrir (~tzafrir@212.179.75.202)
23:46.18spenguin[w0rk]same as 2.6.31.8-0.1
23:46.21[TK]D-Fenderbecause you need the matching HEADERS
23:46.30raden_worki will double check
23:47.44raden_work.12 instead of 8 thanks :)

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