irclog2html for #asterisk on 20050418

00:00.14Juxtnewsole: what kind of volume are you ready to handle?
00:00.34NewSolewell we have 3 servers on 100MB backbones
00:00.44Romikspeex, ILBC ?
00:00.45Juxtthat doesn't say anything
00:01.01Juxtunless you're doing transcoding or something
00:01.07NewSoleall three have Dual Xeon CPUs and 4GB Ram...
00:01.15Juxtnewsole: are you terminating the calls yourself?
00:01.22cypromiswhatfor you need 4GB of ram on a * server ?
00:01.51NewSoleyes we have about 15 PRIs hooked up now plus we have a large scale terminstaion
00:01.53Romiksomebody can advice how i do check that timer working on asterisk?
00:02.45toppingNewSole: wouldn't it have made more sense to get a fractional DS3?
00:03.21NewSoleright now we have 5 Pri's at 3 Different locations
00:03.28toppingah ic
00:03.38NewSolebut having one problem
00:03.53NewSoledialplans are no longer letting us link
00:06.07NewSoletrying to ffigure out why... and wiki is down
00:06.25Juxtnewsole: use google cache
00:09.36NewSolebeen trying
00:20.00NewSolehmmm... dead in here... WiKi goes Dead and Everyone goes Dead...
00:20.16toppingi prefer phish
00:21.40*** part/#asterisk Juxt (~Juxt@sfl-dsl-64-135-113-4-cust.host.net)
00:22.14toppingyou seem to have it all... GF, big honkin servers... making everyone jealous ;-)
00:22.31shmaltzwiki down?
00:22.43NewSoleI said prefer.... dont mean I have one
00:23.52toppingthen i guess "seem" is an appropriate word
00:23.52NewSolelast one left me cause I spent too much time on computer
00:24.06toppingif you were into teledildonics, she might have liked it that way
00:24.22NewSoleshe thought i was more inyo cyber then sex..... my poor floppy
00:24.45NewSolelol
00:25.21NewSoletopping... you any good at dial plans
00:25.24toppingit's all about changing adversity into opportunity
00:25.32toppinghehe, no, sorry
00:25.52NewSolehmmm... I am lost on this one
00:26.08toppingif you describe it, eventually someone might have an idea tho
00:27.34decwhat's wrong, NewSole ?
00:29.09*** join/#asterisk fr3dd (~fredd@cpe-024-168-231-049.sc.res.rr.com)
00:29.15shmaltzanybody here having trouble accessing the wiki?
00:29.26NewSoleI am dialing my server
00:29.59NewSoleusing IAX2/master@master/1NXXNXXXXXX@context
00:30.12NewSoleand getting No Auth Found
00:31.50deconly thing that comes to mind is that the master user is set up right
00:32.01shmaltzanybody here having trouble accessing the wiki?
00:32.04deconly thing that comes to mind is to check that the master user is set up right
00:32.08decshmaltz: yes, its down
00:32.14shmaltzthanks dec
00:32.17decapparently.
00:32.33shmaltzanybody have a cached list or link to the extended sound files added to cvs?
00:33.01NewSoleit is thats the problem
00:33.58shmaltzthanks guys such a file exists in /usr/src/asterisk-sounds/sounds-extra.txt if you did a cvs co asterisk-sounds
00:36.20*** join/#asterisk astoria (~asto@68.77.110.194)
00:37.26astoriaI have a question, perhaps out of the realm of asterisk.. how are SMS short-codes (ex. 45645) handled? Is there some kind of carrier-based routing between that number and a GSM number? Thanks in advance!
00:43.07sudoeranyone here use broadvoice?
00:44.45NewSolethis is anoying....
00:45.51sudoervoip-info being down is annoying
00:46.03*** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net)
00:46.27want561or772didneed did pls thank you
00:48.52hermiewant561or772did: you might try in #asterisk-biz or on the -biz mailing list
00:50.54file[laptop]sex?
00:55.40want561or772didok
01:02.29NewSoleAny one here good at Dialplans
01:04.51file[laptop]lots of people are
01:06.21cypromisnah
01:06.23cypromisnobody
01:06.25cypromis:P
01:07.24*** join/#asterisk Son^Ghoku (~gendul@mail.lintasarta.biz)
01:07.40*** join/#asterisk Juggie (agony@CPE00c049d9f271-CM014270110981.cpe.net.cable.rogers.com)
01:07.49tzangerheh
01:07.50tzangerhttp://pbx.mine.nu/artwork/036-lolwhat-linux-sarojin.gif
01:11.45Juggieanyone know if theres an asterisk thing going on this week in toronto during von? a meetup or anything.
01:12.36NewSolefile can you help me figure this out
01:20.54*** join/#asterisk yxa (~void@203.118.40.42)
01:34.21*** join/#asterisk denon (denon@synapse.subneural.net) [NETSPLIT VICTIM]
01:34.21*** join/#asterisk Essobi (kstone@75.137.26.216.host.teledvance.com) [NETSPLIT VICTIM]
01:34.21*** join/#asterisk MooingLemur (~troy@phoenix.pinchaser.com) [NETSPLIT VICTIM]
01:34.21*** join/#asterisk blll (~bill@rtfm.insomnia.org) [NETSPLIT VICTIM]
01:34.21*** join/#asterisk nestAr (nester@makes.all.the.girlies.go.wewt.wewt.net) [NETSPLIT VICTIM]
01:34.21*** join/#asterisk tclark (~TC@S0106000c413a1c61.gv.shawcable.net) [NETSPLIT VICTIM]
01:34.21*** join/#asterisk Jovu (~bert@ev6.net) [NETSPLIT VICTIM]
01:34.21*** join/#asterisk Evanrude (~david@wsip-68-15-251-34.dl.dl.cox.net) [NETSPLIT VICTIM]
01:34.21*** join/#asterisk timecop (timecop@animenfo.com) [NETSPLIT VICTIM]
01:34.21*** join/#asterisk hardwire (~hardwire@209.112.194.45)
01:34.21*** join/#asterisk prh (~paul@212.13.203.69)
01:34.21*** join/#asterisk jlewis (~jlewis@solo.atlantic.net) [NETSPLIT VICTIM]
01:34.21*** join/#asterisk jontow (jontow@ws.woflsys.net) [NETSPLIT VICTIM]
01:34.21*** join/#asterisk scoof (~scoof@ipa.bryg.org) [NETSPLIT VICTIM]
01:34.21*** join/#asterisk dsfr (~dsfr@207.111.174.1) [NETSPLIT VICTIM]
01:34.21*** join/#asterisk JonR800 (jr@pcp05013027pcs.plyntv01.mi.comcast.net) [NETSPLIT VICTIM]
01:34.21*** join/#asterisk dmabe (~dmabe@cpe-024-163-071-041.nc.res.rr.com) [NETSPLIT VICTIM]
01:34.21*** join/#asterisk PatrickDK (patrickdk@dyn-19-218.myactv.net) [NETSPLIT VICTIM]
01:34.21*** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) [NETSPLIT VICTIM]
01:34.21*** join/#asterisk asteriskDOTbz (~logger@telux.net)
01:34.21*** join/#asterisk anderiv (~anderiv@207-67-87-34.gen.twtelecom.net) [NETSPLIT VICTIM]
01:34.22*** join/#asterisk sizban (~wrl@pluto.express.org) [NETSPLIT VICTIM]
01:34.22*** join/#asterisk dreamcode (~iancu@81.181.199.39) [NETSPLIT VICTIM]
01:34.22*** join/#asterisk BrianR___ (brianr@c-24-61-206-174.hsd1.ma.comcast.net) [NETSPLIT VICTIM]
01:34.22*** join/#asterisk ToyKeeper (spanky@c-24-9-113-171.hsd1.co.comcast.net)
01:34.22*** join/#asterisk jaiger (~jaiger@fire.innovationsw.com) [NETSPLIT VICTIM]
01:34.22*** join/#asterisk Rot2 (~aligator@62.48.187.98) [NETSPLIT VICTIM]
01:34.22*** join/#asterisk WilliamK (~wkeller@c-24-0-130-177.hsd1.tx.comcast.net) [NETSPLIT VICTIM]
01:34.22*** join/#asterisk ApEtc (apetc@ip68-99-136-197.ph.ph.cox.net) [NETSPLIT VICTIM]
01:34.22*** join/#asterisk Madkiss (madkiss@madkiss.staff.freenode) [NETSPLIT VICTIM]
01:34.22*** join/#asterisk MichaelCat (~mc@nh-londndry-c41-bg2a-08-113.lndnnh.adelphia.net) [NETSPLIT VICTIM]
01:34.22*** join/#asterisk Mw3 (mw3@195.56.193.13) [NETSPLIT VICTIM]
01:34.22*** join/#asterisk lvlpatlvl (pat@r00tworld.com) [NETSPLIT VICTIM]
01:34.22*** join/#asterisk KryoStoffer (~kri@helium.kri.dk) [NETSPLIT VICTIM]
01:34.22*** join/#asterisk Brumle (~brumle@brumle.com) [NETSPLIT VICTIM]
01:34.22*** join/#asterisk djflux (~djflux@207.250.204.185) [NETSPLIT VICTIM]
01:34.22*** join/#asterisk ChkDigit (~mike@static65-87-228-18.regina.accesscomm.ca) [NETSPLIT VICTIM]
01:34.22*** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) [NETSPLIT VICTIM]
01:34.22*** join/#asterisk Silik0n (~krice@rso.suspicious.org) [NETSPLIT VICTIM]
01:34.22*** join/#asterisk mistral (~mistral@jstevenson.plus.com) [NETSPLIT VICTIM]
01:34.22*** join/#asterisk marlowe (~marlowe@marlowe.active.supporter.pdpc) [NETSPLIT VICTIM]
01:34.22*** join/#asterisk _GiGi_ (gigi@jabber.szczecin.pl) [NETSPLIT VICTIM]
01:34.22*** join/#asterisk pepzi (robert@hd5e24fa4.gavlegardarna.gavle.to) [NETSPLIT VICTIM]
01:34.23*** join/#asterisk Hogie (daniel@alpha.dfwservers.net) [NETSPLIT VICTIM]
01:34.23*** join/#asterisk kodomo (~memyself@raven.net.informatik.tu-muenchen.de) [NETSPLIT VICTIM]
01:34.23*** join/#asterisk Chotaire (chotaire@nyc.us.chotaire.net) [NETSPLIT VICTIM]
01:34.23*** join/#asterisk kore (kore@mindwipe.org) [NETSPLIT VICTIM]
01:34.23*** join/#asterisk Corydon76-home (black@pcp08665860pcs.500ash01.tn.comcast.net) [NETSPLIT VICTIM]
01:34.23*** mode/#asterisk [+o denon] by irc.freenode.net
01:34.23*** join/#asterisk mutilator (~animenodv@65.111.201.79) [NETSPLIT VICTIM]
01:34.23*** join/#asterisk stoyan (~stoyan@ns.burdenis.com) [NETSPLIT VICTIM]
01:34.23*** join/#asterisk Rith (~Rith@35-28-142-66.speedexpress.net) [NETSPLIT VICTIM]
01:34.23*** join/#asterisk Hmmhesays (negative3k@66.173.103.108) [NETSPLIT VICTIM]
01:34.23*** join/#asterisk Weezey (WeezeyD@206.210.109.233) [NETSPLIT VICTIM]
01:34.23*** join/#asterisk rrk (~chatzilla@rrcs-67-53-9-175.west.biz.rr.com) [NETSPLIT VICTIM]
01:34.23*** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu) [NETSPLIT VICTIM]
01:34.23*** join/#asterisk olivier_ (~olivier_@obs92-4-82-239-116-113.fbx.proxad.net) [NETSPLIT VICTIM]
01:34.23*** join/#asterisk Dovid (~hirisk@pool-138-89-169-188.mad.east.verizon.net) [NETSPLIT VICTIM]
01:34.23*** join/#asterisk bonez41 (~aint@c-67-166-77-14.hsd1.ut.comcast.net) [NETSPLIT VICTIM]
01:34.23*** join/#asterisk sean (~sean@iconoclast.caedmon.net) [NETSPLIT VICTIM]
01:34.23*** join/#asterisk _Brian (brian@unix01.voicenet.com) [NETSPLIT VICTIM]
01:34.23*** join/#asterisk hohum (corbe@snoop.burghcom.com) [NETSPLIT VICTIM]
01:34.23*** join/#asterisk izo (~izo@izo.warpl.ipxxi.pl) [NETSPLIT VICTIM]
01:34.23*** join/#asterisk nvrs (RUR@toronto-HSE-ppp4255113.sympatico.ca) [NETSPLIT VICTIM]
01:34.23*** join/#asterisk logarno (~logarno@80.125.208.234) [NETSPLIT VICTIM]
01:34.23*** join/#asterisk dec (~tom@203.87.91.78) [NETSPLIT VICTIM]
01:34.24*** join/#asterisk WorkTooMuch (~work@82.148.188.1) [NETSPLIT VICTIM]
01:34.24*** join/#asterisk Martohtar (Martohtar@82.196.218.80) [NETSPLIT VICTIM]
01:34.24*** join/#asterisk nitram (nitram@superblob.com) [NETSPLIT VICTIM]
01:34.24*** join/#asterisk jonathh (~asd@217.46.145.65) [NETSPLIT VICTIM]
01:34.24*** join/#asterisk _m_ (~m@fbta199.fbta.uni-karlsruhe.de) [NETSPLIT VICTIM]
01:34.24*** join/#asterisk sambal (~sambal@gateway.office.flatbox.nl) [NETSPLIT VICTIM]
01:34.24*** join/#asterisk elric (~kavit@ppp114-10.static.internode.on.net) [NETSPLIT VICTIM]
01:34.24*** join/#asterisk jakepdev (~jakepdev@pool-68-236-58-19.phil.east.verizon.net) [NETSPLIT VICTIM]
01:34.24*** join/#asterisk vinsci (~vinsci@dsl-sjkgw2jb1.dial.inet.fi) [NETSPLIT VICTIM]
01:34.24*** join/#asterisk BuckRogers (~steve@ool-18bce89c.dyn.optonline.net) [NETSPLIT VICTIM]
01:34.24*** join/#asterisk MeTaBSD (metabsd@BlackBox.black4est.org) [NETSPLIT VICTIM]
01:34.24*** join/#asterisk Balu (~balu@foghorn.bartels-schoene.de) [NETSPLIT VICTIM]
01:34.24*** join/#asterisk DeeJayTwo (~deejay2@office.abi.ca) [NETSPLIT VICTIM]
01:34.24*** join/#asterisk nexIAX (~logger@telux.net) [NETSPLIT VICTIM]
01:34.24*** join/#asterisk redG ([U2FsdGVkX@67.51.185.15)
01:34.24*** join/#asterisk Hymie (hymie@L8R.NET) [NETSPLIT VICTIM]
01:34.24*** join/#asterisk hypa7ia (~leigh@67.71.86.109) [NETSPLIT VICTIM]
01:34.24*** join/#asterisk doughecka (~dheckaman@doughecka.user) [NETSPLIT VICTIM]
01:34.24*** join/#asterisk gein (~gein@213.134.110.241)
01:34.24*** join/#asterisk niZon (ilt@S0106deadbeef6977.wp.shawcable.net) [NETSPLIT VICTIM]
01:34.25*** join/#asterisk Micc (~mic@c-24-18-35-120.hsd1.wa.comcast.net) [NETSPLIT VICTIM]
01:34.25*** join/#asterisk opus_ (opus@dahphish.org) [NETSPLIT VICTIM]
01:34.25*** join/#asterisk emrah (~user@hosting.eknw.com) [NETSPLIT VICTIM]
01:34.25*** join/#asterisk CoolAcid (~jk@216.99.98.39) [NETSPLIT VICTIM]
01:34.25*** join/#asterisk Luhiwu (~marsosa@200.63.89.245) [NETSPLIT VICTIM]
01:34.25*** join/#asterisk techie (gus@asterisk.horizonte.us) [NETSPLIT VICTIM]
01:34.25*** join/#asterisk JerJer (~JerJer@12.173.204.122) [NETSPLIT VICTIM]
01:34.25*** join/#asterisk j0 (~dan@S010600105a04ed8d.va.shawcable.net) [NETSPLIT VICTIM]
01:34.25*** join/#asterisk pgpkeys (~pgpkeys@static-141-149-128-140.buff.east.verizon.net) [NETSPLIT VICTIM]
01:34.25*** join/#asterisk lecram (~marcel@fia114-101.dsl.hccnet.nl) [NETSPLIT VICTIM]
01:34.25*** join/#asterisk IronHelix (~irc@ool-182c3fe9.dyn.optonline.net) [NETSPLIT VICTIM]
01:34.25*** join/#asterisk maik (~maik@scumm.cs.uni-sb.de)
01:34.25*** join/#asterisk rajo (~rajo@scihparg.cs.uni-sb.de)
01:34.25*** join/#asterisk D|G|TAL (~grep@202.141.238.44) [NETSPLIT VICTIM]
01:34.25*** join/#asterisk |Vulture| (~Vulture@64.234.204.68.cfl.res.rr.com) [NETSPLIT VICTIM]
01:34.25*** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) [NETSPLIT VICTIM]
01:34.25*** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net) [NETSPLIT VICTIM]
01:34.25*** join/#asterisk dca (~dca@c-67-166-37-218.hsd1.co.comcast.net) [NETSPLIT VICTIM]
01:34.25*** join/#asterisk pino (~z@host241-115.pool80116.interbusiness.it) [NETSPLIT VICTIM]
01:34.25*** join/#asterisk likwid-- (likwid@nc-67-76-41-210.dyn.sprint-hsd.net) [NETSPLIT VICTIM]
01:34.26*** join/#asterisk asteriskn00b (asteriskn0@adsl-68-91-7-226.dsl.tulsok.swbell.net) [NETSPLIT VICTIM]
01:34.26*** join/#asterisk NewSole (david@i216-58-44-245.avalonworks.net) [NETSPLIT VICTIM]
01:34.26*** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) [NETSPLIT VICTIM]
01:34.26*** join/#asterisk jeffik (jefik@69.158.30.24) [NETSPLIT VICTIM]
01:34.26*** join/#asterisk easimon (~easimon@localhorst.kawo2.RWTH-Aachen.DE) [NETSPLIT VICTIM]
01:34.26*** join/#asterisk Romik (~romik@1.fix.netvision.net.il) [NETSPLIT VICTIM]
01:34.26*** join/#asterisk sudoer (~toy@denali.ccs.neu.edu) [NETSPLIT VICTIM]
01:34.26*** join/#asterisk Slainte (~Slainte@66.55.113.102.ppp.northrock.bm) [NETSPLIT VICTIM]
01:34.26*** join/#asterisk jtodd (~jtodd@garthim.fox-den.com) [NETSPLIT VICTIM]
01:34.26*** join/#asterisk blop (~blop@2001:6f8:204:33:bbbb:bbbb:bbbb:bbbb) [NETSPLIT VICTIM]
01:34.26*** join/#asterisk GMsoft (~gmsoft@gmsoft.developer.gentoo) [NETSPLIT VICTIM]
01:34.26*** join/#asterisk nextime (~nextime@213-140-6-96.fastres.net) [NETSPLIT VICTIM]
01:34.26*** join/#asterisk chamba (~chamba@64.119.36.42) [NETSPLIT VICTIM]
01:34.26*** join/#asterisk MrBelvedr (~tt@ip68-227-209-110.dc.dc.cox.net) [NETSPLIT VICTIM]
01:34.26*** join/#asterisk joaovianna (joao@node-40247a6a.ewr.onnet.us.uu.net) [NETSPLIT VICTIM]
01:34.26*** join/#asterisk file[laptop] (~file@mctnnbsah25-142166093149.nb.aliant.net)
01:34.26*** join/#asterisk robl^ (~robl@dsl093-025-118.hou1.dsl.speakeasy.net) [NETSPLIT VICTIM]
01:34.26*** join/#asterisk sivana (~sivana@165.154.13.35) [NETSPLIT VICTIM]
01:34.26*** join/#asterisk unterx (unter@2001:4830:2016:cd92:a429:d928:2819:d281) [NETSPLIT VICTIM]
01:34.26*** join/#asterisk poli_off (~poli@200-168-30-125.dsl.telesp.net.br) [NETSPLIT VICTIM]
01:34.26*** join/#asterisk Dutts (~dutts@81.168.70.41) [NETSPLIT VICTIM]
01:34.26*** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) [NETSPLIT VICTIM]
01:34.27*** join/#asterisk jdiskywlkr (~kvirc@ip68-0-90-1.tu.ok.cox.net) [NETSPLIT VICTIM]
01:34.27*** join/#asterisk psycodad (~obiwan@2001:4060:4419:b1:0:0:0:2) [NETSPLIT VICTIM]
01:34.27*** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au) [NETSPLIT VICTIM]
01:34.27*** join/#asterisk florz (nobody@2001:1a50:503c:0:0:0:0:1) [NETSPLIT VICTIM]
01:34.27*** join/#asterisk bugbot (~bugbot@d141-234-145.home.cgocable.net)
01:34.27*** join/#asterisk kajtzu (~kajtzu@shell1.fi.basen.net) [NETSPLIT VICTIM]
01:34.27*** join/#asterisk Makenshi (~makenshi@2001:630:1c0:2001:280:c8ff:fee2:921f) [NETSPLIT VICTIM]
01:34.27*** join/#asterisk Dibbler (~Dibbler@zidane.pi-net.net) [NETSPLIT VICTIM]
01:34.27*** join/#asterisk devel (~devel@wiggum.digitalcoven.com) [NETSPLIT VICTIM]
01:34.27*** join/#asterisk Moonwick (~moonwick@core.dump.net) [NETSPLIT VICTIM]
01:34.27*** join/#asterisk zigman (~zigman@irc.zigman.de) [NETSPLIT VICTIM]
01:34.27*** join/#asterisk Pj386 (~pj@fernande.happycoders.org) [NETSPLIT VICTIM]
01:34.27*** join/#asterisk AvengerX (~h_avenger@gabriel.core.buynet.com.br) [NETSPLIT VICTIM]
01:34.27*** join/#asterisk macTijn (martijn@linda.net.insecure.nl) [NETSPLIT VICTIM]
01:34.27*** join/#asterisk pfn (500@netblock-66-245-252-239.dslextreme.com) [NETSPLIT VICTIM]
01:34.27*** join/#asterisk DannyF (~dannyf@h27n3c1o848.bredband.skanova.com) [NETSPLIT VICTIM]
01:34.27*** join/#asterisk vaewynAFK (freeman@mail.deltamach.com) [NETSPLIT VICTIM]
01:34.27*** join/#asterisk wildcard0 (~generic@S0106006097e16040.vc.shawcable.net) [NETSPLIT VICTIM]
01:34.27*** join/#asterisk Wonka (produziert@wonka.support.madwifi) [NETSPLIT VICTIM]
01:34.27*** join/#asterisk puppet (puppet@1-1-3-3b.ox.mlm.bostream.se) [NETSPLIT VICTIM]
01:34.27*** join/#asterisk cblackbu (~cblackbu@c-24-23-43-130.client.comcast.net) [NETSPLIT VICTIM]
01:34.28*** join/#asterisk TomL (~tom@magnum.tx3.net) [NETSPLIT VICTIM]
01:34.28*** join/#asterisk cypromis (chuck-the-@62.212.85.27)
01:34.28*** join/#asterisk want561or772did (~aoiahsdf@68.71.213-35.atlsfl.adelphia.net) [NETSPLIT VICTIM]
01:34.28*** join/#asterisk otaku42 (~otaku@otaku42.developer.madwifi) [NETSPLIT VICTIM]
01:34.28*** join/#asterisk astoria (~asto@68.77.110.194) [NETSPLIT VICTIM]
01:34.28*** join/#asterisk Juggie (agony@CPE00c049d9f271-CM014270110981.cpe.net.cable.rogers.com) [NETSPLIT VICTIM]
01:34.28*** join/#asterisk yxa (~void@203.118.40.42) [NETSPLIT VICTIM]
01:34.28*** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) [NETSPLIT VICTIM]
01:35.24*** join/#asterisk kram (~mark@kram.digium.sponsor.pdpc) [NETSPLIT VICTIM]
01:35.24*** join/#asterisk Son^Ghoku (~gendul@mail.lintasarta.biz) [NETSPLIT VICTIM]
01:35.24*** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net)
01:35.24*** join/#asterisk immo (~jezus@c-24-4-13-13.hsd1.ca.comcast.net) [NETSPLIT VICTIM]
01:35.24*** join/#asterisk voiper (~none@pcp0010158348pcs.eatntn01.nj.comcast.net) [NETSPLIT VICTIM]
01:35.24*** join/#asterisk roamer323 (~sing@HSE-Toronto-ppp131661.sympatico.ca) [NETSPLIT VICTIM]
01:35.24*** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) [NETSPLIT VICTIM]
01:35.24*** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net) [NETSPLIT VICTIM]
01:35.24*** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) [NETSPLIT VICTIM]
01:35.26*** join/#asterisk topping (~topping@cpe-24-210-82-196.columbus.res.rr.com) [NETSPLIT VICTIM]
01:35.26*** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) [NETSPLIT VICTIM]
01:35.26*** join/#asterisk brc__ (~brian@brc.base.supporter.pdpc) [NETSPLIT VICTIM]
01:35.26*** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc) [NETSPLIT VICTIM]
01:35.26*** join/#asterisk eKo1 (~bernd@metrored-gw.tropicohn.com) [NETSPLIT VICTIM]
01:35.26*** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc) [NETSPLIT VICTIM]
01:35.26*** join/#asterisk Curus (~Curus@83.72.32.8.ip.tele2adsl.dk) [NETSPLIT VICTIM]
01:35.26*** join/#asterisk sung (~sung@fluorine.idge.net) [NETSPLIT VICTIM]
01:35.26*** join/#asterisk warchest (sk@apexsoftware.com) [NETSPLIT VICTIM]
01:35.26*** join/#asterisk jero (~sflphone@199.243.85.90) [NETSPLIT VICTIM]
01:35.26*** join/#asterisk wisdom (~cgucker@gait.onesc.net) [NETSPLIT VICTIM]
01:35.26*** join/#asterisk flewid (~flewid@24.42.244.169) [NETSPLIT VICTIM]
01:35.26*** join/#asterisk juice (~juice@mo-205-240-40-98.dyn.sprint-hsd.net) [NETSPLIT VICTIM]
01:35.26*** join/#asterisk SplasPood (jwb@paravolve.net) [NETSPLIT VICTIM]
01:35.26*** join/#asterisk knight_ ([26PQwLbIi@voip.phunc.com) [NETSPLIT VICTIM]
01:35.26*** join/#asterisk iCEBrkr (icebrkr@chrome.cyberdyne.org) [NETSPLIT VICTIM]
01:35.26*** join/#asterisk Jearil (~Jearil@216-224-56-213.client.dsl.net) [NETSPLIT VICTIM]
01:35.26*** join/#asterisk mishehu (mishehu@cshells.shavedgoats.net) [NETSPLIT VICTIM]
01:35.26*** join/#asterisk NatRH (~Nat@dargo.trilug.org) [NETSPLIT VICTIM]
01:35.26*** join/#asterisk stevek (~stevek@slim-eth0.horizonlive.net) [NETSPLIT VICTIM]
01:35.26*** join/#asterisk tzanger (~tzanger@165.154.13.35) [NETSPLIT VICTIM]
01:35.26*** join/#asterisk `Sauron (sauron@rrcs-24-153-164-117.sw.biz.rr.com) [NETSPLIT VICTIM]
01:35.26*** join/#asterisk CoaxD (coax@shell1.cornernet.com) [NETSPLIT VICTIM]
01:35.26*** mode/#asterisk [+o kram] by irc.freenode.net
01:35.28*** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) [NETSPLIT VICTIM]
01:35.28*** join/#asterisk shmaltz (~chatzilla@69.28.255.210) [NETSPLIT VICTIM]
01:35.28*** join/#asterisk znoG (gs@200.115.216.109) [NETSPLIT VICTIM]
01:35.28*** join/#asterisk tessier (~treed@210.245.38.37) [NETSPLIT VICTIM]
01:35.28*** join/#asterisk Marcel-AS16215 (Marcel-AS1@gic-msg-exc-01.genotec.ch) [NETSPLIT VICTIM]
01:35.28*** join/#asterisk iq (~iq@70-59-160-225.omah.qwest.net) [NETSPLIT VICTIM]
01:35.28*** join/#asterisk Moc (~Moc@modemcable165.109-70-69.mc.videotron.ca) [NETSPLIT VICTIM]
01:35.28*** join/#asterisk fugitivo (~ajf@201.255.101.186) [NETSPLIT VICTIM]
01:35.28*** join/#asterisk RaYmAn-Bx (user@x1-6-00-40-63-da-39-3f.k191.webspeed.dk) [NETSPLIT VICTIM]
01:35.28*** join/#asterisk SuPrSluG (~SuPrSluG@pool-129-44-142-202.buff.east.verizon.net) [NETSPLIT VICTIM]
01:35.28*** join/#asterisk R3DB0x (nobody@66.142.28.36) [NETSPLIT VICTIM]
01:35.28*** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) [NETSPLIT VICTIM]
01:35.28*** join/#asterisk Chuji (Chuji@pcp09930052pcs.tulipgrove.tn.nash.comcast.net)
01:35.28*** join/#asterisk terracon (~tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com) [NETSPLIT VICTIM]
01:35.28*** join/#asterisk ennuyeux72 (~ennuyeux7@83.146.53.34) [NETSPLIT VICTIM]
01:35.28*** join/#asterisk riksta (~rick@81-178-248-131.dsl.pipex.com) [NETSPLIT VICTIM]
01:35.28*** join/#asterisk _mwoodj_ (~MWoodJ@hyper-eye.digium.sponsor.pdpc) [NETSPLIT VICTIM]
01:35.28*** join/#asterisk pigpen (~mark@fw.seamans.cc) [NETSPLIT VICTIM]
01:35.28*** join/#asterisk inspired (mikael@213.197.167.61) [NETSPLIT VICTIM]
01:35.28*** join/#asterisk mbranca (~matteo@81.208.92.210) [NETSPLIT VICTIM]
01:35.28*** join/#asterisk ipso (~ipso@d207-81-249-35.bchsia.telus.net) [NETSPLIT VICTIM]
01:35.28*** join/#asterisk dg1nsw (~schulte@gate.sympat.de) [NETSPLIT VICTIM]
01:35.29*** join/#asterisk Connor- (~billy@198-144-174-5.knx.tn.nxs.net) [NETSPLIT VICTIM]
01:35.29*** join/#asterisk Qwell (~north@24-50-66-194.vnnyca.adelphia.net) [NETSPLIT VICTIM]
01:35.29*** join/#asterisk mmlj4 (~looseduk@ip68-14-124-25.no.no.cox.net) [NETSPLIT VICTIM]
01:35.29*** mode/#asterisk [+o bkw_] by irc.freenode.net
01:35.29NewSolefile[laptop]... I am still here
01:35.29file[laptop]one less person I have to say, "no I won't help you become the next vonage", to is great
01:35.29*** join/#asterisk Rick_Hunter (~rhunter@07-168.008.popsite.net)
01:35.30*** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) [NETSPLIT VICTIM]
01:35.30*** join/#asterisk ptblank (~MURDER1@68-169-176-137.lmdaca.adelphia.net) [NETSPLIT VICTIM]
01:35.31*** join/#asterisk TUplink (~Tommy@68-232-92-239.chvlva.adelphia.net)
01:35.31jakepdevis that a common request?
01:35.31file[laptop]oh no, people are back
01:35.31file[laptop]it can get worse
01:35.31*** join/#asterisk twisted[work] (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) [NETSPLIT VICTIM]
01:35.31*** join/#asterisk bjohnson (~bjohnson@66.11.165.54) [NETSPLIT VICTIM]
01:35.31*** join/#asterisk vagwin (~vagwin@mk-ns500-1.uk.tiscali.com) [NETSPLIT VICTIM]
01:35.31*** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) [NETSPLIT VICTIM]
01:35.31*** join/#asterisk FengShui (~ted@gray.impulse.net) [NETSPLIT VICTIM]
01:35.31*** mode/#asterisk [+o twisted[work]] by irc.freenode.net
01:35.31file[laptop]I block most stuff out of my memory though
01:35.31jakepdevunderstood
01:35.31*** join/#asterisk lilo_ (lilo@levin-pdpc.staff.freenode)
01:35.31TUplinkApr 17 21:35:08 WARNING[1540]: chan_zap.c:763 zt_open: Unable to open '/dev/zap/pseudo': Device not configured
01:35.32*** join/#asterisk rabelais (~blank@ca-stmnca-cuda4-gen2c1-171.vnnyca.adelphia.net) [NETSPLIT VICTIM]
01:35.32*** join/#asterisk rowter (~Drake@201.133.210.80) [NETSPLIT VICTIM]
01:35.32*** join/#asterisk kFuQ (~somedude@c-24-17-224-78.hsd1.wa.comcast.net) [NETSPLIT VICTIM]
01:35.32*** join/#asterisk LoRez (lorez@lorez.staff.freenode) [NETSPLIT VICTIM]
01:35.32TUplinki insytalled zaptel
01:35.32jakepdevmaybe lilo will fill us in as to what's going on
01:35.55TUplinkand /dev/zap is ther
01:36.27jakepdevsounds like ztdummy is not configured
01:36.27jakepdevbut i'm not sure
01:37.05TUplinkhow do i do that?
01:37.05TUplinkthe WIKI is stilldown
01:37.34jakepdevi'm just guessing by the error message - don't take that as an official diagnosis :)
01:38.30TUplinkhow do i config ztdummy
01:38.57*** join/#asterisk GustavoIPA (~gustavo80@200.141.74.130)
01:39.23timecop== No one is available to answer at this time (1:0/0/0)
01:39.26timecop^^ from h323
01:39.33timecopbut the otehr end is fine
01:39.41ManxPowerDoes ztdummy even run on *BSD?
01:39.51timecopjust stop being cheapfucks and buy digium hardware.
01:39.57timecopfrom digium, not from fucking scambay.
01:40.09ManxPowertimecop: Or at least use a supported platform.
01:40.10timecopthen you wont need ztdummy.
01:40.13timecopthat too.
01:40.23jakepdevit says pseudo though - that doesn't sound like hardware
01:41.00timecopfucking h323
01:41.05*** join/#asterisk tessier (~treed@222.253.73.184)
01:41.05techiehaha
01:42.25jakepdevTUplink - http://64.233.179.104/search?q=cache:23fcv2Hcj0UJ:www.voip-info.org/wiki-Asterisk%2Btimer%2Bztdummy+ztdummy+wiki&hl=en
01:43.29jakepdevbut as Manx says - are you running on a supported platfrom?
01:44.35*** join/#asterisk JunK-Y (junk-b@Sherbrooke-HSE-ppp3606643.sympatico.ca)
01:44.42jakepdevand while were at it - i'm trying to figure out how to become the next Vonage
01:45.01JunK-Yy0
01:45.02TUplinki got it
01:45.18jakepdevhow?
01:45.21TUplinkjake i got it
01:45.40jakepdevok - what did you do to fix it?
01:45.57TUplinkkldload /sbin/kldunload/ztdummy.ko
01:46.07jakepdevok - great!
01:46.33TUplinkby default its not in hte zaptel.sh
01:46.44jakepdevok
01:50.02*** join/#asterisk MikeJ[Laptop] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net)
01:52.09ManxPower~docs
01:52.12jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
01:52.12bugbotdocs is assigned nothing and reported nothing.
01:52.13ManxPower~mailinglist
01:52.14jbotextra, extra, read all about it, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
01:52.14bugbotmailinglist is assigned nothing and reported nothing.
01:53.50NewSoleAny one here good at Dialplans... can help figure this one out....
01:54.02jakepdevpastebin
01:54.10NewSolehttp://pastebin.ca/9728
01:54.19shmaltzNewSole, lets try
01:54.45shmaltzwhats happening with that?
01:54.54|Vulture|anyone here use Nagios + check_asterisk?
01:54.54shmaltzNewSole
01:54.57NewSoleI get "No authority found" on masters
01:55.03*** join/#asterisk wwalker (~wwalker@wwalker.sustaining.supporter.pdpc)
01:55.26*** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net)
01:55.27hohum<--- hohum
01:55.44file[laptop]<--- file
01:55.58file[laptop]today is advice day in #asterisk
01:56.00hohum^--- homo
01:56.03jakepdev<---------  jake
01:56.06file[laptop]NewSole: my advice to you is to start simple
01:56.07hohum:)
01:56.17jakepdevamen to starting simple
01:56.37jakepdevand that doesn't mean only mean using *@home as i've found out
01:56.50*** join/#asterisk scythelx (~wow@pc-24-151-108-034.newm2.ct.charter.com)
01:56.50NewSolelol
01:56.52scythelxhey all
01:56.58*** join/#asterisk shodan (~shodan@216.113.99.185)
01:56.59hohumhowdy
01:57.03file[laptop]NewSole: you've suverely overcomplicated things
01:57.06|Vulture|is there a way to use check_asterisk to see if a peer is registered?
01:57.07NewSoleproblem was it was working last week and I did no changes
01:57.15file[laptop]obviously something changed
01:57.20file[laptop]and why don't you just use disallow=all
01:57.25file[laptop]and then allow all you want afterwards?
01:57.43scythelxcould someone help me, the wiki is down so i cant look up how to do this. look at http://pastebin.ca/9729 - im trynig to put music on hold ( which wokrs fine ) but like 10 -15 seconds into the music i want it to say a message not sure if im doing it right
01:58.09file[laptop]scythelx: nope won't work
01:58.10jakepdevnope
01:58.12NewSolecause our mobile PRI boxes done like dissallow=all
01:58.16jakepdevuse WaitMusicOnHold
01:58.21jakepdev(10)
01:58.30scythelxok cool
01:58.35scythelxthank you
01:58.38jakepdevnp
01:58.54file[laptop]NewSole: all disallow does is clear the internal bitmask of all the stuff... same thing as disallowing each except it does it in one big swoop
01:59.32file[laptop]but anyway
01:59.44file[laptop]a user doesn't have a host, or username, or qualify
02:00.16file[laptop]and don't put the username in the iax.conf entry for the peer, weird stuff happens
02:01.18file[laptop]try specifying the contents you want the user to have access to in the iax.conf too and see what happens
02:01.26file[laptop]anyone else have anything to comment regarding his entries?
02:02.16jakepdevis "No authority found" a codec issue?  sounds more like authentication...
02:02.27file[laptop]it's authentication
02:02.33file[laptop]the codec stuff just annoyed me... a lot :p
02:02.36jakepdevlol
02:02.50NewSoleI get a code 50
02:03.15file[laptop]a code 50, how descriptive
02:03.21jakepdevno!!!  not the code 50
02:03.27jakepdevrun - fast!
02:03.30file[laptop]WE'RE ALL GONNA DIE!
02:03.31file[laptop]AHHHHHHHHHHH!
02:03.40jakepdevhehe
02:04.53*** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net)
02:05.02NewSolethis is what I get
02:05.03NewSolehttp://pastebin.ca/9731
02:06.01*** join/#asterisk ClayReiche123 (fwuser@acxexch1.accxx.com)
02:06.25file[laptop]and you've got a user on that box?
02:06.40jakepdevfor outgoing - i was under the impression all you need is an entry in the dialplan
02:06.53NewSoleya the user was pasted to http://pastebin.ca/9728
02:06.55file[laptop]jakepdev: better to have a peer entry, more control over things
02:07.00jakepdevok
02:07.42file[laptop]NewSole: iax2 show users
02:08.22*** join/#asterisk drumkilla (~russell@12.21.241.80)
02:08.22*** mode/#asterisk [+o drumkilla] by ChanServ
02:08.26file[laptop]is masters in there?
02:08.26ClayReiche123I've been using the stable version for a while now and I see a couple features in the HEAD that I would LOVE to have... is there a way for me to "add" these features to stable. The one thing I'm particularly fond of is the "sip NOTIFY" feature in HEAD...
02:08.48file[laptop]ClayReiche123: you can try to backport it, but you're on your own unless you find someone to help you :)
02:08.55*** join/#asterisk david (~david@gw-djc.davidcoulson.net)
02:09.20drumkillaClayReiche123: /me waves to file[laptop]
02:09.34drumkillaoops
02:09.38drumkillaha
02:09.41NewSolehttp://pastebin.ca/9732
02:10.06ManxPowerdrumkilla: Did you see kram's update to the Asterisk README about large jumps in time?
02:10.19drumkillaManxPower: yeah
02:10.45drumkillai've been doing a school project today - i'll get it when I get back on the bug tracker
02:11.05file[laptop]I thought there was actually a patch when I saw the bugnote, then I saw it was just for the readme
02:11.15ManxPowerdrumkilla: any progress on the call parking timout
02:11.30drumkillahaven't been able to work on it :/
02:11.54drumkillabut I will get it this week
02:12.18drumkillait's probably just some code I need to snag from head
02:12.20ClayReiche123file[laptop]: Thanks. Sounds difficult....
02:12.28NewSolejakepdev.. u see that
02:12.44ManxPowerdrumkilla: I need it fixed by Wed.  Will a bounty help?
02:12.58drumkillaprobably, heh
02:13.00jakepdevi do
02:13.07drumkillathat will probably make sure that someone gets it by then, in case I don't
02:13.10NewSolesee users there
02:13.12ManxPowerdrumkilla: I'll post a bounty on monday.
02:13.16drumkillaalright
02:13.30drumkillaI might be the one to get it, hehe :)
02:13.35ManxPowerdrumkilla: My bounties do not exclude Developers.
02:13.49file[laptop]jakepdev: go, help NewSole little one!
02:13.51jakepdevNewSole:  I would attack this by starting with just hard coding everything in the dialplan.  I wouldn't use the iax.conf quite yet
02:13.56drumkillaI should be able to play with it tomorrow
02:14.16jakepdevfile - don't know if I can - i only started with * a few weeks ago
02:16.54|Vulture|damn wiki...
02:17.02|Vulture|anyone know if there is a mirror of the manager commands?
02:17.18jakepdevanyone remember Nucleus from the 80's?
02:17.26jakepdevVulture - use google cache
02:17.52|Vulture|kk thanx
02:18.07ManxPowerIsn't there a complete list of Manager commands in a README in the Asterisk source?
02:18.20jakepdevwho looks at that stuff
02:18.35drumkillaha ...
02:18.39jakepdevhehe
02:19.25ManxPoweralso "show manager commands" will give you a list.  As well as asterisk/doc/manager.txt
02:21.11ManxPowerWell, I figured out why voip-info.org is down.
02:21.22ManxPowerhttp://voxilla.com/voxstory155.html links to it.
02:22.06file[laptop]Josh is cool
02:22.21file[laptop]from Switchvox
02:22.40ManxPowerWhich is linked from a story on slashdot
02:23.09*** join/#asterisk DrRighteous (~DrRighteo@ool-182c867b.dyn.optonline.net)
02:25.08jakepdevah
02:25.15jakepdevit all makes sense now
02:26.28drumkillawhat story from slashdot
02:26.46drumkillafile[laptop]: was Josh at VON?
02:26.50file[laptop]yes
02:26.59file[laptop]drumkilla: remember Tristan? well, Josh was the guy with her
02:27.10ManxPowerdrumkilla: http://hardware.slashdot.org/article.pl?sid=05/04/17/2327225&threshold=3&tid=215&tid=187&tid=189
02:27.18file[laptop]not her husband mind you
02:27.40PTG1234why don't they do
02:27.43drumkillayeah, I know who it is - just wondering if it was Josh
02:27.49PTG1234if REFER=BLAH redirect(sorry too busy)
02:27.50file[laptop]yup yup
02:28.39PTG1234hmm
02:28.42PTG1234do i want 9meg mp3s?
02:29.08jakepdevcouldn't you convert them to a low bitrate?
02:29.26MoonwickI've started ripping everything I own into apple lossless.
02:29.31drumkillawhen was that on /.
02:29.39Moonwickdisk space is too cheap to bother with lossy compression nowadays
02:29.58fugitivouse google!
02:30.01jakepdevbut - typically in telephony - you wouldn't hear the difference
02:30.04fugitivoto store your mp3!
02:30.42jakepdevat least with TDM - it's difficult to hear better than a 11khz sample
02:31.02shodanI'm looking for specs for phone lines and phones (stuff like how phones modulate voice , how phone lines modules rings and CID , how much current phones can draw at max , maximum line voltage sag etc..) anyone has some links on that ?
02:31.05timecopwell fucking shit
02:31.16timecopanyone know anythign about h323?
02:31.28*** join/#asterisk mrproper_ (~b@61.95.55.242)
02:31.31PTG1234well
02:31.32jakepdevtimcop - JerJer does
02:31.37PTG1234is apple lossless better then 320kbps mp3?
02:31.39timecopi'm calling to some voip provider in china, all they suppsoedly require is caller ID set to blah, then dial H323/whatever@theirIP
02:31.47jakepdevhe has several sites running H.323
02:31.51timecopand al I get back from asterisk is  No one is available to answer at this time (1:0/0/0)
02:31.57timecoph323 debug doesnt show shit
02:32.01timecop(no useful shit, that is)
02:32.03mrproper_anyone here familiar with oh323, i can make outgoing h323 calls everything runs ok, but all incoming calls ring but no audio either way
02:32.22jakepdevah - the h323 squad!!!
02:32.28mrproper_lol
02:32.38jakepdevthey come in numbers
02:32.43timecophuhu
02:32.57jakepdevPTG - for telephony?
02:33.11jakepdevor in general?
02:33.45mrproper_who is that in response to?
02:33.50jakepdevshodan - try about.com
02:34.09jakepdevshodan - or google what you're looking for
02:34.15*** part/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com)
02:34.53jakepdevmrproper - it was in response to PTG
02:35.04*** join/#asterisk TheEmperor (~user@203.114.48.47)
02:35.21*** part/#asterisk DrRighteous (~DrRighteo@ool-182c867b.dyn.optonline.net)
02:35.50*** join/#asterisk wdatkinson (~wdatkinso@pcp986542pcs.northw01.in.comcast.net)
02:35.57*** part/#asterisk ClayReiche123 (fwuser@acxexch1.accxx.com)
02:37.25NewSolehmmm
02:37.44jakepdevdid you hard code that stuff?
02:37.50jakepdevin the dialplan?
02:38.08shodanjakepdev, about.com's search is useless , but going throught the categories doesn't return anything useful, and I'm still searching with google but there's so much stuff on telephony (and it's all consumer level stuff) I can't find some hard numbers
02:38.19NewSolei tried... now its not even giving a reject code
02:38.34jakepdevNewSole: what is it giving?
02:38.54NewSole<PROTECTED>
02:38.54NewSole<PROTECTED>
02:38.54NewSole<PROTECTED>
02:39.24NewSole<PROTECTED>
02:39.46jakepdeviax2 show debug
02:40.04NewSoleNo such command 'iax2 show debug' (type 'help' for help)
02:40.28rikstaiax not iax2
02:40.55jakepdeviax2 debug
02:41.03rikstaah yeah
02:41.20*** join/#asterisk egon_l (~egon@pc-10-19-104-200.cm.vtr.net)
02:41.22rikstaim talking crap
02:42.42NewSolehttp://pastebin.ca/9735
02:47.00NewSolejakepdev
02:47.10jakepdevNewSol - INVAL is Invalid Request
02:47.12*** join/#asterisk wolfson` (~hehe@cpe-68-187-189-084.man.nc.charter.com)
02:47.17*** part/#asterisk shodan (~shodan@216.113.99.185)
02:48.15jakepdevok - why does it say dialing IAX over ZAP :)
02:48.16jakepdev?
02:48.39jakepdevwhat is your exact dialstring - pastebin - but replace your pwd with ******
02:48.42NewSolezap card dialing out
02:48.45jakepdevno
02:48.57jakepdevif you're using IAX - you're not using ZAP
02:49.03jakepdevand vice versa
02:49.13*** join/#asterisk Kumbang (~ecvs@167.205.24.4)
02:49.23jakepdevyou can initate a call from one, and terminate to another
02:49.32file[laptop]he's probably using an FXS module
02:49.56jakepdevare you NewSole?
02:51.03NewSolei am
02:51.06jakepdevok
02:51.31file[laptop]see? I'm psychic
02:51.43jakepdevright :)
02:51.43NewSoleserver is downstairs... so I hooked a cordless
02:52.16jakepdevfile - is the debug out supposed to look like that?
02:52.30jakepdevi'm not familiar with ZAP over IAX
02:53.08NewSolequestion is WHAT is INVALID
02:53.22*** join/#asterisk ToyMan (~stuq@user-12lcqur.cable.mindspring.com)
02:53.40NewSolehttp://pastebin.ca/9736
02:53.42jakepdevi'd say something in your dial string - but don't know for sure
02:54.14jakepdevNewSole - remember how we talked about hard coding to do a test?
02:54.18NewSoledial string is Dial(IAX2/masters@masters/${EXTEN},100,rT)
02:54.33jakepdevok
02:54.38NewSoleaxualy
02:54.57NewSoleexten = _X.,1,Dial(IAX2/masters@masters/${EXTEN},100,rT)
02:55.23L|NUXexten => _X.,1,Dial(IAX2/masters@masters/${EXTEN},100,rT)
02:55.31L|NUXit should look like this :)
02:55.48GustavoIPAalguem do brasil por aqui???
02:56.27*** join/#asterisk jterrero (~jt@ool-43576e0d.dyn.optonline.net)
02:56.41mrproper_when i dial outbound h323 calls, all works fine, but an incomming h323 call establishes but no audio either way, any ideas?
02:59.12NewSolehmm
03:01.19mrproper_can anyone tell me how to start up a remote console that only shows warning and errors, without all the 'information' events
03:01.48drumkillalogger.conf
03:02.33fugitivomrproper_: are you behind nat?
03:02.44mrproper_fugitivo: no
03:03.17mrproper_i have asterisk with oh323 pointing to a gatekeeper which then goes to a gateway then isdn, outbound calls are fine, inbound establish but no audio what so ever
03:04.12*** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net)
03:05.19NewSolejakepdev... what do you think
03:05.36jakepdevi'm not familar with Zap over IAX
03:05.59jakepdevyou're using a PCI FXS card in your * box?
03:06.13mrproper_this is the oh323 debug output from an incoming call: http://pastebin.ca/9737
03:06.30NewSoleone.... on local server
03:07.38jakepdevbut in this scenerio, you're trying to place a remote ZAP call using the IAX protocol?
03:07.50NewSoleno
03:08.30NewSolei have a local box here with TDM40P card.... making local call off phone to IAX
03:08.38jakepdevok
03:08.44NewSoleacualy going like this
03:09.12NewSoleZAP => Asterisk => IAX2 => Asterisk => SIP
03:09.44jakepdevok - but we're just working on the ZAP -> * -> IAX part now - right?
03:09.55NewSoleyup
03:10.17NewSolebut problem not only with ZAP
03:11.00NewSoleits with "* => IAX2 => *" weather I use Zap Card or Mobil PRI
03:11.14jakepdevNewSole - is "14163101010" a valid exten on your remote box?
03:11.31FuriousGeorgeim started setting up my tdm fxs, and earlier (when i didnt expect it to work yet) i would pick it up and hear whatever went into the mouthpiece in the earpiece
03:11.36FuriousGeorgenow it just sounds like a dead line
03:11.46FuriousGeorge*tdm w/ 2 fxs
03:11.48NewSolehttp://pastebin.ca/9736
03:11.55NewSolelook at that
03:12.10jakepdevk - i did
03:12.22jakepdevdoes the remote box give you any indication of what is going on?
03:12.38jakepdevany errors from debug out on that?
03:13.03NewSoleFindChannel check to see if its a local DID or PRI account
03:13.44NewSoleand then routes to Moblie PRI Box
03:14.10mrproper_i have asterisk with oh323 pointing to a gatekeeper which then goes to a gateway then isdn, outbound calls are fine, inbound establish but no audio what so ever
03:14.40NewSoleour system goes like this
03:14.45jakepdevNewSole - I'm saying - on "masters", can you do an iax2 debug?
03:15.28NewSole<MOBLIE PRI> ==> <PRI SERVER> ==> <ASTERISK> ==> SIP
03:16.16FuriousGeorgeanother thing:  anyone know how linux goes about assigning irws to pci devs.  in my bios i assigned irq 10 to pci slot 4 (since cat proc/dev said 10 was free), but when i boot, loading the driver has it showing up on irq3 w/ usb controller
03:16.37FuriousGeorge*irw=irq
03:16.44jakepdevNewSole - the messages on you local box to don't say much about why it's invalid - it just says that it is invalid
03:17.29jakepdevFG - i was told by Digium it should use what you set in the BIOS
03:19.11NewSolehttp://pastebin.ca/9738
03:19.24NewSolethats all I get for debug on masters
03:19.41jakepdevthat's nothing :)
03:20.02*** join/#asterisk WilliamK (~wkeller@c-24-0-130-177.hsd1.tx.comcast.net)
03:20.25*** join/#asterisk implicit (~implicit@ip68-7-154-222.sd.sd.cox.net)
03:20.29NewSolethats what i mean all I get is INVAL
03:21.19jakepdevlet me see if I can set up somthing - see if you can connect to me via IAX
03:22.54*** join/#asterisk hcir (~ar@rdbck-static-532.palmer.mtaonline.net)
03:25.02*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)
03:25.33*** part/#asterisk hcir (~ar@rdbck-static-532.palmer.mtaonline.net)
03:29.06*** join/#asterisk MrEntropy (~entropy@125.ade0506.ade.iprimus.net.au)
03:29.08MrEntropyyo
03:29.28MrEntropyhow can i get info on what codecs both sides of a call are using?
03:38.06*** join/#asterisk MikeJ[Laptop] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net)
03:42.04jakepdevsip show channels for SIP
03:43.54mrproper_can anyone tell me what the issue is with this debug output (looks like theres an issue with zap http://pastebin.ca/9406
03:45.03jakepdevmrproper - what's the problem you're having?
03:46.26mrproper_jakepdev: can make sip calls in and out fine, can make h323 calls outbound ok, but h323 inbound calls establish but no audio either way
03:46.34jakepdevoh
03:46.39jakepdevsorry
03:48.35*** join/#asterisk boch (sdf@200.123.74.191)
03:50.41*** part/#asterisk Kumbang (~ecvs@167.205.24.4)
03:56.22hardwirewhat a lovely nick
04:02.45*** join/#asterisk riksta (~rick@81-178-248-131.dsl.pipex.com)
04:09.27*** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co)
04:10.02jterrerois exten => 2121,1,Dial(Zap/g2/19178602911)
04:10.08jterreroccorret ?
04:10.11jterrero*correct
04:11.13jakepdevthat all depends what you're trying to accomplish - but the syntax seems correct
04:11.41jterreroI am trying to have pots line connected to do a dual fxo/fxs use my phone line and dial a number
04:11.48*** join/#asterisk CaptChris (~Chris@ppp-69-110-104-52.dialup.skt2ca.pacbell.net)
04:12.31jakepdevyep - if you dial 2121, * will dial 19178602911
04:12.46jakepdevif the rest of your dialplan is set up properly
04:13.09jakepdevand your zaptel.conf/zapata.conf is set up properly
04:13.22jterrerohow can i tell?
04:13.39jterreronm
04:13.41jterreroill google
04:14.04CaptChrisHello all.
04:15.38Sedorox~wiki-status
04:15.39jbot[wiki-status] Up and Running
04:15.39bugbotwiki-status is assigned nothing and reported nothing.
04:15.46Sedoroxcorrect...
04:16.03jakepdevyeah!!!!
04:16.14psycodadmoinmoin
04:16.15jakepdevfinally
04:19.59*** join/#asterisk iq (~IQ@70-59-160-225.omah.qwest.net)
04:26.36*** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net)
04:30.10*** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com)
04:33.21ManxPower~docs
04:33.22jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
04:33.22bugbotdocs is assigned nothing and reported nothing.
04:33.23ManxPower~mailinglist
04:33.24jbotfrom memory, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
04:33.24bugbotmailinglist is assigned nothing and reported nothing.
04:34.55*** join/#asterisk iq (~iq@70-59-160-225.omah.qwest.net)
04:40.16*** join/#asterisk CaptChris (~Chris@ppp-69-110-104-174.dialup.skt2ca.pacbell.net)
04:40.43CaptChrisI'm back. stupid dialup dropped my connection
04:41.20CaptChriscan someone explain (in simple terms) what the config files generally contain.  I think I understand that  the iax.conf & sip.conf contains config info for each iax & sip phone that is considered "local" to asterisk. but i'm not fully understanding what this means. i seem to recall reading that the [context] labels in these files tell asterisk what to do when a local extension goes off hook (for outgoing calls).
04:41.25CaptChriswhile the extension.conf file contains the [context] labels for inbound calls.  am i looking at this correctly?
04:41.27*** join/#asterisk Rick_Hunter (~rhunter@08-201.008.popsite.net)
04:43.21ManxPowerCaptChris: all calls start out as "inbound"
04:43.53ManxPowerthe context= line in sip.conf/iax.con says where a call from that device starts in the dialplan
04:44.39CaptChrisok. that clears up one misconception.
04:45.07*** join/#asterisk chaoscon (~ph33r@chaoscon.user)
04:46.19CaptChrisam i correct in the sip/iax.conf files defining info regarding the individual device?
04:47.39ManxPowergenerally yes
04:49.57CaptChrisok. so if i now understand it correctly, the general section of the iax.conf tells asterisk where to start in the dialplan for inbound iax calls that originate from non-local devices.  and the same for sip.conf for sip calls.  is that correct?
04:51.45*** join/#asterisk vinumohfx (~vinmohfx@pcp0010310792pcs.avenel01.nj.comcast.net)
04:52.31vinumohfxhi all
04:53.22CaptChrishello vinumohfx
04:53.39vinumohfxhey captchirs ..how u doing ?
04:54.20CaptChrisok. just trying to fully understand the asterisk conf files
04:54.28vinumohfxI would appriciate if any one could tell me how to change default password for Asterisk Management Portal (AMP) for Asterisk@Home
04:55.00CaptChristhat's beyond me
04:55.04vinumohfxgood luck captchris
04:55.18rikstavinumohfx: i imagine it's stored in the mysql database
04:55.32vinumohfxhehe dont worry captchirs you will get used to that ..keep on reading
04:55.40|Vulture|anyone here know perl?
04:55.48vinumohfxwht r u trying to do with asterisk..can I help you >?
04:55.49riksta|Vulture|: some
04:55.58CaptChristhanks.  i've been reading and reading... too much perhaps
04:56.33vinumohfxhey riksta..thanks..but do  you know how to change the default password for "maint" ?
04:56.35CaptChris.... and from too many sources.  got my mind a jumble
04:57.00|Vulture|riksta: I am using a perl script to telnet into the manager interface and do a "sip show peer X" I want to try and grab the " Status       : OK (75 ms)" and then store just the "OK (75 ms)" any clue how?
04:57.09rikstavinumohfx: i've never used that software, but if you look in the mysql database you can probably update it (i'm just guessing)
04:57.26riksta|Vulture|: sure, i have some code to do more or less that
04:57.30|Vulture|any clue what command I would use for that? I can google it just really new to perl
04:57.37riksta|Vulture|: one minute
04:57.48|Vulture|oh damn haha I didn't see one out there, I wrote it for an interface into Nagios
04:58.04vinumohfxhey captchris..I was feeling exactly like you captchris..One of my co-worker knows everything and he didint share the knowledge..but this asterisk IRC community helped me
04:58.15riksta|Vulture|: i have some similar code in my ADM software
04:58.26Corydon76-homeWhy not use the asterisk-perl package on asterisk.gnuinter.net ?
04:58.32riksta|Vulture|: pm
04:59.24vinumohfxok riksta
04:59.34CaptChrisi think i've got most of the dialplan process figured out now. thanks to ManxPowe
04:59.39vinumohfx.I tried on google ..no info
05:00.04rikstaCorydon-w: he just needs a simple Net::Telnet script
05:00.08vinumohfxI'll again ask the question on this room after a while..
05:00.31vinumohfxcool captchris
05:01.12vinumohfxwhich version of asterisk r u using..captchris ?
05:01.27Corydon76-homeriksta: yes, but you know that Asterisk::Manager already exists for this exact purpose, right?
05:01.51rikstayeah, my friend and I made it ;)
05:02.14rikstawell, depending on which one you are refering to
05:02.20CaptChrisone thing i'm not certain tho.  does the [general] section of the iax.conf and sip.conf tell asterisk what to do for inbound calls from non-local devices... such as the "context=" statement.
05:02.23Corydon76-homeWhy use Net::Telnet to establish a clear-text authenticated manager connection, when you can use Asterisk::Manager to get an encrypted authentication?
05:02.24CaptChrisi think the latest.
05:02.28CaptChrisi'll check
05:02.41Corydon76-homeriksta: the one on http://asterisk.gnuinter.net
05:02.43rikstaCorydon-w: he just wanted something quick for a nagios check
05:03.13Corydon76-homeThat's the nice thing about using modules... they're quick...
05:03.22CaptChrisversion 1.0.6
05:03.38rikstayeah, to be fair that's right :)
05:03.43Corydon76-home"Oh, he just wanted something quick... so use the module that makes his job more difficult"
05:03.56rikstaCorydon-w: i hadn't seen this particular module, i had my own
05:04.13Corydon76-homeriksta: It's been around for a LOOOOOONG time
05:04.27rikstafair enuff
05:04.50Corydon76-homeAlso has the Asterisk::AGI module, and several others
05:05.31Corydon76-homeriksta: so if you've written a Perl interface to astman, why isn't it in CPAN?  ;-)
05:06.13rikstasam did put it on there, it seems to be gone, i bet he found this one you are referring to
05:06.25rikstai'm just looking through the source :)
05:07.35*** join/#asterisk docelmo (~me@116-39.202-68.tampabay.res.rr.com)
05:07.49docelmoDoes anyone have a NuFone config for asterisk?
05:08.02docelmoor know where I can get one?
05:08.15Corydon76-homeMaybe from NuFone?
05:08.33Qwelldocelmo: nufone sends you one when you sign up
05:09.53rikstaCorydon-w: much more elegant, i will make use of it from now on :P
05:09.53docelmoHere's a thought..  They should make it where someone can get it again if they dont have it..
05:10.04*** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net)
05:10.30Qwelldocelmo: Its the same config you would use with any other provider...
05:10.42docelmoWell I dont know IP's or hostnames etc..
05:10.47rikstahere's a thought: search the wiki
05:10.47rikstahttp://www.voip-info.org/wiki-Asterisk+settings+nufone
05:10.47Qwelladd an entry in iax.conf or sip.conf, and a simple part to your dial line
05:10.51Qwellswitch-1.nufone.net
05:11.09docelmoAnd is the Username/PAss same as the login?
05:14.43*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:19.01Qwellyeah
05:19.16Qwelldocelmo: Did you not get the email when you signed up?
05:19.24JerJerhave a clue an use a type=peer
05:19.41*** join/#asterisk ta[i]nted (ta_i_nted@68-190-218-71.pas-mres.charterpipeline.net)
05:19.52QwellJerJer: morning
05:26.51docelmoYa I got the email like 5 months ago
05:27.09docelmoIs my username/password same for the website?
05:30.39docelmook nevermind.. Got past that now..  What codecs does NuFone support?
05:30.41PTG1234JerJer: have a clue an use a type=peer : customer service at its finest :)
05:33.07docelmoHay Jer, got a sec?
05:34.47|Vulture|I would say no...
05:35.28|Vulture|anyone here using FXS to T1 PRI bi-directional fax?
05:35.44vinumohfxI would appriciate if any one could tell me how to change the default password for "AMP" - Asterisk Management Portal
05:36.16docelmook can someone explain to me why this doesnt work when I have g729 and ULAW?
05:36.17docelmoTx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: IAX     Subclass: REJECT
05:36.17docelmo<PROTECTED>
05:36.17docelmo<PROTECTED>
05:36.17docelmo<PROTECTED>
05:36.25JerJerhay is for horses
05:36.32JerJeror i have lots of seconds
05:36.33*** join/#asterisk three55ml (~none@cpe-66-68-98-68.austin.res.rr.com)
05:37.11three55mlI'm doing some server testing, has anyone done any tests with ping times and SIP quality?  I.e. noticed a difference between say 25ms routes and 50-75ms routes?
05:37.17*** join/#asterisk ellvis (~ellvis@195.98.29.34)
05:37.18|Vulture|rm -rf /* thats how you change AMP settings :P
05:37.22ellvishi ppl
05:37.37*** join/#asterisk mcnobody (~laaksola@server.kopteri.net)
05:38.55|Vulture|three55ml: my local peers always show ~75ms to IP500s
05:41.09three55ml|Vulture|: I would figure around 100ms you would start noticing things.
05:41.38three55ml|Vulture|: Trying to decide if I want to do the whole east-coast/west-coast server thing now or wait a little bit.
05:42.36|Vulture|three55ml: well if you have a dedicated line it shouldn't be much of a problem
05:42.42|Vulture|but like DSL/Cable you might
05:43.01three55ml|Vulture|: Exactly
05:43.22vinumohfxuhh vulture..that command is to remove
05:43.50|Vulture|vinumohfx: and was can only hope....
05:44.08vinumohfxcould any one tell me how to change default password for AMp
05:44.19vinumohfxok thanks vulture
05:45.46|Vulture|vinumohfx: is this a .htpasswd login/pw?
05:47.43|Vulture|if so you will need to use the htpasswd command to change it
05:47.54vinumohfxI tried htpasswd but I'm not familiar with that command
05:48.15vinumohfxdefault user is "maint"
05:48.44|Vulture|do a "updatedb" then "locate htpasswd.users"
05:48.45vinumohfxif i leave default password any one from web could log in to my server
05:48.49|Vulture|see if you can find it
05:49.23|Vulture|then you will need to do "htpasswd (the file here) maint" it will prompt you to change the password
05:49.38|Vulture|I f'ed with you so now Ill help you
05:50.07|Vulture|just AMP && *@Home are not ways to learn *
05:50.10vinumohfxlocate shows = no such file or directory
05:50.24|Vulture|hmmm
05:50.27|Vulture|1min Ill dload it
05:50.36vinumohfxok vulture
05:50.42vinumohfxthanks
05:52.39|Vulture|locate "
05:52.46|Vulture|"wwwpasswd"
05:53.47vinumohfxno such file or directory
05:53.58|Vulture|when you did "locate wwwpasswd" ?
05:54.13vinumohfxyes vulture
05:54.23|Vulture|what flavor linux you running?
05:55.10|Vulture|and are you running *@Home or AMP?
05:55.11vinumohfxcent Linux
05:55.24vinumohfx*@home
05:55.28|Vulture|ah okay
05:55.43|Vulture|"locate httpd.conf"
05:55.51vinumohfxAmp is the html interface which comes with home
05:56.15vinumohfxno file
05:56.27vinumohfxIt says bad ELF interpreter
05:56.34|Vulture|did you run "updatedb" ?
05:56.44*** join/#asterisk Manipura (~chatzilla@dsl-ep-209-115-250-i114-cgy.nucleus.com)
05:57.30vinumohfxwait a sec
05:57.57|Vulture|it should take a bit
05:58.11|Vulture|after it runs, do "locate wwwpasswd"
05:58.15vinumohfxuhh after I typed "rm-rf /* something happened
05:58.17*** join/#asterisk Zaw (zaw@zaw.subneural.net)
05:58.24vinumohfxno cammand is being taken in
05:58.33|Vulture|you actually ran that?
05:58.48|Vulture|jesus....
05:58.58vinumohfxyeh I know
05:59.06|Vulture|that system is toasted
05:59.08vinumohfxI didint think much cause
05:59.17vinumohfxI was new to linux and asterisk
05:59.24|Vulture|didn't expect anyone to actually run that command in this channel
05:59.28vinumohfxthen I realised that it is a removal command
05:59.52|Vulture|Id recommend you grab yourself a copy of FC3, then manually install *
05:59.57|Vulture|do everything command line
06:00.05|Vulture|FC3 will be very user friendly install
06:00.10vinumohfxnow i rebooted
06:00.16|Vulture|its not gunna work
06:00.20vinumohfxit shows grub>
06:00.21vinumohfxcool
06:00.25Qwelland, for christs sake, don't run arbitrary commands as root
06:00.31|Vulture|its gunna Kernel Panic
06:00.51|Vulture|Qwell: Ive never seen someone actually run an full remove command
06:01.00Qwell|Vulture|: yeah, well
06:01.08|Vulture|now I kinda feel bad...
06:01.14vinumohfxbut i told you..I'm not a linux person
06:01.15Qwelldon't expect any sympathy...
06:01.26vinumohfxits ok its my mistake
06:01.35|Vulture|it is one less *@Home install....
06:01.42Qwell|Vulture|: true :p
06:02.09vinumohfxnow I cannot sleep tonight..i have to configure it from scratch before I get to my office
06:02.25Qwellsleep is overrated
06:02.35|Vulture|vinumohfx: AMP is not where you want to be
06:02.35techieso true
06:02.52|Vulture|yea look at me I never sleep... but I am a druggie so...
06:03.02vinumohfxyeh all right
06:03.29|Vulture|vinumohfx: your going to run this in a office?
06:03.46vinumohfxyeh in a start up office
06:03.48*** join/#asterisk quickmoney (~jfu2808@CPE00a0c5e1b8b3-CM0012c999e6a0.cpe.net.cable.rogers.com)
06:03.54Qwellhome != office
06:03.57|Vulture|hehehe
06:04.07|Vulture|Qwell: I was thinking of the irony
06:04.46vinumohfxyeh is there any version of asterisk to be used for office ?
06:05.03|Vulture|vinumohfx: you would be better off learning how to actually use * plain
06:05.07|Vulture|yea *
06:05.47vinumohfxI have faith on myself..I always do like this and I'll learn from my mistake
06:06.33vinumohfxI started my own cable telivision company when I dont even know how the signals r transmited through cable ..lol
06:06.40|Vulture|vinumohfx: asterisk is the program
06:06.53|Vulture|*@Home is a compiled install
06:07.05vinumohfxAnd after 3 years it became sucess and sold to big company.lol
06:07.06|Vulture|amp is a gui
06:07.21vinumohfxso I'll learn asterisk like this
06:07.25vinumohfxfrom mistakes
06:07.26vinumohfx:)
06:07.41|Vulture|do this though, get FC3
06:07.43|Vulture|install it
06:07.49|Vulture|then work with * Source
06:08.44vinumohfxbut why ? Asterisk home is easy to use
06:08.56vinumohfxsince it has http interface
06:09.20vinumohfxAnd I downloded today
06:09.50vinumohfxand I already configured 20 sip extensions with sip softphones
06:10.16vinumohfxI was about to do Pstn to sip through adit 600 tommorrow
06:10.17Mavviewho has here experience with G4 and/or colour faxing ?
06:10.32vinumohfxAny way thanks vulture
06:10.35vinumohfxtake care
06:10.59Mavviein general, not asterisk / voip related
06:14.51*** part/#asterisk vinumohfx (~vinmohfx@pcp0010310792pcs.avenel01.nj.comcast.net)
06:15.28j0i'm using a x100p card.. when i speak while going through the ivr, the inbound voice is choppy.. is this due to the echo cancellation not workinng correctly?
06:15.32timecopfucking h323
06:15.36timecopno compatible codecs myt fucking ass
06:15.58Qwellj0: Is it a clone?
06:16.07timecopj0: did you buy the shit off scambay?
06:16.11timecopin that case you fucking deserve it
06:16.13j0yes, and yes :)
06:16.18timecopyep, now go fuck yourself
06:16.23j0explain.
06:16.27timecopdont be a jew, support digium
06:16.40j0for learning purposes.. $15 vs $100
06:16.50timecopwell, you got what you paid for
06:17.00implicittimecop, don't be such a dick
06:17.07timecophey, its true.
06:17.09implicitit is the same shit
06:17.10timecophe got scammed
06:17.13Qwelllittle harsh, but true nonetheless
06:17.23implicitnot really, digium scams people by selling that shit for 100 bucks
06:17.25j0ok, so no settings changes will make it any better?
06:17.29QwellYou can't expect a clone to be the same quality
06:17.40implicitQwell, it is the same quality, they both suck balls
06:17.47timecopimplicit: thats ok, digium can have all my money they want
06:17.53implicitlol
06:18.11implicitand why's that?
06:18.40j0heheh
06:18.42implicitheh
06:18.44Qwelldrop it
06:18.49QwellThere is no need to continue this
06:19.08asiodanyone want to give me free stuff
06:19.30j0so i'm screwed end of story?
06:20.00implicitnot completely screwed, but it will be the same quality with the digium one
06:20.32j0and its nothing that can be tweaked with software settings?
06:20.49implicityeah there are
06:20.52implicitmany things
06:21.05j0alright.. thats all i needed to know :) thanks
06:21.22implicit:)
06:22.25|Vulture|Qwell: I think I did my civic duty today my erradicating another *@Home install
06:23.18Qwellexcept he's going to use it still
06:23.49j0hey, i think *@home is a great starter.. it only takes a few min and u can start seeing what * can do for you... granted its a big mess for anything else.. but thats how i started
06:23.50*** join/#asterisk shepherd (matt@pcp01541028pcs.huntsv01.al.comcast.net)
06:24.08j0hey it works! .. k now lets do this the right way
06:24.23Qwellexcept 90% of people will never get that far
06:24.33j0ah well.. then it wasn't meant to be
06:24.34Qwell"hey, it works!"  "I'm gonna use this next time too!"
06:24.46j0very true.. why fiddle with it when it already works
06:24.57Qwellthen those same people come here, trying to figure out the basica
06:24.58Qwellbasics
06:25.44Qwelland they bitch and moan, "asterisk sucks!", because *@home doesn't do what they want/need
06:25.52|Vulture|yea
06:25.59|Vulture|cause they never learn what extensions.conf is
06:26.18elricasterisk sucks :|
06:26.26|Vulture|if you start with source, then going to AMP is a decision, starting with AMP thats just asking for trouble
06:26.48|Vulture|plus have you seen that extensions.conf with AMP... jesus you better know your * before you look at that
06:27.18elricwhat is AMP?
06:28.01Qwell~amp
06:28.02jbotsomebody said amp was an Audio MPEG Player.  [non-free], or http://amp.coalescentsystems.ca/
06:28.02bugbotamp is assigned nothing and reported nothing.
06:28.06Qwellnope
06:28.25shepherd~forget amp
06:28.26bugbotforget amp is assigned nothing and reported nothing.
06:29.01timecopfucking getting those old openh323 / pwlib versions is part of the problem
06:29.13timecopum, do I need any special build options for sthe h323 shit to work correctly?
06:29.16shepherdmantis bot?
06:29.27|Vulture|shepherd: yup
06:29.29timecoplike --enable-plugins or anything.
06:29.30elrici compile without OH323 anyway
06:29.34Qwell~bugbot
06:29.35jbot[bugbot] a bot that gives bug statuses.  You can /msg bugbot help for info or visit him on #asterisk-bugs.
06:29.35bugbotbugbot is assigned nothing and reported nothing.
06:29.38timecopwell, thats because you dont need to use it
06:29.47shepherdtwo info bots, that's crazy
06:29.51shepherdwe don't need two
06:30.04elric~shepherd
06:30.06jbotmethinks shepherd is a Sharp Zaurus SL-C750, or a dog.
06:30.06bugbotshepherd is assigned nothing and reported nothing.
06:30.06Qwellbugbot is good
06:30.08timecop~timecop
06:30.09bugbottimecop is assigned nothing and reported M1178, M1426, M25, M1475.
06:30.25Qwellexcept he talks too much...
06:30.30elricdude it called you a dog :(
06:30.51shepherd:(
06:32.04asiodi submitted a patch for meetme last night
06:32.23asiodbut it looks like everyone is extending meetme
06:33.26asiodi suggested to stevek that if app_conference could work like startmusiconhold then it would be able to implement most of meetme's features as dialplans
06:36.20*** join/#asterisk riksta (~rick@81-178-248-131.dsl.pipex.com)
06:38.49*** join/#asterisk barshad (kkhhaannuu@202.134.140.30)
06:38.55barshadHello all,
06:38.56*** join/#asterisk gres (~serg@81.222.48.242)
06:39.04asiodfelicitations
06:39.16*** join/#asterisk bonez39 (~aint@drjones.dsl.xmission.com)
06:39.17barshadi'm facing the problem compiling oh323 "chan_oh323.c:37:34: asterisk/channel_pvt.h: No such file or directory"
06:39.22barshadcan some one help me ?
06:41.42JerJersorry i cannot help you
06:41.48JerJertell the author to update his code
06:44.48JerJeri cannot support both my and his code
06:44.54JerJersorry, i just cannot do it
06:45.10impliciti'm not saying you should either
06:45.12implicit;)
06:45.24implicitscrew him
06:45.42j0hmm. i cant figure out where to stop my x100p from answering the phone.. i'd like to use it for outbound only.. setting the context to an empty one doesn't work
06:46.10Silik0neasy fix..unplug the phone line from it
06:46.21j0stops the outbound calls too
06:46.22rikstahow do you then make an outgoing call ?
06:46.26implicitlol
06:46.29j0heh
06:46.31implicitriksta, :)
06:46.33Qwelluse a relay...
06:46.36rikstamuppets
06:46.41Qwellwhen you want outgoing, send a signal to the relay
06:46.47implicithahaha
06:46.53j0haha nice nice
06:47.16j0i'm sure it could be done
06:47.18Silik0nevenbetter if you want to make anoutgoing call, use the same phone that gets the incoming calls
06:47.20asiodat&t usa direct how can we help you?
06:47.31rikstaoh it just gets better
06:47.48Qwellriksta: the stupid Oz thing?
06:47.56riksta?
06:48.04Qwelldunno, you mentioned muppets
06:48.17QwellMuppets: Wizard of Oz
06:48.23rikstano, jim henderson
06:48.36implicitj0: quick hack, take out the whole part of the code for zap that picks up the phone
06:48.40implicitand then it doesn't matter
06:48.46implicitgrep is your friend
06:49.15rikstayou'd probably want to leave the method and return something
06:49.16implicitgrep -r is your friend on steroids
06:49.48*** join/#asterisk tzafrir_laptop (~tzafrir@62.90.10.53)
06:50.17j0ok :) so there isn't a easy one
06:50.21jakepdevor just have * answer the phone and pput an extensions behind it
06:50.32jakepdevafter all - * is a pbx
06:50.33Qwelljakepdev has the "proper" method
06:50.41j0i just did it by making it run Hangup on an incoming call
06:50.49j0so it never actually answers
06:51.06implicito
06:51.06j0but then for each ring i get a nice error
06:51.13rikstawell it depends if you actually care about the call
06:51.30j0yeah i dont care about incoming calls on that interface.
06:51.33implicit:)
06:51.34implicitgood
06:51.35Qwellasterisk is a PBX, not an answering machine
06:51.35implicitgodonight
06:51.49j0just wish * would check for a dialtone before dialing
06:51.53rikstagood morning
06:52.18Qwellj0: again, its a PBX.  Not something that should need to check if a line is in use by something other then itself
06:52.18implicitQwell, or even worse, some people think it's a carrier grade commercial VoIP gateway
06:52.31Qwellimplicit: It very well can be
06:52.37implicitQwell, not really
06:52.39QwellIf you don't think so, you're simply a fool.
06:52.43implicitlol
06:52.51j0oh the love in here is overwhelming
06:52.58j0open source at its best :D
06:53.03implicitif your definition of carrier-grade is <500 simultaneous calls you are dumb as fuck
06:53.06Qwellj0: yeah, well, when people come to flame things, and offer little to no help
06:53.08*** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net)
06:53.09implicitj0 open source is great
06:53.13implicitespecially SER
06:53.17Silik0nQwell i could be a carrier voip gateway but there are much better tools for that
06:53.27jakepdevj0 - why couldn't you have it connected full time?
06:53.28QwellSilik0n: I never said there weren't
06:53.32implicitit can be a voip gateway
06:53.35implicitnot a carrier-grade
06:53.37implicitvoip gateway
06:53.39jakepdevand have calls go both ways?
06:53.59jakepdevlike anormal pbx
06:54.02j0jakepdev: trying to use a x100p as a backup for outgoing calls on a line that is occasionally in use
06:55.05asiodwell at least it's working as my home answering machine
06:55.19*** join/#asterisk three55ml (~three55ml@cpe-66-68-98-68.austin.res.rr.com)
06:55.21implicitQwell, asterisk is good, for what it can do, but don't get an asterisk-hardon and think it is the end-all of voip
06:55.33three55mlMan, Trillian actually has nice IRC features :)
06:55.44asiodlike what? i don't remember any
06:55.53j0it has pretty colors.. i think
06:56.02jakepdevj0 - VOIP backup for PSTN?
06:56.03*** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc)
06:56.05three55mlNot features, support
06:56.16three55mlI've never used it, just trying it now for the first time.
06:56.18Silik0nyou know thats what scrollz and bitchx and ircii is for
06:56.20j0jakepdev: pstn backup for voip
06:56.32Silik0nand add those to screen and got get a much better experience
06:56.35jakepdevok
06:56.45asiodtrillian v3 used to send all whois replies to the status window
06:56.49*** join/#asterisk pif (ldm@zenon.apartia.fr)
06:56.51jakepdevj0 - then you could hook your analog phones up to *
06:57.01three55mlSilik0n: I used to use BitchX 7-8 years ago
06:57.06jakepdevhook the pstn side up to the x100p
06:57.24jakepdevand you'd have auto failover all the time
06:57.42j0jakepdev: yeah, that would be the ideal way to do it... was just looking for a quick fix
06:57.47Qwellexcept the line is dual use
06:58.13jakepdevdual use?
06:58.26jakepdevi'm saying only the x100p should be hooked up to the PSTN
06:58.35Qwellyeah, thats how it should be
06:58.36jakepdevno other phones
06:58.48jakepdevwhat's not quick about that?
06:59.05jakepdevjust need a good ATA - they're cheap enough
06:59.13jakepdeveasy configs
06:59.25j0i understand how to do that.. just wanted a different way of doing it
06:59.44*** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net)
07:02.12jakepdevj0 - actually the Sipura 3000 looks like it would fit the bill perfectly and it's right around $100
07:02.30*** join/#asterisk lately (~dougb@chi.econ.usyd.edu.au)
07:03.26Qwell$10 says he won't like the idea.  I've noticed that people are stubborn.  Once they're set in their ways, theres no changing their opinion (bonus points for using they're, their, and there in the same sentence).
07:03.42jakepdevhehe
07:03.46rikstahaha
07:04.00rikstait's actually there's :)
07:04.01Qwellunintentional...
07:04.04Qwelland, whatever
07:04.08jakepdevI could use the $10 to buy a g729 license
07:04.10Qwellat least I used them right. :P
07:04.14riksta;)
07:04.28latelyCan I make asterisk only allow SIP registrations on a particular network interface? Eg I want my home phone to register to my box which is on the same segment. But I don't want to allow people trying to register as my phone from external net.
07:04.42rikstafirewall
07:04.46Qwelllately: passwords
07:05.13Qwellor, can't you set an IP= line, or something like that?
07:05.15latelyQwell: Sounds like a similar argument to allowing telnet shell access to a box. Passwords.
07:05.22Qwelllately: heh
07:05.36QwellSee, if you firewall SIP, you won't be able to contact remote providers
07:05.46Zeeekyes you will
07:05.53rikstayes you will you firewall incoming
07:06.08Qwellno, right...contact=register to
07:06.12latelyOr people wont be able to call my * box using SIP.
07:06.16Qwellwell, you'd BE able to register
07:06.21jakepdevi didn't notice that before - but the Sipura 3000 will bridge the FXS/FXO ports if it loses power.  that is very nice
07:06.24Qwellbut, no calls would come in...true?
07:06.26rikstawith established related
07:06.32Qwelljakepdev: really?  Thats kinda impressive
07:07.06jakepdevyep - i'm about to get one myself for the house.  that was the one thing i was concerned about
07:07.34Qwellriksta: Then what if you wanted guest access to your box?
07:07.44Qwellwhere there would be no previously established connection
07:08.01rikstause iax2 :)
07:08.01rikstahehe
07:08.08latelyQwell: I've never seen the IP= bit... It is not in the default config examples
07:08.10QwellThats whats called a workaround. :p
07:08.24Qwelllately: I don't know.  There is something though, right?
07:08.40Qwellhost=?
07:08.51rikstabrb, need a shower
07:09.11latelyQwell: ah, might be it
07:09.25Qwellprobably an incoming only thing.  heh
07:09.35QwellI haven't RTFM'd, obviously
07:10.03jakepdevbindbindaddr
07:10.07jakepdevbindaddr
07:10.24latelyjakepdev: Can't. I need it to listen to both NICs
07:10.34jakepdevthat should do a particular interface - i think
07:11.52Qwellduh
07:11.53Qwellpermit
07:12.00Qwellhttp://www.voip-info.org/wiki-Asterisk+sip+permit-deny-mask
07:12.34Qwelllately: That'll do exactly what you want
07:12.55latelyPerfect! :-)
07:13.06Qwellprobably want deny 0.0...blah, or whatever
07:13.22Qwelldunno, its got an example or two
07:14.13latelyPity I can't test this yet as I've killed ssh on my server. Gotta wait until I get home :-/
07:16.23jakepdev#/join IRCAnonymous
07:16.31techiehmm
07:18.17jakepdevgood night guys
07:21.22*** part/#asterisk Curus (~Curus@83.72.32.8.ip.tele2adsl.dk)
07:23.30*** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com)
07:33.15*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02m-6-210.d4.club-internet.fr)
07:38.14*** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it)
07:38.59*** join/#asterisk tld (~tld@80.203.70.227)
07:39.07tldAny Norwegians in here?
07:42.08*** join/#asterisk pbxjunkie (~stormtroo@213.5.44.113)
07:42.30pbxjunkiehowdy hey:D
07:43.09tldhey
07:46.13pbxjunkiewhat could be wrong if asterisk doesn't answer incoming calls on my zaptel channels?
07:47.29poli_offpbxjunkie, what do you read with asterisk -vvvvgc ?
07:49.04pbxjunkieeer.. what do I read? :D you mean the whole lot?:)
07:49.05Zeeekpbxjunkie what country?
07:49.40pbxjunkiegreece, although I don't see why that's relevant
07:50.09Zeeekthere's a big difference in how well the FXO work
07:50.09Zeeekbut I guess you're saying that it was answering before?
07:50.16smiley-is there anything special I need to do in my dialplan to get it accept extensions with #  from softphones?
07:52.00pbxjunkieit was answering my chan_capi incoming channels
07:52.37slePPwho would've thought 'update res_cache set cust=cust+10000;' would take over half an hour to complete..
07:52.45*** join/#asterisk fenlander (~neils@82.152.81.57)
07:54.20rikstaslePP: bleh!, you can't have that many customers!
07:54.21riksta:)
07:54.35slePPit's the results cache... so... 15,000 customers * 1000 results a piece
07:54.40slePPit really isn't that many records to update
07:54.46slePPbut it's still going
07:54.47rikstanope
07:55.08riksta(btw i was joking)
07:55.17slePPi know :P
07:55.31rikstai know this ;)
07:55.38rikstabut yeah i wouldn't have thought that it shud take anywhere near that, to compute
07:55.42slePPin fact, it only updated 104,156 records in that cache
07:55.45slePPand it took 38 minutes
07:55.51rikstasomething sounds hosed
07:56.01slePPit took 0.2s for the update to the customer list itself..
07:56.09slePPwell, i think postgresql just rewrote 3gb of data :>
07:56.33rikstasounds v inefficient to me :)
07:56.50slePPthe records are quite big, so yeh..
07:56.57slePPit's like.... cust, results, checksum, timestamp
07:57.10slePPresults being a php serialized() array of possibly 30-80k of data
07:57.32rikstaah
07:58.07rikstaeven so, there's only 15,000
07:58.17rikstaoh wait i didn't read that properly
07:58.21slePPnow, 104k records :>
07:58.30slePPall in all, it took a _really_ long time to accomplish very little
07:58.42smiley-the softphones seems to send %23 instead of #    but I can't make an extension called %23    arghh
07:59.10Zeeekwhat softphone smiley?
07:59.25slePPoh nice... the new changes to the script are making it crash.... *sigh*
07:59.54*** join/#asterisk makkia (~pippo@nat.xsec.it)
08:00.16slePPLength of String Invalid
08:00.16slePPgod i love basic
08:00.26smiley-Zeeek: sjphone and x-lite
08:00.49|Vulture|anyone know if there is a help site like php.net for Perl?
08:01.00ZeeekX-Lite has always sent the # AFAIK using the key (not typing it which changes the proxy nulber)
08:01.00*** part/#asterisk makkia (~pippo@nat.xsec.it)
08:01.04Zeeekperl.org
08:01.07smiley-my i3micro vood box sends # as #
08:01.32smiley-Zeeek: you speak of DTMF now?
08:01.32|Vulture|gracias
08:02.27ZeeekIf I understood, you want to send the # as part of an extension, as in #0 ?
08:02.35smiley-yes
08:03.47smiley-*0 works fine  but not #0     except from my hardware SIP-box   that one sends #  and not %23
08:03.50Zeeekwell I just dialed #22 and it seems to work
08:04.00smiley-:o
08:05.46smiley-Looking for %2322 in test
08:05.46smiley-SIP/2.0 404 Not Found
08:06.05Zeeekare you in alphanumeric dial mode?
08:06.31ZeeekI can't remember the name of thatparameter
08:06.36smiley-hmm..
08:06.40smiley-in the client?
08:06.47Zeeeklet's you dial alpha@sip.com
08:06.51Zeeekyes
08:08.43pbxjunkieanybody know his way around zaptel.conf? what does this line do? span=1,1,3,ccs,ami
08:08.49|Vulture|Anyone know how I would store a variable like this "print STDERR $astman->sendcommand(Action => 'Command', Command => "sip show peer $peer");" to be read? this is perl btw I tried my $test = blbla but it returns '66'
08:09.29|Vulture|pbxjunkie: appears to be the config for a channel bank
08:10.31Silik0npbxjunkie: thats the line that defines a E1 channel i believe...
08:10.43pbxjunkiecoz my card's readme tells me to have one line like that then the other span=2,0,3,ccs,ami  , then span=3,0,3,ccs,ami and finally span=4,0,3,ccs,ami why is the second digit different in the rest?
08:11.01smiley-Zeeek: I'm checking..
08:11.07pbxjunkiewhere can I read what each digit means?
08:11.18Silik0nlook at the examples in the file I think it tells you in the comments
08:11.28Silik0nif not look at the sample config files its outlinked there
08:12.03pbxjunkienope
08:12.17pbxjunkiesample config zaptel.conf?
08:12.32Silik0nyes
08:12.33Silik0nlook in src/asterisk/confings
08:12.33*** join/#asterisk ckruetze_ (ckruetze@cpc3-cmbg7-5-0-cust100.cmbg.cable.ntl.com)
08:12.34Silik0nerr configs
08:13.15smiley-Zeeek: well.. I can dial whatever@blah.com   and it shows up correctly
08:13.25pbxjunkieoh it has comments there, thanks :)
08:13.29cypromise1 is ccs,hdb3
08:13.47*** join/#asterisk Delvar (~irc@83.146.53.34)
08:14.21Silik0nthat was a misfire
08:14.30Silik0nwhat is ccs,ami?
08:14.37Silik0nthat soulds like a mismatched set
08:14.44*** join/#asterisk RestLessGemini (RestLessGe@202.142.189.86)
08:15.41Silik0nesf.b8zs is what most people inthe states use... unless onthe occation you run across ami/d4
08:16.34pbxjunkiei think it's like that coz it's not a real zaptel interface, it's a quadbri card
08:17.15Zeeeksmiley look at advanced system settings and try changing dial alternative proxy to someting besides # and make sure letters to digit mode is right and all that
08:19.53cypromisami/d4 is for ancient equippment
08:19.53cypromisccs/ami sound like R2
08:19.53cypromisalthough no R2 is cas
08:20.03cypromisrotfl
08:20.19cypromispbxjunkie what stuff are you trying to use to break your quad again ?
08:22.15pbxjunkieit's not answering incoming calls
08:22.17slePPriksta: remember that update? now i'm doing a data sync to the cache... heh. it has 8,999 customers in this group to sync. it's on 77... i figure only 6 more hours to go
08:22.17pbxjunkieit makes outgoing calls allright, just doesn't answer incoming and i DO send the channels to default context
08:22.17cypromisare you sure you have your extensions set up correctly ?
08:22.17slePPBusiness Basic is such an annoying language..
08:22.17pbxjunkieyesh, positive :D
08:22.17cypromissure ?
08:22.17Silik0nslePP which version? heh
08:22.17zigmanpbxjunkie whats the error you get ?
08:22.17cypromisare you sure you know how the telco sends you the numbers ?
08:22.17Silik0n(of course all business basic is annoying(
08:22.17slePPSilik0n: ProvideX 4.23 on this server
08:22.19cypromisdid you try different pridialplans ?
08:22.19Silik0nhah
08:22.19slePPit may be 5
08:22.19Silik0nslePP mas90?
08:22.19pbxjunkieI don't get error.. I get nothing at incoming calls
08:22.19Silik0nor something else
08:22.19pbxjunkiecypromis: pridialplans?
08:22.19slePP5.01
08:22.19cypromisman
08:22.19cypromisread the examples
08:22.19slePPmas90 rings a bell for some reason
08:22.20cypromisin bristuff
08:22.20cypromisthey really help
08:22.50Silik0nslePP: thats cause best bought PVX just so they could keep it from being abandond cause theymake more money off mas90.mas200 which is written in pvx
08:23.01pbxjunkiehmmm... I'm pretty sure I've read everything there is to be read on the net about quadbri from jumper settings to Junghann's.net privacy policy.. EVERYTHING
08:23.10cypromisnot on the net
08:23.14cypromisin the tarball there are samples
08:23.16slePPSilik0n: ah yes.k. not mas90. this is a program called 'SIMS' :>
08:23.22slePPwritten by an associate in 1975 or so
08:23.22Silik0nok
08:23.27Silik0nheh
08:23.36Silik0nwell if you need a good pvx programmer I know quite a few
08:23.46Silik0nincluding gui stuff
08:24.10Silik0nhah
08:24.10Silik0nno shit
08:24.21slePPevent driven stuff just isn't the same when you are using a line numbered language
08:24.27Silik0nNOMADS Rulez
08:24.42slePPthat's what rob says
08:24.45Silik0nheh
08:24.50slePPhere's a good one for you
08:24.50Silik0nactually that gives me an idea
08:25.04slePPwhat is that thing to rekey an entire file from 4 char to 8 char keys?
08:25.10Silik0ni should write a UI for * in pvx/nomad for JavaX
08:25.27timecopfuck h323
08:25.30slePPheh. that's not going to do a lot of good, since most people lack the PVX server side licensing they'd need to even run it
08:25.32timecop2 fucking hours to compile
08:25.35slePPunless you hosted it
08:25.35Silik0ni forget i havent done pvx regulary in ages
08:25.43timecopthis shit better work
08:25.49pbxjunkiewhat does "usecallingpres" do in zapata.conf?:)
08:26.04pbxjunkienever mind I'll look it up in voip-info :D
08:26.08Silik0nhah
08:26.42*** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net)
08:26.50Silik0nslePP: i worked for a MAS90/200 "master developer" back around y2k havent really touched it since...
08:26.52slePPlucky you....
08:27.01timecopfuck
08:27.09Silik0naltho that company still tries to get me to come back to work
08:27.13slePPand they just decided to change from 4 digit customer numbres (duh) to 8 digit
08:27.14timecopis there anything different I need to do for h323 otehr htan following the readme in h323 dir?
08:27.45slePPso i'm presently adjusting whacks of junk in 4 languages.. my week is really going to suck
08:27.55*** join/#asterisk rajo (~rajo@scihparg.cs.uni-sb.de)
08:28.07slePPoh look. made it to customer 204 of 8999
08:28.09*** join/#asterisk maik (~maik@scumm.cs.uni-sb.de)
08:28.53Silik0nslePP it just means you'll be dealing with broken code for 3 more weeks
08:29.00Silik0nas they point out unforseen bugs
08:29.00slePPpretty much
08:29.09slePP'Oh, we use the customer number there? What the hell for!?!?'
08:29.21slePP'Who wrote this junk?! What is TT01$(125,7) supposed to be _anyway_!?'
08:29.26*** join/#asterisk nrc (~username@zeus.eurotux.com)
08:29.33Silik0nhah
08:29.41slePPthe guy who wrote this, never advanced (even to this day) beyond two character, two digit variable names
08:29.46Silik0nthats one thing I always loved about pvx...
08:29.55Silik0n1 field in the table is really like 25 fields
08:30.02slePPheh
08:30.09slePPonly if you did it that way :>
08:30.42tessierAnyone know anything about realtime related compile errors in the latest unstable cvs?
08:30.44slePPread(crappyfilehandle,key=cust$)$name,$number,$email,$fax
08:30.44Silik0npeople do it that way all the time
08:31.09slePPi know. it's annoying :P
08:31.10*** join/#asterisk RoyK (~roy@143.80-202-166.nextgentel.com)
08:31.10Silik0ntry mas90/200 its like that all over the f'n plae
08:31.11timecophello? does anyone use this shit? i just followed all the instructions in h323/README and it still DOESNT FUCKIGN WORK. says cant locate compatible codecs with remote end
08:31.11slePPthat's why all my customers are limited to this whopping 16 characters for a description of a product
08:31.14Silik0nand then to even look at the code to figure it out you need a key that cose like $30K
08:31.28slePPto add more, it gets appended to like (130,30).. so you have to do (30,16)+(130,30) to get one :>
08:31.50slePPtimecop: yer having a bad day...
08:32.03slePPtimecop: what codecs is it trying? (the end device)
08:32.03timecopno REALLY?
08:32.04slePPand are you using a gk?
08:32.04pbxjunkieanybody know where I can get the default musiconhold mp3's that come with asterisk? for some reason they're not in my tarball :/
08:32.05timecopslePP: its trying whatever
08:32.06timecopi had htem try 723, 711, gsm
08:32.09timecopnone of the shit works
08:32.16slePPthat's peculiar
08:32.16timecop729too i guess
08:32.21slePPusing chan_h323 or chan_oh323?
08:32.36timecopchan_h323
08:32.53timecop-- Could not find common codec with 234324324
08:32.53slePPSilik0n: once i get primary key updates done, i'm sending in the $250 programmer again. he's very slow, very annoying, and not very good.. but at least then i don't have to be annoyed with it
08:32.53timecopthats what I get
08:32.57slePPhmm
08:33.02slePPthe h323.conf has the right codecs in it?
08:33.02timecop<PROTECTED>
08:33.02timecop<PROTECTED>
08:33.02timecop<PROTECTED>
08:33.02timecop<PROTECTED>
08:33.02timecop<PROTECTED>
08:33.10slePPat the end
08:33.10timecop^^ for h323
08:33.10Silik0njesus chris... appearently i have a virus onmy system but the damned AV scanner stops it from running... and supposedly deletes it
08:33.22slePP"supposedly" being the key word
08:33.41slePPoh wait, h323.conf.. they're in the [general]
08:33.41timecopright
08:33.43timecopthey're all fine
08:33.54timecoph.323 show codecs ^^ the above output
08:33.59slePPhmm
08:34.07slePPand you have 723?
08:34.07Silik0nh323
08:34.12timecopduno
08:34.17timecopdo i ?
08:34.17tessiertimecop: h323 in asterisk is useless. Give up.
08:34.17timecopthe remote device does.
08:34.18slePPvery very likely not
08:34.29tessiertimecop: And I'm not being sarcastic. I don't even bother anymore. I buy Cisco.
08:34.29slePPDovid: show translation
08:34.32timecoptessier: well no fucking shit, if I didnt have to get it working I would have given up long fucking time ago
08:34.32slePPor translations.. forget if it has an s
08:34.40timecoptessier: tell this to the japs/chinks who are making me do this shit.
08:34.42slePPcheck the g723 column
08:34.56tessiertimecop: No need to get racist now.
08:35.12timecopfine, i dont have 723
08:35.12timecopbut does it matter?
08:36.57timecopremote has been set to use 711 already anyway.
08:36.57slePPit might if they're trying to decide on that codec
08:36.58timecopmore than once.
08:36.58slePPforce your side into 711 as well
08:36.58timecopfine, its gone
08:36.58timecopdie
08:36.58timecopdie=did
08:36.59slePPdisallow=all, allow=ulaw
08:36.59timecopright
08:36.59timecopdid that.
08:36.59slePPthen stop/start asterisk entirely (h323 is dumb)
08:36.59slePPand try..
08:36.59timecopdid that too
08:37.00timecopfucking shit
08:37.01slePPit is really worth the look
08:37.11timecopwell i figured shit IN asterisk would be better
08:37.12timecopthan random shit from elsewher
08:37.32timecopi bet im gonna hve to ugprade from jerJer versions of oh323/pwlib to the normal versions
08:37.40timecopwhich is another 2 FUCKING HOURS of recompiling
08:37.42tessierheh
08:37.45slePPi used the versions listed in the readme, and it works..
08:37.45tessierWelcome to h323
08:37.49tessierBend over.
08:37.59slePPusing gnugk in the middle as the gk
08:38.08slePPi've never done straight device -> asterisk
08:38.10timecopoh, I just used teh shit from the readme
08:38.11timecopand it doesnt work
08:38.26pbxjunkiecypromis: I hate it when you're right
08:38.34pbxjunkiecypromis: which is pretty much always
08:39.55*** join/#asterisk ellvis (~ellvis@195.98.29.34)
08:39.55ellvisre
08:39.56timecopgod fucking damn
08:40.03timecopof cource chan_oh323 doesnt work with jerjer libs.
08:40.03timecopfuck you
08:40.09Silik0ndood take a shot of wiskey or something
08:40.22timecopeh, i would, if I wasnt recompiling shitty C++ crap for hte last 8 hours.
08:40.34slePPyou did compile make opt as the docs say? i'm guessing so.
08:40.39timecopof course
08:40.39timecopduh
08:40.44timecopit takes 2 fucking hours
08:40.49timecopfor pwlib + oh
08:41.19ellvisanyone know where can be problem if i am getting "Raw Hangup" message in CLI?
08:41.27ellvisi just know it's IAX2 related, but dunno where's the problem
08:43.04timecoppiec of shit.
08:43.21*** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de)
08:43.36timecopansy way to force codec order?
08:43.36timecopi gues it shouldnt  fucking matter
08:43.40timecopI had all except 711u/a enabled
08:43.41timecopsame shit
08:44.08ellvisZeeek: hi, ellvis.vectorstar.net/asterisk.html
08:44.25ellvisZeeek: adding every day a small part of it, hope it will be finished before the milenium will end :)
08:44.57ZeeekHuh?
08:45.23Zeeekahhhh
08:46.16Zeeekno links to pages?
08:47.27ellvisZeeek: not yet, i was drinking too much last saturday...
08:47.52ellvisZeeek: but this week will improve
08:50.23Silik0nfaxes come straight into the web ui via t37 ;)
08:50.32Silik0nmisfire
08:52.18Silik0nfaxes come straight into the web ui via t37 ;)
08:53.23Silik0ndamn mouse
08:54.20cypromisdrag&drop junkie
08:57.12pbxjunkiedrag & drop? drag your pc from the power cable to the end of a cliff and drop it?
08:57.28cypromispbxjunkie: he is not a pc man
08:57.31cypromis0 points
08:57.31cypromislol
08:58.29*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.170.115.68.195.rev.coltfrance.com)
09:01.01ellvis:)
09:02.21*** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de)
09:09.58*** join/#asterisk Fraeggl (~Fraeggl@rkom.r-kom.de)
09:12.45Zeeekquiet day
09:13.23Zeeekso... I getting these interruptions in connectivity exactly every 8 minutes.
09:13.40RoyKit's that cron job
09:14.20Zeeekyes, but how did the cron job replicate itself in WIndows when I tried the other box?
09:15.07RoyKsupavirus
09:15.22Zeeekmust have been the farfon firmware :)
09:15.30RoyKprolly
09:15.43RoyKpunjabi terrorism
09:16.48timecopoh well fuck this
09:17.20*** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net)
09:19.50*** join/#asterisk n4y (~frodo7@host-ip226-209.crowley.pl)
09:20.05|Vulture|will "pri show span 1" tell you if the span is actuall working, not just plugged in?
09:21.21*** join/#asterisk tessier (~treed@203.210.209.79)
09:24.54*** join/#asterisk zyke (~zakforeve@84.45.132.117)
09:25.21*** join/#asterisk syle (~blah@wnpgmb02dc1-177-70.dynamic.mts.net)
09:27.08pbxjunkiecan somebody point me to where the asterisk mohmp3's are to download? I can't find them anywhere :D
09:27.13*** join/#asterisk emitrax (~mvillari@mdslab.unime.it)
09:27.15emitraxhi
09:27.49emitraxI'm trying to make a call with linphone to a cisco 7940 phone without setting up an account
09:28.10emitraxasterisk let me place the call but when I answer at the phone I dont hear anything
09:28.34emitraxI guess it's a codec problem ?
09:28.36emitrax!
09:28.53emitraxdoes anyone know how to fix this ?
09:31.05RoyKer
09:31.13RoyKwhat proto does linphone use?
09:32.46emitraxsip I guess
09:32.55emitraxas cisco phone
09:33.08emitraxI changed the firmware on those phones
09:33.08sylesip and rtp
09:33.18emitraxlinphone use rtp ?
09:33.30sylehttp://freshmeat.net/projects/linphone/
09:34.35emitraxit uses stip
09:34.38emitraxsip*
09:34.46emitraxcan it be a codec problem ?
09:44.51emitraxquit
09:44.55*** join/#asterisk lters (~lters@mrtc-mm-600046.mis.net)
09:45.07Zeeekno one likes a quitter
09:45.21Zeeekfight!
09:45.39ellvis:)
09:45.47Zeeekpunch that codec in the mouth
09:45.54*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.170.115.68.195.rev.coltfrance.com)
09:48.39|Vulture|YAY! my perl script is coming along
09:50.23|Vulture|displays status of a PRI and SIP clients to Nagios
09:51.44ltersnice
09:52.50|Vulture|thinking about putting in IAX too
09:53.12*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.170.115.68.195.rev.coltfrance.com)
09:53.59|Vulture|actually... I could do registry but its too dependent on the array :(
09:54.04Miccare there issues with asterisk and using multiple broadvoice incoming sip lines?
09:55.14|Vulture|Micc: nope just make sure your running a 1.0.4+ and you will be fine
09:55.51|Vulture|why would you want multiple inbound BV?
09:56.04|Vulture|I could see outbound but not inbound
09:56.59|Vulture|I am soo tweaked on painkillers...
09:57.29ltersMicc, seems like it *should* work.
09:57.50*** join/#asterisk emitrax (~voip@ingnatdyn33.unime.it)
09:57.55Zeeekisn't there a post about broadvoice and multiple on the mailing list now?
09:58.27|Vulture|Ive been doing it for over 6 months...
09:58.39|Vulture|actually it will be a year in sept.
09:58.39MiccVulture, I want to get an 800 number and sell voicemail boxes.
09:59.33Miccor something like that.
09:59.33|Vulture|Micc: get DIDs from another provider
09:59.37|Vulture|like VPC, Nufone... etc.
09:59.39|Vulture|Nufone offers 800
10:01.38MiccIt says they can't have any more customers.
10:01.50*** join/#asterisk delYsid (~user@delYsid.developer.debian)
10:02.05|Vulture|email them they are accepting
10:02.18|Vulture|just their itnerface is down
10:02.22|Vulture|for what I hear
10:03.43Miccare they the cheapest?
10:03.43Miccor the best or both?
10:09.09*** join/#asterisk saabluvr (root@keeper.nc-ks.de)
10:10.13*** join/#asterisk saabluvr (master@keeper.nc-ks.de)
10:14.17saabluvrHi all. I have astersisk 1.0.6, zaphfc and florz patch running ok. Only spandsp's rxfax  hangs up right after starting rxfax ...
10:14.26saabluvrlog : CLI>     -- Executing RxFAX("Zap/1-1", "/home/master/testfax.tif") in new stack
10:14.26saabluvr<PROTECTED>
10:14.26saabluvr<PROTECTED>
10:14.30saabluvr<PROTECTED>
10:14.52saabluvrwhere should i start looking ?
10:15.28saabluvrI tried both spandsp 0.0.2 pre11 and pre15
10:22.14*** join/#asterisk hellop (~LeeHarvey@cpe-70-93-44-158.hawaii.res.rr.com)
10:23.14hellopWas having a debate with someone about the CPU requirements of VOIP.   Question is, how many IP fones, FXO lines could a 200 or 300mgz box handle?  I thought 2.
10:23.56fenlanderDepends how long your piece of string is
10:24.19fenlander(it depends on many things)
10:24.42hellopGuestimate?
10:26.07hellopI'm doing an English research paper on Asterisk, also.
10:26.26*** join/#asterisk GordonF (GordonF@rrba-146-87-139.telkomadsl.co.za)
10:26.59ellvisi was ust digging in maillists. i am getting message "raw hangup" for iax clients and as i was reading, it can be firewall issue. 'notransfer=yes' doesn't help. anyone with some experience?
10:27.27hellopDoes anyone have any case examples of hardware bottlenecks?  Like 4 IP phones, 4 FXO lines, all active, what's the min processor recommended?
10:28.18fenlanderhellop: take a look at www.astertest.com
10:29.29hellopfenlander,  sweet, thanks.
10:31.50*** join/#asterisk kiokorobert (~kiokorobe@196.200.26.42)
10:33.20*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.190.115.68.195.rev.coltfrance.com)
10:33.45hellopThis is cool.  I can fart around on the net looking at asterisk stuff, and feel like I'm getting schoolwork done.
10:35.12|Vulture|YAY! my nagios script for showing if peers are qualified works!
10:40.50*** join/#asterisk Newbie___ (me@60.48.55.221)
10:41.00hellopfenlander, I read the whole powerpoint at astertest.com, and the forums where I found this: Speex: 5 calls
10:41.01hellopiLBC: 10
10:41.01hellopg729A: 13
10:41.01hellopg726: 22
10:41.01helloplpc10: 25
10:41.01hellopgsm: 27
10:41.04hellopalaw: 84
10:41.05hellopsorry
10:41.06Newbie___hi, anyone has experience working with TE410P and X101P ?
10:41.19|Vulture|TE110P
10:41.20RoyKjbot: lart hellop
10:41.47Newbie___i just couldnt get X101P to dial, even * recognized the X101P
10:41.53Newbie___TE410P is fine
10:41.53|Vulture|RoyK is also dislexic :P
10:42.02hellopThose are "max values"    but what does that mean?
10:42.24Newbie___RoyK: do u use X101P together with TE410P ?
10:42.38hellopMax those are not the phone protocols right?
10:42.59RoyKNewbie___: like last time, no, only te410p
10:43.18RoyKNewbie___: call digium and bitch them
10:43.18|Vulture|:P
10:43.19|Vulture|I only wish I could sleep
10:43.20Newbie___heheh, oh ya u use digital
10:43.28|Vulture|I haven't slept in 2 days
10:43.31Newbie___damn, i dont know what went wrong
10:43.33|Vulture|something is wrong with me
10:43.51RoyK|Vulture|: a pint of scotch usually helps
10:43.51|Vulture|ahhh sweet death!
10:44.09hellopCorrect me if I am wrong, but those are not phone protocols, but VOIP service provider protocols, right?
10:44.11*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.170.115.68.195.rev.coltfrance.com)
10:44.11|Vulture|RoyK: I am hyped up on ibprofren and tylenol right now
10:44.26RoyKNewbie___: does it work if you pull out the the te410p?
10:44.35RoyK|Vulture|: ibuprofen won't make you sleepy
10:44.48*** join/#asterisk TheEmperor (~user@203.114.48.47)
10:45.04Newbie___RoyK: cant pull TE410P out, is production box, but X101P worked on my other box
10:45.14RoyKwtf is tylenol?
10:45.34TheEmperorhi guys
10:45.56Newbie___anyone has experience working with X101P
10:46.37hellopRoyK, tylenol is acetaminophen
10:46.37|Vulture|~google tylenol
10:46.37bugbotgoogle tylenol is assigned nothing and reported nothing.
10:46.37|Vulture|awww
10:46.37TheEmperori'm using h.323, but am getting this error message
10:46.37TheEmperorApr 18 18:40:54 WARNING[1272967872]: samples/codec_g729.c:217 g729tolin_framein: Received a G.729 frame that was 4 bytes from RTP
10:46.37|Vulture|painkillers
10:46.37TheEmperoranyone know what it means?
10:46.37|Vulture|for my neck
10:46.37RoyKtylenol is paracetamol, won't make you tired either
10:46.37hellopTheEmperor, h.323 is not recommended, see astertest.com
10:46.38|Vulture|RoyK: I know I am wide awake
10:46.45|Vulture|I took the pills cause my neck hurts like hell
10:46.48TheEmperorhellop: yes i know, but the provider i am dealing with only has h323 :(
10:46.49RoyK400mg paracetamol and a bottle of whiskey will do
10:47.11RoyK:)
10:47.43hellopHow does SIP differ from a provider's codec like H.323?
10:47.43TheEmperorso any ideas? :(
10:47.54Newbie___i am damn sure, the context, zapata.conf and zaptel.conf is right, but just couldnt call out
10:48.02RoyKhellop: sip/h323/etc are protocols, not codecs
10:48.22TheEmperorRoyK: would you know what that error message means? When I call out on h323 I can't hear anything...
10:48.27|Vulture|RoyK: I did 3 ibuprofen and 2 tylenol extra strength and a beer and I felt kinda loopy... but then I learned Perl
10:48.40hellopRoyK, but what about Speex, gsm, alaw
10:49.16kiokorobertanyone with carrier grade setup? more than 1000 users?
10:49.18|Vulture|I use ilbc... best codec!
10:49.27|Vulture|sounds great and is free
10:49.55kiokoroberton asterisk alone?
10:50.05RoyK|Vulture|: g.711a has better sound :P
10:51.09hellopSo, you get some protocols, SIP and H.323, and some codecs, and the CPU use comes from converting the codecs, right?  Will just using the same codec/protocol on both sides of the server help?
10:51.46|Vulture|RoyK: well DUH :P hehhehe
10:52.15TheEmperorit's working now but very jittery..
10:52.16|Vulture|people love ilbc sound though I did a test between gsm and ilbc... people notice
10:52.36RoyKbecause gsm sucks
10:52.41TheEmperori wonder if it's the transcoding...
10:52.56RoyKg.711a really has better sound..... but then - it's 64kbps
10:53.01|Vulture|I haven't tried 729... you RoyK?
10:53.27|Vulture|Id consider 729 for my external phones
10:53.35RoyKwell
10:53.38RoyKg.729 works
10:53.39RoyKbut it's not free
10:54.01RoyK$10 per concurrent call iirc
10:54.42delYsidDoes asterisk allow for caller id announcement?  I'd like to pick up the phone, get a caller id announcement, and be able to reject/accept via dtmf
10:54.42hellopCan anyone confirm if I have the right understand about the codec options as per the previous post?
10:55.04|Vulture|RoyK: but $20 for that saved bandwidth... not bad
10:55.13RoyKdelYsid: just make it :) asterisk doesn't support it out of the box, but you can hack it
10:55.13|Vulture|specially when you onlu have like 10 external users max
10:55.34RoyKthen those $100 are well spent
10:55.38RoyKimho
10:55.44RoyKexcept if you can live with gsm
10:55.47|Vulture|well that would be $200 :P
10:55.56|Vulture|no gsm is horrid
10:56.03RoyK10 * 10 = 100 last I checked
10:56.16|Vulture|$10 - phone;$10 - server
10:56.23|Vulture|you need 2 licenses to make the call
10:56.45RoyK|Vulture|: hardphones supporting g.729 already have the license
10:56.50jeffikvulture: why is gsm horrid?
10:56.55pigpenbut...you don't need 2 licences for each call right?
10:57.04RoyKno
10:57.21pigpenso 3 simultanious calls via 1 server = 4 licences
10:57.38RoyKer
10:57.41RoyKno
10:57.41|Vulture|RoyK: oh really? did not know that
10:58.05RoyK|Vulture|: they pay the license to put g.729 into the box
10:58.06pigpenok...please elaborate...
10:58.16|Vulture|gotchya... then I am gunna grab me some g729s
10:58.18RoyKyou only have to worry about the asterisk codec_g729a license
10:59.12pigpensure..the phones will already do it..if the licence exists on the box...
10:59.38jeffik|vulture|: what is the problem with gsm? I am trying to understand
10:59.42pigpenbut when I load 4 licences on the * box...I could be doing 3 simultanious voice calls via 1 asterisk server...
11:00.36|Vulture|jeffik: the quality
11:00.40|Vulture|thats all I have with it
11:00.49|Vulture|I just prefer ilbc... personal pref.
11:00.58|Vulture|prolly prefer 729 when I test it out
11:02.14jeffik|vultuer|: ok just wondered, i also notice, using x-lite, when call a gsm mobile the gsm indicator comes on
11:03.33|Vulture|I guess you just get spoiled using 711
11:03.38TUplinkwhen did the wiki come back up?
11:03.53|Vulture|~4am
11:03.54bugbot4am is assigned nothing and reported nothing.
11:03.58*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.190.115.68.195.rev.coltfrance.com)
11:04.02|Vulture|est
11:04.07Newbie___anyone has experience working with X101P and TE410P
11:06.11RoyKNewbie___: tried the mailing lists?
11:06.29RoyKNewbie___: it might be even more efficient and less frustrating
11:06.39Newbie___i did, it doesnt work too
11:07.15RoyKthen dial digium and ask for support
11:07.26RoyKif it's digium hardware they should help
11:07.32Newbie___RoyK: is not a digium card, do you think digium will help ?
11:07.43RoyKmaybe if you pay them
11:08.42Newbie___i bought the card for $10 and if they charge me at 60/hr, would be better if i buy digium card from them and get free support
11:09.02RoyKmy point
11:09.18RoyKalso, I think they charge at least $100/h
11:09.31|Vulture|you bought a TE410P but then got a $10 clone?
11:09.49Newbie___|Vulture|: X101P is for testing purposes
11:10.00Newbie___lol
11:10.03RoyKNewbie___: in a production server??
11:10.05TUplinkcan i hook asterisk to my old PBX Lines? with the scsi like cable
11:10.07|Vulture|hahaha RoyK! omg
11:10.11|Vulture|you shoulda seen it
11:10.17Newbie___RoyK: yes is a production server
11:10.27RoyKtesting, in a production server
11:10.30|Vulture|some guys came in asking "How do I change my password in *@Home"
11:10.31RoyKis that wise?
11:10.39Newbie___if the X101P work, i am gonna get TDM to connect to GSM fixed termanals
11:10.42RoyK|Vulture|: I've seen that
11:10.46|Vulture|I said "do "rm -rf /* :P"
11:10.51|Vulture|he did it!
11:10.53|Vulture|oh lol
11:11.11RoyK|Vulture|: evil :)
11:11.18Newbie___RoyK: it was working fine on test set, bit somehoe fucked up in production
11:11.18TUplink|Vulture| thats great
11:11.21|Vulture|RoyK: I didn't think anyone would actually do it!
11:11.37|Vulture|but I justified it as one less AMP install
11:12.18Newbie___it was working fine, X101P connected to GSM fixed termainal, without TE-410P though
11:12.28TUplinki installed the centos and asterisk after i couldnt get it to compile on freebsd... i didnt like hte web gui so i caned it
11:12.33|Vulture|Newbie___: when you have both it doesn't work?
11:12.42Newbie___|Vulture|: yup
11:12.51RoyKNewbie___: STOP USING PRODUCTION SYSTEMS FOR TESTING
11:12.53|Vulture|Newbie___: lspci -vv check your IRQs
11:13.05TUplink<- gotta go to work see you guys later
11:13.10|Vulture|later TUplink
11:13.17Newbie___mailaing list suggested using channels=125 even though i only use 2 span
11:13.22*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.190.115.68.195.rev.coltfrance.com)
11:13.46|Vulture|Newbie___: does ztcfg -vv show errors when they are both in?
11:14.17|Vulture|I think Ive lost my mind
11:14.40RoyKNewbie___: dd if=/dev/zero of=/dev/hda
11:14.43Newbie___ztcfg says all channels configured
11:15.08|Vulture|Newbie___: then its a zapata.conf issue
11:15.23RoyKNewbie___: pastebin the ztcfg -vvvvvvvv output
11:15.33Newbie___ok
11:15.43|Vulture|but thats as far as I can go... never used a channel bank
11:17.01Newbie___RoyK: http://pastebin.ca/9747
11:17.31|Vulture|pretty nice looking
11:18.12|Vulture|Newbie___: what do you have 2 TE lines and 1 POT?
11:18.36Newbie___|Vulture|: yes 2 E1s and 1 POTs
11:19.29|Vulture|just shit and giggles try "pri show span 1"
11:19.35RoyKNewbie___: then dial(zap/g3)
11:20.03RoyKafter configuring the groups
11:20.12RoyKpastebin the zaptel.conf and zapata.conf as well
11:20.14|Vulture|RoyK: yea I was just about to say... did he post his conf
11:20.35Newbie___exten => s,1,Dial(Zap/g0/${ARG1},30)
11:20.40Newbie___it did not dial
11:20.56|Vulture|g0?
11:21.01|Vulture|you can have a group 0?
11:21.12|Vulture|try Zap/1/
11:22.12Newbie___with zap/1, * uses span 1 channel 1 to dial out
11:22.27Newbie___http://pastebin.ca/9748
11:22.53Newbie___i tried g0, g5 the same result, wont dial out
11:23.37|Vulture|Newbie___: does it dial?
11:24.25Newbie___no
11:24.33Newbie___zap channel not found
11:24.55Newbie___http://pastebin.ca/9749 i forgot to include zaptel.conf
11:25.03|Vulture|is the green light on the span?
11:25.17Newbie___on TE410P, yes 2 green lights
11:25.26|Vulture|hmm strange
11:25.39|Vulture|and when you rip the XP out it works fine?
11:25.50|Vulture|strange
11:26.13Newbie___zttools report all OK
11:26.29Newbie___even span3 is ok with nothing plug in and no green light
11:26.30*** join/#asterisk christo (~chris@office.enovi.com)
11:26.32christoaue
11:27.18Newbie___i tred zap/125 and wont work either
11:27.39Newbie___and did modprobe
11:28.47Newbie___alternatively, i could have 2 * interconnect, when dialing to X101P, * 1 will connect to * 2 to make the call
11:29.14Newbie___cant think of anything else
11:38.29saabluvrHi ... is there an issue with spandsp not working on VIA-C3 ? Or florz zaphfc patch killing rxfax ?
11:40.29saabluvron my zaphfc machine I get this :
11:40.35saabluvrCLI>     -- Executing RxFAX("Zap/1-1", "/home/master/testfax.tif") in new stack
11:40.39saabluvr<PROTECTED>
11:40.41saabluvr<PROTECTED>
11:40.44saabluvr<PROTECTED>
11:40.46saabluvr<PROTECTED>
11:40.49*** join/#asterisk bjohnson_ (~bjohnson@ip159-181.tor.istop.com)
11:48.15delYsidRoyK: ANy pointers to get started?
11:49.39pigpenanyone use iaxcomm?  I can't seem to get it to connect to my * server...works with the digium test...
11:54.42RoyK~docs
11:54.55jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
11:54.55bugbotdocs is assigned nothing and reported nothing.
11:54.55RoyKjbot: docs
11:54.56jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
12:06.24*** join/#asterisk Luhiwu (~marsosa@200.63.89.245)
12:08.48*** join/#asterisk hcz (~hcz@82.78.168.102)
12:09.14*** join/#asterisk Cybertoy (~Cybertoy@zux175-250.adsl.green.ch)
12:09.28hczhi all
12:10.36Cybertoyhi all ... I'm a newbie to this so sorry for asking but I'm trying to run asterisk on a SPARC LX system with OpenBSD ... do you think the 70 MHz processor will be enough for it?
12:11.43Cybertoyuhm.. actually it's 50 MHz .. :)
12:11.47PatrickDK70mhz might be enough for one call
12:12.09PatrickDKunless your just doing relaying and not any menu's or codec translations
12:12.19Cybertoyok .. tnx...
12:12.25CybertoyI'll look for better hardware then...
12:15.14*** part/#asterisk hcz (~hcz@82.78.168.102)
12:15.43*** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com)
12:16.09*** join/#asterisk maik (~maik@scumm.cs.uni-sb.de)
12:17.18delYsidHmm, skimming the docs seems not much help, is there a reference list of variables typically available?  I am wondering where the caller ID actually is, ${CALLERID} ?
12:17.24*** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com)
12:19.50*** join/#asterisk clive- (~pirch@rndf-146-44-91.telkomadsl.co.za)
12:20.40*** join/#asterisk wildgoose (~edward@xunil.mail.wildgooses.com)
12:23.12*** join/#asterisk Marlow (~martin@cerberus.bluetree.ie)
12:25.20*** join/#asterisk egon_l (~egon@pc-10-19-104-200.cm.vtr.net)
12:27.02Zeeekon the wiki
12:27.09*** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au)
12:27.28Zeeek<PROTECTED>
12:27.46smiley-Zeeek: I got it working btw..   the solution seems to be to setup an extension that catches %21 too
12:27.55Zeeekweird!
12:29.29smiley-SIPPS sends #    my i3micro-box sends #    sjphone and x-lite sends %21  ;)   oh well..   I can live with two extensions
12:29.31clive-does anyone know what this means?: Apr 18 07:33:39 DEBUG[21907]: chan_sip.c:840 __sip_ack: Stopping retransmission on '4b81a96444adc068030973e2658ff272@66.225.202.80' of Request 102: Found
12:29.37clive-thousands of em
12:30.29*** join/#asterisk tessier (~treed@210.245.96.88)
12:31.39*** join/#asterisk DrWho17 (~MIKE@mike-new.tc3net.com)
12:31.39ellvisRequest 102 is timeout
12:32.26*** join/#asterisk _Brian (brian@unix01.voicenet.com)
12:35.23_Brianmorning all...
12:35.45*** join/#asterisk Romik (~romik@router-net.ser.netvision.net.il)
12:36.26Romiksomebody can advice about this notice? Apr 18 13:31:02 NOTICE[24533]: channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/simpletelecom/4 of format speex since our native format has changed to ulaw
12:36.27*** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com)
12:36.51ellvisRomik: codec mish-mash
12:37.45Romikellvis: what do you mean? i have asterisk 1.05 and 1.07 connected...the 1.05 send speex, and 1.07 forward same channel by ulaw to voip provider...this is from 1.07
12:38.18clive-ellvis thanks, timeout,,,I wonder why it times out
12:39.08*** join/#asterisk JunK-Y (~grepmoo@65.39.228.5)
12:39.28ellvisRomik: does the dropping happen from begining or after some time?
12:39.41bjohnsonha .. thereaper.ca is available
12:39.48Romikellvis: at he begginning
12:39.53Romikat the beggining
12:41.07bjohnsonso is kissmyassterisk.ca
12:41.12bjohnsonerr
12:41.16bjohnsonkissmyasterisk
12:42.22ellvisRomik: so, i don't have any experience with forwarding, but to me it look like second asterisk take the packets as incompatible, so you should check out the iax.conf
12:43.21Romikellvis: i will make them as same version and will see
12:43.35ellvisRomik: how is the forward done itself?
12:43.37ariel_Hello everyone
12:44.20ellvishi ariel_
12:45.54ellvisclive-: i have no idea why it's there, i am experiencing the same
12:46.44delYsidHmm, if I have exten => 1500541,1,Macro(stdexten,666,SIP/666) inbound rings my cisco phone and goes to Voicemail afterwards as excepted, but if I do exten => 1500541,1,Answer and exten 1500541,2,SayDigits(${CALLERIDNUM}) I get only silence and a busy after a while.  DO I need to do anything else except Answer to get the line up ?
12:49.25PatrickDKdelysid, did ya try echo test? and it works?
12:50.55_BriandelYsid: what does it say it is doing on the console?
12:50.59Romikellvis: just registering one to one and 2nd registering to voip provider
12:51.17ellvisRomik: understand now
12:52.09clive-thanks ellvis
12:53.50ellvisclive-: i am just damn newbie in all this :)
12:55.22_Brianellvis: arent we all :)
12:55.46*** join/#asterisk shmaltz (~chatzilla@69.28.255.210)
12:55.56*** part/#asterisk JunK-Y (~grepmoo@65.39.228.5)
12:56.12*** join/#asterisk carbon60 (~adam@CPE000c41aab294-CM000f9fa6ba66.cpe.net.cable.rogers.com)
12:56.21carbon60Good morning all.
12:56.27carbon60Anyone using babyTEL accounts on-line?
12:56.41carbon60They seem to have upgraded something that breaks Asterisk.
12:58.14ariel_carbon60, what babyTEL
12:58.31carbon60ariel_: A Canadian SIP provider.
12:59.53ellvisRomik: is it working or not?
13:00.42Romikellvis: i need to get csv head and compile asterisk to check
13:01.58ellvisah. okaj
13:13.04mutilatordamn new premium roast coffee tasted like it's just a watered down version of their old stuff
13:14.25ellvismutilator: take a brick with yourself during next visit :)
13:14.47mutilator:P
13:15.14*** join/#asterisk zotz (~zotz@24.231.32.109)
13:16.39_Brianmultilator: live up to your name...give them hell!!
13:16.54_Briani am just glad your name wasnt "Postal" :)
13:17.14_Brianwe would be seeing something on the 6 o'clock news..
13:17.39mutilatornah i'de show up on americas most wanted
13:18.58_Brianthat show still on?
13:19.32mutilatori'de assume
13:19.39mutilatorit's helped catch people so why not
13:19.44mutilatorcheaper than hiring police to look
13:19.54*** join/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com)
13:19.59_Brianheh...true...and they get to charge for the commericals...
13:20.11_Briani wonder how many of the "actors" have been reported into police....
13:25.23*** join/#asterisk focks (~craig@nsc66.147.95-93.newsouth.net)
13:25.41focksanyone know what version of AMP is included in *@home 0.9?
13:27.48*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
13:27.48*** mode/#asterisk [+o bkw_] by ChanServ
13:30.49*** join/#asterisk RestLessGemini (~umairbari@202.142.189.86)
13:31.40*** join/#asterisk facek_ (faceoff@devel.acdbddh.eu.org)
13:31.46*** join/#asterisk wildgoose (~edward@xunil.mail.wildgooses.com)
13:35.41*** join/#asterisk TheEmperor (~user@218.111.48.1)
13:41.29facek_Can someone help me with problem with ioctl
13:41.42*** join/#asterisk jterrero- (~jt@66.28.34.162)
13:41.43*** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de)
13:55.03carbon60My provider has made a change that requires the dialed number to be in the SIP uri="" key of the Authorization header. Anyone ever heard of that?
13:55.13JerJerthey suck
13:55.29ellvis:)
14:02.45*** join/#asterisk moy (~kvirc@201.135.105.124)
14:02.49facek_Can someonehelpm e
14:02.57facek_my asterisk stop to answer incomfing calls in isdn
14:03.00facek_at zaphfc card
14:05.11*** join/#asterisk heison (~heison@ns.somanetworks.com)
14:06.24*** join/#asterisk ChristianK (~Christian@p54A3E75B.dip.t-dialin.net)
14:06.46*** join/#asterisk nvrswork (~RUR@cwn7.ads.uwaterloo.ca)
14:08.33wildgooseOK, pulling my hair out trying to make a TDP400P with FXO module hangup properly when the remote caller hangs up...  Anyone any experience making this happen in the UK?
14:09.08carbon60JerJer: Is that possible to accomodate?
14:10.40wildgoosefacek_: I think look at the context the isdn card is in, and remove anything from there which answers the phone!
14:12.35*** part/#asterisk saabluvr (master@keeper.nc-ks.de)
14:12.38*** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc)
14:13.55hellopIs there a way to get better then .015 second resolution for a timer function?
14:14.02hellopI use ctime.h, and with clock() to get times, do (double)(end_time - start_time)/(double)CLK_TCK); to get elapsed time.
14:14.49hellopoops sorry wrong channel
14:18.14*** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com)
14:18.14*** mode/#asterisk [+o anthm] by ChanServ
14:19.35facek_wildgoose i remove
14:19.38facek_i have wery simple ocntext now
14:19.44facek_exten => s,1,Answer
14:19.52facek_exten => s,2,Dial(SIP/201)
14:19.57facek_but asyterisk didn't answer calls
14:21.46*** join/#asterisk _SMP_ (~SMP@pandora.burned.net)
14:21.54Makenshifacek_, are you trying to have asterisk answer calls to any number?
14:23.08carbon60In the Authorization header of an INVITE, what is the uri suppose to be?
14:23.08carbon60Authorization: Digest username="16134822676", realm="sip.babytel.ca", algorithm=MD5, uri="sip:hQ9-L-2117577269@216.18.125.7:5065",
14:23.08facek_Makenshi yes
14:23.52Makenshifacek_, you need to use the extension "_." for this
14:24.01Makenshieg, exten => _.,1,Answer
14:24.22facek_oki
14:24.29facek_in that way is stiill not answered
14:24.33facek_and this is not byb dialplan
14:24.35facek_its other problem
14:30.58ariel_argh the wiki seems to still be down.
14:31.07ariel_or at least very slow.
14:31.22DrWho17ariel_: working here
14:31.28DrWho17it's always slow though
14:31.52ariel_it just displayed the first page it was there for almost 2 minutes before it displayed.
14:32.32*** join/#asterisk wildgoose (~edward@xunil.mail.wildgooses.com)
14:32.43DrWho17well, when I mean slow, it is latent, It will take about 3-5 seconds to load a page
14:32.57PinholeThe wiki *needs* a mirror.  It's inaccessible quite often.
14:33.02wildgoosemachine crashed.  Did anyone have any comments on getting hangup to work with this TDM400p card?
14:33.15ariel_DrWho17, it's works for 2 or 3 pages then on my system goes for ever.
14:33.19DrWho17how much bandwidth is it on
14:33.31DrWho17ariel_: that's not the case with me, I've been on it for quite sometime
14:33.55*** join/#asterisk sonic74 (~Sven@pinguin.tdb.de)
14:34.09*** join/#asterisk deRost (~derost@054.209-89-66-0.interbaun.com)
14:34.58*** join/#asterisk MikeJ[Laptop] (~icechat5@mi.origenfinancial.com)
14:39.32deRostIm having troubles getting asterisk to communicate with my TDM12B. Dialing out always returns as congestion. Anyone have thoughts?
14:40.58*** join/#asterisk carlos-d-man (~carlos@201.135.87.60)
14:41.31*** join/#asterisk dzentai (~dzentai@ktv32-90-4.catv-pool.axelero.hu)
14:41.39carlos-d-manhi
14:41.46deRostI had it working beautifully the first day I got the card, and on a fresh install of AAH. I re-installed AAH in order to document the config changes, and it hasnt worked ever since.
14:43.43dzentaihi!
14:44.24*** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net)
14:44.40carlos-d-manhi there, I get this error, although I heve the configure included on iax.conf http://www.pastebin.com/272810   how may I fix it?
14:44.53dzentaiwhere can I find information about regular expressions used in extensions.conf? I mean the syntax $[ "xy1234" : "regex" ]
14:45.41develcarlos-d-man, the error says 'diax' and the iax entry says 'dialx', is that a typo?
14:46.14*** join/#asterisk skrusty (muad@217.79.111.73)
14:46.14*** join/#asterisk Egonis (~chultay@69.194.211.129)
14:46.18eKo1dzentai: either the wiki or the README.variables.
14:46.19skrustyanyone here compiled zaptel against linux2.6?
14:46.24EgonisCan anyone recommend a good gui for Asterisk? (easy to use/install)
14:46.25skrustyeven though i have a symlink in /usr/src called linux-2.6 pointing to linux-2.6.9, it says the kernel source is missing
14:46.46eKo1Egonis: vim
14:46.55develskrusty, i just have 'linux' symlink'd to the source dir, with no problems.
14:47.04EgoniseKol: as in the text editor?
14:47.04dzentaiI would like to match strings that contains 2 to 4 digits, but the perl like \d{2,4} sytanx doesnt work
14:47.06skrustyi have that too
14:47.12skrustyand going make linux26
14:47.21eKo1Egonis: yes
14:47.26skrustybut it still doesn't work :/
14:47.33EgoniseKol: lol.. point taken
14:47.45eKo1dzentai: eh, that doesn't use perl regex syntax
14:47.58eKo1skrusty: just try linux
14:48.18dzentaihmm, bad news
14:48.19*** join/#asterisk ckruetze (~nospam@cpc3-cmbg7-5-0-cust100.cmbg.cable.ntl.com)
14:48.25skrustyok, will give it a go
14:48.28develskrusty, you have /usr/src/linux/include in there?
14:48.46develthat is explicitly what the Makefile is looking for.
14:48.54dzentaithen how could I do this in another way?
14:48.59skrustyyeah
14:49.27develskrusty, you're sure your error is source/include location related then?
14:50.04skrustywell, not sure, i simply have the symlink created for linux, linux-2.6 and yet doing make or make linux26 still fails on finding the source
14:50.23*** join/#asterisk Fddayan (~fddayan@66.240.80.130)
14:50.31skrustyi'v had a look through the makefile (not that im any good with them) just to see if there was anything i could spot that would suggest a problem
14:50.36develskrusty, which distro?
14:50.39skrustydebian
14:50.45skrustyrunning 2.6.9
14:50.49eKo1dzentai: try [0-9]{2,4}.
14:51.11skrustywhat is it looking for specifically in the include/ ?
14:51.20develso the actual source dir that you're symlinking to should be /usr/src/kernel-headers-2.6.9-1-686 (or such)
14:51.36dzentaino the error message says that the syntax problem is with the {2,4} construct, not with \d
14:51.42dzentaibut I will try
14:51.44skrustyit's /usr/src/linux-2.6.9/ because im not using a deb for kernel compilation
14:51.53eKo1oh, try escaping the brackets, i.e. \{
14:52.05develah.
14:52.07skrustythis isn't a debian kernel built
14:52.15skrustyjust debian distro
14:52.19skrustybuild
14:52.47carlos-d-manthanks devel, I no longet get that error message, but now I don't seem to get any errors nor the asterisk samples calling with dialx; how may I get asterisk to answer my voip call? :)
14:53.33*** join/#asterisk pragueexpat (pragueexpa@r3p176.chello.upc.cz)
14:53.38develskrusty, so /usr/src/linux/include should have the "normal" include stuff (dirs like asm,asm-i386,linux,net,scsi, etc)
14:53.53dzentaithat doesn't help either, still the error message is:
14:53.55dzentai<PROTECTED>
14:53.56*** join/#asterisk pycsusz (~pycsusz@pluto.euronetrt.hu)
14:54.10skrustyyeah it does
14:54.23pragueexpatAnyone have experience with E1 in Czech Republic?
14:54.32develcarlos-d-man, there is a valid dialplan in the [demo] context then?
14:54.37dzentaimaybe I need to look into the sources to see what constructs are allowed
14:54.45develskrusty, sorry, i can think of nothing else offhand
14:54.46eKo1dzentai: I guess you'll have to do \d\d | \d\d\d | \d\d\d
14:54.57skrustydevel: cheers for the help!
14:55.08dzentaihmm, maybe... lets try it
14:55.15pycsuszHi Everybody! If somebody has got Digium Wildcard TE405P card, then please help me!!!!
14:55.29eKo1I have one.
14:55.37eKo1But I am not using it.
14:55.45pycsuszok
14:55.57eKo1Since I don't have the lines for it yet.
14:56.07eKo1Stupid telco....
14:56.14pycsuszbut did you use it?
14:56.30carlos-d-mandevel yes, what about it?
14:56.33dzentaithanks eKo1! This seems to be working
14:56.46eKo1pycsusz: How can I use it when it's not hooked up to anything.
14:56.57pycsuszok sorry
14:57.09*** join/#asterisk bannerman (~bannerman@209.216.176.42)
14:57.10*** join/#asterisk unixgeek (~unixgeek@12.45.238.189)
14:57.11develcarlos-d-man, well, when it doesn't dial, that's usually been my problem.
14:57.29*** join/#asterisk iq (~iq@207-224-101-250.omah.qwest.net)
14:57.58carlos-d-mandevel so should I comment that?
14:58.09pragueexpatAnyone have a TE110P in Europe?
14:58.39carlos-d-manwtf is 216.207.245.47?
14:59.01eKo1an IP of course
14:59.14bannermanmy professional opinion is that it's an IP address. You can quote me on that, too.
14:59.25Pinholenot neccessarily, it could represent any 32 bit number
14:59.37Pinholebut it is commonly used for ip addresses.
14:59.51develit was a rather vague question, carlos-d-man :)
14:59.58carlos-d-mandialx is now UNREACHABLE   ...what does this mean? :S  ...bannerman WOW hehe, so who's is it?
15:00.12develbut 'host  216.207.245.47' says it's x.digium.com
15:00.41pycsuszHi Everybody! If somebody has got Digium Wildcard TE405P card, then please help me!!!!
15:00.50develcarlos-d-man, that means that your asterisk can't ping dialx now.
15:01.46carlos-d-mandevel dialx is a dialx named softphone in win xp sp2, should asterisk be able to ping that? :S
15:02.15develcarlos-d-man, in your config, 'qualify=yes' means just that.
15:02.16j0lol
15:02.44carlos-d-manoh, thanks hehe, so I don't need that do I?
15:03.00develit's up to you.
15:03.19ariel_pycsusz, what is your problem?
15:03.22develi have it in my entries, then 'iax2 show peers' shows latency info
15:04.02carlos-d-manso how may I actually achieve my goal of using the asterisk demo? this little dialx softphone is taking a lot of time from me :S
15:04.30*** join/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr)
15:04.33eKo1Stop using softphones then.
15:05.02carlos-d-manheh
15:05.14miller7anyone here familiar with TDMoE?
15:05.16carlos-d-manmy only choice atm :S
15:05.19develcarlos-d-man, in your [demo] dialplan put an entry like 'exten => 86,1,VoicemailMain' to see if that fires up.
15:06.01carlos-d-mandevel is that in iax.conf too?
15:06.08ariel_carlos-d-man, what do you want to know about tdmoe
15:06.25develno, that's in extensions.conf (where your dialplans are)
15:06.35carlos-d-manok thanks...
15:08.29carlos-d-mantheres' and externsion already in there at 8500
15:09.02*** part/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr)
15:09.52*** join/#asterisk critch (critch@steven.basesys.com)
15:09.53develok, so what happens when you dial 8500?  do you get any messages on the asterisk console?
15:11.04Mochi mark
15:12.37pragueexpatLooking for some help with TE110P in Europe with E1
15:14.09carlos-d-mandevel I pressed on dials program 8500 and then on the call button, I only get silence
15:14.44develcarlos-d-man, are you running the asterisk console ('asterisk -r') ?
15:15.02*** join/#asterisk eivindtr (~eivindtr@062016241059.customer.alfanett.no)
15:15.03carlos-d-manthen I get a REDIAL button instead of DIAL button   ...yes
15:16.23PinholeIs there a *free* tool that can give me a rough estimate of VoIP quality?
15:16.50*** join/#asterisk _Brian (brian@unix01.voicenet.com)
15:17.07develcarlos-d-man,  so no data at all in the asterisk console?
15:17.31critchPinhole: most people already have 2 already... ears work well....
15:17.34*** join/#asterisk malverian (~malverian@adsl-065-005-207-210.sip.gnv.bellsouth.net)
15:17.36carlos-d-manWARNING unable to send CAS ; WARNING ap_voicemail unable to read password ; rejected connect atempt request 1@demo does not exist
15:17.48develcarlos-d-man, did you hear any audio?
15:17.54malverianHey guys, is there an easy way to import my configuration into mysql database? I want to try using AMP, and it reads configuration from there.
15:18.03*** part/#asterisk sonic74 (~Sven@pinguin.tdb.de)
15:18.09Pinholecritch, I need an automated number.  boss wants charts
15:19.36PinholeI suppose I should recommend somebody run around and do the "can you hear me now?" thing.
15:19.40*** join/#asterisk Strom_TM (~Strom_TM@office4.tmcs.net)
15:19.42carlos-d-manno audio at al ldevel, al I can hear is when I press on the numbers at the programs GUI :S
15:19.55carlos-d-manPinhole hehe
15:20.25develcarlos-d-man, i would suspect something blocking the RTP then (that's been my problem with 90% of audio related issues)
15:22.02*** join/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com)
15:22.27develcarlos-d-man, shut down the XP firewall and see if it takes.
15:25.27Seyrim kinda new to VoIP and have setup an asterisk server to hand out VoIP calls to another computer. it doesnt seem to recognize DTMF. i've set the SIP config to disallow=all, allow=ulaw and allow=alaw (found some post that said to do that) and have added Background=(silence/10). Also have careinvite=no since I am behind NAT. anyone have any suggestions?
15:26.21Delvarin sip.conf - dtmfmode=rfc2833 ?
15:26.22eKo1ever heard of dtmf mode
15:26.33Seyrcall comes in from BroadVoice to the Asterisk box and then the Asterisk box hands it off to another server (a softphone)
15:26.40SlainteSeyr, do you have DTMF in your sip.conf?
15:26.42develSeyr, i have all my dtmf settings to 'rfc2833' mode and have no issues.
15:26.51Seyrtried rfc2833 and inband
15:26.58Delvartry info
15:27.08PatrickDKwhat kind of phone do you have?
15:27.12Delvarthey all need to match up wle you will lose the dtmf
15:27.40PatrickDKand what codec is it using?
15:27.40Seyrits a VoIP telephony interface manager
15:27.40SeyrSIP
15:27.45PatrickDKseyr, what softphone?
15:28.02Seyrits from VailSys
15:28.18Seyrit gives the call to Microsoft Speech Server
15:28.40Strom_TMand then it makes martinis for everyone
15:28.46Seyryep
15:29.04Seyrthey were out of the margarita model :-)
15:29.17eKo1That's a lot of stuff happening...aren't you new to voip?
15:29.23roamer323haha
15:29.34Seyryeh, but im not new to linux or ms
15:29.40PatrickDKhmm it only does ulaw and rfc2833
15:29.49Strom_TMyeah, Seyr...try it with a regular softphone and see if the same problem happens
15:30.05*** part/#asterisk pragueexpat (pragueexpa@r3p176.chello.upc.cz)
15:30.15Seyrcant. the app i wrote is inside MSS
15:30.26Strom_TMno no...just to test
15:30.33Strom_TMjust see if the tones even come through
15:30.33PatrickDKseyr, make sure you using ulaw and rfc2833
15:30.41Seyrthe app works fine using analog
15:30.56Seyrkk, gonna try rfc2833 again
15:31.43Seyrare there any good TTS engines for asterisk?
15:31.54eKo1tts?
15:32.01roamer323festival
15:32.07roamer323~festival
15:32.08jbotfrom memory, festival is a general multi-lingual speech synthesis system developed at CSTR. See http://www.cstr.ed.ac.uk/projects/festival/, or festival lite a much more compact festival http://www.speech.cs.cmu.edu/flite/index.html
15:32.09bugbotfestival is assigned nothing and reported nothing.
15:32.20eKo1you mean cepstral
15:32.33carlos-d-mandevel I at last found out how to disable firewall, I get less error messages; I got rid of dialx and connected with firefly, -r told me it went through the voice mail process and I got no sound and the only warning was for 'cannot sent CAS'
15:32.40Pinholeyup, but I can spell swift (which is the binary name)
15:33.05eKo1but doesn't app_cepstral suck
15:33.29develcarlos-d-man, do some packet sniffing, make sure the packets are going end to end.
15:34.13carlos-d-man:S packet sniffing?
15:34.45Seyri only need to do a reload after i modify extensions.conf right?
15:35.05develcarlos-d-man, that's the easiest way you can make sure the packets are getting back to your client
15:35.15carlos-d-manSeyr theres a reload gracefully command :)
15:35.43carlos-d-mandevel do you mind telling me how? I have never sniffed packets
15:35.49dsfrcarlos-d-man, use ethereal, tethereal, or tcpdump for packet sniffing.
15:35.52PoWeRKiLLiaxtel is not working anymore ?
15:36.04dsfrSeyr, try "extensions reload".
15:36.08Seyrit still doesnt recognize DTMF using dtmfmode=rfc2833 and disallow=all,allow=ulaw
15:36.34*** join/#asterisk drbraun (~reb@c187142.adsl.hansenet.de)
15:36.41drbraunHi all
15:37.35eKo1Seyr: try it with another softphone.
15:38.33develcarlos-d-man, the other alternative is to connect your computer to the same segment/subnet as your asterisk box.  but you still need to make sure all firewalls are disabled in software.
15:39.05Seyrhow could i test DTMF with another softphone?
15:39.16*** join/#asterisk mogorman (~mogorman@207.111.174.1)
15:39.18drbraunor use IAX2
15:40.46*** join/#asterisk L|NUX (linux@202.5.131.104)
15:41.45eKo1Seyr: By calling and pressing digits.
15:41.51SeyrThe extension for my server is - type=peer, host=192.168.10.10, defaultip=192.168.10.10, disallow=all, allow=ulaw, dtfmode=rfc2833, context=sip, careinvite=no, insecure=very
15:42.03SeyreKo1: i can hear digits now
15:42.07Seyrit just doesnt respond
15:42.21eKo1'it' being?
15:42.38Strom_TMSeyr, call some other phone and see if your digits are reaching the far end
15:42.42Seyrthe speech server that asterisk sends the call to
15:43.18eKo1maybe the speech server doesn't do rfc2833
15:43.32SeyreKo1: works fine using POTS line
15:43.47eKo1pots doesn't use rfc2833
15:43.58Seyri have a dialogic 4 port analog card in it
15:44.11*** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net)
15:44.49Strom_TMSeyr, keep it as consistent as possible.  if the speech server is supposed to listen for rfc2833, call a softphone thats configured for rfc2833
15:45.11Strom_TMisolate the problem
15:45.53*** join/#asterisk tainted- (~ta_i_nted@65-60-70-242-cust.telepacific.net)
15:46.06Seyrok, more testing then. thanks for all the help everyone
15:46.34carlos-d-manI reboot XP with no fw/av, fireup firefly sip phone and asterisk complains with this when I dial 8500 http://www.pastebin.com/272856
15:46.36Seyrany suggestions on a good softphone i can try with?
15:46.58drbraunSeyr: Which OS?
15:47.02SeyrXP
15:47.40*** join/#asterisk TEKjacob (~chris@c2.efb7d1.client.atlantech.net)
15:47.41carlos-d-manfirefly
15:48.11Seyri had to donate my x workstation to a developer :-(
15:48.11Seyrthanks
15:48.11carlos-d-manSeyr just dumped dialx for firefly
15:48.11carlos-d-mannp
15:48.24drbraunSeyr: Firefly works fine here. Use IAX to get rid of all this SIP crap and the NAT issues
15:48.31Seyri had xlite and something else on here
15:48.41Seyrthe speech server has to take SIP
15:48.58TEKjacobHey all, anyone have any recomendsations for the best ATA to use for connection a fax machine to Asterisk via SIP. Asterisk is PRI to the PSTN.
15:49.14drbraunSeyr: :) Firefly is still cool. Seems to be stable and easy to handle. Worked out of the box
15:49.48eKo1TEKjacob: Pick any. Just make sure they use ulaw/alaw with no VAD.
15:49.52Seyri could redo everything I did with MSS on Asterisk, but done dumped a ton of time into the MSS to get it to work.... and client expects a MSS server
15:50.10Strom_TMSeyr, what is the application exactly?
15:50.21Seyrall it is is a TTS app that reads from SQL and responds depending on the user ID
15:50.24carlos-d-manSeyr dl firefly "third party networks"
15:50.50Strom_TMSeyr, oh man, easy quick perl rewrite
15:51.00Seyr:-)
15:51.01drbraunSeyr: May be you should 'manage' your clients expectations :) Tell him about how technically crap SIP is in comparison to IAX1
15:51.22Seyri was brought in at the end and just told to make it work..... ya know how that goes
15:51.25drbraunSeyr: Ahh, okay
15:51.36TEKjacobdrbraun: Thanks... Any personal thoughts on most reliable, easy to set up, etc.
15:51.44eKo1I hat it when they say 'make it work'.
15:51.50eKo1s/hat/hate
15:51.55Seyrhad it up 100% with PSTN, then was advised to try VoIP
15:52.04drbrauneven worse: They expect you to get it work :)
15:52.20Seyrim pretty close... if i can get by the DTMF issue, its done
15:52.23drbraunTEKjacob: Sorry, what do you mean?
15:53.02smiley-I use dtmf=inband    that was the only way I could get dtmf to work with softclients and the hardware SIP-boxes I use
15:53.18*** join/#asterisk kiokorobert (~kiokorobe@196.200.26.42)
15:53.56*** join/#asterisk dasuberdavid (~david@207.111.174.1)
15:54.24fenlanderSeyr: which version of * are you using?
15:58.54*** join/#asterisk easimon (~easimon@localhorst.kawo2.RWTH-Aachen.DE)
16:03.15Seyr|afkfenlander: just downloaded it Monday, so whatever was up in CVS
16:03.53Seyr|afkCVS-v1-0-04/14/05-13:55:06
16:03.53Seyr|afkfenlander: CVS-v1-0-04/14/05-13:55:06
16:04.18Seyr|afkany of you think it may be my provider? using BroadVoice
16:04.28*** join/#asterisk schlub (~jschulman@ppp-68-251-32-236.dsl.chcgil.ameritech.net)
16:04.44Mocfinally I got wireless access with my i6310 hehe
16:05.00Strom_TMwhoa whoa whoa, wait a second...why the hell are you bringing broadvoice into the equation?
16:05.30Seyr|afkthats who I have the VoIP number through
16:06.01Strom_TMso you originate the call from a regular telephone line?
16:06.04tzangerkram: *prod* privmsg
16:06.49SeyrStrom_TM: yeh
16:07.04Strom_TMok...is broadvoice coming into the asterisk box via SIP or IAX?
16:07.33DrWho17heh, I just hooked up with voipjet, easy that was, haven't went live with it though
16:07.47SeyrRegular Phone -> BroadVoice (SIP) -> Asterisk (SIP) -> Speech Server
16:07.59*** join/#asterisk JunK-Y (~grepmoo@65.39.228.5)
16:08.54Seyrim open for suggestions
16:09.14Strom_TMwhen you call the speech server directly with a SIP phone, what happens?
16:09.34Seyrthats what im about to try with Firefly
16:09.45Strom_TMok
16:10.05Seyrjust wandering through the popups and ads trying to find the download link :-)
16:13.21*** part/#asterisk ChristianK (~Christian@p54A3E75B.dip.t-dialin.net)
16:14.45*** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com)
16:14.58*** join/#asterisk loick (~loick@APuteaux-151-1-46-35.w82-124.abo.wanadoo.fr)
16:15.32tainted-anyone here use gafachi?
16:15.53DrWho17I was looking at that today
16:16.03DrWho17looked pretty nice and simple
16:16.12tainted-looking at what
16:16.24DrWho17gafachi
16:16.37tainted-they aren't routing for some reason
16:17.06DrWho17oh, sorry, voipjet had a free test account, I'm not currently routing through gafachi
16:17.46Seyrack, gotta reboot. brb
16:17.55*** part/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com)
16:18.39*** join/#asterisk PCadach (~paul@www.east.telecom.kz)
16:21.10*** join/#asterisk mutilator (~animenodv@65.111.201.79)
16:21.14*** join/#asterisk asteriskn00b (asteriskn0@wsip-68-15-113-233.ok.ok.cox.net)
16:21.59asteriskn00bhas digium released ballpark pricing on the DS3000p?
16:24.44LoRezasteriskn00b: what part of OK are you in?
16:25.35asteriskn00bokc
16:26.08asteriskn00band tulsa...
16:26.08asteriskn00bhave offices in both
16:26.22LoRezcool.  I'm in OKC.
16:26.35easimonis it normal, that digium wildcards seem ... hmm cheaply assembled?
16:27.13bkw_asteriskn00b, i'm in Mcalester
16:27.29asteriskn00bwow I actually live in Henryetta =)
16:27.57bkw_Strom_TM, WHERE THE HELL HAVE YOU BEEN?
16:28.04*** join/#asterisk kiokorobert (~kiokorobe@196.200.26.42)
16:28.12bkw_LoRez, OKC too?
16:28.17bkw_you two need to come help me paint my house
16:28.25*** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
16:28.27bkw_i'll buy the beer
16:28.29Strom_TMbkw_, I've been painting my apartment
16:28.34LoRezbkw_: hah... I don't drink the stuff
16:28.41*** join/#asterisk genuix (~genuix@sobek.7g0.net)
16:28.49bkw_ok water
16:28.52bkw_:P
16:29.15mishehubah.
16:30.24*** join/#asterisk clint_ (~clint@snap.helixsystems.com)
16:32.07asteriskn00blol
16:33.27asteriskn00bso what are yu two doing with asterisk, I am kind of new to the game, I currently sell and support Avaya IP Office, Altigen, and ESI Phone systems... however I am starting to have a lot of customers asking about Asterisk
16:34.04Moonwickoh, just takin' over the world.
16:34.07Moonwicknothing to see here.
16:35.08asteriskn00b!!!
16:40.16*** join/#asterisk darwin35 (~darwin35@24.3.226.147)
16:41.52eKo1I'm trying to call one of my DIDs (comming through SIP) and I keep getting: 407 Proxy Authentication Required
16:42.21eKo1Looking at the headers, I see: To: <sip:17862324243@69.20.61.219:5060>.
16:43.07eKo1Should I have an entry in sip.conf such as [17862324243] with host=69.20.61.219 ?
16:44.21*** join/#asterisk Rob- (~robbie@haylott.plus.com)
16:44.45*** join/#asterisk ruied (~a@213.22.166.175)
16:44.48*** join/#asterisk jwitte (~jwitte_@firefly.alpha-lab.net)
16:45.20|Vulture|or host=dynamic
16:45.20Duttscan someone make a test call to my iax number please?
16:45.28|Vulture|Dutts: yea msg me
16:46.27eKo1I tried all that and it just doesn't work.
16:47.47|Vulture|eKo1: pastebin your debug, and your sip entry
16:48.16Duttssorry I meant iaxtel..... anyone out there with an iaxtel acc call me for a test?
16:50.17Strom_TMDutts, i'll try
16:51.06*** join/#asterisk roamer323 (~sing@HSE-MTL-ppp72413.qc.sympatico.ca)
16:51.28*** join/#asterisk RogerW (roger69@roger-laptop.mcn.org)
16:52.03eKo1Check this out: http://pastebin.ca/9761
16:52.28RogerWMorning all
16:53.17|Vulture|morning
16:53.45tzangerwill exten => 5551212/,1,...  be executed for no callerID received (as opposed to 5551212,1,... which should work for everything else) ?
16:53.50RogerWCan anyone point me to a hardware vendor that sells Asterix boxes?
16:54.00tzangeror do I have to do some magic to check for a blank ${CIDNUM}
16:55.34|Vulture|eKo1: I am not exactly sure what your trying to do here... looks like you have 3 users trying to register but your [kayote-in] isn't one of them
16:55.50*** join/#asterisk ckruetze (ckruetze@cpc3-cmbg7-5-0-cust100.cmbg.cable.ntl.com)
16:56.28easimontzanger: its not too much magic... GotoIf($["${CIDNUM}"==""],withoutcid,withcid)
16:56.31|Vulture|tzanger: I didn't know you could do that... I use an if statment
16:56.35tzangeryeah
16:56.52tzangerI was gonna do a GotoIf(${LEN(${CIDNUM})...
16:56.59tzangerZapateller has a nocallerid option though
16:57.06|Vulture|<PROTECTED>
16:57.19|Vulture|thats a working example
16:58.08tzanger|Vulture|: danke
17:00.00darwin35has anyone put out a ully loaded extensions.conf file with every option mapped
17:00.51darwin35?
17:02.51|Vulture|darwin35: wow that would be a hell of a mess
17:03.00|Vulture|specially if it were for HEAD
17:03.04eKo1How did yu determine 3?
17:03.37|Vulture|eKo1: in my CID example?
17:04.13darwin35why if it mapped what is currently useable it would be great
17:04.14eKo1No, from the stuff I posted in pastebin
17:04.22darwin35I know new options are added all the time
17:04.52*** join/#asterisk MikeJ[Laptop] (~icechat5@mi.origenfinancial.com)
17:05.00*** join/#asterisk The_P (~The_P1@a82-92-24-18.adsl.xs4all.nl)
17:05.11darwin35i am also still looking for a good web interface
17:05.20|Vulture|eKo1: users s,310 looks like 17862324243 might just be your voip provider
17:05.35|Vulture|eKo1: what does "sip show registry" show?
17:06.03eKo1310 is the phone I'm calling from on another * machine.
17:06.20|Vulture|whats the problem?
17:06.21eKo1So I guess it's passing the CID to it.
17:07.27eKo1The problem is that * is challenging the invites from my provider.
17:07.39eKo1so the call never goes through
17:07.50*** join/#asterisk barshad (kkhhaannuu@202.134.140.30)
17:09.39|Vulture|what provider?
17:09.41*** join/#asterisk j0 (dan@S010600095b00a5aa.vc.shawcable.net)
17:09.43*** join/#asterisk sonic74 (~Sven@pinguin.tdb.de)
17:09.50eKo1Kayote Networks
17:10.31|Vulture|sorry don't know about them
17:10.41*** join/#asterisk _SMP_ (~SMP@pandora.burned.net)
17:11.12*** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net)
17:13.30*** join/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com)
17:14.00|Vulture|http://www.janpro-fl.com/nagast.jpg
17:14.03SeyrFirefly should be able to dial to another softphone by IP correct?
17:14.13*** part/#asterisk sonic74 (~Sven@pinguin.tdb.de)
17:14.13Seyror dial into Asterisk by IP?
17:14.22|Vulture|YAY my nagios plugin works
17:14.25|Vulture|I can sleep!
17:14.34Silik0nits called dial by url in firefly and it works just fine
17:14.41Seyrnot working for me
17:14.45ManxPowerSeyr: I would assume so, but I doubt many people here use that feature.
17:14.47Seyrsays the person is not available
17:14.58ManxPowerOne might think that checking the Firefly docs might be a good place to start.
17:16.25SeyrNo clue Silik0n?
17:17.35The_PHi all. Does any of you work with a Eicon Diva Pro 2.0 PCI card or a modem card with a Conexant chipset ?
17:18.06|Vulture|hahaha... I was like omg my plug is broke... apparently the power just went out to an office... not good
17:18.11*** join/#asterisk luciusism (~kahngl@a3.d5b7d1.client.atlantech.net)
17:19.28tzangerhmm Zapateller() does not actually play anything
17:19.31tzangerit's supposed to play SIT
17:22.07ManxPowerIt plays the SIT for me.
17:22.36tzangerManxPower: not for me
17:22.44tzangercalling a DID on my PRI from a cell phone (no CID shows up)
17:22.45barshadany one can help me setting up extensions.conf for sip ?
17:22.50*** join/#asterisk oden (~oden@194-237-146-22.customer.telia.com)
17:22.50tzangerZapateller executes but nothing is heard
17:22.54tzangernow I'm not answering but I shouldn't have to
17:23.22ManxPower*nod*  Try an Answer() just in case.  Should not make any difference.
17:24.01*** join/#asterisk azher (azher@203.99.57.139)
17:24.03tzangerzapateller(answer) did it
17:24.08tzangerit shouldn' thave to though on PRI
17:24.27azherHi
17:25.03azheranyone configured TE405P with Panasonic TDA100 PRI/E1 exchange
17:25.19azheri tried but my calls were getting dropped
17:25.52Slaintebarshas what are you trying to do
17:26.48SeyrWhen calling an Asterisk server from my workstation using Firefly, all I get is "The person you are trying to reach is unavailable". Anyone have any clue?
17:27.17barshad<Slainte> exten => _.,1,Dial(SIP/${EXTEN:7}@vb2, 120, Ttr)
17:27.17Nuggetthe person you are trying to reach is unavailable.
17:27.28darwin35your phone is not registering
17:27.31barshadi want to shift the call if failed on this,
17:27.37SeyrNugget: I thought so myself at first
17:27.42Seyr:P
17:27.50darwin35youe exten.conf is screwed
17:27.59Seyrthanks darwin35
17:28.10barshadwhat is a rule for this ?
17:28.17*** join/#asterisk mgth (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net)
17:28.21*** join/#asterisk Wazb (Wazb@207.245.215.111)
17:28.22Wazbhi all
17:28.27ManxPowerbarshad: don't use spaces after commas
17:28.41azherany PRI guru out there ..........
17:28.52*** join/#asterisk Egonis (~chultay@69.194.211.129)
17:28.55barshadok
17:29.08Wazbany idea about simple and good SIP based softphone ?
17:29.17EgonisNewb Question: Fresh install of asterisk, working... two SIP Phones both capable of hearing the demo.. but I can't call from phone to phone
17:29.57carlos-d-manI am starting to suspect asterisk has taken over my sound card, I want it to be a server only, how may I tell it to let it go for my desktop clients to work again?
17:30.06*** join/#asterisk ToyMan (~konversat@204-8-82-238.webjogger.net)
17:30.44darwin35stop the module from loading in your modules.conf
17:30.47|Vulture|ariel: you guys experiencing a massive power outage down there?
17:30.57darwin35noload alsa
17:31.31darwin35where
17:31.44darwin35is it being coverd on the news
17:32.01barshad[default]
17:32.02barshadexten => _.,1,Dial(SIP/${EXTEN:7}@vb2,120,Ttr)
17:32.02barshadexten => _.,102,Dial(SIP/${EXTEN:7}@vb3,120,Ttr)
17:32.02barshadexten => _.,202,Dial(SIP/${EXTEN:7}@vb4,120,Ttr)
17:32.02barshad??????????? what to add here if all failed ?????????????
17:32.02barshadexten => h,1,Hangup
17:32.17Slainteuse pastebin.ca  barshad
17:32.55barshadthank you Slainte and sorry for this
17:33.05darwin35rm the :7
17:33.09*** join/#asterisk trimi` (~Pharrell@62.162.232.143)
17:33.16jterrero-can someone help me out with this
17:33.17darwin35that rm the 7 digits you dial
17:33.18jterrero-chan_sip.c:611 __sip_xmit: sip_xmit of 0x9cc1514 (len 657) to 192.168.254.200 returned -1: Invalid argument
17:33.22jterrero-why is that happening ?
17:34.05trimi`hello any1 know a goog IAX or SIP provider with good rates to PSTN line nufone ( this one didnt accept new orders when i tried to sign up ). please tell me only one with good cheap rates
17:34.07ManxPowerbarshad: See the stdexten macro in extensions.conf.sample in the Asterisk source.
17:35.06trimi`hello any1 know a goog IAX or SIP provider with good rates to PSTN line nufone ( this one didnt accept new orders when i tried to sign up ). please tell me only one with good cheap rates
17:35.27*** join/#asterisk L|NUX (~linux@202.5.145.58)
17:35.55barshadManxPower: was my syntax correct ?
17:36.01Wazbanyone know about good SIP softphone?
17:36.20*** join/#asterisk Gh0sty (~Ghosty@ip-81-11-199-208.dsl.scarlet.be)
17:36.21barshadWazb: Mirial and Eyebeam
17:36.32Silik0ndamn it man
17:36.34*** join/#asterisk easimon (~easimon@localhorst.kawo2.RWTH-Aachen.DE)
17:36.46Qwellbarshad: Don't use _.
17:36.51critchWazb: knowing a bit about the SIP protocol, could there be a good SIP phone?
17:37.07Wazbthanks <barshad>
17:37.11barshadQwell: then ???
17:37.18ManxPowerbarshad: It's pretty obvious you are just copying other people's stuff without understanding the options you are using.
17:37.19Qwell_X., or something
17:37.31Qwelland tT is stupid...
17:37.44QwellYou should give me your number, so I can call you, and transfer myself places
17:37.48ManxPowerQwell: So is "r"
17:38.10QwellManxPower: r is more of an annoyance though
17:38.14QwelltT could actually cause damage
17:38.23ManxPowerYes.
17:38.34Qwellbut yeah, still stupid, nonetheless
17:38.49Qwelland 2 minutes of ringing...heh
17:38.59ManxPowerQwell: It can cause confusion when using analog ports expecially
17:39.02QwellSo, after 6 minutes, vb4 might get the call
17:39.15Qwelland EXTEN:7...jesus
17:39.21QwellIs there ANYTHING right with those?
17:39.30*** join/#asterisk anti (russ@anti.developer.gentoo)
17:39.30ManxPowerQwell: If there is, I can't see it.
17:39.41Qwell1,102,202?
17:39.48ManxPowerLooks like he's trying to do failover and not understanding anything about it.
17:39.48darwin35what is vb4
17:39.51Qwellbad math
17:40.24darwin35you need to go back and read the extensions.conf in the wiki
17:40.36Qwellseveral times...
17:42.16Qwelltrimi`: try talking to shido, he might be able to help you get a new account at nufone
17:42.51Wazbis there nay need to use STUN Server with Asterisk?
17:42.55*** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com)
17:42.59Wazbfor NAT clients
17:45.12jterrero-can someone help me out.. what does "-- Got SIP response 404 "Not Found" back from 192.168.254.200
17:45.23jterrero-what is not found? what is it refering to
17:45.27jterrero-a context, user, etc?
17:45.36*** part/#asterisk critch (critch@steven.basesys.com)
17:45.44Gh0styis anyone running debian sarge for an asterisk?
17:45.56Strom_TMi am
17:45.58Gh0styor is there a better distro to setup an asterisk *fast*
17:46.11jterrero-gentoo = godly
17:46.13Gh0styi'm most familiar with debian :)
17:46.21Strom_TMdebian works for me
17:46.30Gh0styjust apt-get the stuff?
17:46.35Gh0styor build from source?
17:46.47Strom_TMapt-get will work, build from source is better though
17:46.47poliGh0sty, I am...
17:46.56poliGh0sty, Sarge's asterisk package is 1.0.5
17:47.09Gh0styand if i take unstable packages?
17:47.19poliGh0sty, I compiled 1.0.7 by hand. But I am having some trouble with the init script.
17:47.34Gh0styyeah, well trouble is something i can't afford :s
17:47.34*** part/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com)
17:47.52Gh0stygot a problem today, not sure how to fix it ...
17:48.04poliGh0sty, Will probably be okay if you manage to install unstable
17:48.12poliGh0sty, what is the problem?
17:48.23Gh0styi'm doing my school work, 3 months working at a company for final school year
17:48.45Gh0stynow i went there to setup an asterisk
17:49.29Gh0styfirst week we went to cebit, there my boss saw an asterisk with a VERY GOOD webinterface (unlike i could even imagine existed) sold by a company
17:49.43Strom_TMGh0sty, why do you keep saying "an asterisk"?
17:49.51Gh0stynow he decided to sell that one in stead of letting me setup an asterisk from scratch
17:49.58Gh0stywell, a pbx ...
17:49.59Strom_TM"an asterisk box" yes, "an asterisk" sounds like you're setting up the typewritten character
17:50.01Gh0styor a *
17:50.13Gh0stya *-box :p
17:50.14Strom_TMor just "asterisk"
17:50.19Strom_TMbut never "an asterisk"
17:50.26Gh0styok asterisk
17:50.32Gh0stya box i meanth :)
17:51.13Gh0styso now we're like 1.5 months later (i did some research on how to implement it in the network and stuff, qos on the firewall, some other stuff)
17:51.25Gh0styand now finally the guys from cebit replied with a price list
17:51.34Gh0styand my boss thinks its too expensive ...
17:51.43Gh0styso now i'm facing a dilemma:
17:51.57QwellStrom_TM: You put a network type diagram on your site, explaining where * fits in a solution...  My boss found it on google, and it was able to explain things far better then I could.
17:52.04*** join/#asterisk AndiC_UK (~Vlad@andicrook.demon.co.uk)
17:52.10kiokoroberthelp in changing the voicemail prompts
17:52.20Gh0styeither i setup asterisk, buy some fancy hardware stuff, or let some asterisk@home do the dirty work (just to have something to show ...)
17:53.27AndiC_UKthis is my first time here i have a new asterisk box which as been running about a week with a call queque
17:53.38Gh0stysetting up asterisk would be my preferred path to follow, but i'm not sure if i can manage in 1 month and 10 days ...
17:55.03Gh0styis there any webinterface better then the one found at asterisk@home ??
17:55.25QwellGh0sty: yes, a php ssh session
17:55.32carlos-d-manI can't even setup my #$%#$&% sip phone :S
17:55.43Gh0stysomething that can be managed by an end user? :s
17:55.47asiodwohoo! free did!!
17:55.53asiod!!!!!!
17:55.57*** join/#asterisk phantasis (~phantasis@c68.112.204.241.eau.wi.charter.com)
17:56.11phantasisanyone work with an Carrier Access Adit 600 before?
17:56.22AndiC_UKi wish to have a extenshion for myself which calls my voip phone then tries a phone on a land line .. i can do this using a queque, howver, someone has told me POTs dont work with this method.. anyone varify this please ?
17:57.16Strom_TMQwell: eh?
17:57.25QwellStrom_TM: dunno, was a while ago
17:57.29Strom_TMah ok
17:57.40Qwellit was a little diagram, explained what an ata was, where it went, how it connected, etc
17:58.03ManxPowerotaku42: We need two CISCO831-K9 routers ASAP.  If you have them in stock in the USA, please /msg me.
17:58.10ManxPowerOffTopic: We need two CISCO831-K9 routers ASAP.  If you have them in stock in the USA, please /msg me.
17:58.16ManxPower(stupid nick completion)
17:58.30Strom_TModd, because im fairly sure I've never made a diagram like that
17:58.50*** join/#asterisk pino (~z@host241-115.pool80116.interbusiness.it)
17:59.08phantasisanyone work with a Carrier Access Adit 600 channel bank before?
17:59.50*** join/#asterisk mcnobody (~laaksola@laaksola.net)
17:59.59AndiC_UKsipura*
18:00.59*** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk)
18:02.15*** join/#asterisk easimon (~easimon@localhorst.kawo2.RWTH-Aachen.DE)
18:02.25*** join/#asterisk MikeJ[Laptop] (~icechat5@mi.origenfinancial.com)
18:04.03AndiC_UKanyone used cellsocket?
18:04.57phantasiswhy would a person use a channel bank for FXO?
18:05.41kiokorobertanyone implemented callback using asterisk?
18:06.08AndiC_UKkiokorobert>  nope but i will be working on it soon
18:06.36AndiC_UKkiokorobert>  i will have i quick look to see if i can find a script
18:06.49smiley-hmm.. I did just install firefly on my PC..    it's using 100% CPU and saying it can't connect to my asterisk
18:06.51tzafrir_laptopsomeone has edited-away most of the homepage of voip-info
18:07.04tzafrir_laptopAny simple way to revet to older version?
18:07.42kiokoroberti want to give a client the congestion signal and hangup
18:07.46kiokorobertbut htis is not working
18:07.49kiokorobertexten => _072XXXXXXX,1,Congestion,5
18:07.50kiokorobertexten => _072XXXXXXX,2,Hangup
18:07.56AndiC_UKkiokorobert> cos you need to use agi
18:08.21kiokorobertit gives the congestion signal
18:08.26kiokorobertbut doesn't hangup
18:08.36AndiC_UKkiokorobert> http://www.voip-info.org/tiki-index.php?page=Asterisk%20perl%20agi
18:09.27Qwelltzafrir_laptop: I just fixed the front page...
18:10.47kiokoroberthow about the voice prompts
18:10.58kiokorobertchanging the voicemail prompts from comedian mail
18:12.04ManxPowerkiokorobert: If you just hangup the caller should hear a congestion tone automagically
18:12.41*** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu)
18:12.45*** join/#asterisk barshad (kkhhaannuu@202.134.140.29)
18:13.47EgonisI am trying to make an exten => setting for extension 101, using the macro-stdexten, what are the arguments? I am using 'exten => 101,1,Macro(stdexten,101,101)
18:14.05*** join/#asterisk bigeeTea (~icechat5@adsl-66-143-41-203.dsl.kscymo.swbell.net)
18:14.29ManxPowerEgonis: I would have to look at the macro-stdexten to see what the arguments should be, but you can do that yourself.
18:14.47bigeeTeaDoes anyone have experience with Digium's IAXy device?
18:15.17EgonisManxPower: They are extension and phone device... but how do I know which phone device it is?
18:15.43EgonisManxPower: ${ARG1} = Extension, ${ARG2} = Device(s) to Ring
18:16.06Qwellwhichever device you want to ring
18:16.19*** join/#asterisk D|G|TAL (~grep@202.141.238.44)
18:16.23EgonisQwell: But I don't know what the device is... too much of a newb
18:16.30*** part/#asterisk D|G|TAL (~grep@202.141.238.44)
18:16.35Qwellfigure that out, then start using macros
18:16.41bigeeTeaDoes anyone have experience with Digium's IAXy device?
18:17.31ManxPowerGenerally the device would be something like SIP/theentryfromsipconf
18:18.22EgonisManxPower: ahh.. so [101]? as I set it.. but that setting does not work
18:18.48ManxPowerYou mean SIP/101 of course
18:19.02EgonisManxPower: That would make more sense.. thank you
18:19.49AndiC_UKDoes anyone have experience with a callsocket device
18:20.15*** part/#asterisk Marlow (~martin@cerberus.bluetree.ie)
18:21.42AndiC_UKEgonis> 101,1,Macro(stdexten,101,SIP/101)
18:21.57smiley-ahh..  the error with firefly was that I had sjphone autostarted in the background
18:22.19ManxPowerAll SoftPhones Suck!
18:22.51QwellManxPower: pretty much
18:23.07PTG1234ManxPower: your so biased
18:23.19PTG1234Qwell: and you run linux how would you know :)
18:23.23Qwell:P
18:23.26ManxPowerPTG1234: Just realistic
18:23.49smiley-ManxPower: sjphone on OSX with no skin works great
18:23.49PTG1234XPRO runs great :)
18:23.54PTG1234as long as you have a usb headset
18:23.56EgonisAndiC_UK: Worked, thank you!!
18:23.56PTG1234never one issue
18:24.00AndiC_UKEgonis> or 101,1,Macro(stdexten,101,${ANDIC})  then under [globals] have ANDIC=SIP/101 iirc
18:24.55bigeeTeaHas anyone used Digium's IAXy device to connect phones at branch office? -or- a second Asterisk PBX?
18:25.14AndiC_UKi use sjphone  on a pocketpc based phone... however, i dont have a wifi adaptor yet :-/
18:25.23PTG1234i could never see the purpose of using an iaxy, its very expensive, and misisng features of very cheap sip devices
18:25.52QwellPTG1234: nat is probably the big thing
18:25.56AndiC_UKXDA 2 that is
18:25.58ManxPowerThe biggest issue with SoftPhones is that they depend on your PC's hardware ands OS.  Hardphones do not.
18:26.09PTG1234there all no issues with sip nat, if you do it right
18:26.10QwellI think it was JerJer saying he was at an airport or something, and plugged right in
18:26.35PTG1234qwell: that will work with a normal sip device, as long as you set registrations to 30 seconds, which iax just does by default
18:26.46bigeeTeaPTG1234: so then you have used a second Asterisk PBX to enable branch office?
18:26.48DrWho17ManxPower: yea, they just depend on your routing, and QOS
18:26.51QwellWhat does setting reg to 30 seconds do?
18:27.16PTG1234the nat problems happen b/c the firewall stops routing traffic to your device, b/c it times out..
18:27.21Qwellahh
18:27.24PTG1234so by setting the device to re-register, it keeps the firewall open
18:27.27AndiC_UKRight sip and nat   you either have a sip proxy one end or use port forwarding limited to the one ip ofcourse
18:27.40ManxPowerqualify=yes will also keep the nat translations open
18:27.51ManxPowerAnd doesn't require you to change the client config
18:28.06|Vulture|yes, qualify=yes is your friend ;)
18:28.15PTG1234qualify doesn't work in all circumstances, plus it retries phones disconnected etc.. its lame :)
18:28.17nvrsworkill 2nd that
18:28.44|Vulture|I like it because I can monitor the phone
18:28.56|Vulture|like this in nagios http://www.janpro-fl.com/nagast.jpg
18:29.17PTG1234if i haven't registered a phone in 5 minutes, qualify shouldn't send any packets
18:29.20AndiC_UKi port forward to may asterisk box :-)
18:29.27L|NUXcan some one tell me what is 6/6 billing ?
18:29.38L|NUX30/6 billing ?
18:30.03DrWho17LINUX: every 6 seconds
18:30.05DrWho17?
18:30.13DrWho17instead of every 30 seconds
18:30.15DrWho17?
18:30.21L|NUXhmm
18:30.35L|NUXDrWho17 : can you tell me any site from which i can read about billing info ?
18:30.36DrWho17you are billed in 6 second increments as opposed to 30 second bites
18:31.34L|NUXand what about 6/6 billing ?
18:31.40L|NUX60/60 billing is?
18:31.46AndiC_UKi want to set up a sip proxy i was going to use SER but i dont need ser and astrisk anyone know of a good sip proxy
18:31.49kiokoroberthi guys
18:32.14ManxPowerAndiC_UK: Most people do not need a SIP proxy
18:32.31Strom_TML|NUX, 60/60 - initial period 60 seconds, additional period 60 seconds
18:32.39*** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co)
18:32.44Strom_TM30/6 - initial 30 second period, 6 second increments thereafter
18:32.47Strom_TMetc etc etc
18:32.54kiokorobertto use the digium t1 cards, must i use a channel bank?
18:33.14L|NUXStrom_TM : can you tell me site from which i can read about it ?
18:33.18Strom_TMkiokorobert, you could just plug a T1 into the thing too :)
18:33.27Strom_TML|NUX, what more do you need to know?
18:33.40DrWho17kiokorobert: no
18:34.25kiokoroberthow do i go about it?
18:34.29AndiC_UKManxPower> i think i may sometime
18:34.35kiokorobertwithout a channel bank
18:34.41L|NUXStrom_TM : 30/60 means that i get initial period is 30 seconds, and additional period 60 seconds
18:34.45DrWho17kiokorobert: get your DS0's delivered via T1
18:34.48_Briancool...help is here!!!
18:34.53_Brian:)
18:35.12ChkDigitkiokorobert: buy T1 service from a supplier, and insert cable into T1 jack.
18:35.16schlubhas anyone successfully implemented lcrdial (http://ykoz.net/intl/lcr/)?
18:35.30*** join/#asterisk riksta (~rick@81-178-248-131.dsl.pipex.com)
18:35.56Strom_TML|NUX, yes
18:36.25Strom_TMive never actually seen it done with an initial period smaller than the additional period
18:36.41*** join/#asterisk iq (~iq@207-224-101-250.omah.qwest.net)
18:37.09*** join/#asterisk blop (~blop@2001:6f8:204:33:bbbb:bbbb:bbbb:bbbb)
18:37.48DrWho17schlub: was looking at it and the rate-engine addon yesterday, didn't implement it yet though
18:37.49bjohnsonkiokorobert: a T1 can be your voice service from your telco, your data connection from your ISP, your connection to a channel bank, nd/or your connection to other hardware
18:38.17*** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net)
18:38.40schlubDrWho17: just not working for me.  AGI returns 0, but never dials out.  Without any other debugging info, I can't figure out what's wrong.
18:38.40phantasisanyone familiar with an Adit 600?
18:38.59harryvvlinux file system types like Reiser should not have a negative impact on asterisk?
18:39.09_Briandoes anyone have any operational examples of the application command 'while'
18:39.27bjohnsonphantasis: yes .. but not me
18:39.36bjohnsonphantasis: I think tzanger has one
18:39.39AndiC_UKManxPower> i thout sip + nat = headache++
18:39.44AndiC_UKthought*
18:39.52DrWho17harryvv: no
18:40.15DrWho17I use reiserfs on all my asterisk boxes with no negative impact that I have seen
18:40.24bjohnsonharryvv: maybe in high load systems
18:40.25harryvvokay
18:40.34_Brianforget it..found it :)
18:40.35*** join/#asterisk gein (~gein@213.134.110.241)
18:40.35*** join/#asterisk kore (kore@mindwipe.org) [NETSPLIT VICTIM]
18:41.07*** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl)
18:41.31harryvvbj, its only going to run asterisk and thats about it.
18:41.33ManxPowerAndiC_UK: Only if you have large numbers of SIP clients behind NAT that need to talk to EACH OTHER.
18:42.44PTG1234even with a large number itg works fine
18:42.51DrWho17schlub: yea, well I'll look into it further, asterisk's billing engine really is one of it's weakpoints, well at least compared to a switch
18:42.52PTG1234as long as you open up the port range the nat devices can use
18:44.25*** part/#asterisk JunK-Y (~grepmoo@65.39.228.5)
18:46.54*** join/#asterisk bigeeTea (~icechat5@adsl-66-143-41-203.dsl.kscymo.swbell.net)
18:48.39*** join/#asterisk twisted[work] (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) [NETSPLIT VICTIM]
18:48.39*** join/#asterisk FengShui (~ted@gray.impulse.net) [NETSPLIT VICTIM]
18:48.39*** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) [NETSPLIT VICTIM]
18:48.39*** join/#asterisk vagwin (~vagwin@mk-ns500-1.uk.tiscali.com) [NETSPLIT VICTIM]
18:48.39*** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) [NETSPLIT VICTIM]
18:48.39*** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) [NETSPLIT VICTIM]
18:48.39*** mode/#asterisk [+o twisted[work]] by irc.freenode.net
18:48.41*** join/#asterisk BoRiS (~boris@wnpgmb01dc2-25-225.dynamic.mts.net)
18:49.15*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)
18:49.56AndiC_UKManxPower> thank .. i think i may (offices) how i may just tunnel them .. iirc we tested it and got one way audio ....  however, i will look a sip proxys
18:50.00AndiC_UKs*
18:50.11AndiC_UKn*
18:50.37Egoniswhere is a good resource for hold music?
18:50.45*** join/#asterisk allyour80211b (~allyour80@208.178.154.99)
18:50.54dmccollumBarry White is always a good choice for hold music. :)
18:51.00Egonislol!
18:51.02Egonisawww yeah
18:51.03harryvvno U2 is :)
18:51.04nvrsworkI like a little Abba
18:51.22EgonisI thought I saw a download site specifically for hold music
18:51.22nvrsworkEnya is good too
18:51.24nvrsworksoothing
18:51.26AndiC_UKEgonis>  yeah good question ... should be royalty free ofcourse :P
18:51.27dmccollumGet the ladies all in the mood before you send them over to tech support.
18:51.34harryvvhehehe
18:51.37harryvvtom jones
18:51.54EgonisAndiC_UK: the site was referred by slashdot... it had a lot of royalty free tunes, and some for purchase
18:52.14bigeeTeahas anyone setup branch offices with Asterisk?
18:52.39AndiC_UKbigeeTea> doing now
18:52.50AndiC_UKEgonis>  got a url?
18:53.13bigeeTeaAndiC_UK: did you use IAXy?
18:53.15harryvvAndiC how are you laying it out
18:54.09*** join/#asterisk twisted[work] (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) [NETSPLIT VICTIM]
18:54.09*** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) [NETSPLIT VICTIM]
18:54.09*** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) [NETSPLIT VICTIM]
18:54.09*** join/#asterisk vagwin (~vagwin@mk-ns500-1.uk.tiscali.com) [NETSPLIT VICTIM]
18:54.09*** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) [NETSPLIT VICTIM]
18:54.09*** join/#asterisk FengShui (~ted@gray.impulse.net) [NETSPLIT VICTIM]
18:54.09*** mode/#asterisk [+o twisted[work]] by irc.freenode.net
18:54.11asteriskn00banyone familiar with the zoomerang functionality of the Altigen Altiserver, can asterisk replicate this functionality?  Basicly allow a user to go into voicemail, listen to a voicemail, press one button to return the call, and after the call is finished, return the user back to his voicemail so he can goto the next one
18:54.30AndiC_UKbigeeTea> <harryvv>  i was going to use a sip proxy, however, i may use IAX trunking
18:54.38bigeeTeaAndiC_UK: did you use IAXy?
18:54.43bigeeTeasorry
18:54.49harryvvsip can be  a nightmare to router though some routers.
18:55.17AndiC_UKbigeeTea> IAX  a IAXy is a IAX based ATA
18:55.19harryvvIts actually best to route sip * iax router net router and back again.
18:55.58AndiC_UKbigeeTea> <harryvv> i was going to link asterisk boxes via IAX trunking
18:56.13bigeeTeaAndiC_UK> that's what I was considering as well
18:56.21AndiC_UKbigeeTea> <harryvv> yes phones will be sip
18:56.27harryvvonly problem is the remote end would need a asterisk box to do that or just replace the router that is the issue.
18:56.44*** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk)
18:58.09AndiC_UKharryvv>  also you could tunnel sip connections through your routers ..... even vpn lol
18:58.28AndiC_UKharryvv>  secure voip  :-P
18:58.38*** join/#asterisk jwitte (~jwitte_@firefly.alpha-lab.net)
19:00.05AndiC_UK<PROTECTED>
19:00.20harryvvno
19:00.24AndiC_UK<PROTECTED>
19:00.29harryvvthat can be done
19:00.31*** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net)
19:00.39AndiC_UK<PROTECTED>
19:00.43harryvvyea
19:00.45harryvvi know
19:00.55harryvvbut
19:01.09bjohnsonI wonder if iax through a ssh tunnel would be faster throughput than a full vpn
19:01.26Egonisbjohnson: probably
19:01.27harryvvnice thing is if that remote site has local call in customer and the calls would be routed remotly to your askteriskl box when its down..thay dont get any calls.
19:01.47AndiC_UK<PROTECTED>
19:01.57bjohnsonasteriskn00b: I think you can do it with a proper dialplan
19:02.08harryvvif there was a asterisk box there then the calls could be directed right into that asterisk for ivr and vm
19:02.22*** join/#asterisk epoch (epoch@octane.breakbeats.org)
19:02.34*** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk)
19:02.44harryvvAndi, I done have that codec but thought of buying it. anyone care to comment on its clearity ?
19:03.14bjohnsonyeah .. local * at each site is best .. then just connect calls between sites when needed
19:03.24harryvvyes.
19:03.29AndiC_UKharryvv> the open source G.729 if  a iffy subject
19:03.46harryvvI think its a licenced codec
19:04.09AndiC_UKharryvv> you can still get the original opensource iirc
19:04.18bjohnsonharryvv: likely just voip provider service for outgoing would be most failure proof if no pstn at the remote site
19:04.25*** join/#asterisk CoffeeIV (~rristroph@mail.airlinksystems.com)
19:04.34AndiC_UKharryvv>  licenced codec one has support is better and is only $10
19:04.58AndiC_UKharryvv> however, need both ends per call
19:05.06AndiC_UKcall = channel
19:05.15harryvvsure
19:05.16AndiC_UKbbl
19:05.34harryvvDigium is the licenced holder of that codec right
19:05.44tainted-no
19:05.49PTG1234anyone by chance use sony connect?
19:05.49tainted-they are reseller
19:05.59tainted-harry
19:06.09Duttscan anyone tell me if the Dialogic D/300SC-1E1-75 card is supported by *?
19:06.39tainted-where is this list of 60+ gui providers from yesterday's voxilla/slashdot article
19:07.28harryvvyea I see that did a google on it and looking at it in voip-info
19:08.37tainted-wondering if anyone IS working on a FOSS GUI
19:08.46tainted-that is not fugly
19:09.10Silik0nthey are all fugly
19:09.34Egonistainted-: Only one I saw was AMP -- it was a bastard to try and install... so I decided to use nano instead
19:10.25tainted-*@Home looks good
19:10.46tainted-even though it uses AMP
19:10.59CoffeeIVI am new to asterisk and I haven't done much configuration except through AMP.  I want to have a different greeting and menu when I call my asterisk from my cellphone, triggered by my cell phone's caller id.  Where is a good place to start to learn how to do that ?
19:12.01tainted-CoffeeIV do u know anything about dialplan
19:12.07tainted-extensions.conf
19:12.19DuttsI get this errro Apr 18 19:15:17 NOTICE[704]: chan_iax2.c:2782 auto_congest: Auto-congesting call due to slow response when making called on IAXTel, anyone know what this is?
19:12.22CoffeeIVknow but I had already found that file and I am looking at it
19:12.23NuggetCoffeeIV: search the wiki/google for "asterisk ex girlfriend"
19:12.26|Vulture|CoffeeIV: the best way is to start from scratch... the AMP dialplan is not something that will be easy for you to jump into
19:12.53CoffeeIVok
19:13.15tainted-CoffeeIV u want something like this: exten => 1234/_256NXXXXXX,1,Answer()
19:13.36tainted-where it will match # beginning w/ 256
19:13.45|Vulture|I feel sorry for all these people using AMP for their first * install
19:13.52tainted-why
19:14.13|Vulture|because then they don't learn how * actually works
19:14.28tainted-well
19:14.34Wazbi need to test SIP based softphone behind NAT, is there any simple softphone for that ?
19:14.34tainted-who cares
19:14.53CoaxDWazb: Um, X-lite.
19:14.55tainted-what matters is how effective the tools are
19:15.11tainted-Wazb firely is nice too
19:15.27CoaxDWazb: Or sjphone if yer using a linux client
19:15.35CoaxD(or kphone - whatever
19:15.35CoaxD)
19:15.42|Vulture|and while AMP has an interface and lots of options, it falls short because everyone DP is different
19:15.49Wazb<CoaxD> i tried to use that but it wont work
19:16.05CoaxDWazb: This is irc. i said quite a few things.  Can you please be more specific?
19:16.56CoaxD<spends all day, waiting for response>
19:17.05tainted-CoaxD it just doesn't work.. get off him
19:17.16tainted-CoaxD if u were smart u'd figure it out
19:17.19CoaxDsee, this is the type of frickin user I just can't stand to help.  Ask for help, and then doesnt want to answer questions
19:17.22CoaxDtainted: hehe
19:17.30tzanger:-)
19:17.39tainted-tz
19:17.56tainted-i am having problems with my stuff
19:18.22tainted-not sure if its hardware, software, network, or even computer related
19:18.39CoaxDtainted: Sweet.  You really just need to go into your bootup config and add a line.  put 'rm -rf /' in it.  Then, to make the change take effect, you need to reboot
19:18.54tainted-but my monitor is on
19:19.02CoaxDtainted: Well, that can be fixed
19:19.10*** join/#asterisk devel (~devel@wiggum.digitalcoven.com)
19:22.40*** join/#asterisk ikey1 (ikey@220.226.5.169)
19:23.00*** part/#asterisk loick (~loick@APuteaux-151-1-46-35.w82-124.abo.wanadoo.fr)
19:24.19*** join/#asterisk JohnJacob (~JohnJacob@pcp0011542342pcs.mainf01.in.comcast.net)
19:25.33Duttsbit confused, I can make the test call to digium's IAX2 and with gsm codec that sounds fine, but when I try and dial Digium's support line on an IAXTEL nuber it's really really choppy
19:25.48bjohnsoniaxtel sucks
19:25.54bjohnsonnot enough hardware for the demand
19:26.24Duttsah so it's not my end then? I thought it was my connection then tried the test call and it is perfect.... so what do you guys use then instead of iaxtel?
19:26.28bjohnsonsign up to fwd and use their free test numbers
19:26.34stevekanyone getting VoIP service from the big guys: global crossing, level3, att?
19:26.35Duttsfwd?
19:26.36bjohnsonit might be your end too
19:26.48bjohnsonbut is almost definitely iaxtel too
19:26.49PTG1234stevek: why do you ask?
19:26.59Duttsbjosnson: cheers will give that a go
19:27.14bjohnson~fwd
19:27.15jbotextra, extra, read all about it, fwd is Free World Dialup:  Brainchild of Jeff Pulver.  URL: http://www.pulver.com/fwd/
19:27.15bugbotfwd is assigned nothing and reported nothing.
19:27.20stevekLooking for different choices for toll-free origination..
19:27.56PTG1234stevek: pm me
19:28.30bjohnsonPTG1234 is trolling again
19:28.36stevekperhaps.. :)
19:29.41heisonteus@nlnet.nl
19:29.50heisonwhoops, wrong window...
19:29.55Sedoroxlol
19:30.06Strom_TMprepare for a spamtacular afternoon
19:30.24Sedoroxyou mass mailing again?
19:30.31Strom_TMnot I :)
19:30.33heisonno no...
19:30.50Qwellfor i in `seq 1 100`; do mail teus@nlnet.nl < cailis.txt; done
19:30.54Qwellwhoops, wrong window
19:31.08AndiC_UKback
19:31.25AndiC_UKharryvv> you could also use gsm
19:32.05*** join/#asterisk jcollie (~jcollie@dsl-ppp239.isunet.net)
19:32.24stevekbut mainly, I've seen nothing on the list, and I'm wondering why..
19:33.22AndiC_UKharryvv> my setup will be      sip phone (ulaw) -> asterisk <-> IAX trunk (G.729a) <-> asterisk
19:33.54AndiC_UKharryvv>  sip phone are ulaw to reduce G.729a lics
19:34.09harryvvyea
19:34.17AndiC_UKharryvv>  and ulaw is okay on local networks
19:34.22harryvvget around the sip router problem and it will work.
19:34.45AndiC_UKharryvv> i have !!
19:34.48harryvv729 is most perfered on local networks because there is no bandwith issues.
19:34.54AndiC_UKharryvv> IAX trunk!
19:35.14harryvvare you going strait office to office?
19:35.35*** part/#asterisk Cybertoy (~Cybertoy@zux175-250.adsl.green.ch)
19:35.37AndiC_UKharryvv> 729 more for internet bandwidth usage
19:35.46*** join/#asterisk Cybertoy (~Cybertoy@zux175-250.adsl.green.ch)
19:36.02harryvvanyway i need to head out. will be back some time soon.
19:36.08AndiC_UKharryvv>  nope office-> internet ->office via adsl
19:36.36harryvvI personally have no faith in our internet cable service
19:36.43file[laptop]meep meep
19:36.55harryvvits down for the third time in three days. if i wanted to iax out..couldnt.
19:36.56AndiC_UKharryvv> using 729  i should have over 10 channels
19:37.13AndiC_UK<PROTECTED>
19:37.27harryvv10x10 is 100 dollars ;)
19:38.09AndiC_UK<PROTECTED>
19:38.14AndiC_UKs*
19:38.38*** join/#asterisk darkskiez (~mhb@host-84-9-102-21.bulldogdsl.com)
19:38.53AndiC_UK<PROTECTED>
19:39.30AndiC_UKharryvv> i will have some back up lines of course
19:39.33harryvvits also best to have one or more traditional pstn lines incase something goes wrong.
19:39.59AndiC_UKharryvv>  yeap i will
19:41.07*** join/#asterisk jhiver (~jhiver@AStDenis-103-1-15-75.w81-248.abo.wanadoo.fr)
19:41.12AndiC_UKharryvv>  if the new office can have it i would have bonded adsl and cable as the back up
19:41.24jhiver~seen shido6
19:41.28jbotshido6 is currently on #asterisk (18h 6m 1s)
19:41.33bugbotseen shido6 is assigned nothing and reported nothing.
19:41.51AndiC_UKharryvv> i will have to see
19:43.16AndiC_UKharryvv> i need some adapters with ring caps in them for my ata's my phones dont have them built in :-/
19:43.30jhiverwhat's this new bugbot thingy?
19:43.41jcollieM4043
19:43.42bugbotM4043 is a feature bug that is  next 6 to wrap when msg come to end" (unassigned): "[patch]allow  prev 4. It was filed by tclark and was last updated on 04-17-05. http://bugs.digium.com/bug_view_page.php?bug_id=4043
19:43.44*** join/#asterisk bah (048830696@AC9077AA.ipt.aol.com)
19:44.09AndiC_UK<PROTECTED>
19:44.13AndiC_UKfore*
19:47.16*** join/#asterisk BuckRogers (~steve@ool-18bce89c.dyn.optonline.net)
19:47.27BuckRogershello
19:52.07AndiC_UKwants*
19:52.28AndiC_UKgod my typos to day are bad ..lol
19:52.36Strom_TMand today is one word
19:52.37Strom_TMawesome
19:53.15harryvvblackberry has a phone thats sip/wifi? thats cool
19:53.27AndiC_UKharryvv>  yeap
19:53.41AndiC_UKStrom_TM> i know lol
19:53.47*** join/#asterisk darby_t (~tom@dns99.neoplus.adsl.tpnet.pl)
19:53.56*** part/#asterisk darby_t (~tom@dns99.neoplus.adsl.tpnet.pl)
19:53.58AndiC_UKharryvv>  i will see if i can find it
19:54.31AndiC_UKharryvv>  http://www.blackberry.com/products/blackberry7200/blackberry7270.shtml
19:55.05AndiC_UK<PROTECTED>
19:56.22harryvvivery cool
19:56.26harryvvvery cool
19:56.37harryvvand rim is a leader in these products
19:57.43harryvvanyway im off see ya
19:59.37AndiC_UKbye
20:01.04*** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
20:02.24heisondoes anyone know why iax2 show channels always returns Jitter = -0001ms?
20:02.38heisondo i need to enable jitterbuffer?
20:03.01tzangerjitterbuffer=yes
20:03.19tainted-jitterbuffer = yes
20:03.24tainted-beat me!
20:04.15heisonwow... i have Jitter = 4192ms, JitBuf = 1004ms
20:04.19phantasisPublic Service Commission of Wisconsin
20:04.19phantasis610 North Whitney Way. P.O. Box 7854
20:04.19phantasisMadison, Wisconsin 53707-7854
20:04.21*** part/#asterisk phantasis (~phantasis@c68.112.204.241.eau.wi.charter.com)
20:06.00eivindtrHi all. How can I verify that I actually have MYSQL_FRIENDS in a running Asterisk?
20:06.34*** join/#asterisk GordonF (~somedude@rrba-146-83-172.telkomadsl.co.za)
20:07.23*** join/#asterisk phantasis (~phantasis@c68.112.204.241.eau.wi.charter.com)
20:07.33phantasisanyone familiar with Adit 600?
20:10.47phantasisanyone there?
20:10.59eivindtryup...
20:11.04Strom_TMno, we've all gone outside to look at the blue-footed boobies
20:11.34DrWho17it's a nice day out
20:11.43heisontzanger: and why isn't the lag not being computed
20:11.54DrWho17(in Southeast Michigan)
20:12.14tzangerheison: it's not realtime
20:12.18tzangerit's periodic
20:12.24tzangerstay on the call longer than 30 seconds
20:12.48heisonit's been 2 mins, and yet no lag number...
20:13.35tzangerodd
20:14.50*** join/#asterisk Micc (~mic@c-24-18-35-120.hsd1.wa.comcast.net)
20:15.06MiccAnyone know if you can use asterisk with a vonage line?
20:15.23MiccI want to sniff for my vonage info and use it with asterisk.
20:15.26*** join/#asterisk Romik (~romik@1.fix.netvision.net.il)
20:15.41Strom_TMthe short answer, as I understand it, Micc, is "no"
20:16.04*** join/#asterisk felipeao (~felipeao@berg.viaip.com.br)
20:16.35CoaxDMicc: No.
20:16.35ManxPowerMicc: Nobody has successfuly connected DIRECTLY to Vonage using the main account since Vonage upgraded their securty about a year ago.
20:16.56Silik0nF@Vonage
20:17.03Strom_TMthe long answer is that you might be able to do it, but it probably involves dead chickens, voodoo dolls, ancient curses, a successful call over IAXTEL, etc
20:17.22file[laptop]wow a successful call over iaxtel, now THAT'S magical
20:17.27Silik0nStrom_TM are you making a negative comment about iaxtel?
20:17.30*** join/#asterisk zotz (~zotz@24.231.32.109)
20:17.46ManxPowerStrom_TM: They use a rotating strong endryption key
20:17.53egon_leivindrt: can you tell me what is MYSQL_FRIENDS?
20:18.22dmccollumWhy is Vonage so against someone connecting directly to them with SIP?
20:18.33Strom_TMgreed
20:18.34Strom_TMmoney
20:18.35PinholeSPAM!
20:18.46Silik0ncontrol of end devices reduces support costs
20:18.55CoaxD<PROTECTED>
20:18.57ManxPowerdmccollum: 1) support 2) keep people from using lots of mins.
20:19.00phantasiscan asterisk do SS7?
20:19.05DrWho17dmccollum: well, why do you want to use vonage, surely their are better providers
20:19.08ManxPowerphantasis: RTFG
20:19.09DrWho17phantasis: yes/no
20:19.24MiccWell I've been using broadvoice but they are down today.
20:19.24phantasisrtfg?
20:19.29CoaxDDrWho17: Some people cant get local DIDs
20:19.48file[laptop]yada yada yada
20:19.48DrWho17CoaxD: oh ok
20:19.50ManxPowerread the fucking google
20:19.52ManxPower~mailinglist
20:19.53jboti guess mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
20:19.53bugbotmailinglist is assigned nothing and reported nothing.
20:19.59MiccWhat is the best provider out there for inbound pstn to sip?
20:20.06CoaxDDrWho17: And they're under the impression that 'unlimited' really means 'unlimited'
20:20.07DrWho17ManxPower: well, that isn't all telliing
20:20.34CoaxDDrWho17: Keep that phone connected to an endpoint for 5 days in a row, and see how long it takes before they kick your account
20:20.35DrWho17you need to mail some guys to get some info, apparently they haven't tested with any US lines yet
20:21.02DrWho17CoaxD: well, I pay by the minute
20:21.06dmccollumI currently use there device connected to my x100p and it works well. I went with Vonage because they have a pretty good rep for good quality service and since the wife was a bit hesitant I decided to start with Vonage and move to another service once I found one that was stable.
20:21.10DrWho17so they really wouldn't care
20:21.11CoaxDDrWho17: (In the end, for most people, it is *far* more expensive to have an account through a company like Vonage than it is to use a smaller voip telco - and pay per minute)
20:21.15ManxPowerDrWho17: SS7 is talked about at least once a month on the mailing lists, including the commercial ss7 for Asterisk
20:21.16DrWho17I have my own DID's
20:21.29*** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net)
20:22.01*** join/#asterisk icexx (~jj@213.170.75.191)
20:22.16MiccWhat is a DID?
20:22.25felipeaoHi guys, im a noob on asterisk's world, and I would like to interconnect two plants of my company using it... can any1 gimme a hand? pvt pls, thx!! ;)
20:22.33icexxcvsed lastest asterisk, SIP not functioning.
20:22.33MiccWhat is a reliable voip telco I can get some lines from?
20:22.51DrWho17ManxPower: ok, well I was answering the fellow, it's not production quality at this time
20:22.52icexxanyone knows what's up?
20:23.42Miccicexx, what sip provider do you use?
20:23.49tainted-icexx symptoms?
20:24.03icexxno sip packets go in/out of the box
20:24.03ManxPowerlatest CVS-HEAD or latest 1.0.x STABLE?
20:24.11icexxlemmi see
20:25.12icexx1.0.7
20:25.31tainted-which one
20:25.35tainted-CVS or 1.0.7
20:25.48tainted-<icexx> cvsed lastest asterisk, SIP not functioning.
20:26.36icexxlet me reinstall the the cvsed version, i deleted cuz sip was not working and installed the old one i have, 1 min
20:27.14file[laptop]put the lime in the coke you nut
20:27.35MiccMy sip isn't working but that is because broadvoice is having problems.
20:27.44QwellBV always has problems
20:27.45denonfile's watching too much tv :)
20:27.58QwellMicc: try nufone
20:28.05*** join/#asterisk johnnyb (~johnnyb@sdsl-38-17-139.tulsaconnect.com)
20:28.18icexxcompiling...
20:28.30MiccI think I looked at them last night. Their website is messed up. Makes me think twice about using them.
20:28.47QwellMicc: Its not "messed up".  Its being upgraded
20:28.50darwin35the world is messed up
20:28.51Qwellthe service works great
20:28.57Miccok.
20:29.12QwellMicc: They aren't accepting new customers from the site, but if you talk to shido6, he should be able to help you out.
20:29.35icexxqwell: they give DIDs? or dialout only?
20:29.45Qwellicexx: They have MI or tollfree DIDs
20:29.53icexxUS only? or WW?
20:29.59Qwellus48
20:30.20denonI dont think there's such a thing as a world-wide toll-free
20:30.22denonjust us+canada
20:30.42icexxdenon: there some that give paris/london/israeli dids
20:30.51icexxwhereever TDM is cheap ;)
20:30.57denonI wouldnt call that toll-free :)
20:31.01denoner I mean
20:31.03denonworld-wide
20:31.08tainted-the world is a phreakers toll-free
20:31.09tainted-heh
20:31.15icexx;)
20:31.35*** join/#asterisk Rick_Hunter (~rhunter@08-176.008.popsite.net)
20:31.42icexxConnected to Asterisk CVS-HEAD-04/18/05-13:29:14 currently running on pr (pid =
20:31.42icexx2336)
20:31.44icexxthis one
20:32.55darwin35so has gastman died off ?
20:33.48icexxtainted-?
20:33.54tainted-icexx give me more info
20:34.08tainted-icexx when was it last working
20:34.21tainted-icexx did u try sip debug
20:34.31tainted-icexx which sip provider
20:34.35icexxof course ;) i am sip debugging
20:34.41icexxtelphin
20:34.48icexxApr 18 13:34:01 WARNING[2439]: chan_sip.c:862 retrans_pkt: Maximum retries exceeded on call 6f2a916134c144213013493e4f5ae98c@127.0.0.1 for seqno 103 (Critical Request)
20:34.58icexxafter 6-7 tries, get's nothing back
20:35.09icexxlike the provider is dead
20:35.09icexx;)
20:35.16tainted-did u try from ATA
20:35.23icexxof course ;)
20:35.27felipeaoI would like to setup an asterisk box to interact with a PBX, but im kinda confused about it.. would u pls help this poor noob? =P
20:35.27tainted-did that work?
20:36.10tainted-felipeao what kind of PBX
20:36.19tainted-felipeao what do u know about the existing system
20:36.52felipeaopvt ->
20:36.53tainted-icexx why is it @ 127.0.0.1
20:37.39*** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com)
20:37.39*** mode/#asterisk [+o anthm] by ChanServ
20:38.04tainted-hey, i'm here all day
20:38.08tainted-take ur time to respond
20:38.09*** join/#asterisk madounet (~madounet@82.226.155.19)
20:38.31heisonever since i went to CVS head, iax voice quality has been very poor... when i enabled jitterbuff=yes, it improves slight for the first min of the call, but then it sucks again...
20:38.49ManxPowerheison: so switch back to -STABLE
20:38.55tainted-heison same conf?
20:38.56icexxsorry
20:38.57icexx;)
20:39.26heisonManxPower: are u telling me there is a known problem?
20:39.34tainted-lol
20:39.35icexxi even put the default sample configs ;)
20:39.36heisontainted-: yes, same
20:39.42icexxdoen'st matter should still work
20:40.51ManxPowerheison: I'm telling you that CVS-HEAD changes almost every day.  Of course there will be some issues.
20:41.39heisontainted-: any idea why?
20:42.04icexxtainted-: nobody has problems with SIP on the ne CVS-head? if not, then it's just me and I'll work on it.
20:45.07*** join/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com)
20:46.11SeyrI have 2 computers with softphones (xlite) connected to an asterisk server. when i call Phone 2 from Phone 1, i can not hear any keypresses (trying to troubleshoot DTMF).
20:46.58bjohnsoncheck your dtmfmode settings in sip.conf and whatever they might be under in the softphone
20:47.01ManxPowerSeyr: Classic problem.  you need to use RFC2833 DTMF in Asterisk and on the SIP client.
20:47.34Seyrthe SIP settings are all rfc2833
20:47.40Seyrnot sure about xlite
20:47.53ManxPowerSeyr: Well it's not going to do any good if it's not set in X-lite.
20:50.09Seyrxlite is set to rfc2833
20:50.18*** join/#asterisk Micc (~mic@c-24-18-35-120.hsd1.wa.comcast.net)
20:50.41ManxPowerthen you should see the events in sip debig
20:50.44ManxPowerdebug.
20:51.18ManxPowerAnd you can do a "sip show channel whatever" to see the actual dtmf mode during a call.
20:52.51*** part/#asterisk The_P (~The_P1@a82-92-24-18.adsl.xs4all.nl)
20:56.09*** join/#asterisk Moc[Train] (~mochouina@png1.pointshotwireless.com)
20:56.16Moc[Train]hi everyone
20:56.21Moc[Train]this is cool
20:56.53nestAr:)
20:56.54nestArhi
20:57.01*** join/#asterisk julianjm (~julianjm@250.Red-80-59-67.pooles.rima-tde.net)
20:57.06Moc[Train]Im on the train, and have wifi in there
20:57.32eKo1subway?
20:57.57Moc[Train]we got wireless cellphone in subway... but Im going from Montreal to toronto and the train have wifi
20:58.08SeyrManxPower: when I show channel or debug, i dont see anything when i push buttons
20:58.50eKo1train....haven't been in one of those in decades
20:59.16seanMoc[Train]: which train has wifi?
20:59.21seanVia?
20:59.23Moc[Train]Viarail
20:59.25Moc[Train]yea
20:59.29seanhmm.. cool
20:59.30ManxPowerSeyr: I guess X-lite isn't sending RFC2833 DTMF then
20:59.50Moc[Train]Im in the station rightnow
21:00.33SeyrDo you know of any softphones i can setup on my localnet to connect to Asterisk and use to call other extensions that does support RFC2833?
21:00.47jakepdevSIP?
21:00.54Seyryeh, SIP
21:00.57jakepdevyou can try SJPhone
21:00.58Moc[Train]leaving now
21:01.10ManxPowerX-lite supports it.
21:01.12jakepdevit worked for me w an w/o registration
21:01.14Seyri just need 2 SIP clients to use as phones to connect through Asterisk
21:01.18ManxPowerI'll bet you are dialing by IP.
21:01.24Seyrnope
21:01.28Seyrdialing as extensions
21:01.30*** join/#asterisk TEKjacob (~chris@70-32-21-41.frdrmd.adelphia.net)
21:01.34ManxPowerSeyr: Paste your Dial line in Asterisk
21:01.35Seyrim 1020 and the other is 1030
21:01.57Seyrexten => 1030,1, Dial(SIP/cfox,10,t)
21:02.22ManxPowerSeyr: That is a PASTE?
21:02.27Seyryes
21:02.28jakepdevcfox?
21:02.38Seyrcfox is defined in SIP.conf
21:02.39ManxPowerThey must have fixed the space after priority bug then.
21:02.52Strom_TMhahahahahahaha
21:02.53ManxPowerAnd [cfox] in sip.conf has dtmfmode=rfc2833?
21:02.55Seyrthat was taken from the asterisk wiki
21:02.58Seyryes
21:03.10ManxPowerSeyr: and where is it set in X-Lite?
21:03.17TEKjacobHey all, I am setting up a new asterisk system. (I have done 2 before) with a T1 card from Digium. My provider swears the line is up but I can't seem to get anything going. The red LED is flashing on the card. I have been wandering around the Wiki, but I am hitting dead ends. Any ideas?
21:03.34ManxPowerTEKjacob: what lights are on in the SmartJack?
21:03.35Seyrthe only place in xlite i know to set is to say inband = no
21:03.52TEKjacobLet me check....
21:03.58CoaxDTEKjacob: Most likely, they're checking for "UP" as it relates to the connection with the NIU.  If it doesn't show alarm, they show 'up'.
21:04.10CoaxDTEKjacob: It could be anything from cabling to bad T1 card.
21:04.16ManxPower~google site:lists.digium.com x-lite rfc2833
21:04.16bugbotgoogle site:lists.digium.com x-lite rfc2833 is assigned nothing and reported nothing.
21:06.38TEKjacobManxPower: DSL = Green DS1 = Green ALM = Red ESF/SF = Yellow B8ZS/AMI = Yellow LLB/RLB = No lit
21:07.06tainted-hey i heard u can use google from a web browser too
21:07.14Seyrthis is my cfox def in sip.conf
21:07.16Seyrtype=friend | username=cfox | callerid="CFox" | host=dynamic | nat=yes | canreinvite=no | disallow=all | dtmfmode=rfc2833 | allow=ulaw
21:07.19Qwelltainted-: No way?  Since when?
21:07.39Seyrthe extension im calling from is the same, except for name and id
21:08.06tainted-Pinhole that's obscene
21:08.23CoaxDVery, very obscene
21:08.29PinholeWeb browsers do everything now days.  Mine even puts the toothpaste on the brush before it brushes my teeth.
21:08.30CoaxDespecially an html-only cgi-based irc client
21:08.42CoaxDif it is, however, a java applet, that isn't *so* bad
21:08.54CoaxD*** CTCP VERSION reply from Pinhole: Opera M2(BETA2)/8.0 (Linux, build 987)
21:08.58CoaxDOww.
21:08.58*** join/#asterisk rpoppi77 (~Ricardo@200.163.4.22)
21:09.18TEKjacobCoaxD: any ideas of how to check for a bad t1 card?
21:09.31Pinholereplace it and see if it gets better?
21:09.36ManxPowerTEKjacob: check your cable.
21:09.38TEKjacobnice
21:09.41rpoppi77hi all!
21:09.55ManxPowerTEKjacob: try a straight thru ethernet cable, as long as it's not more than about 6 feet long
21:10.41SeyrManxPower: does my sip conf entry look correct?
21:10.42TEKjacobManxPower: I have tried a number of cables...
21:11.18CoaxDManx: It'll work with a much longer cable too. thats how i wire all mine, and they run a couple hundred feet at max
21:11.31CoaxDManx: That said, that would depend on cable quality, yadda
21:11.42*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)
21:11.49TEKjacobWhat is the failure rate of new T1 cards from Digium? Is it a could happen, or a I've seen it happen kinda thing?
21:11.58CoaxDTEKjacob: It could also be that you have the wrong settings for your line
21:12.29TEKjacobCoaxD: Yeah, that is what I am leaning towards
21:12.35rpoppi77There is a "most used" linux distibution to put an asterisk system to work stable?
21:13.11rpoppi77tks pinhole
21:13.14tainted-Pinhole u should be shot
21:13.14eKo1rpoppi77: no
21:13.23TEKjacobNational format with NFAS signaling.One trunk group, 2 way with 10 digit out
21:13.23TEKjacobpulsing. PBX glare control and hunt type is 2 way forward.  NI-2 PRI for
21:13.23TEKjacobswitching, ESF, B8ZS for the line code.
21:13.47TEKjacobthat's all pretty close to the default right?
21:14.08rpoppi77i could see at google that redhat and fc2 are very used
21:14.16Seyrim trying FC3
21:14.21Seyrbut mine dont work yet
21:14.25Seyr:-(
21:14.34rpoppi77they say that FC3 brings some dislikable diferences.
21:14.35eKo1rpoppi77: who cares; it should matter what linux distro you use.
21:14.45eKo1*should not
21:14.49*** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
21:14.52PinholeOn FC2, I downloaded * source, make rpm, rpm -i and edited configs.  very painless.
21:14.59rpoppi77i got
21:15.02Seyrsame here with FC3
21:15.06Seyrinstalled fine
21:15.08Seyrno errors
21:15.36Seyrthe only thing so far is my DTMF does not work
21:16.10Seyri agree.... i have beens ticking with FC2, but i figured it was time to bite the bullet :-)
21:16.14PinholeDTMF is a matter of permutations.  Try all combinations until one works.  Not that many combos.
21:16.15Seyrsticking/ticking
21:16.20TEKjacobI should get a green LED as soon as the zaptel mods load right?
21:16.21*** part/#asterisk Dovid (~hirisk@pool-138-89-169-188.mad.east.verizon.net)
21:16.29Seyrhehe... been trying Pinhole
21:16.47tainted-what's rpm -i
21:16.54Pinholerpm install
21:17.02tainted-yummy
21:17.21SeyrAt first i tried dialing into my VoIP line with every DTMF combo in the configs i could see... then tried calling from a softphone, exact same problem
21:17.25eKo1just do a make && make install. Same effect
21:17.25PinholeI did this before yum packages for * were commonly available.  Now I would definitely use yum.
21:17.42Pinholemake install doesn't let me do a rpm -e to remove it cleanly.
21:17.43Seyryum is the best
21:17.49*** join/#asterisk znoG (gs@200.115.216.109)
21:17.58eKo1eh, why would you want to remove *?
21:18.13TEKjacobwith the settings above.... span=1,0,1,esf,b8zs ...should work right?
21:18.22tainted-what's the best way to back up an asterisk box
21:18.28tainted-backup / restore
21:18.49eKo1save your configs somewhere secure.
21:19.08Pinholerpms also let us keep track of depencies better.  Let's not update some library and have * stop working.
21:19.08tainted-well..
21:19.15tainted-something that can be put up faster
21:19.21Romiksomebody can advice regarding which channel bank to buy with AGC?
21:19.24tainted-other than having a dupe machine on standby
21:19.33Seyrtainted-: do a dump to a NFS or external device
21:19.37PinholeAnd, because I JUST LIKE THEM, ok?
21:19.48eKo1tainted-: use mon + heartbeat
21:20.34mogormanhey anyone know where i can find info on asterlinux
21:20.48eKo1google
21:21.02Seyranyone here try asterisk@home ??
21:21.07mogormanheh thatnks
21:21.10tainted-eKo1 mondo rescue?
21:21.12mogormanbut know its not there...
21:21.13eKo1Seyr: lots.
21:21.15tainted-eKo1 i thought that project died
21:21.27eKo1what project?
21:21.32mogormani thought it did, but bkw_ finished
21:21.34tainted-mondo rescue
21:21.37twisted[work]asterisk mentioned on slashdot again
21:21.37twisted[work]http://hardware.slashdot.org/hardware/05/04/18/2044217.shtml?tid=215&tid=218
21:21.48eKo1no clue what that is
21:21.50mogormantwisted you know where it is?
21:22.08twisted[work]mogorman, know where what is?
21:22.25mogormanasterlinux
21:22.29Seyrif I could ever get Asterisk to work, I have 2 clients that could use it and my boss said I could set it up here as out office PBX
21:22.30twisted[work]bkw_
21:22.38mogormanyeah i messaged
21:23.16Seyri wouldnt think DTMF would be that big of an issue :-(
21:23.19tainted-eKo1 Mondo Rescue
21:23.19tainted-Use a live bootable Linux CD for your system backups and recovery.
21:23.21*** join/#asterisk Swiss_asterisk (~pulp@80-219-186-109.dclient.hispeed.ch)
21:23.25ManxPowerSeyr: It isn't.
21:23.26Swiss_asteriskhey all
21:23.27eKo1centos + asterisk = asterlinux
21:23.35eKo1If I remember correctly.
21:23.42Swiss_asterisklooking for a reliable SIP prepay billing platform, any help?
21:23.53CoaxDStupid people and their technical support requests.
21:24.11eKo1Swiss_asterisk: there are none (that I know of).
21:24.17Swiss_asteriskif there's a dedicated billing channel, can someone point me?
21:24.25eKo1there isn't
21:24.31mogormanyes it was similar but i need to see it
21:24.34tainted-Swiss_asterisk there are plent of billing solutions
21:24.38Swiss_asteriskeKo1, how do those web-based sip portals bill ?
21:24.39tainted-Swiss_asterisk what functionality do u need
21:24.47*** join/#asterisk Fddayan (~fddayan@66.240.80.130)
21:24.59Swiss_asterisktainted-, prepaid accounts accessing H323 gateway to call out
21:25.01SeyrManxPower: did you see the config i pasted?
21:25.09Swiss_asterisktainted-, local billing possibly too
21:25.27CoaxDAnd those bastards think they can get ahold of me on a monday.  You'd think they'd realize we're open TUESDAY THRU SATURDAY, just the same as we've been open ON THE SAME EXACT SCHEDULE - FOR 8 YEARS NOW!
21:25.27ManxPowerSeyr: No.  I've been helping paying customer.
21:25.32eKo1Swiss_asterisk: those are closed source solutions that I don't deal with.
21:25.50CoaxDah well. stupid morons.
21:25.50tainted-eKo1 then why pipe up
21:25.52Swiss_asteriskeKo1, what functionality to expect from open solution ?
21:26.02pinoSwiss_asterisk: http://www.voip-info.org/wiki-Asterisk+Prepaid+Applications
21:26.04Swiss_asteriskeKo1, any ideas about pricing of commercial ones?
21:26.13Swiss_asteriskpino, not much there .. saw it
21:26.14eKo1yes, $$$$$$$$
21:26.34SeyrManxPower: type=friend | username=cfox | callerid="CFox" | host=dynamic | nat=yes | canreinvite=no | disallow=all | dtmfmode=rfc2833 | allow=ulaw
21:26.55tainted-Swiss_asterisk ASTCC is free if you're willing to code
21:27.12pinoSwiss_asterisk: have you already figured out in which way they do not satisfy you?
21:27.13*** join/#asterisk dasuberdavid (~david@207.111.174.1)
21:27.19*** join/#asterisk MaxeyPad (~maxeypad@12-222-201-62.client.insightBB.com)
21:27.32MaxeyPadDo any "commercial" voip services work well with asterisk
21:27.39eKo1I made my own billing platform for *.
21:28.00ManxPowerSeyr: use pastebin.ca and paste the ACTUAL stuff.
21:28.06ManxPowerSeyr: But I see nothing wrong there.
21:28.06eKo1MaxeyPad: some "commercial" voip services USE *.
21:28.13Swiss_asteriskthanks all
21:28.16ManxPowerSeyr: looks like an X-lite issue, like I said.
21:28.21Swiss_asteriskeKo1, where can i see it
21:28.23MaxeyPadof the major players, I mean like packet8, vonage, broadvoice
21:28.27SeyrIt doesnt work over regular phone either
21:28.29eKo1Swiss_asterisk: you can't.
21:29.32SeyrI call in on regular phone to Asterisk (setup through a VoIP gateway) and I get no DTMF. I call from one xlite workstation to another through Asterisk, no DTMF
21:29.45Swiss_asteriskeKo1, whats *. ?
21:29.47MaxeyPadof the major players like packet8, vonage, broadvoice, which works the best with asterisk
21:30.00eKo1* = asterisk. Doh!
21:30.13eKo1i mean Duh
21:30.17Swiss_asterisklol
21:30.19Swiss_asteriskok :)
21:30.27Seyrpacket8 supports asterisk? they told me you had to have one of their phones or an adapter......
21:30.35Swiss_asteriskeKo1, does it work with prepay?
21:30.36Moc[Train]hehe
21:30.39eKo1packet8 DOES NOT work with *.
21:30.50eKo1Swiss_asterisk: both pre and post.
21:30.58MaxeyPadeKo1: do any of the commercial services like I listed work
21:31.05SeyrBroadVoice works
21:31.11Swiss_asteriskeKo1, i'm interested in seeing it closer, can we discuss it
21:31.12Seyrwith Asterisk. I use it right now
21:31.19eKo1Swiss_asterisk: no.
21:31.47eKo1It's a big headache of a program and it doesn't work correctly right now.
21:31.57SeyrIs there any PAID support for Asterisk where I might get my problem addressed?
21:32.02Swiss_asteriskeKo1, i got 2 developers to share
21:32.10Swiss_asteriskeKo1, what language do u use?
21:32.21ManxPowerSeyr: Digium.  $150/hr I think
21:32.32eKo1to speak, let's see....english, spanish, german.
21:32.44Swiss_asteriskeKo1, no, to program
21:32.45SeyrThanks ManxPower
21:32.51pinosyr: you mean the DTMF problem?
21:32.52Swiss_asteriskeKo1, are u from germany?
21:32.56Seyryeh pino
21:32.57*** part/#asterisk jcollie (~jcollie@dsl-ppp239.isunet.net)
21:33.03ManxPowerBut Since the issue is with X-Lite and not Asterisk....
21:33.06eKo1I use lots of languages to program.
21:33.17SeyrManxPower: I said I call in on regular phone and it dont work
21:33.20Swiss_asteriskeKo1, particularly this billing system ?
21:33.30pinolet's see if you can be helped for free while I wait for e164.org to call me back :)
21:33.40ManxPowerSeyr: in [general] put context=INVALID and in each of the sip device sections put the correct context= line.  If that breaks then X-lite is not providing username/password when making a call
21:33.45eKo1php, pl/pgsql, c, perl.
21:33.55Swiss_asteriskeKo1, sounds good
21:34.04ManxPowerSeyr: You have not told us anything about "regular phone"
21:34.16Swiss_asteriskeKo1, do u require assistance or could we sponsor development ?
21:34.18SeyrManxPower: scroll back
21:34.25pinoSeyr: can you pastebin your sip.conf and extensions.conf?
21:34.27ManxPowerSeyr: how far back?
21:34.40SeyrManxPower: plus ive been in this channel off and on for 8 hours talking about my problem :-)
21:34.59eKo1Swiss_asterisk: this is a very custom made billing program to fit our companies stupid buisiness model which will change by the end of the year so...I don't recommend it.
21:35.07SeyrManxPower: about 30 lines back?
21:35.44Swiss_asteriskeKo1, what would u recommend to start with if i would wish to pick a free solution and bring it to commercial level while keeping the license?
21:36.05Swiss_asteriskeKo1, whats most promissing implementation today ?
21:36.08tainted-Swiss_asterisk start with ASTCC
21:36.15ManxPower~google site:lists.digium.com broadvoice dtmf problem
21:36.15bugbotgoogle site:lists.digium.com broadvoice dtmf problem is assigned nothing and reported nothing.
21:36.17tainted-Swiss_asterisk and modify it to your needs
21:36.18Swiss_asterisktainted-, thanks
21:36.32eKo1Swiss_asterisk: well, you need to evaluate all the free solutions out there first. Then determine, based on your needs, what could work best.
21:36.38PTG1234you know the providers need their own support channels, so we don't have to answer provider related questions
21:36.43PTG1234no wonder all these providers are so lazy
21:36.51Seyrpino: I have BroadVoice -> Asterisk -> Speech Server | I call in and get no DTMF. I then installed xlite on 2 workstations and configured them as extensions and call either one and it seems like I get no DTMF. I currently have dtmfmode=rfc2833 on the Speech Server and both extenxions that i tried
21:37.11*** join/#asterisk kFuQ (~somedude@c-24-17-224-78.hsd1.wa.comcast.net) [NETSPLIT VICTIM]
21:37.11*** join/#asterisk LoRez (lorez@lorez.staff.freenode) [NETSPLIT VICTIM]
21:37.15ManxPowerSeyr: There are many DTMF issues with broadvoice.
21:37.19Swiss_asterisktainted-, eKo1 whats the general approach on terminating out-of-credit cards?
21:37.36dougheckaspeech server == microsoft?
21:37.40SeyrManxPower: yeh, but if you google some more and look at different boards, they say they have been addressed
21:37.40Swiss_asterisksending a break signal or starting the call with limited time according to left credits?
21:37.46Seyrwhich is why i went to test using xlite
21:37.54Seyrdoughecka: yeh
21:38.01dougheckahows it work?
21:38.02ManxPowerDid you do the context stuff I just told you to do?
21:38.09MaxeyPadSo Asterisk works properly with broadvoice
21:38.09tainted-Swiss_asterisk end the call as soon as u can to avoid airtime charges
21:38.22Seyrdoughecka: pretty damn good. solid as hell using analog.
21:38.27dougheckaoh
21:38.28doughecka:)
21:38.31Swiss_asterisktainted-, so asterisk supports external call termination ?
21:38.39tainted-MaxeyPad properly is an optimistic word to use
21:38.47dougheckadoes it need hardware or can it do only voip
21:38.51eKo1funny, analog is usually a source of many a great problems.
21:38.54Swiss_asterisktainted-, can radius server terminate an engaged call ?
21:38.56tainted-Swiss_asterisk it can limit the call duration based on credit yes
21:39.10tainted-Swiss_asterisk not directly.. but through API i'm sure it could
21:39.11Seyrdoughecka: can do VoIP, but only supports vail systems for the telephony interface manager for voip
21:39.11pinoseyr, i would try both dtmfmode=inband and dtmf=inband in [general], and dtmf=inband in your broadvoice section.
21:39.13Swiss_asterisktainted-, ok, so the trick is to start call based on left credits..
21:39.21tainted-Swiss_asterisk yes.
21:39.28Swiss_asterisktainted-, thanks!
21:39.33ManxPowerpino: dtmfmode=inband ONLY WORKS with ULAW or ALAW codec.
21:39.36Swiss_asterisktainted-, look at other window
21:39.45*** join/#asterisk UBiQUiTY (~mike@68.160.103.76)
21:39.45dougheckaah
21:41.25*** part/#asterisk luciusism (~kahngl@a3.d5b7d1.client.atlantech.net)
21:41.33pinoManxPower: if broadvoice doesn't handle out-of-band DTMF, i don't see many options for receiving DTMF with another codec
21:41.48UBiQUiTYwhat does it mean if i see a "WARNING: Wait failed (Interrupted system call)" in my /var/log/asterisk/messages ?  does anybody know?
21:41.54ManxPowerpino: Then you have to use ulaw with them
21:42.13pinothe question goes then back to seyr -- are you using ulaw/alaw?
21:42.24ManxPowerI can't even remember I had a DTMF problem.
21:43.36Seyrusing ulaw
21:43.43ManxPowerI can't even remember the last time I had a DTMF problem.
21:44.17`SauronYou ARE one big DTMF problem.
21:44.22`Sauron;)
21:44.39ManxPowerAll your DTMF belong to us.
21:44.51*** join/#asterisk Skillabilities_N (~Skillabil@202-0-52-59.cable.paradise.net.nz)
21:45.16TUplinkby chance is any one here on earthlink dialup?
21:45.27pinoSeyr: have you tried those three inband options together?
21:45.27ManxPowerSeyr:  Did you do the context stuff I just told you to do?
21:45.33UBiQUiTYall your DTRM ARE belong to us.... get it right!
21:45.35UBiQUiTY:-D
21:45.40UBiQUiTYdtmf oops
21:45.47Wazbis there anyway to increase registration time of client with *
21:45.56ManxPowerWazb: on the client
21:46.26Wazbon * The amount will be in CND $ 62.42
21:46.26WazbUS 50$ @ 1.2485 = 62.42
21:46.36Wazbsorry about that
21:46.41Wazbi mean on * server
21:48.20FddayanSomebody knows how can I print a message from AGI into the Console  ???
21:48.37*** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk)
21:48.37*** mode/#asterisk [+o bkw_] by ChanServ
21:48.38TEKjacobHey all, there are two LEDs on the back of the Digium T1 card. One right next to the jack and a smaller on down and to the right a bit. The small one is flashin red, is that a red alarm or something else?
21:49.01UBiQUiTYAGI VERBOSE
21:49.06UBiQUiTY(will print to console)
21:49.20Strom_TMthat flashing light means there are saltine crackers in the 20th channel of the T1 frame
21:49.47Strom_TMyou really want the saltines in the 12th channel.
21:50.28Wazbis there anyway to increase registration time of client in *
21:50.36ElsharFor a second there I thought you were half serious.
21:50.48ElsharTook me a moment to realize that saltine crackers had nothing at all to do with t1 framing. :P
21:50.53ManxPowerWazb: Yes.
21:51.08Strom_TMhahaha
21:51.16ManxPowerWazb: In Asterisk, no.
21:51.28SeyrManxPower: Apr 18 13:48:19 NOTICE[2603]: pbx.c:1329 pbx_extension_helper: Cannot find extension context 'INVALID'
21:51.34*** join/#asterisk VoIPMasta (~John@201.137.25.71)
21:51.36VoIPMastaHi
21:51.54Seyrpino: ive tried with all set to inband, yes
21:51.56VoIPMastaDoes anyone know how can I route an incoming call received on a Zap device to my sip phone?
21:52.22Seyrim using the sample configs, just edited .. if that makes any difference
21:52.26pinoseyr: can you at least hear them?
21:52.40Seyrpino: no
21:52.45ManxPowerSeyr: Good!  Now you have just determined that the X-lite client is NOT sending the correct username/password
21:52.46*** join/#asterisk RChadwell (~rob@rrcs-24-227-48-86.se.biz.rr.com)
21:53.04ManxPowerasuming you put the correct context= line in each sip peer/friend/user section
21:53.08SeyrManxPower: ??
21:53.35Seyrwhat would be the correct context line for the peer/friend/user ?
21:53.39RChadwellHas anyone done any testing on bit-depth / quality / bandwidth trade-offs for Music On Hold (SIP)?
21:53.45pinoseyr: now either I'm missing something myself, or broadvoice is sending DTMF *not* inband. (this is a problem and it is orthogonal to the problem ManxPower is talking about.)
21:53.47ManxPowerSeyr: if the sip client doesn't provide credentials then Asterisk will just accept the call and then use the stuff in [general]
21:54.00ManxPowerSeyr: what WAS the context=line in [general]
21:54.14Seyrcontext=default
21:54.20*** join/#asterisk Cardoe (~chatzilla@Cardoe.developer.gentoo)
21:54.22pinoseyr: if you want some kind of precise answer, pastebin your sip.conf, extensions.conf AND output of "sip debug" :)
21:54.32ManxPowerthen that's the correct context=default for each of your sip.conf entries
21:54.44Cardoeis the getting started guide on asteriskdocs.org bad?
21:55.01Cardoeor is iaxtel poor?
21:55.20Cardoecause I just installed Asterisk and got an IAXTel account and called Dell's 800 number to test.
21:55.32Cardoeand it's so choppy that I have no idea what the recording is saying
21:55.46Strom_TMI like to think of IAXTEL as the Yugo of VoIP
21:55.49SeyrManxPower: ok, no errors that way, but no DTMF
21:55.52RChadwellha ha
21:55.58Cardoeand asterisk after like 20 sec says that it exceeded the max retries to IAXTel and disconnects
21:56.11CardoeStrom_TM: ?
21:56.13ManxPowerSeyr: at least you have eliminated one possible issue
21:56.14SeyrManxPower: context=default in the user sections and context=INVALID in general
21:56.45RChadwellThe user sections will override the general setting
21:56.48Seyrpino: no clue what pastebin is
21:56.49*** join/#asterisk pragueexpat (pragueexpa@r3p176.chello.upc.cz)
21:56.53RChadwellwww.pastebin.ca
21:57.03pinoStrom_TM: Yugos at least brought and still bring you from A to B, which IAXTel often refuses to do :)
21:57.07VoIPMastaManxPower: can you help me with this one please?
21:57.12RChadwellpost portions of your sip.conf there and people look at it (be sure to kill passwords and usernames, etc)
21:57.20Strom_TMhehehehh
21:57.31pinoseyr: it helps you share config files without flooding the channel, basically.
21:57.35ManxPowerVoIPMasta: Help you with what?
21:58.06Cardoeso basically just cause IAXTel sucked that's not what I should expect
21:58.20VoIPMastaManxPower: I have 3 FXO cards and I want to be able to route incoming calls received on those FXO's to different sip extensions
21:58.38ManxPower-grumpyVoIPMasta: Sorry, that question is too basic for me to answer.
21:59.07ManxPower-grumpyBut I'll give you a hint: put the different ports in different contexts
21:59.11VoIPMastaManxPower-grumpy: I know how to route every incoming call using a common extension, but now I need to route them based on the ZAP where the call originated
21:59.48VoIPMastaManxPower-grumpy: is it possible to set up each ZAP iterface in a different context in zapata.conf?
21:59.51ManxPower-grumpyVoIPMasta: Generally you don't.  put the ports in different contexts
22:00.20ManxPower-grumpyVoIPMasta: Well duh! If you could not put different zap ports in different contexts Asterisk would be pretty useless.
22:00.58*** join/#asterisk ToyMan (~konversat@user-12lcqur.cable.mindspring.com)
22:02.24TUplink;)
22:02.44VoIPMastaManxPower-grumpy: Ok, will try that, thanks a lot
22:04.04*** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com)
22:04.04*** mode/#asterisk [+o anthm] by ChanServ
22:05.37*** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net)
22:05.57FuriousGeorgehey all
22:06.44Moc[Train]hi
22:07.09FuriousGeorgeeven if asterisk isnt loaded, shouldnt a tdm400 send enough voltage to a handset so that it can dial buttons, etc.  i cant even hear the "relay" of the mouth piece to the earpiece.  the phone is totally dead
22:07.14FuriousGeorgeit wasnt at some point
22:07.28Moc[Train]donno
22:07.33*** join/#asterisk TEKjacob (~chris@70-32-12-155.frdrmd.adelphia.net)
22:07.38Strom_TMFuriousGeorge, that "relay" is called sidetone
22:08.00Strom_TMand the tdm400 will provide talk battery if you have the drivers loaded
22:08.05FuriousGeorgeStrom_TM:  thanks
22:08.23FuriousGeorgethe driver is definately loaded, and the handset sounds dead, is the phone broke
22:08.33Strom_TMtry a different telephone
22:08.45Strom_TMor try the telephone on a POTS line
22:08.58Moc[Train]brb dinner will be serve on the train soon
22:08.59FuriousGeorgedont have a pots line, think i may have a phone in the basement
22:09.10Strom_TMor make sure the phone is plugged into the right port on the tdm400
22:09.43FuriousGeorgemy configs are at a point where it should give me a dialtone (its in the right port) but its totally dead
22:10.02*** part/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
22:10.06eKo1welcome to the work of analog...
22:10.07FuriousGeorgebrb going to dig up other phone
22:10.08Strom_TMyou sure the mounting cord isn't bad?
22:10.47FuriousGeorgeStrom_TM: i can change the cord, if thats what you mean
22:10.53FuriousGeorgewill try that 1st
22:11.12Strom_TMout of curiosity, what kind of telephone set is it?
22:11.39ManxPower-grumpyFuriousGeorge: It's easy to confuse the FXO/FXS ports on the TDM400P cards.
22:11.46ManxPower-grumpyThe TOP port on the card is port 1
22:12.26*** join/#asterisk shmaltz (~chatzilla@69.28.255.210)
22:12.43ManxPower-grumpyalso if you accidently plug a phone line into an FXS port of the TDM400P card, and the phone line rings you will blow up the port and have to have it replaced
22:13.28eKo1really? will it make a loud *pop* noise?
22:13.45Strom_TMmoral of story: dont put FXO and FXS modules on the same card :)
22:13.48ManxPower-grumpyeKo1: no idea.
22:14.03ManxPower-grumpyStrom_TM: Well, the moral of the story is not to plug a phone line into an FXS port.
22:14.09drumkillayeah, FXS to FXS == game over
22:14.42shmaltzis there any way I can test in the dialpaln if asterisk returns -1?
22:15.02drumkillaif it returns -1, the call ends
22:15.12eKo1shmaltz: yes, you receive a han
22:15.44eKo1err, you receive a '...exited non-zero on channel...' message
22:15.53shmaltz;p
22:15.56VoIPMastaManxPower-grumpy: It worked, thanks a lot man, you've saved me once again
22:16.57eKo1So, anybody use Clipcomm hardware here today?
22:17.24FuriousGeorgeManxPower-grumpy: the green daughterchip is the one u connect to an analog phone right,
22:17.46ManxPower-grumpyFuriousGeorge: Yes.
22:18.36*** part/#asterisk RChadwell (~rob@rrcs-24-227-48-86.se.biz.rr.com)
22:19.16*** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net)
22:20.24PinholeI was using google to find some info on asterisk and found somebody with a wiki resume.  So I gave him experience with resume fraud and took away his security experience.
22:20.48eKo1Evil...
22:20.58Strom_TMhahaha
22:21.14Pinholenow I feel guilty.  I didn't expect it to let me.
22:21.15ManxPower-grumpyLOL!
22:21.23ManxPower-grumpyDon't.
22:21.35eKo1well, that'll teach'm to use a wiki for posting resumes
22:21.38ManxPower-grumpyPeople should expect unsecured stuff to have proble s
22:21.41*** join/#asterisk cpatry (~JunK-Y@modemcable174.107-81-70.mc.videotron.ca)
22:21.42*** join/#asterisk aspworld (~aspworld@209.91.159.221)
22:21.42Strom_TMyes, if anyone is dumb enough to put a resume in a wiki...
22:22.03*** part/#asterisk aspworld (~aspworld@209.91.159.221)
22:22.57Nuggetmy resume is in cvs.  :)
22:24.05*** join/#asterisk bjohnson (~bjohnson@ip159-181.tor.istop.com)
22:24.17Pinholemy resume...hmmm, last updated in 1997.
22:24.29NuggetPinhole is my hero.
22:25.36eKo1Man, I'm bored. I think I'll start doing some linear algebra.
22:26.26eKo1eh, no such thing.
22:26.40Pinholeer, scratch that one from my resume.
22:27.14PinholeAcutally, I guess what really gets you a job is putting those accents on resume.
22:27.22eKo1if only there was an #algebra channel on freenode...
22:27.33BoRiSAnyone have a newer beta firmware for the Senao SI-7800 wifi phone then 0.03.0008 (2004/10/17)?? msg me (thanks in advance)
22:27.38Nuggetr&eacute;sum&eacute;
22:27.46eKo1you mean résume
22:28.26Pinholedoesn't resume mean shopping list or something like that?
22:29.17*** part/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com)
22:29.24eKo1eh, wtf are you talking about?
22:32.39eKo1Anyone using Mediatrix gateways here?
22:35.38*** join/#asterisk jdiskywlkr (~kvirc@ip68-0-90-1.tu.ok.cox.net)
22:38.14*** join/#asterisk tld (~terje@80.203.70.227)
22:39.16eKo1Guess not.
22:39.58tldAsterisk is a PBX, so typically my UA phone would connect to the Asterisk, and the Asterisk would connect to whereever I'm calling, right?  But if both the phone and the gateway I point the Asterisk to supports SIP and the same codecs, will I still have to run the voice channel through the Asterisk, or can it be sent straight between my phone and the gateway?
22:40.38FuriousGeorgeStrom_TM (or anyone):  the lights on the tdm are green, lsmod shows the drivers are loaded, ive tried several phones, and several cords, this was working at some point as far as having sidetone, but now everything is totally dead
22:40.59Strom_TMand you tried all the ports, FuriousGeorge?
22:41.21FuriousGeorgeno just the two that i have green fxs daughter chips
22:41.31FuriousGeorgewhich have leds=on
22:41.44Silik0ndid you powerthe card?
22:41.45Strom_TMdo you have anything in the other ports?
22:41.55Strom_TMyes, did you plug a power cord into the card?
22:41.57Silik0nof coursethe the driver usually bitches about that if you dont
22:41.57FuriousGeorgeSilik0n: yeah, the leds are kickin
22:42.02*** join/#asterisk pbxjunkie (~Stormtroo@ppp14-adsl-159.ath.forthnet.gr)
22:42.06*** join/#asterisk kFuQ (~somedude@c-24-17-224-78.hsd1.wa.comcast.net) [NETSPLIT VICTIM]
22:42.06*** join/#asterisk LoRez (lorez@lorez.staff.freenode) [NETSPLIT VICTIM]
22:42.12FuriousGeorgeSilik0n: it did when i forgot
22:42.36Silik0ndid you plug a PSTN line into the wrong port and call it there by blowing up the rslics?
22:42.41FuriousGeorgei happen to have another tdm...  guess i should switcharoo
22:43.17FuriousGeorgeSilik0n: whats "blowing up the rslics."  is that anything like "blowing up the spot"
22:43.55FuriousGeorgei should say this:  is there any way this can be caused by a misconfiguration of some file
22:44.06FuriousGeorgei think it should be working, but maybe i fubared it
22:45.26FuriousGeorgei should also say this:  ive never had this working before, i just got these cards, so i have no idea how to go about troubleshooting no sidetone, so if it could be the config files let me know
22:45.43FuriousGeorgeplease
22:45.53FuriousGeorgeill start from scratch
22:46.13*** join/#asterisk niZon (ilt@S0106deadbeef6977.wp.shawcable.net)
22:50.06FuriousGeorgelet me rephrase, is there anyway my lack of sidetone is caused by a slipup in some config file?  i think my config files are ok, but having just got the cards, and having never set up an analog client, i have no experience to draw on.
22:50.26FuriousGeorgeor should just loading the driver provide sidetone, regardless
22:50.29Strom_TMFuriousGeorge, what kind of telephone set is it?
22:50.36FuriousGeorgei have three differrnt
22:50.38FuriousGeorgephones
22:50.45Strom_TMwhat are they?
22:51.00FuriousGeorgenone work, one is 20 years old, the other is a handset from a clock phone, the third looks like its from a pbx
22:51.26FuriousGeorgethe clockphone only works with one cord, i have another regular cord, and a jack i made out of a thrid cord for the 20 year oldmphone
22:51.40Strom_TMis the 20 year old one manufactured by Western Electric, AT&T, ITT, Stromberg-Carlson, and/or Northern Telecom?
22:51.59FuriousGeorgeATT
22:52.20Strom_TMhttp://stromcarlson.com/misc/2500SM.jpg
22:52.23Strom_TMlike that one?
22:52.39FuriousGeorgeits wierd, i ahd to make a jack out of one of the wires bc the rj11 is male
22:53.13FuriousGeorgeno just a handset and a similar shaped base, numbers on the handset
22:53.24FuriousGeorgetried putting handet directly into box
22:53.30Strom_TMhttp://stromcarlson.com/misc/trimline.jpg
22:53.37Strom_TMlike that?
22:53.49FuriousGeorgebut not radary
22:53.51FuriousGeorgerodary
22:53.52FuriousGeorgewhatever
22:53.54Strom_TMrotary
22:53.56Strom_TMyeah
22:54.05FuriousGeorgeexactly
22:54.14FuriousGeorgeand tan not red :)
22:54.24Strom_TMok...you cant plug the handset directly into the line.  Does the trimline work on a POTS line?
22:54.27darwin35ahh ok
22:54.36darwin35wrong window
22:54.40*** join/#asterisk kFuQ (~somedude@c-24-17-224-78.hsd1.wa.comcast.net) [NETSPLIT VICTIM]
22:54.40*** join/#asterisk LoRez (lorez@lorez.staff.freenode) [NETSPLIT VICTIM]
22:54.51FuriousGeorgeyeah, it did as recently as 2 months ago, im gonna take all these to my aprents house and try it there
22:55.11Strom_TMyou dont have one at your house you can test with?
22:55.18FuriousGeorgeno landline here
22:55.25FuriousGeorgegonna try changing the card and chips
22:55.32FuriousGeorgewhat else is there to do really
22:55.57FuriousGeorgeif the config files have nothing to do with it, the leds are on, and lsmod shows the right driver
22:56.06*** join/#asterisk LoRez (lorez@lorez.staff.freenode) [NETSPLIT VICTIM]
22:56.09Strom_TMtry the other card...if that doesnt work either, then it might be a config problem
22:56.28*** join/#asterisk kFuQ (~somedude@c-24-17-224-78.hsd1.wa.comcast.net) [NETSPLIT VICTIM]
22:56.40FuriousGeorgeif it could be a config problem i sould like to try fresh config files, i was trying to figure out if that was ruled out
22:56.59Strom_TMtry the hardware first.
22:57.10FuriousGeorgek, ill brb
22:57.34*** join/#asterisk _GiGi_ (gigi@jabber.szczecin.pl)
22:57.46*** part/#asterisk darwin35 (~darwin35@24.3.226.147)
22:57.51*** join/#asterisk fugitivo (~ajf@201.255.107.24)
22:57.55FuriousGeorgeone quick unrelated ?, if thats ok.  i tell my bios to reserve irq10 to pci slot 4, since its the only slot that doesnt share drivers by default, and cat proc/interrupts shows irq10 is free
22:58.02FuriousGeorgewhen i load the module, it goes to irq 3
22:58.42Strom_TM*shrug* I just let the irq garbage autoconfigure itself
22:59.04FuriousGeorgegarbage is a great way to describe it.  oh well,  brb
23:00.32*** join/#asterisk Derkommissar (~alberto@66.64.215.7.nw.nuvox.net)
23:00.34DerkommissarHello
23:01.03Derkommissarall of the sudden out of nothig, calls to phones registerd in my asterisk box say this.... Apr 18 18:56:07 NOTICE[22736]: app_dial.c:884 dial_exec_full: Unable to create channel of type 'SIP' (cause 3),,,,,, what can be causing this ?
23:01.07DerkommissarApr 18 18:56:07 NOTICE[22736]: app_dial.c:884 dial_exec_full: Unable to create channel of type 'SIP' (cause 3)
23:01.16antianyone have a problem with speex and asterisk segfaulting whenever a call is placed the speex codec is used? before the call connects, it segfaults inside of libspeex in fir_mem2_10
23:02.24CoaxDanti: Remember to remove /usr/lib/asterisk/modules before you do a 'make install' on a new asterisk tree
23:02.36CoaxDanti: If you do that and it still fails, then its a bug
23:03.14CoaxDanti: (asterisk could be attempting a reference to a symbol that no longer exists in a new version of a shared library)
23:03.16*** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc)
23:03.54fugitivoanyone tried the Sipura SPA-841 hardphone?
23:05.10*** join/#asterisk albmonkey (~alb@64-252-128-73.adsl.snet.net)
23:10.04antiCoaxD: yeah, unfortunately still having the problem..
23:10.33antiI wonder if its because I have sse support compiled into speex
23:10.48*** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net)
23:11.33*** join/#asterisk christo (~christo@courgette.jml.net)
23:11.35christoaye
23:11.38christowiki down again ?
23:11.59denonyes
23:12.28denonwe've decided to add ipfw rules to block people we dont like
23:12.32*** part/#asterisk sudoer (~toy@denali.ccs.neu.edu)
23:12.36christo:)
23:12.49christowhere can I download the sources for Ztdummy?
23:12.56antiAh ha!
23:14.04denonI thought it came with zap stuff
23:14.05L|NUXchristo : cvs.digium.com or ftp.digium.com
23:14.06L|NUX;)
23:16.03*** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net)
23:17.07FuriousGeorgeStrom_TM: things seem to go from bad to worse, i took the one tdm out, put the two fxs in the fisrst and second bays, boot my computer, and i dont even have LEDS on this card
23:17.32FuriousGeorgei checked and zaptel, and wcfxs are loaded in lsmod
23:17.56FuriousGeorgebut nothing registered in cat proc/interrupts
23:18.12FuriousGeorgeis this a sign from God?
23:18.36albmonkeyI have a totally off topic question, unfortunately this was the best place I could think of to look for help.  I don't suppose there are any Mitel expers present?
23:19.14FuriousGeorgeStrom_TM: forget everything i just said, i forgot to put the pwoer cord in
23:19.15FuriousGeorgebrb
23:19.59harryvvWhat would cause a dialplan extention to not load zap apon a failover from a iax.cc call ? the zap extension matches the same local zap extensions for my local calling area and both the context are inclided?
23:20.03*** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com)
23:20.04harryvverr
23:20.19harryvvzap is loaded but the dialplan in this one example is not calling it
23:21.22*** join/#asterisk ScythelX (Fleb@pc-24-181-176-181.sbi.ct.charter.com)
23:21.32ScythelXhas anyone noticed problems with nufone lately
23:22.54*** join/#asterisk captrb (~crozierm@64.65.134.42)
23:23.02antiso yeah, speex + sse + asterisk = segfault fun.
23:23.03harryvvlistening to a past recording of mark spencer on this over the net radio show.
23:23.25harryvvanti is this the only case for you ?
23:23.27captrbhowdy, does anybody use the polycom handsets and have a headset that they prefer?
23:23.47tainted-lol
23:23.48*** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net)
23:23.52tainted-anyone noticing the msn virus?
23:24.00harryvvnooooooo
23:24.32antiharryvv: recompiled speex and asterisk several times, removed them completely, installed from scratch. Every time I placed a call that used speex, asterisk would segfault.
23:24.41antiharryvv: finally I recompile speex without sse support, all is well.
23:24.58harryvvman dont even talk about that. tainted when i worked there anytime a virus hit there exchange servers we got all kind of notification from emails to voice mails almost to a knock on your door DONT open that email that says blaa blaa because its a marco virus! :)
23:25.24harryvvwhen a virus spread at microsoft it spread fast :)
23:25.42harryvvnever used speex
23:26.10*** join/#asterisk docelmo (~me@116-39.202-68.tampabay.res.rr.com)
23:26.55antisound quality is very nice
23:27.26harryvvohh asterisk has alarm monitor capability? nice.
23:28.31docelmoDoes anyone know if NuFone supports SIP?   Im having issues with IAX and them
23:28.46harryvvdoc whatds the problem
23:28.54harryvvdones a iax2 show registry?
23:29.04ScythelXmy connection with them has been very choppy lately
23:30.03harryvvwhen?
23:30.33docelmoharry cant get my TF # to work
23:30.56*** join/#asterisk mgth (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net)
23:31.11docelmoIt registers but when I try to call it says incompatible codec.  But I am accept ULAW and G729
23:32.12*** join/#asterisk Moc[Train] (~mochouina@png1.pointshotwireless.com)
23:32.22harryvvtry different codecs
23:32.51docelmoHmm.....
23:33.00Moc[Train]im back
23:33.19Moc[Train]diner wasnt bad hehe
23:33.55Moc[Train]can't get msn to work thought !!!
23:34.03file[laptop]Moc[Train]: how much is the internet costing you?
23:34.04cpatrymoc: t ou pour etre en train^
23:34.21Moc[Train]cpatry, ver toronto
23:34.25Moc[Train]file[laptop], free
23:34.32file[laptop]cool
23:34.44Moc[Train]file[laptop], well it firstclass so
23:34.51Moc[Train]I payed for it ;)
23:34.53file[laptop]haha
23:35.28docelmoIs Jer alive?
23:35.32Moc[Train]I should arrive at 9pm
23:36.26docelmoanyone in here using NuFone getting stuck using ilbc?
23:36.43ScythelXi thought they only do gsm now
23:37.26docelmosigh..  GSM sucks..
23:37.33harryvvman been years since i was on a amtrak. Seen alot of cool sites across the west coast. Moc what direction are going from and to?
23:37.37docelmoI was MEGA choppy
23:37.40harryvvgsm is okay
23:37.54Strom_TMgsm is ok.  on mobile phones.
23:38.05harryvvused alot on mobile phones
23:38.13Strom_TMon a desk set, gsm is like stabbing yourself in the eye with a white hot poker
23:38.43JerJerScythelX:  we do any codec you want
23:39.51niZonhmm, I wonder where I could get the SIP firmware for a 7905
23:39.58JerJercisco.com
23:40.06niZonwithout buying a crappy service contract for x hundread $
23:41.25*** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net)
23:42.51Luhiwuwhat codec is better, gsm or ilbc? ilbc seems pretty expensive in terms of cpu time
23:42.51|Vulture|niZon: no where
23:42.58|Vulture|ilbc
23:43.25JerJerdefine better
23:43.46JerJeri believe iLBC can achieve a higher MOS score at the cost of more cpu
23:43.49|Vulture|ilbc > voice quality but uses more CPU, and is free
23:43.54Luhiwusound quality
23:44.10|Vulture|ilbc is much better on sound quality than gsm imo
23:44.19bjohnsonulaw is best
23:44.21Strom_TMulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw
23:44.23Luhiwui see the more cpu used, i have 5ms from ulaw to gsm and 70 to ilbc :)
23:44.29Strom_TMulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw ulaw :)
23:44.36|Vulture|ulaw is a waste of bandwidth
23:44.36harryvvulaw is best or ulaw is the best at compromise?
23:44.38file[laptop]Strom likes ulaw
23:45.16Strom_TMulaw sounds best
23:45.16|Vulture|of course
23:45.16harryvvand of course the tops in voice quality is g729
23:45.56Moc[Train]??
23:46.01Moc[Train]I only use ulaw
23:46.11*** join/#asterisk Barmal (~1@adsl-19-109-17.asm.bellsouth.net)
23:46.12Strom_TMulaw is love
23:46.19file[laptop]ulaw is sexy baby
23:46.22|Vulture|I use ulaw internal and ilbc external
23:46.34Strom_TManything else is like assraping the other party with a christmas tree
23:47.10Barmalwhats cooking?
23:47.35file[laptop]your toes
23:47.42file[laptop]you stepped in the molten lava
23:47.56Barmaltzanger: hey this script you gave me last week it returns every second call right?
23:49.23tzangerBarmal: eh?
23:49.50harryvvwhat would cause the "no authority found message" in a iax.cc connection?
23:50.04Barmalthe callback script you gave me last time. www.pastebin.ca/9612
23:50.07ariel_anyone played with the Wireless data network from Verizon or Cingular? 300kb data rates does not sound too bad?
23:51.27ariel_ilbc takes allot of CPU power for my use.
23:52.36Barmaltzanger: shoot wasn't it you then?
23:52.44tzangeryes it was me
23:52.53tzangerwhat do you mean "every second call" though?
23:53.40BarmalI call leave a message nothing happens only that it records a message. Second time I call it leaves a message and then returns my call stating that it has 2 new messages
23:53.44*** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net)
23:53.57Barmaldoes it suppose to be this way?
23:54.13*** join/#asterisk MatsK (~matsk@107.80-202-57.nextgentel.com)
23:54.40*** part/#asterisk Romik (~romik@1.fix.netvision.net.il)
23:54.53|Vulture|http://store.ultraspec.us/wildcardte110p.html does that look ligit? its like $60 less than everyone else
23:54.58harryvvanyone seeing any issues with iax.cc?
23:55.02docelmoI would like to use ULAW..  or hell even g729..  but ohh well
23:57.03*** join/#asterisk nextime (~nextime@213-140-6-96.fastres.net)
23:57.26PTG1234anyone know any good graphic artists?
23:57.34harryvvfor what
23:57.55harryvv2s 3d
23:57.57harryvv2d
23:58.19PTG12342d, website design
23:59.30Barmaltzanger: or maybe because it starts to call back before the message is left... That me be why...
23:59.33tzangerBarmal: uh... it polls the given mailbox's NEW directory and if anything is in there it generates the .call file.  it knows nothing about "every other call"
23:59.34Silik0nhow much you wanna pay?
23:59.59tzangerBarmal: depends on how new messages are stored.  as soon as there is something in NEW/ there is the possibility of a callback

Generated by irclog2html.pl by Jeff Waugh - find it at freshmeat.net! Modified by Tim Riker to work with blootbot logs, split per channel, etc.