irclog2html for #asterisk on 20051018

00:00.54JohnnyC"Open of mISDN Failed" anyone working with mISDN ?
00:04.15*** join/#asterisk litage (n=nick@203.220.55.70)
00:11.54*** join/#asterisk coppice (n=chatzill@123.192.17.210.dyn.pacific.net.hk)
00:13.04galelsome body know why i can register to a iax-truk ... this is what it shows:XX.XX.XX.XX:4569    master      <Unregistered>             60  Rejected
00:14.21galeli have to change something in manager.conf
00:14.22galel?
00:14.25*** part/#asterisk SarahEmm (n=sarahemm@Toronto-HSE-ppp3685577.sympatico.ca)
00:14.41galeli changed to accept:0.0.0.0
00:14.50galelsome one?
00:15.04*** join/#asterisk fugitivo (n=ajf@209.13.243.83)
00:18.01*** join/#asterisk zotz (n=zotz@24.231.36.100)
00:22.27halorgiumwhat computer  specs would be bare minimum to support asterisk + a x100p fxo?
00:22.30*** join/#asterisk file (n=jcolp@mctnnbsa31w-142166095097.nb.aliant.net)
00:23.22Nuggeta cpu, some ram, and a unix.
00:23.28halorgium:P
00:23.31Nuggetreally
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00:23.54Nuggetwell, actually, s/unix/linux/ for the zaptel card.
00:24.35xhelioxThe IAXy is requesting a registration period of 0 seconds, when Asterisk's default is 60 seconds. Is there any way to tell the IAXy how long to register for? And yes, I know the minreg can be set in iax.conf, but that creates a different issue, which isn't really relevant. :)
00:28.08*** join/#asterisk XTR-II (n=xtr@staff-nat.netnation.com)
00:28.08*** join/#asterisk adker (n=adker@67-51-239-111.dsl1.glv.ny.frontiernet.net)
00:41.24*** join/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net)
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01:04.32Icemaannafter a call to Set(GROUP()=TEST), GROUP_COUNT(TEST) still returns 0. Is this the expected behavior?
01:04.43Icemaannusing 1.2beta1
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01:22.16drumkillaso chatty in here ...
01:22.21Juxtyep
01:22.26drumkillarussell-: ?!
01:22.26brookshirerussell.. you're a nubb ;)
01:22.28JerJercan't keep up
01:22.30drumkillaTHAT'S MY NAME
01:22.33Juxtanyone have a take on goiax.com
01:22.42brookshirejust kidding
01:22.43brookshire<3
01:22.45Juxti am seriously confused of how they can afford to do what they do
01:22.48drumkillabrookshire: it's nub.
01:22.59brookshirelol
01:23.04brookshirewell.. ok
01:23.07brookshirei really don't care
01:23.35drumkillayour mom doesn't care ...
01:23.37JerJer404
01:23.48drumkillaFOUR OH FOUR!
01:24.01Juxtwww.goiax.com
01:24.13drumkillai barely slept that entire week.
01:24.14fileI remember reading that on the mailing list
01:24.30JerJeri missed it  :(
01:25.06*** join/#asterisk dalfry (n=dalfry@ool-435285b1.dyn.optonline.net)
01:25.55Juxtthey do not pass caller id but the quality is very nice
01:28.17bweschkeJuxt: they're a clec... the termination is cheap and it helps them to work on a scalable iax platform for their paying clients
01:29.02Juxtso they are taking this traffic so they could test scalability of their system?
01:29.55bweschkeyes - that is the goal.. they want to work on their iax platform so it scales horizontally.... they will be making changes to it over time to see how they can get it to perform better to suit their needs.
01:30.09bweschkeso we're guinea pigs essentially, and in exchange, we get the free termination
01:30.16Juxthow do you know this
01:30.59bweschkebecause I'm one of their paying customers
01:31.13Juxtoh, what's the name of the company?
01:31.16bweschkeand the clec stated his intentions with the platform on the forums a few weeks back
01:31.25bweschkeused to be txlink, now acquired by CommPartners
01:32.09Juxtgotcha
01:32.15Juxtthey are brave :-)
01:32.59bweschkethey're also smart... how many minutes you think they need to serve up before they equal the cost of an Empirix Hammer tester and even then how much time/labor would you expend to configure the hammer tester to simulate real world load?
01:33.16bweschkethey're getting real world load for only the cost of the term which is sub a penny per min for them
01:33.27Juxtyeah it makes sense
01:34.19hardwireblah
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01:48.02lilneonhey everyone
01:50.09*** join/#asterisk babak (n=root@r-72-244-140-113.nycmny83.covad.net)
01:50.14lilneongot a quick problem guys, reinstalled rh9 on my asterisk box (got bigger hard drive).. and added a tdm11b card, keep getting a dma_timer_expiry error followed by a too much work on the interrupt abut 8 times on the screen when linux bootss
01:50.20lilneonany help?
01:50.44lilneonis my hard drive bad? or is the digium card grabbing all the interrupts leaving none for the network card?
01:51.28lilneonhello?? n e one home?
01:51.52*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
01:52.07lilneonwhere is everyone?
01:52.08babaki need help: i have an asterisk system with tdm400p with 4 fxs modules. when i dial out to the pstn i almost always get a "please dial 1 before making this call message".
01:52.24babakcan anyone help?
01:52.41mog_homeadd wwww before the ${EXTEN}
01:52.47mog_homeim here lilneon
01:52.55lilneonhi mog_home
01:53.03lilneondid u see my problem?
01:53.07mog_homenope
01:53.28babakhi mog_home. i'll try wwww.
01:53.28lilneongetting a dma_timer_expiry error after reinstalling rh9 on my asterisk box
01:53.47mog_homewhen and where?
01:54.11lilneonduring first boot
01:54.17lilneonand well every boot up after that
01:54.39mog_homedmesg says that?
01:55.04lilneonduring boot up
01:55.18lilneondidn't bother check dmesg
01:55.19brookshiremog!
01:55.26mog_homebrooks
01:55.32brookshiregtknubbjab
01:55.45mog_homeword
01:56.09brookshiregtkastnubbjab
01:56.09mog_homewe need to hook that up
01:56.10brookshire:D
01:56.14mog_homelol
01:56.17mog_homebetter
01:56.18lilneonmy network card stopped wrking as well after i added a tdm11B card.. says too much work for the interuupt
01:56.21file256 area code omg
01:56.38brookshiregtknubbastjab
01:56.47brookshirefile what what?
01:57.02mog_homeno gtkastnubbjab
01:57.57lilneon?? whats with the gtkastnubbjab?
01:58.01Juxtlooks like you need to go into the bios and resuffle the interrupts
01:58.20mog_homeits a top secret project.....
01:58.24lilneonit doesnt allow me to do that juxt
01:58.47lilneonjust has plug and play OS yes/No.. and then it assigns interrupts to what ever device it likes..
01:59.04Icemaanncan someone explain to me how GROUP() and GROUP_COUNT work? If I do, Set(GROUP()=TEST), and then, NoOp(${GROUP_COUNT(TEST)}) it returns 0 for the group...
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02:00.35*** part/#asterisk alibby (n=alibby@pcp01412034pcs.phnixv01.pa.comcast.net)
02:00.38babakhi mog. thank you, thank you, thank you!!!! that did it.
02:00.55babakthank you!!!!!!
02:00.56mog_homeno prob
02:01.09mog_homethe line was seazing to fast
02:01.10babakso, i guess wwww add a wait before dialing?
02:01.19mog_homethe www add 300 millaseconds of wait
02:01.25mog_homew = 100 millasecond wait
02:01.38babakwhat does seazing mean. i've seen it here and here...
02:01.58mog_homeit grabs the line and then dials
02:02.01Icemaannit actually seems to start the count at 0, which is fine, but when doing a transfer it seems it starts the count at 1 lol. I can work around it with variables, but I wanted to see if Im using the functions right. If I am I will file a bug report
02:02.04babakah
02:02.14mog_homenow  what happens is it grabs the line waits 300 millaseconds then dials
02:02.24babakthis is great.
02:03.02babakis there a line seaz setting that i can set (on my phone or asterisk) or should i just add "w"
02:03.42mog_homewell you could
02:03.45mog_homebut its lame
02:03.47babakactually, if i think about it, its not my phone...
02:03.49mog_homebetter stick with wwww
02:03.52babakyeah.
02:04.01babakthank you very much.
02:04.18babakbye
02:04.25*** join/#asterisk brad_mssw (n=brad@ip24-170-193-14.ga.at.cox.net)
02:05.39Brijnlilnoen: Did you look at the noapic kernel option.. Just guessing, but there was something you could feed the kernel at boot time
02:07.01Corydon76-homeIcemaann: file a bug report
02:07.23IcemaannCorydon76-home: k
02:10.41lilneonbrijn... um..  you know what command line parameter?
02:12.37Brijnlinux noapic at the boot prompt
02:13.12lilneonbrijn:googling.. as i ahve no idea what it is
02:13.41brad_msswanyone here use teliax?  any idea how to improve DTMF code reliability?  seems like sometimes when you dial extension 112, it'll think you dial 111 or similar (seems to mainly mess up from a cell phone) ...
02:13.48_ThorWhat port is used by CLI?
02:15.03Brijnme neither, but i know it has something to do with irq routing
02:16.43lilneonbrad_mssw : i use em, but mainly dial from analog phones.. havent had problems.. some users say they have a hard time entering their pin# though when they attempt a call fro their cellphone.. but i have never checked it out.. try mailing them
02:18.44brad_msswlilneon, just signed up today, was experiencing the same issues with vonage ... which is why i'm trying teliax ... i guess i'll research it more
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02:22.51lilneonbrad_mssw : yeah cool, they are ok..  if u find anything shout me, cant test now cuz my asterisk test box is down.. sigh
02:27.03*** part/#asterisk schuylerdigium (n=Bosco@pcp03052091pcs.huntsv01.al.comcast.net)
02:27.31*** join/#asterisk alephcom (n=Miranda@207.34.97.130)
02:27.32pauldydoes teliax support *
02:27.36*** join/#asterisk SplasPood (n=sp@brooklyn.paravolve.net)
02:27.51pauldyI noticed you can get vontage working if you don't mind their bs 500 minute softphone limit
02:27.59lilneonpauldy : um yeah.. u can connect via iax, sip etc
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02:30.45brad_msswpauldy, teliax supports both SIP and IAX2 for asterisk ....  pauldy, vonage has a business plus offering they don't advertise that they allow direct sip
02:31.30pauldyyea just read the rates though and in comparison to broadvoice teliax is much higher
02:31.40pauldyreally brad_mssw got any links or rates
02:33.05pauldyjust currious been wadding through provider hell trying to find somewhere to park a client I did an asterisk install for
02:33.21pauldywanted to go forward with broadvoice and now they don't have any local dids for their area
02:34.19pauldynot to mention lnp hell because they want to port over three numbers
02:34.34brad_msswpauldy, just do a google search for 'vonage business plus', you'll find a bunch of resellers ... vonage doesn't seem to want to sell direct
02:34.56brad_msswpauldy, minimum plan is 4 simultaneous lines, 5k minutes, $150/mo
02:36.00*** join/#asterisk file[laptop] (n=jcolp@mctnnbsa31w-142166095097.nb.aliant.net)
02:36.17pauldyyea to much for a soho setup
02:36.41pauldythey are paying just a bit more than that right now for their service
02:36.49brad_msswteliax pay as you go is a really nice plan
02:36.51pauldywith POTS lines
02:36.58brad_msswjust waiting to see if it's reliable enough
02:37.14pauldyright now I cna't seem to beat the offer broadvoice has
02:37.43brad_msswbasically it's only $5/mo + 0.02 per minute ... and unlimited simultaneous calls
02:38.00brad_msswdon't think broadvoice allows more than 1 simultaneous call
02:38.05pauldyI can move their other lines to buisness lines and do a tele-branch on the numbers they can't port and still be under 70 bucks a month
02:38.28pauldyI've had 6 simultanious calls in a meetme conference over bv
02:38.36IcemaannCorydon76-home: I filed a bug report, 5453, I falsely put it under core functions instead of New Functions, sorry.
02:38.51brad_msswpauldy, 6 simultaneous calls on a single broadvoice phone number?
02:39.02pauldyyup
02:39.04brad_msswpauldy, I have broadvoice for home service, didn't think it was possible
02:39.13pauldyvery much so
02:39.39pauldyI've routinly had more than one the 6 was a special occassion to see if it would kill my broadband
02:39.41pauldyand it did
02:39.55brad_msswhmm, is that only the business plans then ?
02:39.55Icemaannpauldy: I've heard Telasip is good, havent tried them yet
02:40.24pauldybrad_mssw, nope mine is residential because I just use it to fool around with
02:40.47pauldyyea I saw a review for them on bbreports
02:43.05*** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com)
02:43.08brad_msswpauldy, interesting ...
02:43.47pauldyyou trying it now
02:43.49pauldyhehe
02:44.44pauldyyea telasip to expensive for a small home buisness
02:45.36Icemaannyea their business rates do seem hi. I have emailed their tech support and they (Gene) is very responsive. He may work with you on the pricing. The website leaves a lot to be desired
02:46.00file[laptop]Gene... where have I heard that name
02:46.11file[laptop]oh I remember
02:46.12pauldyYea they have a premium on some of their hardware to which makes me think they are a third party provider
02:46.25alephcomGene is working on doing some nice stuff with the rates as well as the packages I believe.  I'm not sure exactly when though.
02:46.53alephcomThey resell for Level3 I believe.  I heard that from one of their customers.
02:47.03Icemaannyea they use level3
02:47.13alephcomI really don't know what they have for their own lines....
02:47.16IcemaannI will probably setup an acct with them in the next couple of weeks
02:47.50alephcomHe will hopefully have online signup by then.  He said he was working on it as I don't think they have that right now.
02:48.00Icemaannno they done
02:48.03Icemaanndont*
02:50.50Juxttelasip does level3?
02:51.41crash3m__<PROTECTED>
02:53.04pauldyanyone here happen to know why number are portable to some providers and not to others
02:53.37JonR800i think some providers lie because they don't want to deal with the hassle. :)
02:54.32pauldythats what I think to but I figure someone else might know if there is a reason so I cans top being pissed at them for not doing it
02:57.34JonR800i spoke with 4 providers who were all able to provide me with a number in my area code/exchange.. 2 of the companies known for their support said porting would be no problem, 1 said no, and the last didn't reply.
02:58.35JonR800i was left with the impression that the last two just didn't give a shit.
03:00.55Kattyhi lads.
03:07.37*** join/#asterisk mcadory (n=mcadory@208.149.64.28)
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03:11.04n3u7wow
03:11.19n3u7I finally have asterisk installed on SuSE9.3
03:11.51n3u7so now I'm this far:
03:11.55n3u7usr/lib/asterisk/modules/chan_modem_i4l.so: undefined symbol: ast_unregister_modem_driver
03:11.55n3u7Loading module chan_modem_i4l.so failed!
03:15.57pauldyvi /etc/asterisk/moduls.conf add the line "noload => chan_modem_i4l.so" to the end of the file
03:16.12*** join/#asterisk twisted[digium] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted)
03:16.12*** mode/#asterisk [+o twisted[digium]] by ChanServ
03:16.48drumkillatwisted[digium]: nub
03:17.29file[laptop]O.O
03:18.22pauldyhow do you get info on the modules that start with pbx_
03:18.43pauldylike show applications
03:18.50file[laptop]Katty: nooooooo
03:20.00drumkillai hope it's good
03:20.38file[laptop]drumkilla: how is my favorite Russell doing?
03:21.45drumkillagood
03:22.17*** join/#asterisk Moc__ (n=mochouin@modemcable181.215-82-70.mc.videotron.ca)
03:23.10jake1932has anyone recorded prompts that are faster spoken than the defaults for voicemail?
03:27.00Kattydrumkilla: Russell stover, i do hope
03:27.00*** part/#asterisk mcadory (n=mcadory@208.149.64.28)
03:28.35n3u7false allarm SuSE9.3 install failing on account of libnewt
03:30.53*** part/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net)
03:31.45Qwelljake1932: Allison can, for a small fee
03:33.26Kattynitenite.
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03:41.55n3u7file this is exausting
03:42.11n3u7Apthis makes Ubuntu and SuSE
03:42.19_ThorAnybody knows what pport is used by the CLI?
03:42.29file[laptop]the CLI doesn't use a port
03:42.36file[laptop]it uses a unix socket
03:42.38_Thor?
03:42.50_Thorhow come?... the manager uses a port
03:43.21file[laptop]because it doesn't.
03:43.42_Thorwell file... coming from you, I'll take your word for it :)
03:44.04twisted[digium]file[laptop], you poked?
03:44.09_Thorthank you
03:44.43file[laptop]twisted[digium]: your Powerbook is showing.
03:45.26_Thorummm.. I got the answer now!... because it never really sends anything out of the box....
03:45.37twisted[digium]file[laptop], oops.
03:45.50*** join/#asterisk dos000 (n=dos000@i216-58-9-73.cybersurf.com)
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03:46.36Qwellhmm
03:46.36meanphilI keep getting "Polarity reversed (-1 -> 1)" and then the same thing but with the numbers the other way around
03:46.41meanphilanyone know what this means?
03:46.56QwellAre there any programs/daemons that will connect to a socket, and open a port for said socket?
03:46.58Qwellor something
03:46.58meanphil(when using a Digium TDM11B)
03:48.27arp2qwell, you mean like netcat?
03:48.28*** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net)
03:48.40arp2what do you mean 'connect to a socket' and 'open a port for said socket'?
03:48.43Qwellarp2: yeah, probably
03:48.52Qwellarp2: dunno
03:49.43n3u7where can i get the patch to make asterisk compile without libnewt?
03:50.06*** part/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net)
03:50.06Qwelln3u7: installing libnewt would be easier...and supported
03:50.08lilneonhey guys, anyone know what siocsifflags incorrect? my network card stopped wrking
03:50.27lilneonis there a way to see a list of IRQs and devices they are assigned to in RH9?
03:50.57Qwellcat /proc/interrupts
03:51.05*** join/#asterisk bmg505 (n=leon@rndf-146-57-153.telkomadsl.co.za)
03:51.31n3u7edited the make file to include OPTIONS=none
03:51.34n3u7it worked
03:51.36n3u7but
03:51.42n3u7cli.o err!
03:53.29lilneonthnx Qwell
03:53.35dos000anyone know what it takes to mimick a CO interface to a modem that talks v22.bis ? i want to avoid renting lines from the CO
03:53.59dos000best being using digium cards
03:54.01*** join/#asterisk KaBewM (n=kabewm@66-215-7-106.dhcp.psdn.ca.charter.com)
03:55.19Kyreethdos: fxs ports emulate a CO.. You can plug phones or modems into them.
03:55.40n3u7OMG...it worked
03:55.53n3u7ASETERISK on SuSE9.3
03:55.57n3u7!
03:56.00n3u7::)\
03:57.03pauldyn3u7, now go cross compile it for the tivo
03:57.22Igbothom_IIIand the imate JasJar
03:57.45BrijnIs there a reason to keep the default Zap/go channel?
03:57.48Brijng0
03:57.57pauldyasterisk at home?
03:58.07BrijnManual MAP install
03:58.10BrijnAMP
03:58.26pauldyBrijn, if you just using voip then nope
03:59.00pauldyI think it sets it up as the default trunk
03:59.04BrijnKill kill kill :)
03:59.07n3u7ps -aux
03:59.13dos000Kyreeth, nice ... any idea what modem protocol the digium cards support ?
04:00.20n3u7awwwwww
04:00.25n3u7false alarm
04:00.28Kyreethdos: Ah, you need it to decode the modem protocol? Hm.
04:00.53n3u7make
04:01.19n3u7gcc: cannot specify -o with -c or -S and multiple compilations
04:01.19n3u7make: *** [cli.o] Error 1
04:01.25n3u7:0
04:01.28pauldynow whats with the fisher price voip phones http://www.voipsupply.com/popup_image.php?pID=557
04:01.30dos000Kyreeth, mind you i do not have a lot of data
04:02.29*** part/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net)
04:03.03BrijnI just received my SIP details from Dolphintel(.com), outbound calls are working OK.. But if I dial the number I get extention not available.. I don't see anyhin on the * console. So it's Dolphintel I guess.. What is needed to accept incoming calls?
04:03.20*** join/#asterisk copantl (n=copantl@205.240.200.98)
04:04.13ManxPowerBrijn, do a sib debug then try calling the number
04:04.28Kyreethdos: You'll probably need the calls to come in over ISDN, maybe from an FXS channel-bank, and look into this: http://www.voip-info.org/wiki/view/Asterisk+ISDN4Linux
04:04.50Kyreethdos: Alas, I'm not an expert, and haven't done what you're wanting to do.
04:04.53copantlany body knows howto interconnect 2 asterisk with iax trunk?
04:04.56pauldyBrijn, you may need to setup a DID
04:05.09pauldyinbound sip are rejected by default
04:05.47pauldycopantl, there is a tut in the documentation part of sourceforge for AMP
04:05.59BrijnManxPower: Nothing logged.. I wonder, I don't see a "registered xxx" for the ip of the provider. I also have a fwd trunk, and i see that IP
04:06.21ManxPowerBrijn, do a "sip show registry"
04:06.24Brijnpauldy: i added a did (i think), in extentions_additional.conf there is a entry for the DID
04:06.52copantli tried pauldy but i got an error: autentication reject ... any idea?
04:06.52BrijnmanxPower: Ahhh, it registered ok
04:06.54pauldyhttp://sourceforge.net/docman/display_doc.php?docid=26418&group_id=121515
04:08.00pauldycopantl, make sure you have the authentication details matched up correctly ie peer should have the username for the remote connection on each config
04:08.02BrijnHmmmm, sip debug doesn't show any activity if I dial the number.. Anything else that I need to check on my side.. Or is it likely the other end forgot something?
04:08.53pauldycopantl, that was the only trouble I had following those docs + a context needs to be set where you see a c
04:08.53*** join/#asterisk nexis (n=nexis@12-219-60-252.client.mchsi.com)
04:09.27copantlcan you show me a example?
04:09.49pauldycopantl, the eample is on the site I pasted
04:09.56copantlok
04:10.07pauldyBrijn, what did you have for sip show registry
04:10.27nexisis it possable to run meetme if you have no ZAP interfaces?
04:10.49pauldynexis, yes just use the ztdummy
04:10.56BrijnHost                            Username       Refresh State
04:10.56Brijnvoip.lightspeed.ca:5060         001404640234       105 Registered
04:11.15copantlthere is my registry : 63.245.93.140:4569    master      <Unregistered>             60  Rejected
04:11.40nexispauldy, yea, i have ztdummy in the kernel, it still complains about channel.c:2206 ast_request: No channel type registered for 'zap', then says invalid confrence, and drops it.
04:12.06Brijnpauldy: does the "user" sip entry need a secret entry? Or can it be left out?
04:13.03copantlpauldy: what context do you use?
04:13.26n3u7how often ids the cvs updated?
04:13.30pauldynexis, wierd working here with that config do you have a trunk setup for zap or something
04:13.51*** part/#asterisk Moc__ (n=mochouin@modemcable181.215-82-70.mc.videotron.ca)
04:13.52n3u7could this error be fixed in two nights?
04:13.56n3u7gcc: cannot specify -o with -c or -S and multiple compilations
04:14.12twisted[digium]n3u7, cvs is updated regularly
04:14.25pauldyBrijn, you need the secret for the peer section
04:14.29twisted[digium]please check out a new version, make clean, and try again.
04:14.36pauldycopantl, context=from-internal
04:14.40n3u7:)\
04:14.50copantlin both sides?
04:14.57pauldyfor testing yes
04:15.05pauldyjsut to make sure you can dial one from the other
04:15.25pauldythey have another part that says outbound dial prefix don't worry about that
04:15.45pauldyinstead setup an outbound route to use the trunk with a dialing rule of 7|xxx or something
04:15.47Brijnpauldy: not for user, because we will decide by defining an did context if we want to accept the call?
04:15.52nexisodd, nothing is documented that you have to have chan_zap.so loaded to use meetme.
04:15.53copantli made the 4 version from voip-info
04:16.14pauldyBrianR, not for user because you don't want to authenticate the incomming call
04:16.36pauldyprobably best if you just leave the incomming settings blank
04:16.47pauldyjust enter the info in peer details
04:16.50n3u7inux:/usr/src/asterisk # cvs checkout asterisk
04:16.50n3u7cvs checkout: cannot chdir to asterisk: Not a directory
04:16.51n3u7cvs checkout: ignoring module asterisk
04:17.00n3u7?
04:17.14Brijn[frm-dolphin]
04:17.14Brijntype=user
04:17.14Brijncontext=from-pstn
04:17.32pauldyBrijn, probably not a good idea to paste the whole seciton in here
04:17.41pauldyesp the password etc...
04:17.45Brijnis what I have nowm even if there would be mistajkes there, i should see the call coming in on the console (verbose 30, sip debug)??
04:17.54twisted[digium]n3u7, try cvs update -d
04:17.59twisted[digium]from within your asterisk source
04:18.00BrijnI left the password out :)
04:18.07pauldyBrianR, if you aren't registered properly you might have problems
04:18.28Brijnpauldy: the sip how registered looked ok?
04:18.31n3u7that worked!
04:18.35pauldyfrom my experience I use the following register string and it has worked well for me on several installs
04:19.01pauldynumber@host:pass:number@host
04:19.26drumkillatwisted[digium]: make update, fool
04:19.33Corydon76-hometwisted[digium]: shouldn't that be:  make update
04:19.39drumkillaI win!
04:19.41twisted[digium]it works either way
04:19.46twisted[digium]silly rabbits
04:19.53file[laptop]tricks are for kids?
04:19.54drumkillayou forgot -P
04:19.55copantlpauldy : do i need to change something  in /etc/asterisk?
04:19.57Brijnpauldy: i guess one of the "numbers" is user?
04:19.59Corydon76-homeI think 'make update' does something extra
04:20.01drumkillathat's why you should just use make update
04:20.05n3u7:(
04:20.05drumkillaand there are other reasons, as well ...
04:20.07twisted[digium]sure.
04:20.09n3u7gcc: cannot specify -o with -c or -S and multiple compilations
04:20.09n3u7make[1]: *** [localtime.o] Error 1
04:20.18twisted[digium]n3u7, did you modify your makefile?
04:20.22pauldycopantl, if you have amp installed no
04:20.28pauldyBrijn, yes
04:20.38n3u7yes to compile with out libnewt
04:20.44n3u7options=none
04:20.48twisted[digium]drumkilla, i fixed your streamplayer for osx earlier.
04:20.52twisted[digium]:P
04:20.56drumkillahm?
04:21.00copantlasterisk@home
04:21.01n3u7libnewt is broken in SuSE9,3
04:21.12twisted[digium]simple fix, but fixed nonetheless
04:21.25drumkillai build on osx all the time ...
04:21.31pauldycopantl, the web based interface for AAH is AMP so no you shouldn't have to edit any of the files by hand just use the wb interface
04:21.38twisted[digium]drumkilla, it wasn't building
04:21.45copantlok
04:22.14Brijnpauldy: sorry, you wrote number@host:pass:number@host.. I guess it's number@user:pass:number@host or something.. Where should user be in your string?
04:22.17twisted[digium]two different macs, two different versions of osx
04:22.37drumkillai don't see your commit
04:22.44twisted[digium]it went in
04:22.47file[laptop]hrm
04:22.49twisted[digium]it's probably hung in mailmain again
04:22.52twisted[digium]er mailman
04:22.57drumkillawhat did you change
04:22.58pauldyBrijn, replace the number string with user host is the remote connection and pass is your sip secret
04:23.07twisted[digium]drumkilla, in streamplayer.c, line 42
04:23.29drumkillaok
04:23.33drumkillaannnd
04:23.36twisted[digium]err not 42
04:23.38pauldycopantl, another thing is just make sure that the two machines are talking to eachother by using iax debug
04:23.47twisted[digium]wtf
04:23.49twisted[digium]i commited it
04:24.02twisted[digium]strange.
04:24.05pauldyBrijn, is your inbound number not the user
04:24.31twisted[digium]lemme try again
04:24.52drumkillahappily builds for me.
04:24.57twisted[digium]because yea, the checkout didn't have the change
04:24.59Brijnpauldy: Hmmm, one thing, my password contains an @, would that eb a problem?? Do I need to escape that
04:25.09drumkillai think you're on crack.
04:25.10twisted[digium]no
04:25.19twisted[digium]i was at ABC with brookshire and all
04:25.27Brijnpauldy: it's a number yes
04:25.35ManxPowerBrijn, Yes, passwords with @ are bad
04:25.46BrijnI guessed that much :(
04:25.49pauldyBrijn, or get them to set another pass for you
04:25.49ManxPowerBut if sip shows registry shows you registered.....
04:25.50twisted[digium]it even gave me a rev bump
04:25.52twisted[digium]*shrugs*
04:26.03twisted[digium]I saw it not build.  Made the change, and it built
04:26.05drumkillait probably failed the up to date check
04:26.15JunK-Yi cant sleep.
04:26.17twisted[digium]shouldn't it have complained though?
04:26.19drumkillaI changed something this weekend at some point
04:26.23JunK-Ysup here?
04:26.24drumkillait's easy to miss
04:26.31twisted[digium]ah... could have been thn
04:26.33twisted[digium]er then
04:26.52pauldyI spent 8 hours trying to figure out why my meetme conferences suddenly stopped working only to find someone removed a # from an include line
04:26.59twisted[digium]i'll recommit as soon as i retest with this fresh tree..
04:27.00twisted[digium]grr
04:27.18twisted[digium]it should make zero functionality change, but allow it to be built on older versions of osx as well as newer ones
04:27.22*** join/#asterisk file[laptop] (n=jcolp@mctnnbsa31w-142166095097.nb.aliant.net)
04:28.09drumkillawhat's the change
04:28.18pauldyhey twisted how well do you think asterisk would run on a g4 433hz with 384MB RAM running 10.3.9
04:28.22drumkillai'm running tiger.
04:28.34twisted[digium]pauldy, *shrug*
04:28.37drumkillapastebin it or something
04:28.43twisted[digium]drumkilla, me too, but the two boxes i tested were panther and jaguar
04:28.57drumkillawas it that ifdef line?
04:28.57twisted[digium]it built fine on tiger all three tries
04:29.01twisted[digium]yea
04:29.02drumkillato include that extra header?
04:29.08BrijnManxPower/pauldy: If I escape the @ in the pwd it keeps hanging in Sent auth.. So I guess the @ is ok
04:29.22drumkillaso you just added, || defined(__Darwin__)
04:29.23drumkilla?
04:29.25JunK-Ywow, http://www.futureshop.ca/catalog/proddetail.asp?sku_id=0665000FS10058254&atab=&spviewed=&newlang=EN&logon=&langid=FR pretty cheap
04:29.27twisted[digium]*nods*
04:29.32drumkillaok, i'll add it ...
04:29.40twisted[digium]ok
04:30.06ManxPowerAnyone that pays $500 for portable audio is an idiot
04:30.18twisted[digium]confirmed - no difference in osx 10.4.2
04:30.52drumkillak, it's in
04:30.52pauldyManxPower, I fail too see the economic sense in ipods too
04:30.53drumkilla:)
04:31.00JunK-YManxPower: thats ur opinion :)
04:31.02twisted[digium]gratsi
04:31.08pauldyesp when PDAs are cheaper and will perform the same functions
04:31.12twisted[digium]did you commit from your ipod? :P
04:31.18drumkillaof course
04:31.29JunK-Ydrumkilla: i wonder if i sould take the new one, with video streaming.
04:31.31*** join/#asterisk fugitivo (n=ajf@209.13.243.217)
04:31.34ManxPowerpauldy, I can see spending $150 for such a thing if you really love music and are not near your computer.
04:31.36drumkillamy ipod is in my compilation farm
04:32.13ManxPowerthat reminds me, I need to re-down.oad my music collection to my laptop.
04:32.31JunK-Ypauldy: when i do jogging, i prefer having an ipod in my adidas pants instead of a pda :)
04:32.38pauldyyea my pda is now a sip extension, gps, movie player, ipod lookalike, and day planner
04:33.06ManxPowerI keep thinking about getting a PDA
04:33.16ManxPowerbut I need a car and a perm place to live first.
04:33.21pauldyJunK-Y, to each his own I just can't phathom such a purchase with its limited functionality
04:33.28drumkillag'night all
04:33.37JunK-Ysee ya drumkilla
04:33.46fugitivoi want a nokia 770
04:34.01ManxPowerpauldy, the thing is, I doubt Junky jogs for more than an 3 hours and most cheap audio players can handle at least that mucn.
04:34.33pauldymy PDA will play for 6 hours if I tune down the speed from 400Mhz to 100
04:34.42*** part/#asterisk zedas (i=zedshawc@pizarro.dreamhost.com)
04:34.48fugitivopauldy: what pda?
04:34.59pauldytoshiba e805
04:35.06fugitivohow much is that thing?
04:35.40pauldystill about 500 bucks
04:35.57fugitivowhat OS?
04:36.01pauldyif you cna find it and they don't make them any more
04:36.09pauldywinblows ce
04:36.43BrijnDoes exten => 6046289655,3,Goto(ext-local,201,1) mean jump to ext-local context, extention 201 after 1 second?
04:36.49pauldyspyros been working on a port of linux since early 2004 but its barely more than halfway complet
04:36.57fugitivowww.nokia.com/770
04:37.00fugitivoit runs linux
04:37.11fugitivoit's not really a pda, it's an internet tablet
04:37.15ManxPowerBrijn, no, itmeans jump to context ext-local, extension 201, priority 1
04:37.24n3u7hem astman.o
04:37.27pauldyI have an old 9200 that was the last nokia product I ever purchased
04:37.30ManxPowerof course it's not going to work if you don't have priorities 1 and 2 for that extension
04:37.46JunK-YBrijn: polux*CLI> show application goto will give u all the answers.
04:37.53n3u7trying to get around libnewt so that I can get this going on SuSE
04:37.54pauldyManxPower, shouldn't it be prefixed with an _ too
04:38.36ManxPowerpauldy, it's not a pattern so it doesn't need a _
04:38.46pauldyoh the other tweak I do to maximize usage is to turn off the PDAs screen
04:39.32n3u7what di I do now?
04:39.54pauldyn3u7, you need a bigger hammer to get that round peg in the square hole
04:40.08*** join/#asterisk pigpen2 (n=mark@fw.seamans.cc)
04:40.13n3u7heh, like editing the make file
04:40.27n3u7but I wouldneed lots of helpz0rs
04:41.14pauldy<PROTECTED>
04:41.29fugitivopauldy: i'm waiting the release to get one
04:41.52fugitivopauldy: www.maemo.org
04:41.53BrijnWhat does the hint part in exten => 201,hint,SIP/201 mean??
04:42.01ManxPowerDoes anyone have recommendations for a podcast client for wn32?
04:42.03pauldyany idea on what the hardware specs are nokia is rather limited
04:42.06BrijnI tried shop app hint ;)
04:42.52fugitivopauldy: http://europe.nokia.com/nokia/0,,75023,00.html
04:43.08BrijnBrijn, but it's not a command so that is not surpising
04:43.31fugitivopauldy: version 2006 of the os will have voip
04:43.33pauldyfugitivo, yea no cpu specs etc...
04:44.15fugitivopauldy: check www.maemo.org, it's for developers
04:44.29pauldyfound it 220Mhz OMAP 1710 powered by an ARM9 core
04:44.36pauldywierd is that not a TI chip?
04:44.57fugitivoi think i've read it's slow
04:45.03pauldyIt is it says it further down the page
04:45.17pauldyyea that thing isn't meant to handle a display that large
04:46.07pauldybut I believe if it is the chip I think it is there is a nice programmable DSP piggy backing it
04:46.15fugitivohttp://maemo.org/maemowiki/Nokia_770_Hardware_Specification
04:46.24pauldywhich would be nice for compression codecs for audio
04:47.00fugitivono PIM
04:47.04fugitivothat's bad
04:47.13copantlpauldy: i tried the configuration and the aix2 trunk debug say: Rx-Frame Retry[Yes] -- OSeqno: 001 ISeqno: 002 Type: IAX     Subclass: REJECT
04:47.13copantl<PROTECTED>
04:47.13copantl<PROTECTED>
04:47.31copantli used from-internal
04:48.17pauldycontext=from-internal
04:48.29copantlyes
04:48.38pauldythat goes in the user settings not peer
04:49.20copantlright
04:49.43pauldyyea fugitivo to bad this device will be pretty much outa date before it ships hopefuly the next revision will sport a faster cpu and built in video acceleration
04:49.53fugitivogimp will run on that device
04:50.00pauldynice
04:50.24pauldycopantl, run this cd /etc/asterisk;grep [from-internal] *
04:50.45pauldyno wait
04:50.49Corydon76-homeHeh
04:50.57Corydon76-homeDoing a character class, eh?
04:51.12pauldycd /etc/asterisk;grep "\[from\-internal\]" *
04:51.25Corydon76-homeYou need to double the backslashes
04:51.37Corydon76-homeOnce for the shell, once for grep
04:51.51*** join/#asterisk NoRemorse (n=axel@202.161.68.6)
04:51.53pauldyCorydon76-home, I just ran it and it worked fine
04:52.08pauldynot using egrep
04:52.10NoRemorsehi all, how do I recompile asterisk with a RF2833 dtmf payload of 99 please?
04:52.31pauldyit just hit me after I pasted that those brackets will cause issues
04:52.38Icemaannis markster in here?
04:52.39Corydon76-homeYeppers
04:52.54Corydon76-homeMark is probably asleep
04:53.03Icemaannnah, he is replying to bugs ;-)
04:53.08fugitivoscummvm will run on the nokia 700
04:53.10fugitivo770
04:53.14fugitivoso we can play monkey island
04:53.25pauldyhehe
04:53.40pauldyI had mame running on my e805 for a while it was neat
04:54.10pauldybut since I actually use my pda for productive things it came off as soon as the novelty wore off
04:54.37fugitivotoo bad no PIM is included
04:54.53copantlpauldy: a lot info :))!
04:55.09pauldyyea copantl did you try my revised version of the command
04:55.32pauldyjust looking to make sure it finds the line in extensions
04:55.36NoRemorsehi all, how do I recompile asterisk with a RF2833 dtmf payload of 99 please?
04:56.19pauldyfugitivo, http://oss.kernelconcepts.de/maemo/ there you go
04:58.00fugitivoniiiiiice
04:58.22fugitivonow we need a calendar and it'll be perfect
04:58.34pauldythe only drawback is the device is seriously underpowered from a cpu perspective
04:58.57pauldyfugitivo, like this http://www.steinbauer.org/matthias/computer/linux/gpe-calender-770/
05:01.10NoRemorseyou are a font. not.
05:01.12*** part/#asterisk NoRemorse (n=axel@202.161.68.6)
05:01.25*** join/#asterisk websae (n=websae@CPE-24-167-206-63.wi.res.rr.com)
05:02.14websaeanyone here have experience with new WIFI phones?
05:02.32websaeif so how are they with asterisk--and what seems to be the best one?
05:02.37copantlzyxel?
05:03.18*** join/#asterisk pigpen2 (n=mark@fw.seamans.cc)
05:03.45KyreethI've got one of the zyxel wifi phones, but it's having trouble registering with DHCP at work. Seems to work fine at home.
05:03.48*** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1)
05:04.11copantli got one too
05:04.54copantlbut myone works... but don't now how tranfer a call in that phone
05:06.06pauldyhave you tryed just pressing pound with it
05:06.17*** join/#asterisk erickj_az (n=erickj_a@wsip-68-98-222-74.ph.ph.cox.net)
05:06.37erickj_azIs there a channel dedicated to DUNDi
05:07.23pauldyhave you tried #dundi
05:07.52copantlyes
05:08.04copantlbut hangup the call
05:08.16pauldyhaha well thats not good then
05:08.27copantlright
05:08.44*** join/#asterisk ag0ny_ (n=ag0ny@ares.sengawa-networks.com)
05:09.03*** join/#asterisk logicalonline (n=Ken@209.242.52.25)
05:09.14copantlpauldy, i can't found what's wrong with my iax2 trunk setup
05:10.35pauldycan you do some screenies of the two trunk pages and put them up somewhere
05:11.29erickj_azDoes anyone here know anything about dundi?
05:11.34pauldywen can get the config fixed tehn you can change the pass info or whatever
05:13.00*** join/#asterisk rking (n=rking@ip68-105-231-56.lu.dl.cox.net)
05:16.04copantlok
05:20.56copantlpauldy : what do you need?... just tell me?
05:22.24*** part/#asterisk copantl (n=copantl@205.240.200.98)
05:22.41*** join/#asterisk copantl (n=copantl@205.240.200.98)
05:23.18digimeanyone recommend a good SIP softphone
05:23.22digimethat is open source
05:23.26digimebesides SJlabs
05:24.12copantlis there in digium some one called kenny?
05:25.20wunderkinno.. they killed kenny! those bastards!
05:26.29copantli need support from digium
05:26.53copantli bought a te110p
05:26.54wunderkinand only someone named kenny can do that?
05:27.23copantli'm not shure of hes name
05:27.26copantlbut no
05:27.38wunderkinok and?
05:27.55wunderkin<insert problem here>
05:28.48copantlok this guy kenny or kenneth or whatever , fix me a PRI/isdn issue with my telco
05:29.21copantland now my cdr not work
05:29.51wunderkinand define by not work , what happens, maybe someone can help you then
05:30.50copantli'm using asterisk@home
05:31.43copantland when i go to amp for see my cdr reports... they giveme the reports from 2 days ago
05:31.59*** join/#asterisk SarahEmm (n=sarahemm@Toronto-HSE-ppp3685577.sympatico.ca)
05:34.19copantland i made a  cdr status and give me this: cdr status
05:34.20copantlCDR logging: enabled
05:34.20copantlCDR mode: simple
05:34.20copantlCDR registered backend: cdr_manager
05:34.20copantlCDR registered backend: csv
05:35.17JamesDotComand they're meant to be logging to a database?
05:36.07JamesDotComoh
05:36.09JamesDotComa@h
05:36.12JamesDotCommeh
05:36.13JamesDotComlearn asterisk
05:36.18*** join/#asterisk konfuzed (n=konfuzed@H129.C72.B0.tor.eicat.ca)
05:36.28copantlthanxs.... i know
05:37.40copantlbut for a "simple mortal " like me is a good beginning
05:37.56*** join/#asterisk argos73 (n=mike@65-85-207-101.client.dsl.net)
05:40.32copantlthe other problem is the ANI dont pass thought the telco
05:40.37*** join/#asterisk bweschke (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net)
05:42.01copantli configured 2 ATA's with  2605600 and 2605601
05:42.54*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:43.00copantland when i placed a call from the 2605601 in  the CID shows 2605600
05:43.56copantlsorry for my english is not my first language !! :))
05:44.14n3u7hoe do I compile asterisk without astman?
05:44.24n3u7hoe do I compile asterisk withoutlibnewt?
05:45.24*** part/#asterisk logicalonline (n=Ken@209.242.52.25)
05:45.46L|NUXhave any one treid braintel DID here ?
05:46.04JamesDotComcopantl: a@h adds a whole layer of complexity above asterisk, you need a@h support
05:46.34n3u7I've been tyrying to install for 2 weeks
05:46.42n3u7two different distros
05:47.39*** join/#asterisk enota (i=dimka@freelsd.net)
05:48.31copantldo you know the channel?
05:49.16JamesDotComwhat distro now n3u7? why not just install the libs?
05:49.24JamesDotComcopantl: no idea sorry, might mention something on their website
05:49.59copantli tried to install asterisk - debian - amp really painfull!!!
05:53.13*** join/#asterisk dasuberdavid (n=egg@pcp01534754pcs.huntsv01.al.comcast.net)
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06:00.39X-Robn3u7 -why not install newt and astman?
06:00.54X-Robusually 20 seconds of isntallation is worth more than 2 weeks of scratching your head.
06:01.03pauldydamb I bet he never thought of doing that
06:03.18*** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye)
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06:15.13*** mode/#asterisk [+o kram] by ChanServ
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06:15.32*** mode/#asterisk [+o twisted[laptop]] by ChanServ
06:18.40*** part/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted)
06:20.03tzafrir_laptopconsidering that there are (unofficial) amportal debs for debian, it can't be that painful.
06:20.44wunderkinwb kram
06:21.31brookshiretwisted's new laptop is hot
06:21.36brookshirespeaking of laptops
06:21.38brookshireheh
06:23.17kramthanks wunderkin, but i'm headed to bed
06:23.24wunderkinbed? wow
06:23.39wunderkinbut its only 02:23
06:23.39wunderkin:D
06:24.29wunderkin- another stupid mistake by me.. whee
06:29.53*** join/#asterisk mmmToop (n=chatzill@196.31.11.194)
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06:42.59argos73ick - puppy runny poo at 2:45 am.  fun
06:51.32mmmTooplets keep it clean  ;  )
06:55.26*** join/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted)
06:55.26*** mode/#asterisk [+o twisted] by ChanServ
06:56.30twistedi am the nat king.
06:56.53twistedcisco 7960 --SIP--> NAT1 --> NAT2 --> Server
06:56.59twistedworks perfectly, with two way audio
06:57.51*** join/#asterisk af_ (n=af@ip-142-250.sn1.eutelia.it)
07:00.30X-Robtwisted - re #5150
07:00.38X-Roboej said 'next week'
07:00.43X-Roband it was meant to be in 1.2beta
07:00.45twisted5150...
07:01.04X-Rob<PROTECTED>
07:01.09X-Rob<PROTECTED>
07:01.11X-Robwups
07:01.29twistedyou can't submit bug reports on code that is not in the codebase
07:01.34X-RobI didn't
07:01.36X-RobOEJ did
07:01.51X-Robnote the 'Hey, I could make a copy!'
07:02.01twistedwhy would he set you as the reporter?
07:02.04X-RobFIIK
07:02.14twistedlol
07:02.14X-Robbecause on 4877 I was the reporter
07:02.17X-Rob?
07:02.28X-Robor maybe I wasn't
07:02.30twisted4877 was gst
07:03.08X-Robno idea
07:03.30*** part/#asterisk SarahEmm (n=sarahemm@Toronto-HSE-ppp3685577.sympatico.ca)
07:03.50*** part/#asterisk twisted[laptop] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted)
07:04.43X-Robso what should I do - I've already asked OEJ to comment, and there's been no responce.
07:04.50X-Robit does, definately, cause * to crash.
07:05.02X-Roball AMP distros do NoCDR/ResetCDR
07:05.21twisteddidn't you see markster's notes?
07:05.44X-Robyesh
07:05.50X-Roband then I commented immediately after it
07:05.54X-Robsaying 'Olle, any comments?'
07:06.08X-Roblike 5 minutes after he posted
07:09.12twistedokay, well, the bug is on code that isn't in the codebase apparently
07:09.17twistedand we cannot reproduce it
07:09.40twistedbut i'm not going to take any action on it until OEJ responds
07:09.54twistedsince you did the right thing and got ahold of me;)
07:11.48*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
07:11.48*** topic/#asterisk is Asterisk 1.2.0 Beta1 - http://www.asterisk.org || Astricon 2005 was a great success! Thanks to everyone who made it happen, as well as to everyone who made it out!
07:12.29infinity1hmm ..when dialing a gizmo #, i'm getting Got SIP response 488 "Not Acceptable Here" back from 198.65.166.131.
07:12.35*** join/#asterisk swm_ (n=admin@digitaldatabits.net)
07:13.00swm_Anyone know why Linux shows 4 Processors on a 2 Processor System under /proc/cpuinfo ?
07:13.14infinity1swm_: hyperthread?
07:13.20brookshirehyperthreading
07:13.22swm_Yeah It's hyperthreading
07:13.48swm_Linux 8.x didn't show 4, but Linux 10.x shows 4 of them ... amazing
07:14.08brookshirelinux 8.x?
07:14.14brookshiresuse?
07:14.20swm_Slackware
07:14.25brookshireahh
07:14.33swm_*slutware*
07:14.48swm_Not I have to find something to use all that power... Hmm
07:15.11brookshirewell.. it's actually still only two processors
07:15.50brookshirebut you can balance tasks on it better
07:15.52brookshirelol
07:16.01swm_balance is nice
07:16.04e-HernickI laugh at you and your weak linux 8.0! I am using Turbo Linux2000 X11 XR2006
07:16.30e-HernickAdvanced Edition !
07:16.32brookshiregentoo!
07:16.34swm_Turbo linux? lol
07:16.41brookshirei don't think it has a version
07:17.13e-HernickWell, I'm using the 2005.0 and 2005.1 profiles
07:17.32e-HernickI may still have a 2004.3 somewhere but I think they're all upgraded.
07:18.22swm_My only problem is I cannot get KDM or XDM to run . I dont care which one is running as long as I can access it via remote
07:19.04e-HernickDon't you like freenx and vnc ?
07:19.25swm_dont know much about the linux graphical side
07:21.02pauldyswm_, tried piping the display or running vnc
07:21.22pauldyor are you trying to get a remote x sesson via XDMC or domething
07:21.58swm_oh it got it working no problem, somone generated a shitty client version and it donesnt work. Reflection for Windows connects me fine
07:22.15pauldykewl
07:23.25infinity1does anyone here have asterisk connected to gizmo?
07:23.31infinity1..and it works?
07:23.59*** join/#asterisk szer (n=Miranda@217.116.36.22)
07:24.09szerhi all
07:24.16pauldysip phone for macintosh?
07:26.21*** join/#asterisk Mw3 (i=mw3@national.t-error.hu)
07:27.33pauldyinfinity1, are you talking about the sip phone for the macintosh
07:29.01JerJerCaptain Crunch is working on a sip phone for OS X
07:29.20pauldyso what you gota whistle to make it work
07:30.50JerJer2600hz unlock tone to launch the app
07:30.56*** join/#asterisk LANmower (n=LANmower@ndn-165-153-187.telkomadsl.co.za)
07:31.04LANmowerhi there
07:31.07JerJermoo
07:31.33JerJerasterisk is not an operating system
07:31.41pauldyneat I've only seen asterisk as an app until now
07:31.43infinity1pauldy: no.
07:31.49LANmowerthats why I need to make one
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07:32.04JerJerthe fact remains asterisk is not an operating system
07:32.04pauldyinfinity1, ok there are several tutorials on getting it to work
07:32.15infinity1pauldy: gizmo is some free sip voip service
07:32.26JerJerLinux is an operating system - and there are already a dozen micro distros out there suitable for Asterisk
07:32.36infinity1pauldy: well, not totally free, but if you're talking to another gizmo user it is free
07:32.57infinity1pauldy: there is probably a mac client as well. i'm just trying to get it to work with asterisk instead of their client.
07:33.18pauldyI came across a couple of totorials on getting it setup
07:33.34infinity1pauldy: yea. there is a page on voip-info, but i can't get it to work. :/
07:33.47LANmowerjerejer, I have my reasons
07:33.53*** join/#asterisk Akelavlk (n=jansun@82.119.239.141)
07:33.58LANmoweranyway I have 2 qstns
07:34.00LANmowerwhat ap does asterisk use to mail? and how do you set what server mail forwards to?
07:34.30AkelavlkHello guys, I just installed spanDSP + RxFAX and TxFAX applications and it's working properly.. :-)
07:34.40pauldyyour lying
07:35.04pauldyhaha Akelavlk do you have it working over a BRI or SIP
07:35.23AkelavlkBRi.. I don
07:35.33AkelavlkI don't have SIP client yet..
07:35.37AkelavlkDo you have one?
07:36.04pauldyyup tried to get it working finally gave up and setup a hylafax server with a hardware modem
07:36.34pauldyI got one of those damb linspire boxes from frys for 151 so I figured I wasn't out much
07:36.53AkelavlkWhat?
07:37.18AkelavlkYou used SIP hardware phones for faxing?
07:37.30pauldyyea
07:37.50AkelavlkHmm, how much does it cost?
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07:38.17AkelavlkI rather test it with software SIP client . It's cheaper.
07:38.27pauldytotal setup was 200 bucks to build a hylafax server and get the HT386 to emulate a phone line
07:39.03LANmowersnom makes a nice windows-based sip client
07:39.06infinity1if someone has a sec, could they look at this debug and see whats wrong? i'm at a loss
07:39.10infinity1http://pastebin.ca/25817
07:39.36LANmowerif your target is the technically-apt, sjphone is better though
07:39.42AkelavlkMy setup cost 100 $. One hardware fax and spanDSP.
07:40.14pauldywow nice what kind of computer
07:40.17AkelavlkHylaFAX support also SIP FAX-ing?
07:40.42JerJerthere is no such thing as 'sip faxing'
07:40.47AkelavlkYou mean what PC i used for PBX?
07:41.29pauldyI"m going to try and get it working now
07:41.29AkelavlkJerJer, you should check last version of txfax and rxfax application..
07:41.39JerJerthat is called T.38
07:41.43JerJernot 'sip faxing'
07:41.48AkelavlkYes exactly..
07:42.08AkelavlkAs I know, it's FAX extension for SIP.
07:42.11JerJerthen use that term
07:42.16JerJersip faxing means nothing
07:42.27AkelavlkHow do you mean that?
07:42.31JerJerT.38
07:42.35JerJerits not 'sip faxing'
07:42.40pauldyI wsa doing faxes over a sip connection
07:42.47pauldyT.38 was disabled on the ht386
07:43.04pauldyI can't think of what else to call that but sip faxing
07:43.07JerJerT.38 is a codec encapsulation
07:43.13JerJerum T.38
07:43.53AkelavlkYes, but T.38 allow you send fax over SIP as I know..
07:44.08JerJerbut sip has nothing to do with it
07:44.15pauldyJerJer, read what I did if you still think it is T.38 ok
07:44.23JerJeryou can do T.38 over any signaling transport that supports it
07:44.33AkelavlkYes, of course SIP is such as TCP/IP in this case.
07:45.46LANmoweroh well, as usual, no help here
07:45.55LANmowerthx anyway
07:45.57JerJerLANmower:  then leave
07:47.44pauldywell gizmo is a wash not going to bother with something that tells me my username is taken just because it doesn't have numbers in it
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07:49.49AkelavlkHow can I create correct TIFF image for faxing.. I use FAX ccitt 3 compresion but it's not working.. What compresion should I use?
07:50.46*** join/#asterisk Willem_ZA (n=willem@wbs-146-146-209.telkomadsl.co.za)
07:51.03JerJeruse a solution that can deal with PDFs
07:51.47pauldyor postscript
07:51.52AkelavlkHmm, at this time rxfax and txfax support just TIFF....
07:52.26*** join/#asterisk MuppetMaster (n=MuppetMa@169.red-81-184-73.user.auna.net)
07:52.43AkelavlkHylaFAX has also PDF support?
07:52.49MuppetMasterHi
07:54.19MuppetMasterAnyone know the ETA for Asterisk1.2beta to come out of beta?
07:55.43*** join/#asterisk szer (n=Miranda@217.116.36.22)
07:55.47Ahrimanes1.2 Beta1 is out
07:55.59MuppetMasterIndeed, but when is it due to come out of beta?
07:56.06JerJerMuppetMaster: when its ready to come out
07:56.11MuppetMasterie - Generally Availablle
07:57.32MuppetMasterI understand that.
07:57.36MuppetMasterBut any estimates?
07:57.39MuppetMasterWhat is holding it up?
07:58.14AkelavlkMuppetMatster, I am not sure if there are some betas versions. You know it's development.. I am using lastest version and basic functions are working properly..
07:58.53MuppetMasterYes, I have been using the 1.2beta.  The only problems I have had are some crashes on reload at the CLI, other than that functionality is fine.
07:59.50JerJerthen report the crashes
07:59.59JerJeror use a more sane method to reload your config
08:00.04AkelavlkYes, what kind of problems do you have exactly.
08:00.47MuppetMasterCorrect.  I did not report the crashes as there are already appear to be bug reports on the subject.  Don't want to go and open dups.
08:00.59MuppetMasterAkelavik:  Not reproduceable and appear to be somewhat random.
08:01.13MuppetMasterI use Realtime in conjunction with the required text files of course.
08:01.54JerJerno wonder
08:02.26MuppetMasterWhy all of the ill will towards Realtime?
08:02.31MuppetMasterI have seen quite a bit.
08:02.32AkelavlkAha, so problem happend in any config file?
08:02.36MuppetMasterShould Digium have not allowed it into the core?
08:02.47MuppetMasterAkelavik:  Not sure, making changes to the likes of sip.conf and iax.conf.
08:02.54JerJerAkelavlk: yes HylaFAX can deal with PDF
08:02.54MuppetMasterIf I don't do reloads, very stable.
08:03.13*** join/#asterisk terracon (n=tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com)
08:03.20JerJerhow about realtime is simply crap
08:03.38MuppetMasterJerJer:  Then why did Mark Spencer/Digium allow it in the core?
08:03.48MuppetMasterAs opposed to say an add-on?
08:03.56AkelavlkI had quite same problem with sip and zapata.conf. Then I found problem with config parameters..
08:04.01MuppetMasterAnd why is it crap?
08:04.09MuppetMasterHonest questions.
08:04.16JerJerit was disclaimed and mark felt having some sort of database based config was better than nothing
08:04.27JerJerbut has also admitted that it needs serious work
08:04.37MuppetMasterWhat are the key issues with it?
08:04.47Ahrimaneshence still only beta1
08:04.52JerJerdependence on an external system
08:05.05JerJermassive abuse of system resources
08:05.08MuppetMasterWell, that is the definition of database driven configuration, no?
08:05.11AkelavlkAs I remember, there was some problem with parsing of file..
08:05.15JerJerlack of sanity in processing
08:05.17MuppetMasterAh, so takes too much CPU...
08:05.29MuppetMasterSo, may work for a small system but does not scale well.
08:05.53MuppetMasterI do not believe the crashes are related to realtime, but with config file parsing of Asterisk.  As nothing changed in extensions.conf with the crashes occured.
08:05.59JerJerhow about 4 SQL queries per extension priority
08:06.00MuppetMasterAlso, if I stopped and started it worked fine.
08:06.08MuppetMasterJerJer:  I see, that is excessive.
08:06.24AkelavlkMay be you should download lastest Asterisk version..
08:06.45MuppetMasterI am not too worried about it, this is not a production system.
08:06.54*** join/#asterisk _m_ (n=m@nat-ph3-1.rz.uni-karlsruhe.de)
08:07.02MuppetMasterJust curious if there is any expectant ETA on the GA for 1.2...
08:07.59AkelavlkAsterisk is in very early stage, so there are some basic functionality bugs, so I recommend update to lastest CVS version.
08:08.18MuppetMasterAh, too often you catch the CVS mid-bug and can't even compile.
08:08.25JerJerbullshit
08:08.29JerJercvs always compiles
08:08.29MuppetMasterNot bullshit.
08:08.45MuppetMasterEven sent in a bug report which Mark Spencer immediately fixed and apologized for it.
08:08.46AkelavlkI had few problems with ZAPATA ports and parsing config files etc etc.. But last version seems to be pretty stable..
08:08.58MuppetMasterI was impressed, he did it within 10 minutes.
08:09.00JerJerthen re-check it out and it compiles
08:09.04MuppetMasterBut it was not compiling in those 10 minutes.
08:09.05JerJerproblem solved, as you stated
08:09.08MuppetMasterAfter it was fixed.
08:09.09JerJerMOVE ON
08:09.13MuppetMasterBut not during.
08:09.18JerJerit was a mistake
08:09.21MuppetMasterAnd the point is, it does happen.
08:09.22MuppetMasterThat is fine.
08:09.24JerJerMOVE THE FUCK ON
08:09.27MuppetMasterThe point is, the CVS HEAD is a moving target.
08:09.36JerJerand the problem is
08:09.38JerJer?
08:09.47AkelavlkHmm, this should never happend. But you know shit happend..
08:09.49MuppetMasterYou were the one who said it was bullshit.  Simply correcting your bullshit.
08:09.51MuppetMasterNow, carry on.
08:10.45JerJeri'm done
08:10.48MuppetMasterOnce 1.2 is out it has all of the functionality I currently require.
08:10.58MuppetMasterSo, happy to wait for that and suffer some reload issues in the meantime on a lab system.
08:10.58*** part/#asterisk JerJer (n=JerJer@pdpc/supporter/bronze/jerjer)
08:11.23AkelavlkWhat OS do you use?
08:11.45MuppetMasterI use OpenSuSE v9.3 for my production/lab systems.  And also play around on OSX.
08:12.21AkelavlkI am using Fedora Core 2..
08:12.52MuppetMasterAh, I had problems with FC and various stages.  Tried it, didn't like it, so was very happy when SuSE made their Pro version availabe.
08:13.09MuppetMasterI prefer the green lizard to the red fedora...  ;)
08:13.15Akelavlk:-)
08:13.38*** join/#asterisk psk (n=psk@golia.caltanet.it)
08:13.53AkelavlkI know, FC is not such stable, but I think for Asterisk it's quite good platform.
08:14.23MuppetMasterYes, it works.  I just had strange issues at one point (this was a while ago with FC2 and a pre-1.0 Asterisk platform) and when I switched to SuSE those problems dissappeared.
08:14.30MuppetMasterAt that time it was SuSE Pro v9.1
08:14.32pauldyhrm I cna';t even get this pos to auth
08:15.09AkelavlkBTW, what system is Mark using for developing?
08:15.27MuppetMasterNot sure to be honest, I think Redhat though.
08:15.44MuppetMasterThe new O'Reilly Asterisk book is also predicated on installing with Redhat.
08:15.52MuppetMasterSeems to be prevalent in the community.
08:15.55*** join/#asterisk grumpie (n=vijay@dsl093-139-122.sfo4.dsl.speakeasy.net)
08:16.12grumpiehi twisted
08:17.00Akelavlk:-) You know, they use redhat because it's pretty good software..
08:17.08AkelavlkI prefer Gentoo anyway..
08:17.32MuppetMasterYes, it is one of the best commericial distros.
08:17.41MuppetMasterI am telling you, I just think that Red Fedora is one of the ugliest logs around.
08:17.47MuppetMasterGreen lizard much, much better.
08:17.49pauldywow who was asking about gettin gizmo to work with asterisk?
08:17.59MuppetMasterOther than that, they are much of the same.  Although I do like YaST config under Suse better.
08:18.10MuppetMasterpauldy:  Why do you ask?
08:18.22MuppetMasterI use them together, although the Gizmo client does not connect directly to the Asterisk box.
08:18.31MuppetMasterAlso, there is an entry on the wiki that someone kindly put up.
08:19.00*** join/#asterisk BladeRunner05 (n=feelme@adsl-82-213.37-151.net24.it)
08:19.05pauldyyea but it doesn't tell you that you have to use the 1
08:19.18MuppetMasterpauldy:  The ?
08:19.20MuppetMaster1?
08:19.38pauldyyea for your number i.e. 1747XXXXXXX
08:19.52MuppetMasterAh, I see.
08:19.56MuppetMasterWell the Wiki may be updated.
08:19.59MuppetMasterI will do that.
08:20.34pauldyI had to bust open etherreal to see how the app was registering then customize asterisk to it then everything worked
08:21.16MuppetMasterThose details are in the Gizmo knowledgebase.
08:21.30MuppetMasterAs well as on the SIPPhone website, which is the proxy you are actually using with Asterisk.
08:22.38MuppetMasterAdded a little blurb under Outbound Calls:  http://www.voip-info.org/wiki/view/Asterisk+settings+Gizmo
08:23.08MuppetMasterAlso, you can see the way it parses the example of extensions.conf under Outbound SIP Calls that it is providing the 1.
08:23.56AkelavlkWhat is Gizmo project exactly?
08:24.38*** join/#asterisk tessier (n=treed@wsip-68-15-4-13.sd.sd.cox.net)
08:26.03pauldy99-3 erro neat
08:26.56*** join/#asterisk tobiasWolf (n=konversa@195.162.255.10)
08:27.26*** join/#asterisk damned (n=vpol@prior.lanck.net)
08:30.05MuppetMasterhttp://www.gizmoproject.com
08:30.19MuppetMasterA SIP softphone with XMPP for Presence developed by the folks from http://www.sipphone.com.
08:30.27MuppetMasterMeant to be a Skype killer, but not really up to par yet
08:31.32AkelavlkHmm, I see. It seems to be quite same.. Also prices are same..
08:33.30MuppetMasterExcept it is open standards based SIP/XMPP
08:33.37MuppetMasterMay dial SIP URIs
08:33.43MuppetMasterAnd have your Asterisk registry with the proxy as welll
08:33.47snitti like gizmo conference rooms
08:34.19MuppetMasterYes, especially since they also integrated with http://www.freeconferencecall.com
08:34.20pauldywow I never realized what a difference a codec could make
08:34.23MuppetMasterIt is great for large conferences
08:34.40pauldyjust runnig ilbc with gizmo reduces the latency by a ton vs ulaw
08:34.51manyi know its a meta question, but did *anyone* connect a fax device behind a analogue card like tdm400?
08:34.56MuppetMasterYeap, a nice thing.
08:35.00MuppetMasterGizmo also likes gsm
08:35.05MuppetMasterGotta run.  Bye
08:35.07*** part/#asterisk MuppetMaster (n=MuppetMa@169.red-81-184-73.user.auna.net)
08:35.14pauldyme too bed time
08:37.05*** join/#asterisk ianrid (i=hidden-u@213-131-100-29.onyx.net)
08:37.36*** join/#asterisk FreezeS (n=gido_b@82.208.156.94)
08:37.46FreezeShey guys
08:37.55FreezeSI've got a problem
08:38.00FreezeShttp://lists.digium.com/pipermail/asterisk-users/2004-October/067978.html
08:38.11FreezeSaparently it happened to this guy last year
08:38.21FreezeSbut nobody answered then
08:40.03ianridHi, Do any of you guys use an E1 card on a UK BT network?
08:43.59*** join/#asterisk apardo (n=w0w0@49.Red-83-41-11.dynamicIP.rima-tde.net)
08:44.41*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
08:44.44puzzledmorning
08:45.06*** part/#asterisk Akelavlk (n=jansun@82.119.239.141)
08:51.42*** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62)
08:55.05nfi|ermeshi all
08:55.15nfi|ermesi have some problem with nat and xlite
08:55.31nfi|ermesi cant register with asterisk
08:56.13nfi|ermesthe client send request with NAT=pubblic_address and not his address in the subnet
09:26.06*** join/#asterisk Starmaker (n=magnus@85.8.2.169)
09:29.27*** join/#asterisk szer (n=Miranda@217.116.36.22)
09:29.31szerhi all
09:40.55*** join/#asterisk folsson (n=filip@h147n1fls32o985.telia.com)
09:42.46*** join/#asterisk littleball (n=littleba@bb219-75-114-120.singnet.com.sg)
09:46.28Starmakerhi, does anyone know if there is a way to build chan_bluetooth without rebuilding asterisk?
09:47.52manymh.
09:47.56manyi think there is
09:48.34StarmakerI would guess it is possible by just including the headers for asterisk and some bluetooth library and just compiling it?
09:48.42manyhowever you need the source
09:48.50Starmakerof course
09:49.18manywhat i did was t take asterisk source, copy blt into channels/ and then toplevel 'make subdirs'
09:49.34nfi|ermesit's incredible
09:49.46nfi|ermesxlite send a request to a web server
09:49.49manyand then cp the chan_blt.so manually.
09:50.02Starmakernfi|ermes, yeah, I noticed that too
09:50.10StarmakerI denied that using little snitch :)
09:50.16nfi|ermesto know your pubblic ip address
09:50.18nfi|ermesi too
09:50.39nfi|ermesi closed all request to thAT address in my firewal
09:50.49*** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
09:50.52*** join/#asterisk RoyK (n=roy@80.239.107.70)
09:52.19cryzeckRoyk :)
09:52.40manystarmaker: you have to beware anyway, not every blt headset works.
09:53.02manyi even crashed one headset to be FUBAR with chan_blt :)
09:53.37Starmakermany, i was thinking about connecting a cell phone to my asterisk
09:53.44littleballmany, what is the function of chan_blt?
09:53.50manyah. that should make less problems
09:54.07manylittleball: bluetooth. connect bluetooth headsets/handies to asterisk
09:54.25littleballwhat is the special hardware requirement of the asterisk server?
09:54.27manybt hs obviously is a wireless headset then,  bt handy connects asterisk to gsm
09:54.40manys/handy/mobile/gi
09:54.50Starmakersince my operator has free phonecalls within it's network, and I have a pre-paid subscription too, i was thinking about how to get cheaper cell phone calls
09:54.52manygermanisms. :)
09:56.01littleballmany, i think a special hardware needed to hook onto the asterisk server to use your bluetooth headset, right?
09:56.28Starmakerlittleball, a bluetooth interface
09:56.32Starmakerof course
09:56.43*** join/#asterisk DonDonnie (n=don@ip51cd13bc.speed.planet.nl)
09:57.08manylittleball: a BT adapter, my usb/BT dongle works fine
09:57.08littleballStarmaker, can you recommend a device(which i can purchase) so that i can try it out myself.
09:57.15RoyKhi
09:57.22littleballok.
09:57.50*** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net)
09:58.25Starmakeryeah, just about any usb bluetooth dongle works
09:59.04manyiam just pondering how many headsets can be talking at the same time over one dongle.
09:59.18Starmakerdepends on the dongle
09:59.20littleballmany, i want to ask the same question :)
09:59.45Starmakerand of course the implementation
09:59.46manyi was confused since there are only 12 or so RF Channels
10:00.05manyand i thought one rf channel has to be dedicated to one connection
10:00.17Starmakerbut there are dongles that allows like 8 concurrent connections, and there are those that allows only one concurrent connection
10:00.23many(and afaik even some devices are hardcoded to channels)
10:01.51manywell. :)
10:02.34*** join/#asterisk Willem_ZA (n=willem_Z@wbs-146-146-209.telkomadsl.co.za)
10:02.43manynow if only fax device would be working.
10:03.34Willem_ZAhi, could anyone help me with a digium tdm400p?
10:04.04manydepends on what yuor problem is
10:05.11Willem_ZAwell, i just like to know, i have a 3x fxs and a 1x fxo config and.
10:05.44Willem_ZAif i plug a phone into the fxs i do get power on the phone, but no dialtone,
10:06.17Willem_ZAaccording to asterisk everything is installed and running properly.
10:06.21*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
10:07.02Willem_ZAand when i dial the extention, i get a busy tone, but when i call the fxo from my pbx, it just keep ringing.. any suggestions?
10:07.31DonDonnie@Willem_ZA, do you see anything in the asterisk console?
10:07.49Willem_ZAyes, hold on, i will give you an example.
10:08.02DonDonniek
10:08.47Willem_ZADial("SIP/louis-318b", "Zap/1|20") in new stack
10:08.47Willem_ZAOct 18 12:08:21 WARNING[3402]: channel.c:2249 ast_request: No channel type registered for 'Zap'
10:08.47Willem_ZAOct 18 12:08:21 NOTICE[3402]: app_dial.c:1109 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented)
10:08.47Willem_ZA<PROTECTED>
10:08.47Willem_ZA<PROTECTED>
10:08.48Willem_ZA<PROTECTED>
10:08.50Willem_ZA<PROTECTED>
10:08.52Willem_ZA<PROTECTED>
10:08.58Willem_ZA<PROTECTED>
10:09.00Willem_ZA<PROTECTED>
10:09.02Willem_ZA<PROTECTED>
10:09.08Willem_ZA<PROTECTED>
10:09.10Willem_ZA<PROTECTED>
10:09.12Willem_ZA<PROTECTED>
10:09.14Willem_ZA<PROTECTED>
10:09.16Willem_ZAthefly*CLI>
10:09.22Willem_ZAthe ZAP error i did not get the last time..
10:10.08DonDonniewell, obviously there is no Zap/1 channel avaialble
10:11.09Willem_ZAk, will take a look quickly, thanx
10:13.55*** join/#asterisk lters (n=lters@mrtcdsl-034.mis.net)
10:14.23*** join/#asterisk Tili (i=Tili@202-133-67-23-dialup.sat.net.pk)
10:14.28Willem_ZAwhy will this work?:
10:14.39Willem_ZAor not work?
10:14.39Willem_ZAdial 1000
10:14.40Willem_ZA<PROTECTED>
10:14.40Willem_ZA<PROTECTED>
10:14.40Willem_ZA<PROTECTED>
10:14.40Willem_ZAUse EXIT or QUIT to exit the asterisk console
10:14.41Willem_ZA<PROTECTED>
10:14.43Willem_ZA<PROTECTED>
10:14.45Willem_ZA<PROTECTED>
10:14.47Willem_ZA<PROTECTED>
10:14.49Willem_ZA<PROTECTED>
10:14.51Willem_ZA<PROTECTED>
10:14.53Willem_ZA<PROTECTED>
10:15.41Willem_ZAbut should this not acctually ring on the extention?
10:16.16DonDonnieyou could try to make a SIP extension, with a softphone and dial Zap/1 with your Sip softphone
10:16.36DonDonnielike: exten => 100,1,Dial(ZAP/1,20)
10:16.59DonDonnieyou sure Zap is loaded in asterisk? just for the record ;)
10:17.40Willem_ZAhow can i be sure its loaded?
10:18.06X-Rob'zap show channels'
10:18.47DonDonniefrom the cli
10:20.03Willem_ZAok
10:20.09*** join/#asterisk kippi (n=chrisfro@untrust-gct.equinoxit.net)
10:20.11kippihey
10:20.17kippiis there a echo test?
10:21.45X-RobYes.
10:22.11X-Robkippi - at the astersk*CLI> prompt, type 'show applications'
10:22.23X-Robthose are all the toys you that come with asterisk.
10:23.17Willem_ZAok, i ran the command, but i got this error: zap show channels
10:23.18Willem_ZANo such command 'zap' (type 'help' for help)
10:23.18Willem_ZA, so i tried:show channels
10:23.18Willem_ZAand got:Channel              Location             State   Application(Data)
10:23.18Willem_ZA0 active channels
10:23.18Willem_ZA0 active calls
10:23.47X-Robno such command 'zap' means that Zap is not loaded.
10:26.08ianridAny ISDN30 experts about?
10:27.09DonDonnie@Willem, what is the output when you run /sbin/ztcfg -vvvv
10:27.22DonDonniefrom your linux shell
10:28.12Willem_ZAlet me check
10:28.38Willem_ZAZaptel Configuration
10:28.39Willem_ZA======================
10:28.39Willem_ZAChannel map:
10:28.39Willem_ZAChannel 01: FXO Kewlstart (Default) (Slaves: 01)
10:28.39Willem_ZAChannel 02: FXO Kewlstart (Default) (Slaves: 02)
10:28.39Willem_ZAChannel 03: FXO Kewlstart (Default) (Slaves: 03)
10:28.41Willem_ZAChannel 04: FXS Kewlstart (Default) (Slaves: 04)
10:28.43Willem_ZA4 channels configured.
10:29.11DonDonnieok, did you reloaded asterisk?
10:29.20Willem_ZAi did yes.
10:29.22DonDonnierestarted actually
10:29.25Willem_ZArestart now**
10:29.27DonDonnieok
10:29.38ianridWe've had an ISDN30 box installed and the line tests fine with a loop test althoug the service has yet to be activated. What should the status of our E1 card be when we plug this into the box with a stright through RJ45 cable
10:30.32ianridWe have red LEDs by the way withan alarm on the BT ISDN box.
10:30.46DonDonnie@Willem, check this: http://www.digium.com/downloads/hw_article (scroll down a little) it has configuration files for TDM* cards
10:31.20Willem_ZAok, thanx
10:31.40DonDonnieGood luck, I am off for now
10:31.52ianridA looped cable into the ISDN box gives green LEDs. A looped cable into the E1 card gives green LEDs. But a straight through between the two gives red LEDs
10:31.58DonDonnieyour cards are installed well btw
10:32.13DonDonnieas ztcfg shows no errors
10:32.23X-Robianrid - where are you?
10:32.57ianridUK
10:33.50X-Rob~pb
10:33.51jbot[pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.ca/
10:33.58X-Robyour /etc/zaptel.conf
10:37.04ianridhttp://pastebin.ca/25822
10:47.51*** join/#asterisk dreamler (n=bill@195.28.52.162)
11:08.59*** part/#asterisk dreamler (n=bill@195.28.52.162)
11:09.43*** join/#asterisk ful|work (n=fulgas@213.58.130.46)
11:27.50*** join/#asterisk MuppetMaster (n=MuppetMa@62.37.169.84)
11:27.54MuppetMasterHello
11:28.00MuppetMasterIAX and DTMF.
11:28.13MuppetMasterIf it is always inband, how does one use a low bandwidth codec?
11:28.16MuppetMasterLike g729a?
11:28.34*** join/#asterisk rowter (n=SilverDr@201.135.26.195)
11:28.41*** join/#asterisk [Airwolf] (n=airwolf@airwolf.xs4all.nl)
11:30.27*** join/#asterisk hypomanic (n=foo@FC.wayout.net)
11:30.43MuppetMasterAnyone have an idea on low band codecs and DTMF on IAX channels?
11:30.57X-Robyes. They don't work. use inband.
11:31.17MuppetMasterI see that, but then how does one use inband with a codec like g729 which destroys them?
11:31.26MuppetMasterIAX only works reliably with DTMF over alaw/ulaw?
11:31.31MuppetMasterSeems like a real deficiency.
11:31.51X-Robuh
11:31.56MuppetMaster?
11:31.59X-Robthere was a silent 'don't' there
11:32.02X-Rob'don't use inband'
11:32.24MuppetMasterBut, according to the Wiki, it only does inband:  http://www.voip-info.org/wiki-IAX
11:32.24X-Robsorry
11:32.25*** join/#asterisk Talnakh (n=Talnakh@217.22.177.17)
11:32.27MuppetMasterHow to get it out of band?
11:33.08X-Robafaik it _doesn't_ use it inband.
11:33.29MuppetMaster?
11:33.54X-Robit's always oob
11:34.05X-Robwhat's your problem?
11:34.14TalnakhHi all. can when i am trying to call to asterisk though PSTN line, if i hang the phone, asterisk still tries to record a voice mail, leaving a voicemeil of approximately 1 minute with hang tones. how can i fix it?
11:34.32*** join/#asterisk skiold (n=userid@84-121-64-126.onocable.ono.com)
11:34.43X-RobTalnakh - enable busydetect in zapata.conf. It's all documented
11:35.07TalnakhX-Rob, i did that :-(
11:35.32Talnakhok, although i reloaded astrisk, i ll restart the pbx now
11:35.49Starmakercrappy phone
11:35.53Starmakeri have a mt
11:35.54Starmakerot
11:36.01Starmakermotorola a925
11:36.08MuppetMasterX-Rob:  You have confused me.  Does IAX only do inband and therefore can not use codecs like g729.  Or is there a way to get it to do out-of-band so that it may use the lower bandwidth codecs?
11:36.13Starmakerand it won't work with chan_bluetooth
11:36.33*** join/#asterisk kippi (n=chrisfro@untrust-gct.equinoxit.net)
11:36.33X-RobIAX does _not_ use in band signalling.
11:36.41MuppetMasterWhat does it use?
11:36.46X-Roboob
11:36.48X-Robout of band
11:36.49X-Robas I said
11:36.51X-Robwhat is your problem?
11:37.13MuppetMasterMy DTMF on an IAX channel between two Asterisk boxes is not being transmitted effectively using g729 codecs.
11:37.23MuppetMasterFor example, unable to enter a uname/passwd in the voicemail app.
11:37.30MuppetMasterBut if I use alaw, no problem.
11:37.39MuppetMasterAnd then I read the wiki, and it says that IAX does inband ONLY.
11:37.52X-Robno
11:37.54X-Robit sends it inline
11:37.57X-Robnot in band.
11:38.07MuppetMasterWhat is the difference between inline and inband?
11:38.23*** join/#asterisk X-Rob (n=rob@dsl-202-173-151-24.qld.westnet.com.au)
11:38.24X-RobWups
11:38.28MuppetMasterWelcome back.
11:38.28ianridX-Rob, Did you spot anything in our zaptel.conf?
11:38.38MuppetMasterWhat is the delta between inline/inband?
11:38.41MuppetMasterAs the wiki says inband.
11:38.46*** join/#asterisk sivana (n=sivana@mixdown.ca)
11:38.49X-Robthe wiki says INLINE
11:38.50MuppetMasterAnd I do not understand the differentiation.  But I can be dense.
11:39.00MuppetMasterYou are right.,
11:39.03MuppetMasterReading what I wanted to.
11:39.06MuppetMasterBut what is the difference?
11:39.06X-Robinband == as an audio stream
11:39.14X-Robinline == 'the user pushed 2'
11:39.45X-Robit's bad phrasing
11:39.53MuppetMasterCorrect, which can not be used over g729, which destroys DTMF
11:40.08X-Robthere's another problem.
11:40.16X-Robthat's not it.
11:40.39X-Robiax2 does not, repeat, not, encode DTMF as audio at any stage of the transmission.
11:40.57MuppetMasterI see, so it is inline, in terms of using the same UDP stream.
11:41.02MuppetMasterNow I am beginning to get it.
11:41.12MuppetMasterSo why would it be coming through garbled?
11:41.21X-Robthat's the next question.
11:41.27X-Robpossibly something else is encoding it.
11:41.41TalnakhX-Rob, busydetect worked, everything is ok now.
11:42.02MuppetMasterHmmmm.....
11:42.18*** join/#asterisk zotz (n=zotz@24.231.36.100)
11:42.31X-Robianrid - sorry, what's your pastebin?
11:42.40MuppetMasterIf it is IAX2 between the 2, and I can see both are connecting g729.  What else could be encoding?  (BTW - Both machines have g729 licenses)
11:43.06*** join/#asterisk Willem_ZA (n=willem_Z@wbs-146-146-209.telkomadsl.co.za)
11:43.17X-Robthe end device is using g729 and deciding that it wants to send dtmf inband?
11:43.32MuppetMasterNo, IAX2 in Asterisk on both sides, v1.2beta
11:44.30X-Rob'the end device' == 'the thing that is pushing '2''
11:44.36MuppetMasterBut, the endpoint on the Asterisk A intiating the dial is a SIP/Avaya 4602SW IP Phone and then Asterisk A is sending to AsteriskB.
11:44.49MuppetMasterSo, is Asterisk not suppose to translate between SIP and IAX2 seemlessly?
11:45.07MuppetMasterAnd AsteriskB is getting the confusing stuff in the voicemail app.
11:48.26MuppetMasterSo IAX2 does not transmit DTMF if the originator is a SIP endpoint on an Asterisk box?
11:48.35MuppetMasterTransmits, but not properly.
11:48.37X-RobNo. Something else is wrong.
11:48.39X-RobI said that before
11:48.42MuppetMasterThe local voicemail app works on the Asterisk A.
11:48.43X-Robyou'll have to do some debugging.
11:48.50MuppetMasterAh, I see, maybe a bug in the beta.
11:48.57MuppetMasterThanks
11:49.10RoyK~seen coppice
11:49.13jbotcoppice <n=chatzill@123.192.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 17h 55m 48s ago, saying: 'nortel used to be a really great company. something went horribly wrong'.
11:49.19RoyK~seen jesus
11:49.20jbotjesus <i=jesus@84-122-34-139.onocable.ono.com> was last seen on IRC in channel #debian, 54d 17h 34m 49s ago, saying: 'hi all'.
11:49.42X-Rob~seen thelight
11:49.44jbotX-Rob: i haven't seen 'thelight'
11:49.52X-RobMWahahah
11:49.59RoyK~seen elvis
11:50.00jbotelvis <~elvis@9-151.tr.cgocable.ca> was last seen on IRC in channel #kde, 176d 18h 22m 40s ago, saying: 'StevenR, thx a lot bro'.
11:50.06X-Rob~seen A good movie recently
11:50.07jbotX-Rob: i haven't seen 'a good movie recently'
11:50.15Willem_ZAdoes anyone know anything about the chan_zap.so module?
11:50.18RoyK~seen the light
11:50.19jbotRoyK: i haven't seen 'the light'
11:50.27X-RobTHE BAND!!!
11:50.38ianridX-Rob, http://pastebin.ca/25822
11:51.19X-Robianrid - it shuld be 1,1,0... but apart from that it's fine.
11:51.31X-Robwhat card do you have? te110?
11:51.40X-Robuh, span=2, means TE400?
11:52.09X-Robwct4xx even
11:52.16*** part/#asterisk MuppetMaster (n=MuppetMa@62.37.169.84)
11:52.33ianridThe card is a Junghanns Double E1
11:53.26X-Robdon't know anything about them, sorry 8-(
11:56.30*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no)
11:56.35rowteranyone has an experience similar to http://bugs.digium.com/view.php?id=5406?
11:59.51*** join/#asterisk spiekey (n=spiekey@p549D1B88.dip0.t-ipconnect.de)
11:59.54spiekeyhello!
12:00.14spiekeycan i put Asterisk between my telco and my PBX with two isdn cards?
12:04.49TalnakhX-Rob, i hear an echo when i use SIP phone with asterisk. could you suggest which part of config should i dig?
12:05.02Talnakhto get rid of echo
12:08.31*** join/#asterisk FastJack (i=fastjack@2001:8d0:20ff:3:0:0:0:1)
12:09.26*** join/#asterisk victormedrano (n=vmedrano@196.32.128.206)
12:10.44X-RobTalnakh - are you using asterisk 1.2 or CVS-HEAD?
12:11.08Talnakhi use stable
12:11.12X-Robno
12:11.14X-Robyou use 1.0
12:11.31X-Robthe name 'stable' was depreciated about 6 months ago.
12:11.35Talnakhok, i use 1.0.9 i think
12:11.44Talnakh:-)
12:11.58*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
12:12.02X-RobI suggest upgrading to current CVS-HEAD, which will enable the new KB1 echo cancellation
12:12.05X-Robworks well
12:12.24Talnakhok, thanks. i ll think about it. :-)
12:12.50*** join/#asterisk gonzo- (n=gonzo@web.portaone.com)
12:13.43gonzo-Hi everybody. Doas anybody know where i can get PCI specs on AVM Fritz! card?
12:14.02Talnakhtheir website does not have?
12:15.34*** join/#asterisk ToR\L (i=toril@cpe-24-58-23-240.twcny.res.rr.com)
12:15.53ToR\Lquick stupid question
12:16.01ToR\Lthere a way to announce the name before an extension?
12:17.47Dr_Raywhen you call an extension?
12:19.44*** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com)
12:20.06ToR\Lyea
12:20.24Dr_Rayjust put a sound file with their names in your extensions.conf
12:20.32ToR\Lhmm
12:20.48*** join/#asterisk falbala- (i=falbala@bne75-1-81-57-10-154.fbx.proxad.net)
12:21.44*** part/#asterisk falbala- (i=falbala@bne75-1-81-57-10-154.fbx.proxad.net)
12:25.11*** join/#asterisk coppice (n=chatzill@123.192.17.210.dyn.pacific.net.hk)
12:26.45jake1932ToR\L: or if you want to get fancy and the user already recorded their name in voicemail, find out where the VM names are and play them directly from there
12:28.46Dr_Raymmm... fancy
12:36.45ToR\Lyea thats what I was thinking
12:37.26ToR\LI'm asking for a friend... I have voip through home (on asterisk)
12:37.38ToR\Land he setup some an asterisk box for someone
12:37.49ToR\Lso I'm trying to figure out exactly what hes trying to do
12:38.07ToR\LI remember recording my name through ivr for my vm on my box
12:39.52*** join/#asterisk oej (n=Olle@apollo.webway.se)
12:40.12*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
12:41.34jake1932ToR\L: if you haven't found it already, the names are in /var/spool/asterisk/voicemail/[vm-context]/[exten]/greet.gsm (or.wav)
12:44.58*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
12:48.55Talnakhi want to make asterisk server to be online at all times. does anyone know how i can make PC to be always switched on. If lets say there is a powercut, the pc will switch on when electricity appears.
12:49.15jake1932Talnakh: use a UPS
12:49.28FreezeSjake1932: my words exactly :)
12:50.11Talnakhjake1932, FreezeS, bad idea if there is no power for 4 hours.
12:50.35jake1932Talnakh:  http://cableorganizer.com/generators/propane-generators-guardian-plus.htm
12:50.39FreezeSTalnakh: you have 2 solutions. Use a VERY large battery or a generator
12:51.09FreezeSactually, there is a third sollution: use some extremely low power computer
12:51.17FreezeSlike a laptop or something...
12:51.24Talnakhbut can i make linux to switch on automatically?
12:51.31Dr_Rayyour bios might be setable to power on
12:52.02Talnakhon old TX cases unless the switch is off, pc will go back to on when power appears.
12:52.05FreezeSTalnakh, simplest solution: AT power supply :)
12:52.11jake1932Talnakh: you don't have to worry about it turning on if it never turns off
12:52.39*** join/#asterisk tdonahue (n=tdonahue@64.201.13.50)
12:53.01tdonahuegood morning all
12:53.04mutilatorusually the bios will let ya do that anymore
12:53.13Talnakhyes, but motherboard require a different power connector for ATX and TX power supplies
12:53.32Dr_Raymy motherboard has a bios setting to auto power on
12:53.34FreezeSof course Talnakh. Just use an AT compatible mobo
12:54.07TalnakhFreezeS, megalol. it wont be good enough for me now. wit crappy CPU and memory :-)
12:54.49Talnakhbut i think i have an idea how to make POWER supply to be switched on at all times.
12:55.00*** join/#asterisk pif (n=ldm@zenon.apartia.fr)
12:55.01FreezeSTalnakh: well, in this case buy a VERY large battery for your UPS
12:55.08Dr_RayTalnakh - check your bios
12:55.10FreezeSor link several UPS'es
12:55.24mutilator^ wouldn't work
12:55.40FastJacksome bioses have a "power up after power failure" option
12:55.43mutilatorwell it would to an extent i guess
12:55.59FreezeSyou can link many UPS'es
12:56.26FastJackFreezeS: imagine a beowulf cluster of UPSes ;)
12:56.27Talnakhno, i ll shortcircuit a couple of pins on power supply connector. I ll try it now and i ll let you all know if it works. it might be helpful
12:56.59FreezeSTalnakh, since you didn't know about UPS or a generator, I guess you're no electronics specialist
12:57.10FreezeSso I would recommend you DON'T DO THAT !
12:57.14TalnakhU assume too much :-)
12:57.23FreezeSbetter check your bios options
12:57.25Kattymew.
12:57.26*** join/#asterisk Lathos42 (n=Lathos42@adsl-69-208-243-229.dsl.lgtpmi.ameritech.net)
12:57.36KattyLathos42: :>
12:57.36Dr_Raylord forbid you do the easy/correct way first..
12:58.00Lathos42Katty: Good Morning
12:58.01Talnakhi knew about UPS and generator and all that stuff. i just like electronics haking
12:58.15Talnakhhacking i mean
12:58.41pifhi, anyone tried the swissvoice IP10 phone?
12:58.45KattyLathos42: mew.
13:01.07FreezeSwhat happends Lathos42 ? :)
13:01.15tdonahueis there any way to use queues to ring people in the same order every time?
13:01.29FreezeStdonahue, it's very simple even
13:01.39Lathos42FreezeS: I suddenly found myself in #asterisk-unregistered as Lathos42_ :)
13:01.42FreezeSexten => 1,1,Dial(guy1)
13:01.47FreezeSexten => 1,2,Dial(guy2)
13:01.57FreezeSexten => 1,3,Dial(guy3)
13:02.00FreezeSetc...
13:02.26tdonahueFreezeS, What happens when guy1 is on the phone?
13:02.32FreezeSguy2 is called
13:02.37*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
13:02.57tdonahueok, i'll try that out.  thanks
13:03.03FreezeSglad to help
13:03.38Kattythe world is coming to an end :<
13:03.54FreezeSof course Katty
13:04.22FreezeSactually, everything comes to an end eventually :)
13:04.56*** join/#asterisk docE (n=docE@66.237.242.41.ptr.us.xo.net)
13:05.09docEMorning!
13:05.28jake1932not worried so much about the world - just keep my asterisk box up and running
13:05.35FreezeSdocE: It's 4:04 PM in Romania :D
13:05.51FreezeSjake1932, what are you using * for ?
13:05.59RoyKdorstopper.....
13:06.01jake1932i have a few
13:06.06RoyKs/dor/door
13:06.23jake1932one is at a client site as an IVR
13:06.32Lathos42I use asterisk so I can be hip like all the Cool kids
13:06.54Kattysad.
13:07.14FreezeSLathos42: buy an iPod better
13:07.17jake1932i have one here at home so i'm not paying the big money to the bell company
13:08.18Ariel_morning folks
13:08.30Ariel_Katty, it's not coming to an end yet.
13:09.05KattyAriel_: that's not what the JWs are trying to cram down my throat.
13:10.36jake1932that's 242 hugs - should be a while
13:10.49docEya well..  Ill get around to it.
13:14.42ManxPowerUgh.  I really should start packing.
13:15.31Ariel_ManxPower, yes you should.  (just got 10 boxes and will be getting another 10 boxes on Thurday to finish up packing).
13:16.07ManxPowerAriel_, I just need to pack up the laptop and a suitcase.
13:16.08*** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com)
13:16.38Ariel_ManxPower, nice. I have a 4 bedroom house to pack. Then get a storage unit to store the stuff then move into a 2 bedroom appartment.
13:17.12ManxPowerAriel_, Ah.  I'm heading to Covington/Gulfport this week.
13:17.16Ariel_Sold the house in 3 days with just a sign out front. (Strange I feel I asked way to low).
13:17.19X-RobKatty - kick the JW's in the nads.
13:17.32*** join/#asterisk pointer (i=pointer@64.18.103.6)
13:18.00KattyX-Rob: I shall not.
13:18.03KattyX-Rob: She is my mother.
13:18.10X-RobBugger.
13:18.11jake1932hah
13:18.12X-Robyou poor thing.
13:18.22KattyX-Rob: hardly.
13:18.26KattyX-Rob: I learned to think.
13:18.43KattyX-Rob: If I hadn't, I'd still be a good little JW girl.
13:18.59jake1932does she still come to your door with a booklet?
13:19.19X-RobKatty - onya.
13:19.19Kattyjake1932: no, she pesters me everytime I visit.
13:19.23FreezeSwhat's JW ?
13:19.37X-RobKatty - tell her you've deicded to worship the FSM
13:19.44FreezeSFSM rule !
13:19.44KattyX-Rob: Stopping fixing me.
13:19.59KattyX-Rob: I didn't ask for help or counsel.
13:20.00FreezeSoh, Jehova's Whitnesses
13:20.15X-RobKatty - sorry, I'm just trying to be humourous.
13:20.16X-RobGeez.
13:20.21Kattyk
13:20.42FreezeSKatty, do you know what FSM is ?
13:21.05KattyFreezeS: of course.
13:21.19FreezeSok, I only found out about 2 weeks ago
13:23.11KattyAriel_: that's what google is for.
13:23.58FreezeSAriel_, it's the TRUE religion
13:24.13FreezeSpeople finally found out the REAL TRUTH
13:24.14X-RobFreezeS - that's TRUE(*) religion
13:24.24X-Rob(*) where true == as true as any
13:24.38Talnakhhi all, i am back to life
13:24.39Dr_Raywell, as untrue as the all are
13:24.41Ariel_well I got two main ones from google the Flying Spaghetti Monsterism or the Free Speach Movement.
13:24.56Talnakhi just got a really bad electric shock.
13:25.16*** join/#asterisk pigpen2 (n=mark@fw.seamans.cc)
13:25.27FreezeSX-Rob, there is only ONE truth
13:25.34jake1932Talnakh: didn't your parents ever tell you not to mess with electricity
13:25.38FreezeSand finally the people found it
13:25.51Dr_Raythere is no dog
13:27.19Talnakhah, i am just kidding. but remember i said that it is possible to keep ATX motherboard powered on without BIOS settings. well, it actually works. my mobo does not support to switch on using bios.
13:27.33Ariel_X-Rob, truth is that they all lie, they all have secrets and you can't belive any of them.
13:28.23FreezeSAriel_, have you found the TRUE religion ?
13:29.28Ariel_FreezeS, yes I do belive.
13:30.55pointerthis is a silly newbie question, but...when you change the mp3s in your MOH dir for a context, how do you you tell asterisk to restart/notify the mpg123 threads?  will a reload work or will it require a restart (when convenient)?
13:30.56FreezeSone more pastafarian joins the Holy Church
13:31.35FreezeSpointer: musiconhold.conf
13:31.36iCEBrkrpointer: I'd just make sure all instances of mpg123 are dead.  Then when Asterisk fires it back up, it'll collect the new filenames.
13:32.11pointerFreezeS: I know what config file to edit...in this case, I shouldn't have to touch it though...
13:32.31FreezeSoh, I misunderstood the question
13:32.39pointeriCEBrkr: so it only fires up mpg123 threads when a channel is put on hold?
13:32.46iCEBrkrpointer: Yeah
13:33.07iCEBrkrpointer: But sometimes there will be two mpg123 processes out there running even though no one is on hold.
13:33.14iCEBrkrSo just be sure to kill them off.
13:33.33pointeriCEBrkr: ah, ok....didn't realize that.  Can I safely kill running mpg123 processes if I know that nobody is on hold?
13:33.34*** join/#asterisk bob_too (n=chris@rrcs-24-153-179-246.sw.biz.rr.com)
13:33.44iCEBrkrpointer: Sure
13:34.09pointeriCEBrkr: thanks, I really didn't have time to test that one and risk the worst
13:34.15iCEBrkr:)
13:38.30*** part/#asterisk pointer (i=pointer@64.18.103.6)
13:39.47docESay anyone know anything about static agents in app_queue or could point me somewhere to find the info.  I tried the WIKI and Google with no information
13:40.23RoyKstatic agents?
13:40.31RoyKwaddayamean?
13:40.44FreezeSdocE, you could just use lines instead of agents
13:40.49FreezeSlike sip/user
13:40.49iCEBrkrRoyK: They all have afro's from the static electricity
13:40.53FreezeSor zap/1
13:40.55Ahrimaneslol
13:41.04RoyKSIP/user
13:41.06RoyKnot sip/user
13:41.06RoyK:P
13:41.38skyenwhat's wrong when debug says "Cannot cretate channel of type SIP", when the call is patched throuh?
13:41.43FreezeSRoyK: hopefully, he got the idea
13:41.57skyeneverything works as a charm, but that errormsg keeps appearing
13:42.17*** join/#asterisk pigpen2 (n=mark@fw.seamans.cc)
13:43.04FreezeSskyen: maybe the problem is on the client part. Are they registered on the server ?
13:43.15*** join/#asterisk gambolputty (n=gambolpu@72.240.241.108)
13:43.33*** join/#asterisk jimmy_deanPB (n=jhodapp@72.244.232.226)
13:44.32skyenFreezeS: yes
13:44.57FreezeScan they call ?
13:45.06skyeneverything works.
13:45.12skyenjust complaints in the log
13:45.21FreezeSwhen do these messages appear ?
13:45.28skyenjust as the call is patched
13:45.34skyenwhen the phone starts ringing
13:45.55docEWell here is the synario of what I have now..  its working but doesnt work when the system reboots.   I have 2 SIP extensions not registered to asterisk so that when a call is in queue and its routed to one of the end points it dials a SIP URI and sends the call of.  How do I make it so users dont have to actually login to the queue?
13:46.47FreezeSdocE: you need the extensions to be registered to asterisk
13:47.09FreezeSwhat telephone are you using for the agents ?
13:48.18docE2 Cisco 3660's
13:48.20docE:)
13:49.04Ariel_I really think that the press is evil. The just want to report doom and gloom.  There trying to make everyone think that this Dandemic for the bird flue is here or close to being here in the states. argh
13:49.21*** join/#asterisk nomazda (i=nyyankee@user-0c6tnqf.cable.mindspring.com)
13:49.44docEThe 1 Cisco 3660 accepts SIP and is routed into a NEC PBX via CAS signaling..   The other sits in Melbourne, AUS and routes to a DID on the PSTN
13:50.07jake1932damn it
13:50.13Ariel_skyen, are you using canreinvite=yes on your system?
13:50.32docEDamn what?
13:50.34skyenno, all clients are forced to no on that option
13:50.54jake1932i was on a call and the guy i was talking to (SIP to SIP) got another call, he couldn't hear me after the call waiting beep
13:51.24docEAsterisk call?
13:51.28jake1932yep
13:51.42docEinteresting..
13:51.54docEWhat version?   I am using head and dont have the issue with call waiting
13:52.00skyenAriel_: could that be my problem?
13:52.18jake1932HEAD as of 9/15
13:52.31Ariel_skyen, no it should not be. If it works just leave it along in my view.
13:52.49Ariel_jake1932, head has had allot of fixes since then.
13:52.56jake1932ok - i'm upgrading
13:53.21Ariel_jake1932, backup first
13:53.29jake1932will do - tnx
13:53.44*** join/#asterisk synthetiq (n=roger@64.201.13.50)
13:54.01synthetiqwhat other options than SER do i have for load balancing on cloned machines
13:54.03Ariel_why did Mediatix make the 1204.. I know to make my life a living hell. Yes that is it.
13:54.48Ariel_cloned machine... hum ... hartbeat for starters.  hummm look at the wiki for more info
13:54.52*** join/#asterisk ikey (i=ikey@220.226.13.53)
13:56.26*** join/#asterisk trym (n=trym@cD9088B17.sdsl.catch.no)
13:57.07*** join/#asterisk paryl (n=paryl@209.236.78.59)
13:58.01trymwhat was the name of that handy app that converted sound files into .gsm again?
13:58.18trymsox
13:58.22Ahrimanestrym: sox?
13:58.25trymindeed
13:58.29*** join/#asterisk mkrufky (n=mk@68.160.103.77)
13:59.28*** join/#asterisk KriS83 (n=KriS@212.202.141.92)
13:59.36KriS83Hi
13:59.39*** join/#asterisk lars-- (n=lars@lars.debian.us)
13:59.43Kattyhi.
13:59.45*** join/#asterisk joelsolanki (i=joelsola@202.160.161.93)
14:01.02*** join/#asterisk wmandra (n=me@pcp04943183pcs.verona01.nj.comcast.net)
14:01.07wmandramorning all
14:01.37wmandrais anyone else having trouble logging into the wiki???
14:03.33Kattymorning.
14:05.05trymfile.c:492 ast_openstream_full: File velkommen does not exist in any format
14:05.13trymfile.c:804 ast_streamfile: Unable to open velkommen (format ulaw): No such file or directory
14:05.15trymim very sure it exists
14:06.57*** join/#asterisk Meaty (n=cp_simbu@office.abi.ca)
14:07.09szerknows anybody something from bug 5442?
14:08.28*** join/#asterisk pigpen2 (n=mark@fw.seamans.cc)
14:08.53pskanyone with problems sending faxes with asterisk and getting T4 timeout?
14:10.23IronHelixwmandra- try logging out on every machine/browser you are logged in with, then login again
14:10.34paryli've got a wierd problem... when a call comes in from an analog line, i have it ring all SIP phones (gxp-2000's)... when one person is on line1, the incoming call rings to their second line, but the other phones ring on line1 AND line2.. i can't figure out why
14:11.17IronHelixthat is very odd
14:11.25IronHelixand if nobody is on the line, then only line1 rings?
14:11.26jake1932does line 1 support call waiting?
14:12.53*** join/#asterisk clennon (n=clennon@dialup144.ts009.bmt.esat.net)
14:13.15KriS83I need a hint.. I have just compiled * beta 1.2.0 because I need the mysql cmd feature. Everything seems to work fine. Also compile chan-capi (using a AVM B1 card) set incomingmsn=27 in /etc/capi.conf but when I call my MSN 27 I get the following: Oct 18 13:51:38 NOTICE[15395]: pbx.c:1688 pbx_extension_helper: Cannot find extension context 'capi-in'
14:13.16KriS83Oct 18 13:51:38 ERROR[15395]: chan_capi.c:2012 start_pbx_on_match: ISDN1: did not find exten for '27', ignoring call.
14:13.48IronHelixkris- you have capi setup right, but maybe not extensions.conf
14:13.53JuggieKriS83, you essentially answered your question
14:13.57Juggiecreate a context in extensions.conf
14:14.02IronHelixcalled capi-in
14:14.06Juggie[capi-in]
14:14.09Juggieinside it
14:14.14Juggieadd your extensions
14:14.18Juggieeg
14:14.23KriS83ok so If I renamed [demo] to [capi-in] it should work right?
14:14.26Juggieexten=> 27,1,Answer
14:14.43Juggiepossibly... but your not going to learn anything that way
14:15.02Juggietry
14:15.06Juggie[capi-in]
14:15.08IronHelixkris- yup, or you could make a [capi-in], and then put exten =>27,1,Goto(demo,s,1)
14:15.29KriS83Right I'll try that
14:15.30Juggieeither way, your missing the context
14:15.35Juggiewhich is why its broken
14:17.52KriS83works fine ;)
14:18.32joelsolankiHello anybody used the http://voip-info.org/wiki/view/Asterisk+CDR+csv+mysql+import to import the text cdr from /var/log/asterisk/**
14:19.25KriS83one more thing... When I create a IVR and add exten => 27,4,wait(3.0) for example because I want to have a Pause in the Text, in this pause the caller can not press any keys? is that correct?
14:19.29*** join/#asterisk ikey1 (i=ikey@220.226.23.83)
14:20.16IronHelixas i recall it depends on what is happening
14:20.33IronHelixlike if you do ringing() then wait() in my experience it doesnt hear dtmf
14:21.04IronHelixbut if you just wait() it might
14:21.06KriS83IronHelix, what I'm doing is just playing like for TEchnical support -> wait
14:21.10KriS83press 1
14:21.16KriS83wait(3.0)
14:21.26*** join/#asterisk jsiddall (n=jsiddall@206-248-134-222.dsl.teksavvy.com)
14:21.29KriS83for sales -> wait(0.3) press 2
14:21.30tzafrir_laptopjoelsolanki, CSV is CSV, it is data. Filter it to your favorite format
14:21.32IronHelix3 seconds is a lot of time on the phone
14:21.41KriS83IronHelix, I know was only an example
14:21.44IronHelixah
14:21.52KriS83but I have like 1.5
14:22.10KriS83and when pressing 1 in this 1.5 second period it is not accepted
14:22.14*** join/#asterisk dalabera (n=Dalabera@pmr.pmrtechnologies.com)
14:22.19IronHelixgive it a shot, make something like background(hello) wait(30) and see if it will listen to dtmf
14:22.42joelsolankitzafir_laptop: not able to understand u. I need to convert csv cdrs in to mysql and put into cdr tables. so that i can be able to do billing.
14:22.51IronHelixthen maybe another way to do it (ugly hack) make a 1.5 second long blank gsm file and background() it
14:23.02joelsolankiCan u please give some suggestion on this.
14:23.23IronHelixahhh
14:23.25KriS83IronHelix, ok...
14:23.27IronHelixyou want WaitExten()
14:23.29IronHelixthere you go
14:23.32IronHelixthat will wait and listen
14:23.39KriS83Ok thx
14:23.51KriS83Put that down on a piece of paper ;)
14:23.57wmandraironhelix: closing all browser windows and trying to login again didn't work (couldn't log out, cause I was never logged in) I even tried to create a new account, got the registration email, clicked the link, entered a new password, no joy - back to the homepage without being able to log in.
14:24.01*** join/#asterisk Anthro (i=gss@pdpc/supporter/active/Anthro)
14:24.14IronHelixtry clearing all cookies?
14:24.26ManxPowerasterisk-sounds has silent .gsm files of various lengths
14:24.54KriS83IronHelix, now the only thing I'll be fighting with again is Passing my Calls from Line 1 (incoming) to Line 2 -> MSN 17 for example (via CAPI)
14:25.12IronHelixi suggested that cuz its done this a few times, but I use both IE and firefox.  sometimes it wont log me in on IE, then i just logout in firefox and it works again
14:25.27IronHelixyou just need to dial()
14:25.35IronHelixget the right channel and you should be set
14:26.05*** join/#asterisk [Airwolf] (n=airwolf@airwolf.xs4all.nl)
14:26.18KriS83IronHelix, I did that yesterday (with the old 1.0.6 install) and It connected to 17 but so short that it didn't even ring
14:26.29IronHelixany errors in the log?
14:26.37KriS83I only saw it on my display ->Missed calls
14:26.58KriS83This morning I found a capiHOLD and capiEXT function
14:27.24*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
14:27.27KriS83this worked with one minus point... it always told me the number of the calling person before giving me the call on 17 ;)
14:27.29KriS83No no error
14:27.42KriS83But I'll try again now with the fresh install
14:28.05AnthroI would like a box with Asterisk, a place to plug in my home's phone wiring (RJ11, I assume), a place to plug in ethernet (RJ45), and a nice web interface to configure the more common aspects (SIP<->POTS provider information, mapping phone numbers to SIP addresses, etc.). Does such a thing exist? If not, would there be an interest in it once I develop it?
14:28.10IronHelixmaybe then the dial plan isnt doing anything interesting with it?
14:28.19IronHelixanthro- it does
14:28.32AnthroIronHelix: URL?
14:28.36IronHelixyou need an old pc, a copy of asterisk@home, and a digium TDMxx board
14:28.44jsiddallIs there any solution yet for the choppy audio when using ztdummy on the latest FC4 2.6.13 kernel?
14:28.48wmandraironhelix: go figure, IE works fine - just can't login with FF
14:28.57IronHelixhttp://asteriskathome.sourceforge.net/
14:29.54IronHelixhttp://www.voipsupply.com/index.php?cPath=99_103  FXO ports connect to phone lines, FXS ports connect to phones (like your existing phone handsets)
14:29.59AnthroIronHelix: I'm thinking of a turnkey solution that people like my parents would be able to install on their own. Something as easy to plug in and start using as a Linksys router.
14:30.10synthetiqi have an interesting situation...and am lookign for suggestions.... we have 3 cloned servers which we want to load balance.....the issue is, what if i need a two phones to make calls to each other when the two phones are registered on different machines?
14:30.38synthetiqwith out going to out termination provider and back in
14:30.43IronHelixsyn- use realtime?
14:30.50IronHelixif there is a single database it can store registrations
14:30.53IronHelixor replicate the db
14:31.16IronHelixanthro- its hard to make * linksys simple without ripping out most of it
14:31.25synthetiqwe will be using realtime
14:31.34AnthroIronHelix: Ripping out or just setting up plausible defaults?
14:31.40IronHelixAAH certainly isnt linksys simple, you need to understand a bunch of stuff
14:31.50IronHelixboth
14:32.14IronHelixsyn then store the registrations in a db that all the servers get access to, so a registration will be valid on any server
14:32.34trymI cant seem to convert my .wav to .gsm and make it work in asterisk
14:33.01jontowtrym; have you seen the wiki page on that topic?
14:33.18trymhmm on sox and gsm yes
14:33.21trymim using the same options
14:33.54*** join/#asterisk yartelecom (n=no-email@62.33.183.215)
14:34.08AnthroIronHelix: Mm. Well, I intend to learn what I need to know. Hardware-wise, is a middle of the road VIA box with one PCI slot going to be sufficient for the task? I haven't bought any hardware, so I'm aiming for low cost, small size, and low power consumption/heat dissipation.
14:34.53IronHelixyou'll need to both rip out much of the un needed stuff (few usa users will want capi, for example) and set defaults for the rest (voice mail etc)
14:35.00trymah lol
14:35.00trymnm
14:35.08IronHelixvia should be nice but check the incompatibility lists first.
14:35.25IronHelix1 pci slot will accomidate up to 4 POTS lines, either fxo or fxs depending on how you load the card
14:35.33*** join/#asterisk ayano_ (n=erik_lee@68-117-160-098.static.chtn.wv.charter.com)
14:35.34IronHelixcheck the voipsupply link i posted above
14:35.40AnthroIronHelix: Which incompatibility lists?
14:35.47IronHelixyou can get a digium card with any combination of fxo or fxs modules
14:35.47*** part/#asterisk ayano_ (n=erik_lee@68-117-160-098.static.chtn.wv.charter.com)
14:36.17*** part/#asterisk case__ (n=case@mailhost.seeft.com)
14:36.19IronHelixthere's a list on voip-info.org
14:36.26IronHelixmostly old Dell systems
14:36.34IronHelixas long as the card can get a free IRQ you should be fine
14:36.47synthetiqironhelix how do i go about storign registraions realtime
14:36.52IronHelixdigium cards generate tons of irq traffic
14:37.43*** join/#asterisk l1nux (n=moi@lns-bzn-4-82-250-119-242.adsl.proxad.net)
14:37.49l1nuxhi
14:38.00*** join/#asterisk MGSsancho (n=user@ppp-67-126-240-180.dsl.irvnca.pacbell.net)
14:38.03synthetiqdell machines lvoe to sue the same irqs for ...everything
14:38.05AnthroIronHelix: I see. What kind of power consumption/heat dissipation should I expect from a Digium card? They aren't like modern video cards that need their own fans or anything, right?
14:38.05synthetiquse
14:38.35synthetiqthe digium cards will melt your heatsink :-P
14:38.43*** join/#asterisk LostFrog (n=really@69-174-51-210.chvlva.adelphia.net)
14:39.04*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
14:39.09Anthrosynthetiq: Eh? That's not so good. So a fanless system is out of the question?
14:39.20l1nuxhowto add "busy tone" if  "SIP response 404 "Not Found" ?
14:39.28IronHelixhmmm, syn not sure about realtime registrations
14:39.31IronHelixi had thought it was possible
14:39.33LostFrogIs there anything short of the SNOM 4S ITSP that you can put in front of asterisk to support SSIP/SRTP?
14:40.10LostFrogLike, does the 4S Proxy support SSIP/SRTP?
14:40.15spiekeycan i put Asterisk between my telco and my PBX with two isdn cards? is that possible?
14:40.20IronHelixsure
14:40.25IronHelixer
14:40.34IronHelixi dunno if an isdn card can 'serve'
14:40.44IronHelixbut why would you want to do that?
14:40.55MGSsanchonot enough bw
14:40.57MGSsanchoi think
14:41.52ManxPowerisdn bri or isdn pri?
14:42.33*** join/#asterisk lehel (n=asd@82.79.20.17)
14:42.57IronHelixhehe
14:43.05IronHelixi think i was the only US user of ISDN
14:43.14AnthroIronHelix, synthetiq: Do the Digium cards really run that hot?
14:43.25synthetiqim joking man
14:43.36iCEBrkrIronHelix: Are you kidding? ISDN sold pretty good in NE. Ohio--  If you were a telco guy or geek.
14:43.37ManxPowerIronHelix, you are 8-)
14:43.38synthetiqthey dont run hot at all
14:43.44Anthrosynthetiq: Ah, good.
14:43.48MGSsanchostill works with the 8 switches on it
14:44.00iCEBrkrI used ISDN for about 5yrs.
14:44.08IronHelixim not anymore, had it for a few years before cable came out
14:44.14MGSsanchogood times
14:44.21IronHelixback then it was the shit, connect in like 3 seconds and WOW 64 REAL KBPS!
14:44.25lehelhello
14:44.30IronHelixhi
14:44.32MGSsancholol
14:44.35MGSsanchohi
14:45.07iCEBrkrIronHelix: I liked the fact that I could set a threshold on my Cisco ISDN router for more bandwidth and then have it drop the second line when I was done leeching
14:45.15IronHelixhttp://www.digium.com/index.php?menu=compatibility   digium card compatibility notes
14:45.21iCEBrkrOr if a call came in, drop the second channel and ring my phone
14:45.29IronHelixyeah multilink was cool, i never got it to work tho
14:45.36IronHelixi think my isp didn't support it
14:45.41iCEBrkrAhh
14:45.51MGSsanchoAOL doesnt :( i called
14:45.59iCEBrkrI worked for an ISP at  the time.  I was kinda a test case.
14:46.07MGSsancho>_>
14:46.30clennonhi
14:46.38*** join/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net)
14:46.38iCEBrkrMy first ISDN terminal adaptor was this Hayes Modem.  Sucked.
14:46.39MGSsanchonow if only they mde T1 the same price as cabel or dsl
14:46.40*** join/#asterisk galel (n=galel@63.245.93.138)
14:46.45MGSsanchoor even $40 a month
14:46.50IronHelixi got a nastygram from my isp once complaining that i was dialed in too much
14:46.56IronHelixyeah cheap pri would rock
14:46.57*** join/#asterisk galel (n=galel@63.245.93.138)
14:47.04IronHelixeven if it only had 2-3 channels
14:47.17*** join/#asterisk copantl (n=galel@63.245.93.138)
14:47.24*** join/#asterisk copantl (n=galel@63.245.93.138)
14:47.38AnthroIronHelix: So Asterisk@home is a web administration interface, basically?
14:47.43MGSsanchoi cant find drivers for my old hayes supermodem
14:47.44MGSsanchoyup
14:47.46IronHelixno
14:47.50iCEBrkrDrivers?
14:47.53MGSsanchoand it automates a lot of stuff
14:47.55spiekeyermm..sorry, had to grep something to eat...
14:47.59iCEBrkrSince when does a serial device require drivers :P
14:48.00spiekeyi use bri ISDN
14:48.01MGSsancholol
14:48.01IronHelixasterisk@home is a distribution of linux which includes a web interface, asterisk, and a bunch of other stuff
14:48.17IronHelixit includes Asterisk Management Portal (AMP) which is the web interface
14:48.24spiekeya normal german ISDN line with two "channels"
14:48.25IronHelixit is (relatively) easy to configure
14:48.26AnthroIronHelix: Ah, I see. I was confused.
14:48.30iCEBrkrWindows made modems more difficult to deal with/install
14:48.40*** join/#asterisk heison (n=heison@ns.somanetworks.com)
14:48.43IronHelixno worries
14:48.47IronHelixyeah i had hayes too
14:48.50LostFrogNo help on SSIP/SRTP??
14:48.56IronHelixcourier i-modem isdn with v.34
14:49.01IronHelixlost- its hard
14:49.03MGSsanchoT_T
14:49.08IronHelixthere is no real srtp support for * atm
14:49.13iCEBrkrSpeaking of which, I should get this PRI hooked up to my asterisk box to start configuring it.
14:49.13IronHelixthere are plenty of bounties for it
14:49.16LostFrogIronHelix, that I know..
14:49.28LostFrogI would be willing to put another software packages in front of it.
14:49.29IronHelixiax2 has some sort of encryption routine but i dont know how well developed or secure it is
14:49.35IronHelixand thats really only for site to site
14:49.52jake1932Anthro: if you're confused already - this is nothing
14:49.57LostFrogI want to secure my work-from-home users using snom phones.
14:50.11IronHelixheres another idea
14:50.12LostFrogWithout out using IPsec to secure their whole networks.
14:50.20IronHelixget them all Linksys wrt54g's
14:50.28IronHelixload linux on them (sveasoft, openwrt, whatever)
14:50.31MGSsancholol
14:50.46IronHelixset them as pptp clients, and instruct the users to plug only their work computer and phone into them
14:50.57IronHelixdeny phone registrations from the Internet, only allow from behind pptp
14:51.10IronHelixclumsy, but works and lets them secure other stuff too
14:51.25AnthroI don't know a whole lot about phone systems (yet). The wiring in my house is designed to work with POTS, of course. Is there a wire somewhere in the house that I can plug into a port on a Digium card to provide phone service to all of the extensions in the house?
14:51.31LostFrogHmm.. I thought about that.
14:51.37IronHelixanthro- anywhere
14:51.44IronHelixpots phone wiring is a loop
14:51.57IronHelixthat means any port is electrically the same as any other port
14:51.59LostFrogAnthro, make sure you disconnect the connection from your phone service, first.
14:52.06IronHelixjust make VERY VERY SURE that you unplug your telco
14:52.18LostFrog75-90 Volts to a FXS port can suck.
14:52.18IronHelixbecause ringing voltage from the telco will possibly fry something expensive
14:52.26iCEBrkr120v DC.
14:52.30MGSsancho50V usealy
14:52.34MGSsanchoin LA
14:52.35LostFrogno, ringing is AC.
14:52.36iCEBrkr50?
14:52.41AnthroIronHelix: So I could have my Asterisk box in my office, plug it into the phone extension there, and provide service to the entire house? Sweet!
14:52.46jake1932i got a shock from a ring before - not fun
14:52.49MGSsanchomy lines are 48V and 49.2 at home
14:52.52iCEBrkrUnless they changed it.. It's DC
14:53.01*** join/#asterisk scoates (n=sean@iconoclast.caedmon.net)
14:53.14IronHelixanthro- yes, sort of
14:53.14AnthroIronHelix: At the moment I have no landline service.
14:53.16scoatesanyone know of a mirror for app_conference? sourceforge's CVS isn't playing nice.
14:53.19IronHelixdepends on what you are trying to do
14:53.22LostFrogok, iCEBrkr, that explains why they have sine generators in FXS channel banks.. :)
14:53.47AnthroIronHelix: I basically want all calls to route through SIP, but I want the interface to be identical to POTS from my wife's perspective.
14:53.58IronHelixyou dont need asterisk to do that
14:54.05IronHelixbut you can use it
14:54.13AnthroIronHelix: What would you use, if not Asterisk?
14:54.39IronHelixthe cheapest way- sign up with any voip provider, they will ship you a little box called an ata (analog telephony adapter).  it has ethernet on one end and 1-2 fxs ports on the other
14:54.41iCEBrkrhrrm.
14:54.50*** join/#asterisk litage (n=nick@203.220.55.70)
14:54.50iCEBrkrGrandpa's memory is going..
14:54.51iCEBrkr<==
14:55.02IronHelixdisconnect the telco interface, plug the ata's phone port into the wall
14:55.02LostFrogStupid question: I shouldn't have any problems with assisted transfer with snoms and Asterisk, in regards to having control of the call if the remote extension doesn't answer, should I?
14:55.11*** join/#asterisk [Airwolf] (n=airwolf@airwolf.xs4all.nl)
14:55.15IronHelixyour wife won't know the difference, she will get dialtone, ringing, etc etc
14:55.17nexisi hate to spit it out in here, but is anyone else seeing problems with goiax today?
14:55.50IronHelixi suggest get a provider that supports BYOD (bring your own device) so if you decide to upgrade to * later you can
14:55.59AnthroIronHelix: Hrm, okay. That sounds pretty good. Is there a place with a good list of providers?
14:56.04scoates<PROTECTED>
14:56.05LostFrogMmmm.. broadvoice. :)
14:56.07IronHelixby putting the SIP settings from the adapter straight into the asterisk box
14:56.07Kattyhmm.
14:56.12nexisyea, vonage is evil, except they helped me get my hands on a grip of ATAs for about 2 bucks each.
14:56.15kiko69Earthlink
14:56.19IronHelixi recommend broadvoice or quantumvoice if you are in the USA
14:56.23nexisnufone is my choice.
14:56.35jake1932callvantage is good but expensive
14:56.37IronHelixquantumvoice especially, their webiste looks like ass but they have a good forum and always answer emails within a day
14:56.41iCEBrkrhttp://www.textfiles.com/phreak/black3.txt
14:56.43IronHelixcant say the same for broadvoice
14:56.44synthetiqi dont think asterisk registers phones via realtime
14:56.50iCEBrkrTHOSE where the days
14:56.55nexisIronHelix, whats the pricing like on quantum?
14:56.59*** join/#asterisk power1 (n=marktren@rndf-146-36-59.telkomadsl.co.za)
14:57.05AnthroIronHelix,nexis: I'll look into them, thanks!
14:57.09IronHelixnot too bad, 25/mo unlimited usa as i recall
14:57.09LostFrogWhy can't linksys put the the version numbers of their products on the packaging???
14:57.11Nuggetjason's "bbs documentary" dvd set is amazing.  buy it.  :)
14:57.11heisoncan anyone recommend a good TTS software that works well with Asterisk and supports Cantonese?
14:57.41nexisyea, nufone is 2c a min outgoing/incoming 800, and a 25 one time setup for a MI DID
14:57.45IronHelixlostfrog- i wish they'd stop with version numbers all together
14:57.55nexisbut you get full control over your call with setcidnum ect.
14:58.02kiko69Earthlink 14.95 for 500 minutes, or 24.95 unlimited
14:58.07synthetiqspeakign of phreakign i know  way to make free voip calls and avoid keepign registration on a server
14:58.09synthetiq=]
14:58.16IronHelixhehe
14:58.25nexissynthetiq, you mean free outgoing or incoming?
14:58.26power1Hey all, Ive just got asterisk @ home 1.5 up and running. everything is working correctly except for voicemessages going out as emails, where do i tell asterisk to use my internal mail server as its outgoing smtp?
14:58.26synthetiqcdrs dont get written
14:58.30*** join/#asterisk mogorman (n=mogorman@gateway.digium.com)
14:58.32synthetiqoutgoing
14:59.44*** join/#asterisk hypa7ia (i=hypatia@wsip-24-234-241-145.lv.lv.cox.net)
14:59.46nexiswell, i suppose you could put like dial([IAX2|SIP]/username:password@host/exten) if they support that.
15:00.03nexisand just never register
15:00.10LostFrogYou don't have to register for outbound calls.
15:00.21IronHelixpower1- use the webmin console and configure sendmail to use a smart relay host
15:00.44IronHelixregister is just a 'hey im here send me my calls' for incoming
15:00.46LostFrogok.. guess I will flash this brand-new WRT54G V4.0
15:00.57MGSsancholol
15:01.02*** join/#asterisk oej (n=Olle@apollo.webway.se)
15:01.02IronHelixthey're up to 4.0 now?
15:01.03IronHelixgah
15:01.11LostFrogup to 5.0, I believe.
15:01.11MGSsanchobad?
15:01.13IronHelixwhy cant they stick with ONE design...
15:01.19MGSsancholol
15:01.19IronHelixesp with network cards
15:01.30power1IronHelix, thanks....could you point me in the direction of the correct sendmail conf file?
15:01.31LostFrogUSB wireless is the worst.
15:01.38IronHelixif i know i have a linksys lne100tx i shouldnt have to open my pc and take the card out to figure out which card i have
15:01.40LostFrogOne card will work and then the next wont.
15:01.53IronHelixpower i mean use the webmin thing to configure sendmail
15:01.54AnthroOkay, so this might be naive, but... IIRC, iChat AV uses SIP. Is there a way to set things up so an iChat AV user can attempt to call me and have an Asterisk server treat it as any other SIP call?
15:02.03IronHelixit should have a sendmail setup thing as i recall
15:02.11IronHelixanthro- not naieve at all
15:02.15*** part/#asterisk oej (n=Olle@apollo.webway.se)
15:02.25LostFrogI guess I should download the linksys firmware first.. :)
15:03.13power1IronHelix, webmin thing : does a default @ home iso install run webmin on port 10000 ?
15:03.15MGSsanchoya
15:03.17*** join/#asterisk Meaty-Wrk (n=cp_simbu@office.abi.ca)
15:03.55mmlj4power1: sendmail is not trivial to configure... your best bet is to use webmin if at all possible
15:03.56IronHelixanthro- turns out ichat uses some sip derivative, which is not actual sip and thus not compatible with asterisk
15:04.09AnthroIronHelix: Oh. How disappointing.
15:04.27IronHelixyeah :(
15:05.07AnthroOkay, I'll have more questions (naive or otherwise) later. For now I have to get back to work. Thanks for all your help!
15:05.13*** part/#asterisk Anthro (i=gss@pdpc/supporter/active/Anthro)
15:05.14power1IronHelix, oh ok,,,thanks..is webmin installed by default on an asterisk @ home distro?
15:05.20IronHelixdont think so
15:05.25IronHelixhttp://www.voip-info.org/wiki/view/Asterisk@home+Handbook+Wiki  chapter 6.3
15:05.28IronHelixhas a command to grab it
15:05.31IronHelixi had thought it was
15:05.37IronHelixno problem anthro
15:06.06mmlj4power1: you can determine what port webmin runs on by using the netstat command from the console or a login prompt:   "netstat -an | grep LISTEN" will tell you what ports are open, and you can deduce from that
15:06.24power1mmlj4, thanks..will have a look
15:09.22LostFrogwhy not 'netstat -tan'?
15:09.27LostFrogwebmin is TCP
15:09.38*** join/#asterisk CBTCWwW (n=CBTCWwW@165.154.121.241)
15:09.39*** join/#asterisk frenzy (n=frenzy@193.220.82.108)
15:10.24nexiswoot, i now have full control over my X10 from my phone
15:10.47Ahrimanesbluetooth phone and xten eyebeam?
15:11.42*** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com)
15:12.12nexisno, X10 home automation
15:12.26Ahrimanesah
15:12.27Ahrimanescrap
15:12.28Ahrimaneshehe
15:12.53nexisi can pick up a phone, turn on and off lights ect
15:13.01nexishas motion and heat detectors
15:13.31Ahrimaneshow much did you spend for that?
15:13.31IronHelixhehe
15:13.34IronHelixthose are fun
15:13.49IronHelixdid you use the x10 dtmf module or something more exotic?
15:13.58nexisIronHelix, X10 usb module
15:14.11IronHelixahh
15:14.11Kattywhat's for lunch?
15:14.20IronHelixpancakes
15:14.26nexisi got a lot of it from rat shack on a sale
15:14.56nexisall in all, my asterisk box, all the modules in the system, the 2 voip phones, and the stack of linksys PAP2s, i have about 300 bucks tied up.
15:15.31Ahrimanesnice
15:15.54Kattyany /other/ lunch suggestions?
15:16.12trymWhen I reach agent login (AgentCallbackLogin), I enter my agent number with #, but nothing happens.. it just hansg up
15:16.13nexisKatty, chineese sounds good.
15:16.14IronHelixpancakes
15:16.19IronHelix:)
15:16.25nexisohh, chineese pancakes.
15:16.28NuggetI've got leftover manicotti.
15:16.31Nuggetcome on by.  :)
15:16.41KattyNugget: it has cheese in it.
15:16.46Nuggetfeh
15:16.53nomazdawhat's wrong w/ cheese?
15:16.54jovuanyone got any suggestions as to why my cisco 7960 wouldn`t pass dtmf tones through correctly?
15:17.15IronHelixtrym- your asterisk box doesnt like you.  buy it a Digium board to win its good graces back
15:17.24IronHelix/or/ anything on the console?
15:17.39IronHelixkatty- yeah get scallion pancakes
15:17.42IronHelixor lo mein
15:17.43trymill put it on a pasteshiznit.. gimme a sec
15:17.44IronHelixthose are good
15:18.13KattyIronHelix: i'm not interested in pancakes or lo mein for lunch. thanks anyway.
15:18.13nexisjovu, try dtmfmode=rfc2833
15:18.23trymhttp://pastebot.nd.edu/235
15:18.27IronHelixoh well
15:18.30IronHelixwas worth a shot
15:18.35Kattynomazda: i do not eat cheese.
15:19.02znoGdoes anyone know how I could find out the admin password to a sipura unit?
15:19.03nexisjovu, http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx
15:20.58IronHelixbbiab
15:21.08nexisnow, if i could only figure out how to build a robot to bring me a beer from the fridge
15:21.16jovunexis, i tried rfc2833, didnt work.. is there anything i need to set on the phone too?
15:21.31LostFrognexis: get married.
15:21.45mcf3782cheaper to just get the beer yourself.
15:22.12nexisi think i may train a dog to do it.
15:22.30LostFrogBut, then there will be fang holes in it.
15:22.47LostFrogNot to mention the pool of beer by the fridge.
15:23.06Dr_Raywhy not just install a beer keg at your cpu desk
15:23.24*** join/#asterisk marc324 (n=marc3234@206-248-159-56.dsl.teksavvy.com)
15:23.29LostFrogYeah.. and use the cooling unit to cool your cpu/mb too.
15:23.40lunkDr_Ray: almost a good idea, except you don't want ice melting everywhere
15:24.17*** join/#asterisk myiagy (n=myiagy@200.138.215.78)
15:24.18LostFrogYou would have to drink a lot of beer to justify a keg.
15:24.55*** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se)
15:25.58Starmakeri'm having a problem with my timer
15:25.59*** join/#asterisk feist (n=feist@nat-pool-msp.redhat.com)
15:26.04LostFrogI hate to say it, but I need another windows machine.
15:26.17StarmakerI don't have a zaptel-card, so I should use ztdummy, right?
15:26.21nexiswell, i could put in a keg-o-rator here at the desk
15:26.27LostFrogHmm.. then again.. maybe I can use vmware.
15:26.37*** join/#asterisk brookshire (n=pfffft@gateway.digium.com)
15:26.49nexisLostFrog, i drink a 30 of long necks a week, at least
15:27.27synthetiqdrink beer to ahve stomach lookign liek a keg
15:27.27nomazdamust have a gut like a trampoline
15:27.31trymIronHelix: any ideas ?
15:27.44Starmakerthe problem is, with ztdummy it's even worse than without. without it i can't use MP3Player(), because it plays REALLY slow, but with ztdummy, it doesn't fix the MP3Player()-problem, and it screws up my voicemail
15:27.49nexisi have a bit of a ponch, but not a geer gut.
15:27.59*** join/#asterisk ursuspacificus (n=paul@wsip-24-249-27-197.ri.ri.cox.net)
15:27.59LostFrogWhat is the small distribution of windows 98 that you can run from CD called?
15:28.20nexisim not aware of a livecd windows.
15:28.27jsiddall<PROTECTED>
15:28.32Starmaker2.6.13.4
15:28.34LostFrogTNot officially, nexis, but there is.
15:28.51LostFroghmm.. 2.6 doesn't even need UHCI for ztdummy, supposedly.
15:28.53trymhttp://pastebot.nd.edu/235 <-- trying to login as an agent.. I enter my agent ID after the prompt and hit #, but after a few seconds asterisk just hangs up
15:28.54StarmakerI'm using a 250 Hz kernel
15:29.04jsiddallYup, same here, same problem.  Don't know when it appeared, no one seems to have a solution short of buying a digium card :(
15:29.13*** join/#asterisk darwin35 (n=darwin35@208.139.193.178)
15:29.23*** join/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net)
15:29.26jsiddallHow do you know you have a 250 Hz kernel?
15:29.27RoyKdoes zaptel work with 2.6.13 yet?
15:29.31*** part/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net)
15:29.33nexisim runnin 2.4.31 on my asterisk box.
15:29.45Starmakerjsiddall, because I compiled it with that setting?
15:30.01szerbye al
15:30.02szerl
15:30.06nexisRoyK, it only runs on the asterisk box, the rest are running 2.6 or freebsd
15:30.19ursuspacificusHello, all... anyone have any experience getting a TDM22B working on WhiteBox Enterprise Linux 4 (Full install, plain vanilla, stock SMP kernel on P4 w/HT)?
15:30.32RoyKnexis: why 2.4?
15:30.50jsiddallHmmm... I'm using RPM'd kernel (yeah, yeah) but I'm betting the ztdummy expects 1024 Hz from the kernel
15:30.54RoyKwhy oh why.....
15:30.59Starmakerjsiddall, 1000 hz
15:31.02Starmakernot 1024
15:31.17nexisRoyK, cas before it was a asterisk box, it was my router, sat in the closet for a while, then came out of retirement for a asterisk box
15:31.24Starmakerhmm, i could 1. try ztrtc
15:31.25jsiddallAFAIK it has to be a power of 2, so ztdummy drops a few hz to compensate
15:31.35marc324not everything is in power of two.
15:31.47Starmakeror 2. recompile my kernel
15:31.52RoyKdoes zaptel ztdummy from -stable support the 2.6 timer?
15:31.56jsiddallno, but the kernel RTC does apparently
15:32.12RoyKjsiddall: was that to me?
15:32.40jsiddallNo, sorry, that was to marc324.  I'm using the ztdummy from stable
15:33.00RoyKjsiddall: with the kernel timer or usb?
15:33.02darwin35use rtc on 2.6 not ztdummy
15:33.24nexiswonder if you can pull timer data off the intel v92 card
15:33.25jsiddallkernel.  I don't think the 2.6 ztdummy uses USB at all
15:33.36RoyKdarwin35: rtc instead of ztdummy???
15:33.48RoyKdarwin35: you mean ztdummy using rtc.....
15:33.51jsiddallWhere is ztrtc now?
15:33.56RoyKer
15:33.58RoyKztrtc?
15:33.58Starmakerztdummy for 2.6 does not use USB
15:34.42jsiddallthere was a ztrtc, but I thought that was like 2.6 ztdummy for 2.4 (ie: used kernel timer instead of usb)
15:34.42SwK[Work]ztdummy uses RTC on 2.6
15:34.43*** join/#asterisk facecake (n=facecake@81.29.64.26)
15:35.01jsiddallYeah, but if doesn't work in 2.6.13 anyway :(
15:35.02nexishttp://www.voip-info.org/wiki-Asterisk+timer
15:35.29RoyKjsiddall: use 2.6.12, then
15:35.48*** join/#asterisk hellagony (n=egutierr@irc.americatelnet.com.pe)
15:35.57jsiddallThat might be the path of least resistance.  Aside from compiling the kernel, how can you tell what frequency the kernel RTC runs at?
15:36.00facecakeHi, just wondering if anyones experienced any issues with the te110p's where they refuse to notice that the nte is pluged in however with a loopback cable it gives the green light
15:36.00nexisZaprtc will not work on SMP systems
15:36.09RoyKnexis: que?
15:36.16RoyKnexis: why?
15:36.50ursuspacificusThe problem I'm running into, after having followed the "quick install guide" is that when I modprobe wctdm, I get a large pile of newlines, then "line 0: Unable to open master device '/dev/zap/ctl'".  When I look in /dev, I see zap1, zap2, zap3, zap4, zapchannel, zapctl, zappseudo and zaptimer... but no /dev/zap/... and, as you might expect, nothing in the nonexistent /dev/zap/
15:37.05nexisRoyK, from the looks of it, SMP systems are locking the RTC for SMP timing
15:37.20*** join/#asterisk n3u7 (n=neutrin0@CPE000d8802a707-CM0011e6c7edb1.cpe.net.cable.rogers.com)
15:37.43nexisursuspacificus, which distro?
15:37.58*** join/#asterisk ayano_ (n=erik_lee@68-117-160-098.static.chtn.wv.charter.com)
15:38.13ursuspacificusWhite Box Enterprise Linux 4 full install, plain vanilla with SMP kernel
15:38.15nexiscas it seems the files that it expects to see in /dev/zap are being made in /dev/
15:38.33*** part/#asterisk ayano_ (n=erik_lee@68-117-160-098.static.chtn.wv.charter.com)
15:38.36n3u7is there anyway to compile asterisk without libnewt?
15:38.53nexisalso, are you running stable, or cvs head, are you using the same version of zaptel drivers?
15:39.22ursuspacificusnexis:  I did a CVS checkout, as per the "quick install guide"
15:39.28jsiddallMake sure you read README.udev if your system has udev
15:39.32Ariel_ursuspacificus, use udev
15:40.09*** join/#asterisk RussC (n=face@216.157.205.211)
15:40.13jsiddallHmmm, tried cat /proc/driver/rtc
15:40.15nexisyea, udev will screw with ya if its not setup correctly.
15:40.27jsiddallNoticed periodic_freq   : 1024
15:40.41Ariel_ursuspacificus, http://www.voip-info.org/wiki/view/Asterisk+OS+Platforms  pick the closes which would be CentOS 4.1
15:40.47RussCHello I am getting an error pbx.c:1331 pbx_extension_helper: Cannot find extension context 'default' when ever I try to connect to a sip on an internal network
15:40.50jsiddallUnfortunately that should be the right setting
15:40.57RussCAny help would be great
15:41.22*** join/#asterisk razu_ (n=razu@ip58.cab60.mus.starman.ee)
15:41.30nexisRussC, you are missing a context.
15:41.32Ariel_RussC, it's saying you don't have any context default.  Check your settings
15:41.36n3u7nexis:I just had that problem
15:42.17n3u7used sym links
15:42.25n3u7to dev/zapctl
15:42.48RussCAriel_: context defaults in wich confs extensions?
15:43.02RussCor sip those are the only two I have edited
15:43.05nexisn3u7, thats one way of doing it, untill you restart.
15:43.09Ariel_RussC, your settings might not have the proper context setup
15:43.35RussCAriel_: thank you
15:43.45iCEBrkrI'm going to assume the zaptel stuff can't see my TP100
15:43.45iCEBrkrZT_SPANCONFIG failed on span 1: No such device or address (6)
15:43.47iCEBrkr??
15:43.54Ariel_RussC, you can see example of context in /usr/src/asterisk/configs/extensions.conf.sample
15:44.07*** part/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net)
15:44.39*** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com)
15:44.55*** join/#asterisk oej (n=Olle@apollo.webway.se)
15:46.19jlewisanyone ever seen asterisk "slow down" at which point after calls are connected, it's a few secs before you can hear each other...and you can watch dialplan logic execute relatively slowly when normally it would fly by in the CLI?
15:46.20n3u7libnewt is not cooperating
15:46.53iCEBrkrjlewis: check the load on the machine?
15:47.02eKo1What could be causing the primary d-channel to go up and down all the time. WTF?!
15:47.04jlewiswe've had this problem on and off (using various CVS stable snapshots...usually only happens after * has been running for a "long" time
15:47.10jlewisload is negligible
15:47.29nexisjlewis, also, check your ram.
15:47.41iCEBrkrfree -m
15:47.48jlewisdual HT 2.8ghz xeon...97+% idle
15:47.52ursuspacificusAriel_: Thanks for the link.  I got everything compiled and all... It looks like a udev config issue....  I will have to play with it a bit.
15:48.11jlewis800mb of buffers+cache
15:48.16nexisman, thats a lil over kill for a asterisk box
15:48.50jlewisits got 1gb ram...it's the PBX for around 100 users and has 1 PRI to the PSTN
15:48.55iCEBrkrjlewis: naaaa, what's it say for mem: free?
15:49.11jlewis<PROTECTED>
15:49.11jlewisMem:       1025392     993272      32120          0      84364     723640
15:49.11jlewis-/+ buffers/cache:     185268     840124
15:49.11jlewisSwap:      2104496      20184    2084312
15:49.57iCEBrkrThat's not bad
15:50.04jlewiswhen this happens, restarting * always clears it up
15:50.16nexisa reload fix it?
15:50.39iCEBrkruse top and sort by mem usage
15:50.48mutilatorw00t
15:50.55jlewisreload in cli?...I don't remember if we tried that before
15:50.58n3u7I'm getting a load of error messages on install to the extent:
15:51.02mutilatori get a whole week!
15:51.16ursuspacificusAriel_:  One thing that seems to be eluding me is the names of the devices that wctdm expects to see... I mean, I got /zap/ctl figured out from the modprobe error message... but does the "zap" at the beginning of all the devices I'm seeing correspond to the directory they should be in in /dev and should be stripped off on the actual device names in the directory?
15:51.30*** join/#asterisk LostFrog (n=really@69-174-51-210.chvlva.adelphia.net)
15:51.35mutilatororder the server today and build next week when it comes in, then build asterisk on it, and then hook it to 4 24 port channel banks
15:51.41mutilatorand figure out some dialplan
15:51.52mutilatorthen cross my fingers and hope nothing goes wrong
15:52.06mutilatori love doing stuff at the last minute
15:52.06LostFrogThat sounds like my MO
15:52.08jsiddallursuspacificus: README.udev should tell you what to config
15:52.27mutilatorit goes online to about 50 users on the 31st
15:52.27LostFrogyeah.. magneto-optical. :)
15:52.33mutilatoralong with sdsl
15:52.55sylewish i could get sdsl
15:53.16mutilatormove here
15:53.17mutilatori'll give ya soe
15:53.21mutilatorsome
15:53.27LostFrogwish I could afford a DS3 in my house. :)
15:53.32LostFrogWishing doesn't really work.
15:53.34mutilatorwin the lotto
15:53.44mutilatori'de do it if i did
15:53.48n3u7libnet.a(scrollbar.o)(.text+0x237):scrollbar.c indefined reference to 'SLsmg_write_chr'
15:53.48n3u7*libnewt
15:53.49n3u7hoe do I comppile without libnewt for SuSE9.3
15:53.51n3u7ther is no working patch for SuSE
15:53.53mutilatorthen spend a crap load on a few servers
15:53.58mutilatorand just host game servers or some crap
15:54.02nexisLostFrog, only a DS3?
15:54.09*** join/#asterisk gaspiz (i=gaspi@86.34.6.164)
15:54.16gaspizhi there
15:54.22iCEBrkrWhat the hell is libnewt?
15:54.29nexismutilator, whats your budget like?
15:54.29LostFrogOnly???
15:54.30iCEBrkrand what's it gotta do with Asterisk? :P
15:54.46n3u7libnewt are libraries
15:54.48mutilatorbudget?
15:54.49mutilatorum
15:54.58nexisLostFrog, if your going for overkill, bust out a OC-786
15:55.00iCEBrkrn3u7: ok, I got that much, but what's it for?
15:55.02gaspizcan someone help me with a realtime question?
15:55.06mutilatorno budjet
15:55.17mutilatorbudget*
15:55.18syleif i won the lotto i think i would play xbox fulltime
15:55.19nexiscall center, or office?
15:55.21syle:)
15:55.25mutilatora town
15:55.30ful|worki got a 100mbits line for a voip provider
15:55.44sylehow much?
15:55.45nexiseww, so billing and everything.
15:55.50mutilatoroh yea
15:56.00LostFrogI don't think I could saturate an DS-3, let alone anything higher.
15:56.01nexisfirst expernece with asterisk?
15:56.04mutilatorno
15:56.07ursuspacificusAriel_: Aaaahhh!  (1,000,000,000 candlepower light shines from behind me, through a foggy mist as it all becomes clear to me).  Many thanks, Ariel_!
15:56.34syle100 megabit line, damn nice :)
15:56.36r0d3ntlibnewt is console graphic libraries for zttool, astman and several other utilities that get compiled with asterisk and zaptel if you have newt libararies available @ compilation,
15:56.48sylein canada costs us about 2k for a 10 megabit line unmetered
15:56.56mutilatorbeen using it for sip -> sip -> pstn for a year or so
15:56.58r0d3ntsorta like ncurses, but extra special crappy from RedHat/Fedora.
15:57.17mutilatorinstalled a channel bank with a single slot at a local college a few months ago
15:57.20trymHow can I make AgentCallbacklogin use the agent id as the extension to execute ?
15:57.23nexishave the FXS banks yet?
15:57.31mutilatorno
15:57.38sylerhino?
15:57.42mutilatoradtran
15:58.01nexiscas if you have to order the server today, i was gonna sugest find a p3 500 or something, put asterisk on it, and start working on your dialplan now
15:58.04sylei'd stick with rhino
15:58.18nexisget a week extra.
15:58.24mutilatorextra week
15:58.31mutilatorwe've already got customers signed up
15:58.37mutilatorand ready to go online the 31st
15:58.45syleSER+asterisk?
15:58.52mutilatorthats our "turn on date" we've been advertising
15:59.02nexisyea, but you can start working the dial plan now, not when you get the server built
15:59.08LostFrogHe means, you can have the week that you would have spent waiting on the server to work on your dialplan.
15:59.19n3u7iCEBrkr: :it's a development library for text user inferaces
15:59.32sylei spent 3 months solid on my dialplans
15:59.36mutilatori dun have an extra machine
15:59.43mutilatori could vmware one i guess
15:59.46LostFrogGo to walmart.. :)
15:59.52nexisyour telling me you cant even scrounge up a p2 350?
16:00.02mutilatorno
16:00.09nexiswhat kind of geek are you?
16:00.18sylei got a 800 mhz machine just lieing in parts on my floor
16:00.21mutilatorthey wouldn't out out and get somethin like that cause i needed it
16:00.25sylejust needs a harddrive
16:00.38mutilatorgo out
16:00.43mutilator*
16:00.50nexissyle, i have 10 p2 350 HP Vectra's in my closet, unused
16:00.57mutilatori live in my car right now nexis
16:01.04mutilatorit's full of everythin o own
16:01.06mutilatori own
16:01.15nexisouch
16:01.17sylegirlfriend kick you out?
16:01.24mutilatoryep
16:01.26mutilatorsure did
16:01.34gaspizhow can I jump to another realtime context from a realtime context?
16:01.35eKo1chan_zap.c:1938 pri_find_dchan: No D-channels available!  Using Primary on channel anyway 16! <--- wtf?! why am I getting this all of a sudden
16:01.36LostFrogSteal her computer. :)
16:01.47nexisi take it your working at a telco or a ISP right?
16:01.56mutilatorisp yea
16:02.09nexisthey should have a extra box you can steal.
16:02.16mutilatornope
16:02.26sylehe needs to rent an apartment before he can do anything lol
16:02.32nexisyea, that too
16:02.42nexistime to call CDW and order a fully built server for next day air.
16:02.46mutilatori don't make enough to rent
16:02.55LostFrogGeez.. where are you working, mutilator, bumfcuk??
16:02.56sylethen switch jobs
16:03.02mutilatori've tried
16:03.18mutilatoronly gotten 1 interview in like 2 years of trying
16:03.26mutilatorphone interview at that
16:03.30nexisman, i make paper, and i have no money problems, you would think that a ISP would pay better then a factory.
16:03.39mutilatornope
16:03.46syleyou don;t know how to get a job then hehe
16:03.55mutilatorsyle: obviously
16:04.30sylepersonally if i wanted a job, i just walked into a place i wanted to work, asked for owner, and just had a chat with him
16:04.43sylesometimes you get lucky
16:05.27iCEBrkrI make good cash, and I have all sorts of money problems :P
16:05.56sylerent out a bedroom hehe
16:05.57iCEBrkrmutilator: Sell yourself like a whore, man..
16:06.15mutilatori;'ve tried that too syle
16:06.27mutilatorthis is the only place i landed a job doing that
16:06.37mutilatorand i'm like their genius slave labor
16:06.47sylecan you program in php?
16:06.54mutilatori do like 100x more than i was hired to do
16:06.55mutilatoryea
16:06.56syleyou can get jobs off irc if you can
16:07.17sylecamp in #php on efnet
16:07.26nexisheh
16:07.31iCEBrkrmutilator: Monster.com works
16:07.38sylein past i;ve done about 10k in contract work just from people in that channel
16:07.41sylealso from diff sites
16:07.43mutilatoriCEBrkr: not really
16:07.45nexisthats soo true syle, people come in there saying ill pay 300 bucks for somone to code this up.
16:07.53mutilatorthats where i got my 1 phone interview
16:07.54iCEBrkrmutilator: Not really?  I've gotten 2 jobs off it.
16:07.59mutilatorout of i think 800 apps i put it on there
16:08.05iCEBrkrmutilator: and I dunno how many face to face intervies.
16:08.22mutilatoru probly have a degree too
16:08.29iCEBrkrmutilator: Nope
16:08.35mutilatorwell i dunno then
16:08.37sylewell my last job i got from #php to, they offered me fulltime 60k a year position
16:08.42sylei worked there for 2 years
16:08.46iCEBrkrmutilator:  apt-get install confidence
16:08.49mutilatori apply for every job from help desk to network engineer
16:08.57nexisi have no degree, got my high school deploma last year, and i have no problems gettin jobs.
16:09.00LostFrogThat works, iCEBrkr??
16:09.07mutilatoriCEBrkr eh..
16:09.10iCEBrkrLostFrog: It was worth a try :P
16:09.12LostFrogI've been trying rpm -i confidence.rpm
16:09.21LostFrogIt keeps saying 'file not found'
16:09.27ikey1rt
16:09.30nexisiCEBrkr, i would be happier with apt-get install girlfriend
16:09.40iCEBrkrmutilator: Seriously, you gotta go in there and tell them what you know and be confident in what you're telling them.
16:09.44eKo1hmm...looks like a full reboot did the trick
16:09.46*** join/#asterisk philm (n=Phil@73.236.204.68.cfl.res.rr.com)
16:09.47iCEBrkrnexis: That'd be sweet!
16:09.48IronHelixyou need to understand that package first, do man woman
16:09.50eKo1wtf man
16:09.59iCEBrkrIronHelix: lol
16:10.04LostFrogman man should be illegal. :)
16:10.04mutilatoryeh
16:10.09mutilatorif i got the interviews i could be
16:10.11mutilatorbut i don't get em
16:10.37nexisor
16:10.53nexisapt-get remove --purge dishes
16:11.01iCEBrkrmutilator: then you need to go over your resume format.
16:11.06sylebtw reason i left my job at 60k a year was dude woudl never give me a raise, so if you own a company don;t forget to do that :)
16:11.19mutilatoriCEBrkr: ya i've been thinking of saving up and blowing cash on the monster.com resume writer ppl
16:11.44nexismutilator, no need for that, you have a local community college right?
16:11.44iCEBrkrmutilator: ehhh, I wouldn't go that far. Just look around and find a nice looking for mat.
16:11.49LostFrogI don't make shit, but I'm happy.. (except for the bills)
16:11.57mutilatori make $8.50/hr
16:11.58mutilator:)
16:12.08LostFrogI make $27.5k
16:12.08nexisdamn, where in the hell do you live?
16:12.13iCEBrkrYeah, really. You could probably get a nice looking resume from a college student for $20
16:12.18LostFrogYou can make $10 at McDonalds
16:12.24sylei make 0.00/hr right now, still trying to get my site up
16:12.26mutilatorya LostFrog
16:12.33mutilatori could go do a factory job for like $12
16:12.37mutilatorand get insurance and shit too
16:12.38Ariel_10 dollars at McDonalds???
16:12.48mutilatorbut that isn't what i wanna do
16:12.51LostFrogHere you can, areski.
16:12.51philmDoes anyone know a site with information on hardware hacking for voip, like connecting a fxs port to a loud speaker or microphone?
16:12.51mutilatorso i stick around here
16:12.54LostFrogAriel, even.
16:12.57nexiswell, i was saying, the community colleges usualy offer free classes on how to write a resume
16:13.09syle10 at mcdicks? that must be california
16:13.11nexisphilm, why, use a sound card
16:13.20nexischan_oss or chan_alsa
16:13.20LostFrogNorthern Virginia
16:13.21mutilatori live in northern michigan
16:13.29nexisdamn dude
16:13.35syleisn;t minimum wage there like 10-11 bucks an hour now
16:13.36philmI want to havee allot of them.
16:13.46nexisno, min wage is still 5.25
16:13.51LostFrog$7.75, I think.
16:13.55mutilatorfederal
16:14.01iCEBrkrIt's more than 5.25 anymore..
16:14.04mutilatorstate to state is diff
16:14.04nexisi havent made min wage sense i was like 16
16:14.17LostFrogI never made minimum wage.
16:14.31mutilatorsoon as hardwire gets here
16:14.35*** join/#asterisk xunil (i=xunil@66.194.40.30)
16:14.37mutilatori'm goin to take up his job offer
16:14.41mutilatorsee if he's still lookin
16:14.44nexisehh, i washed dishes in a kitchen when i was 14 for min wage.
16:14.49sylei don;t even want to remember jobs i had at 16, picking weeds out of strawberry farms, and picking up horse shit lol
16:14.52*** join/#asterisk gaggaman (n=leo@host-82-135-28-39.customer.m-online.net)
16:14.59mutilatori'll just be on a frigid island off alaska
16:15.19LostFrogewww.. $5.15
16:15.21nexismutilator, dude, alaska is killer
16:15.32nexisits like .7:1 ratio males to females.
16:15.35iCEBrkrLostFrog: $5.15 > $.0.00
16:15.37mutilatorya
16:15.43mutilatorbut the femals are all fat samoans
16:16.03sylemales to females or females to males?
16:16.21LostFrogI thought samoa was in the south pacific..
16:16.29mutilatorit is
16:16.41mutilatorbut anchorage anyway is filled with em
16:16.44sylecanadian maritimes is where you want to be: 9 to 1 girls to guys
16:16.48mutilatorsamoans and asians
16:17.28sylethen there is always the phillipines and tailand hehe
16:17.38LostFrogNote to self: vmware doesn't like it when you change the MAC for your ethernet card without rebooting.
16:17.44mutilatorhm
16:17.50mutilatorwell none of my bosses are around
16:17.55ursuspacificusAriel_: well... /dev/zap is now appearing as it should, but when I modprobe wctdm, I still get the same error "line 0: Unable to open master device '/dev/zap/ctl'"
16:17.59mutilatori'm going to sneak out to town and get food
16:18.03mutilatorbbl
16:18.06*** join/#asterisk Utah_Dave (n=boucha@0-1pool138-165.nas28.salt-lake-city1.ut.us.da.qwest.net)
16:18.16sylelostfrog, stealing ip addresses again?
16:18.18mutilatormaybe i'll get boardwalk at mcdonalds!
16:18.32mutilator;)
16:18.32LostFrogno.. fooling a MAC filter.
16:19.06syleahh, licensed software
16:19.15iCEBrkrgrrrr. lspci doesn't show my TP100 card
16:19.31LostFrogIt must not exist then.. :)
16:19.38LostFrogIt was all your imagination.
16:19.43iCEBrkrI guess so
16:19.56LostFrogSue Digium for vaporware. :)
16:19.59iCEBrkrhaha
16:20.06iCEBrkrThis is weird.
16:20.14n3u7I'm not sure that asterisk can be installed on SuSE9.3
16:21.21LostFrogI'm sure it can be installed, running sucessfully is a fish of a different color.
16:21.46gaspizhow can I jump to another realtime context from a realtime context?
16:21.56LostFrogIf I wasn't lazy, I would download it and help you out.
16:22.55*** join/#asterisk pjz (n=pj@place.org)
16:24.02iCEBrkrYa know. It helps if the card is pushed in all the way
16:24.11LostFroglol.. yes it does.
16:25.03iCEBrkrw00t! configured!
16:25.22LostFrogGrrr.. why does RendezvoisProxy not save it's configuration when it exits??
16:25.26LostFrog-i
16:25.38*** join/#asterisk tainted_ (n=somewher@mail.k2usa.com)
16:25.41*** part/#asterisk pjz (n=pj@place.org)
16:26.11*** join/#asterisk in (n=int@secure.wifihacker.org)
16:27.13iCEBrkrok, now how the hell do I Dial() through this thing :P
16:27.53LostFrogDial(Zap/<x>,<number>)?
16:28.21iCEBrkrI'm lost on the <x> part... I figured Zap/
16:28.29iCEBrkrDoc's claim g1
16:28.32*** join/#asterisk smcmahon (n=admin@digitaldatabits.net)
16:28.48*** join/#asterisk damned (n=vpol@damned.vpol.org.ru)
16:28.59LostFrogWhatever port it is.
16:29.04LostFrogStarting with 0
16:29.04iCEBrkrhaha
16:29.08iCEBrkrI dunno
16:29.14iCEBrkrI've never done PRI stuff with Asterisk
16:29.18LostFrogzap show channels?
16:29.24iCEBrkroh, duh!!
16:29.34darwin35ouch
16:29.40darwin35thats my forhead
16:29.46iCEBrkrSorry
16:29.47LostFroglol.. I knew that was coming..
16:30.07LostFrogBetter your forehead than your foreskin.
16:30.11iCEBrkrdoh
16:30.19smcmahonWow running Xwindows platform over Windows 2000 is pretty cool. Looking for a better X-Client to use tho besides reflection
16:30.22darwin35hey keep my foreskin out of this
16:30.36darwin35kde
16:30.39iCEBrkrIs there a zap module?  Cuz I don't have the zap command available.
16:30.41darwin35xfce
16:30.48LostFrogchan_zap.so?
16:30.56Ahrimanessmcmahon: xfree's win32 edition?
16:30.58darwin35hey hey hey
16:31.10smcmahonheh sorry had to darwin35
16:31.15darwin35leave it where it is
16:31.19LostFrogYou know what the best thing about being a rabbi is?
16:31.24smcmahonbut that would be limp right?
16:31.25LostFrogYou get to keep the tip.
16:31.37darwin35hahahah
16:32.44nextimecan * detect a pulse tone like a dtmf on a pri zap channel?
16:33.19*** join/#asterisk hugo-v6 (n=hugo@195.140.232.114)
16:33.31hugo-v6hiho
16:33.45azzienextime, pulse over PRI ? that's funny :)
16:34.16*** join/#asterisk [Airwolf] (n=airwolf@airwolf.xs4all.nl)
16:34.49nextimeazzie : i know, but i have some ivr apps with a large user base to support, and about 1% of those users can use only a pulse phone line
16:35.04iCEBrkrnextime: Sucks to be them!
16:35.10sylewonder what would happen if i used a rotory phone and dialed 2 to a PRI line
16:35.25iCEBrkrnextime: When was the last time you ever heard of a IVR system that supported pulse-dial?
16:35.40azzienextime, my guess - you'll be getting flash events for every digit...
16:35.46gaggamanhi!
16:35.58nextimeiCEBrkr : i've no hear about that, i'm asking if it is possible
16:36.09iCEBrkrnextime: I don't think it is.
16:36.16gaggamancould maybe somebody help me with my bristuffed asterisk 1.09 and call Pickup?
16:36.24jarrodim getting droped voice in the middle of phone calls.. i guess thats the RTP stream and it only happens one way
16:36.26iCEBrkrchan_zap.so is being stuborn
16:36.28jarrodanyone ever heard of this
16:36.42gaggamanwhen I do a Pickup, i get;
16:36.44iCEBrkrload_modules: Loading module chan_zap.so failed!
16:36.47gaggaman<PROTECTED>
16:36.47gaggaman<PROTECTED>
16:36.47gaggaman<PROTECTED>
16:36.50iCEBrkrblah blah blah
16:37.20nextimeiCEBrkr : ok thanks, i think the same.
16:37.37gaggamanand then, the picked up channel (Zap/1-1) is ZOMBIE
16:37.47ursuspacificusw00000000000000000000000000000000000000t.
16:38.07azzienextime, try here: http://www.voip-info.org/wiki-Dial+Pulse+to+Touchtone+DTMF+Converters
16:38.23ursuspacificusmust modprobe zaptel first, then modprobe wctdm.
16:38.24tainted_what is parkandannounce used for?
16:39.04iCEBrkrweird.. Seems as if I put a noload => chan_oss.so it worked.
16:39.10iCEBrkrI guess I could test that.
16:39.22gaggamannobody?
16:40.01gaggamanwhat does "<MASQ>" in Hungup 'SIP/61-a2fe<MASQ>' mean?
16:40.21nextimeazzie : the pulse to dtmf converters are for a phone line attached to a fxs on the * server, not for an external line come in from a remote and unknown user to an E1 line
16:40.23sylemeans incorrent symbol for NASDAQ trade
16:40.45iCEBrkrsyle: boooo hissss :)
16:40.59gaggamancould have guessed this :-)
16:42.09LostFrogThere is no such thing as a pulse-only phone line.
16:42.27LostFrogMake your users buy dual-mode phones with a switch for pulse/DTMF.
16:43.12LostFrogThat's what I used to do when the phone company wanted to charge me extra for DTMF dialing.
16:43.19nextimeLostFrog : i can't ask to all italian phone users to buy a device for call my service 1 time
16:43.20*** part/#asterisk scoates (n=sean@iconoclast.caedmon.net)
16:43.49nextimethey are about 60 milion of people
16:44.08nextimeand i don't know which of them will call my ivr
16:44.10LostFrog1% would only be 600,000.
16:44.22iCEBrkrnextime: Seriously man, it ain't happening.
16:44.40*** join/#asterisk zedas (i=zedshawc@pizarro.dreamhost.com)
16:44.43*** part/#asterisk darwin35 (n=darwin35@208.139.193.178)
16:46.46iCEBrkr*CLI> pri show span 1
16:46.46iCEBrkrStatus: Provisioned, In Alarm, Down, Active
16:46.52iCEBrkrErrrrrm.
16:47.25nextimeiCEBrkr : my problem is that the ivr that i'm talking about serve a service like a "lottery" over a premium number, so, if a user call and can't play to the game, he pay for the service but he can't win anything...
16:47.55iCEBrkrWelcome to 2005
16:48.01nextimeanyway, i will put a "check if your phone is tone compatible" menu on the ivr app
16:48.03*** join/#asterisk mhnoyes (n=mhnoyes@user-38lc1bi.dialup.mindspring.com)
16:48.04iCEBrkrThat'd be my greeting
16:48.09nextimethis sound like the only one solution
16:48.54iCEBrkrThat's how everyone used to do it
16:53.43mutilatoromg mcdonalds monopoly sucks
16:53.46mutilatorevery, and i mean every time i've gone i get B&O Railroad
16:53.50mutilatori have like 9 sittin in my car right now
16:54.16brookshire:(
16:54.23brookshireyou need to buy more mcdonalds
16:54.25brookshirehehe
16:54.52brookshirei wonder if you can buy stamps on ebay
16:55.14mutilatormost of the time i got get breakfast
16:55.28mutilatoractaully went for lunch today
16:56.16syleignorepat => 9
16:56.21syleam i missing something
16:56.26sylewhy doesn;t this work
16:57.51*** join/#asterisk fugitivo (n=ajf@209.13.241.249)
16:58.05pauldyanyone know why did doesn't seem to work right and if there are any examples on how to make it work right
16:58.19pauldysorry i should add with incomming sip calls
17:00.23*** join/#asterisk nobell (n=jdegraff@67.137.31.58)
17:02.59mutilatorwonder if i can instant win anything
17:05.22*** join/#asterisk AgiNamu (n=AgiNamu@dsl081-096-215.den1.dsl.speakeasy.net)
17:06.42*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)
17:08.43AgiNamushit, configuring a cisco as5X makes configuring asterisk look easier than anything
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17:22.18*** part/#asterisk bkw_ (n=bkw_@adsl-69-148-34-63.dsl.tulsok.swbell.net)
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17:30.39nexisblah, that sucked
17:32.27*** join/#asterisk phb (n=phb@c213-100-46-84.swipnet.se)
17:34.10nextimedetach
17:36.12eKo1wtf, this is the second time * crashes on me. It always crashes after "-- Stopped music on hold on ..."
17:36.39*** join/#asterisk lars-ut (n=me@67.137.31.58)
17:38.25gaggamancould maybe somebody help me with my bristuffed asterisk 1.09 and call Pickup?
17:38.31*** join/#asterisk pnviking (n=pnviking@c83-248-7-150.bredband.comhem.se)
17:38.33gaggamanwhen I do a Pickup, i get:
17:38.46gaggaman<PROTECTED>
17:38.55gaggaman<PROTECTED>
17:39.02gaggaman<PROTECTED>
17:40.05*** join/#asterisk phb (n=phb@c213-100-46-84.swipnet.se)
17:40.14snittafaik picking up does not work between different technologies
17:40.26snitta sip exten can only pickup an other ringing sip exten
17:40.38*** join/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com)
17:41.25nexisman, aint cell proviers supposed to have battery backups and crap for when the power dies?
17:41.36*** part/#asterisk frenzy (n=frenzy@193.220.82.108)
17:41.52*** join/#asterisk asterisk99 (n=dunoowhy@modemcable169.194-130-66.mc.videotron.ca)
17:42.28asterisk99anyone know if ztfcg is supposed to be run after every reboot??
17:42.45asterisk99(amke the ztcfg)
17:42.56eKo1yes
17:43.18asterisk99eKol: any reason why it would sudennly stop running?
17:43.43eKo1what suddenly stopped running?
17:44.23*** join/#asterisk malverian[work] (n=pawalls@pawalls.teamgleim.com)
17:44.33asterisk99eKol: it used to work... it would reboot fine... now it doesn't --- I have to manually run ztcfg and start asterisk
17:45.13asterisk99eKol: I wonder if I should d/l and install the latest Zaptel driver
17:45.45nexisasterisk99, sounds like your init script is gone, or messed up
17:45.53asterisk99hmmmmm
17:46.40g__Strageness: I'm calling a 1-888 number on our PRI, and it times out after 45 seconds, even though the call is still in progress.
17:47.37*** join/#asterisk JohnnyC (n=JoaoCorr@195-23-115-68.net.novis.pt)
17:48.05g__Since it's a PRI.. all the signallying should be happening out of bound.. so how could asterisk not know the phone call has been answered?
17:48.15g__s/bound/band
17:48.52*** join/#asterisk loick (n=loick@APuteaux-151-1-30-110.w82-124.abo.wanadoo.fr)
17:49.29g__Does anyone know what could be going on?
17:50.03*** join/#asterisk malverian[work] (n=pawalls@pawalls.teamgleim.com)
17:50.23malverian[work]If I set a variable in sip.conf for a phone using setvar=FOO=bar, how can I access that from the dialplan?
17:51.24*** join/#asterisk pehrjansson (n=pehr@adsl-68-90-188-15.dsl.austtx.swbell.net)
17:51.58pehrjanssonDare I ask a newbie question?
17:52.00develword, everybody.  can i have music-on-hold while ringing?
17:52.11develpehrjansson, have at it.
17:52.25*** join/#asterisk darkskiez (n=darkskie@host86-138-169-183.range86-138.btcentralplus.com)
17:52.32wunderking__: the other end isnt answering the line, the easiest way i know of to fix it is to answer it on your side, you can set it to just the specific numbers you have problems with.. i think there are other ways but dunno
17:52.45malverian[work]devel, ?
17:53.00malverian[work]devel, Yeah... Dial(...,,m[(class)])
17:53.17JohnnyC<PROTECTED>
17:53.18JohnnyCOpen of mISDN Failed
17:53.24JohnnyCIm getting mISDN failed
17:53.52develmalverian[work], i'm sorry, i meant while ringing via transfer
17:53.58g__wunderkin: I'll look into it.. I *just* tripped over a mailing list post that describes my problem.
17:54.45develyou hear the MOH while they're actually dialing the exten, then goes to dead air until answered....
17:55.43pehrjanssonI have a couple of Grandstream Budge Tone and cannot get the VM to work.  Changed the dtmf in sip_additoinal.conf dot "dtmfmode=Info".  Still it will not work to access the vm mailboxes from the telephone set.
17:56.38develpehrjansson, i use dtmfmode=rfc2833 all the way around with no problems
17:57.04develpehrjansson, both on the grandstream side and the asterisk side (set for each exten in sip.conf)
17:57.44pehrjanssonin the Asterisk hitchhiker's guide it is stated "Info
17:57.44pehrjanssonThe Info method sends SIP Info messages from phone to Asterisk with the text of the buttons pressed on the phone. SIP Info tends to be a better choice than Inband because the key-presses are sent textually, independent of the audio codec. Users of Grandstream phones should use Info for the Asterisk voicemail; the other methods won't work.
17:57.49pehrjansson"
17:58.32develbah.
17:58.48pehrjanssondo you use grandstream phones?
17:58.55*** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1)
17:59.01develyes.  both budgetone and handytone
17:59.10ful|workhaven't problems compiling oh323 with asterisk -> http://pastebin.ca/25865
17:59.18ful|work*i'm having
17:59.21pehrjanssonand you have no problem with rfc2833?
17:59.41ful|workanyone knows how to fix this ?
17:59.42develthat is affirmative, pehrjansson, so long as it is set the same in both the phone and in sip.conf
18:00.04wunderkinexten => 1,n,Monitor(raw|/home/dialer/monitor/${leadid}-${UNIQUEID}|b)
18:00.04wunderkinexten => 1,n,Dial(SIP/2|18)
18:00.12wunderkinthat should be ok right? its only recording 0.040 sec hmm :(
18:00.23pehrjanssonthe sip.conf I know how to manipulate but I haven't found dtmf on the web interface for the telephone set
18:01.19develpehrjansson, it's labeled "Send DTMF"
18:01.39pehrjanssoni see it now.
18:01.47pehrjanssonthanks.  i'll try it.
18:01.52wunderkinhey mark think i found a bug with the queue monitoring.. it core'd on me when the agent answers.. using 10/15 have to check to see if anything has changed on that yet :D  trying to work around the problem now
18:02.41develso, nothing about MOH while ringing (via transfer)?
18:03.24*** join/#asterisk junbug (i=junya@adsl-144-134-75.mia.bellsouth.net)
18:03.33develheck, even ringing would be acceptable (i just get dead air)
18:03.50wunderkindevel, maybe the moh is getting stopped early?
18:04.19develwunderkin, it acts like as soon as the "transfer" takes place (i.e. first phone releases call) the MOH stops
18:04.32wunderkinwell i meant check the cli :D
18:04.37iCEBrkrAlrighty, this blows.
18:04.42develi am.  i watch it stop.
18:04.51develso now i need to know how to make it unstop.
18:05.03iCEBrkrTP100P says my PRI is in alarm, but the same PRI just worked on our PoS Dialogic box
18:05.44wunderkindevel, you have moh on the dial for who you are dialing?
18:06.17develi don't, in fact.
18:06.24pehrjanssondevel - that worked.  thanks again.
18:06.32develbut shouldn't i at least be getting the ringing?
18:06.36*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
18:06.36develno problem, pehrjansson
18:06.51*** join/#asterisk Godsey (i=lanny@pdpc/supporter/sustaining/Godsey)
18:06.59sylei prefer
18:07.15syleMonitor(wav|/home/dialer/monitor/${leadid}-${UNIQUEID}|bm)
18:07.17sylein your case
18:07.23wunderkindevel, ive heard people say sometimes they dont get ringing.. im not sure about all the internals so i dunno just giving hints where to look
18:07.27wunderkinsyle, yeah i know about m :P
18:07.36`Sauronhm.
18:07.39*** join/#asterisk dasuberdavid (n=david@gateway.digium.com)
18:07.42`SauronMonitor() lets you record calls?
18:07.51AgiNamuyea
18:07.55sylewav i can;t listen to from my windows computer
18:07.55*** join/#asterisk AlexCTI (n=alex@weston-69.65.86.197.myacc.net)
18:07.56sylecan
18:07.59develwunderkin, i'll force the m(class) for that exten, since it's a group.  thanks for the pointer.
18:08.03wunderkinsyle, im just going to have a script to mix and convert to mp3 at the end of the day
18:08.03`SauronApparently I should update my * install. :)
18:08.21pehrjanssonnext question, which I believe relates to txgain, people we talk to complain about the volume being too muffled.  I ran ztmonitor and it did appear that the levels were very low.  how do I best increase the output volume.
18:08.26*** join/#asterisk AlexCTI (n=alex@weston-69.65.86.197.myacc.net)
18:08.36wunderkinsyle, my problem is that it didnt record a darn thing though :P
18:09.25sylestrange
18:09.39develpehrjansson, that's in /etc/asterisk/zapata.conf (rxgain/txgain)
18:09.40syleare you sure that dir is writeable by asterisk user?
18:09.50syledid you check debug log?
18:10.04wunderkinsyle, yeah well the file did get written only 0.040 sec tho
18:10.25AlexCTIHi... Some can help me to get a good quality on Music on Hold, I tried raw and mpg123 but the quality is not good.
18:10.27sylewhat version of asterisk
18:10.30develpehrjansson, i think you have to actually restart asterisk after that change (rather than just reload)
18:10.34wunderkinsyle, head 10/15
18:10.51g__wunderkin:it sounds like I'm having problems related to "Answer Supervision".. I'm just reading up on it now.
18:10.57Renacorugh what am I doing wrong with this polycom phone, I can dial out on it, but when I dial it's extension it goes to VM
18:11.15sylestrange, well worked for my zap and sip tests
18:11.15*** join/#asterisk mmmToop (n=chatzill@rrba-146-64-241.telkomadsl.co.za)
18:11.23wunderking__: yeah.. sometimes the other end doesnt send an answer while in the ivr i guess
18:11.24Renacorit waits but the phone doesn't ring
18:11.41g__Yeah.. do you know (off the top of your head) what the fix is?
18:11.41sylei;m using latest cvs head as well
18:11.55wunderkinnot sure what i changed i thought i had this working already
18:12.09sylehmmmmm
18:12.24sylei have a feeling maybe it can;t find sox in the path
18:12.27sylecheck
18:12.29sylewhich sox
18:12.30wunderkinim not using m
18:12.37*** join/#asterisk ic (n=ic@staff.rbi.speka.net)
18:12.40ichi there
18:12.55syletry it
18:13.03sylewhat you got to loose
18:13.06wunderkinit shouldnt be running sox until afterwords anyways though, the file should be several seconds
18:13.07g__Wunderkin: I presume you have a T1 as well?  Would you care to trade zapatel.conf files?
18:13.10sylefix by trial and error
18:13.37wunderking__: yeah i have a pri, nothing special in the config though, i commented out everything and only added in callerpres
18:13.59wunderkini can't call 800 numbers though since its only LD, and i dont call ivrs :D
18:14.01g__Thanks, I'll compare mine with the default then.
18:14.14syledo you own the PRI?
18:14.23wunderkinyeah its mine
18:14.24develwunderkin, that's the stuff.  after adding the m(class) to the group extens all works as expected.  thanks.
18:14.26ichas someone here interfaced asterisk (h323) with a eads telecom ipbx with a pt2 card ?
18:14.29g__Not personally, but my company does.
18:14.35wunderkinoh him
18:14.49g__why?
18:14.49wunderkindevel: ok thats what i was wondering, not sure about no ringing though
18:16.22*** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net)
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18:20.02*** part/#asterisk T-Squared (n=ted@hidden.serreyn.com)
18:21.22wunderkinnot working w/o b either
18:21.46wunderkinweird :(
18:21.58wunderkinit stops monitoring after the dial probably
18:22.08wunderkini mean after answer
18:22.13wunderkinwhy?
18:22.50*** join/#asterisk zeedo (n=zeedo@80.68.92.188)
18:25.46Renacorargh why wont this damn phone ring
18:26.18nexisanyone using goiax?
18:26.49*** join/#asterisk ronaldl79 (n=chatzill@c-24-8-54-203.hsd1.co.comcast.net)
18:28.37ronaldl79How are you guys dealing with clients who are Qwest, SBC, BellSouth or Verizon subscribers that require white page listings? After porting a clients number, are you finding these companies still list your clients number and address, or not?
18:31.16Ariel_ronaldl79, when I setup any system all my customers keep one or 2 lines with the local phone co.  Then we forward the local number to the voip one.
18:31.32Ariel_other reason for this is faxes and 911
18:33.21*** join/#asterisk vader-wrk (n=johndoe@204.183.88.101)
18:33.27vader-wrkhello
18:35.39vader-wrkdoes anyone in here use ADIT 600 channel banks in their setup?
18:38.06ronaldl79Hmmm, Ariel. I don't want the telco anywhere in the picture. There are sold on the cost savings of VoIP alone.
18:38.14ronaldl79They*
18:38.46Ariel_ronaldl79, well don't have any suggestion. Due to all my setups must have one pots. for 911. It's just makes no sence in my view not to have it.
18:39.09g__911 is a very good reason..
18:39.19g__Safety first..
18:39.26AlexCTIAriel_, do you know how can i do to get better quality on MOH?
18:40.07Ariel_AlexCTI, I use normal mpg123 in the setups seems to work fine.
18:41.07infinity1Ariel_: who do you use for voip gateway usually?
18:41.52Ariel_infinity1, depending on the setup I use either a sipura 3000 or tdm400p.  Also channel banks like the Adtran. It really depends on the setup.
18:41.53AlexCTII did, but the quality is not the best, and in some parts of the song it sounds like underwater.. or noise, i'm using G729 i don't if it affect the quality.
18:42.42Ariel_AlexCTI, ahh then convert them to g729 or native (raw) if your using cvs head. I only use stable at present.
18:42.52infinity1Ariel_: what about voip to pots service? like teliax and such?
18:43.32Ariel_infinity1, they work I use them. I have some customers using teliax, VoicePulse, Nufone asterlink. But I always maintain one pots line.
18:44.21infinity1Ariel_: have you found an inexpensive solution for incoming callerid info over voip?
18:44.23Ariel_Not everyone needs the same type of service.
18:44.34AlexCTIAriel_, How can I know if i'm using cvs head?
18:44.53AlexCTIi'm new.. sorry :-(
18:44.54ronaldl79Speaking of G729, who's using it for themselves or a client for termination?
18:44.58Ariel_infinity1, that is a loaded question. Due it's more of what they need then just inexpensive
18:44.59infinity1AlexCTI: did you download a tgz to use cvs to download it?
18:45.09Ariel_show version
18:45.52AlexCTItgz
18:45.57*** part/#asterisk junbug (i=junya@adsl-144-134-75.mia.bellsouth.net)
18:46.25infinity1Ariel_: the CNAM issue i guess it is called. thats what i find as a major drawback with teliax
18:46.42Ariel_there is a tgz of 1.2beta1 so that is not a valid.  AlexCTI in the CLI do show version
18:46.52AlexCTIok
18:47.11AlexCTIAsterisk 1.0.9 built by root@ACS on a i686 running Linux
18:47.22Ariel_infinity1, I don't have much problems with them.  Since I am only changing the number not the name when we use them for dial out.
18:47.34*** join/#asterisk scoates (n=sean@iconoclast.caedmon.net)
18:47.39Ariel_AlexCTI, you have stable.
18:47.44infinity1Ariel_: oh . do you use voip for incoming calls?
18:47.52Ariel_infinity1, yes I do
18:48.06AlexCTIAriel, that's the same like yours..correct?
18:48.24Ariel_Yes
18:48.36infinity1Ariel_: is there a solution for callerid in incoming calls?
18:48.49scoatesI'm getting what I can only describe as "DTMF doubling" iax->asterisk->SIP-><conferencing bridge>  (it detects a single "4" as "44") -- any ideas? I don't mind reading, I just don't know what to look for.
18:48.49Ariel_But I have not run into any problems do you have enough cpu power for the transcoding?
18:49.50Ariel_infinity1, Yes there is. My home I and work I have an did from connect.voicepulse.com Which gives me full caller ID info. Also allot of others do as well.
18:50.02iCEBrkrSo is there some special cable I'm supposed to use with this T100P card?
18:50.16Ariel_iCEBrkr, t1 crossover cable
18:50.19iCEBrkrThe cable I have now seems to work just fine in another box with a Dialogic card in it.
18:50.35infinity1Ariel_: guess i just got unlucky with teliax :)
18:50.37iCEBrkrI'm assuming this little adaptor thing is the crossover.
18:50.38Ariel_dialogic card is not a digium card
18:50.38xhelioxThe IAXy is requesting a registration period of 0 seconds, when Asterisk's default is 60 seconds. Is there any way to tell the IAXy how long to register for? And yes, I know the minreg can be set in iax.conf, but that creates a different issue, which isn't really relevant. :)
18:50.51iCEBrkrAriel_: I understand that.. I'm trying to rid this company of Dialogic cards
18:50.51AlexCTIAriel_, yes, the CPU is enough
18:50.53iCEBrkr:P
18:51.03*** join/#asterisk Inv_arp (i=junya@adsl-144-134-75.mia.bellsouth.net)
18:51.22iCEBrkrI've had nothing but headaches with Dialogic cards in the 15yrs I've been around phone 'systems'
18:51.27Ariel_AlexCTI, are you using mpg123 or mpg321? which version of mpg123?
18:51.47iCEBrkrnot saying I know anything about phone systems-- But I'm an expert that Dialogic cards are nothing but problems :P
18:52.04Ariel_iCEBrkr, http://www2.adtran.com/support/technotes/t1ddsadptxvr/
18:52.14AlexCTIRight now, I switch it by rawplayer, it sound a little bite better, but i was using mpg123.
18:52.16Inv_arpAriel_: sup man
18:52.23iCEBrkrAdtran?  had one of those at the last job...
18:52.35Ariel_iCEBrkr, I have about 50 of the 4pci and dialog4 boards. I think i know them also
18:52.52iCEBrkr:)
18:53.11infinity1Ariel_: wow. 11 / month for a voicepulse did. but free incoming minutes!
18:53.26Inv_arpAriel_: the client i was talking to you about never got the phones in yet so thats why i havent updated you on anything yet
18:53.45*** join/#asterisk nextime (n=nextime@213-140-6-96.ip.fastwebnet.it)
18:53.56Ariel_Inv_arp, not much just stopping by to answer some questions. No problem
18:54.14AlexCTIAriel, how can i get the mpg123 version?
18:54.26Inv_arpinfinity1: inphonex offers 7.95 incoming DID ...   SIP/lIbc/g729 support only tho
18:54.32Ariel_infinity1, I have used them for over 2 years now and it's been rock solid. There not the cheapest but they have worked.
18:54.47Ariel_AlexCTI, at the linux prompt do mpg123
18:54.51infinity1Ariel_: well the fact that they have callerid is huge.
18:55.08ronaldl79I'm actually looking for a rock solid provider as well for a client.
18:55.19Ariel_infinity1, not all sections or calls will give you caller ID
18:55.26AlexCTIHigh Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3.
18:55.26AlexCTIVersion 0.59r (1999/Jun/15).
18:55.30Ariel_some are blocked depends on who is calling.
18:55.42ronaldl79I cannot risk these guys with BroadVoice, they need a solid sip/iax provider ... my reptuation is on the line .. and so is VoIP.
18:55.49Ariel_AlexCTI, that is the correct one
18:56.12scoatesheh.. it was echo
18:56.15scoatessilly moi
18:56.44AlexCTIlet me give you the config.. hold on
18:56.48Ariel_ronaldl79, look at voicepulse, teliax, then.
18:56.57iCEBrkrOk, the cable I have is a straight-through... I'm going to have to assume this little blackbox hang'n off the interface here is the crossover
18:57.05Ariel_AlexCTI, I have to go back to work I will be back later.
18:57.12AlexCTIok..
18:57.17iCEBrkrronaldl79: Good luck
18:57.21Ariel_iCEBrkr, never assume
18:57.27iCEBrkrAriel_: agreed :P
18:57.52ronaldl79hmmmm
18:58.07iCEBrkrI guess all I need to know is what kinda cable I need for this Digium card.  I briefly looked on Digium's site, and didn't find anything.. Tho, I didn't look hard
18:58.28infinity1Ariel_: do they let you pick your did #? how much of a selection do they have?
18:58.29g__After reading more posts on the Internet, the only solution to calling big companies with IVRs that don't send their a "Answer Supervision" until you're speaking to a real person is either: navigate the IVR menu really fast, or make special dialing rules that Answer() the before Dial().  Is there no other solution?
18:58.32Ariel_iCEBrkr, did you not get my link
18:58.32ronaldl79I don't see anyone else offering BroadVoice's service plans though ... damn, what a wedge to be in...
18:58.44iCEBrkrFigures I get all the zaptel/libpri stuff compiled and working and now I'm gonna have issues with hardware :-/
18:58.47Ariel_infinity1, you pick from a list
18:58.51iCEBrkrAriel_: Yeah, what am I looking for on this thing?
18:58.58*** join/#asterisk Anthro (i=gss@pdpc/supporter/active/Anthro)
18:59.07Ariel_I have to go see you all later.
18:59.08AlexCTIIf I use mp3 files wich is the correct format of these files?
18:59.31Ariel_~docs
18:59.32jbotsomebody said docs was Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
19:00.31iCEBrkrI'm quite aware of the docs and the wiki
19:00.53AlexCTIcf-usa-en => custom:/var/lib/asterisk/mohmp3/cf-usa-en,/usr/local/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s
19:02.23*** join/#asterisk ap0ught (n=ap0ught@208.19.226.94)
19:04.06*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
19:05.26*** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com)
19:07.06wunderkinlooks to me like * is sending the rtp to the right ip and port why aint it workin :(
19:07.27infinity1i'm trying to connect * to gizmo. when making a call, i get an error. i haven't been able to figure out a working configuration. error: Got SIP response 488 "Not Acceptable Here" back from 198.65.166.131
19:11.13pauldyahh infinity1 after you left last night I set it up using the config from the voip-info site
19:11.25pauldyworks fine
19:11.31infinity1pauldy: no )(#*) way!
19:11.37pauldyyup
19:11.52pauldyyou running amp?
19:11.57infinity1amp?
19:12.07pauldyasterisk management portal
19:12.08infinity1can you post your sip.conf and your extenion to pastebin.ca
19:12.11infinity1pauldy: no.
19:12.29pauldyit is the sme as whats on voip-info
19:12.42infinity1pauldy: i've tried that! hrmz.
19:12.48pauldyhttp://www.voip-info.org/tiki-index.php?page=Asterisk+settings+Gizmo
19:12.49infinity1pauldy: and many variations
19:13.04*** join/#asterisk fulgas (n=fulgas@a81-84-116-219.cpe.netcabo.pt)
19:13.06infinity1pauldy: what version of * are you useing?
19:13.20pauldy1.0.9
19:13.28infinity1pauldy: i'm using head.
19:13.31pauldywell no scratch that I updated to head last night
19:13.43infinity1ahh ...now i must hurt you :)
19:14.00infinity1can you post your sip.conf and extension anyway?
19:16.02infinity1it works for me if i use xten client. but not * ...heh
19:17.12pauldywhy not post yours to see what is wrong
19:17.16pauldymine has way to much for me to cherry pick out user/pass pairs
19:17.37Renacoranybody know why a polycom phone will not ring?
19:17.59pauldyvolume turned down, no ones calling it, is it plugged in
19:18.44harryvvRenacor it wont ring because you dont have it configured and or dont have a ftp server it log into and copy the image files.
19:20.12infinity1pauldy: http://pastebin.ca/25871
19:20.20infinity1pauldy: mine is a mess. i've tried everything
19:20.44vader-wrkdoes anyone in here use ADIT 600 channel banks in their setup?
19:20.44infinity1pauldy: just post your gizmo and your register line
19:20.46Renacorharryvv it is set up
19:21.27Renacorharryvv: it never gets to SIP/phonename-xxx is ringing
19:21.38Renacorbut it executes the rest of the dialplan
19:25.25*** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net)
19:25.46*** join/#asterisk Los415 (n=los415@64.201.104.186)
19:25.58vader-wrkcan an FXO card only accept phone calls?
19:26.38syleanalog trunk
19:27.45*** part/#asterisk lars-ut (n=me@67.137.31.58)
19:27.46vader-wrksay if i have 8 analog trunk lines
19:28.11vader-wrkdo i need (2) quad FXO cards and (2) quad FXS cards?
19:28.16infinity1anyone have a gizmo phone # i can try
19:28.57snittinfinity1: i have one
19:29.07snittcan you send dtmf inband?
19:29.28snittif no, my number would ring at the livingroom
19:29.39infinity1hmm ..can i try?
19:30.34snittdtmfmode=rfc2833
19:30.38snittwell..
19:30.48snitt1-747-622-2653
19:30.51*** join/#asterisk wzlwzl- (n=wzlwzl@wsip-70-183-60-181.oc.oc.cox.net)
19:30.54wzlwzl-hey all.. have a * setup using 3 broadvoice lines and Polycom IP300 phones... We have 1.5Mbit up and down via cable. 40ms (ave) pings to the bv proxy and no packetloss. The phones sound like cell phones. The person on the other end complains about it cutting in and out. Any help troubleshooting this plz?
19:30.59snittdial 2 when you hear voice
19:31.49snittFUCK
19:31.54infinity1?
19:31.56snitti told u to dial or hangup
19:32.03snittAsterisk Ready.
19:32.09infinity1i dialed. didn't hear anytihng
19:32.19*** join/#asterisk mogorman (n=mogorman@gateway.digium.com) [NETSPLIT VICTIM]
19:32.28snittno. i dont believe that
19:32.37snittit works, maybe not at your side
19:32.39infinity1either my side is broke. or your side is :)
19:35.46*** join/#asterisk Goshen (n=Goshen@c-67-172-238-57.hsd1.ut.comcast.net)
19:38.35infinity1hmm. when calling into mine using gizmo, i don't even get anything on the cli
19:39.13*** part/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com)
19:40.29*** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) [NETSPLIT VICTIM]
19:42.42infinity1i appear to be able to call gizmo users. they just cant call me.
19:43.42*** join/#asterisk juice (n=juice@mo-67-77-176-229.dyn.sprint-hsd.net)
19:46.30*** join/#asterisk YaP (n=YaP@ppp-60-43.27-151.libero.it)
19:46.38YaPhi
19:46.39YaPis it possible to authenticate iax phones using their mac address?
19:47.10AgiNamuwell, if you're on a single ethernet network
19:47.19AgiNamuyou could hack asterisk to do that
19:47.37AgiNamubut mac addresses aren'tr transmitted into TCPIP so...
19:48.46Nuggetwell, there's also the challenge that the server doesn't necessarily see the mac address of the phone.
19:48.51Nuggetit would be quite limiting to presume so
19:50.20*** join/#asterisk zotz (n=zotz@24.231.36.100)
19:51.32YaPNugget: with hack you mean modify asterisk code?
19:51.56YaPi need to authenticate the phone, do you have any other idea?
19:53.50*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
19:55.50*** join/#asterisk wmandra (n=me@pcp04943183pcs.verona01.nj.comcast.net)
19:56.19Igbothom_IIItzafrir_laptop; art thee here?
19:57.51*** join/#asterisk Nexis (n=nexis@12-207-56-108.client.mchsi.com)
19:58.01Igbothom_IIItzafrir_laptop; art thou here?
19:58.54*** join/#asterisk jsaunders (n=jsaunder@d154-5-198-14.bchsia.telus.net)
20:00.52vader-wrkany of you work with channel banks?
20:01.12harryvvnot yet
20:01.26harryvvI hear the rhino channel bank is one of the easiest to work with.
20:01.45vader-wrkid like to do an intermix of channel banks and ip phones/software
20:01.46*** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)
20:01.49generalhanwhats up everyone !
20:02.12vader-wrkive only found a few 24-48 channel banks but i need more connections
20:02.16generalhanim having a serious issue with transfering calls right now between my users ... im getting this NOTICE
20:02.16generalhanOct 18 13:01:07 NOTICE[3710]: app_dial.c:764 dial_exec: Unable to create channel of type 'SIP'
20:02.16generalhan<PROTECTED>
20:02.18vader-wrkcan you hook up more than one channel bank
20:02.39generalhananyone know what the deal is ?
20:03.52Nexisvader-wrk, correct, you can hook up as many banks as you would like
20:04.04Nexisgeneralhan, you do have chan_sip.so loaded correct/
20:04.57*** part/#asterisk jsaunders (n=jsaunder@d154-5-198-14.bchsia.telus.net)
20:04.59generalhanlol i dont know, ive had this asterisk setup running nonstop for the past month with no changes and this just started happening today
20:05.12generalhanshould i just reload everything and see if that fixes it ?
20:05.47generalhanwell that didnt help any so scratch that idea ... crap
20:06.06*** join/#asterisk paryl (n=paryl@209.236.78.59)
20:06.09Nexistry a restart
20:06.16generalhancrap
20:06.16hardwirenever!
20:06.17generalhani cant do that
20:06.19hardwireCRAP!
20:06.23hardwireNOOOOOO!O!O!!
20:06.23generalhani have 20 peopole on the phones right now
20:06.23Nexiswhy cant you restart?
20:06.29Nexiseww
20:06.33generalhanyea no good
20:06.38generalhanlemme see if i can tell them all to get off ! lol
20:06.45hardwireheh
20:06.47Nexiscall center, or can you ask them to logoff for like 3 min?
20:06.49hardwireyou tell them like this
20:06.51vader-wrkwat do you guys think of the ADIT 600 E1?
20:06.52hardwirerestart now
20:07.05vader-wrki was looking at the E1 because you don't need a PRI card supposly just a network card
20:07.23generalhanwhen i restart it do i do a "shutdown now" then a "safe
20:07.27generalhan_asterisk"
20:07.34generalhanor is there a better way ?
20:07.42Nexisstop gracefully
20:07.53Nexisare you running asterisk as a non privlaged user?
20:07.59generalhannope
20:08.01generalhanas root
20:08.08Nexiswow, you have balls.
20:08.19generalhan??
20:08.26hardwireheh
20:08.27Nexisthen yea, you would execute safe_asterisk
20:08.31hardwirerestrart when convenient
20:08.33hardwireor gracefully
20:08.34hardwirenever works
20:08.38hardwirenor shutdown when ...
20:08.50hardwirenever ever has worked correctly for me
20:08.53hardwirealways drops the channels
20:09.03Nexisstop when convenent works good
20:09.03paryli'm looking for a SIP phone with a nice price/feature point for a rollout of about 40 stations.  i tried the gxp2000 in a satellite location, and they just seem a little too flimsy.  any suggestions?
20:09.17hardwireNexis: CVS?
20:09.21Nexisyea
20:09.28hardwirein 1.0.0 1.2.0 and CVS it has never worked correctly
20:09.41hardwirethere will be bridged calls.. then poof.. no calls
20:09.43Nexisparyl, do you need any special features off a sip phone?
20:09.45mutilatori run asterisk with daemon-tools
20:09.47mutilatornever dies
20:09.49mutilator:P
20:10.14Nexisand are your users going to be abusive to the hardware?
20:10.37parylnexis: 2+ lines, standard headset capable, speakerphone
20:10.53paryland, no, they won't be terribly abusive
20:10.57Igbothom_IIIparyl; polycom 501
20:11.03*** join/#asterisk jsaunders (n=jsaunder@d154-5-198-14.bchsia.telus.net)
20:11.08paryli'm going to buy some of the 2000's for the warehouse, etc.
20:11.11Igbothom_IIIor snom 360
20:11.30Nexisparyl, a channel bank would be a choice.
20:11.53paryla channel bank?
20:11.57Igbothom_IIIbut channel banks only allow old analog phones to be used, don't they?
20:12.03parylah
20:12.10Nexiscorrect
20:12.24parylcurrently we're on a siemens system, and i'm replacing it with asterisk... so the phones go with it
20:12.25Nexisbut he could plug old phones in, and drop normal phones around
20:12.27*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal)
20:12.33Nexisahh
20:12.35Igbothom_IIIactually, do they allow old PABX analog phones to be used?
20:12.57*** join/#asterisk newl (n=newlook@203-59-214-216.dyn.iinet.net.au)
20:13.10jsaundersIn a sip 200 OK after a REGISTER from client, is the "Contact" header required?
20:13.11Nexisnot sure, i havent looked into them much
20:13.50paryligbothom: i was looing at the 501... you like it?
20:14.33Nexisparyl, dont know if you have looked at the stuff on voip-info, but that will give you a idea as to how well phones would work.
20:14.34Nexishttp://www.voip-info.org/wiki/view/Asterisk+phones
20:15.25parylyeah... i've done a lot of searching... just wanted to see if anyone could give me real-world views on it
20:15.31harryvvcan anyone recomend a voip phone that has the best tx/rx range?
20:15.35parylpeople swore i'd love the grandstream phone :)
20:15.45harryvvvoip wifi phone with those ranges?
20:15.55Nexishttp://www.mail-archive.com/asterisk-users@lists.digium.com/msg38942.html
20:16.08Nexisthats a good review there also, real world experence.
20:17.11generalhanNexis:i restarted it all and still no go. same error about creating a SIP channel
20:17.19Nexisodd
20:17.50Nexisits only when they try to transfer, or?
20:17.56generalhani dont understand it ! lol. its been working great for over a month now
20:18.24generalhanwheather you try and transfer or just 4 digit extension dial it goes directly to their VM Box and i get that error in the console
20:18.49Nexisand nothing has changed?
20:18.54generalhannope
20:18.55*** part/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
20:19.17Nexiscvs head, or 1.0?
20:19.21generalhannot with asterisk anyway. i just finished installing Festival, but i dont know why that would change anything with asterisk
20:19.25generalhan1.0.9
20:20.05*** join/#asterisk Connor- (n=billy@198-144-174-5.knx.tn.nxs.net)
20:20.20Connor-Hey guys, how much difference is there with HEAD vs 1.2Beta 1 right now?
20:20.32Connor-Was 1.2B1 just a snapshot of the HEAD tree?
20:21.01Nexiscan you set verbose in 1.0.9?
20:21.07generalhanyep
20:21.11generalhanits at 42 right now ! lol
20:21.16Nexisok, set debug 50
20:21.21generalhanlol hang on
20:21.34Nexisshould be able to do it from the CLI
20:21.54generalhank
20:22.05Nexistry it
20:22.06generalhan:Verbosity was 42 and is now 103:
20:22.07Nexissee what you get
20:23.19*** join/#asterisk JASON-0 (n=jason@jason.unitz.ca)
20:23.35generalhanNexis: this is what i get in the console http://generalhan.pastebin.ca/25881
20:23.43JASON-0I am compiling asterisk but it fails at :  /usr/bin/ld: cannot find -lssl
20:24.07harryvvssl modual not installed?
20:24.14NexisJASON-0, install the ssl dev package
20:24.14JASON-0i installed OpenSSL
20:24.18JASON-0and it still does it
20:24.25NexisJASON-0, distro?
20:24.29JASON-0Mandrake
20:24.29generalhanNexis: see anything out of the ordinary ?
20:24.31Nexisgeneralhan, do you have a sip phone on your desk?
20:24.58generalhankinda !
20:25.02generalhanit is now
20:25.05generalhanCisco 7960
20:25.24Nexisreboot it
20:25.44*** join/#asterisk kippi (n=chrisfro@cpc3-hatf3-6-0-cust49.lutn.cable.ntl.com)
20:25.50generalhanok why ?!
20:26.11JASON-0Nexis: I'm using Mandrake
20:26.26Nexisi have a suspicion
20:26.32generalhanwhat am i looking for though ?>
20:26.33NexisJASON-0, you need the ssl dev libs
20:26.40JASON-0where do I get that?
20:26.51Nexisgeneralhan, we are looking to see if the phone is not registered in asterisk
20:27.03Nexisif you can make calls, but not recieve them, it kinda points to 1 of 2 things
20:27.07Nexis1. asterisk needs restarted
20:27.14Nexis2. your phone is not registering with asterisk
20:27.33generalhani wish i would have rebooted a different phone then
20:27.51generalhanmy cisco takes like 3 minutes to start up. the other 25 Aastra phones i have reboot in like 10 seconds
20:27.57Nexisahh
20:28.31JASON-0Nexis: Do you know where I get the SSL dev libs?
20:28.47NexisJASON-0, no, i dont, i dont use a rpm based distro
20:28.58Nexis#mandrake or the forums could help im sure
20:29.05JASON-0ok thank you :)
20:29.19JASON-0#mandrake
20:29.23JASON-0oops
20:30.25generalhanNexis: i rebooted the phone
20:30.31eKo1#mandrake...i thought that distro was dead
20:30.58generalhanbut i had about 20 calls comming in and going out on the console so i couldnt see if it said anything about my phone registering
20:32.03harryvvGeneral. what astra phones u using?
20:32.56generalhan9112i
20:33.06Nexisgeneralhan, any luck?
20:33.16generalhanNexis: nope, still cant transfer inside
20:33.23*** join/#asterisk damned (n=vpol@prior.lanck.net)
20:33.37generalhani cant even call the autoattendant and dial an extension there
20:34.37harryvvgeneralhan so you system is not up and running yet. what did that phone cost you?
20:34.52*** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
20:35.18generalhanthe phones cost only $120 and were well worth it
20:35.28generalhanmy system is having issues right now thats all
20:36.01generalhanNexis: ok new info for you. i can get the transfer to work to 3 of the 30 numbers that i have. so why would it allow me to transfer to 3 people and not the other 27 ?
20:36.12harryvvgeneral not a bad price.
20:36.24harryvvAre you seeing this system up for your office?
20:37.00vader-wrkwhats the different between IAX and SIP?
20:37.27harryvvfirewall issues vader
20:37.31*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
20:38.05generalhanharryvv: no the price is great for what it is. its only a one line phone but it works great. and i have already set my system up, its been running great for over a month now, and just today something went haywire
20:39.07harryvvgeneralhan do you always write down the steps you do in asterisk?
20:39.21generalhanno
20:39.25generalhanwhat steps ?
20:40.14generalhanok i hae to go figure this out ... this is bad ill be back in a bit guys
20:41.18harryvvsee, most errors are created by the person who configures the software. I dont know in if this is the case.
20:41.36*** join/#asterisk Katty (n=katrina@68-112-15-110.dhcp.cpgr.mo.charter.com)
20:41.47Igbothom_IIImeow
20:42.01Kattymew.
20:42.10Igbothom_IIImeowning
20:42.13wzlwzl-hey all.. have a * setup using 3 broadvoice lines and Polycom IP300 phones... We have 1.5Mbit up and down via cable. 40ms (ave) pings to the bv proxy and no packetloss. Using ulaw. The phones sound like cell phones. The person on the other end complains about it cutting in and out. Any help troubleshooting this plz?
20:42.27generalhanharryvv: i agree with you, but this time, i didnt do anything to it. its the same today as it has been for the past month and it just messed up
20:42.33wzlwzl-within the office, ip300->ip300, it sounds great
20:43.39harryvvWz, have you done any preliminary bandwith calculations and mearsurment test before implementing the ipphone system?
20:43.58Igbothom_IIIwzlwzl-; dedicated 1.5/1.5, or shared with other Internet traffic?
20:44.25harryvvgeneralhan look at the error logs on what is is doing.
20:45.06wzlwzl-Igbothom_III: shared, but its the same when all other traffic is stopped... also running QoS (more details of config here: http://www.ospiron.com/ast.txt )
20:45.27Igbothom_III404
20:45.46wzlwzl-you sure? i can get there just fine..
20:46.03*** join/#asterisk tainted_ (n=somewher@mail.k2usa.com)
20:46.12Igbothom_IIIactually, the main page is 404 as well
20:46.16wmandrawz: connection refused
20:46.21wzlwzl-hrm.. 1 sec
20:46.30generalhanharryvv: the error that is comming up is that it cant creat a channel SIP everyone is busy/congested at this time
20:46.33Igbothom_IIIlooks like your website is offline to the ret of the world!  :(
20:46.44wzlwzl-dns issue on that domain
20:46.45*** join/#asterisk Caede (n=caede@sentry.zoom.com)
20:46.49wzlwzl-www.beachrentalsearch.com/ast.txt
20:47.03Igbothom_IIIaha - that damn Level3/Whomever bitchfight is rearing its ugly head again
20:47.22Igbothom_IIIbeachblah works fine
20:47.30wzlwzl-k
20:47.37Igbothom_III~pb
20:47.39jbothmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/
20:47.39harryvvgeneral, busy means that the other line is in use.
20:47.53harryvvor the line is not connected.
20:47.54generalhanright, but they arent
20:47.58generalhanand thats the issue
20:47.58wzlwzl-jbot: yea.. shoulda used that instead
20:48.07groogswzlwzl-: maybe you're not actually getting 1.5mbps
20:48.36Igbothom_IIIno chance of 1.5/1.5 on a 1.5/1.5 pipe anyway
20:48.38Igbothom_III90% tops
20:48.38CaedeAnyone know what might cause a "T1: Lost our place, resyncing" on a zaptel card?
20:48.42generalhanthose phones can make calls and recieve them if they are called directly, but they cant be the recipiant of transfer or an extension dial
20:48.44*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfmuo.dialup.mindspring.com)
20:48.50groogswzlwzl-: try connecting to another SIP/IAX number
20:48.54Caede(one that is repeated about 10,000 times an hour)
20:49.05wzlwzl-1.1mbit according to various speed tests
20:49.10Igbothom_IIIand your overall pipe size needs to be a touch under the maximum speed you get - tried d/ling the latest kernel ands seeing the actual speed?
20:49.33groogswzlwzl-: maybe it's just your connection to broadvoice then. check some tracerts against places you know you can get hte bandwidth
20:49.41wmandrawz, 1) QoS will only work if your ISP supports it. 2) What cable provider are you using??
20:49.42Igbothom_IIIso, your pipe limit set to just under that?
20:49.42groogswzlwzl-: and 1.1mbit down, but what about up?
20:49.50wzlwzl-wmandra: cox
20:49.52wzlwzl-business
20:50.03groogswmandra: no, it will work on your network regardless of your ISP
20:50.26Igbothom_IIIexactly - it will only work OUTSIDE of your LAN if your ISP supports it  :)
20:50.30groogswmandra: ie, if someone on his net is trying to download something, the QoS will place priority on the asterisk traffic and the download will slow down
20:50.38wmandragroogs: it will work on your network yes, but once the packets leave hisr router the qos tags will be ignored
20:51.01groogsyes. and in a perfect world that wouldn't matter, since he has 1.5mbit to the isp ...
20:51.21wmandratrue, 3 calls should only be using about 240kbps
20:51.28Igbothom_IIIbut few ISPs fully implement QoS support
20:51.40wzlwzl-Your download speed : 1396 kbps or 174.5 KB/sec.
20:51.40wzlwzl-Your upload speed : 387 kbps or 48.4 KB/sec.
20:51.45Igbothom_III3 calls should use a LOT less than that if using G.729
20:51.54wzlwzl-guess upstream is getting capped
20:52.01wmandrahe said he was using g.711u
20:52.09Igbothom_IIIaha
20:52.11wzlwzl-but in anycase... even with just 1 call
20:52.13Igbothom_IIIfurry muff
20:52.25wzlwzl-it shouldn't sound choppy/cell-phone-like
20:52.28wzlwzl-no?
20:52.44Igbothom_IIIdepends on issues between your modem and the end user's handset
20:53.00Igbothom_IIIthere are MANY things than can cause issues, unfortunately
20:53.13wmandrawz, to be honest I was using cable for a while here (comcast) and had the same problem, switched to dsl and the problem went away
20:53.54groogsalso they don't really have 1.5 mbit .. ie, if there's 100 subscribers in a given area, they don't have 150mbit dedicated to it .. they have maybe like, 5mbit that has to be shared
20:54.08wzlwzl-groogs: cox business is dedicated
20:54.12Igbothom_IIIissue is possibly that cable is based on the old hub-style architecture - when your neighbor goes at downloads hammer and tongs, you suffer
20:54.32Igbothom_IIIwzlwzl-; no, it isn't
20:54.39groogswzlwzl-: perhaps
20:54.40Igbothom_IIIthey sell it to you that way...
20:54.45*** part/#asterisk Caede (n=caede@sentry.zoom.com)
20:55.03wzlwzl-anyhow, that's a tangent...
20:55.09*** join/#asterisk jbenson (n=jbenson@genpubad.gotadsl.co.uk)
20:55.09groogsthey might also allocate 2 mbit of a 5 mbit for business only .. and call it dedicated
20:55.09wzlwzl-is there any way to troubleshoot this
20:55.10groogswho know
20:55.12wmandrawz: my first suggestion would be to try g729
20:55.12wzlwzl-to isolate the problem
20:55.17*** join/#asterisk spiekey (n=spiekey@p549D1B88.dip0.t-ipconnect.de)
20:55.26groogswzlwzl-: yes, call another SIP/IAX number/service and see if the same thing happens
20:55.35Igbothom_IIIalso, try calling someone else on Cox business and see if that's fine
20:55.35groogslike the digium test number
20:55.47Kattyheh, i read that as digium - take a number.
20:55.48wmandrawz: or try a call through a different provider
20:55.53Igbothom_IIIif that works, then it is the link between Cox Business and the rest of the world where the issue is
20:56.13Kattytwisted[asteria]: or more like, asteria - take a number and get in line for hysteria!
20:56.19groogsIgbothom_III: or the link between cox and his provider
20:56.19spiekeyzapta.conf is for the hardware configuration only, right?
20:56.24Kattytwisted[asteria]: oops, did i say that outloud?
20:56.34Igbothom_IIIgroogs; his provider IS cox
20:56.43groogssorry, i mean voip provider
20:56.48Igbothom_III:)
20:57.03Igbothom_IIIyeah, thsat is the link between Cox and the outside world, still
20:57.04jbensonHi All - I am trying to set up a connection to FWD via IAX, but I am getting an error on the console: - Host 'iax2.fwdnet.net/496460' not found at line 17.  Is anyone else having this problem please?
20:57.50groogsIgbothom_III: or PART of the outside world.. presumably a 'big' provider like cox (big enough that i've heard of them before, though i live in canada) has more than one connection
20:58.10Igbothom_IIIyeah, still relates to their connection to the outside world!
20:58.39wmandrawz: does the problem happen all the time or only during peek hours?
20:58.48wzlwzl-wmandra: all the time
20:58.57*** join/#asterisk folsson (n=filip@lund-meje-sr0-vl101-249.perspektivbredband.net)
20:59.03wzlwzl-even when everything else is taken off the network, and i make a single call
20:59.15jbensonor, does anyone here connect to FWD via IAX, please?
20:59.19Igbothom_IIIsure your users aren't playing with you and talking in a choppy, low quality voice?  :)
20:59.29wzlwzl-lol... yea, im sure =)
20:59.33Igbothom_IIIhehe
20:59.40wmandraigbothom: lmao
20:59.56scoatesjbenson: http://www.freeworlddialup.com/community/support/iax.php
21:00.05wzlwzl-the other problem i have, and i dunno if its related or not... on inbound calls from the bv numbers into the auto attendant... it picks up duplicate DTMF
21:00.29wzlwzl-but somewhere i read that's just an issue with bv and not much can be done about it
21:01.01wmandrawz: using bv here with no problems
21:01.03Inv_arphmm.. cant get this sometimes asterisk just gets unregistered to my sip provider , sip reload fixes it
21:01.16jbensonthanks scoates.  I have that all set up, but I get an error Host 'iax2.fwdnet.net/496460' not found at line 17.  What does that mean exactly?  The host does "exist", but is the login being rejected?
21:03.24wmandrawz: you can try setting qualify=yes under the section for bv in your sip.conf then use sip show peers from the CLI
21:03.26spiekeyzapta.conf is for the hardware configuration only, right?
21:03.26Inv_arpjbenson: hmm the host is iax2.fwdnet.net  not iax2.fwdnet.net/496460
21:05.07spiekeyok...zapta.conf is the config file i will have to edit when i first install/configure asterisk. or where woudl you start?
21:06.07Igbothom_IIIspiekey; zapata.conf is to configure the Zaptel ATA card
21:06.27Igbothom_IIIsip.conf and extensions.conf are two of the main files
21:07.28wmandraspiekey: http://voip-info.org/wiki/view/Asterisk+config+files
21:09.39Renacoranybody familiar with polycom phones know how to make the phone re-read the .cfg file when you reboot it? for some reason it is not updating it from the tftp
21:12.06*** join/#asterisk JohnnyC (n=JoaoCorr@195-23-115-68.net.novis.pt)
21:12.12JohnnyChellop
21:16.22*** join/#asterisk ohad (n=ohad@19-231-13-72.cosmoweb.net)
21:16.44ohadhow do i add more ringtons to my BT 101?
21:17.16Igbothom_IIIBT 101, as it is the cheapest of the cheap, I'd doubt you can
21:20.40*** join/#asterisk dos000 (n=dos000@i216-58-60-251.cybersurf.com)
21:23.56ohadIgbothom_III, i am sure i can. the question is how:)
21:27.33*** join/#asterisk SarahEmm (n=sarahemm@Toronto-HSE-ppp3685577.sympatico.ca)
21:28.10tehdelygah
21:28.11tehdelydfasdfasdf
21:28.19tehdelydigium replaced my two TDM400Ps
21:28.21tehdelyinsalled the new ones today
21:28.23tehdelystill crackling
21:28.24tehdely:(
21:28.30tehdelyzttest gives consistent 99.98 - 100%
21:29.51*** join/#asterisk delink (n=delink@ziegchen.delink.net)
21:31.17ursuspacificusRenacor: Re Polycom phone config files... if you're using TFTP, you need to change the filenames on the bootserver... I usually just symlink... so... FE if you changed file 'phone1234.cfg' you will need to 'ln -s phone1234.cfg phone1234.cfg.r001' or something like that and update 000412345678.cfg to point to the updated filename.
21:32.13*** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca)
21:33.10steve___Is there something other than callerid that a telco can send that isn't blockable?
21:33.44*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
21:35.05enderFtp is the way to go for Polycom.  not tftp
21:35.53Kattytftp--
21:35.55Katty^- skeery.
21:37.04ohadanyone, help with canceling  echo on asterisk?
21:37.16ohadi am using PRI and therefore shouldn't have any ech
21:37.16ohado
21:39.59Nexisgeneralhan, sorry about runnin off, you figure it out?
21:41.19*** join/#asterisk diclophis (n=diclophi@adsl-69-238-124-226.dsl.pltn13.pacbell.net)
21:41.22diclophishello
21:41.27*** join/#asterisk darkskiez (n=darkskie@host86-138-169-183.range86-138.btcentralplus.com)
21:41.34diclophiswhat is the standard way of switching contexts in a call?
21:41.49devela question that somebody may be able to address quickly: we have a t1 (not PRI), and no callerid data.  the calls come in with a text label of "asterisk", where does that come from?
21:41.50diclophisi thought it was the Goto command, however I keep getting this error
21:42.03diclophisWARNING[3170]: pbx.c:2278 __ast_pbx_run: Channel 'SIP/212-62d9' sent into invalid extension '600' in context 'alt', but no invalid handler
21:42.06spiekeyi have asterisk here with a ISDN card. i finished configuring zapata.conf. Which config file should i configure next?
21:42.14diclophisAlso note that I am using asterisk Realtime config...
21:43.14Renacoranybody familiar with polycom phones know how to make the phone re-read the .cfg file when you reboot it? for some reason it is not updating it from the tftp
21:43.24darkskiezhas anyone used asterisk head on osx in the last couple of weeks?
21:45.00enderRenacor: tftp the file would have to be renamed.
21:45.08enderRenacor: honestly, you _really_ should use ftp instead.
21:45.34enderRenacor: ftp will work off of timestamps and work easier for uploading files/configs/directories/etc...
21:49.27*** join/#asterisk wmandra (n=me@pcp04943183pcs.verona01.nj.comcast.net)
21:49.42asterisk99Q:   I am trying to d/l STABLE 1.2 from CVS --- I am geting an error "no such tag v1-2"   Ideas? Hints?
21:50.08MikeJ[Laptop]there is only 1.2beta1
21:50.13MikeJ[Laptop]and it is not called stable
21:50.21MikeJ[Laptop]in fact, there is nothing called stable
21:50.42MikeJ[Laptop]you can snag the tarball for beta1 off the ftp site or grab it by it's tag
21:52.26kippidid anyone have problems when installing mysql? for AMP?
21:52.56*** join/#asterisk ManxPower (n=ewieling@adsl-67-65-233-194.dsl.lgvwtx.swbell.net)
21:55.55*** join/#asterisk wunderkin (i=kev@0-1pool199-201.nas26.tempe1.az.us.da.qwest.net)
21:56.14*** join/#asterisk scud (n=scud@12-214-190-139.client.mchsi.com)
21:56.32mmlj4hey PacketLoss :-)
21:57.14pauldykippi, I had major problems because I decided to clear a table after I installed it
21:57.34pauldyapparently it needs it default data or it starts acting wierd
21:59.32nobellany suggestions on a good web site that has perl agi examples for ivr stuff?
21:59.36spiekeywhats app_capiCD.so good for?
21:59.55spiekeyi get a "Loading module app_capiCD.so failed!" but i am not even sure if i need that module
22:01.32kippii just keep on getting this damm error Timeout error occurred trying to start MySQL Daemon. and it is stoping my asterisk time :(
22:03.39pauldykippi, sounds like you don't even have asterisk running right
22:03.52pauldycheck your my.conf and make sure everythings good
22:03.55pauldyhave to run again
22:06.28spiekeywhat does that mean? "loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/app_capiCD.so: undefined symbol: ast_capi_MessageNumber"
22:06.38spiekeyundefined symbol?!
22:07.44diclophisanyone using extension realtime config?
22:07.47*** join/#asterisk Miggidy (n=Miggidy@203-59-9-189.perm.iinet.net.au)
22:07.59Igbothom_IIIundefined symbol - the source code is buggy
22:08.00MikeJ[Laptop]it means it does not know what that is
22:08.07MikeJ[Laptop]where is that function?
22:08.26MikeJ[Laptop]somthing you are not including in -l when you are compiling\linking
22:08.32*** join/#asterisk Shaun2222 (n=ndci@ip68-111-70-41.oc.oc.cox.net)
22:08.35spiekeyi am currently using the version which comes with debian sarge
22:09.06spiekeyi guess i will have to switch to the currend tarballed version ;)
22:16.31zedasspiekey, see if debian has a bunch of other required packages you need to install first.
22:16.47zedasspiekey, debian is famous for doing lots of little packages without dependencies.
22:17.36hardwireblah
22:20.55*** join/#asterisk santiago (n=santiago@63.245.87.62)
22:21.24*** join/#asterisk mxmasster (n=mxmasste@pitbull.media.net)
22:21.26mxmassterhi all
22:21.47mxmassteri am looking for docs on how to configure DID to people's extensions - can anyone point me to a url?
22:22.28Inv_arpmxmasster: do you have the DID yet?
22:22.35mxmassteryes
22:22.49Inv_arpmxmasster: sip or iax?
22:22.59mxmassterthat is not the problem, given a block of x sip did's what is the easiest way to sent them to a phone
22:24.30*** part/#asterisk bob_too (n=chris@rrcs-24-153-179-246.sw.biz.rr.com)
22:24.50generalhanmxmasster: exten => YOUR_DID_NUMBER,1,Dial(SIP/User,20,tT) will work just fine
22:25.22mxmasstergeneralhan: is their a shortcut for this - i would assume that a larger organization wouldn't define 100+ did's this way
22:25.43*** part/#asterisk mkrufky (n=mk@68.160.103.77)
22:26.01generalhannot that i know of ... my company isnt HUGE but i have a block of 50 DIDs and i have programmed 20 of them in this manner
22:26.35pauldyyou can us a macro and pattern
22:26.55generalhaneww thats right, but youll need to talk to someone else about macros ... i dont touch them ! lol
22:26.56pauldy_55555512XX,1
22:27.43*** join/#asterisk cp5 (n=samy@adsl-69-110-135-49.dsl.irvnca.pacbell.net)
22:27.58generalhanpauldy: but that is 100 possible numbers what if all those 100 numbers he has and needs each to go to a different place ?
22:28.20hardwirehmm
22:28.27hardwirechannel based volume normalizationw ould rule
22:29.12generalhanpauldy: not that i know how to do it or anything but i would love to know how for the future. we might be getting another 50 DIDs and i would like to know how to get it done properly
22:31.02mxmassterpauldy: do you know of a public example configuration that show's how to do this?
22:31.25pauldynope
22:31.43cp5is CVS zaptel's fxsdump broken?
22:31.48pauldyjust know thats one way to do it if you were one to one mapping extensions to a did
22:31.50cp5it's expecting a missing header
22:33.25Renacoranybody know where I can get the xml example for the polycom setup cfg's?
22:33.28cp5coeffs.h
22:34.16pauldyif your doing 100 dids though a simple script could generate the dial lines for you
22:35.07wzlwzl-when editing config files like extensions.conf... can i do reload now, or do i need to restart now
22:37.34*** join/#asterisk K-Bear (n=k-bear@h29n2fls32o815.telia.com)
22:37.36*** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net)
22:39.19*** part/#asterisk jsaunders (n=jsaunder@d154-5-198-14.bchsia.telus.net)
22:42.22K-BearI'm having a pretty odd problem. I'm setting up an IAXy. I've manage to provision it and I've set up my Asterisk server to have it accepted. It's working to the point that I get a dial tone, I can dial all the extensions on the PBX. But when a call is connected, even when it's local in my own network, I don't get any sound from the device connected to the IAXy. I can hear the person at the extension I dialed, but they can't hear the IAXy. And
22:42.22K-Bearafter 15s Asterisk hangs up. Anyone know what's wrong?
22:44.13syleit doesn't like you?
22:44.21K-BearHeh :-)
22:45.02*** join/#asterisk nagl (n=nagl@213.235.241.6)
22:46.01*** join/#asterisk stevek (n=stevek@slim-eth0.horizonlive.net)
22:49.18*** part/#asterisk Utah_Dave (n=boucha@0-1pool138-165.nas28.salt-lake-city1.ut.us.da.qwest.net)
22:50.16*** join/#asterisk n3u7 (n=neutrin0@CPE000d8802a707-CM0011e6c7edb1.cpe.net.cable.rogers.com)
22:51.05cp5anyone familiar with the queue/agent deadlocking bugs?
22:51.17n3u7lokking for answers:
22:51.20n3u7http://forums.digium.com/viewtopic.php?t=1973&highlight=suse9+3
22:56.36*** join/#asterisk jbenson (n=jbenson@genpubad.gotadsl.co.uk)
23:00.50*** join/#asterisk _tekati_ (n=captain@cpe-66-75-215-63.bak.res.rr.com)
23:01.59*** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker)
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23:05.24*** join/#asterisk bob_too (n=chris@rrcs-24-153-179-246.sw.biz.rr.com)
23:07.08l1nuxis possible use modem without sound (without load oss or alsa modules) ?
23:09.08*** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net)
23:10.46Kattymrow.
23:11.11SarahEmmkatty!
23:11.12SarahEmmmeow!
23:11.29SarahEmm*purrrr*
23:12.09*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
23:12.18KattyAriel_: hihi
23:12.28Ariel_Katty, hello
23:13.13*** join/#asterisk christo (n=chris@brezhnev.spiration.co.uk)
23:13.16*** part/#asterisk nobell (n=jdegraff@67.137.31.58)
23:13.43*** join/#asterisk lilneon (n=tj_r3@cuscon12932.tstt.net.tt)
23:14.13lilneonhey everyone.. my RH9 asterisk installation not detecting my new digium tdm21B card
23:14.17lilneonany ideas?
23:14.39Ariel_just stopping by to say hello. On my way to see mom.
23:15.03KattyAriel_: talk to you later (=
23:15.32lilneonok so no one else ever ran into this problem?
23:15.38lilneonis my card bad?
23:15.59Kattylilneon: perhaps you should call digium.
23:16.41lilneonemailed em.. no response yet
23:16.59Kattyk
23:17.04*** join/#asterisk K-Bear (n=k-bear@h29n2fls32o815.telia.com)
23:17.56twisted[asteria]lilneon, what version pci bus?
23:18.07twisted[asteria]lilneon, and does the card show up in lspci?
23:18.16lilneonhave no idea really..  cant find the manual for this old board
23:18.18twisted[asteria]also, did you build and install, and the load the zaptel modules?
23:18.35lilneonyeah modprobe zapata goes by without a hitch
23:18.42twisted[asteria]it's not zapata
23:19.04lilneoni mean zaptel
23:19.04*** join/#asterisk bjohnson (n=bjohnson@i216-58-16-41.cybersurf.com)
23:19.05lilneonmodprobe zaptel
23:19.08twisted[asteria]you still need to actually load the driver for the card
23:19.10twisted[asteria]not just zaptel
23:19.12*** join/#asterisk wmandra (n=me@pcp04943183pcs.verona01.nj.comcast.net)
23:19.20twisted[asteria]in this case, wctdm
23:19.30lilneonthat's when i get errors
23:19.43lilneoninsmod failed... blah blah
23:19.58lilneonso i've been trying to tweak but nothing
23:20.11twisted[asteria]does it show up in lspci?
23:20.16twisted[asteria]do you have power connected?
23:20.18lilneonno it doesnt
23:20.24lilneonyes power is connected to it
23:20.25twisted[asteria]ahhh
23:20.29lilneoneven switched the pci slots
23:20.31twisted[asteria]it doesn't even show up in lspci?
23:20.36lilneonnope
23:20.37Kattylilneon: are they on the same iqr?
23:20.39Kattyi mean irq
23:20.41Kattyif i could type.
23:20.43twisted[asteria]chances are then that you have an older pci rev
23:20.49twisted[asteria]and it won't support the card
23:20.55SarahEmmit sounds like it's not the right PCI rev, and you can't run that board then
23:21.05twisted[asteria]SarahEmm, :P
23:21.11lilneondamn it....
23:21.15wwalkerDoes anyone use asterisk on Windows servers?  I'm a Linux/*nix/*BSD biggot and need someone else's viewpoint.
23:21.40lilneongot another pc but that is only for windows.. and running linux in vmware on the box i heard asterisk wont wrk
23:21.46X-Robwwalker - it doesn't work on windows.
23:21.47twisted[asteria]that's correct
23:21.57twisted[asteria]linux in vmware will not work since you do not have direct access to the hardware
23:21.58twisted[asteria]well
23:22.00twisted[asteria]let me rephrase
23:22.05lilneonso basically i am screwed
23:22.06twisted[asteria]zaptel under linux in vmware
23:22.17Kattyfood network is telling me i must have two different types of peelers in the house at all times.
23:22.19lilneonoh ok..
23:22.27wwalkerX-Rob SCORE!
23:22.36wwalkerX-Rob, Thank you!!!!!!!
23:22.49twisted[asteria]X-Rob, well, actually.....
23:22.54twisted[asteria]X-Rob, it does
23:22.58twisted[asteria]X-Rob, but not natively
23:23.11lilneondoes digium have a list of mobos their cards work on?
23:23.26lilneondont want to go get another mb and run into the same problem
23:23.28twisted[asteria]lilneon, no, but they have a list of systems it won't
23:23.42twisted[asteria]lilneon, make sure you have pci rev 2.2 or higher
23:24.05*** join/#asterisk E|nyPRI_ (n=les@205-200-14-92.static.mts.net)
23:24.08lilneontwisted:.. how would i tell the difference?
23:24.10wwalkertwisted, what do you mean?
23:24.11E|nyPRI_hi
23:24.19twisted[asteria]lilneon, it'll be in your documentation, and possibly on your bios/post screen
23:24.34lilneonk
23:24.40twisted[asteria]wwalker, astwind
23:24.47twisted[asteria]wwalker, er, astwin
23:24.54twisted[asteria]google for it
23:24.57twisted[asteria]asterisk on windos
23:24.59twisted[asteria]windows, too.
23:25.08lilneonastwin doesnt allow u to add any hardware though..
23:25.14twisted[asteria]correct
23:25.15lilneonright twisted?
23:25.31twisted[asteria]YEAH TOAST!
23:25.36lilneonthis sucks sooo bad
23:25.51E|nyPRI_any canadians here?
23:25.54twisted[asteria]lilneon, next time, read the manual :P
23:26.06twisted[asteria]lilneon, most newer mobo's have pci 2.2/2.3
23:26.25lilneonwell like i said this was an old mb..
23:26.31lilneonwas wrking fine with a quicknet phonejack
23:26.37twisted[asteria]lilneon, does it have ISA?
23:26.37Kattytwisted[asteria]: bob and tom are in cape, kthx.
23:26.40lilneondecided to try the digium cads
23:26.44twisted[asteria]Katty, *nods*
23:26.46lilneonyeah it does
23:26.55twisted[asteria]ahh, yeah, probably 2.1 or 2.0
23:27.05lilneonk\
23:27.24twisted[asteria]best way to actually check the card though, is to download a live cd, and then boot a newer box to the livecd.
23:27.36twisted[asteria]if it shows up in lspci from the livecd, you'll know the card is good
23:27.45twisted[asteria]well, at least, that it is detectable
23:28.03*** join/#asterisk marc324 (n=marc3234@206-248-135-84.dsl.teksavvy.com)
23:28.06*** join/#asterisk wolfson (n=hehe@usr-kdh-208-6-58-26.beachlink.com)
23:28.14twisted[asteria]because there is the very rare possibility that the card is just hosed, but it's doubtful based on what you've stated here
23:28.39*** join/#asterisk Clearwire (n=a@206.173.47.7)
23:29.31lilneonok
23:29.40lilneonwait.. can i run live cd from inside windows?
23:29.42lilneon??
23:30.00X-Robno
23:30.01sylehmm strange thought
23:30.07syleanyone met a programmer over 40?
23:30.08X-Robyou don't run anything 'inside' windows.
23:30.16lilneonsyle : yeah
23:30.21Ariel_syle, yes
23:30.24syleseems people have stopped programming alot by that age to me
23:30.35X-Robyou boot from a CD that contains a different operating system and doesn't write to your hard drive
23:30.43ClearwireHello, can anyone tell me why the 'Read' application command stops execution in the dial plan if it returns 'User Disconnected'?
23:30.52*** join/#asterisk _DAW (n=bob@adsl-222-51-199.msy.bellsouth.net)
23:31.27dos000syle, by that time they already moved to management or got nixed
23:32.23syletrue enough
23:32.36sylethey becomes teachers or owners usually
23:33.30syleidk i;ll see where i am in 10 years and get back to you on it lol
23:33.40dos000syle, i have seen one tho. until he was shown the door for trying to stick to old teknology !
23:34.09*** part/#asterisk diclophis (n=diclophi@adsl-69-238-124-226.dsl.pltn13.pacbell.net)
23:34.25sylei think at a certain age you want the learning to stop
23:34.36sylethis is wrong field for that
23:34.47E|nyPRI_at a certain age you realise its all bullshit
23:35.12E|nyPRI_is this #philosophy? or #asterisk :P
23:35.29sylei;ll get back to you philosophy when i have smoke a joint or 2
23:35.33rayvdMars.
23:36.39wwalkersyle, I'm over 40, only been programming since I was 13...(29 years)
23:36.56syleno shit
23:36.57marc324programming dulls the mind.
23:37.09sylei bet you can program in any language then
23:37.53wwalkersyle, you are correct.  We get paid less than plumbers but work 40 to 80 hours a week and have to "learn/keep up with new tech" between 10 and 30 hours a week.
23:38.16lilneonyou said it wwalker
23:38.45syleyeah the keeping up sucks
23:38.54sylerather watch tv for those 10-30 hours somedays
23:38.58lilneonplus with all the  opensource and sites like hotscripts etc.. my 'boss' usually thinks its just a matter of googling and copying and pasting the code
23:39.02wwalkerA few.  6502, x86 (barely), Pascal, Perl, C, C++, Java, *sh, but spend 98% of my programming in perl
23:39.17lilneonso he expects alot to be done in one day
23:39.32wwalkerWe are the lowest paid workers on the market with any actual skill
23:39.57sylehave you considered working for company making games?
23:40.14sylethats where the money is
23:40.29wwalkerHahahahahahahahahahahahahahahahahahahahahahahahahahahahahahaha
23:40.37wwalkersyle, THAT's funny!
23:40.46sylec++ and some grade 11 trig seems to be
23:40.46ClearwireOnly if he writes/publishes/markets it himself
23:40.59sylefriend makes about 150k a year is toronto c programming games
23:41.05lilneonhere... only really got web application opportunities with a few console / network  apps popping up here and there
23:41.20lilneonno games.. besides simple flash and  or java crap for me..
23:41.42lilneonbut wwalker.. u take the cake man.. been programming 29 yrs? i am not even 29 yrs old
23:41.49wwalkerIf you worked for Id and all the money is shared between 8 guys, BIG money.  But EA and the rest of the market make game developers into slave labor.  $40K US for 80-100 work weeks, and fire them the day the game ships
23:42.18syleouch
23:42.20lilneonouch
23:42.24sylewell thats more like contract
23:42.29wwalkersyle, Your friend must be  VERY good to be able to get any serious money out of a gaming dev company.  That is NOT the norm
23:42.39lilneonwait but doesnt salary bracket increase with experience wwalker?
23:42.44syleyeah he was in MIT at 13
23:42.53wwalkersenior game devs get $48K
23:42.57lilneonsyle-wat?
23:42.57*** join/#asterisk danowoz (n=danowoz@wsip-24-249-162-89.ph.ph.cox.net)
23:43.00E|nyPRI_an MIT grad only making 150k ?
23:43.21lilneonmy life is soo unfulfilled now.. sigh
23:43.27enderwwalker: that really depends on the dev house.
23:43.41sylei didn;t say he still does it lol
23:43.42enderwwalker: we're a game dev house but our devs and artists and even IT guys (me) are very well paid.
23:43.52sylenow he usually day trades on the stock market
23:43.56X-Rob"GET /gxp2000.bin HTTP/1.0" 200 12695 "-" "Grandstream GXP2000 1.0.1.13"
23:43.59X-RobWEeeeee!
23:44.03wwalkerlilneon, That's OK, I listened to Robert Frost - "took the road less traveled by"  I am still a developer, not a lead, not a manager
23:44.34*** join/#asterisk deezed (i=none@adsl-065-006-189-182.sip.bct.bellsouth.net)
23:44.46danowozHi, I have asterisk box set up with two extensions. both extensions are registered and I am able to make outbound calls via an outbound trunk. but whne I recieve incomming calls from an external service and when I dial extension to extension my calls are sent staight to voicemail. can anyone help?
23:44.53wwalkerender - sweet.  I know some places do that.  My President ran two game companies and paid his people well, but it's Not the industry norm :(
23:44.55lilneonwell i like programming..
23:45.17X-Robdanowoz - sounds like you're using asterisk@home.
23:45.22lilneonfeel like i actually did somethingafter the project done and i go thru it
23:45.33danowoz:/
23:45.40wwalkerI still program because I enjoy it.  The hours suck, but I do half of them on the couch with Buffy, SG1 or the like as video wallpaper.
23:45.40enderwwalker: right.  We're Casual games too, so it's a different market.
23:45.50danowozyes, how can I fix this problem?
23:45.52enderwwalker: still very good incom.
23:46.12syleidea of a programmer to begin with is you learn the trade to create a company of programmers down the road to build some nice software
23:46.17X-Robdanowoz - you _are_ using asterisk@home?
23:46.29danowozX-Rob - yes
23:46.49lilneonsyle : kinda what most programmers end up doing
23:46.53wwalkerender, glad someone has a good gaming desgin gig.  But I got sick of running purify and valgrind type tools long ago.  As I aged I got lazy.  Perl, backend work, 1/10 the lines, 1/5 the debugging.
23:47.03lilneonbut there are some who just cant stop coding
23:47.04X-Robdanowoz - try #amportal
23:47.10X-RobI'm on there but I Can't help you. Someone elsemight.
23:47.39syleunfortunately that means building some nice income on the side these days while working at your day job to do that
23:48.06danowozX-Rob - is this a problem with @home? or amp in particular, cause I am not agains editing config files by hand.
23:48.18lilneonsyle: kinda burns you out after a while though
23:48.35wwalkerif I won the lottery tomorrow, I'd still be in the office Th and Fr (except for meeting with an accountan t and a lawyer).  Then I'd invest enough in my current gig to control dev.  Then I'd sit back down and start typing again.
23:48.44lilneonit will be like u have three jobs.. each demanding 40-80hrs per week
23:49.09sylei think that is how the american dream is built ....good paying job, develop hobby site on the side, site or products grows over the years, don't need dayjob anymore--quit!, hire people, incorporate
23:49.15wwalker"running your own business" == "you can work whichever 100 hours a week you wish"
23:49.21X-Robdanowoz - it's probably a problem with dialparties.agi
23:49.36X-Robbut still, get someone to help you on #amportal
23:50.16danowozX-Rob - thanks
23:50.44sylewwalker is that any different from working 8 hours a day programming, and how many hours keeping up on tech?
23:50.53lilneonwwalker: yeah was hoping asterisk adn voip would have been a nice side business.. to grow...  spending more tiem on it than i do @wrk
23:53.23znoGargh, if only my provider didn't use https to download its config
23:53.29znoGi could have the sipura config in my hands right nowwwww
23:54.36enderwwalker: hehe, yeah, thats why I do IT and tools development, rather than being a full time developer.
23:54.57lilneonender:IT and tools??? explain
23:55.18file[laptop]I have mail, who wants to delete it for me?
23:55.29enderlilneon: Information Technology, the name that gets slapped on system admins.  Tools are jsut that.  Tools that a sysadmin uses to complete tasks.
23:55.36enderlilneon: I manage servers.
23:55.38X-Robfile, whats your pop3 passworD? 8)
23:55.42X-RobI'll sort it out for you!
23:55.45file[laptop]pop3? pfft, I use imap
23:55.53X-Rob1 LOGIN file@is.hawt.com
23:56.00X-Rob1 AUTH s3xx0r
23:56.03lilneonender: oh.. net Admin??
23:56.22enderlilneon: net, phone, hardware, etc...
23:56.37sylei think we know how to read an rfc but thx for update hehe
23:56.41twisted[asteria]oh no
23:56.43enderX-Rob: I'd like to see you type it out in Imaps  (:
23:56.45twisted[asteria]not more file hittingupon
23:57.02file[laptop]my poor inodes
23:57.21file[laptop]what
23:57.23file[laptop]did it really?
23:57.26twisted[asteria]haha yeah
23:57.30file[laptop]you bastard
23:57.32file[laptop]you killed a Powerbook!
23:57.35twisted[asteria]it went to sleep
23:57.37twisted[asteria]and didnt' wake up
23:57.47twisted[asteria]i didn't kill it
23:57.57syleidk if imap is that popular though...i seen qmail+vpopmail+pop3 but not imap
23:57.58twisted[asteria]it was sitting here just fine, and then fell asleep, and that was the end
23:58.00file[laptop]killall -9 powerbook
23:58.11file[laptop]syle: I like imap... I never used to
23:58.16twisted[asteria]file, i'm getting a brand new one
23:58.23file[laptop]but I'm at three computers usually and they all stay in sync
23:58.24twisted[asteria]apple is announcing the new powerbook line tomorrow
23:58.25n3u7i'm working on installing Asterisk on SuSE9.3
23:58.35twisted[asteria]and the store is going to replace mine with one of the newly announced ones
23:58.36n3u7:)
23:58.41syleanyone have any idea what digium and intel are up to?
23:58.41file[laptop]twisted[asteria]: yayyyyy
23:58.45twisted[asteria]file[laptop], yea :)
23:59.04twisted[asteria]might have to wait a few days to get it
23:59.08twisted[asteria]but it'll be worth it in the long run
23:59.23twisted[asteria]1900x1200 native resolution :)
23:59.29file[laptop]mmm
23:59.57twisted[asteria]thats a lot of pixels

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