IRC log for #asterisk on 20090923

00:00.18Kattyi believe it is a candy.
00:01.12eppigyIT IS MY WORD
00:01.16eppigyAND THEY CANNOT HAVE IT
00:01.21Kattyyes, dear. of course.
00:01.55*** join/#asterisk kazaa_lite (n=msaleem@cpc1-lamb4-0-0-cust590.bmly.cable.ntl.com)
00:02.15[8none1]eppigy: Sounds like you have a good legal case
00:02.31Kattyvolunteers to be lawyer.
00:03.06eppigyyes you may have something
00:03.27jayteewow, I remember Kazaa. for a P2P music swapping site that thing was like a hooker with every STD known to man
00:03.43eppigythrowback
00:03.48Kattyeww.
00:03.54radenLMAO
00:04.00radeni hate cisco ASA :(
00:04.01[8none1]As a friend of mine says, digital herpes
00:04.05eppigyraden: why?
00:04.10Kattythe Hiv.
00:04.17jayteeever see Ice Pirates?
00:04.23[8none1]no
00:04.38radenepigy in freaking ASDM i cannot figure out how to route internal lan to outide :(
00:04.45eppigyson
00:04.54eppigyto start close the asdm immediately
00:05.01radenokk
00:05.03jayteereally bad scifi movie with Robert Urich and the girl from Little House on the Prairie. They had Space Herpes in it.
00:05.04eppigygoogle asa command reference
00:05.16eppigythen
00:05.20radeni had da books in front of me
00:05.21Kattychecks to see if that's on netflix
00:05.22eppigywhat is your topology like?
00:05.44eppigyare there any other routers, or do they already have default routes to the asa?
00:05.51Kattywoo! it's on netflix
00:05.55Kattyqueues it up for later tonight
00:06.16jayteeKatty, it's a crap movie
00:06.24[8none1]wow, Robert Urich . . . girl . . Little . . Herpes
00:06.43Kattythat's okay.
00:06.50Kattyi need something to put me sleep anyway.
00:06.52radenepigy , i have a DSL modem bridged on 0/0 a wifi bridged on 0/1 << thoose are to 2 diffrent ISP's
00:07.02jayteenot exactly Dread Pirate Roberts
00:07.05radeni have a procurve 1800 linked to ASA
00:07.21eppigyraden: you would want a default route going out the outside interface
00:07.25Kattyjaytee: i watched a movie about a giant shark and octopus one night.
00:07.36radenepigy 0/1 is up at the moment
00:07.40radenoutside2
00:07.41[8none1]jaytee: You have 6 fingers on your right hand
00:07.51Kattyjaytee: and a cheesy subtitled chinese movie about a chick who lost an arm and had a machine gun graphed on it.
00:08.12eppigyroute <outisde int name> 0.0.0.0 0.0.0.0 <next hop ip> 1
00:08.57eppigyas in the isp router
00:09.02eppigyfor next hop ip
00:10.24radendont i use my wan IP ?
00:10.54eppigyalso if you get both isp interfaces up you can create a policy map to route certain traffic over each connection
00:11.08eppigyraden: well your outisde interface should have your wan ip
00:11.17eppigyfrom dhcp
00:11.52radenwe have static IP's
00:11.57eppigyright
00:12.09eppigyit should still via dhcp
00:12.19eppigyI mean is this consumer internet?
00:13.02radenour wifi our static completly diffrent they authenticate via MAC
00:13.10radenbusiness grade on both
00:13.14eppigyraden: what is the ip address and mask of your outside interface?
00:13.16radenDSL pppoe
00:13.17*** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com)
00:13.28radenwifi
00:13.36raden64.108.141.237
00:13.40*** join/#asterisk ming_zym (n=ming_zym@124.127.101.0)
00:13.41raden255.255.255.128
00:13.54eppigythat is a lot of ip's
00:14.02radenwhat u mean ?
00:14.10radenwe have 1
00:14.13eppigythat is 127 ip's
00:14.16eppigyin that subnet
00:14.20eppigy126
00:14.21eppigyI mean
00:14.23radenyeah
00:14.33radenspread between 2 towns
00:14.46radenDSL is
00:14.50eppigywhat is the gateway ip for that?
00:15.00eppigy64.108.141.129?
00:15.05radengate 64.108.141.128
00:15.15eppigy128 is the subnet number
00:15.30radenthats what they gave eme
00:15.37radensorry keyboar messed up
00:15.56radenDSL
00:16.04raden69.179.99.17
00:16.17raden255.255.255.255 << which does not make sense
00:16.26raden69.29.188.6
00:16.42eppigylets do this
00:16.52eppigyinstead of making everyone in the channel want to kill use
00:16.53eppigyus
00:17.07eppigypastbin the output of show int
00:17.09CareBear\that's not a useful subnet mask
00:17.11eppigyand show route
00:17.50jayteedave's my buddy, I'd never want to kill him
00:18.04*** join/#asterisk darkdrgn2k3 (n=darkdrgn@70.31.3.81)
00:18.21darkdrgn2k3hey guys, whats the major dif between 1.6 and 1.4? is there a big differnce
00:18.52russellb~asterisk16
00:18.53infobotnew features in Asterisk 1.6 are listed at http://svn.digium.com/view/asterisk/trunk/CHANGES?view=markup
00:19.47darkdrgn2k3tips hat
00:19.49darkdrgn2k3thanx
00:20.09CareBear\I didn't get a lot of response to my observation of what might be a bug in 1.6.2.0-rc1: http://pastie.org/626741  the A has been chopped off the Allow header
00:20.56Kattyand i'd tip my HAT
00:20.57russellbalso, diffstat between 1.4 and trunk (dev tree for upcoming 1.6.3):  1020 files changed, 477432 insertions(+), 113817 deletions(-)
00:21.02Kattyimagine THAT
00:21.05russellbIn short, a ton has changed :-)
00:23.47CareBear\Katty : was that a response to me? :) I was hoping for something more technical. Maybe even confirmation that it's a known issue.
00:23.56CareBear\or even
00:23.58CareBear\~tracker
00:24.28CareBear\?
00:25.00russellbCareBear\: that's pretty bizarre.
00:25.15*** join/#asterisk coppice (n=chatzill@157.202.17.210.dyn.pacific.net.hk)
00:25.16russellbrc2 was just released, try updating to that before reporting anything
00:25.21russellbbut the issue doesn't sound familiar ..
00:25.29CareBear\russellb : could it be a freak thing with sip debugging?
00:25.34russellbit's possible
00:25.39russellbdoes it happen every time?
00:25.42russellbor did you see it just once?
00:25.50CareBear\sadly not every time, no
00:25.57russellbmore than once?
00:25.58*** join/#asterisk De_Mon (i=de_mon@fl-69-34-134-91.dhcp.embarqhsd.net)
00:26.19CareBear\I've only noticed it once, the first time was very recently
00:26.31CareBear\let me try to reproduce
00:26.35russellbOkay.
00:26.55*** join/#asterisk coppice (n=chatzill@157.202.17.210.dyn.pacific.net.hk)
00:27.08*** join/#asterisk mrbnet (n=mrbnet@c-75-73-142-28.hsd1.mn.comcast.net)
00:27.22darkdrgn2k3any one know of  good place to get config for a spa3102?
00:27.28darkdrgn2k3?? spa3102
00:28.18mrbnetI have installed asterisk on a debian system. It appears to work but every day or two I need to restart asterisk so calls will come in.
00:28.28mrbnetAny suggestions?
00:28.52jayteeRHEL 5.2 FTW!
00:29.17russellbmrbnet: something is locking up.  Try installing the latest version, and if you still have a problem, join #asterisk-bugs for additional help.
00:29.27russellblatest version from asterisk.org, that is.
00:29.31*** join/#asterisk manxpower (n=EWieling@24.42.221.26)
00:29.33[8none1]mrbnet: I had a similar issue on a debian system with a packaged install.
00:29.43manxpower~answers
00:29.44infobotextra, extra, read all about it, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki:  http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt
00:30.00[8none1]It had something to do with recording ODBC CDR data and would lock channels
00:30.14[8none1]Just install from source and it should work better
00:30.25manxpowerThe only GOOD Asterisk package is an UNINSTALLED Asterisk package
00:31.25mrbnet8none1: I had so much trouble with the packaged version that I installed from source.
00:31.44mrbnet8none1: What version are you running? I am on 1.4.26
00:31.59[8none1]I'm using 1.6 right now
00:32.09[8none1]May not be the same issue since you compiled from source
00:35.56*** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
00:35.56*** join/#asterisk darkdrgn2k3 (n=darkdrgn@70.31.3.81) [NETSPLIT VICTIM]
00:35.56*** join/#asterisk CareBear\ (i=peter@stuge.se) [NETSPLIT VICTIM]
00:35.56*** join/#asterisk Iamnacho (i=Iamnacho@98.186.180.143)
00:35.56*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) [NETSPLIT VICTIM]
00:35.56*** join/#asterisk errr (n=errr@fedora/errr) [NETSPLIT VICTIM]
00:35.56*** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1)
00:35.56*** join/#asterisk evil_gordita (n=evilgord@ip24-254-160-77.rn.hr.cox.net) [NETSPLIT VICTIM]
00:35.56*** join/#asterisk jlnt (n=jlnt@ppp-70-242-97-37.dsl.rcsntx.swbell.net) [NETSPLIT VICTIM]
00:35.56*** join/#asterisk Winkie (n=urmom@ur.fa.gs) [NETSPLIT VICTIM]
00:35.56*** join/#asterisk fuxu2 (i=iconicfl@www.kevinlynn.com) [NETSPLIT VICTIM]
00:35.56*** join/#asterisk thansen (n=thansen@76.27.110.194) [NETSPLIT VICTIM]
00:35.56*** join/#asterisk fskrotzki (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) [NETSPLIT VICTIM]
00:35.56*** join/#asterisk Maliuta (n=scooby@59.167.214.92) [NETSPLIT VICTIM]
00:35.56*** join/#asterisk brut- (n=brut-@h66-173-4-254.mntimn.dedicated.static.tds.net) [NETSPLIT VICTIM]
00:35.56*** join/#asterisk raden_work (n=tanning@69-179-99-17.stat.centurytel.net) [NETSPLIT VICTIM]
00:35.56*** join/#asterisk miloux (n=KVIrc@milu.rit.se) [NETSPLIT VICTIM]
00:35.56*** join/#asterisk many (i=many@argabuthon.ukeer.de)
00:35.56*** join/#asterisk Zhad (n=tom@server30261.uk2net.com) [NETSPLIT VICTIM]
00:35.56*** join/#asterisk lirakis_ (n=lirakis@ool-45760ef7.dyn.optonline.net) [NETSPLIT VICTIM]
00:35.56*** join/#asterisk carrar (i=tim@osburn.com) [NETSPLIT VICTIM]
00:35.56*** join/#asterisk TSM (n=the_soft@fw-lon1.wenn.com) [NETSPLIT VICTIM]
00:35.57*** join/#asterisk luckyaba (n=lucky@ip72-194-215-55.sb.sd.cox.net) [NETSPLIT VICTIM]
00:35.57*** join/#asterisk linagee (n=linagee@about/linux/staff/linagee) [NETSPLIT VICTIM]
00:36.38*** join/#asterisk Firass-z0r (n=asadf@c-67-201-205-34.reshall.wwu.edu) [NETSPLIT VICTIM]
00:36.38*** join/#asterisk mutante (i=mutante@wiktionary/Mutante) [NETSPLIT VICTIM]
00:36.38*** join/#asterisk blackest_mamba (n=blackest@c-71-239-160-143.hsd1.il.comcast.net) [NETSPLIT VICTIM]
00:36.39*** join/#asterisk witchdoc (n=witchdoc@unaffiliated/witchdoc) [NETSPLIT VICTIM]
00:36.39*** join/#asterisk ajohnson (n=ajohnson@65-122-4-130.dia.static.qwest.net) [NETSPLIT VICTIM]
00:36.39*** join/#asterisk t (i=tom@freenode/staff/tomaw)
00:36.39*** join/#asterisk LtScarr (i=benno@palm.hoeg.nl) [NETSPLIT VICTIM]
00:36.39*** join/#asterisk engrxyz (n=sfgsfgs@92.237.248.183) [NETSPLIT VICTIM]
00:36.39*** join/#asterisk eliasp (n=quassel@HSI-KBW-095-208-045-212.hsi5.kabel-badenwuerttemberg.de) [NETSPLIT VICTIM]
00:36.39*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) [NETSPLIT VICTIM]
00:36.39*** join/#asterisk dandre (n=daniel@82.236.48.30) [NETSPLIT VICTIM]
00:36.39*** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) [NETSPLIT VICTIM]
00:36.39*** join/#asterisk gewt (i=b4@you.need.to.stfu-kthx.net) [NETSPLIT VICTIM]
00:36.39*** join/#asterisk Subdolus (i=dexterit@203.82.208.13) [NETSPLIT VICTIM]
00:36.39*** join/#asterisk raspi (i=raspi@62.204.2.215) [NETSPLIT VICTIM]
00:36.39*** join/#asterisk path (i=path@server1.bshellz.net) [NETSPLIT VICTIM]
00:36.39*** join/#asterisk qdk (n=qdk@87.61.141.139)
00:36.39*** join/#asterisk Linuturk (n=linuturk@unaffiliated/linuturk) [NETSPLIT VICTIM]
00:36.39*** join/#asterisk vader-- (n=me@c-68-36-9-8.hsd1.nj.comcast.net) [NETSPLIT VICTIM]
00:36.39*** join/#asterisk zamba (i=marius@flage.org) [NETSPLIT VICTIM]
00:36.39*** join/#asterisk max_ep (n=max@c-98-220-76-124.hsd1.in.comcast.net) [NETSPLIT VICTIM]
00:36.39*** join/#asterisk iksik (i=xk@livedata.pl) [NETSPLIT VICTIM]
00:36.39*** join/#asterisk Cuz (n=plastik@mail.gradeatechs.com) [NETSPLIT VICTIM]
00:36.39*** join/#asterisk creativx (n=creadure@82.134.19.197) [NETSPLIT VICTIM]
00:36.39*** join/#asterisk felipe_ (n=felipe@my.nada.kth.se)
00:36.39*** join/#asterisk Takapa (i=vegard@junior.svanberg.no)
00:36.39*** mode/#asterisk [+o leifmadsen] by irc.freenode.net
00:36.39*** join/#asterisk |omni| (n=rob@67.185.91.139) [NETSPLIT VICTIM]
00:36.39*** join/#asterisk DigitalFlux-AFK (n=DigitalF@unaffiliated/digitalflux) [NETSPLIT VICTIM]
00:36.39*** join/#asterisk fenlander (n=fenlande@82.152.81.57)
00:36.40*** join/#asterisk lirakis (n=lirakis@65.200.191.241) [NETSPLIT VICTIM]
00:36.40*** join/#asterisk Woody2143 (n=Woody214@209.244.4.189) [NETSPLIT VICTIM]
00:36.41*** join/#asterisk pfn (n=pfnguyen@66.245.252.239)
00:36.41*** join/#asterisk p3nguin (i=BuhPjX1J@asterisk-klv5.a2infotech.com) [NETSPLIT VICTIM]
00:36.41*** join/#asterisk docelmo (n=chatzill@65.114.160.138) [NETSPLIT VICTIM]
00:36.41*** join/#asterisk friehmaen (i=freeman@xers.de) [NETSPLIT VICTIM]
00:36.41*** join/#asterisk GNU\colossus (i=colo@truschnigg.info) [NETSPLIT VICTIM]
00:36.41*** join/#asterisk KingOfDos (n=irssi@ht1-ix.kingofdos.net) [NETSPLIT VICTIM]
00:36.41*** join/#asterisk viq (n=viq@unaffiliated/viq) [NETSPLIT VICTIM]
00:36.41*** join/#asterisk Pagautas (n=bigman@ns.voip.ktu.lt)
00:36.41*** join/#asterisk wimt (i=wimt@freenode/staff/wikipedia.wimt) [NETSPLIT VICTIM]
00:36.41*** join/#asterisk MT`AwAy (n=MagicalT@shigoto.ookoo.org) [NETSPLIT VICTIM]
00:36.41*** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) [NETSPLIT VICTIM]
00:36.41*** join/#asterisk razu (n=razu@razu.data.ee) [NETSPLIT VICTIM]
00:36.41*** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) [NETSPLIT VICTIM]
00:36.41*** join/#asterisk lftsy (n=lftsy@ratatouille.leurent.eu) [NETSPLIT VICTIM]
00:36.41*** join/#asterisk ian6 (i=blank@slu.ms) [NETSPLIT VICTIM]
00:36.41*** join/#asterisk MarcWeber (n=marc@88.80.200.63) [NETSPLIT VICTIM]
00:36.41*** join/#asterisk lipek (i=lipek@lipek.pl) [NETSPLIT VICTIM]
00:36.41*** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright)
00:36.41*** join/#asterisk thehar (i=thehar@thehar.xmission.com) [NETSPLIT VICTIM]
00:36.41*** join/#asterisk giovani (n=giovani@unaffiliated/giovani) [NETSPLIT VICTIM]
00:36.41*** join/#asterisk Slashman (n=Slash@ariane.fimasys.com) [NETSPLIT VICTIM]
00:36.41*** join/#asterisk dwayne (n=dwayne@c-71-228-179-90.hsd1.al.comcast.net)
00:36.41*** join/#asterisk Failrar (n=Failrar@2001:470:1f15:316:2a0:d1ff:fe4e:e802) [NETSPLIT VICTIM]
00:36.41*** join/#asterisk BlargMaN00 (n=blarg@12.234.16.130)
00:36.41*** join/#asterisk KyleK (n=Kyle@64.114.61.6)
00:36.41*** join/#asterisk tris (i=tristan@camel.ethereal.net)
00:36.42*** join/#asterisk mbranca (n=matteo@2001:1418:130:0:21e:8cff:fe51:5b05)
00:36.42*** join/#asterisk fnordus (n=dnall@70.70.0.215) [NETSPLIT VICTIM]
00:36.42*** join/#asterisk Kobaz (n=kobaz@its.kobaz.net)
00:36.42*** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org) [NETSPLIT VICTIM]
00:36.42*** join/#asterisk tompaw (n=tompaw@91.121.184.63) [NETSPLIT VICTIM]
00:36.42*** join/#asterisk mace (n=mace@debian/developer/mace) [NETSPLIT VICTIM]
00:36.42*** join/#asterisk bpgoldsb (n=bpgoldsb@209.208.68.1) [NETSPLIT VICTIM]
00:36.42*** join/#asterisk keith4 (n=keith@unaffiliated/keith4)
00:36.42*** join/#asterisk eharris (i=eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) [NETSPLIT VICTIM]
00:36.42*** join/#asterisk yang (i=yang@freenode/sponsor/cacert.assurer.yang)
00:36.42*** join/#asterisk mangala (i=mangala@208.68.95.138) [NETSPLIT VICTIM]
00:37.07[8none1]Katty: I just stumbled upon this from your comment about the move you saw : http://craphound.com/images/bearsharktopus-30363-1253244193-27.jpg
00:37.07*** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
00:37.07darkdrgn2k3welcome back
00:37.09*** join/#asterisk Iamnach0 (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net)
00:37.20Kattyhttp://www.imdb.com/title/tt1350498/ <- Movie.
00:38.22KattyFrom the movie: http://www.dreadcentral.com/img/reviews/megashark1b.jpg
00:39.22[8none1]I saw that in the trailer :D
00:39.35[8none1]funny
00:39.37*** join/#asterisk memph (n=memph@sd1438.sivit.org)
00:39.54KattyMegaladon is a very fascinating creature.
00:40.39Kattyhttp://upload.wikimedia.org/wikipedia/commons/thumb/0/07/Megalodon_scale1.png/800px-Megalodon_scale1.png <- for size reference.
00:41.30Kattyit oculd eat a Blue Whale in a couple chomps.
00:43.10*** part/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
00:44.16Kattyand the Plesiosaur isn't extinct either.
00:44.25KattyOne was found off the coast of Japan in 1977, rotting.
00:45.48Kattyhttp://www.discoverynews.us/DISCOVERY%20MUSEUM/CreaturesFromTheDeep/CreaturesIMAGES/Plesiosaur_4_large.jpg <- Plesiosaur
00:46.23coppicethat report from japan was bogus
00:46.54kevinh90howdy
00:46.55Kattylink?
00:47.37coppicetry looking up anything about plesiosaur and japan. the sensationalist ones says its a sea monster. most say it was a basking shark
00:47.59Kattygoogles.
00:51.12*** join/#asterisk mumtazah (n=mumtazah@142.78.48.60.wmu01-home.tm.net.my)
00:53.20*** join/#asterisk ingenius (n=alektro@host172.190-231-93.telecom.net.ar)
00:54.42*** join/#asterisk jmacz (n=jmacz@190.144.75.22)
00:56.11*** join/#asterisk micols (n=mio@rlogin.dk)
00:56.52*** join/#asterisk bradleyprice86 (n=bradleyp@75-120-244-130.dyn.centurytel.net)
00:58.12eppigyNEIN
00:58.33*** join/#asterisk shmaltz (n=chatzill@mail.dmaven.com)
00:59.42leifmadsenHI!
00:59.56carrarHI!!
01:01.05*** join/#asterisk phunyguy (n=phunyguy@h69-130-65-171.kgldga.dsl.dynamic.tds.net)
01:02.25shmaltzhi!!!!!!!!!!!
01:02.58[8none1]|-| !
01:04.21shmaltzhow come I can see calls being made to sip/5371 and I don't have a sip/5371?
01:09.58*** join/#asterisk jpcansa (n=jpbenavi@201.200.55.138)
01:10.36shmaltz~anyone
01:10.37infobotrumour has it, anyone is Instead of looking for mentors from specific projects here, try the project's IRC channel, likely on this server as well.
01:10.49shmaltz~sleep
01:10.50infobotwell, sleep is overrated, and a poor substitute for caffeine.
01:11.53jpcansadoes anybody has an example of code to store a number dialed byt a called on an IVR and then dial it thru a different gateway??
01:13.04shmaltzjpcansa, example
01:14.22jpcansashmaltz, someone calls to my IVR from pstn and dials a number, i want * to dial that number thru VoIP gateway
01:14.42shmaltzjpcansa, then just use app_dial
01:14.49shmaltzsomething along these lines:
01:14.56shmaltzcaller is dumped in s,1
01:15.39shmaltzs,n,Read(NUMTODIAL|plsenternum)
01:15.40shmaltzs,n,Dial(Sip/Gatway/${NUMTODIAL})
01:16.44jpcansagood, let me try that
01:17.36jpcansashmaltz: "plsenternum" is a recording?
01:17.54*** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose)
01:18.00shmaltzjpcansa, if you record it yes
01:18.22shmaltzjpcansa, please check docs for exact args for each app
01:18.44jpcansayeah i got it
01:18.55jpcansathanks for your help
01:25.36shmaltzanyone here as bored as I am?
01:26.15shmaltz~bored
01:26.16infobotLa ... lalalala ... beer!
01:26.22shmaltz~beer
01:26.23infobotACTION has disconnected (Read error: 99 (Connection reset by beer))
01:26.33shmaltz~reset
01:26.34infobotwell, reset is not influencing the message about the kernel on hyper terminal
01:26.44shmaltz~sex
01:26.44infobot[~sex] updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; emerge --oneshot condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; emerge -C condom; make clean; sleep
01:27.40shmaltz~prenant
01:27.45shmaltz~nyc
01:27.46infobotmethinks nyc is where the secret organization l.a.u.r.a. is headquartered (i'm on to you tzu!)
01:27.57shmaltz~ny
01:27.58infobotny is, like, a place where they make the best pizza, the best hot dogs, and the nicest hookers
01:28.00shmaltz~nj
01:28.01infobothmm... nj is home to the sopranos.  Fogedaboudit!
01:28.19shmaltz~wa
01:28.20infobotwha? what in $DIETY's name are you talking about? wtf?
01:28.28shmaltz~uk
01:28.29infobotwell, uk is a place where they don't know english to well
01:28.36shmaltz~paris
01:28.37infobot[paris] Capital of France, or Paris AZ, MI, NE, OR, WA.
01:28.46shmaltz~AZ
01:28.47infobot[az] Azerbaijan
01:28.55shmaltz~obama
01:28.56infobotwell, obama is the 44th and current president of the United States of America
01:29.03shmaltz~clinton
01:29.04infobothmm... clinton is the best president we've had since george bush, dammit! or the only president we've had since george bush, dammit! or the Pants Dropper in Chief or the guy blowing away small countries or the president whose staff continues to classify cryptographic software as munitions or a shitty human being or stupid or in control of nuclear weapons or a psychopath, or clit-ton
01:29.07manxpowerThere's nothing wrong with playing with your bot as long as you do it in private and wash your hands after.
01:29.10L2Logic~pastebin
01:29.10infobot[~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
01:29.28manxpoweryou can use /msg to talk to infobot too
01:29.50shmaltzoh so some ppl are alive
01:30.01shmaltzyes but it's way more fun to talk in public
01:30.31shmaltz~rush limbo
01:30.32infobotACTION sees limbo is dawdling and runs over and pushes limbo out the door
01:30.51shmaltz~lol
01:30.51infobotlol is probably stands for Laughs Out Loud. It is grammatically incorrect to use LOL in the first person; use 'heh' or 'haha' instead. If you want to use LOL, do '/me lol' instead.
01:31.03shmaltz~biden
01:31.23shmaltz~iraq
01:31.24infobot[iraq] a country
01:31.24L2Logiccheck this out...     it's probably do what you're wanting to do to specificy a specific CID for a DISA application  (and record the call)   http://pastebin.com/d6ea6194
01:31.41L2Logicit'll vs it's
01:32.23L2Logicwe've had a president since Reagan ?
01:33.00shmaltzL2Logic, :)
01:33.11shmaltzwell we don't currently have one that is sure
01:33.27*** join/#asterisk s0lid (n=s0lid@112.200.180.234)
01:35.23L2Logici don't think Arnold S. gets as much airtime as Mr. O....  even with terminator reruns
01:36.39L2Logici remember when I was a kid..  and we had three channels..   "the president is on, every channel"..  now it's, "the president's on..   all the time..."
01:37.31L2Logicnext thing we hear...   "We interrupt this program to share with you that Mr. Obama will not be making an appearance on the news today"
01:37.53*** join/#asterisk scalex000 (n=chatzill@95puntacana02.codetel.net.do)
01:41.31shmaltz:P
01:45.14*** join/#asterisk Carlos_PHX (n=carlos@ip68-108-193-174.ph.ph.cox.net)
01:45.34*** join/#asterisk voipmonk (n=voipmonk@69.172.93.45)
01:47.30*** join/#asterisk TJNII_WORK (n=TJNII@207.189.199.62)
01:47.35radennight Katty
01:47.39*** part/#asterisk Carlos_PHX (n=carlos@ip68-108-193-174.ph.ph.cox.net)
01:50.56*** join/#asterisk Iamnacho (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net)
01:56.19*** join/#asterisk OrNix (n=ornix@l151-249-47.static.cn.ru)
02:00.52*** part/#asterisk manxpower (n=EWieling@24.42.221.26)
02:12.59*** join/#asterisk [8none1] (n=[8none1]@c-76-22-141-39.hsd1.tn.comcast.net)
02:14.21*** join/#asterisk korolev (i=korolev@c-75-74-122-15.hsd1.fl.comcast.net)
02:23.23*** join/#asterisk ingenius (n=alektro@186.136.12.198)
02:26.29*** join/#asterisk sahX (n=Bawbatos@99-40-7-14.lightspeed.sntcca.sbcglobal.net)
02:41.56*** join/#asterisk denon (i=denon@sassinak.net)
02:41.56*** mode/#asterisk [+o denon] by ChanServ
02:48.14*** join/#asterisk s0lid (n=s0lid@119.92.130.74)
02:52.37*** join/#asterisk geneticx (n=geneticx@adsl-2-59-65.mia.bellsouth.net)
03:02.14*** join/#asterisk bradleyprice86 (n=bradleyp@75-120-244-130.dyn.centurytel.net)
03:09.42*** join/#asterisk s0lid (n=s0lid@119.92.130.74)
03:15.06*** join/#asterisk parker (n=ti@189.24.135.78)
03:16.09p3nguinHere's a registration debug from a problematic softphone:  http://pastebin.ca/1575494   Does everything look okay/normal as far as the registration goes?
03:17.44[TK]D-Fenderp3nguin: Yes
03:18.44p3nguinIf it registered correctly, I don't understand why it doesn't make calls.
03:18.50p3nguinIt receives them.
03:22.22*** join/#asterisk dssman (n=dssman@CPE001d7e602900-CM0011aec52a9c.cpe.net.cable.rogers.com)
03:23.24dssmanHey all, have a quick question... if I followme an exten is there any way to make it blind to the call creator... IE only play back a ringing tone rather then "please hold while we try to locate the person you are calling [moh]"
03:24.21[TK]D-Fenderp3nguin: Registeration has jak-all to do with placing a call
03:30.32p3nguinDoes it not show that SIP packets are able to reach the server from the client?
03:31.23[TK]D-Fenderp3nguin: Looks lie so far.
03:31.39[TK]D-Fenderp3nguin: of course it could be using split info between registrar and call servers
03:31.48[TK]D-Fenderp3nguin: amongst a ton of other possibilities.
03:32.16p3nguinI'm so puzzled about it.
03:32.21[TK]D-Fenderp3nguin: I don't know twinkle specifically
03:32.33[TK]D-Fenderp3nguin: try another soft-phone in the meanwhile
03:32.56p3nguinTried ekiga, too.  It's in the same situation.  Can register and receive calls, can't make calls.
03:34.16[TK]D-Fenderp3nguin: You get NO packets on call attempts
03:34.18[TK]D-Fender?
03:35.03p3nguincorrect
03:36.04p3nguinWhen I showed you guys earlier what debug there was when he tried making a call, the info was from qualify=yes.  I turned that off, and no debug info came out when he tried making a call.
03:36.36parkerhy, i'm newer with asterisk and i've have one problem, i'm using asterisk 1.4 with dahdi 2.2 and one interface E1 (INTELBRAS), i have one queue named RECEPCAO, when a call come througth E1 link the call stay mute and in 5 seconds drop it, but if a dial to the same queue by internal side, everything is OK, can anyone help me?
03:38.38[TK]D-Fenderp3nguin: this tells me that yrou * side probably isn't properly forwarded
03:39.25p3nguinThe particular Linux computer that these two softphones are on has a direct connection to the internet, and it acts as a gateway for one Windows computer.  It's using iptables/masquerade to create NAT for the other computer.  Running zoiper on the Windows computer which connects through the Linux box works correctly.  Zoiper makes and receives calls.
03:40.26p3nguinThat makes me think that * is correctly communicating with that remote network.
03:40.40[TK]D-Fenderp3nguin: I thinking your * side is bad.  not the client
03:40.57[TK]D-Fenderp3nguin: And what are you doing with 2 softphones on 1 PC?
03:41.07[TK]D-Fenderp3nguin: that is in itself a disaster waiting to happen
03:41.12p3nguinTrying to solve this probably, obviously.
03:41.14[TK]D-Fenderp3nguin: as they fight for SIP portm etc
03:41.24[TK]D-Fenderp3nguin: You are polluting your test
03:41.52p3nguinYou suggested another SIP phone.  Why would you suggest that if it pollutes the test?
03:42.23p3nguinWe obviously have no reason to run twinkly and ekiga at the same time.  That would be bad.  And useless.
03:42.27[TK]D-Fenderp3nguin: Hold on.. no simultaneous?
03:42.32p3nguinno no
03:43.00p3nguinOnly for testing.  DIdn't even run zoiper on the other machine until we went troubleshooting.
03:43.06p3nguinnever two at once
03:44.29parkerhy, i'm newer with asterisk and i've have one problem, i'm using asterisk 1.4 with dahdi 2.2 and one interface E1 (INTELBRAS), i have one queue named RECEPCAO, when a call come througth E1 link the call stay mute and in 5 seconds drop it, but if a dial to the same queue by internal side, everything is OK, can anyone help me?
03:44.34p3nguinI also have another Windows client on a completely different network behind NAT which runs zoiper successfully.  Is it possible that zoiper is more "forgiving" about some screwed up setting that I might have?
03:46.12p3nguinAnd could a debug of a good call using zoiper help to diagnose anything?  I don't recall if I provided that debug earlier or not.  I just know there was no debug when trying to call out using neither twinkle nor ekiga.
03:46.21*** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com)
03:46.37[TK]D-Fenderp3nguin: what do the  softphoens tell you when you try to dial?
03:47.38p3nguinThe status on the window shows that it is trying to place a call, but then about 30 seconds later it says that it timed out.
03:48.03*** join/#asterisk Cresl1n_ (n=matt@asterisk/libpri-and-libss7-expert/Cresl1n)
03:48.03*** mode/#asterisk [+o Cresl1n_] by ChanServ
03:49.18[TK]D-Fenderp3nguin: Still screams networking error
03:49.31*** join/#asterisk brunner (n=chris@68.119.87.106)
03:49.40brunnerasterisk is working well with 200 callers =D
03:50.22p3nguinI'll check anything you want checked if you have an idea where to look or what to look for.
03:51.27p3nguinI wonder if it's possible to run zoiper through wine.  That could be an interesting test.
03:52.49*** join/#asterisk bradleyprice86 (n=bradleyp@75-120-244-130.dyn.centurytel.net)
04:00.56kevinh90hi
04:02.27*** part/#asterisk Cresl1n_ (n=matt@asterisk/libpri-and-libss7-expert/Cresl1n)
04:29.31*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)
04:33.03*** join/#asterisk xpot (n=james@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net)
04:34.16*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
04:41.58*** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano)
04:47.46CareBear\Where does asterisk get the username that it puts into From: when sending an INVITE ?
04:49.39[TK]D-FenderCareBear\: what do you see in there?
04:49.44CareBear\asterisk
04:51.01CareBear\am using asterisk as client
04:52.48[TK]D-FenderCareBear\: pastebin a complete call attempt with SIP debug enabled along with your peer config masing only passwords
04:52.53[TK]D-Fender~pb
04:53.03infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org ,  http://bin.cakephp.org/ , or apt-get install pastebinit
04:58.16CareBear\http://pastie.org/private/sjoq4a3eszwotwv2ja5g
04:59.28CareBear\is it a bad idea to use the same hostname both on client and on server - although the client registers with the server and is reachable through there with the hostname?
05:00.17*** part/#asterisk mumtazah (n=mumtazah@142.78.48.60.wmu01-home.tm.net.my)
05:00.50[TK]D-FenderCareBear\: You are showing the call being received.  I want to see the call being SENT
05:01.21CareBear\from .9 is what is SENT
05:01.26CareBear\the server is .7
05:02.25[TK]D-FenderCareBear\: You aer showing me debug from the receiving server.  I want to see the sender's configs & debug
05:03.31CareBear\the debug from the sender looks just the same though..
05:03.46CareBear\(as for what goes onto the wire)
05:07.24CareBear\http://pastie.org/private/u3jkoyie9ahvicseqyhtq has it, along with the extension I dial
05:08.00CareBear\a new call-id, but otherwise same
05:12.49[TK]D-FenderCareBear\: exten => tablet,1,Dial(SIP/tablet@stuge.se,15)
05:13.11[TK]D-FenderCareBear\: You are directly dialing the host.  This means you are not using your peer, and that means there is no auth attached
05:13.11CareBear\yes?
05:13.16[TK]D-FenderCareBear\: And also no identity
05:13.24[TK]D-FenderCareBear\: hence the "asterisk"
05:13.49[TK]D-FenderCareBear\: Dial(SIP/apeernamewithusernameetc/numbertodial)
05:14.07CareBear\but that just moves the problem one step, doesn't it?
05:14.20[TK]D-FenderCareBear\: like Dial(SIP/peter/tablet)
05:14.35[TK]D-FenderCareBear\: No, it should not show "asterisk"
05:15.06[TK]D-FenderCareBear\: Of course it'd also be good to see the callerid of the originating call (before you call from that server to the other
05:15.20Carlos_TicoTK now i am getting this ...[Sep 23 00:14:52] NOTICE[159]: chan_sip.c:14035 handle_request_invite: Call from '6002' to extension '4549142' rejected because extension not found.
05:15.58Carlos_Tico4549142 is the number i want to dial out from the sp3k
05:16.00[TK]D-FenderCarlos_Tico: And it means just what it says
05:16.21CareBear\D-Fender : asterisk is the originating call, I set "Peter Stuge" in oss.conf
05:16.33CareBear\okey, let's see.
05:16.42[TK]D-FenderCarlos_Tico: this is * refusing the call you are placing.  this has nothing to do with calling OUT.  * is refusing the call IN
05:17.49*** join/#asterisk fiddur (n=fiddur@dhcp08.textalk.com)
05:20.30CareBear\.9 sends peter@ to .7
05:20.36CareBear\.7 says Found peer 'peter' for 'peter' from 213.88.146.9:5060
05:21.01CareBear\followed by 'Failed to authenticate device "Peter Stuge" <sip:peter@...'
05:23.49*** join/#asterisk blkry (n=chatzill@96.37.27.72)
05:24.06*** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1)
05:25.49CareBear\D-Fender : I get the idea. I have to dial out through where I am known as peter@stuge.se before From will actually show that.
05:26.41CareBear\Makes good sense for a PBX, but maybe slightly less for a softphone, which is what I'm using asterisk on my laptop as. :)
05:27.09[TK]D-FenderCareBear\: add these to your peer : fromuser=peter , sendrpid=yes , trustrpid= yes
05:27.20[TK]D-FenderCareBear\: add those last 2 lines to the other side
05:28.00CareBear\still get the forbidden
05:28.18CareBear\the server has type=friend for this registration btw
05:28.33CareBear\would that be bad?
05:31.19CareBear\this looks incomplete: Remote-Party-ID: "Peter Stuge" <sip:@stuge.se>;privacy=off;screen=no
05:31.47[TK]D-FenderCareBear\: put "username=peter for the other side
05:32.37CareBear\no difference - because this header is coming from my laptop
05:32.41CareBear\maybe?
05:33.03CareBear\(there I already have defaultuser=)
05:33.10CareBear\=peter
05:33.37[TK]D-Fenderbad parm... don't use it
05:33.46[TK]D-Fenderok, all I have time for for tonight...
05:33.52[TK]D-Fenderkeep at it.. you';re almost there
05:34.00CareBear\thanks for the help so far. :)
05:42.55*** join/#asterisk tjz (n=tjz@bb220-255-158-226.singnet.com.sg)
05:48.42*** join/#asterisk scardinal (n=supreme@0905ds1-rdo.0.fullrate.dk)
05:55.35*** join/#asterisk maour_ (n=gnu@unaffiliated/maour)
05:58.40*** join/#asterisk |Cybex| (n=John@atwork-26.r-212.178.82.atwork.nl)
05:59.37*** join/#asterisk flohack (n=fhackenb@84.115.131.198)
06:11.43*** join/#asterisk kc2tnk (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com)
06:12.06*** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com)
06:26.00*** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com)
06:27.23*** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com)
06:32.54*** join/#asterisk xrmx__ (n=rm@host103-251-dynamic.15-87-r.retail.telecomitalia.it)
06:35.22*** join/#asterisk brut- (n=brut-@h66-173-4-254.mntimn.dedicated.static.tds.net)
06:36.00*** join/#asterisk TommyBotten (n=tommy@217-14-12-26-dhcp-osl.bbse.no)
06:41.41*** join/#asterisk CrawZ (n=chrisc@129.70.96.58.static.exetel.com.au)
06:43.20*** join/#asterisk maour (n=gnu@unaffiliated/maour)
06:46.58*** join/#asterisk grEvenX (n=even@apb99b.ip.ssc.net)
06:53.18*** join/#asterisk mikkel (n=mikkel@130.226.36.170)
06:56.00*** join/#asterisk brut- (n=brut-@h66-173-4-254.mntimn.dedicated.static.tds.net)
06:59.15*** join/#asterisk mintos (n=mvaliyav@nat/redhat-us/x-rytiklkspkqhofhs)
06:59.23*** join/#asterisk maour (n=gnu@unaffiliated/maour)
06:59.53*** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com)
07:07.45*** join/#asterisk MaliutaLap (n=biteme@203.171.192.191)
07:16.48*** join/#asterisk lordmortis (n=lordmort@203-8-160-250.secure.com.au)
07:18.04*** join/#asterisk bluOxigen (n=asad@static-host119-73-71-143.link.net.pk)
07:19.57*** join/#asterisk wildzero-cw (n=chatzill@p50997436.dip0.t-ipconnect.de)
07:20.19*** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:74b4:c5ae:25f3:6a02)
07:29.41ThoMehello
07:29.47ThoMegood morning.
07:30.15*** join/#asterisk tamiel (n=tamiel@213.30.183.226)
07:38.59*** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net)
07:39.54*** join/#asterisk JT (n=j@unaffiliated/jt)
07:40.32*** join/#asterisk unasi7 (n=unasi7@84-72-46-20.dclient.hispeed.ch)
07:48.36*** join/#asterisk ice_croft (n=nolan@mail.kubkurort.ru)
07:48.40*** part/#asterisk ice_croft (n=nolan@mail.kubkurort.ru)
07:50.20*** join/#asterisk MaliutaLap (n=biteme@203.171.195.86)
07:52.44*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
07:52.53*** join/#asterisk war9407 (i=war@liquidswords.org)
08:00.29*** join/#asterisk Dovid (n=annon@213.8.121.90)
08:01.12ThoMetzafrir_laptop: hello.
08:01.21tzafrir_laptopho
08:01.25ThoMetzafrir_laptop: can you help me with dahdi?
08:01.33tzafrir_laptophopefully :-)
08:01.38ThoMei would like use chanspy (i need dahdi for this?)
08:02.12tzafrir_laptopI don't think so
08:02.13ThoMei have this problem, when i use the function ",w" (whipser) i can only hearing but not speak.
08:02.18ThoMenow I have found this: https://issues.asterisk.org/view.php?id=15660
08:02.35ThoMecan you help me with get this branches?
08:02.44ThoMei havent found a doku
08:04.28ThoMeoh ok http://www.digium.com/en/docs/asterisk_handbook/downloading_compiling.html
08:04.35ThoMesvn co http://svn.digium.com/svn/asterisk/trunk asterisk <<is this ok tzafrir_laptop ?
08:06.16tzafrir_laptopBTW: (unrelated) any comments on http://docs.tzafrir.org.il/dahdi-linux/#_live_install ?
08:07.29tzafrir_laptopThoMe, it's fixed in 1.4 , so by now latest 1.4 tarball should include it
08:08.06tzafrir_laptopThat handbook page needs some updating
08:08.16ThoMehm. have 1.4.26.2
08:08.17ThoMehmm.
08:08.19tzafrir_laptopWon't actually work as-is
08:11.30*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
08:13.03*** join/#asterisk bluOxigen (n=asad@static-host119-73-71-143.link.net.pk)
08:23.54ThoMetzafrir_laptop: hmm.
08:23.57*** join/#asterisk thansen (n=thansen@76.27.110.194)
08:23.57ThoMeproblem fixed
08:24.02ThoMenot in 1.4.26.2 :_(
08:24.10ThoMenow i have
08:24.10ThoMeAsterisk SVN-branch-1.4-r219816 built by root @ asterisk01 on a i686 running Linux on 2009-09-23 08:16:46 UTC
08:24.45tzafrir_laptopit was commited on late August
08:27.28ThoMehmm. and version 1.4.26.2 ?
08:29.51ThoMetzafrir_laptop: is it posible in a macro when hangup this includet:
08:29.52ThoMeexten => h,1,NoOp(Coaching Start)
08:29.52ThoMeexten => h,n,Set(DB(coach-eins/${nebenstelle})=nein)
08:29.53ThoMe?
08:29.56ThoMea h-extention?
08:31.37*** join/#asterisk wonderworld (n=w@ip-62-143-22-226.unitymediagroup.de)
08:35.17cjkhi, i have not voice on local channels, except if i do something like playpack(beep) which answers the calls before dialing another party. any idea why? this problem only exists if all call legs are on SIP. If there is some ZAP in the middle it works perfectly
08:48.03*** join/#asterisk Maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com)
08:49.06*** join/#asterisk pif (n=ldm@zenon.apartia.fr)
08:53.35*** join/#asterisk jgoo (n=lxuser@gw1.mycosmos.gr)
08:53.42ThoMeemmm
08:53.57ThoMeis it posible ${SIP_HEADER(TO)} <<here extract only digits
08:53.58ThoMe?
08:54.40jgooHej people - I was having an issue with reload commands not working, so following a google find, i did asterisk -vvvvvvvvc - it loads instantly - however reload command sticks at Reloading module 'chan_sip.so' (...) Parsing... manager*.confs
08:55.25kaldemarThoMe: use func CUT
08:55.45ThoMekaldemar: hm.
08:55.56jgoohttp://pastebin.me/25a3f01cca2f883313c8f80cb458dd3a
08:57.58ThoMekaldemar: emm
08:58.00ThoMekaldemar: I use
08:58.01ThoMe<PROTECTED>
08:58.01ThoMe<PROTECTED>
08:58.03ThoMe<PROTECTED>
08:58.11ThoMeis it posible in one step extract the number?
08:59.57cjkhi, i have a call coming over ZAP to asterisk1 then it goes over SIP to asterisk2 and then it goes back over SIP to asterisk1 where I execute the echo application. if i do not answer the call on asterisk2, before going back to asterisk1 i have no voice at all. any idea?
09:00.44kaldemarThoMe: combine those lines
09:01.36kaldemaror maybe you can't, since CUT wants a variable name
09:03.54*** join/#asterisk Chinorro (n=Chino@19.226.117.91.dynamic.mundo-r.com)
09:05.19ThoMekaldemar: ah, works fine
09:05.19ThoMe<PROTECTED>
09:10.24*** join/#asterisk TheOpenSourcerer (n=TheOpenS@81.178.65.1)
09:10.33*** part/#asterisk TheOpenSourcerer (n=TheOpenS@81.178.65.1)
09:17.58*** join/#asterisk dymaxion (n=gohsac@host86-172-50-54.range86-172.btcentralplus.com)
09:19.56*** join/#asterisk Greek-B0y (n=greek@41.188.154.137)
09:33.48*** join/#asterisk MaliutaLap (n=biteme@203.171.195.204)
09:36.42*** join/#asterisk flohack (n=fhackenb@chello213047219150.surfer.at)
09:57.49*** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy)
10:02.26ThoMehello?
10:03.42*** join/#asterisk so_solid_moo (n=nmoo@fsf/member/alexhudson)
10:04.54*** part/#asterisk so_solid_moo (n=nmoo@fsf/member/alexhudson)
10:05.58*** join/#asterisk so_solid_moo (n=nmoo@fsf/member/alexhudson)
10:06.13so_solid_mooHi; could anyone tell me how long it's supposed to take an Astribank to load its firmware?
10:06.59tzafrir_laptopA few seconds
10:07.06so_solid_moooh dear :S
10:07.27tzafrir_laptopwhat's the output of dahdi_hardware   #?
10:07.55so_solid_mooah; I don't have dahdi installed - using zaptel
10:08.32so_solid_mooit's seeing a "e4e4:1160 Astribank-modular no-firmware" and the xpp_fxloader script is writing the firmware to the right device
10:17.52tzafrir_laptopso_solid_moo, do you have fxload installed?
10:18.08so_solid_mootzafrir_laptop: yup
10:18.42tzafrir_laptophttp://docs.tzafrir.org.il/dahdi-tools/README.Astribank.html#_lsusb_test
10:18.54tzafrir_laptopyou installed it after you connected the astribank?
10:19.22so_solid_moono, it was installed from the start
10:19.35so_solid_moothe usb id is as above - e4e4:1160
10:19.52so_solid_mooI have fpga_load available too
10:19.57tzafrir_laptopWhat system is it? What linux distro?
10:20.06tzafrir_laptopfpga_load is not relevant to that
10:20.17so_solid_mooOh, sorry - it's debian lenny
10:20.40tzafrir_laptopdebian lenny packages?
10:20.45so_solid_mooyeah
10:20.55tzafrir_laptop(zaptel 1.4.12 ?)
10:21.01tzafrir_laptoptwo issues:
10:21.03so_solid_moo1.4.11 even I think
10:21.21tzafrir_laptop1. that version does not support those newer astribanks
10:21.56tzafrir_laptop2. those packages don't include the firmware . the firmware is not DFSG and thus belongs in non-free (in a separate source package)
10:22.21tzafrir_laptoptry the packages from:
10:22.33ThoMetzafrir_laptop: hello.
10:22.52tzafrir_laptopThoMe, hi
10:23.13tzafrir_laptopso_solid_moo, hmm.. I see I didn't yet add them to my main "unofficial" source
10:23.19so_solid_moo:)
10:23.21ThoMetzafrir_laptop: one question: when i use the dial app and i have a call with a friend (connected) is it posible when i press example "8" then hangup by my friend and run a macro X ?
10:23.28so_solid_mootzafrir_laptop: I'd figured out 2., but not 1. - my fault, sorry
10:25.01tzafrir_laptoptry packages (zaptel , zaptel-firmware) from:   deb http://updates.xorcom.com/pkg-voip/repo-i386-lenny/ unstable main
10:28.04*** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com)
10:28.56ThoMetzafrir_laptop: is it posible press a digit when i have a call
10:29.00*** join/#asterisk Moz (n=me@81.179.238.144)
10:29.16ThoMeand when i press example the digit 8 then run a macro
10:29.17ThoMe?
10:29.56tzafrir_laptopThoMe, look into feature codes (feature.conf)
10:30.17so_solid_mootzafrir_laptop: thanks, sadly no difference - I just get lots of '....' on the console :S
10:31.16ThoMetzafrir_laptop: yes but is it posible to define a own code with a macro?
10:31.26tzafrir_laptopdpkg -l zaptel | grep ^i
10:31.32tzafrir_laptopsorry
10:31.36tzafrir_laptopdpkg -l zaptel\* | grep ^i
10:32.24tzafrir_laptopactually:
10:32.32tzafrir_laptopdpkg -l zaptel zaptel-firmware | grep ^i
10:33.05ThoMehow i can reload the features.conf?
10:33.05so_solid_moo1:1.4.12.9.svn.r4649-0.7280 and 1:1.4.12.9.svn.r4649~dfsg-0.7280
10:33.29kaldemarThoMe: features reload
10:33.40ThoMeNo such command 'features reload' (type 'help features reload' for other possible commands)
10:33.45ThoMeNo such command 'feature reload' (type 'help feature reload' for other possible commands)
10:34.42ThoMeah
10:34.43ThoMeDynamic Feature           Default Current
10:34.43ThoMe---------------           ------- -------
10:34.43ThoMetestfeature               no def  #9
10:36.03ThoMehave
10:36.04ThoMetestfeature => #9,self,Playback,tt-monkeys
10:36.13ThoMebut but no playback tt-monkes
10:36.14ThoMehmm
10:40.27ThoMeSet(DYNAMIC_FEATURES=testfeature)
10:40.27ThoMetestfeature => #9,callee,Playback,tt-monkeys
10:40.30ThoMeis it correct?
10:40.39ThoMeaeh testfeature => #9,self,Playback,tt-monkeys
10:42.47*** join/#asterisk ccesario (n=ccesario@189-19-6-236.dsl.telesp.net.br)
10:43.58*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
10:57.35ThoMehow i can hangup the calling party but not my line
10:58.03ThoMeI would like jump to X (example meetmeroom) when i hangup the callinguser
10:59.56nextimeis there a way to get the current asterisk version ( 1.4 or 1.6 ) from the manager interface?
11:00.37troffaskycore show version
11:00.51troffaskyoh, manager interface
11:00.53troffaskynm
11:01.42*** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com)
11:02.20*** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com)
11:02.57nextimeor from the dialplan, so from the manager i can use EXEC or GETVAR
11:04.26ThoMeemm
11:05.59ThoMeemm, have a problem: i would like when i call with user X and I press the "1" hangup and jump to a meetme-room
11:06.04ThoMei try: http://paste.keks.be/2052/txt
11:06.07*** join/#asterisk garymc (n=garymc@host86-163-43-91.range86-163.btcentralplus.com)
11:07.38ThoMebut when press "1" jump to  my macro and hangup() then i cant then jump to my room
11:08.12ThoMeany ideas?
11:08.30ThoMeI would like only hangup the callee not my line
11:08.35*** join/#asterisk m0bius (n=mobius@mailexchange.realize.gr)
11:13.16kaldemarnextime: manager command CoreSettings will show the version
11:14.06kaldemarnextime: func VERSION will return asterisk version in dialplan
11:15.38ThoMewhen i use dial() and go to my cli
11:15.40ThoMeasterisk01*CLI> soft hangup SIP/
11:15.40ThoMeSIP/gw_7_asterisk01-08885300  SIP/53002-08824e18
11:15.49ThoMehow i can get this SIP/gw_7_asterisk01-08885300 ?
11:16.01ThoMe$CHANNEL is SIP/53002-08824e18
11:16.04ThoMeAND SIP/gw_7_asterisk01-08885300 ?
11:17.10kaldemarget where?
11:18.02ThoMewhere, yes
11:18.17nextimekaldemar : thanks
11:18.53kaldemarThoMe: where are you trying to get it?
11:19.10ThoMekaldemar: hm, i would like hangup only the callee channel
11:19.10nextimekaldemar : are they present also on 1.4?
11:19.19ThoMeis it posible?
11:19.42kaldemarThoMe: core show application SoftHangup
11:20.09ThoMekaldemar: jep, but when i Use SoftHangup how i can hangup the callee??
11:21.15kaldemarThoMe: SoftHangup takes the channel as a parameter, as you should understand from the application doc
11:22.26kaldemarget the channel name by some other means, for example using core show channels concise
11:22.39ThoMekaldemar: yes  but kaldemar i would like hangup the callee
11:22.50ThoMewhen i use: core show channels
11:22.53ThoMeSIP/gw_7_asterisk01- (None)               Up      AppDial((Outgoing Line))
11:22.54ThoMeSIP/53022-08824e18   s@macro-dial-gateway Up      Dial(SIP/461@gw_7_asterisk01|1
11:23.02ThoMehow i can get theis SIP/gw_7_asterisk01-.... value?
11:24.28kaldemarcore show channels concise
11:25.48*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
11:25.49ThoMekaldemar: jep, but how i can get this in my dialplan?
11:26.32kaldemarThoMe: with a shell script for example
11:27.14kaldemarnextime: blasted, bot seem to be in 1.6 only
11:27.18kaldemarboth
11:28.25*** join/#asterisk wr| (n=wr@p54BE7426.dip.t-dialin.net)
11:31.25nextimekaldemar : ok, so if i don't find the manager command, i can suppose is <= 1.4
11:31.27nextimeright, thanks
11:32.52*** join/#asterisk voipmonk (n=voipmonk@69.172.93.45)
11:32.54renzoeguys. im trying to compile dahdi but my kernel is newer that what dahdi is looking for
11:32.57kaldemarThoMe: something like Set(callee=${SHELL(asterisk -rx 'core show channels concise' | grep ^${CHANNEL} | awk -F\! '{ print $7 }')})
11:33.01renzoedahdi says "You do not appear to have the sources for the 2.6.18-92.el5 kernel installed."
11:33.08kaldemarThoMe: might be a better way to do it
11:34.05tzafrir_laptoprenzoe, you need linux-devel 2.6.18-92.el5
11:34.08renzoebut i got: kernel-devel-2.6.18-164.el5
11:34.19renzoehmm ok will try to yum it
11:34.43ThoMekaldemar: but when i have more as one call, but which one?
11:34.44renzoeare you sure its linux-devel?
11:34.49renzoeim usinf centos 5.2
11:34.59ThoMekaldemar: example
11:35.00ThoMeasterisk01:/etc/asterisk# asterisk -rx 'core show channels concise' | grep ^${CHANNEL} | awk -F\! '{ print $7 }'
11:35.03*** join/#asterisk bluOxigen (n=asad@static-host119-73-71-143.link.net.pk)
11:35.04ThoMe(Outgoing Line)
11:35.06ThoMeSIP/442@gw_7_asterisk01|180
11:35.09ThoMe(Outgoing Line)
11:35.11ThoMeSIP/456@gw_7_asterisk01|180
11:35.19renzoethere's no packaged named linux-devel
11:35.39*** join/#asterisk s0lid (n=s0lid@61.28.187.210)
11:37.10renzoetzafrir_laptop, any more ideas?
11:37.18kaldemarThoMe: that's what the grep is for. take the line that you need to get.
11:38.09renzoei cannot move on to asterisk until i install dahdi
11:39.19tzafrir_laptoprenzoe, the package you need is http://vault.centos.org/5.2/os/i386/CentOS/kernel-devel-2.6.18-92.el5.i686.rpm
11:40.08renzoebut i already have a newer kernel-devel
11:40.31kaldemardo you have a newer kernel in use also?
11:40.41renzoeif i need to downgrade, how will i do that?
11:40.55*** join/#asterisk ice_croft (n=nolan@mail.kubkurort.ru)
11:41.13renzoei have this one installed: kernel-devel-2.6.18-164.el5
11:41.51renzoei am using 64 bit version
11:41.53*** join/#asterisk [8none1] (n=[8none1]@c-76-22-141-39.hsd1.tn.comcast.net)
11:41.57renzoeof centos 5.2
11:42.52renzoeif i dont install this. will zaptel still work?
11:43.13kaldemaryou already got a link to the right package
11:44.05kaldemardahdi is the new name for zaptel.
11:44.19renzoeso i dont need to install dahdi?
11:44.42renzoei am trying to follow this tutorial: http://www.voip-info.org/wiki/view/CentOS+5.2+and+Asterisk+1.6.x+installation
11:45.05kaldemarzaptel doesn't work with 1.6. you need dahdi for it.
11:45.30renzoeits just dahdi is getting my way
11:45.38kaldemarif you want to use telephony hardware and a dahdi clock, then you need it. if not, then don't install it.
11:45.54renzoeso how can i install dahdi on the newer kernel?
11:46.11kaldemardahdi clock is needed for example in meetme conferences and iax2 trunking.
11:46.47kaldemardoes uname -r match the header package version you have?
11:46.49*** join/#asterisk maour (n=gnu@unaffiliated/maour)
11:46.54renzoelemme check
11:47.12renzoeno
11:47.23renzoeuname -r says its 92.e15
11:47.30renzoehow come
11:47.46renzoeshould i uninstall the kernel devel?
11:47.52renzoethen try to reinstall it?
11:47.53kaldemarthen your header package version is wrong. install the package tzafrir linked to you and proceed with the compiling
11:48.16renzoebut can i uninstall it first?
11:48.31kaldemarsure
11:49.37renzoethanks kaldemar
11:49.50ThoMeah, kaldemar this is the variable ${DIALEDPEERNAME}
11:50.34kaldemarThoMe: is it the whole channel name or just the peer name?
11:51.05ThoMethe value is: SIP/gw_7_asterisk01-0882dd40
11:52.05ThoMebut when i use     SoftHangup(${DIALEDPEERNAME}|a);
11:52.08ThoMethe callee is not hanging up
11:52.12ThoMehmm :-(
11:53.10*** join/#asterisk mikkel (n=mikkel@84.238.113.66)
11:53.12*** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw)
11:58.10renzoesigh
11:58.22renzoei have been installing i386 version
11:58.27renzoei have 64bit
12:00.43renzoefinally.. thanks kaldemar and tzafrir_laptop
12:00.59renzoeits doing make all now
12:03.40*** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net)
12:17.05*** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163)
12:22.49Dovidanyone know what would cause this ?
12:22.49Dovidhttp://pastebin.com/m2672c754
12:27.21*** join/#asterisk scalex000 (n=chatzill@95puntacana02.codetel.net.do)
12:27.24*** join/#asterisk Naikrovek (n=jjohnson@63-252-251-77.ip.mcleodusa.net)
12:27.31scalex000good morning
12:27.52cjkhi, is there a way to send RTP before the call is answered?
12:28.20cjki mean to the called party
12:32.00*** join/#asterisk denon (i=denon@sassinak.net)
12:32.00*** mode/#asterisk [+o denon] by ChanServ
12:32.56m0biuscjk: yes
12:33.19cjkm0bius, not to the calling party, but to the destination
12:33.21m0biuscjk: you might need to set Progress() before doing Playback but make sure you set Playback to play even on un-answered channels
12:34.12m0biushum, but how will the other party hear the RTP audio if he hasn't pickup the phone?
12:34.15Naikrovekwonder if you could send a secondary data stream to another voip user.  ooh
12:34.32*** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee)
12:34.35cjkm0bius, its not important if he doesnt hear it
12:34.46cjkbut i need to bypass nat problem in one special scenario
12:34.47Naikroveksome sort of collaborative notetaking via RTP during a call...
12:35.29m0biushum
12:35.48cjkthe problem is this
12:35.49Naikrovekor even just for file transfers, for things like business cards and the like
12:36.00m0biusif the problem is nat, why don't you enable qualify to force an open connection to the router?
12:36.04cjkcall comes from asterisk1 to asterisk2 and asterisk2 sends it back to asterisk1
12:36.12cjkbetween ast1 and ast2 is a NAT
12:36.24cjkif i do not answer on ast2 to create some RTP, then i will not have voice
12:36.31cjkthis scenario happens for call redirects
12:37.52m0biushmmm
12:37.58m0biusthat doesn't sound logical
12:38.04*** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at)
12:38.17m0biusif nat is your problem you should focus on solving the nat
12:38.24cjkits not the NAT
12:38.31cjki tested with all nat configuraitons
12:38.42cjkits the fact that the one that answeres the call is outside the nat
12:39.11cjkif i answer the call before redirecting wiht playback(beep) then it works
12:39.19cjkso its not 100% a nat issue
12:39.48troffaskyI would take some packet captures if I were you
12:41.34m0biusI've tested multiple setups where asterisk has been in or out of nat but I never had issues
12:41.41cjki didn the problem sounds no easy to understand, difficult to explain and more difficult to solve
12:41.53m0biusso I guess taking a capture could shed some light
12:42.17troffaskycjk, maybe a diagram would help?
12:42.23cjkyes
12:42.25troffaskypicture being worth 1000 words and all that
12:42.25cjki create one
12:45.47*** part/#asterisk so_solid_moo (n=nmoo@fsf/member/alexhudson)
12:46.10*** join/#asterisk tcseke (n=chatzill@217.20.134.239)
12:46.32jayteewow! my old AOL account back in the 90's was mobius812
12:48.06m0biusreally?
12:48.31m0biusI've been using since the 90s as well :)
12:48.35cjkhere is a quick schema: http://www.wanter-trail.org/natredirect.pdf
12:48.57cjki showed 4 scenarios out of which 3 works, just to show that i do not have a general nat problem
12:49.19m0biusdo you have reinvite on on asterisk 2?
12:49.24cjkno of course not
12:49.52cjkthe problem is that in the last scenario no RTP is comming from the inside of the nat
12:50.13cjkand as such, it doesnt accept RTP from the outside, and so the conversation never takes place
12:52.18kaldemarcjk: what application are you using for the "redirect"? a Dial?
12:52.20*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
12:52.26cjkkaldemar, yes
12:52.35cjkjust normal dialplan logic
12:53.03kaldemaryou could use app transfer to send a SIP 302 to the originating asterisk
12:53.05*** part/#asterisk ice_croft (n=nolan@mail.kubkurort.ru)
12:53.28cjkkaldemar, not an option, it should look like a normal call
12:53.49kaldemarso you want the signaling to go through both servers?
12:53.57cjkyes
12:54.18cjkkaldemar, most of the time the problem does not exist because incoming calls come from isdn or pri on asterisk2 so there is no problem
12:54.23kaldemarthen disable re-invites or answer the leg.
12:54.38cjkkaldemar, reinvites are disabled
12:54.44cjkanswereing has billing issues
12:55.30cjkis there a solution to do something like playback beep on answer?
12:55.41m0biusyou can play an audio on answer
12:55.47m0biuson the calling party
12:55.53m0biusand the called party
12:56.04m0biusbut I am not sure that would solve your problem
12:56.50*** join/#asterisk jsmith (n=jsmith@asterisk/training-and-documentation-guru/jsmith)
12:56.50*** mode/#asterisk [+o jsmith] by ChanServ
12:56.56cjkif asterisk2 plays it i think it will, let me check
12:56.59*** join/#asterisk defswork (n=andy@mx2.3gcomms.co.uk)
12:57.09*** topic/#asterisk by jsmith -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.6 (2009/09/03), 1.6.0.15 (2009/09/03), 1.4.26.2 (2009/09/03), *-Addons 1.6.1.1 (2009/07/24), 1.6.0.3 (2009/07/24), 1.4.9 (2009/07/24), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.1 (2009/07/02) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev, AstriCon Dis
12:57.11ThoMehmm, i try:
12:57.17ThoMe[Sep 23 14:55:33] WARNING[27056]: pbx.c:2437 __ast_pbx_run: Channel 'SIP/53002-b6701a08' sent into invalid extension 's' in context 'from-internal-users', but no invalid handler
12:57.26ThoMe<PROTECTED>
12:57.29ThoMeis it wrong?
12:57.51*** topic/#asterisk by jsmith -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.6 (2009/09/03), 1.6.0.15 (2009/09/03), 1.4.26.2 (2009/09/03), *-Addons 1.6.1.1 (2009/07/24), 1.6.0.3 (2009/07/24), 1.4.9 (2009/07/24), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.1 (2009/07/02) -=- Related channels: #asterisknow #switchvox #asterisk-bugs AstriCon Discount code: ac09digi
12:58.22*** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
12:58.22*** mode/#asterisk [+o leifmadsen] by ChanServ
12:58.23*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
12:59.36*** join/#asterisk voipmonk (n=voipmonk@69.172.93.45)
13:00.11*** join/#asterisk Gumug (n=Gumug@nmd.sbx09566.joplimo.wayport.net)
13:00.18Gumughowdy
13:00.42Gumugi need some advice
13:00.43*** join/#asterisk Iamnach0 (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net)
13:02.02Naikrovekask your question
13:02.04jayteejsmith, hi!!!
13:02.22Gumugalrighty
13:02.40Kattymew.
13:02.47*** join/#asterisk engrxyz (n=zcvzxcvx@host81-143-50-89.in-addr.btopenworld.com)
13:02.55jayteemornin Katty
13:03.02Kattyhi
13:03.03jsmithjaytee: Long time, no chat... what's up?
13:03.38jayteejsmith, not too much. getting ready to move my installation to 1.6.0.15
13:03.56jayteejsmith, how's things in your neck of the woods?
13:04.49Gumugi am starting work this week at a new business.  We have 9 offices located in missouri and arkansas.  We have 16 phone lines.  Each line costs $60ish a piece to AT&T.  I want to see if i can lower the costs of communication, while also implementing ACD, voice prompts, etc.
13:05.10jayteesends jsmith a mango smoothee using app_virtual_beverage.so
13:05.13Gumugall of our business is over the phone
13:05.13jsmithGumug: Then you're in the right place :-)
13:05.15*** join/#asterisk lanning (n=lanning@212.183.136.194)
13:05.19KattyGumug: Missouri, huh?
13:05.22jsmithjaytee: Thanks!
13:05.24Gumugyep
13:05.37jsmithjaytee: Busy here (as always) with the Asterisk training... and I'm getting geared up for AstriCon
13:05.45Kattysmall world.
13:05.52Gumugsouthwest missouri infact
13:05.56jayteejsmith, you in Huntsville or teaching a class back east?
13:06.09KattyGumug: i'm over in cape.
13:06.12*** join/#asterisk jamesh1 (n=jhenders@xob.neospire.net)
13:06.13jsmithjaytee: I'm at home until Friday, and then I fly to San Diego
13:06.16Gumugright on
13:06.34jayteejsmith, I'm so jealous!! San Diego is a great city
13:06.35Gumugi've been looking at these hosted pbx's
13:06.51Gumugsome cost a ton of money each month, for the amount of talk time we are going to have
13:06.57[TK]D-FenderGumug: Is that 16 lines spread around?
13:07.05Gumugyes
13:07.17Gumugsome offices have 2
13:07.19Gumugsome have 3
13:07.21[TK]D-FenderGumug: AKA each site having a line or two
13:07.25Gumugyes
13:07.44[TK]D-FenderGumug: S2 or 3?  at 16 for 9 offices, sounds like some have NONE :)
13:08.19Gumugyou know
13:08.24Gumugi think we have a lot more
13:08.29Gumugsome have 2 actual phones
13:08.33Gumugsome have three phones
13:08.34GumugBUT
13:08.39Gumugthey are 4 line phones
13:08.49Gumugso we may have a lot more phone lines then i think
13:08.58Gumugif they are able to keep 3 or 4 calls going at once
13:09.22Gumugall i know is right now its expensive with the costs of having each line per store
13:09.36Kattystore?
13:09.39Kattyis this a retail chain?
13:09.46Gumuginsurance broker
13:09.51Kattybummer :<
13:09.56Gumugpeople call in, or walk in for quotes/policies
13:09.58Kattyi was excited about going shopping there for a minute.
13:10.02Naikrovekbummer?  ah
13:10.02Gumuglol
13:10.14Gumugsell you some SR22 insurance
13:10.35[TK]D-FenderGumug: Each site have broadband?
13:10.41Gumugyes
13:10.53Kattyi've not been in any major accidents.
13:11.02[TK]D-FenderGumug: then you can consolidate to a central server
13:11.03Naikrovekthis is shaping up almost to be a case study for asterisk deployment
13:11.04Gumugi'm glad
13:11.30Kattyany chance your locations are connected through a VPN?
13:11.42Naikrovekoh that would make it even easier
13:12.04*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
13:12.47Gumugi have no idea
13:12.55Gumugi doubt it
13:13.00Gumugi have an IT background
13:13.10Gumugi know i could figure it out if VPN were necessary
13:13.17Naikrovekthat's something else you'd want to implement probably
13:13.27Kattyit has lots of benefits.
13:13.32Gumugsuch as?
13:13.49Kattyphone systems can act quirky with routing when you're going through a firewall.
13:13.58Gumugok
13:13.59Gumugduly noted
13:14.04Kattyplus, being in IT, you know all the benefits of having a VPN connection
13:14.14Kattysecurity, data access, etc
13:14.15Gumugsecurity
13:14.16Gumugetc
13:14.27Gumugafk sec
13:14.27Kattynot to mention you can use VNC to help people
13:14.34Kattyrather than /driving/ there
13:14.49Naikrovekit'll make your asterisk deployment much easier.  secure email exchange between offices, secure file transfer between offices, centralized file server(s) and email server, so on and so forth, etc, etc,
13:15.04Kattydude, i can't function this morning
13:15.11Naikrovekyeah you can dude
13:15.54Kattyi feel like i'm going to fall asleep on my keyboard i'm so tired.
13:16.42Naikrovekdo what i do: "ugh i don't feel good, I need to go to the bathroom" then go take a nap
13:17.05Naikrovekthat's what i used to do anyway, before i got diagnosed with narcolepsy
13:17.24*** part/#asterisk jsmith (n=jsmith@asterisk/training-and-documentation-guru/jsmith)
13:17.56*** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844)
13:17.59Kattyi am NOT going into the bathroom here. it is /gross/
13:18.23Naikrovekeven better "i don't feel good, i'm going home to use the bathroom"  nap
13:18.23*** join/#asterisk troffasky (n=r00t@92-234-126-57.cable.ubr08.gate.blueyonder.co.uk)
13:18.44Naikrovek30 mins is pure dreamy wonderment
13:18.46Gumugright now email is handled by google apps
13:18.54Gumugas well as docs
13:18.57Katty30minutes isn't gonna do anything for me.
13:19.04NaikrovekGumug: ah
13:19.07Kattyneed another 4 hours
13:19.56Gumugright now, each store has its own phone number, which is advertised in the local yellow pages, and radio spots
13:19.58scalex000Hello
13:20.18Gumugwe are thinking about going to 1 or 2 numbers for all the stores though
13:20.22scalex000I need to interconnect BCm and asterisk using h323 or sip any ideas?
13:20.39[TK]D-FenderKatty: Do't need a VPN for VNC...
13:21.14Naikrovekwell if they already have a decentralized infrastructure then a vpn probably won't do much for them
13:21.26*** join/#asterisk spck (n=spck@unioncab.com)
13:22.22Gumugi'm looking at these hosed pbx's, but the cost of minutes is out of this world
13:22.29Naikrovekdon't do hosten
13:22.31Naikrovekhosted
13:22.36[TK]D-Fenderscalex000: Idea : throw the BCM into a dumpster and BURN IT
13:22.47GumugNaikrovek: really?
13:23.36*** join/#asterisk cosmicwombat (n=cosmicwo@69.7.44.68)
13:23.50Naikrovekyou can set up your own server in the biggest or most centralized office, and put phones in each of the other offices, and consolidate those 16 lines into a single box and probably save some money.  i pay $32 per "line" per month using a voip provider
13:24.10Naikrovekunlimited incoming, unlimited free calling to US & canada as well
13:25.22Gumugsave money over time right?
13:25.36Naikrovekyes, definitely
13:25.38Gumugi priced it out and some were going to cost over $500 a month
13:25.49Gumugbecause we are on the phone over 2hrs a day per store
13:26.11Gumug2000 minute plans will be eroded in a day
13:26.11Gumugor 2
13:26.11*** join/#asterisk jcape (n=jcape@209.120.251.81)
13:26.14Naikroveki have asterisk where i work.  one office is in illinois, the other is in India.  one asterisk server
13:26.25Naikrovek60-80 or so extensions
13:26.32scalex000Tk: if I was the owner I will do it but its not my.
13:26.59Kattyscalex000: i think your best bet in this situation would be to hire a consultant.
13:27.08[TK]D-FenderGumug: Where are those 2 hours talking to?
13:27.25Gumugso, would you recommend that we consolidate our #'s to 1 phone #, or would it still be possible to have numbers rerouted
13:27.31troffaskyif it's site2site and you've already got internet in each site...
13:27.44Gumug[TK]D-Fender: the vast majority of our business is incoming calls
13:27.51NaikrovekGumug: each phone could have its own phone number, or its own extension number on a single phone number, or both
13:28.05*** join/#asterisk Orbixx (i=Orbixx@office.exoware.net)
13:28.14Gumugfascinating
13:28.25*** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq)
13:28.43NaikrovekGumug: you're about to enter the rabbit hole buddy :)
13:28.44Gumugpeople call in to get quotes on insurance, change policies, purchase policies, etc
13:28.56GumugNaikrovek: the thing is, my role is really HR
13:29.06OrbixxHow can I record a call that takes place after a client has queued?
13:29.09Naikroveki'm overusing this word, but asterisk is pure telephonic wonderment
13:29.15Gumuglol
13:29.26Gumugis it set it and forget it easy?
13:29.33Orbixxit's more like
13:29.35Gumugi don't have the time to be a sys admin
13:29.55Orbixxset it, set it again, it's still not perfect, set it again, set it some more, then forget it for a long time
13:30.46Gumugyes, like anything
13:30.52Orbixxquite
13:31.26creativxi did a looot of setting
13:31.29creativxi even set too much
13:31.31creativxthen i removed some setting
13:31.38creativxthen i forgot about it and its been what.. 2 years
13:32.01Gumugdo you recommend colocating an asterisk server
13:32.01Kattyfalls asleep on desk
13:32.20Gumugto a more secure area with regulated power, clean connection
13:33.07*** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim)
13:34.19troffaskyif you don't already have a site with decent connectivity then colo may be a good idea
13:34.55Gumugya
13:35.02Gumugsee, if the phones go down
13:35.06Gumugwe are screwed
13:36.04creativxyou dont have.. email?
13:36.07creativx;)
13:36.11creativxthats what you have failover for
13:36.14Gumuglol
13:36.14creativxand redundancy
13:36.45Kattywe use mobile phones for emergency events
13:36.52Kattywe just have the telco forward stuff
13:37.13Kattysomeone please invent a drink with caffeine and melatonin
13:37.31creativxso melanotan and caffeine
13:37.41*** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy)
13:38.05Kattymelatonin effects the pineal gland.
13:38.17Kattywell. not exactly.
13:38.19Naikrovekcaffetonin
13:38.32Kattyusualy the pineal gland secrets melatonin
13:38.36creativxKatty: do you know what melanotan is though :)
13:38.49Kattyit's a bloody awesome hormone
13:38.55*** join/#asterisk denon (i=denon@sassinak.net)
13:38.55*** mode/#asterisk [+o denon] by ChanServ
13:38.58creativxhehe
13:39.02creativximagine drinking it
13:39.02creativx:)
13:39.02NaikrovekGumug: you could put the asterisk server on a UPS, so it could stay up through a couple hour power outage.
13:39.25[TK]D-FenderNaikrovek: Hours for a server?  More like a generator
13:39.27creativxNaikrovek: ever heard the term mission critical
13:39.30creativxand yes
13:39.33Kattyhttp://blogs.babycenter.com/momformation/files/2008/08/melatonin_tablet.jpg <- i guess i could crush it up and put it in my soda.
13:39.33creativxthat would be one big ass ups
13:39.49Naikrovekthough in that case you're likely to need to use a PoE switch, which you would then also put on the UPS to power the phones in that office
13:39.59Naikrovekwell $3k can get you a natural gas powered generator
13:40.15Naikrovekone would only need battery for the 5 seconds it takes for the generator to auto-start
13:40.18creativxand what about the switching
13:40.19troffaskyso, er, yeah, just get it colod
13:40.24creativxin other words yes
13:40.25creativxcolo
13:40.25Naikrovekcreativx: included
13:40.27creativxor
13:40.30creativxbuild yourself a server room
13:40.40Kattyyeah don't put it in a closet :<
13:40.44Kattyor the bathroom.
13:40.44creativxfrom scratch.. with cooling.. batteries.. and redundancy and access control and and and..
13:40.47Kattywhich i /have/ seen.
13:40.56Naikrovekyeah i've seen that also
13:40.59creativxi once called an asterisk number to flush a toilet on a webcam
13:41.00Kattyand MELATONIN!?
13:41.05creativxcuriosity
13:41.06*** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001)
13:41.11Naikrovekvent holes covered by saran wrap to protect them from splashing
13:41.35scalex000this warning what its means "channel.c:3606 ast_channel_make_compatible_helper: No path to translate from H323/test-11(256) to SIP/246-b7812d48(4)"
13:42.52Kattyit means it doesn't have enough melatonin
13:44.13*** join/#asterisk Methose (n=Methose@38.101.237.250)
13:44.21creativxhehehe
13:44.29creativxmelatonintan
13:45.24[TK]D-Fenderscalex000: Means "pick codecs that * can translate.  You allowed either G.729 or G.723 without licenses
13:46.11beekmornin' [TK]D-Fender
13:46.22beekwaves to Katty
13:47.50*** join/#asterisk jmacz (n=jmacz@190.144.75.22)
13:47.53phixwaves to Katty and beek!
13:48.47*** join/#asterisk Yuda-israel1984 (n=yo1984@62.219.144.190)
13:49.04*** join/#asterisk jcape (n=jcape@209.120.251.81)
13:49.22*** join/#asterisk denon (i=denon@sassinak.net)
13:49.22*** mode/#asterisk [+o denon] by ChanServ
13:49.23[TK]D-Fenderbeek: Mornin'
13:49.37Yuda-israel1984hi guys so this is my first time on here and i have been looking for ways to learn asteirks scripting and im finally doing it although im having trouble with something and i was wondering if anyone can help me
13:49.40Kattyhugs beek
13:49.42Kattyphix: hello.
13:50.02Yuda-israel1984is anyone familiar with MyPhoneCompany ?
13:50.02phix:D
13:50.08phixnn
13:50.58Yuda-israel1984anyone familiar then with sip messages?
13:51.13Kattyyawns.
13:51.17Kattyokay, gotta wake up for real.
13:51.45Yuda-israel1984katty u a scripter?
13:51.52KattyYuda-israel1984: no.
13:52.23KattyYuda-israel1984: just about everyone in here is a volunteer. if no one has an answer, i would suggest being patient and asking again in a little bit. otherwise, if its high priority you might consider finding a consultant to assist you.
13:53.10creativxsound advice from Katty there
13:53.12Yuda-israel1984im new here so i dont really know how this IRC works last time i used it was a good 10 years ago at least when i was a child
13:54.00Yuda-israel1984i will ask later i appreciate it
13:54.01Kattykk
13:55.28kaldemarYuda-israel1984: the best way to get an answer is to ask a specific question
13:55.57*** join/#asterisk errr (n=errr@fedora/errr)
13:55.57Kattyspecific question: which has more caffeine? a cup of coffee or a cup of mt dew.
13:56.03Kattyhi errr
13:56.11*** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson)
13:56.11*** mode/#asterisk [+o putnopvut] by ChanServ
13:56.21Kattyhugs putnopvut
13:56.34Naikrovek~book
13:56.35infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
13:56.35putnopvutoh snap
13:56.38NaikrovekYuda-israel1984: ^^^^^^^
13:56.43putnopvuthugs Katty back
13:56.57*** join/#asterisk l2trace99 (n=jr@75.112.140.2)
13:57.32*** join/#asterisk errr (n=errr@fedora/errr)
13:58.44Kattya cup of coffee (not instant) contains 90-150mg of caffeine. mew dt has 36mg per cup.
13:58.52Kattymew dt?
13:59.03Kattydang. i really AM that tired.
13:59.04troffaskyis anybody running SIP over OpenVPN terminated on their * server?
13:59.11Kattytroffasky: i'm not.
13:59.20Naikrovektroffasky: i'm sure someone is
13:59.23Naikrovekbut i'm not
14:00.12Kattyred bull only has 80mg per 8oz
14:00.49Kattythat's insane.
14:01.03Kattywoulda figured redbull has more caffeine than coffee.
14:01.27creativxredbull has lots of sugar
14:01.57Methosehense the crash after 10 mins
14:02.01Kattysugar doesn't do much for me.
14:02.41Kattysugar is horribly bad for you.
14:02.55Kattyso is this diet mt dew i'm drinking.
14:03.01Kattyit will make me hungry in about 15 minutes.
14:03.43troffaskyI wonder if 20ms RTP packets are fattening
14:03.54*** join/#asterisk Carlos_Tico (n=grillo_v@c-98-201-162-34.hsd1.tx.comcast.net)
14:04.15Kattytroffasky: rtp packets are not fattening.
14:04.18Kattyhi carlos.
14:04.40creativxKatty: diet is bad
14:04.45creativxcause of the artificial sugar
14:04.49Kattyyes.
14:04.51Kattyit's horribly bad.
14:04.52creativxit induces hunger
14:04.56Kattyyes, it does.
14:04.59creativxplus you probably get cancer but thats not so important
14:05.03creativx:p
14:05.09creativxaspartam
14:05.18troffaskyif you live in the west, chances are you'll die of cancer before anything else
14:05.19Kattyit also contains phenylalanine
14:05.41leifmadsenI just don't drink pop
14:05.56Naikroveki'm more or less addicted to caffiene free diet coke.  /shame
14:05.57*** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com)
14:06.12leifmadsenI used to drink way too much of it -- I had to quit the addiction :)
14:06.16troffaskycould you explain the point of caffeine free coke?
14:06.18leifmadsenI just drink green tea now
14:06.22Naikrovektroffasky: no idea
14:06.23troffaskydo you just want brown teeth?
14:06.29Kattyit is in a can.
14:06.32Naikrovektastes good?
14:06.32*** join/#asterisk flohack (n=fhackenb@lancelot.acoveo.com)
14:06.32Kattyeasy accessible.
14:06.35Naikrovekplastic bottle
14:06.41Kattydoes not taste of funky city water.
14:07.46Naikroveki used to drink 24 cans of pepsi (not diet) per day
14:07.55Naikrovekthat's why i drink diet now.
14:08.04Naikroveki also have narcolepsy, so caffiene is a no-no
14:08.11Naikrovekthat's why i drink caffiene free
14:08.15creativxahha
14:08.18creativxnarcolepsy sounds fun
14:08.25Naikrovekyeah i about died while driving a few times
14:08.36Chainsawcreativx: It's fun until it happens to you.
14:08.37l2trace99any one have issues with all calls within a queue dropping ?
14:08.39Kattyi'd suggest not doing that anymore.
14:08.45Naikrovekwake up donig 75mph in the grass between lanes on a 4 lane highway
14:08.50Naikrovekthat's a sobering experience
14:09.04Kattyah, sobering experiences.
14:09.07Kattyi've had a few of those.
14:11.55Kattydid i mention my dad's on facebook?
14:11.59Kattythis is weird.
14:12.21creativxouch Naikrovek
14:12.42troffaskyistr you mentioning it yesterday Katty
14:12.51troffaskyit wasn't particularly interesting then either
14:13.26Kattywell perk up buttercup, or life is gonna be awfully boring for you!
14:13.53Kattywe don't do cranky negativism here!
14:13.57creativxhehe
14:14.00creativxbuttercup
14:15.48Kattymy parents were such a cute couple.
14:15.57Kattyhttp://farm4.static.flickr.com/3158/3023014977_5f88fd7035.jpg <- mom.
14:16.09Kattyhttp://farm4.static.flickr.com/3066/3023839552_7be80cd1b5.jpg <- dad.
14:16.16creativxim not even sure why i clicked on a link with a picture of your mom
14:16.26creativxgive me a link and i will click it
14:16.34Naikrovekcreativx: lol same here
14:17.20Kattymaybe it gives you some indication of how i look.
14:17.20Kattymaybe you're just bored.
14:17.23Kattywho knows! it's something intersting to look at.
14:17.24Naikrovekyou posted a pic of yourself once
14:17.27Naikrovekwith your dog or something
14:17.39*** join/#asterisk Deeewayne (n=dwayne@75.76.254.162)
14:17.39*** mode/#asterisk [+o Deeewayne] by ChanServ
14:17.53Kattyhi Deeewayne (=
14:18.11DeeewayneKatty, good morning :-)
14:18.35creativxi should find a really retarded picture link and see how many clicks it
14:18.36Kattycreativx: you can't look at that picture and not smile.
14:19.31spckanyone got a recipe for answering machine detection, i.e. what delays they used?
14:20.17Naikrovekspck: i always thought detection was listening for how long a person talks when they answered
14:21.11SuPrSluGworks ok if you can live with 85% accuracy
14:22.30spckdoing callouts for a taxi company, not mission critical
14:23.24spcknaikrovek: the amd() application has a bunch of options for delay timings, i was wondering if anyone had a set that they had a good experience with
14:23.36Naikrovekah
14:23.38Naikroveki haven't
14:25.13SuPrSluGthese worked ok for me http://pastebin.com/m264227d2
14:28.58spckthanks for the pasty
14:29.20*** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk)
14:29.38*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
14:29.43Kattyhi tony
14:29.58anthmhi
14:30.16Kattyhow're you dear
14:31.50guaxim fine
14:32.11Kattyexcellent.
14:32.22crazybyteHello to all! How can I modify a Digium Wildcard TE110P T1/E1 card sync source settings? I tried in /etc/zaptel.conf but doesn't seem to work. Thank you!
14:32.30*** join/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com)
14:32.55*** join/#asterisk MWE (n=michel@nl06sr01.targetmedia.nl)
14:33.02MWEhi there
14:33.47*** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda)
14:33.47MWEhopefully some could me out...
14:33.47[TK]D-FenderCraIt does.  You also have to stop * and re-init zaptel for that change to take effect
14:33.47MWEhelp*
14:33.48mbrevdadoes dahdi fax detect require answer?
14:33.56[TK]D-Fendercrazybyte: And I don't see your before & after configs
14:34.07[TK]D-FendermbrOf course it does
14:34.16[TK]D-Fendermbrevda: Of course it does
14:34.22[TK]D-FenderGah... Autocomplete fail day
14:34.33crazybyte[TK]D-Fender, one sec (sorry I forgot about them)
14:34.37Katty[TK]D-Fender: you've been doing that a lot lately.
14:34.42Katty[TK]D-Fender: not just today.
14:34.48mbrevda[TK]D-Fender: hey - how've u been? thanks!
14:34.53Katty[TK]D-Fender: but we still love you.
14:35.02mbrevdahow about the new sip fax detect?
14:35.35mbrevdacouldnt find much documentation of using it
14:35.52MWEi put someone in the meetme room with the flag b (run a background script) that script calls somebody with dial and had to set that user in the same meetmeroom. But instead of putting the callee in the meetme room there will be a bridge between the caller and the callee. When the callee hangsup the script is trying to put him again in the meetme room.. IS there a way that the background script is doing the follwing: call the callee and put this in
14:35.52MWE<PROTECTED>
14:36.18*** join/#asterisk |Cybex| (n=John@80.100.126.176)
14:36.36MWEis that with originate?
14:36.44*** join/#asterisk lordmortis (n=lordmort@203.59.207.20)
14:38.47*** join/#asterisk Gumug (n=Gumug@98.98.181.70)
14:38.52Gumugso...
14:38.57Gumuglooking at switchvox
14:38.59Gumugpretty expensive
14:39.05Gumuganyone ever used it?
14:39.21spckoverpriced and i believe they have a yearly subscription requirement
14:39.37crazybyte[TK]D-Fender, here is my zaptel.conf http://pastie.org/627460 and my zapata.conf http://pastie.org/627463
14:39.40troffaskyhave you considered asterisk instead? ;-)
14:40.02Gumugwell, i liked the aspect of it being tied into salesforce, et all
14:40.18CareBear\D-Fender : I was obviously missing auth= for my [peer]
14:40.56crazybyte[TK]D-Fender, also I have the following error (in asterisk console) chan_zap.c:2498 pri_find_dchan: No D-channels available!  Using Primary channel 16 as D-channel anyway!
14:41.08spckpersonally i didn't see any upside to their system other then it being turnkey
14:41.36Gumugsure, i guess thats the nice thing, that it's turn key
14:41.49spckbut even then that's a downside
14:41.56Kattyi'd like to know who invented the phrase Turn Key
14:42.08spckyou have to buy the silver support contract to continue using it
14:42.10Gumughe suggested i got a SMB AA60 Appliance for each office, which would be 9 installs
14:42.24spckdepends on your budget i guess
14:42.29Gumug$3,890.00
14:42.30Gumugeach
14:42.37troffaskyboggles
14:42.46Gumugwould asterisk be less?
14:42.49spcki spent $10k on two dell poweredge 2950
14:42.56KattyGumug: i'm sure someone in here would be happy to get a quote together for you.
14:42.58spck(total overkill for my install)
14:43.06Naikrovek????  compiled and installed asterisk 1.6.1 - where did 'core show translation' go
14:43.16spckasterisk is free
14:43.18GumugKatty: oh i'm sure
14:43.23Gumugso, he suggested i peer the systems
14:43.23spckno strings attached other then the license
14:43.31Kattyi think we spent about 2k on our server.
14:43.36Kattyprobably another 1.5k on the pri card.
14:43.38Gumugso that if one goes down, than, the system doesn't go down as a whole
14:43.40Kattyand around 100 per phone
14:43.50mbrevdagrandstreams?!
14:43.55Kattypolcyom 330s
14:44.00mbrevdaoh, right
14:44.03mbrevdacool
14:44.04Naikrovekpolycom ftw
14:44.06Gumugso, can you peer astrisk like that?
14:44.07Kattymhmm
14:44.09mbrevda+1
14:44.09spckgumug: you can do all that with asterisk
14:44.14Gumugok
14:44.16spckyou just have to setup it yourself
14:44.26Gumugis it rocket surgery?
14:44.27spckswitchvox is basically a supported preinstalled asterisk server
14:44.36KattyGumug: it can feel like it at times.
14:44.39spcklol, no but it can be frustrating
14:44.40Gumuglol
14:44.40Naikrovekjust gotta set up trunks between the systems.  Gumug nope, it's easy to trunk the systems
14:44.50KattyGumug: there are other asterisk based solutions. like Fonality
14:44.51CareBear\Gumug : I started asterisk for the first time about a month ago
14:44.53Naikrovekand we're here to help if you need it
14:45.15Kattyi believe Digium will also help, on a pay per case basis.
14:45.17spcki also have to say polycom's rock
14:45.23Kattymuch like microsoft.
14:45.30Kattyin case you get hung up in your underwear.
14:45.33Gumugright on
14:45.41CareBear\Gumug : I've spent maybe 20 hours on it efficiently and now I have set up a small dial plan for my incoming number, my friends can register with me, and just last night I finally got internal calls between extensions going.
14:45.45spckyou can also put bounty's on features you want
14:45.57Gumugsweet
14:46.00KattyGumug: the only drawback with Asterisk, that i've found...
14:46.11russellbis that it's too awesome?
14:46.13KattyGumug: is if something /breaks/, there isnt' an 800 numbe ryou can call for support.
14:46.17russellbyeah, sorry about that
14:46.24KattyGumug: there's not a Support Contract or anything like that involved.
14:46.35KattyGumug: like you would with a Hosted Solution.
14:46.36russellbYou can purchase a support contract from Digium, actually.  :-)
14:46.41Naikrovekwell there are consultants who lurk in here that
14:46.46Gumugso get used to being in the fetal position crying?
14:46.49spcki would say the real problem with asterisk is that it doesn't have any one specific purpose and thus has been pulled in several directions
14:46.56KattyGumug: no. you just pay for a support contract.
14:47.07KattyGumug: it's not Included(tm)
14:47.12Gumugyes
14:47.19CareBear\Gumug : Nono, support is available, from the makers of asterisk themselves (Digium) as well as from independant contractors.
14:47.26Gumugwell, the problem i have with hosted solutions is, they charge a fortune
14:47.30Kattyrussellb: that's good to know. you have fixed pricing? link?
14:47.33troffaskyspck, s/problem/feature imho :-)
14:47.33NaikrovekGumug: it's a learning curve, but fixes are never far away, if that makes sense.  fixes in asterisk are very quick, with few steps, but it's hard to knwo what to do if you're new
14:47.42CareBear\Gumug : The software you get for no cost, support is sold separately. :)
14:47.43Gumugsure
14:47.52Gumugunderstood
14:47.59NaikrovekGumug: but once you know what to do, it's a pain free setup really
14:48.03Naikrovekmore or less
14:48.04CareBear\Gumug : I think you can get all the support suppliers to work on your own infrastructure.
14:48.08russellbKatty: http://www.digium.com/en/supportcenter/asterisk.php
14:48.09CareBear\Gumug : Most if not all
14:48.19spckgive yourself plenty of time to plan, design, and test
14:48.44leifmadsenand then double it
14:48.45leifmadsen:)
14:48.52Naikrovekheh
14:49.01Kattyrussellb: that pricing is very competative. Samsung is doing 140 per case right now--available only to certified techs.
14:49.02Naikrovek????  compiled and installed asterisk 1.6.1 - where did 'core show translation' go
14:49.06leifmadsenmost people under-estimate how long it can take to build and test a complex solution
14:49.10MWEi put someone in the meetme room with the flag b (run a background script) that script calls somebody with dial and had to set that user in the same meetmeroom. But instead of putting the callee in the meetme room there will be a bridge between the caller and the callee. When the callee hangsup the script is trying to put him again in the meetme room.. IS there a way that the background script is doing the follwing: call the callee and put him in
14:49.10MWEthe meetme room with a pincode instead of connect him directly with the caller
14:49.10[TK]D-Fendercrazybyte: Brand new card?
14:49.23crazybyte[TK]D-Fender, yes, I think so
14:49.36Gumugleifmadsen: exactly
14:49.38CareBear\D-Fender : I didn't get a chance to say thanks last night, so thanks for your help! :)
14:49.39[TK]D-Fendercrazybyte: You probably forgot to set the jumper to E1 like you're supposed to
14:49.47crazybyteI see
14:49.54[TK]D-FenderCareBear\: did you get to finish it successfully?
14:49.55russellbKatty: spread the word ;-)
14:50.08CareBear\D-Fender : Yes, I was missing auth= for my peer, only had credentials in the register=>
14:50.35[TK]D-FenderCareBear\: Excellent.  You should not have cross-site CallerID as well
14:50.52crazybytewell it's not here. it's a remote machine so i suppose they forgot to do it. any other possible reasons (that you can think of) for such issues?
14:51.02Gumugi'm more and more inclined to just do it myself
14:51.17Gumugi don't want to get hosed on hardware, if i could do it myself for 1/4 the cost
14:51.24CareBear\Gumug : If you have a bit of time to spare and don't have to rush it, then it's a great learning experience.
14:51.29Gumugyes
14:51.35Gumugi will be invaluable then as well
14:51.39CareBear\;)
14:51.50p3nguingumug: If you know how to put a computer together, you can certainly build your own asterisk server.
14:52.03Gumugbeen doing that for 15 years
14:52.13p3nguinThen you're already ahead in the game.
14:52.14troffaskyso burn a trixbox ISO and crack on with it
14:52.21Naikrovekyeah
14:52.26Naikroveki was going to suggest asterisknow
14:52.28*** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler)
14:53.20[TK]D-Fender[10:51]<p3nguin>gumug: If you know how to put a computer together, you can certainly build your own asterisk server. <- build yes, configure... that's another matter
14:53.30Gumugyes
14:53.43[TK]D-FenderSILLY RABBIT TRIXBOX IS FOR KIDDIES!
14:53.49Naikrovekwell yes
14:53.51SuPrSluGyeah talk him into a GUI he'll get a lot of help here then
14:53.59p3nguinhahaha
14:54.07Naikrovekbut it's the best place to start if he wants to get going quickly
14:54.10[TK]D-Fenderreaches for his ClueBat (tm)
14:54.20l2trace99does anyone know if asterisk performance  shows correctly in top  ?
14:54.28SuPrSluGer. no
14:54.33Naikrovekl2trace99: why wouldn't it
14:54.34p3nguinI would rather he didn't have a GUI and learned how things work from the CLI first.
14:54.36lirakisits the place to start if you want to double the amount of work you have to do to learn anything
14:54.42*** join/#asterisk coppice (n=chatzill@157.202.17.210.dyn.pacific.net.hk)
14:54.44SuPrSluGthe best place is was and will be
14:54.48SuPrSluG~book
14:54.49infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
14:54.50Kattytakes cluebat away from [TK]D-Fender
14:55.17p3nguinOnce you learn how to do it without a GUI, then you can learn how to use the GUI if you think it would make you more productive.
14:55.20fuxu2fender: I always find a LART stick works better
14:55.22l2trace99is there a threading issue that top incorrectly  reads ?
14:55.25[TK]D-Fenderlocks down a death-grip on his ClueBat
14:55.35[TK]D-FenderKatty: MY PREEEEEECCCCCIIIOOOUUUSSS!!!!!!!!!
14:55.38p3nguinl2trace99: try htop
14:55.50Katty[TK]D-Fender: play nice!!
14:55.54Kattygives cluebat back to [TK]D-Fender
14:56.05Naikrovekyou guys always think gui = stupid.  freepbx shows WHAT can be done, then you get interested in how to do it via conf files.  that how it worked for me, anyway
14:56.24[TK]D-FenderNaikrovek: No... Trixbox is a particularly flaming piece of shit.
14:56.35Naikrovekyes, well i'm talking of freepbx
14:56.35[TK]D-FenderNaikrovek: With f-ing sprinkles on top
14:56.46mbrevdaanyone know where I can find a tiny weeny bit of documentation on sip fax detection in asterisk 1.6.2
14:56.49*** join/#asterisk Juggie (n=Juggie@99.224.81.113)
14:57.00p3nguinOrdinarily, if someone learns by way of GUI, they rely on said GUI and never bother to learn to do it without said GUI.
14:57.04*** join/#asterisk tripps (n=sean@76.31.197.242)
14:57.12[TK]D-FenderNaikrovek: FreePBX instaled yourself on * compiled yourself off standard sources is fine as long as you can live within the constraints of the GUI
14:57.30[TK]D-FenderYou learn jack-shit from a GUI.
14:57.34mbrevdathinks: unless they become a dev of said gui, in which case they need to learn TWO systems
14:57.47Naikroveki learned a lot from freepbx
14:57.51p3nguinThat's why I say learn on the CLI and then move to the GUI if you think it will make you more productive.
14:58.01[TK]D-FenderEvery idiot I've seen who has tried took 10 times as long to UNLEARN garbage than it would for a newb to learn from scratch
14:58.22Naikrovekyou guys are so polarized on the gui thing, there's no way you can be correct
14:58.35Naikrovekit's just not black and white
14:58.36[TK]D-FenderNaikrovek: Or it really is that evident :p
14:58.52mbrevdaNaikrovek: +1
14:58.58mbrevda+1, +1, +1!
14:59.06Naikrovek[TK]D-Fender: that's unlikely, i KNOW in my case, the gui taught me a lot
14:59.07jaytee"Less filling!!!"
14:59.16jaytee"Tastes great!!!"
14:59.17Katty:<
14:59.30p3nguinI've never seen a Linux SysAdmin that relied heavily on any GUI.  It's always the Windows fanboys who can't handle the command line.
14:59.46[TK]D-FenderNaikrovek: before learning *, or after?
14:59.53mbrevdarubs his hand gleefuly waiting to take a big warm bite out of a delecious flame war
14:59.57[TK]D-Fendertakes jaytee's beer away
14:59.58Naikrovekand it's always linux cmdline extremists who call windows users "fanboys"
15:00.07Gumugif i can setup the system for a lot less than a turnkey solution, i'll look like i walk on water
15:00.09mbrevdalol
15:00.10Kattygives jaytee an amberbock
15:00.16Kattyjaytee: don't drink crap!
15:00.18Naikrovek[TK]D-Fender: in tandem
15:00.23jayteesome people like GUI based voip systems while most of us don't. Some people like drinking Miller Genuine Draft while the rest of us don't care for the taste of recycled urine.
15:00.34[TK]D-FenderNaikrovek: I'm talking post.  tandem/post can be OK.
15:00.37Naikrovek[TK]D-Fender: as i saw things in freepbx, I was curious how to do those thigns without freepbx
15:01.02KattyNaikrovek: i did that too (=
15:01.03[TK]D-FenderNaikrovek: Anyone who lets the perverted terminology from GUI's sit in their head are on a LONG road back...
15:01.08KattyNaikrovek: that's how i found asterisk-stat
15:01.17Kattytho not technically part of 'freepbx'
15:01.27p3nguingumug: Cost of hardware + cost of man hours.  Asterisk is free, so you only have to spend for hardware and time.
15:01.37Gumugp3nguin: yes
15:01.38[TK]D-FenderNaikrovek: And yes, bright people starting earlier on can do fine.  the problem is that people are dumb, and GUI's tend to make people lazy....
15:01.40Naikrovek[TK]D-Fender: I agree, but to say that using a GUI automatically makes you stupid is incorrect.  the converse, to say that being stupid makes you use a gui, that's usually correct in my experience
15:01.57Gumughardware i don't think, if i build it all myself
15:02.06Gumugwill cost nearly as much as purchasing a turnkey
15:02.08Naikrovekbut don't get caught in the notion that a mouse dumbs you down
15:02.16NaikrovekGumug: much less
15:02.33Gumugdo i have to buy a special telelphone like card
15:02.40NaikrovekGumug: my * server cost maybe $1500, and could probably handle 500 simultaneous calls
15:02.45[TK]D-FenderNaikrovek: directly no, but it is a common byproduct to expect to be able to jsut click your way through everything and whine when there is no "go" button
15:02.46Gumugwhoa
15:02.57p3nguingumug: It depends on how you plan to configure things.
15:03.02MWEi put someone in the meetme room with the flag b (run a background script) that script calls somebody with dial and had to set that user in the same meetmeroom. But instead of putting the callee in the meetme room there will be a bridge between the caller and the callee. When the callee hangsup the script is trying to put him again in the meetme room.. IS there a way that the background script is doing the follwing: call the callee and put him in
15:03.02MWEthe meetme room with a pincode instead of connect him directly with the caller
15:03.06NaikrovekGumug: if you want to plug into physical phone lines, yes, if you want to buy voip service from smoeone, no special hardware is required
15:03.14GumugVOIP
15:03.33Kattywhat's infobot's trigger for voip telcos?
15:03.37[TK]D-FenderGumug: You only need special cards to interface with physical lines yourself
15:03.42[TK]D-Fender~itsplist-us
15:03.43infobotsomebody said itsplist-us was Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net
15:03.54[TK]D-Fender~itsplist-ca
15:03.55infobot[~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.ca , http://www.les.net , http://www.babytel.ca
15:03.56Gumugok
15:03.56p3nguingumug: I run strictly SIP phones with a SIP trunk.  All I need is a regular computer with adequate resources.
15:04.10KattyGumug: we have a pri and bandwidth.com channels.
15:04.19Naikrovek[TK]D-Fender: yes, fools like guis.  but guis don't make fools.
15:04.29p3nguinsometimes
15:04.49KattyThere are some situations where a GUI can be useful.
15:05.02KattyFor example, Katty is on vacation. and Regular people need to make changes or alterations.
15:05.06[TK]D-FenderNaikrovek: No, but give people TV's and watch them slowly devolve into couch-potatos :)
15:05.28KattyAnd Katty refuses to answer the phone while on the beach at Fiji.
15:05.32KattyExcellent reason to have a GUI.
15:05.34[TK]D-FenderKatty is talking about Katty's self in the third-person again...
15:05.39KattyShe does that a lot.
15:05.40[TK]D-Fendersuggests upping the meds
15:05.50Kattythreats [TK]D-Fender with the bat.
15:05.53TJNIIsmirks as Naikrovek's comments about fools and GUIs after he so firmly defended his using a GUI
15:05.55Kattys/threats/threatens/
15:06.24[TK]D-FenderTJNII : Its the user, not the tool.
15:06.26NaikrovekTJNII: allow me to show you the half dozen putty windows I have open
15:06.31[TK]D-FenderUnless the user IS a tool :p
15:06.35Naikrovekthank you
15:06.35jayteeoh no! they're gonna do a reboot of Highlander
15:06.36[TK]D-FenderWhich happens a lot
15:06.38Naikrovekthat's what i'm trying to say
15:06.48TJNIINaikrovek: I just find it amusing, that's all.  No need to get defensive. :P
15:07.13[TK]D-FenderNaikrovek: You are currently on "exception" status with me.... can't be said of most GUI users...
15:07.16Naikroveki prefer command line for most things, but sometimes, for a quick thing, gui IS faster
15:07.20mbrevdaNaikrovek: so use PuTTY COnection mamanger
15:07.22Gumug$30 is a lot less than our $60 we pay
15:07.35GumugVoIP compared to analog
15:07.38*** join/#asterisk jcape (n=jcape@209.120.251.81)
15:08.57KattyGumug: you can also usually get a 'break' with volume orders.
15:09.04KattyGumug: porting numbers is pretty easy too
15:09.44Kattyeppigy: i saw this and thought of you. http://imgur.com/ePnzw.jpg
15:11.26SuPrSluGmbrevda: give poderosa a whirl
15:11.33p3nguinport rinds?
15:12.08SuPrSluGis that the liitle left overs at the switch
15:15.20scalex000TK: I advance I little ha ha ha
15:18.27*** join/#asterisk m0bius (n=mobius@mailexchange.realize.gr)
15:18.33*** join/#asterisk jmacz (n=jmacz@190.144.75.22)
15:21.52*** join/#asterisk |Cybex| (n=John@80.100.126.176)
15:22.00Kattywhy is it so quiet? :<
15:22.03Kattypokes people.
15:22.09p3nguinack
15:22.16*** join/#asterisk wcselby (n=wcselby@216.110.88.254)
15:22.44p3nguinwcselby: I just learned another one of those words I can't type right.
15:22.50wcselbywhat was updated in the topic?  the astricon discount code?
15:22.55wcselbyp3nguin - what's that?
15:23.48p3nguinwcselby: "port rinds" ... I apparently can't type "pork" correctly.
15:23.55wcselbyhaha
15:24.11p3nguinI never knew about that one before.
15:24.29wcselbylol yeah
15:24.47[TK]D-Fenderport rinds... alcoholic pig?
15:25.01wcselbyi was thinking along the same lines [TK]D-Fender
15:25.07*** join/#asterisk volker- (n=volker@h1311547.stratoserver.net)
15:25.15wcselbypork rinds made that have been marinated in an overly sweet wine
15:25.20volker-hi
15:25.25[TK]D-FenderI'd rather have a bottle in front of me than a frontal lobotomy...
15:25.27wcselbyhowdy volker-
15:26.00volker-I have questions about the SIP Protokoll. And I read the corresponding rfc
15:26.06p3nguinIt's just one of those things, I guess.
15:26.25*** join/#asterisk moy (n=moy@74.12.131.104)
15:26.29volker-I dont know any other irc channels about this topic so I drop my questions here
15:26.56wcselby~ask
15:26.57infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
15:26.58volker-1. does authentification needs encryption?
15:27.12volker-2. how secure is MD5-sess and token?
15:27.48*** join/#asterisk maour (n=gnu@unaffiliated/maour)
15:28.28Kattyhttp://farm1.static.flickr.com/88/247766036_fdeb942446_b.jpg <- check out THAT wiring job.
15:29.17p3nguinYou did a marvelous job with that wiring.
15:29.29Katty:>
15:29.42Kattyi love the picture sub-reddit.
15:30.22wcselbyvolker- - I'm not sure, but I don't think SIP needs encryption for auth
15:30.42wcselbyvolker- - at least not as far as asterisk is concerned
15:31.48wcselby[TK]D-Fender will probably correct me if I'm wrong
15:32.25volker-wcselby: if it needs encryption it should have a "algorithm=none" in the rfc. but it doesn't. I cant dig the information out of the RFC :(
15:32.39Kattyhttp://imgur.com/JjQ2E.jpg :>
15:32.41volker-wcselby: errr, if it needs NO encryption :)
15:32.48Katty"MY FISH"
15:33.06wcselbyKatty - I was thinking someone went otter fishing with really big bait
15:33.57volker-i'm not totaly sure how secure md5 is in connection with sip, can someone tell me?
15:35.15*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
15:35.43Kattydoes anyone here like Lentils?
15:37.22coppiceI've bean eating them for years
15:37.29Kattyhow do you fix them?
15:37.40[TK]D-Fendercoppice: I can't stand much more of this PUNishment....
15:37.46coppiceI don't eat the broken ones
15:37.49*** join/#asterisk errr (n=errr@fedora/errr)
15:38.00Kattyfacepalm
15:40.32SuPrSluGKatty: kinda like risotto, use chicken broth and cook em adding as much as needed.
15:41.31Kattylet me rephrase my question--I know how to cook them.
15:41.34SuPrSluGKatty: or make soup out of em
15:41.43Kattywhat is your prefered recipe that uses them.
15:42.35*** join/#asterisk e4 (n=e4@rrcs-76-79-48-214.west.biz.rr.com)
15:42.38volker-thanks for your help
15:42.39volker-bye :)
15:44.47SuPrSluGa side with lamb
15:45.01CareBear\is off - later people! thanks for the help so far!
15:45.03*** part/#asterisk CareBear\ (i=peter@stuge.se)
15:45.22Kattymkay.
15:45.29Kattynot sure where i can find lamb around here.
15:45.36Kattyit's not a very commonly carried item.
15:46.30[TK]D-FenderKatty: dal makhani
15:46.56Naikrovekmmmm lamb chops
15:47.29wcselbyNaikrovek - you said you have polycom ip 6000's, yes?
15:47.30*** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361)
15:47.43Naikroveki compiled and installed * 1.6.1 from svn a bit earlier - where did 'core show translation' go?  no workie
15:47.46Naikrovekwcselby: yes
15:47.58Katty[TK]D-Fender: is this like...
15:48.01Katty[TK]D-Fender: refried beans?
15:48.13wcselbycan I see how you've got your sip.conf setup for it, and maybe get a copy of your chone configs?
15:48.17[TK]D-FenderKatty: Nope
15:48.17Kattyit looks good, from google images.
15:48.20wcselbyi need to compare what's going wrong here on my end
15:48.29[TK]D-FenderKatty: <3 Indian
15:48.33Naikrovekwcselby: what's the problem
15:48.46*** join/#asterisk jplank (n=GBove@cpe-098-024-169-236.carolina.res.rr.com)
15:48.57e4Apparently overnight our Grandstream phones stopped registering.  It looks like they are sending sip packets to the asterisk server but asterisk logs aren't showing any activity and a tcpdump doesn't reveal any return traffic.  I'm not quite sure where to look next, any ideas?
15:49.09Naikrovekmm grandstream
15:49.14Naikrovekthere's a steaming pile of phone
15:49.18wcselbyNaikrovek - the phone registers, you can call it, it can call out, for about three to five minutes.  then it loses registration with the * server and no more workie
15:49.20*** join/#asterisk aaronr (n=arussell@cpc3-stkn13-2-0-cust249.11-2.cable.virginmedia.com)
15:49.37Naikroveke4: check any firewalls or anything, sometimes they can be blocked by adaptive software
15:49.54e4Naivkrovek:  No firewall on this interface.
15:50.03Naikrovekwcselby: loses registration?! during a call?  wow
15:50.09wcselbyyeah
15:50.11Naikroveke4: are your phones and your asterisk server on the same subnet
15:50.16Naikrovekwcselby: same question
15:50.21e4Naikrovek:  Yes
15:50.22*** part/#asterisk aaronr (n=arussell@cpc3-stkn13-2-0-cust249.11-2.cable.virginmedia.com)
15:50.34Katty[TK]D-Fender: traditionally, are those Lima beans or Kidney beans?
15:50.37e4I do know enough about networking to ensure that's not the case ;)
15:50.42Naikroveke4: okay cool
15:50.57wcselbyNaikrovek - yes.  I had a sip debug yesterday but can't find it.  I'll make a new one today.
15:50.58Naikroveke4: how many phones
15:51.12Naikrovekwcselby: pm me your email and i'll email you my config files
15:51.13e4Naikrovek:  Just 3, we've got a small setup.
15:51.24*** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net)
15:51.32[TK]D-FenderKatty: "Other"
15:52.56wcselbye4 - you've of course rebooted the phones, done a reload on asterisk, etc?
15:53.51e4wcselby:  Yep.
15:56.05e4The thing that I'm stuck on is that Asterisk isn't showing me *anything* in the way of logs, be it rejected connections, failed registry attempts, anything.
15:56.12Kattyponders lunch
15:56.47leifmadsengoes to get lunch!
15:56.49[TK]D-Fendere4: Global SIP debug @ * CLI shows nothing when you reboot a phone?
15:56.52Gumug$11k for fonality solution..
15:56.56Naikrovekone thing that would be nice on polycom phones.  if you could determine the model of phone from the web interface
15:57.09p3nguinwcselby: Sounds as if you're experiencing something equally as annoying as a problem I've been trying to solve.  I've got this one computer which uses a softphone... both twinkle and ekiga can register to * and can receive calls, but neither of the softphones can make calls from that computer.  Zoiper on a Windows box on that same remote network works perfectly, though.
15:57.10Kattyleifmadsen: :<
15:57.13Gumugis $245 per phone a lot?
15:57.14Kattyleifmadsen: TAKE ME WIF YOU
15:57.15wcselbye5 - have you enabled sip debug on one of the xtensions?
15:57.27KattyGumug: depends on the type of phone. desk phone--yeah that's a bit high.
15:57.32Gumugdesk phone
15:57.40KattyGumug: polycom 330s are around 130ish
15:57.48KattyGumug: which is your basic Desk Phone
15:57.49Gumugthose are 2 line phones
15:57.52p3nguinless if you don't want a power supply with it.
15:57.58Gumugwhat about 3 line phones
15:58.02Gumugor 4
15:58.04Kattylines are irrelative.
15:58.07[TK]D-FenderGumug: You don't need
15:58.09Kattyjust put your phone in a group.
15:58.15Kattyand assign as many lines to the group as you want.
15:58.16Gumugok
15:58.21Gumugsounds good
15:58.22Kattyit just picks up the next free one
15:58.32Gumugdoes asterisk have a HUD?
15:58.33Naikroveknone of that "pick up line two" BS
15:58.35Gumugweb hud?
15:58.42NaikrovekGumug: freepbx does :)
15:58.43Kattythere are some 3rd party HUDs.
15:58.48Gumugi need a HUD
15:58.53KattyHUDlite, Isymphony (I'm a big fan) and FOP.
15:58.56KattyFOP isn't so Hot.
15:59.03Kattyaka, Flash Operator Panel.
15:59.03Naikrovekyeah fop sucks
15:59.18NaikrovekGumug: these are what i use: http://www.telephonydepot.com/Catalog/Polycom-Phones/Polycom-Soundpoint-IP-321
15:59.21Kattyit does an Okay job if you're just looking at stuff. it's a bit quirky. not much interaction.
15:59.22e4<[TK]D-Fender>:  Correct :/
15:59.30KattyIsymphony works with the manager. Lets you transfer calls around via drag and drop.
15:59.36Gumugnice
15:59.43KattyWe've been using it for about a year or two now. I really like it.
15:59.48KattyIt's Pretty(tm)
15:59.55[TK]D-Fendere4: Dump your firwall, prove that * has bound the SIP port, etc
15:59.58Kattyalso has support for a jabber server, which lets you IM.
16:00.13Kattyyou can have a 'free' version for 5 clients, limited features. 20 bucks a client for the pro version. i think it's well worth it.
16:00.19Naikrovek[TK]D-Fender: he says he doesn't have a firewall on that interface
16:00.22Kattydoes call recording, call barging with little buttons.
16:00.23Naikrovekbut i dunno...
16:00.29Naikrovekiptables is a monster and i hate it
16:00.30[TK]D-FenderNaikrovek: says != proof
16:00.37Naikrovek[TK]D-Fender: agreed
16:00.55[TK]D-FenderNaikrovek: and iptables has never failed me
16:00.58NaikrovekGumug: those phones are two line phones, with up to two calls per line
16:01.08Gumugya
16:01.15Kattyi wouldn't use a 330 for a receiptionist.
16:01.29Gumugsometimes we get slammed
16:01.31Kattytheir display isn't very big. it's Annoying to transfer calls around.
16:01.36Naikrovek[TK]D-Fender: it's a good firewall, i've just not been able to make it do what i want.  all my machines are unfirewalled and i do security on the border, but not sure if that's a good idea or not
16:01.36Gumugbut that shouldn't be a problem i guess
16:01.39Gumugif 4 people call
16:01.51Kattyif i recall correctly it displays the first caller.
16:01.55NaikrovekKatty: yeah the 330 is not a receptionist phone if you have a busy receptionist
16:01.57[TK]D-FenderNaikrovek: Common argument says "not".
16:01.57Kattythen you have to scroll down to highlight calls after that.
16:02.08Kattyit's just annoying.
16:02.19Naikrovek[TK]D-Fender: awesome.  i still hate iptables though
16:02.21Kattyour receiptionist has something in the 500 line i think.
16:02.30Kattyan older 501 i think
16:02.42Naikrovekthe 601 has some nice expansion panels.
16:02.43Katty450 might have a bigger display. can't recall.
16:02.50p3nguinWhat if iptables had a GUI?  :)
16:02.54Kattyyeah, the 600 line has side cars.
16:02.55[TK]D-FenderNaikrovek: You're quite welcome to.  I know most networking people love pf a heck of a lot more, but I never got around to learning the BSD's
16:02.57Naikroveki still wouldn't lke it
16:03.10Kattywe have no need for side cars here...we use isymphony.
16:03.23wcselbyif I've got a border firewall I dump iptables first chance I get
16:03.25Naikrovekiptables makes sense but i don't know what i'm doing with it yet i guess
16:03.31*** join/#asterisk diatonic1 (n=chillman@mail.clearwater-research.com)
16:03.32Kattymost of the time she just clicks and drags a call to an extension and rarely touches the phone
16:03.33[TK]D-FenderPolycom IP60X = discontinued
16:03.35[TK]D-FenderAvoid
16:03.43*** join/#asterisk _trine (n=psybnc_a@trine1-1-pt.tunnel.tserv5.lon1.ipv6.he.net)
16:04.01p3nguinnaikrovek: Set default policies to DROP.  Add rules to allow what you want.  Done, enjoy.
16:04.06Kattysome people like snomphones.
16:04.10Kattybut they feel cheap.
16:04.20Kattygot one upstairs in the bedroom at my house. just don't like it.
16:04.34_trinehow can I change the smtp port from 25 to something else in asterisk is there a .conf file where you can change it?
16:04.57p3nguin* runs an smtpd?
16:05.05coppiceKatty: they are big on ugly, but the seem more solidly built than most
16:05.06Naikrovek_trine: why would you change it, and why would you have it running on a network port
16:05.07Katty_trine: my guess is that would be your email configs.
16:05.20diatonic1Hey all... I'm wanting to modify sending caller ID when I call a certain area code (in the US). Can anyone point me somewhere that might have the info on how to modify the dial plan? I'm running Trixbox/FreePBX
16:05.32Kattycoppice: it might be that i'm used to a IP501, and this is a snom 190(i think)
16:05.35coppiceKatty: they actually fix reported bugs, too, which is almost unique in this industry
16:05.37wcselbyokay, I've unplugged a sip phone, then turned on SIP debugging for that extension, with the intent of grabbing a SIP debug from boot-up to when my problem occurs.  However, before I plug in the phone, the cli is showing me tons of SIP traffic to the IP address of the phone that's not plugged in....
16:05.38Naikrovek_trine: but to change it would depend on the mailer that is running
16:05.44_trineI want to send the voice messages to my email address but I donr use the standard port
16:05.53Naikrovekwcselby: ip conflict?
16:05.57e4<[TK]D-Fender>:  http://pastie.org/private/dsbebmbdaa2p5fynsazeq
16:06.18Katty_trine: asterisk uses a smtp server to send the messages. it's not an smtp server in and of itself.
16:06.21_trineit's using the default asterisk mailer
16:06.26wcselbyNaikrovek - shouldn't be.  I'm investigating
16:06.36*** join/#asterisk carrar (i=tim@osburn.com)
16:06.46Katty_trine: you need to figure out what it's using and then edit that conf file.
16:06.59_trineKatty: thanks
16:07.11[TK]D-Fendere4: FLUSH THEM
16:07.36wcselby_trine - asterisk doesn't have a default mailer.  It's using sendmail if you haven't setup anything else on the box.
16:07.43_trineI have asterisk running on a pendrive which in turn boots my router
16:07.48e4<[TK]D-Fender>:  Yes, done about a dozen times just in case, heh.
16:07.57_trineI have not set sendmail up
16:08.07wcselby_trine - so go read up on making changes to that.  * does not have a built in smtpd
16:08.09p3nguinMTAs talk to each other on port 25.  You sure you want to change the port?
16:08.18wcselby_trine - sendmail is setup by default on most linux distro's....
16:08.22[TK]D-Fender_trine: * doesn't have a mailer.  It jsut calls a standard sendmail script.  mod it however you want
16:08.24l2trace99mmmmm voip mtas
16:08.35*** join/#asterisk Tim_Toady (n=moi@adsl290-154.kln.forthnet.gr)
16:08.52_trineOK it looks like I will have to install sendmail on my router
16:08.58p3nguinWhy?
16:09.23diatonic1_trine You should have an MTA on the * box - probably sendmail or postfix
16:09.27p3nguinRouters should, well, route.  They usually don't need to be a mail relay.
16:09.41wcselby_trine - I'm not sure you fully understand what you're doing....
16:09.57p3nguinI think you can be positive on that, wcselby.
16:10.04diatonic1_trine see of there is a sendmail.cf file on your * box - if so, you've probably got sendmail
16:10.06GumugiSymphony the best one?
16:10.08_trinewcselby: I not sure I'm actually here so don't worry
16:10.10_trine:P
16:10.22wcselby_trine - check the voicemail.conf file and see what file is being called to send emails
16:10.31p3nguin_trine: Just tell us the problem you're experiencing and what you THINK you want to do to solve it.
16:10.46*** join/#asterisk e4 (n=e4@rrcs-76-79-48-214.west.biz.rr.com)
16:11.18KattyGumug: there's no such thing as The Best
16:11.28Gumugthere needs to be
16:11.31KattyGumug: i'd highly recommend giving them all a try, and finding out what your people like best.
16:11.35Gumugwould make my decision easier
16:11.36_trineI want to send the voicemail received to me email address but I forward port 25 on my router for other reasons
16:11.43*** join/#asterisk carrar (i=tim@osburn.com)
16:11.46_trine/s/me/my
16:11.53KattyGumug: one size does not fit all.
16:12.10_trineso for example I want to send the email to port 4444
16:12.17Gumugsee
16:12.18p3nguin_trine: Port forwarding is for inbound traffic to bypass the firewall.  If your * box is behind that firewall, you don't need to forward anything.
16:12.25Gumugfonality is going to charge $2000 for a HUD
16:12.41Kattythey will also support it.
16:12.44_trinep3nguin: I do
16:12.44Kattyconfigure it. and install it.
16:12.50Gumugtrue
16:13.03p3nguin_trine: Your * box does not have an smtpd, so it cannot receive emails on port 25.
16:13.07Kattyisymphony is around 20 bucks a seat license.
16:13.17Kattythey won't install it for you.
16:13.18Gumugya
16:13.22Kattybut they do offer support.
16:13.23e4[TK]D-Fender:  That was it apparently, is that a fairly common thing?
16:13.23Gumugi know
16:13.24NaikrovekKatty: one size fits none usually
16:13.29coppiceThere are better massage chairs than the iSymphony
16:13.32_trineI don't want to receive any emails on my router
16:13.32Gumugalright i'm out of hear
16:13.36NaikrovekGumug: laters
16:13.39Gumugthanks for the help guys
16:13.39p3nguin_trine: So unless your * box is outside of the firewall and your mail server is inside the firewall, this is a non-issue.
16:13.39KattyGumug: byebye
16:13.41Gumugi'll be back later
16:13.44Gumugi appreciate you all
16:13.46_trineI just want to send them out on a port other than 25
16:13.54Gumugciao
16:14.09[TK]D-Fendere4: People putting STATE BASED rules for **UDP** and running into crazyness?  It happens.  I cname this one guy I know...
16:14.19_trinep3nguin: by box is not really a box it's a router
16:14.29_trinewith asterisk running inside it
16:14.34Naikrovekcoppice: what are some others
16:14.44p3nguin_trine: That's irrelevant at this point.
16:15.01_trinep3nguin: thanks for your help
16:15.19p3nguin_trine: * will send emails out via the local mailer daemon.  Port forwarding is not needed for that to happen.
16:15.44_trinep3nguin: you obviously are not understanding my needs
16:15.46p3nguinPort forwarding is not required for outbound traffic.
16:15.50e4[TK]D-Fender:  Is an iptables accept all state-based?
16:16.24Naikrovek_trine: incoming ports and outgoing ports are different.  mail doesn't LEAVE a machine via port 25, it ARRIVES at a machine on port 25
16:16.28p3nguinIf I'm not understanding it, it is because you haven't adequately described the problem.  I'm trying to guide you into the right path, but you're looking down the road where you currently are.
16:16.30[TK]D-Fendere4: ACCEPT     all  --  anywhere             anywhere            state RELATED,ESTABLISHED  <--------  BIG PRINT
16:16.32*** join/#asterisk jmacz (n=jmacz@190.144.75.22)
16:16.39Katty_trine: pretty much everyone in here is a volunteer. if you feel your Needs are not being taken care of you might wait awhile and ask again later. If it's urgent you might consider finding a consultant to help you.
16:16.46[TK]D-Fendere4: accept all.. WITH BIG ADVERTISED CONDITIONS
16:16.49p3nguinActually, mail does leave on port 25.  But port forwarding has nothing to do with that.
16:16.51_trineNaikrovek: I inderstand that well lets just say 110 then
16:16.54wcselby_trine - when you say you're running a router with asterisk inside of it, what do you mean.  what is your router?
16:16.57*** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net)
16:17.14*** join/#asterisk mumtazah (n=mumtazah@60.53.139.144)
16:17.28Naikrovekwcselby: some dd-wrt firmwares have asterisk in them
16:17.35_trineI have installed asterisk inside my router
16:17.35_trineit runs inside my router
16:17.39e4[TK]D-Fender:  Ah, interesting.  I'll dig into that to fix it, thanks.
16:17.43wcselbywhat is your router _trine?
16:17.48*** join/#asterisk carrar (i=tim@osburn.com)
16:17.57wcselby_trine - is it a linux box?  a cisco device?  what?
16:17.58[TK]D-FenderNaikrovek: See?  FW FAIL :p
16:18.06Naikrovek[TK]D-Fender: yeah
16:18.07_trineit is a full asterisk and it's not ddwrt it's kamikaze
16:18.14Naikrovek_trine: ah cool
16:18.16_trineit's a WRT160NL
16:18.18troffaskywcselby, asterisk on a cisco router? what are you on?
16:18.22wcselby_trine what is kamikaze?  never heard of that router brand?
16:18.33troffaskykamikaze is an openwrt release
16:18.33Naikrovektroffasky: it runs well provided you don't need to transcode anything
16:18.35Kattyi think that's a linksys
16:18.35wcselbytroffasky - I know you can't do that.  I wanted to hear what he was trying to do
16:18.35_trineall running of an 8G pendrive
16:18.39p3nguinSounds like a firmware for a Linksys router.
16:18.49troffaskylike karmic is an ubuntu release
16:19.04Naikroveki said that earlier. you can get open source firmwares for linksys routers that have asterisk in them
16:19.13Naikrovekwhich means, your router runs linux
16:19.21wcselbyNaikrovek - thank you.
16:19.26_trinefrom my understanding asterisk has a default mailer ,, is that correct
16:19.30Naikrovekso he's got an internal mail server he wants to receive mail on
16:19.34wcselbyNaikrovek - which means he's probably got sendmail installed on the router and doesn't know it.
16:19.41Naikrovekand he doesn't want the router to get in the way of that
16:19.41wcselby_trine - NO THAT IS NOT CORRECT
16:19.42Katty_trine: no. it does not have a Default Mailer built in.
16:19.51wcselby_trine - we've told you this multiple times.
16:19.51Kattywcselby: please be polite.
16:20.02p3nguinStill makes no difference on port forwarding.  There is no internal LAN where the port needs to be forwarded from the outside.
16:20.07Chainsawtakes wcselby's megaphone away
16:20.07Kattywcselby: you never know what sort of day a person is having.
16:20.08wcselby_trine - asterisk uses whatever the default mailer is on your linux distro you're using
16:20.13_trinewcselby: keep your knickers on you'll wet yourself shouting like that
16:20.20[TK]D-Fender_trine: This has been explained multiple times.  Lets see if this one sinks in....
16:20.24Naikrovek_trine: asterisk does not have a mailer of its own.  it uses the operating system mailer
16:20.37Naikrovek_trine: if you're running asterisk on a router, it probably does not have a mailer
16:20.38ChainsawProviding a sendmail binary should suffice.
16:20.44wcselbylaughs
16:20.44p3nguinIf the system has no mailer, configure one.
16:20.53[TK]D-Fender_trine: * calls a SENDMAIL BINARY.  It does not COME with a mailer.  If you don't have a script to process that request and send the mail then you have NOTHING
16:21.00*** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net)
16:21.00wcselby_trine - go to your voicemail.conf file, see what it's using to send mail, then go configure that mailer
16:21.13_trineI'm looking now
16:21.15Kattyk
16:21.16p3nguinAnd stop THINKING about port forwarding.
16:21.25wcselbyKatty - i've said it multiple time, I figured maybe caps would get his attention, since he hasn't paid any attention yet
16:21.28Kattyp3nguin: THERE IS A NO YELLING POLICY.
16:21.39p3nguinkatty: OKAY!
16:21.42Naikroveki still don't get why he wants to send mail from a different port than 25.
16:21.43Kattyp3nguin: KTHX
16:21.53wcselbyNaikrovek - because he's already doing something on port 25.
16:21.56KattyNaikrovek: understanding is not required.
16:22.05KattyNaikrovek: we have to assume he has a logicial reason for this.
16:22.06p3nguindoesn't matter if he's using 25 or not.
16:22.08wcselbyNaikrovek - some ISP's don't allow outbound mail on port 25 from residential connections
16:22.24[TK]D-Fenderp3nguin: I seem to have imprinted on you ;)
16:22.32_trineI'm reasonably sure it said somewhere if you don't want to use the default mailer then un comment this and use send mail
16:22.39Naikrovekbut tcp origination ports have nothing to do with destination ports.  when my mail server sends mail it could do it from port 3456 or 55555, it's the destination port that matters
16:22.44p3nguinwcselby: They do, but they force you to run it through their relays.
16:22.48l2trace99i put my mta on 2525
16:22.50Kattyplease play nice while i'm gone.
16:22.53l2trace99but no one emails me
16:22.54Kattygoes to lunch.
16:23.06[TK]D-FenderI think I'm a clone now... there always seems to be two of me around!
16:23.08p3nguinNo one CAN email you when your MTA is on the wrong port.
16:23.12wcselby_trine - what you're reasonably sure about doesn't matter.  read the conf files and find out for yoruself what you're using
16:23.23Naikroveksuch anger for a humpday
16:23.32l2trace99oh I thought no one loved me
16:23.35p3nguinnaikrovek: MTAs send out on 25 and receive on 25.
16:23.38_trinewcselby: are you a felckled face little kid
16:23.39wcselbychannels [TK]D-Fender
16:23.40l2trace99i must be really popular
16:23.42[TK]D-Fenderwcselby: No reason * would be configured for anything except the generic sendmail compatible binary
16:23.45_trinequiet
16:23.50l2trace99and I missed it
16:23.58[TK]D-Fenderwcselby: And whatever is actually going on behind the scenes is completely irrelevant
16:24.25wcselbyam I freckled face little kid for trying to help you out?
16:24.32[TK]D-Fender12:23]<p3nguin>naikrovek: MTAs send out on 25 and receive on 25. <- close.. they sed TO 25 and receive ON 25 :)
16:24.33wcselbyokay, sure....
16:24.35[TK]D-Fendersend*
16:24.44Naikrovek[TK]D-Fender: thank you
16:24.54p3nguin[tk]d-fender: That's what I said.
16:25.10Naikrovekif you do a tcpdump while a mail is being sent, it will almost definitely not COME FROM port 25
16:25.27carrarcorrect
16:26.08[TK]D-Fenderp3nguin: you said send ON 25.  that is the source port, which is pretty much assuredly NOT 25.
16:26.08wcselbybleh, I've got a SIP debug to filter
16:26.08[TK]D-Fenderp3nguin: separate your SRC vs DST ports :)
16:26.08_trinethis is what I have read in the voicemail.conf :-
16:26.08_trine; You can override the default program to send e-mail if you wish, too
16:26.08_trine;
16:26.08_trine;mailcmd=/usr/sbin/sendmail -t
16:26.08_trine;
16:26.08wcselbyhave fun storming the castle boys
16:26.09Naikrovek[TK]D-Fender: i said the same thing but he didn't listen to me
16:26.09[TK]D-Fendergathers the pitchforks & villagers
16:26.13_trinethat's what seems to have been the confusing bit
16:26.14Naikrovekgets a wheelbarrow
16:26.24Naikrovek_trine: your mail will not come from port 25 anyway
16:26.31troffasky_trine, install whatever package has a sendmail in
16:26.41carrarit comes from the INNERTUBES
16:26.47[TK]D-Fender_trine: * is calling sendmail.  Go get a clue on how to run a mail script.
16:26.55diatonic1_trinemodify the sendmail.cf to use a smarthost on whatever port you need, then rebuild the sendmail.mc file
16:26.59[TK]D-Fendercarrar: I love running those down snowy hills!
16:27.11diatonic1_trine: modify the sendmail.cf to use a smarthost on whatever port you need, then rebuild the sendmail.mc file
16:27.12troffaskydiatonic1, openwrt will not have a sendmail.cf in
16:27.19carrarno snow here yet
16:27.19_trineBut you can understand how it gets confusing when this is written in the .conf file for voicemail
16:27.31troffaskybut there will be a package with a sendmail in, which is what _trine needs
16:27.41Naikrovektroffasky: not for a router there won't be
16:27.47troffaskyyes there will
16:27.56diatonic1Ahh, missed the part about it being a linksys router :)
16:27.57_trineyes there will
16:27.57Naikrovekhe'll have to install new firmware with sendmail in it
16:28.12_trineno I don't need to
16:28.19troffaskyNaikrovek, openwrt has a package manager like every other linux distro
16:28.30troffaskyso you don't need to reinstall the whole thing when you want to add new software
16:28.32_trineNaikrovek: the wonders of kamikaze are yet to be revealed to you :P
16:28.55Naikrovekeh
16:29.06Naikroveklast time i tried openwrt or ddwrt they maxed the cpu of my router
16:29.12Naikrovekso i removed them with extreme prejudice
16:29.19_trineon this router I have 8 gigs of space for packages
16:29.35_trinelol
16:29.41Naikrovek_trine: flash drive on USB?
16:29.47_trineusb
16:29.59_trineit is running as pivot-root
16:30.17troffasky_trine, opkg install mini-sendmail
16:30.43_trineyes thanks troffasky I will
16:31.05_trine; You can override the default program to send e-mail if you wish, too
16:31.15_trineit was that one line that through me
16:31.28_trinein voicemail.conf
16:35.01Naikrovek_trine: did you have to solder in a USB port or does that router have one
16:35.27_trineNaikrovek: no it's all done for you they are great routers for the money
16:35.50Naikrovek_trine: I just got a 310N it has no USB
16:36.03_trineNaikrovek: if you need any help getting it going I will be glad to trade info with you as at the moment I know little about asterisk
16:36.19Naikrovek_trine: k
16:36.22_trinethat's the WRT160NL
16:36.30_trineI can help with
16:36.36Naikrovekah i read about that one.  the L means linux
16:36.52_trineyou can install kamikaze on them
16:37.06Naikroveki should have gotten that one, but best buy doesn't carry it locally and my budget was limited to the gift card i had in my hand
16:37.07_trineat the moment though wifi is not working well
16:37.15_trinebut i don't use wifi on it
16:37.37_trineit runs asterisk very well as a single end user
16:37.56_trineI also have freeswitch installed on it
16:39.25_trineNaikrovek: the good thing about them is in my view is that they have 8 flash 32 ram a 400MHz processor and you can boot into a usb drive and use it as root
16:39.42Naikrovekwonders why his asterisk compiled from source does not have ... ooh make menuconfig to turn that on i bet
16:39.46Naikrovek_trine: nice
16:39.53_trineso you are never stuck for space
16:40.39*** join/#asterisk saint_ (n=templar@pdpc/supporter/base/saint)
16:41.14*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)
16:41.31_trineNaikrovek: you need to update the packages with :- ./scripts/feeds update -a, ./scripts/feeds install -a
16:41.35*** join/#asterisk ingenius (n=alektro@host172.190-231-93.telecom.net.ar)
16:41.49_trinethen you get all the other packages
16:42.18_trinethen make menuconfig
16:43.28Naikrovekfeeds?
16:43.33*** join/#asterisk Gumug (n=Gumug@adsl-75-39-194-126.dsl.spfdmo.sbcglobal.net)
16:43.39Gumuganyone used PBX in a flash?
16:43.53NaikrovekGumug: yeah it's not so great from what i understand
16:44.00Gumugok
16:44.05Gumugwhat do you recommend
16:44.34Naikrovekwell if you want to use a distro, asterisknow.  but i would recommend that you install asterisk on a linux system you like rather than that, even
16:44.49Gumugi am indifferent
16:44.52Gumugi literally
16:45.01Gumughaven't run linux since the kernel 2.0.36
16:45.11Gumugthat was the last kernel i ever compiled
16:45.35Naikrovekwell if this is something you want to get into, if you think you may be interested in this telephony thing, i would compile it from source and do it all manually
16:45.47Naikrovekit'll be a much simpler configuration when you're done
16:46.30Gumugnod
16:46.44Gumugwhat is the difference between asterisk and fastswitch?
16:46.50Naikrovekfreeswitch
16:47.07Gumugsorry yes
16:47.13Naikroveknot sure, never used freeswitch.  I believe the consensus is that it's pretty awesome, but doesn't have all the features of asterisk
16:47.24Naikroveki've never used it
16:48.11Gumugright on
16:48.13Gumugafk few
16:50.17saint_is there a physical tool beside wireshark that can be used to analyze SIP traces ?
16:50.51voipmonkfastswitch, nice name
16:51.40dustybinno work tomorrow
16:51.41dustybin:D
16:56.21diatonic1Can anyone tell me how to modify the dial plan based on destination area code? (* newbie) Rinning Trixbox CE 2.4
16:56.23*** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
16:56.34Naikrovekdiatonic1: ask in #trixbox
16:56.54Naikroveki'm in there, i'll help you.
16:57.10Naikrovekbut lets do it in there to keep this channel clear of argument
16:59.38[TK]D-Fenderputs away his rusty nail embedded ClueBat (tm)
17:05.42scalex000TK: do you have document about H323
17:05.43SuPrSluGsaint: you could use ngrep
17:08.37[TK]D-Fenderscalex000: http://en.wikipedia.org/wiki/H.323
17:09.58scalex000TK: thanks, and document of h323.conf
17:10.40[TK]D-Fenderscalex000: Go read the sample configs
17:10.44tzafrir_laptoph323.conf.sample ?
17:11.10*** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br)
17:13.09*** join/#asterisk dandate2 (n=gtejkgjk@112.202.55.116)
17:13.39dandate2http://pastebin.ca/1576719  ext 218 is broken and cannot receive calls from queue or dial out, attemps to reach her ext-ext result in busy signal. any ideaS?
17:15.06[TK]D-Fenderdandate2: Show a failed call.
17:15.09*** join/#asterisk superbeef (n=superbee@74.84.194.4)
17:15.35dandate2crap man we have 10 callers waiting in queue getting told their hold position... i'll try
17:15.45dandate2with ringall...
17:16.02*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
17:16.07[TK]D-Fenderdandate2: Maybe you should should show the QUEUE STATUS while you're at it.
17:16.52jgoohey guys - SPA400's -- I have 3 of them - with 2 the config works, when i setup a third as a trunk, it doesn't load the configuration - why?!! is there some limit?
17:17.04*** join/#asterisk wonderworld (n=w@ip-62-143-22-226.unitymediagroup.de)
17:17.09superbeefI have 2 asterisk boxes linked by a T1 cross over to their sangoma cards......  In a very generic sense, what would I need to do in order to have 1 box ring hte other via the T1s
17:18.00[TK]D-FenderSuPrSluG: Dial(DAHDI/G1/1234567890)
17:18.15[TK]D-Fendersuperbeef: Dial(DAHDI/G1/1234567890)
17:18.33[TK]D-Fendersuperbeef: Assuming you're running DAHDI and grouped your channels as group=1
17:18.48[TK]D-Fendersuperbeef: And taht was actually rather specific...
17:19.09superbeef[TK]D-Fender: Killer... Yeah i'm using dahdi, and i think everthign is group1.....  much simpler than I expected
17:19.10[TK]D-Fendersuperbeef: hard to believe you have 2 probably rather expensive cards and don't know how to make a basic dial command for them...
17:19.25jgoo[TK]D-Fender: why would you think that?
17:19.46[TK]D-Fenderjgoo: who spends thousands without researching the basics?
17:19.47jgoohard to believe that the more expensive a device is, the easier it is to use, or you are expected to already know about it
17:19.58superbeef[TK]D-Fender: haha.... I have this mental block in my head that thigns are different because I'm not tied to a telco on the other side
17:20.07[TK]D-Fenderjgoo: Its called getting a clue about what you are getting yourself into..
17:20.20jgoo[TK]D-Fender: you have a point there - but sometimes knowing it can do something, and being able to make a path of actions to configure it are two different things
17:20.25[TK]D-Fendersuperbeef: And out of curiousity... why are you linking 2 *'s via T1?
17:20.34superbeef[TK]D-Fender: performance
17:20.36superbeef[TK]D-Fender: just kdiding
17:20.54[TK]D-Fendersuperbeef: Migration testing I could understand...
17:20.56superbeef[TK]D-Fender: I have a lot of these guys in production, but I wanted to build a real test environment
17:21.11[TK]D-FenderSuperbTest envirnment.. sure... why not..
17:21.17dandate2http://pastebin.ca/1576732
17:21.22jgoosuperbeef: you want to simulate incoming calls into one of the boxes?
17:21.31dandate2http://pastebin.ca/1576738
17:21.42dandate2queue 292 fails over to queue 290
17:21.57[TK]D-Fenderdandate2: Where is your peer dump?  SIP debug? Queue status dump?
17:22.12superbeefjgoo: yep... i want to test them to as real world as i can get... I got burned last week deploying a box with 2 T1's cuz I couldnt test before hand
17:22.17dandate2i never used that stuff before honestly
17:22.31dandate2i'm just gettin stuck on a problem in a very practical setup heh
17:22.48[TK]D-Fenderdandate2: BS.. you've been run through this before... "sip show peer 218"
17:23.05dandate2i think it is caused by agent 218 being a member of both queues and tried to be rang by both at once
17:23.52dandate2http://pastebin.ca/1576743
17:24.06[TK]D-Fenderdandate2: For a person who isn't looking, now claiming not to know how to, you sure are "thinking" a lot...
17:24.34[TK]D-Fenderdandate2: New call with SIP debug enabled
17:25.46dandate2hmm my cli says that sip debug is deprecated
17:25.58Naikrovekit also says what to use instead
17:26.13Naikrovekcore sip debug <ip address> if i recall
17:26.34wcselbysip set debug peer <peer> or sip set debug ip <ipaddy>
17:26.41dandate2got it
17:26.42Naikrovekthanks
17:26.42jgoo[TK]D-Fender: have you heard or read anything about asterisk only supporting two SPA400's as trunks? not that they are officially suppoted, but damn that 9000 is only useful as a DHCP server when doing linux installs
17:27.09Gumughow powerful a machine do i need to handle 20+ calls?
17:27.17NaikrovekGumug: not too powerful at all
17:27.39[TK]D-Fenderjgoo: PARDON?
17:27.40Naikrovekthere's a voip-info page on performance somewhere
17:27.40Naikroveklet me see if i can dig it up
17:27.40jgooI just cannot understand why my setup works... until i add a third trunk - then it doesn't even show the PBX config in freepbx =( (I know you despise that!)
17:27.43NaikrovekGumug: http://www.voip-info.org/wiki/view/Asterisk+dimensioning
17:27.43Gumugcool, i can find it
17:27.46[TK]D-Fenderjgoo: "supporting as trunks"...
17:28.00jgooas in - using the spa400 as a trunk
17:28.04[TK]D-Fenderjgoo: wrong channel <-
17:28.19jgoohaving 4 channels on each trunk, with spa400s
17:28.25jgoo[TK]D-Fender: how is that?
17:28.43[TK]D-Fender13:27]<jgoo>I just cannot understand why my setup works... until i add a third trunk - then it doesn't even show the PBX config in freepbx =( (I know you despise that!) <- because we don't care what freePBX shows
17:28.59[TK]D-Fenderjgoo: Got something real to show us?
17:29.09jgooI don't think this is a freepbx bug, I wonder if anyone is using SPA400's in this capacity, they are damn useful
17:29.14[TK]D-FenderGumug: Any P4 will do
17:29.20KattyEveryone's playing nice still, Right?
17:29.21jgoo[TK]D-Fender: that is the problem, no error messages, nothing in /asterisk/full
17:29.24Gumugalrighty
17:29.28jgooand nothing on over 900 -vvvvvvv
17:29.38[TK]D-Fenderjgoo: No problem?  Then what are you going on about?
17:29.55jgoo[TK]D-Fender: where did I say no problem?
17:30.03jgoothere is no error message
17:30.08Kattysighs.
17:30.11dandate2http://pastebin.ca/1576747 http://pastebin.ca/1576750  about as much as i could humanly grab without finding the putty log
17:30.12[TK]D-Fenderjgoo: Where is the failed call to look at?
17:30.36jgoothe problem is, the configure isn't loading - and I am trying to find where it might dump the errors - or something is going wrong after I add a third trunk, and I cannot get it to reload after that...
17:30.42jgoo[TK]D-Fender: all calls work
17:30.53[TK]D-Fenderdandate2: That is QUALIFY packets, not a CALL
17:31.08Gumugmy fear is i won't have enough bandwidth at the asterisk server
17:31.09jgoo[TK]D-Fender: asterisk won't load config after adding a third trunk - ergo, I cannot test a call, because calls don't work, because config isn't loading
17:31.14Gumugwhat should i do about that?
17:31.26NaikrovekGumug: you'll have enough.  what kind of bandwidth do the offices have
17:31.29KattyGumug: well. you could get more bandwidth.
17:31.30[TK]D-FenderGumug: Zantac and/or vallium
17:31.39jgoo[TK]D-Fender: I am actually pretty good at debugging (or bad, depending on who I am talking to and what about) - if I know where the error messages are, or if google does
17:31.44KattyGumug: Also, i know bandwidth.com estimates how much bandwidth you will need per phone call.
17:31.49Gumugok
17:31.52KattyGumug: i don't know what it is off the top of my head anymore.
17:31.54Gumugwe have DSL
17:31.58Gumugmaybe 6meg plan
17:31.59[TK]D-Fenderjgoo: You don't seem to be able to show a call that has "failed".
17:32.00NaikrovekGumug: plenty
17:32.06KattyGumug: obviously a lot of it depends on codecs.
17:32.11[TK]D-Fenderjgoo: * CLI has everything we care about.  Log files are meaningless.
17:32.12Gumugyes
17:32.17jgoo[TK]D-Fender: you don't seem to be able to read where I just said the calls aren't failing
17:32.20KattyGumug: the nicer the codec, the bigger the stream.
17:32.25*** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be)
17:32.33jgoo[TK]D-Fender: the problem is, after I put up a third trunk, devices won't resgiter
17:32.33NaikrovekGumug: yes but 6mbit is enough for about 40 G711 calls.  about 320 G729 calls.
17:32.36jgoo*register
17:32.39[TK]D-FenderGumug: dsl is fine for your satelite locations.  your main will need something substantially larger
17:32.55[TK]D-Fenderjgoo: If the device won't register.. thats its problem...
17:32.55Gumugthat's what i figured [TK]D-Fender
17:32.57KattyGumug: usually the VoIP providers will have some sort of formula for you to follow, if you want to ask them. Just remember you will need the same speeds both up and down.
17:33.03[TK]D-Fenderjgoo: Isn't anything on *'s side
17:33.07jgoo[TK]D-Fender: I was quite clear -- anyway, no worries, if you don't want to help, but please don't say I wasn't clear, I appreciate your help for myself and others in here
17:33.30[TK]D-Fenderjgoo: Seems clearer now... SPA-side config (potential) issues...
17:33.39[TK]D-Fenderjgoo: Plenty of reasons we don't recommend that model.
17:33.42KattyGumug: if your Smaller Offices only have a handful of lines, I would say DSL will be perfectly fine.
17:34.01jgoo[TK]D-Fender: ok, I did try /sbin/service asterisk stop \ asterisk -vvvvvc - it loads fine - then at one point I was getting 'previous reload hasn't finished' but i googled that to a few dead ends and other problems
17:34.05KattyGumug: well, depending on your upload speed.
17:34.05Gumugya they only have up to 2 lines
17:34.09KattyGumug: you can't forget about that.
17:34.14Gumugoh, i won't
17:34.18[TK]D-Fenderjgoo: if the dev doesn't reg, thats not *'s fault
17:34.20jgoo[TK]D-Fender: is there another similar device to that?
17:34.45[TK]D-Fenderjgoo: Tons.  Audiocodes / Mediatrix start at 4-port and up
17:34.58jgoo[TK]D-Fender: they do reg, right before I configure another trunk, then all 50 devices stop registering, because I think the config is getting blocked while loading perhaps... I am not sure... no errors as I've said
17:35.00Gumugwish i could just have the asterisk server CALL my analog phones
17:35.17KattyWell, asterisk can call analog phones...there's a card for that.
17:35.17jgooAudiocodes / Mediatrix run on 12v?
17:35.25[TK]D-FenderGumug: ATA <-
17:35.30dandate2ok complete from the putty log http://pastebin.ca/1576759
17:35.38[TK]D-Fenderjgoo: dunno what alternative power options they support
17:35.40scalex000TK: I need a command on h323 to connect both pbx
17:35.41Gumughmmmm
17:35.42[TK]D-Fender~ata
17:35.43infobotfrom memory, ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA
17:35.57[TK]D-FenderGumug: Linksys SPA-2102 for example
17:36.15Gumugsome ata's dont' work with others
17:36.22Gumugone company said it would not work if i wanted to use an ATA
17:36.34[TK]D-FenderGumug: "it"?  what is "it"?
17:36.45Gumugthe linksys spa
17:36.46[TK]D-FenderGumug: And who are "they"?
17:36.50Gumugfinality
17:36.58[TK]D-FenderGumug: BULLSHIT
17:37.00Gumugi wanted to know if i could keep my analogs
17:37.03Katty[TK]D-Fender: if you don't be nice, i'm going to staple a smiley face to your forehead.
17:37.03bmoracafonality are a bunch of idiots
17:37.11Gumugthey wanted to sell me 16 phones for $245 a peice
17:37.16KattyGumug: well of course they do.
17:37.20Gumuglol
17:37.22KattyGumug: they want to make money.
17:37.26[TK]D-FenderGumug: Fonality are picky bastards about what they want to support regardless of what works
17:37.26Gumugof course
17:37.33Gumugi see
17:37.34KattyGumug: most likely they will not be able to Support it, if you do it.
17:37.36bmoracaGumug: what kind of phones do you currently have?
17:37.37sfireis trixbox pretty stable?? I just had a bad experience with asterisknow and I need to change it and get the phone system back working
17:37.40*** join/#asterisk momelod (n=smelo@99.231.22.104)
17:37.44momelodgreetings channel
17:37.47Gumug4 line analogs
17:37.56momeloddoes anyone know of a command-line sip client?
17:37.56Gumugi don't know the type
17:37.59bmoracaGumug: are they really analog or are the digital?
17:38.02*** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
17:38.02Gumugi'm not t an office
17:38.14Gumugbmoraca: good question, i don't know
17:38.16Gumugi'll find out
17:38.17[TK]D-FenderGumug: Call them
17:38.24bmoracathat's a necessary bit of info...
17:38.36Gumug[TK]D-Fender: will do, i'm currently in a web demo
17:38.36Kattyplaces a bet they're toshiba phones.
17:38.42*** join/#asterisk hesco (n=hesco@24.99.160.121)
17:38.44superbeefmomelod: for testing, or are you actually want to do voice with it
17:38.49hesco!paste
17:39.00voipmonktoshiba
17:39.02hesco~paste
17:39.03infobothmm... paste is http://rafb.net/paste/, or see also pb, or http://bin.cakephp.org/
17:39.03voipmonkgood lord
17:39.05*** join/#asterisk grEvenX (n=even@193.90.141.183)
17:39.28voipmonkkatty - have u ever configured the toshiba pbx (cant remember the model name ) to do sip and have it work?
17:39.30superbeef~taco
17:39.31infobotTACO TACO TACO!
17:39.35Kattyyes.
17:39.36bmoracaif they're digital, you may be able to get a Portico TVA, which is a media gateway that emulates for certain digital sets...if they really are analog, any analog media gateway (mediatrix, etc) should work.
17:39.42KattyStrataNet.
17:39.46Kattyand QSIG
17:40.05Kattynot sure if it was actually sip tho ^_-
17:40.07[TK]D-FenderNote : Potico = F-ing expensive and only rarely even worth considering
17:40.11momelodsuperbeef: well im setting up a paging system and thought i could use the company music server.  So i want to install a cli sip client on the music server and set it to auto answer.. then when people talk it just plays over the overhead speakers
17:40.43voipmonkStratanet
17:40.56bmoraca[TK]D-Fender: no doubt...but the fact remains that if your building is wired for digital phones and you have hundreds of them already, they are per-port cheaper than replacing the phones and reterminating the cables (provided it wasn't wired CAT3)
17:41.02Kattyyou know, Stratanet...multiple boxes.
17:41.05Gumugbmoraca: thank you
17:41.07Kattymultiple locations.
17:41.25Kattyi know they were voip phones.
17:41.29wcselbyblast it all
17:41.32Kattybut...yeah. idk if it was sip or not
17:41.43wcselbyi just finally got my sip debug cleaned up for my polycom 6000, and it's over 5000 lines on pastebin
17:42.28wcselbyi don't even want to read through all of that
17:42.58[TK]D-Fenderbmoraca: $125 / port to maintain old crap... hard sell.
17:43.26bmoracaif it's the difference between getting a job and not getting it, i'd sell a company a box of crap and tell them it's a "line conditioner"
17:43.37[TK]D-Fenderbmoraca: You can get cheaper phones with passthrough ports, but yes there are cases where its worth it.... jsut rather rare
17:43.37Kattywhere is dave today.
17:43.40hescoMorning all:  Last night the phone was working fine. When I got up this morning I was unable to make any outbound phone calls. Any ideas why?  *CLI> output (verbose 3) and iax2 show (relevant) peer are posted at:   http://bin.cakephp.org/view/1686740662
17:43.49[TK]D-Fenderbmoraca: "shit" is a condition ;)
17:44.41*** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903)
17:45.18Kattygrooves.
17:45.59*** join/#asterisk errotan (n=errotan@62.201.122.123)
17:47.19bmoracagranted, i'm not a big fan of using media gateways and stuff with Asterisk, but sometimes it's a necessity.
17:48.37bmoracain 1.6, does dbsecret work for SIP peers?
17:49.03*** join/#asterisk casnik (n=Nick@fw1-e0-2.dth.xiocom.net)
17:49.23*** join/#asterisk ajohnson (n=ajohnson@65-122-4-130.dia.static.qwest.net)
17:49.54ehsjoarhesco: ping the IAX provider. It seems you have a problem reaching the IAX provider. So it could be a problem on your network or the IAX provider having issues
17:51.19superbeefmomelod: so your music server would be a client to your asterisk box?
17:52.21hescoehsjoar, I finally did a `stop now`, followed by an `asterisk -vvvgc` and it restored services.
17:52.33ehsjoarhesco: Okay, so it is working now?
17:52.40hescocan't imagine what might have changed to create that issue.
17:52.58hescoyes, at least I was able to reach my cell's voicemail
17:53.02momelodsuperbeef: exactly
17:53.32superbeefmomelod: how will the SIP client interract with your music server
17:53.45superbeefmomelod: the Music server is just hooked up to some sort of PA system right?
17:53.53ehsjoarhesco: Seems like for some reason * was not able to reach the IAX provider (or thought so). Strange that a restart would fix it. Perhaps there is a NAT problem somehow
17:55.42*** join/#asterisk tlarsen (n=tlarsen@71.207.223.160)
17:56.34*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
17:56.41Kattyehsjoar: let us rejoice and have ice cream that your issue is fixed!
17:57.23ehsjoarKatty: It was not my issue, but hesco's. Sounds good with the ice cream though:-)
17:57.24Kattyehsjoar: should it happen again, i'm sure the provider can watch the packets that are going back and forth.
17:57.57Kattythat's the spirit!
17:58.51ehsjoarKatty, hesco: Yeah, I am thinking it could be a firewall / NAT problem hesco is facing. The firewall perhaps keep the port open for a while but closes it if there is no traffic. Pure speculation on my side though
17:59.06momelodsuperbeef: actually the music server is mpd and i control it using the mpc client
17:59.38momelodactually i wrote an app for my cisco7960 handsets where my users can control the overhead speakers using the service button on the phone.
18:00.13*** join/#asterisk Micc (n=Micc@c-98-225-59-171.hsd1.wa.comcast.net)
18:00.21*** join/#asterisk jcape (n=jcape@209.120.251.81)
18:00.27superbeefmomelod: see if pjsip does what you need
18:01.58*** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker)
18:07.51*** part/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek)
18:08.30*** join/#asterisk Greek-Boy (n=greek@41.188.154.137)
18:10.47*** part/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda)
18:15.21bmoracai don't think it's possible for Valcom's website to be any more worthless than it is
18:20.00*** join/#asterisk Gumug (n=Gumug@nmd.sbx09566.joplimo.wayport.net)
18:20.22*** join/#asterisk ingenius (n=alektro@81-152-114-200.fibertel.com.ar)
18:24.47[TK]D-Fenderbmoraca: there is very little in life that's so bad that it can't get worse.
18:24.48bmoracaDUNDi does not have a lot of informational elements to it in the CLI, does it?  one would think that it would be useful to be able to output the entire DUNDi route table, but apparently one cannot
18:25.12[TK]D-Fenderbmoraca: Just when you think you've hit rock bottom, that's why caterpillar makes back-hoes ;)
18:25.18bmoracalol
18:25.53*** join/#asterisk xpot-mobile (n=james@173.8.94.1)
18:27.57kaldemarbmoraca: it's in the database, iirc
18:29.02Kattyi have a weird dns issue i can't wrap my brain around.
18:29.26Kattyit makes my voicemail attachments spaz out when sending to user@domain
18:29.32Kattysaid domain is in the same building. on the same lan.
18:29.37Kattybut it doesn't like domain.com
18:29.40*** join/#asterisk _brent_ (n=_brent_@orem.jiveip.net)
18:30.04Kattywe can't open domain.com in a browser either.
18:30.04Kattywhich might be normal. idk. don't know a lot about dns.
18:30.34Kattywe own several domains which all point back to domain.com, so i just made them go to another alias. works fine.
18:30.39Kattyfigure that one out ^_-
18:31.41Kattyexim4's mainlog is talking about MX records.
18:32.06Kattydoublechecks dns server for mx record for email server
18:32.54bmoracakaldemar: which database?  it's not in astdb
18:34.59kaldemarbmoraca: astdb is what i meant. i recall checking the cache out somehow.
18:35.12*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
18:37.32dustybin[TK]D-Fender: its now fully complete: http://www.thinkdebian.org/archives/828
18:38.05dustybintakes cover from the ClueBat
18:40.13[TK]D-Fenderdustybin: [general] of IAX2 should point to a dead-end context.  Phones should always explicitly refer to their context, and you should NEVER make a context named [default]
18:40.31dustybinmakes note
18:40.37Naikrovekdustybin: you make that?
18:40.44[TK]D-Fenderdustybin: exten => _09XX,1,Dial(IAX2/USERID@voiptalk/${EXTEN}) <-- not abstracted
18:41.16[TK]D-Fenderdustybin: In your IVR, at the end of the menu audio it will HANGUP.
18:41.29[TK]D-Fenderdustybin: And an invalid entry will also hangup on them
18:41.29dustybineeek
18:41.55dustybin[FAILS]
18:42.04[TK]D-Fenderdustybin: exten => 1,1,VoiceMail(1000) <- yous hould also always explicitly include the VM context to use, and you might want to consider playing a PROMPT
18:42.15dustybindouble [FAILS]
18:42.24Naikrovekdustybin: i followed the compiling instructions earlier, they worked great; but you missed one little thing.  this is not a fail, btw
18:43.11dustybinNaikrovek: what did i miss?
18:43.55*** join/#asterisk louben (n=lou@212-70-216-131.ath.static.tee.gr)
18:43.56Naikrovekdustybin: guide is awesome.  i had to do a "apt-get install libncurses5-dev" when i installed build-essentials and subversion and the kernel headers.
18:44.30Naikrovekyou could also change "aptitude install linux-headers-2.6.X-X-common
18:44.30Naikrovek" to "aptitude install linux-headers-`uname -r`"  (those are backticks, not single quotes)
18:45.26dustybinright ok!!
18:45.38dustybinthanks!!
18:45.56Naikrovekeffing awesome guide tho, i found it earlier via google i think and bookmarked it not knowing it was you :)
18:46.12dustybinits on google already?
18:46.23Naikroveki dunno, but it's been in my browser all damn day
18:46.30dustybinwell , you have to thank everybody in this channel, thats where i got most of the info from
18:46.36Naikrovekand i don't think i clicked a link in here
18:47.09dustybinim just saving the headache of finding bits and bobs from all over the place
18:47.10Naikrovekyeah it's on google
18:47.10dustybinok
18:47.14Naikroveksecond page whe i search for "debian asterisk howto"
18:47.24*** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:c42:5117:b2c9:8871)
18:47.25cuscohi
18:47.30cuscoI got a problem with dahdi
18:47.44cusco<PROTECTED>
18:47.51dustybincusco: have you intalled mhomi ?
18:47.51cuscoUnloading DAHDI hardware modules:
18:48.10cuscono what is that?
18:48.15dustybin:P
18:48.21cuscoERROR: Removing 'wcte12xp': Device or resource busy
18:48.22dustybinstops trolling
18:49.10Naikrovekdustybin: would have been better if you'd said "have you been touching mohmi?"
18:49.13Qwellcusco: Did you stop Asterisk first?
18:49.20cuscoyes
18:49.47tzafrir_laptopcusco, it hangs for a while or fails immediately?
18:49.52SuPrSluGlsmod |grep dahdi to see what's using it
18:49.57tzafrir_laptopanything ugly in /var/log/messages ?
18:51.02Naikrovekmmm, snack pack.  cures heartburn nicely for me
18:51.24OrbixxHow can I record a call that takes place after a client has queued?
18:51.41*** join/#asterisk voipmonk (n=voipmonk@69.172.93.45)
18:51.41NaikrovekOrbixx: just always record all calls
18:51.47Naikrovek... i think
18:51.57OrbixxHow can that be done? :P
18:51.58Naikrovekdon't think you can turn recording on mid call
18:52.00Naikrovekbut i could be wrong
18:53.02Naikrovekone sec, Orbixx
18:53.13dustybinNaikrovek: i have fixed the install parts, now for the tricky bits
18:53.22Naikrovekcool beans
18:53.34[TK]D-FenderOrbixx: "core show application monitor"
18:53.47dustybin1 [general] of IAX2 should point to a dead-end context.
18:53.51kaldemarNaikrovek: yes you can
18:53.58Naikrovek[TK]D-Fender: can he turn that on when the call changes contexts
18:54.16Naikroveklike when the call enters a context, and that context has recording set to always, will that start recording?
18:54.17[TK]D-FenderNaikrovek: ... calls... change... CONTEXTS?
18:54.22Naikroveki don't know
18:54.44Naikrovekyou pass a call to a queue, doesn't that queue have a context?
18:54.46kaldemarDial options wWxX and featuremaps in features.conf
18:55.20*** join/#asterisk TimToady_ (n=moi@adsl99-59.kln.forthnet.gr)
18:55.28kaldemara context doesn't have anything, an extension may have.
18:55.44*** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl)
18:55.58[TK]D-FenderNaikrovek: you = very confused
18:56.10Naikrovek[TK]D-Fender: on a few things, yes
18:56.16Naikroveki'm no dial plan master
18:56.28[TK]D-FenderNaikrovek: unfortunate since that's 95% of * :p
18:56.37Naikrovekwell i'm working on it
18:56.43Naikrovekgot a lot of S going down at work
18:57.24OrbixxAfter a call ends, can Asterisk email the recording similarly to that of a voicemail?
18:58.11[TK]D-FenderOrbixx: go read up on Asterisk Standard Extensions
18:59.08Orbixx[TK]D-Fender: Right, action stuff on the hangup extension.
18:59.18jayteeevery time I call my girlfriend from my phone in the [normal] context the call works fine but when I use a feature key to change to the [porn] context I get disconnected. She swears she isn't hanging up on me. sip debug shows nothing unusual
18:59.23OrbixxI'm just asking if it's something that Asterisk can do relatively automatically much like voicemail.conf
19:00.02[TK]D-FenderOrbixx: "core show application monitor" <-
19:00.41OrbixxI don't see anything relevant to the question I'm asking.
19:01.12IBC_jkenneyI have new questions today
19:01.29IBC_jkenneyI want to be able to have in the cdr records better detailed such as who hungup first
19:02.12IBC_jkenneywhere the call went IE it came in on DID XXXX went to main menu s 1 then they pressed 1 then it went to the sales queue thenf rom the queue it went to agent blah
19:02.19IBC_jkenneyis this hard already there
19:02.41[TK]D-FenderOrbixx: Read it AGAIN
19:03.02Orbixx[TK]D-Fender: I've read it three times, I'm obviously missing something.
19:03.09_brent_IBC_jkenney: the second part is possible if you keep track of the traversal yourself (assign it to a variable) and then put it in the CDR with cdr_custom
19:03.18[TK]D-FenderOrbixx: "m"
19:03.35_brent_knowing who hung up might be tougher. i'd love to hear it if someone knows the answer to that part.
19:04.10Orbixx[TK]D-Fender: That just mixes the two recordings. I'm asking if Asterisk provides capability to automatically mail the recording to a mailbox much like voicemail does.
19:04.14*** join/#asterisk Methose (n=Methose@38.101.237.250)
19:04.15[TK]D-FenderOrbixx: Read it AGAIN
19:05.37OrbixxI see nothing relevant to my question.
19:05.37[TK]D-FenderIBC_jkenney: the queue log already tells you who hung up
19:06.14[TK]D-FenderOrbixx: "If the variable MONITOR_EXEC is set, the application referenced in it will be executed instead of soxmix and the raw leg files will NOT be deleted automatically."
19:06.57OrbixxRight, so you're implying that Asterisk doesn't do it automatically like voicemail and I need to invoke something like "mail".
19:07.09[TK]D-FenderOrbixx: WOW.
19:07.35[TK]D-FenderOrbixx: the mere fact VM has this functionaliy semi-built-in alone is rather generous
19:07.50[TK]D-FenderOrbixx: * is up to you to configure.
19:08.26OrbixxThe mere fact that Asterisk exists is generous, but I don't see why I'm being unreasonable if Voicemail does something almost exactly the same.
19:08.32*** join/#asterisk |Rain| (i=rain@ev.il.net)
19:08.39OrbixxI agree, but I was asking a simple question.
19:08.49OrbixxI wasn't saying it was outrageous that it doesn't have such capability.
19:09.04OrbixxI just wanted to *know* for sure.
19:09.22[TK]D-FenderOrbixx: call recording doesn't have a config file to define which calls to record, where to send, and how to manage.
19:09.40OrbixxThank you.
19:09.43OrbixxThat's all I wanted to know.
19:10.14OrbixxFor somebody who is ultimately very helpful, it's very difficult to pry the answers out of you sometimes.
19:10.21Orbixx;)
19:10.48[TK]D-FenderOrbixx: I handed it to you multiple times.  you simply could see the only option on that page which clealy let you do something on end of recording.
19:11.00[TK]D-FenderOrbixx: Don't blame me for your lack of creativity :)
19:11.15jayteesome people want their poop to come out gift wrapped it seems
19:11.27[TK]D-FenderOrbixx: You walked into this expecting a "yes" instead of looking at what the pieces already do
19:11.29OrbixxI already realised it was there.
19:11.34OrbixxAnd I already knew I could do it manually.
19:11.49OrbixxI just wanted to know if Asterisk could do what I could do manually - automatically, much like Voicemail.
19:12.18|Rain|so.  Mr. A (100) calls Mr. B (200), they chitchat for a bit, then Mr. B (200) performs an attended transfer of Mr. A (100) to Mr. C (300).  CDRs are logged for 100->200 and 200->300, but is there any way to determine that 100 talked to 300 from CDRs?
19:12.24OrbixxI don't think I'm being unreasonable?
19:12.29[TK]D-FenderOrbixx: Nope, for all the reasons that are evident by the existance and content of voicemail.conf
19:12.43jaytee|Rain|, what is that? An SAT question?
19:13.01[TK]D-Fender|nope
19:13.08[TK]D-Fender|Rain|: nope
19:13.08|Rain|I hope not, because I fail otherwise
19:13.12|Rain|[TK]D-Fender: boo
19:13.19leifmadsenthere probably is with CEL
19:13.26leifmadsenbut not with CDRs afaict
19:13.52dustybin[TK]D-Fender: why should this be: 1 [general] of IAX2 should point to a dead-end context. should i just make up anything and stick it in the [general] like, context=cluebat ?
19:13.56jayteeCEL? typo? for AEL?
19:13.58*** join/#asterisk el_critter (n=critter@200.8.97.41)
19:14.04leifmadsenjaytee: no, not a typo
19:14.11jayteehmmm
19:14.17leifmadsenChannel Event Logging
19:14.17jayteegoes to Google CEL
19:14.19leifmadsen~cel
19:14.19infobotsomebody said cel was Channel Event Logging
19:14.19|Rain|apparently Channel Event Logging
19:14.20[TK]D-Fenderdustybin: Sure
19:14.28dustybin[TK]D-Fender: is this for security?
19:14.33[TK]D-Fenderdustybin: Yes
19:14.36dustybinaye ok :D
19:14.45el_critterAnyone having problems with asterisk hanging the call but dahdi keeping the PSTN line open?
19:14.55[TK]D-Fenderdustybin: So taht un-authed calls don't get unrestricted access to call out... like they do NOW
19:15.49dustybinright ok
19:16.02*** join/#asterisk nullable_type (n=nullable@hq.verbx.net)
19:16.20nullable_typeHow can i play a music file in the background while doing a DIAL to connect two calls
19:16.28*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
19:16.37leifmadsenDial() with the m flag
19:16.38[TK]D-Fendernullable_type: "core show application dial"
19:17.00nullable_typethanks guys
19:17.08*** part/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
19:17.13|Rain|[TK]D-Fender: thanks
19:17.18*** part/#asterisk |Rain| (i=rain@ev.il.net)
19:18.35dustybin2 Phones should always explicitly refer to their context, and you should NEVER make a context named [default]
19:18.54dustybinwhats wrong with [default] ? im pretty sure [default] was used in the book
19:19.16nullable_typeRE:DIAL, so the m option does it repeat the music file, I do not want that...... Also when i want the music to stop when the first leg can hear ring tone....
19:19.19nullable_typeIs it possible
19:20.32nullable_typemay be I should use the A(x) option for DIAL?
19:20.44[TK]D-Fenderdustybin: Yes, and books can have mistakes as well...
19:21.09[TK]D-Fendernullable_type: that you cannot do.
19:21.34[TK]D-Fendernullable_type: there is no pre-ring/post-ring split
19:21.57*** join/#asterisk geneticx (n=chatzill@host-208-88-126-198.biznesshosting.net)
19:22.03dustybinaye ok!
19:22.08nullable_typeD-Fender, But does the m option loops the file or just play once
19:22.15nullable_typeI want to avoid loop
19:24.07wcselbyhere's a new one
19:24.17[TK]D-Fendernullable_type: make it one long recording.
19:24.19*** join/#asterisk lirakis (n=lirakis@65.200.191.241)
19:25.03geneticxHi you all, we pay for 3 sip trunks how can I configure asterisk so that when those 3 trunks are being used to give a busy signal instead of rejecting the call..?
19:25.29[TK]D-Fendergeneticx: congestion / busy <-
19:25.34[TK]D-Fendergeneticx: AFTER answering
19:27.07wcselbyhere's a new one.....when I ping my polycom 6000 from my asterisk server from the time it boots up, it will stay registered to asterisk (and calls will work etc) for as long as I continue to ping the phone.  Once I stop pinging the phone, the phone stops talking to the asterisk server, and then I can no longer ping it.
19:27.35*** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com)
19:28.03Naikrovekwcselby: i'm starting to think switch now
19:28.21jayteereally? I'm thinking Arby's
19:28.25Naikroveknetwork switch problem maybe
19:28.30Naikrovekmmm arbys
19:28.35Naikrovekbeef-n-cheddar ftw
19:28.41wcselbyNaikrovek - every other phone we have, polycom IP 601's, cisco 7960's and 7961's, and softphones, all work over the same network, same port, etc
19:29.03wcselbybut that doesn't mean it isn't a switch
19:29.04jayteethere's just something about their genetically modified vat grown roast beef that I love
19:29.12wcselbybrb, helping someone in vi
19:29.27dustybin[TK]D-Fender: I have fixed it now, im pretty sure there are still errors in it, I will need to test the whole thing on my setup
19:30.03Naikrovekdustybin: you're lucky to get [TK]D-Fender's opinion on that.  once he gives it his stamp of approval it'll be a really, really good, solid article
19:30.07geneticx[TK]D-Fender: Ok, so let me see if I got this correct, after my Answer ()  statement to have congestion /busy ?
19:30.50dustybinI will put a freenode credit at the end of the article, however, do you want to invite in extra people in this channel?
19:31.38nullable_typeHey guys where do i add an audio file to music on hold classes so i can use it in DIAL with m option? musiconhold.conf doesn't seem to have a [class] section
19:33.00wcselbynullable_type - you create your own [class] section
19:33.22wcselbyso if you want a class named "companyAhold" you would create [companyAhold] and then list the specifics for that class
19:33.47wcselbybut if you just want to add to your default directory
19:34.00wcselbylook for the "directory" directive in your musiconhold.conf file
19:34.11wcselbyon mine, it's /var/lib/asterisk/moh
19:34.28nullable_typeoh cool thank you
19:35.30bmoracammm...kentucky fried rat...
19:35.55geneticx[TK]D-Fender: how would I type it? because congestion /busy sounds a bit too generic
19:37.09bmoracageneticx: http://www.lmgtfy.com/?q=asterisk%20congestion
19:38.45[TK]D-Fendergeneticx: Congestion()
19:39.19geneticx[TK]D-Fender: Got it, thank you.
19:40.29OrbixxMonitor() keeps exiting with non-zero, but it's been able to record before and I can't see why the channel would already be recorded.
19:40.34superbeefwhat's the trick to getting aterisk to compile/run with dahdi support... I built and installed dahdi, adn lib pri, and i have no DAHDI commands in the asterisk cli
19:41.07[TK]D-Fendersuperbeef: Did you recompile * from scratch afterwards?
19:41.21superbeef[TK]D-Fender: i feel like i have 4 times.. going to try once again
19:41.25*** join/#asterisk maour (n=gnu@unaffiliated/maour)
19:41.45[TK]D-Fendersuperbeef: Trash your source folder, and re-extract and in menuselect pay close attention
19:42.00superbeefthere's a menuselect?
19:42.11superbeefi just do ./configre --with-curl then make; make install
19:42.56*** join/#asterisk citywok (n=chatzill@vpn.csgopenline.com)
19:42.59[TK]D-Fendersuperbeef: s/make/make menuselect/
19:43.41citywokis there a way to do something like RemoveQueueMember(*) or All Queues
19:44.20citywokPause & Unpause queue member affects all queues, but i'd like a removequeue members for all queues type function
19:44.51bmoracahow many queues do you have?
19:44.54superbeef[TK]D-Fender: it saw dahdi dependcies from menuselect.... anyway... recompiling
19:45.10bmoracayou could simply add a priority for each queue
19:45.12superbeef<PROTECTED>
19:45.12superbeef<PROTECTED>
19:45.16superbeefif that doesnt work i'm giving up
19:45.32citywokbmoraca: currently in testing 2. planning for ~100
19:45.50bmoracawhy could you possibly need that many queues?
19:46.11[TK]D-Fenderbmoraca: Call center company
19:46.16citywokcall center running multiple projects
19:46.46citywokbut, a priority for each is a viable solution if there isn't a pre-existing function for it
19:46.50[TK]D-Fendercitywok: I see a lot of value in what you'd like to do and see RQM's lack of functionaliy on this side rather disconcerting
19:47.08[TK]D-Fendercitywok: it WOULD be something simple to have created...
19:47.15citywokthe configs are autogenerated so i could do it simply
19:47.27citywokRQM? what does that stand for?
19:47.33[TK]D-Fendercitywok: For now you'd have to make a somewhat more complex script to do this for you.
19:47.39[TK]D-Fendercitywok: RemoveQueueMember
19:47.44bmoracawell, if that's the case, i suspect that you're using pins to authenticate your users...just expand that into an agi application to look and see which queues that user belongs in and use some for-next loops to take care of it
19:47.48citywokoh, har har wow i cant believe i missed that
19:48.55citywokbmoraca: yea, queue show members, all queues member of remove. or set a DB variable and add all the queues you are a member of to that
19:49.03citywokor like your first suggestion, just loop through all of them and remove you manually
19:49.24citywok[TK]D-Fender: do you think that's something i should post as a feature request on the digium site?
19:49.32citywoki've never posted a feature request, just a few bug reports
19:49.49bmoracacitywok: yes, but the second solution (agi) is more flexible...for instance, not all members may be members of all queues
19:50.14*** join/#asterisk hfb (n=hfb@pool-96-247-114-183.lsanca.dsl-w.verizon.net)
19:50.17citywokbmoraca: absolutely, they would likely be a member of 2 or 3 queues max, so it would be inefficient to do 100 commands extra
19:50.42*** join/#asterisk MindTheGap (n=caio@200.251.73.10)
19:51.22bmoracacitywok: it would also be inefficient to create hundreds of different login combinations based on the pins.  you'll need a database backend (for external administration at the very least) and AGI would be the easiest way to interface with that for the purposes of logging in and authenticating users
19:51.31*** join/#asterisk lanning (n=lanning@212.183.134.130)
19:51.40[TK]D-Fendercitywok: Yes, this is worthy of a major mod
19:52.14[TK]D-Fendercitywok: You should be able to remove from all/any queue any device or membername
19:52.24citywokbmoraca: the login stuff is already all handled.  queues are auto-generated, i need to write a query to check for permission to log in to the queue but that's easy enough
19:52.42*** join/#asterisk andres833 (n=andres83@190.144.75.22)
19:53.26bmoracai don't envy that project :P
19:53.50*** join/#asterisk lanning (n=lanning@212.183.134.130)
20:00.07IBC_jkenneyanyone know a good way to convert gsm to mp3 i want  to convert out recordings to mp3
20:00.40wcselbywill lame input gsm?
20:00.59IBC_jkenneyI belive it will
20:01.25citywokbmoraca: it's an interesting project, but we've already been using asterisk for well over a year running all of our calls (PSTN to SIP conversion on our non sip pbx)
20:01.40citywokso running all the projects out of asterisk natively isn't a HUGE step
20:02.29bmoracagotcha
20:02.36*** join/#asterisk kc8pxy (n=gecko@75-145-57-201-utah.hfc.comcastbusiness.net)
20:03.11citywokit's definitely fun.  we're wrapping our pbx around our call center interfaces so that PM's can build their own phone / project queues.
20:03.29jgoo[TK]D-Fender: so - just an update - I've got it to a reproducible test case, from a clean install and setup
20:03.30citywokinstead of a couple day turnaround for a project they can do it themselves in 10 minutes
20:04.25jgoo[TK]D-Fender: I have three trunks configured, none of the 50 extensions register. I disable the third trunk, not delete it, just disable, restart (powering up the 50 extensions after the restart) and all register
20:04.27kc8pxyI'm setting up an asterisk server to handle our offices calls, and i'm just getting it configged...  what are the minimum config files i need for it to come up for sip channels?
20:04.44jgooas soon as I enable the third trunk, all extensions drop... how..
20:04.49kc8pxyconfig of the files, i can do,  but i forget which are the minimum?
20:05.38bmoracajgoo: that might be a question better directed to #freepbx
20:05.38*** join/#asterisk l2trace99 (n=asd@75.112.140.2)
20:06.01jgooperhaps bmoraca , maybe you are right
20:06.03KattyATTENTION.
20:06.06KattyIT IS HUG TIME.
20:06.08Kattyhugs bmoraca
20:06.12jgooKatty: please go away
20:06.15wcselbyhugs Katty
20:06.18bmoracaouch
20:06.20Kattyhugs wcselby
20:06.26IBC_jkenneyhugs katty
20:06.30jgooSo much fail
20:06.32wcselbyhugs the Clue Bat [tm]
20:06.33Kattywe'll have none of your negativism in here, Sir!
20:06.36Kattyhugs jgoo
20:06.41Kattyhugs IBC_jkenney
20:06.46jgooI think that is tantamount to rape
20:06.47IBC_jkenneyagain again
20:06.48jgooand I charge for rape
20:06.59Kattyyou go right ahead and file a complaint.
20:07.11l2trace99how much ?
20:07.11jgoodue to popular demand, second rapes are half price
20:07.17IBC_jkenneyyou can't rape the willing
20:07.19IBC_jkenneyme me me
20:07.21IBC_jkenneyagain again
20:07.23IBC_jkenney;)
20:07.40jgooI know, that is why I can never score a hatrick, I usually have to take out a restraining order against the victim
20:07.45jgoothey get all clingy
20:08.06*** join/#asterisk MindTheGap (n=caio@200.251.73.10)
20:08.06Kattynothing says i love you like a restraining order.
20:08.10Kattyor so [TK]D-Fender says.
20:08.11IBC_jkenneyI didn't know farm animals could file restraining orders
20:08.20KattyIBC_jkenney: okay, that's enough out of you.
20:08.32Kattyno more negative bad thoughts!
20:08.46l2trace99when talking about bench marking would calls be equal to channels ?
20:08.48wcselbywas that a double negative Katty?
20:08.49IBC_jkenneyi thought it was funny
20:08.50bmoracafarm animals...lol
20:09.05KattyIBC_jkenney: that's beside the point! :P
20:09.28kc8pxyKatty:  did you intend to be redundant, or do you believe ther is such a thing as a "negative good thought" ??
20:09.36IBC_jkenneyin an attempt to make friends send jgoo a pass to the petting zoo ;)
20:09.39IBC_jkenneyok i'm done
20:09.40Kattyi forgot a comma, actually.
20:09.49Kattyit was supposed to be two adjetives.
20:09.52IBC_jkenneyi have it out of my system now
20:09.54Kattyi can't spell.
20:10.05IBC_jkenneythats ok i can't read
20:10.06jgookc8pxy: Katty hasn't read my thesis on patheticisms
20:10.07IBC_jkenneydon't judge me
20:10.09IBC_jkenneyclear
20:10.23Kattyjgoo: you are correct.
20:10.24kc8pxyis there any asterisk going on here?
20:10.25[TK]D-FenderKatty: :)
20:10.31Kattykc8pxy: yes, sometimes.
20:10.35IBC_jkenneyok i need to convert gsm to mp3 a whole directory i don't want to change the file names
20:10.38IBC_jkenneyjust the format
20:10.38*** join/#asterisk uqlev (n=yuriy@91.184.221.31)
20:10.45*** join/#asterisk Naikrovek (n=jjohnson@63-252-251-77.ip.mcleodusa.net)
20:10.47IBC_jkenneycan I do it all in one klump
20:10.48IBC_jkenney?
20:11.02Kattysox might do it
20:11.03wcselbyif you write a batch file to script the process, I don't see why not
20:11.06citywok[TK]D-Fender: filed as a feature request 15947
20:11.21[TK]D-Fenderkc8pxy: I don't not never not use double negatives neither!
20:11.39Kattyoh god.
20:11.42Kattythat made my brain hurt.
20:12.57kc8pxy[TK]D-Fender:  she didn't use a double negative.  double negatives contradict. redndant is an excessive reaffirmation. :)
20:13.13jgoo[TK]D-Fender: you are bordering on being a misogynist now (if Katty is female) I say if, but the way you are all pandering to the utter fail in here is reprehensible
20:14.03[TK]D-Fenderkc8pxy: yes, well there was this linguistics class in which the professor explained that unlike most languages, in Russian a double negative remains a negative, yet in no language is a double-positive a negative.
20:14.06kc8pxyjgoo: " pandering to the utter fail" ??
20:14.21[TK]D-Fenderkc8pxy: upon hearing this an attendee exclaims "Yeah right...."
20:14.25jgooshit, we are using gender specific pronouns. yes, I am so glad you can read and quote kc8pxy
20:14.35*** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com)
20:14.44jgooIt is embarrassing to see someone behave like that
20:15.05murdock_utis there a way through dialplan to tell asterisk to hangup a zap/dahdi channel?
20:15.13wcselbyjgoo - behaving like what?
20:15.14[TK]D-Fenderjgoo: Whats wrong with accurate gender-specific pronouns?
20:15.24wcselbymurdock_ut - Hangup() ?
20:15.27Kattymurdock_ut: Hangup() usually
20:15.41*** join/#asterisk Naikrovek (n=jjohnson@63-252-251-77.ip.mcleodusa.net)
20:15.47*** join/#asterisk jdnWEST (n=jdn@mx1.westparkcom.net)
20:15.48Kattywhy are we having this conversation? it seems overly dramatic.
20:15.58Kattylet's not have it anymore.
20:16.09murdock_utI have a zap channel that is constantly getting stuck and I have to keep doing a soft hangup on it.
20:16.18murdock_utIt interfaces with a paging system.
20:16.24[TK]D-Fendermurdock_ut: Figuring you want to trash a call for priority outbound (like 911) - System(asterisk -rx "soft hangup Zap/1-1)
20:16.30[TK]D-Fendermurdock_ut: Figuring you want to trash a call for priority outbound (like 911) - System(asterisk -rx "soft hangup Zap/1-1")
20:17.13murdock_ut[TK]D-Fender: Same basic idea.
20:17.21murdock_utI'll give that a go.
20:18.19kaldemarthere's a SoftHangup application that you can use in the dialplan, no need to put it through System
20:18.33jgooKatty: when people pander to silly little outburts, you may think it is modern and tolerant, actually it is a hate crime, it is because they think you are stupid, therefore they tolerate such ludicrous outburts. If I said, I would get kicked. Fair enough. You say it, you don't you feel good, but what it means is, they think that is what you do. Sad.
20:18.39jgooJust a lesson for you all.
20:19.02bmoracalol
20:19.16murdock_utkaldemar: so there is.
20:19.18murdock_utkaldemar: I
20:19.23el_crittermurdock_ut: can you explain a little bit your problem?
20:19.36murdock_utkaldemar: I'll look at that.
20:19.44Kattyjgoo: that's nice dear. my silly little outbursts will continue.
20:19.54Kattyjgoo: but feel free to put me on ignore.
20:20.50[TK]D-Fenderkaldemar: Depends on which version
20:21.02murdock_utel_critter: I have a fxs port that connections to a port on a Rauland-Borg paging system at a school.  What is happening is that either asterisk or the paging system, I'm not sure which is not releasing the line after the user hangs up the phone.
20:21.45el_crittermurdock_ut: what asterisk/dahdi version?
20:22.05kaldemar[TK]D-Fender: it was in 1.2 already
20:22.47[TK]D-Fenderkaldemar: Ok, will look into for future reference
20:22.49IBC_jkenney<===== jgoo pissing in your cereal bowl  <with love>
20:23.00kaldemar[TK]D-Fender: in 1.0 even
20:23.16bmoracauhg, i'm never using GeoTrust for SSL certificates again
20:23.27KattyIBC_jkenney: what did we say about being so negative!
20:23.36IBC_jkenneysniff sniff
20:23.39IBC_jkenneyi'm sorry
20:23.39murdock_utkaldemar: It is an older version.  I haven't got around to upgrading it.  It is 1.2.22
20:23.45Kattyhugs IBC_jkenney
20:23.53[TK]D-Fenderkaldemar: Wow...
20:24.04[TK]D-Fenderkaldemar: Yup, goes way back... never knew
20:24.06IBC_jkenneynow that is done
20:24.20IBC_jkenneylets focus on my problems beside the mental defects
20:25.01IBC_jkenneydo we know of any good "free" Cheap queue monitoring software
20:25.01kaldemar[TK]D-Fender: even better, it was introduced in 0.4.0 :D
20:25.01IBC_jkenneyto pull stats on agents and all that jazz
20:25.14kaldemarmurdock_ut: you have the application
20:25.27murdock_utkaldemar: ??
20:25.28kc8pxyi have a fresh install of 1.6.2.0. what are the minimum set of config files i need to get sip channels working? i THINK i only need sip.conf and extensions.conf, and IIRC i needed modules.conf..   is that it, or do i need more?
20:26.17kaldemarmurdock_ut: 1.2.22 has app SoftHangup, so no worries there.
20:26.17[TK]D-Fenderkc8pxy: asterisk.conf ... rtp.conf, probably a bunch more...
20:26.18murdock_utkaldemar: Ya.  I saw that.  I'm going to give that a try.  Hopefully it will fix the problem until I can upgrade.
20:28.26[TK]D-Fendercheckout time, BBL
20:29.05*** join/#asterisk maour (n=gnu@unaffiliated/maour)
20:29.56*** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1)
20:32.30tzafrir_laptopmurdock_ut, if the FXO releases the line, the FXS must notice it. The other way around is not guranteed
20:33.40tzafrir_laptopAsterisk's chan_dahdi supports disconnect notification through "kewl-start", that is power denial. But not through polarity reversal
20:36.35murdock_uttzafrir_laptop: Ok, so the I probably have it backwards when I said I had a fxs port.  It is an fxo port connecting to a fxs because the paging system is providing the dialtone.
20:37.17*** join/#asterisk darkdrgn2k3 (n=darkdrgn@208.124.232.58)
20:37.33darkdrgn2k3ok im stuck
20:37.34darkdrgn2k3http://pastebin.ca/1577019
20:37.43*** join/#asterisk MaliutaLap (n=biteme@203.171.192.19)
20:37.43darkdrgn2k3why am i getting "the number you have dialed is not in service"
20:38.07tzafrir_laptopmurdock_ut, does it provide any sort of disconnect notification?
20:38.17tzafrir_laptopIf not, do you use busydetect?
20:39.45murdock_utIt looks like I do not have that set in my zapata.conf
20:40.09murdock_uttzafrir_laptop: Maybe I'll try that.
20:40.35darkdrgn2k3sorry LOL
20:40.41darkdrgn2k3so any idea what i screwed up this time?
20:43.02darkdrgn2k3I see the call come through but then i get the "number not in service" message
20:43.21Kattyblacklist it?
20:44.46darkdrgn2k3wierd.. if i put the inbounce route to an IVR it works.. but not to an extension
20:45.41Kattyany extension? or just a particular one.
20:46.13darkdrgn2k3ok im stuck
20:46.38darkdrgn2k3oops
20:46.40darkdrgn2k3aany one
20:48.58*** join/#asterisk maour_ (n=gnu@unaffiliated/maour)
20:50.14*** join/#asterisk el_critter (n=critter@200.8.97.41)
20:51.45citywokAnother issue with queues i'm running in to is if there is one member in a queue, and they are paused, a caller does not fall out fo the queue with leavewhenempty=yes
20:54.05*** join/#asterisk Cresl1n (n=matt@asterisk/libpri-and-libss7-expert/Cresl1n)
20:54.05*** mode/#asterisk [+o Cresl1n] by ChanServ
20:56.39*** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir)
20:59.24*** join/#asterisk Jymm (i=jim@unaffiliated/jymmm)
21:00.31*** join/#asterisk tamiel (n=tamiel@ip-7.net-81-220-254.rev.numericable.fr)
21:00.35JymmJust curious... Is it possible to setup asterisk to be a fax server without a modem/ata device? Just using SIP credentials
21:00.57Kattyif you figure out how, let me know.
21:01.06darkdrgn2k3me too
21:01.21JymmWell, I can *always* whistle the tones if that helps
21:01.26darkdrgn2k3lol
21:01.27Jymm*almost*
21:01.31darkdrgn2k3apperntly you can also dial 666 :)
21:01.49Jymmwhy would I want to call myself?
21:01.57darkdrgn2k3666->system fax
21:02.05darkdrgn2k3at leastr on some guis..
21:02.07Kattynot on my server!
21:02.16Katty666 goes to the monkies.
21:02.20darkdrgn2k3-> on some guis
21:02.22Kattyor possibly weasels.
21:02.25Jymmdarkdrgn2k3: I guess you were serious, but I have nfc
21:02.40*** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com)
21:03.00darkdrgn2k3on freepbx feature 666 is "Dial System FAX "
21:03.05darkdrgn2k3and you get the high pitch whine of a fax
21:03.10*** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com)
21:03.36Jymmah
21:03.39wcselbyJymm - using iaxmodem and hylafax, it might work
21:03.39Kattydoes fax detect off an IVR work?
21:03.50Kattyyou might be able to point a 'sip trunk' to an IVR which has fax detect.
21:03.53Yuda-israel1984hi guys im a newbie here i have a question for those that understand , i am reading the book the future of telephony 2nd edition and im starting to learn scripting and i added a sip provider and i get a weird message when i call the number it says chan_sip.c:14035 handle_request_invite: Call from '7183057535' to extension 's' rejected because extension not found. and in my script of the context i have it as 7187057535,1,answer()
21:03.55Kattyfor incoming.
21:03.58wcselbythere's also something called spandsp but I don't know what that is, it's just on the fax page of the wiki
21:04.12Kattyi've been looking forever for a pysical fax machine which will take a 'sip trunk'
21:04.43wcselbyYuda-israel1984  - you have different numbers listed "7183057535 != 7187057535"
21:04.44*** join/#asterisk thansen (n=thansen@12.152.165.169)
21:05.15Jymmwcselby: you think it'll be fine for fax only?
21:05.26Yuda-israel1984wcselby i meant it says 7183057535 its the same number
21:05.33wcselbyJymm - i haven't tested, but apparently some people have.
21:05.51Kattyhas never had much luck with asterisk and faxing.
21:06.03wcselbyJymm - http://www.voip-info.org/wiki/view/Asterisk+fax
21:06.13wcselbyi've got faxing working, but it's using a t1 modem and pri's coming
21:06.17wcselbyso it's not really over SIP
21:06.26Jymmwcselby: ah, cool. Wonder if I any of the OpenWRT devices would work with it.
21:07.04wcselbyYuda-israel1984 - check the context on the sip.conf definiton
21:07.04KattyYuda-israel1984: http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
21:07.40wcselbyhis sip.conf context= is different from whereever he put the 7183057535,1,answer() statement
21:07.47Yuda-israel1984i did but i have a feeling that the sip provider isnt passing the correct info
21:08.06wcselbyYuda-israel1984 - paste your sip.conf file and your extensions.conf file to pastebin
21:08.11wcselbyso we can look at the whole thing
21:08.17Yuda-israel1984ok one sec
21:08.27wcselbythat error message for me as always meant the sip.conf context definition is wrong
21:08.31*** part/#asterisk nullable_type (n=nullable@hq.verbx.net)
21:08.42Yuda-israel1984[incoming_calls]
21:08.42Yuda-israel1984exten=>7183057535,1,answer()
21:08.42Yuda-israel1984exten=>7183057535,n,Dial(SIP/MPC/17187150001)
21:08.59wcselbyYuda-israel1984 - please use pastebin
21:09.00wcselby~pb
21:09.01infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org ,  http://bin.cakephp.org/ , or apt-get install pastebinit
21:09.08Yuda-israel1984whats that
21:11.47wcselbyYuda-israel1984 - in your sip.conf file, what's the context= statement for your sip trunk?
21:12.11JymmKatty: slow to load... http://www.faxsuperstore.com/ricoh-4420nf.html
21:12.13wcselbyactualluy, just got to pastebin.com, paste in your sip.conf file and your extensions.conf file, and then click "send"
21:12.22Yuda-israel1984http://pastebin.com/m2fedccc3
21:12.27wcselbythere we go
21:12.46Yuda-israel1984as i said im a newbie im learning and i thank you all for helping me learn
21:13.33wcselbypastebin the cli output from a failed call
21:13.40tzafrir_laptopYuda-israel1984, are you sure a call comes into that extension?
21:13.53tzafrir_laptoptry adding the following extensions:
21:13.54Yuda-israel1984thats what im not sure about
21:14.13Yuda-israel1984when i look at the sip file i get something weird i will add it in pastebin
21:14.35tzafrir_laptopexten => _X.,1,NoOp(In Fallback for ${EXTEN})
21:15.11tzafrir_laptopexten => s,1,NoOp(In default extension s)
21:15.30tzafrir_laptopnow increase verbosity: core set verbose 3
21:15.46tzafrir_laptopreload dialplan: dialplan reload
21:15.56wcselbyalrighty, I'm out
21:15.57tzafrir_laptopand see what happens in an incoming call
21:17.21Yuda-israel1984herre i am updating it
21:17.43*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
21:18.19Yuda-israel1984http://pastebin.com/m7f40c4a9
21:18.42Yuda-israel1984ok i will do that tzafrir
21:20.27Yuda-israel1984== Auto fallthrough, channel 'SIP/7183057535-0825eee0' status is 'UNKNOWN'
21:20.35Yuda-israel1984thats the added part
21:21.40*** join/#asterisk jlnt (n=jlnt@adsl-99-57-151-117.dsl.rcsntx.sbcglobal.net)
21:26.39Yuda-israel1984Executing [s@incoming_calls:1] NoOp("SIP/7183057535-0825eee0", "In default extension s") in new stack
21:26.43Yuda-israel1984sorry i missed that
21:28.19*** join/#asterisk voipmonk (n=voipmonk@69.172.93.45)
21:28.59darkdrgn2k3?? asterisk16
21:31.55*** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com)
21:33.39*** join/#asterisk ebroad (n=EB@72.11.213.195)
21:34.18ebroadwhen used for a peer, is * supposed to use auth=user#md5secret@domain for register?
21:36.58*** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com)
21:37.35Yuda-israel1984tzafrir it seems that the extension goes straight to S
21:37.40Yuda-israel1984not to its DID
21:37.44Yuda-israel1984why would that be?
21:43.16*** join/#asterisk DarkRift (n=dark@modemcable015.68-200-24.mc.videotron.ca)
21:51.16*** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com)
21:54.18*** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194)
21:59.45*** join/#asterisk denon (i=denon@sassinak.net)
21:59.54*** mode/#asterisk [+o denon] by ChanServ
22:00.20*** join/#asterisk josefig (n=JoseFig@189.129.147.130)
22:00.28*** join/#asterisk savageone (n=savageon@static-66-212-194-134.cpe.metrocast.net)
22:00.35savageonedoes anyone know how to monitor asterisk with snmp?
22:01.01Qwellres_snmp?
22:01.51savageoneor net snmp
22:01.56savageonethat's what others are saying they use
22:02.04savageonemy monitoring software wants the oid
22:02.09savageonei don't know what that is hehe
22:02.19savageoneor rather I know what an oid is but not what to use for asterisk
22:02.34*** join/#asterisk Naikrovek (n=jjohnson@98.214.112.102)
22:02.37*** join/#asterisk jcape (n=jcape@209.120.251.81)
22:03.52josefigi have a doubt, i need to send 24V how can I do it with asterisk ?
22:04.18Qwelljosefig: what?  try explaining what you need any why..
22:04.27Qwelland why*
22:04.37josefigokay
22:05.46Naikrovekyou want to power the phone without a power adapter?
22:05.52josefigi have some extensions in my office and i need to open an electric front door with an extension for example 201 for the front door but it opens with 24V how can I do it with asterisk ?
22:06.01Naikrovekah
22:06.12el_critterHi, I have this extension: exten => _9X.,1,Dial(${SNTV}/${EXTEN:1},20). SNTV is PSTN. When I call a number that matches that pattern the call drops with this error: -- Nobody picked up in 20000 ms.
22:06.39Naikrovekjosefig: you will need a door controller; talk to a local alarm company
22:06.49_brent_josefig: http://cyberdata.net/products/voip/digitalanalog/intercom/index.html
22:07.20_brent_josefig: i had a demo unit here a few months back. it's pretty cool.
22:08.45josefigbut how can I send the 24V ? via USB but how with asterisk ?
22:08.57*** join/#asterisk knarfly (n=vlad@c-66-176-177-82.hsd1.fl.comcast.net)
22:09.04Naikrovekasterisk doesn't send the 24V, it controls a device that sends the proper voltage
22:09.29_brent_the device at the link i send has a dry relay switch, this connects the 24v, but it doesn't supply it
22:10.19josefigNaikrovek, yes but how can I do that? because i can get the 5V from USB and get higher to 24V
22:10.24knarflyhelp, I'm setting up my pbx to use bandwidth.com's SIP trunking. Unless I open the firewall wide open I can't receive calls and on some calls the callee cannot hear me. I thought SIP uses 5060-5082 and 8000-20000. I have those open but unless I open all ports I keep getting only partial performance
22:10.54*** join/#asterisk FlaPer87 (n=FlaPer87@unaffiliated/flaper87)
22:10.58ebroadjosefig, check out http://www.phidgets.com/
22:11.09_brent_josefig: read the product info on that intercom. you send DTMF to it, it trips the switch, and you have to have 24v on one side of the switch and your door on the other
22:11.11Naikrovekjosefig: you need a door controller that listens to asterisk, as _brent_ linked earlier
22:11.22knarfly8-)
22:11.25*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)
22:11.28josefigokay
22:11.33josefiglemme chk, thx
22:11.45*** part/#asterisk Cresl1n (n=matt@asterisk/libpri-and-libss7-expert/Cresl1n)
22:11.50FlaPer87hey guys, Is it possible to use raw extension with ast_writefile ?
22:12.30ebroadits a usb relay
22:12.44ebroadyou might be able to trigger it via a System() call
22:17.28JymmJust wire the output of a ATA devce direct... 100Volts should pop that door right open!
22:18.54JymmThat was a joke.... don't try it at home
22:22.10ebroadwhen used for a peer, is * supposed to use auth=user#md5secret@domain for register?
22:23.08*** join/#asterisk geneticx (n=geneticx@adsl-2-59-65.mia.bellsouth.net)
22:23.50Jymm_brent_: That's pretty cool, ugly, but pretty cool =)
22:26.16_brent_yeah, it seemed to work pretty well, too. it didn't do central provisioning, so it was a no-go for us, but it was pretty cool.
22:26.37_brent_cyberdata says they're working on central provisioning
22:26.47Jymm_brent_: "cemtral provisioning" ?
22:27.12_brent_most decent VoIP phones/ATAs will grab their configs via http
22:27.26Naikrovekring voltage is 48V
22:27.31Jymmoh, dhcp
22:27.54_brent_not quite the same as dhcp, but dhcp can push the http URL that the phone should download for its configs
22:28.03JymmNaikrovek: 48V on hook, 88v off hook
22:28.10*** join/#asterisk voxter (n=voxter@76.77.95.2)
22:28.30Naikrovekoff-hook voltage while talking is 88V?
22:28.39Naikrovekpretty sure that's 12V
22:29.01Jymmno, I think it drops belw 48v. ON hook is 48. Ring is 88+volts
22:29.25tzafrir_laptopless than 12V
22:29.42tzafrir_laptopcloser to 5V
22:29.57diatonic1Hey Naikrovek - have you got a sec to take a look at what I posted in #trixbox and see if you have any ideas?
22:30.10Naikrovekring is 48V on the phone system I took apart, because I still have the ring relays
22:30.59Naikrovekthough i'm reading that they can indeed differ across the US
22:31.26Naikrovekthe old bell telephone i'm looking at has a 48V ringer too
22:31.31Jymmhttp://www.epanorama.net/circuits/telephone_ringer.html
22:32.36*** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee)
22:32.54knarflyand in case anyone else needs to be forewarned, one of my grandstream phones freaked out after only minimal use and those bums refused to replace it.
22:33.09knarflyno more gs phones for me
22:33.21Jymmeither way, should energize the solenoid at least once =)
22:33.27_brent_knarfly: not surprising. gs isn't exactly known for making a high quality phone
22:34.00knarflynor taking care of you once you've dropped $100 on one of their phones which stops working
22:34.14[TK]D-Fender~gs
22:34.14infobotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
22:34.20[TK]D-Fender~grandstream
22:34.21infoboti heard grandstream is the Yugo of VoIP hardware.  Run.  Run away now..  Though therealcircut says that they're not that bad
22:34.28[TK]D-Fenderknarfly: SAY IT AIN'T SO!
22:34.31_brent_awesome :-)
22:34.42_brent_if the infobot says it's true...
22:36.25knarflyyep, they have seen the last of my wallet
22:36.25Naikrovekpolycom :)
22:37.34Qwellknarfly: who pays $100 for a gs?
22:37.48Naikrovekno joke
22:37.54Naikrovekwell they do have higher models
22:38.17[TK]D-FenderDeluxeCrap (tm)
22:39.46*** join/#asterisk Carlos_Tico (n=grillo_v@c-98-201-162-34.hsd1.tx.comcast.net)
22:42.44*** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq)
22:47.11nextimemobile voip push system based on asterisk server side and pjsip client side is working!
22:54.33*** join/#asterisk LemensTS (n=customgt@adsl-70-238-189-232.dsl.stlsmo.sbcglobal.net)
22:54.36LemensTShey all
22:55.03Naikrovekyo
22:55.27citywoklol, we have a few grandstreams and i hate them, garbage
22:55.43citywokthe aastra's i've played with have a Phenomenal speakerphone, and great XML interface
22:56.27LemensTSi launch a php script from the dial plan, and when it is done it hangs up and doesnt do the rest of the dial plan after it http://pastebin.com/m7b5eac93
22:56.41LemensTSi thought it would go back to the next step in the dialplan, i tried agi and deadagi
22:56.58Naikrovekpolycom polycom polycom.  they should pay me for the amount of times i say that
22:57.15citywoki played with a cheaper polycom and wasn't all that impressed. though, it was a cheaper one to be fair.
22:57.37QwellNaikrovek: sorry, only one of us are allowed to get paid by Polycom when we say Polycom.
22:57.40QwellI've said too much.
22:57.42citywoki was shocked that it had a horrible speakerphone. i figured any polycom would be able to do that
22:58.08LemensTSthat was with agi debug set on
22:59.21diatonic1I love my Polycoms
22:59.22[TK]D-FenderNaikrovek: Take a number :p
22:59.26*** join/#asterisk darkdrgn2k3 (n=darkdrgn@bas2-toronto44-1176437585.dsl.bell.ca)
22:59.38Naikrovekyeah
22:59.55darkdrgn2k3ok im stumped
23:00.00darkdrgn2k3running asterisk 1.6
23:00.01darkdrgn2k3http://pastebin.ca/1577184
23:00.11darkdrgn2k3calls keep getting "number you dailed is not in service"
23:00.23darkdrgn2k3and frmo time to time a call will get through.... but the next dial is not in service
23:00.37darkdrgn2k3this is only when i send the call to a extension (any extension) or voicemail
23:00.38[TK]D-Fenderdarkdrgn2k3: #freepbx <---- go back to MagicGUILand...
23:00.42darkdrgn2k3if i send the call to ivr it works
23:00.48darkdrgn2k3i been sitting there...
23:00.54darkdrgn2k3but no one seems to have a clue.....
23:00.58Naikrovekone thing grandstream phones do have over polycom is iax2 support
23:01.01[TK]D-Fenderdarkdrgn2k3: This is not 2nd Level FreePBX support
23:01.06darkdrgn2k3hopeing you cli ppl could give me an idea whats wrong
23:01.29[TK]D-FenderNaikrovek: Except Nobody cares about IAX2, and the audio has a bit of a history of instability...
23:01.43nextimeis back to asterisk 1.4 after an hard fight with 1.6 issues in lates few days
23:02.01Naikrovekmy iax2 trunks work great.
23:02.05darkdrgn2k3what does "    -- <SIP/pritrunkdomain-08feaf80>AGI Script recordingcheck completed, returning -1" mean
23:02.41knarflyok the tech from bandwidth.com just called and said to open all udp ports from 1024 thru 65535....is this really a secure way to run an asterisk server?
23:02.46Naikrovekdarkdrgn2k3: depends entirely on what php's return code of -1 means
23:03.01[TK]D-Fenderknarfly: no functional difference
23:03.04Naikrovekknarfly: you just open them from particular hosts, and only udp
23:03.12Yuda-israel1984anyone use MPC here??
23:03.15nextimeknarfly : why ALL > 1024 ports?
23:03.38Qwellnextime: because they are clueless.
23:03.39Naikroveknextime: that's what he's asking
23:03.43knarflyi thought asterisk used 8000-20000 but with that it don't work with bandwidth.com
23:04.01QwellAsterisk uses what you set in rtp.conf.
23:04.21nextimeKrisWillis : just open the ports 5060 and the ports between rtpstart and stop you can read  on rtp.conf
23:04.42knarflyrtpstart=10000
23:04.42knarflyrtpend=20000
23:05.09citywokknarfly: did bandwidth.com not give you a single IP to communicate with?
23:05.10nextimeknarfly : and you can off course also change it to a less large set of ports
23:05.21knarflythat's my rtp.conf but for some reason I'm seeing log entries in my security file that says the DID provider is coming in on 64608
23:05.31Qwellknarfly: That is their source port.
23:05.37Qwell(or yours)
23:05.52Naikrovek"coming in on" would indicate the port he's meant to listen on
23:06.48knarflyit appears than Bandwidth.com uses a bunch of different servers because the traffic comes in from various IP addresses and various IP ports
23:07.06kc8pxyi'm trying to listen to a voicepack i found, and i can't get any of my players to play gsm.   any recommendations?
23:07.17[TK]D-Fenderknarfly: Just do it.  Doesn't make any real difference
23:07.21Qwellkc8pxy: get a player that supports gsm.
23:07.22nextimekc8pxy : use sox
23:07.33nextimeand convert it to anything you can read
23:08.06LemensTShey TKD-Fender i launch a php script from the dial plan, and when it is done it hangs up and doesnt do the rest of the dial plan after it http://pastebin.com/m7b5eac93
23:08.08darkdrgn2k3hmm no matter what i try to modify the script it keeps saying returned -1
23:08.19knarflyyes but I still don't like the ideal of opening up the whole range of ports.
23:09.30nextimemust find the time to open a couple of tickets on mantis
23:10.47knarflyI also don't like the way that once I start chatting on this board, there are a couple of attempts logged at unauthorized users trying to get at my asterisk server
23:11.09citywokwhy are you connecting to this board from your asterisk server?
23:11.17nextimerotfl
23:11.33knarflyhey 189.144.12.47  no habla espanol
23:11.41nextimechatting on irc from a server ( expecially a production one ) isn't a good idea
23:11.54citywokand some services generally check to make sure that you aren't running an open web proxy, and a few other things to make sure you aren't a compromised host
23:12.08knarflyno I'm simply watching the log file on my router
23:12.32citywokyea, that's not all that shocking
23:13.44nextimeknarfly : you can even ask an irc oper to give you an cloack host
23:13.57Naikrovekdon't you have to donate to get that on freenode?
23:14.03nextimeso, no one except irc operators and admins  can see your ip
23:14.06Qwellnot an anon host, no
23:14.23nextimeNaikrovek : no, if you donate you get a different one (supporter)
23:14.39nextimeif you just ask to have one you have one like mine, "unaffiliated"
23:14.39QwellYou all should donate to FreeNode though, of course.
23:14.42Qwell:p
23:14.47nextimei agree :)
23:15.14Jymmmade his check out to lilo
23:15.26knarflyto be honest I'd rather not hang around with a bunch of looser who don't have anything better to do that to try and crack into someone's server they worked very hard to get going.
23:15.40Jymmlol
23:16.09QwellJymm: I sincerely hope you aren't trolling.
23:16.41carrarwait you have a pbx at 66.176.177.82?
23:16.44Yuda-israel1984sorry to write this again but i have a question if on a sip message where it says To: if nothing is written then it wont go to any extension now would it
23:16.47carrarI should try
23:17.05carrarfires up nmap
23:17.44carrardoh
23:17.45carrarhe left
23:18.33[TK]D-FenderYuda-israel1984: It could.  If you made one it could match.
23:19.20Yuda-israel1984TK can u please explain
23:19.20kc8pxynextime:  is it simple to tell it to convert everything in a folder, and rename it file.ogg where it was file.gsm before?
23:20.01[TK]D-FenderYuda-israel1984: Go make an exten it can match
23:21.27Yuda-israel1984but only the provider passes on the TO: message
23:21.28Yuda-israel1984no?
23:21.35*** join/#asterisk ESCulapio__ (n=ESCulapi@barcelord.com)
23:22.00[TK]D-FenderYuda-israel1984: Go look at an actual call attempt and you tell ME
23:22.08nextimekc8pxy : find /path/to/your/folder -name '*.gsm' -type f -exec sox {}  {}.ogg ; rename "s/\.gsm//" {}.ogg \;
23:22.18nextimesomething like that should work
23:22.18Yuda-israel1984i did i get back an error
23:22.30Yuda-israel1984of it not having the S extension
23:22.52Yuda-israel1984meaning i am  not reffering to S i want to know if i can make it pass something else
23:24.12*** join/#asterisk Iamnacho (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net)
23:25.00[TK]D-FenderYuda-israel1984: It its complaining about not finding "s" thats because you didn't tell them what exten to dial... so Asterisk told them for you
23:25.26[TK]D-FenderYuda-israel1984: Hint.... its in your REGISTER
23:26.31kc8pxynextime:  looks like it worked. missing arg to -exec
23:26.35Yuda-israel1984u mean i MUST add in the register line to add the /DID ????
23:26.50kc8pxyother than that, it tried :)
23:26.54Yuda-israel1984but i want it that it should pass me the DID being called not that i have to add each one
23:28.10*** join/#asterisk jcape (n=jcape@adsl-99-37-112-194.dsl.chcgil.sbcglobal.net)
23:28.19nextimeYuda-israel1984 : something like _X. extension and get the ${EXTEN} variable?
23:28.59[TK]D-FenderYuda-israel1984: They probably are passing you the DID... in a HEADER you have to strip off
23:29.16[TK]D-FenderYuda-israel1984: "core show function SIP_HEADER"
23:30.33Yuda-israel1984reading that now in the book
23:31.43[TK]D-Fenderjam time, BBL
23:32.54Yuda-israel1984thanks have a good night yall
23:38.48jgoook, si por
23:38.52jgoofucking keyboard
23:38.55jgoook sip ports :-)
23:39.12jgooI have 8 PAP2Ts, and they were working fnie
23:39.48jgoonow I tried introducing a third SPA400 into the mix, and it seemed to initiate a series of unfortunate events - fixing one problem exposed another - now it seems that it was luck making this work
23:40.04LemensTSdo i need asterisk-addons to use mysql in the dialplan?
23:40.06jgooand it really wants all this on separate ports - what is the requirement here?
23:40.20jgoodoes anyone use this stuff?
23:40.49QwellLemensTS: yes
23:41.17jgooDo I need to have each port on the linksys on a different port, as it is, 5060 / 5061 - and in the extension definition, until now, the port was always 5060, and IT WORKED - now it seems it works if I set it to 5060 / 5061, depending on which port I assign the extension
23:41.18carrarI thought this is a drinking channel
23:41.29jgooI like a fine port
23:43.35jgooback to the issue - can people just say what the hell they do use? nobody seems to use anything, or even use asterisk in this channel half the time - what is it that the cool kids are using on asterisk?
23:43.46*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
23:44.03bmoracajgoo: tcp/ip networking dictates that only one application can be active on one port at a time.  so, yes, each "line" needs to register on a different port.
23:44.09*** join/#asterisk dlynes (n=daniel@bas5-hamilton14-1242444747.dsl.bell.ca)
23:44.29jgoobmoraca: please be more clear - every extension needs a seperate port?
23:44.39jgoo"line" isn't clear, and can mean almst anything - -
23:44.51jgooso, if I have 50 extensions, i need 50 ports?
23:46.06jgooI would love to see the documentation that states this, because I've never seen any documentation talk about more than one extension ever - and how come I've had 5-6 simultaneous calls going just during my own testing using 8 PAP2Ts? is that not runnnig more than one app on a port (also, that doesn't mean it will bind to that port on the server)
23:46.25jgoojust like firefox doesn't bind its ports to 80 no the client - it is just open to 80 on the server
23:47.46jgoobmoraca: tl;dr - what is the way to setup 16 extensions across 8 PAP2Ts
23:47.55*** join/#asterisk tzafrir_laptop (n=tzafrir@212.179.75.202)
23:48.08*** join/#asterisk voipmonk (n=voipmonk@69.172.93.45)
23:49.28jgooanyone? ports? anyone using more than one extension on a pbx? why is this seemingly important, yet no, until it stops magically working, and isn't really documented in the docs I've read?
23:49.56carrarWhy would anyone need more then 1 extension on a pbx!!
23:50.00jgoobmoraca: have you ever setup more than one extension?
23:50.06jgoocarrar: my thoughts exactly
23:50.20jgooa pbx is just a fancy way to plug a phone into a wall
23:50.39jgooone day they will make pbxes that will fit in your phone, and bye bye computer! imagien.
23:50.42carrarYou can convert a soundcard to be a PBX
23:50.42jgoodamn keybaord
23:50.43ehsjoarjgoo: Sorry to jump in so late here. All client will always initiate all SIP on 5060, unless you tell them otherwise. so 1 port
23:51.19jgooso I can run all my PAP2Ts on the same port?
23:51.34LemensTShmmm i did this http://pastebin.com/m48dab0d4 to install asterisk addons, and i rebooted server, do a help mysql at cli and get cmd unknown
23:51.42jgooout of nowhere I am having problems with this config - without changing anything - just restoring from backups...
23:51.43el_critterHi, when Dial sets a call and the other side answers, does it sets a state variable somewhere?
23:51.50*** join/#asterisk wr| (n=wr@p54BE7949.dip.t-dialin.net)
23:51.55ehsjoarjgoo: Yes, if they do SIP they should all use 5060
23:52.28hardwireel_critter: what are you doing that could query the state?
23:52.48jgooehsjoar: why are they preconfigured to use 5060 and 5061 for line 1 / line 2 ?
23:53.20jgooehsjoar: and if I use line 2, the extension I use, should that port match the line 2 port? until now I didn't and it worked. But now I changed it to match *AND IT STILL WORKS* what the hell?
23:53.26LemensTSnevermind i guess thats how it is supposed to be in 1.6
23:53.28ehsjoarjgoo: Do not know. I have installed * for 50 users. All with SIP phones configured to use port 5060
23:53.33jgoomaths would be a lot easier if we just had to approximate our numbers
23:53.34geneticxcan someone shed some light, I'm trying to set up a phone to dial automatically after 10 digits have been entered, is that something you program in asterisk or the phone itself?
23:54.09jgooehsjoar: me too... but it looks like it may be the problem... or at least n3glv said so
23:54.15el_critterhardwire: When one of my SIP extensions call a PSTN line (Digium TDM400P), it seems like sometimes the call is terminated (sometimes by the SIP extension, usually by the destinantion number) and asterisk or dahdi keeps the line open forever. I think maybe is a bug.
23:54.32ehsjoarjgoo: What configuration parameter on the PAP2T are you talking about
23:54.35carrargeneticx, digitmap on the phone
23:54.38*** join/#asterisk MaliutaLap (n=biteme@bne.lentz.com.au)
23:54.46jgooehsjoar: I have 50 extensions, 16 from PAP2Ts that will be strewn around - just 3 of these extensions have issues - - - - - since I restored a valid backup... ffs
23:55.13hardwireel_critter: doh.
23:55.15ehsjoarjgoo: when you say issues, don't they register correctly with * or what issues are you seeing
23:55.27jgooehsjoar: SIP Port
23:55.31geneticxcarrar: could it be under another name with a linksys SPA941 because I don't see anything
23:55.41ehsjoarjgoo: That can safely always be 5060
23:55.47ehsjoarjgoo: Hold on
23:55.50el_critterhardwire: no idea?
23:56.19jgooehsjoar: but on the PAP2T with two ports, the second is configed to 5061 - no docs say to configure the other side like this, and it seems to work either way
23:56.43geneticxcarrar: Ok i found it, thanks
23:56.43ehsjoarjgoo: Unless there is a problem with PAP2T having 2 analogue lines and for whatever reason can't use one SIP port. There is no problem for Asterisk for sure
23:57.17hardwireel_critter: Dial jumps to other extensions based on the status of a call.
23:57.22jgooehsjoar: ... is having two analogue lines a problem?
23:57.30ehsjoarjgoo: No
23:57.51ehsjoarjgoo: If you didn't specifically tell asterisk to listen to 5061 it will only listen on 5060
23:57.52hardwireel_critter: sorry.. sounds like you hit a bug
23:57.57jgooI mean, for the PAP2T - since it does put them on different ports =/
23:58.06ehsjoarjgoo: That is in sip.conf
23:58.18el_critterhardwire: sometimes I put a timeout on Dial, the other side answers the call and then Dial hangs because of timeout. I think maybe Dial is not setting correctly some state
23:58.22ehsjoarjgoo: Perhaps PAP2T is just using that as a backup port or something
23:58.35ehsjoarjgoo: if you change it to just 5061 it will probably not work
23:58.47hardwireel_critter: turn on debug
23:58.57hardwirelook for "remote side answered call" like events in your debug log
23:59.03hardwirelike.. turn debug on in logger.conf
23:59.13el_critterok
23:59.18el_critterbrb then
23:59.19hardwireif you don't see any.. then dahdi is not understanding the call was answered
23:59.23ehsjoarjgoo: I am assuming that this parameter is for what port the sip server should be contacted on and not what port PAP2T is listening on
23:59.31jgooehsjoar: 'listen' is a loaded term for sockets - who is listening when asterisk connects the client? anyway - it has been working with 8 lines on 5061, and all 16 extensions on 5060, and all lines working - how do you explain that?

Generated by irclog2html.pl Modified by Tim Riker to work with infobot.