irclog2html for #asterisk on 20051126

00:00.06mog_homeyes
00:00.08mog_home<PROTECTED>
00:00.24fugitivoDrJES: did you setup zapata.conf correctly?
00:00.58DrJESI believe so, as I have a channel => X for the new card channels, but still no go.
00:01.15Rawplayerdudes: exten => 101,2,Dial(SIP/Kevin1)
00:01.17DrJESChannel => 1,2,3,4 all work
00:01.17Rawplayeri have that
00:01.19fugitivowhat error do you get?
00:01.28DrJESSec...
00:02.07DrJESNov 25 20:01:53 ERROR[6888]: chan_zap.c:10250 setup_zap: Unable to reconfigure channel '7'
00:02.07DrJESNov 25 20:01:53 WARNING[6888]: chan_zap.c:11010 reload: Reload of chan_zap.so is unsuccessful!
00:02.28fugitivopastebin your zaptel.conf and zapata.conf
00:03.23*** join/#asterisk grimse (n=grimse@p5481CCBF.dip.t-dialin.net)
00:03.44DrJESWill that not flood the channel?
00:03.52fugitivo~pastebin
00:03.54jbothmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/
00:04.15DrJESOh, you just want to access it? Sec...
00:05.13DrJESwget http://ednet.ns.ca/~macleajb/z.txt
00:05.27dudesRawplayer - 101,2
00:05.44dudespastebin your extension
00:06.27fugitivoDrJES: you only have 4 channels there
00:06.53DrJESSorry... Sec, I had them commented out....
00:07.47DrJESReget it please :)
00:07.56DrJESAdded sample to channels.
00:10.13DrJESOf course with that config it fails on channel 5 :
00:10.17DrJESNov 25 20:08:08 ERROR[6925]: chan_zap.c:10250 setup_zap: Unable to reconfigure channel '5'
00:10.17DrJESNov 25 20:08:08 WARNING[6925]: chan_zap.c:11010 reload: Reload of chan_zap.so is unsuccessful!
00:10.36Rawplayerdudes: http://www.nomorepasting.com/paste.php?pasteID=53336
00:11.42dudesdoes 100 work
00:11.49h3xwhy dosent lilo just hack ircd to add pastebins to irc itself.
00:11.53fugitivoDrJES: your zaptel.conf
00:12.08DrJESSec....
00:12.29h3xor add a bot for it
00:13.07Qwell[]h3x: /j #h3xpaste
00:13.07DrJESwget http://ednet.ns.ca/~macleajb/zt.txt
00:13.11DrJESComments removed
00:13.23h3xbah.
00:13.24Qwell[]or
00:13.39Qwell[]haha, oops
00:13.43Qwell[]/query h3x - ctrl-v
00:14.13h3xthe only people that complain about channel flooding are assholes with single window clients
00:14.23Rawplayerdudes: you mean in sip.conf?
00:14.29Qwell[]I irc with 640x480
00:14.33Qwell[]one paste kills me
00:15.04dudesRawplayer - when you dial extension 100 does it work
00:15.19fugitivoDrJES: if your modules are fxo, you should use fxsks=5-6
00:16.17Rawplayerdudes: no
00:16.39Rawplayeri get 404 not found
00:16.40dudestry sip debug and see what happens when you dial it
00:16.49dudesthen it's not setup in sip.conf right
00:17.05Rawplayershall i paste that to?
00:17.36dudessure
00:18.12Rawplayerhttp://www.nomorepasting.com/paste.php?pasteID=53337
00:18.42Qwell[]Thats a freakishly long url...
00:18.53Qwell[]almost better to just paste crap into the channel :p
00:19.06Rawplayer:|
00:19.11Rawplayeri'am gonne shower
00:19.14Rawplayerbb in 10 minutes
00:19.15dudescontext=sip
00:19.24dudesin extensions it's in [numbers]
00:19.30Rawplayerhmm
00:19.34dudesso change that and then sip reload
00:19.36Rawplayerit should be 100?
00:21.19dudescontext=numbers
00:21.26DrJESBoth cards are the same
00:21.56fugitivoDrJES: change that in your zaptel.conf
00:22.41DrJESWhat about 7 and 8 ?
00:23.03*** join/#asterisk ginvent (n=joseph@adsl-63-199-244-155.dsl.sndg02.pacbell.net)
00:23.04DrJESztcfg shows:
00:23.14DrJESChannel 01: FXO Kewlstart (Default) (Slaves: 01)
00:23.18DrJESChannel 02: FXO Kewlstart (Default) (Slaves: 02)
00:23.22DrJESChannel 03: FXS Kewlstart (Default) (Slaves: 03)
00:23.24Qwell[]~pb
00:23.25jbotit has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/
00:23.26DrJESChannel 04: FXS Kewlstart (Default) (Slaves: 04)
00:23.26DrJESChannel 05: FXO Kewlstart (Default) (Slaves: 05)
00:23.27DrJESChannel 06: FXO Kewlstart (Default) (Slaves: 06)
00:23.27DrJESChannel 07: FXS Kewlstart (Default) (Slaves: 07)
00:23.27DrJESChannel 08: FXS Kewlstart (Default) (Slaves: 08)
00:23.31ginventNeed a bit of help. I just installed asterisk, and I can't get it running... what is the command to run directly (not as server)?
00:23.41DrJESSo aren't 5-6 FKOs?
00:23.42Qwell[]ginvent: -c will run it in console mode
00:24.06ginventWhat does this mean: ....asterisk: relocation error: /usr/lib/asterisk/modules/res_odbc.so: undefined symbol: ast_load
00:24.06ginventOuch ... error while writing audio data: : Broken pipe
00:24.13test34How can I know if my phone line uses reverse polarity for hangups? (because when somebody gets the voicemail, it doesn't always detect the hangup..)
00:24.28*** join/#asterisk L|NUX (n=linux@202.141.252.82)
00:24.46ginventHow can I fix the pipe?
00:24.53m160858excuse me, i wrote this line register => 3108023056@sip.broadvoice.com:XXXXXXXXX:3108023056@sip.broadvoice.com/201
00:24.55Dr-Linuxi have 2 FXO cards (4 port each), so in the zaptel.conf it will be "fxsks=1-8" or "fxsks=8" or what ?
00:24.58newltest34: you call your carrier and ask if your service is provisioned with reversal on idle.
00:25.04test34ginvent, try: asterisk -cvvvvvv
00:25.07Dr-Linuxi have no FXS cards tho
00:25.15m160858but, i not receive the call and this ext
00:25.27ginventTest34, I get that same response
00:25.38ginvent'Error while writing audio data... broken pipe.
00:25.41m160858which should be my problem
00:25.43newlginvent: that sounds like an mpg123/mpg321 issue.
00:25.47ginventhmmm.
00:25.51test34ok thanks newl
00:25.53ginventI had it running before.
00:26.07ginventI was using asterisk 1.0 no prob.
00:27.05newltest34: most PSTN services don't have it (at least here in .au and in the US).  It's mainly targeted at pay phones and PABX systems.
00:27.36DrJEShttp://pastebin.ca/31263 I think has the failed module load
00:27.45m160858:|
00:27.47test34newl, ok I'm in the US.. is there anything else I should look into ?
00:28.20m160858who speak spanish??
00:28.35newltest34: not that I can think of.  Though that's not to say that someone else here (or perhaps info on the wiki) may suggest something else. :)
00:28.58ginventWhat is a relocation error? relocation error: /usr/lib/asterisk/modules/res_odbc.so: undefined symbol: ast_load
00:29.01m160858please, i need help
00:29.42test34http://www.asteriskguru.com/tutorials/resolving_hangup_detection_problems_fxo_tdm_voicemail.html -> they have a couple of other suggestions there
00:29.55test34like getting rid of my cheap x100p
00:30.15m160858hey? i'm here xD
00:30.41m160858i have problems with this register => 3108023056@sip.broadvoice.com:XXXXXXXXX:3108023056@sip.broadvoice.com/201
00:31.05m160858the calls doesn't send to the ext 201
00:31.57m160858somebody understand this problem?
00:32.04m160858i have 5 accounts
00:32.04newlm160858: is extension 201 in the context assigned to that sip providers entry?  If not, it may be falling into the default context and rejecting the call because it doesn't exist.
00:32.12m160858yes
00:32.24m160858of course, exists
00:33.29newlginvent: that error would probably indicate a mismatch in the asterisk daemon version and the module version.  Rebuild and reinstall fresh.
00:34.46m160858i have 5 account, with 2 ext each 1 .. and 1 context por each account
00:35.11m160858if, i call the ext 201 .. the call send to ext 208
00:36.48*** join/#asterisk muHaarib (n=chatzill@66.237.2.170.ptr.us.xo.net)
00:37.08muHaaribcan anyone help a dumb guy with a quesiton on an incoming context?
00:37.08test34newl, what if I enable hanguponpolarityswitch=yes and my phone company doesn't use it? it will only not detect the hangup or it will cause some more problems ?
00:37.31test34muHaarib, dont ask to ask
00:37.38muHaaribThanks
00:37.58muHaaribI use gafachi iax2 connection with the outside world (PSTN)
00:38.01*** join/#asterisk asteriskmonkey (n=phil@HSE-Toronto-ppp300017.sympatico.ca)
00:38.05m160858someguy can help me?
00:38.16muHaaribI get phone calls routed out perfectly from my phone
00:38.32muHaaribgafachi has provided me with an IAX2 connection
00:38.37muHaaribI set it up in iax.con
00:38.41muHaaribsorry iax.conf
00:38.53muHaariband I set up two contexts in extensions.conf
00:39.01muHaarib[gafachi-outgoing]
00:39.05ginventI keep getting the same error... I just rebuilt fresh... but I keep getting: asterisk: relocation error: /usr/lib/asterisk/modules/res_odbc.so: undefined symbol: ast_load
00:39.05ginventroot@gottlieb3:/usr/src/asterisk-addons-1.2.0 # Ouch ... error while writing audio data: : Broken pipe
00:39.05ginventOuch ... error while writing audio data: : Broken pipe
00:39.12muHaariband [gafachi-incoming]
00:40.17muHaaribI can't for the life of me get [gafachi-incoming] to work ... now I know that its dumb user ... but if somebody could give me an example of an incoming context for an IAX2 connection
00:40.32muHaaribmaybe I could be think better.
00:40.41asteriskmonkeyanyone good with pri's and asterisk?
00:41.00asteriskmonkeymy pri plays busy signals on numbers that are not in service
00:41.17asteriskmonkeysorry pri dosnt asterisk does...
00:42.13test34ginvent, you can try to check google's results: http://www.google.com/search?q=error+while+writing+audio+data%3A+%3A+Broken+pipe+asterisk&start=0&start=0&ie=utf-8&oe=utf-8&client=firefox&rls=org.mozilla:en-US:official
00:42.24DrJESWell, recompiled zaptel. Reloaded modules, stop and started asterisk, now all channels show up... Go figure. But I'm :) happy it's now working.
00:43.31asteriskmonkeykram: drunkkilla: you around?
00:43.46ginventTest, I didn't get any results.
00:43.52Qwell[]drunkkilla...haha
00:43.59Qwell[]drumkilla: ^
00:44.14asteriskmonkeylol
00:44.25bbzWould anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. --
00:44.41asteriskmonkeydamn bbz every time i come in here your blasting that
00:44.43benjkcan we please get this bbz guy banned from this channel
00:44.59asteriskmonkeydont want no advertisments in here please
00:45.00test34./ignore bbz
00:45.01benjksomebody with ops privileges PLEASE
00:45.09asteriskmonkey./ignore bbz
00:45.18benjk./ignore bbz
00:45.24Qwell[]...
00:45.26asteriskmonkeywithout the .
00:45.28benjkhow do I do that?
00:45.31benjkah ok
00:45.44benjknice one, thx
00:46.29asteriskmonkeyanyhow Qwell, you know much about pri and hangup msgs?
00:46.45asteriskmonkeyi get the proper one set but i get a fast bust instead or number out of service
00:47.06Qwell[]asteriskmonkey: no
00:47.15asteriskmonkeydarn it
00:48.27*** part/#asterisk Rav1974 (n=r@pool-68-161-69-3.ny325.east.verizon.net)
00:51.41asteriskmonkeyi need a digium guy anyone of em here?
00:51.51*** join/#asterisk Cresl1n (n=matt@72.146.33.157)
00:52.28asteriskmonkeytwisted: you around
00:52.38asteriskmonkeydrumkilla: you around?
00:52.55benjkhey guys, a friend of mine is trying to come to the channel but he can't seem to log on to freenode.net
00:53.01benjkhe's getting [19:50] Info: Lookup error: Unknown host!!
00:53.07benjkany ideas?
00:53.09Qwell[]benjk: #irc.freenode.net
00:53.36benjkok will ask there
00:53.39benjkthx
00:54.27benjkare you sure that's an existing channel?
00:54.31kippihow can i uninstall asterisk?
00:54.38Qwell[]benjk: no
00:54.41Qwell[]#freenode maybe
00:54.46ginventI think my $150 frys special with ubuntu and asterisk has finally bitten me good. asterisk 1.0 no prob... asterisk 1.2... no workee
00:54.47benjkah ok
00:55.21mog_homeman i wish i had gotten a 150 dollar special...
00:55.24ginventOk... time to try a Domo arigato... Mr. REboot-o
00:55.35Qwell[]ginvent: GQ?
00:55.39ginventmog, they have them everyonce and a while.
00:55.44ginventyeah gq.
00:55.44Qwell[]Shittiest brand ever :p
00:55.55Qwell[]"Great Quality"...yeah, my ass
00:55.55ginventHey... they have been working for me for over 1 year.
00:55.56ginvent:D
00:55.58mog_homeyeah but i dont have a frys out here
00:56.03m160858hello? somebody can help me?
00:56.07ginventmog... ah... that is a prob.
00:56.09mog_homei just need a shitty emachine
00:56.19Qwell[]mog_home: wait for a dell deal
00:56.22mog_hometo test stuff on
00:56.23ginventFor an asterisk server they work great... except when running ubuntu like I am. :D
00:56.25mog_homeyeah
00:56.27Qwell[]mog_home: They sometimes have stuff for ~$250
00:56.27*** join/#asterisk ceph__ (n=amit@adsl-146-57-227.mia.bellsouth.net)
00:56.31Qwell[]WITH a monitor...
00:56.31ginventdeal deal for $150? that will be the day.
00:56.36mog_homei dont need monitor
00:56.36benjkthat really SUXXXX
00:56.39ginventlol
00:56.41ginventbrb
00:56.44mog_homejust need headless box for testing
00:56.46benjkyou cannot write to #freenode
00:56.50benjkonly listen in
00:56.52m160858there exists some irc on spanish ?
00:57.10benjkso how do people get help when they can't get on?
00:57.19Qwell[]benjk: They don't
00:57.21mog_homewhats he using to connect ben
00:57.23Qwell[]problem solved. :P
00:57.25mog_homeits so easy
00:57.26*** part/#asterisk m160858 (n=jsaenz@200.89.12.46)
00:57.31*** join/#asterisk m160858 (n=jsaenz@200.89.12.46)
00:57.33benjkthe same client I am using
00:57.44benjkand I told him to do exactly what I do
00:57.57m160858excuse, some guy speak spanish?
00:58.04benjkit seems that freenode tries to do some reverse lookup of some thing
00:58.08benjk[19:50] Info: Lookup error: Unknown host!!
00:58.17ceph__Hello..can anyone recommend a good online store for IP phones that they've used with *
00:58.27m160858or somebody knows a channel on spanish?
00:58.44benjkm160858: wait a few minutes
00:58.55mog_homeasterisk-es maybe?
00:59.08benjkmy friend is from Colombia
00:59.17benjkhe is tryuing to come to the channel
00:59.24benjkyou can talk to him in Spanish
00:59.53mog_homemaybe russian
00:59.53asteriskmonkeyceph_: massivetel.com they sell voip stuff uber cheap... no onlne shop though :P i buy my digium iaxys from there
01:00.55ceph__*monkey..thanks...have you ever tried voipsupply.com?
01:01.27asteriskmonkeyno from what i head there decent prices but returns on bad product are a headache
01:01.40asteriskmonkeymassivetel sells iaxys cheaper though :)
01:02.04asteriskmonkeyah 1 big difference massivetel=canada voipsupply=us
01:02.56ceph__good to know.
01:03.03test34is iax2 the best protocol to use with asterisk ?
01:03.19Qwell[]test34: there is no "best" protocol to use
01:03.53benjkIAX2 is the best protocol.period
01:04.13mog_homelol
01:04.15Qwell[]each have their benefits - even h323
01:04.19mog_homeno that is h323
01:04.24mog_homeand mgcp
01:04.26mog_homethey own you
01:04.31*** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com)
01:04.31benjkyeah the Ford Etzel also had some good
01:04.34Qwell[]sccp <3
01:04.55mog_homeblech
01:04.56mog_homeqwell you had to go and take it to far
01:05.08Rawplayerdudes: still here?
01:05.11Qwell[]What, got a problem with sccp? :(
01:05.18Qwell[]sccp is hot
01:05.19RoyK~lart himself
01:05.19benjkfreenode really sucks ballz
01:05.19test34ok I guess I will go read some more before I choose then.. so far the only advantage Ive read is that iax2 uses less ports.. which is not a big advantage for me
01:05.22mog_homewell as a fat man i could never like skinny
01:05.45mog_homeyou use the chan_sccp from sourceforge
01:05.47Qwell[]mv chan_sccp chan_fccp
01:05.52mog_homelol
01:05.57Qwell[]oh, wait
01:05.58mog_homefccp kicks ass
01:06.00Qwell[]shit, that was chan_sip
01:06.02Qwell[]:D
01:06.14asteriskmonkeyiax2 is supernice crips sound with ulaw :)
01:06.19Rawplayer[numbers]
01:06.20Rawplayerexten => 100,1,Wait,1
01:06.20Rawplayerexten => 100,2,Dial(SIP/Kevin1)
01:06.20Rawplayerexten => 100,3,Hangup
01:06.22Qwell[]mog_home: Is the sf one the same as the berlios.de one?
01:06.28mog_homeyeah i think so
01:06.29Rawplayerthe phonenumber of kevin is 100 right?
01:06.35mog_homeyes
01:06.41RoyKfccp. what is that? fucking chicks coming personally?
01:06.44mog_homewith a pointless one second wait
01:06.57mog_homeno royk
01:06.59Qwell[]No, it isn't
01:07.06Qwell[]mog_home: different one
01:07.14mog_homeoh really
01:07.21mog_homewhere does berli one come from
01:07.27Qwell[]berlios.de :P
01:07.28RoyKmog_home: too bad
01:07.32Qwell[]chan-sccp.berlios.de I think
01:07.35mog_homewho wrote it?
01:07.58Qwell[]Sergio something maintains it
01:08.03Qwell[]it was skinny, then something else
01:08.14Qwell[]maybe it was derived from the sf one
01:08.30mog_homemmhm
01:08.34mog_homeand its straight gpl?
01:08.44Qwell[]<PROTECTED>
01:08.46Qwell[]<PROTECTED>
01:08.47Qwell[]<PROTECTED>
01:08.50Qwell[]got me...think so
01:08.53mog_homewhat features does it support?
01:08.58Qwell[]like all of sccp, heh
01:09.00mog_homeyeah its based of the sf one
01:09.04mog_homereally?
01:09.06mog_homewow
01:09.07Qwell[]I'd say it's pretty complete.  hints and stuff
01:09.11Qwell[]realtime now :D
01:09.14mog_homedidnt know it was that far along
01:09.18Qwell[]or, soon, if he accepts my patch
01:09.29mog_homei know the chan_skinny doesnt do everything
01:09.44Qwell[]he's done a great job with it, imo
01:09.52mog_homeyeah def.
01:09.56Qwell[]the code is a bit ugly, doesn't follow the * guidelines, but...
01:10.26Qwell[]the config is a bit ugly (it's like the zap configs, where the part that defines what it is is at the end)
01:10.28*** join/#asterisk ginvent (n=joseph@adsl-63-199-244-155.dsl.sndg02.pacbell.net)
01:10.34benjknot that the code which does follow the guidelines would be any prettier
01:10.48test34lots of voip providers offer you pyramid schemes as opportunities ?
01:10.50ginventI am still getting this error  any help?   asterisk: relocation error: /usr/lib/asterisk/modules/res_odbc.so: undefined symbol: ast_load
01:10.56ginventI rebuilt
01:11.00ginventI recompiled clean.
01:11.06ginventnot sure what it could be.
01:11.07Qwell[]in fact, today he's celebrating 6 months of chan_sccp2
01:11.16ginventNo response on google.
01:11.32ginventI don't want to go back to 1.0...
01:11.57Qwell[]ginvent: Did you remove the modules before you upgraded?
01:12.00mog_homechan skinny can act like both sides or just gateway like mgcp?
01:12.07ginventQwell?
01:12.10Qwell[]mog_home: I'm not sure, actually
01:12.15ginventhow would I do that?
01:12.22mog_homecan you talk to cisco gateways
01:12.25mog_homeor just phones
01:12.29Qwell[]ginvent: rm /usr/lib/asterisk/modules/*
01:12.38Qwell[]ginvent: then make, make install
01:12.41Qwell[]mog_home: not sure...
01:12.43mog_homemake clean
01:12.44mog_homemake install
01:12.47Qwell[]right
01:12.51Qwell[]what he said
01:13.02nitramQwell[]: hinting already works in chan_sccp, i have that setup with a 7960 and a 7920 here
01:13.16Qwell[]nitram: yes, thats what I said
01:13.20Qwell[]it supports hints and such
01:13.24nitrammog_home: it's just for phones afaik
01:13.46Qwell[]and hints with 7960+7914 = <3
01:13.48mog_homethats what i thought
01:13.53ginventOK, trying it now... I rm /usr/lib/asterisk/modules/*   then make clean... make... and make install
01:14.06Qwell[]ginvent: make install will compile, you can skip the make
01:14.11ginventoops... hehe
01:14.13ginventtoo late.
01:14.13ginvent:D
01:14.17Qwell[]doesn't matter
01:14.20Qwell[]make install won't recompile
01:14.29ginventk
01:14.31Qwell[]just one less step
01:14.43ginventWell, it can't hurt to try to re-build... again.
01:14.56ginventalthough on this QC pc... it's a tad slow... hehe
01:15.01Qwell[]qc?
01:15.10Qwell[]gq?
01:15.10ginventGQ I mean.
01:15.13ginventlol
01:15.13Qwell[]right
01:15.20ginventI am all doped up on turkey.
01:15.29Qwell[]Is it even a celeron?
01:15.36Qwell[]probably uses a transmeta chip or something, heh
01:15.39ginventSempron... nothing but the best. :D
01:15.46Qwell[]oh, wow
01:15.48ginventwith SIS
01:15.52mog_homesis rules!
01:15.57ginventit's a sempron 2200
01:15.59Qwell[]sempron > duron > celeron
01:16.05mog_homeamen brother
01:16.13ginventI just turned up the overclock a touch. hang on to your shorts!
01:16.24Qwell[]overclocked gq sempron...oh god
01:16.28ginventlol
01:16.38Qwell[]I can just see that opening a portal to hell
01:16.44Qwell[]satan spawn stepping out...
01:16.54ginventI am pretty sure the power supply can not handle it.
01:16.57nitramtoo bad, chan_bluetooth didn't get as far as chan_sccp ;)
01:17.01ginventI have heard horror stories about it.
01:17.10Qwell[]heh, probably what, 250 watts?
01:17.21*** join/#asterisk jaristizabal (n=jaristiz@69.79.133.185)
01:17.24ginventI think they used the supplies from an old amiga 500
01:17.26ginvent:D
01:17.30benjkfinally
01:17.35benjkhi javier
01:17.45jaristizabalhi
01:17.52ginventok, I am on the make install
01:18.06mog_homehehe
01:18.07ginventdone... dare I asterisk -cvvvv
01:18.09jaristizabalgreat!
01:18.27benjkjavier, there is somebody who needs some help in Spanish
01:18.36ginventHoly crap that did it!
01:18.36benjkm160858, are you still here?
01:18.42ginventyou guys are DA BOMB!!!!
01:18.44jaristizabali speak spanish
01:18.49ginventBOMB!!!!!!!!
01:18.51benjkm160858 ???
01:19.00jaristizabalwho need help in spanish??
01:19.04benjkm160858
01:19.07ginventQwell, I should send you a GQ pc just for your tech support.
01:19.15ginventThen you too can have your own gateway to hades.
01:19.20kippigreat!
01:19.20benjkbut it looks like he is gone now
01:19.31kippiI have just done a clean install of asteris
01:19.34mog_homeman think of how much you could sell that for ginvent
01:19.41mog_hometo like a group of gothes
01:19.43mog_homeor some thing
01:19.54benjkgothes?
01:19.57ginventlol mog...
01:20.00kippijust tried to start it and got this error
01:20.07mog_homeyeah i added the e
01:20.16mog_homefor emphasis
01:20.33m160858hey?
01:20.34kippihttp://pastebin.com/438312
01:20.43m160858benjk?
01:21.19mog_homem1
01:22.35jaristizabalhola
01:22.43jaristizabalsi
01:22.48jaristizabalquien te dijo?
01:23.54benjkjavier, you should chat to m160858 in a private chat
01:25.54Qwell[]ginvent: yeah, do that
01:26.48Qwell[]ginvent: There is always paypal too.  My own gateway to hell
01:27.17mog_homehes gone qwell
01:27.24Qwell[]yeah, my client sucks
01:27.32Qwell[]it's still tab completing it :p
01:27.40mog_homeindeed
01:27.48Qwell[]hell, I couldn't /part a channel earlier
01:27.52Qwell[]tip: don't use chatzilla
01:28.12Qwell[]I tried to do a driveby poking of file, but my client didn't leave the channel afterwards
01:29.34benjkQwell, it seems that most irc clients have serious issues
01:29.45Qwell[]benjk: none are this bad
01:30.08benjkmaybe if you are a complete geek
01:30.28benjkif you just want to use it and not get in the way, then they all have serious flaws
01:35.58fugitivobitchx runs whithout problems
01:36.13mog_homei have the best one
01:36.14mog_hometelnet
01:36.17niZonanyone know if verizon phones can be used on other providers, or are they locked?
01:36.25Qwell[]niZon: unlock it
01:36.33fugitivomog_home: ugly :)
01:36.44niZonI'd love to know how :\
01:36.49mog_homeits so easy no wonder its number 1
01:36.50niZonwithout spending more money
01:37.21*** join/#asterisk znoG (n=gs@OL101-122.fibertel.com.ar)
01:38.02Qwell[]niZon: google away
01:38.21mog_homewhy google when someone on ebay will do it for 10 bucks
01:38.39Qwell[]ship it to them, and they'll also clone your phone :p
01:38.47Qwell[]free calls for them :D
01:39.05mog_homelol im all for sharing qwell
01:39.16mog_homethankfully i dont have a phone
01:40.25file[laptop]Verizon is a CDMA provider anyway... what other provider would you use it on?
01:40.31trixterbenjk: you alive?
01:40.40benjkyes I think I am
01:40.53trixterthere are now 580,067 entries in my list, so adding a few wouldnt work :P
01:41.07benjklemme see
01:41.31trixterpulling from my old list to make my current list better, like the netherlands I had 189 in my old one but this one had less info and it went to like 5 or 6 entries
01:41.33benjkyou need to add 8821
01:41.53trixterthat will be tough my old list was only 5500
01:41.54benjkanother 8821 and you get a very lucky number :-)
01:42.32benjkjust add a bunch of e164.org blocks
01:42.58benjkthey give every member a block of 100 number in a non-existing country code
01:43.06benjk80 something
01:43.20benjk882 or something like that
01:43.24trixterha
01:43.29Dr-Linuxi have created an IVR in [ivr] context, i want if the caller dial any extension for  [default] context during greetings, he should redirect to desire extension
01:43.42*** join/#asterisk joat (n=joat@ip70-160-150-20.hr.hr.cox.net)
01:43.45trixterum what?
01:43.47Dr-Linuxmy all user's extensions are in [default] context
01:44.04Dr-Linuxonly [ivr] is there
01:44.08Dr-Linuxany clue?
01:44.12benjkDr-Linux, that's bad practise
01:44.34trixterin default    exten => _X.,1,goto(ivr,1)
01:44.36trixteris that what you want?
01:44.41trixterI really dont understand the question
01:44.55benjkyes but you shouldn't have your users in the default context
01:45.13Dr-Linuxhttp://pastebin.com/438292
01:45.14benjkthats like giving world provileges to your system files
01:45.25trixterthat depends
01:45.28Dr-Linuxthis is my IVR, and it works okey
01:45.42benjkI disagree
01:46.00trixterI think that you can prevent abuse from default if you want to
01:46.01Dr-Linuxbut i have more then 200 extensions,
01:46.06trixterwhether or not people do is a different story
01:46.21benjkbecause even if you start off with a setup where you say "my users have no right to do anything" you may then later add some services which cost you money and you forget to lock it down
01:46.37trixterbenjk: I got a plantronics m2500 bluetooth headset today for $10 ...  one of  those loss leaders to get you in the story
01:46.43trixterI am somewhat happy with my purchase :
01:46.43Dr-Linux:S
01:46.55benjktrix: nice
01:47.02trixter:)
01:47.12benjkdid you already take delivery or did you just order it?
01:47.25trixtercash purchase
01:47.28trixtergot the last one too
01:47.33benjkhehe
01:47.35trixterradio shacks sale today
01:47.38joatclearance?
01:47.39trixterlimit 10 per customer, yes 10
01:47.50trixterno its black friday, so almost every store has a big sale today
01:47.50benjkwow
01:47.57benjkah ok
01:48.06benjkbut its saturday
01:48.11benjk:-)
01:48.16trixter$59.99 - $30 sale == $29.99 with a $20 mail in rebate
01:48.23trixterit was friday :P
01:48.26trixterit still is here
01:48.29joathmm... wonder if the local shack will have it
01:48.36benjkyeah I figured that
01:48.37trixterI expect to see a ton of these on ebay soon
01:48.38trixter$40 or so
01:48.48trixtercause its gonna be a lot of profit for anyone that got any amount
01:48.50benjkI remember this thread a few years ago on Digium's mailing list
01:49.01benjkDigium announced some new product
01:49.07benjkdon't know what it was
01:49.07trixterds3000?
01:49.08asteriskmonkeyok that is simple to do
01:49.09trixter:P
01:49.28trixterthat and their 24 port T1/E1 card I dont think ever actually made it into production
01:49.31benjkanyway, Mark said something like "we should have these new <whatever> out this spring"
01:49.54Qwell[]24 port T1/E1?
01:49.56benjkand some guy from Australia asked "is the northern or southern hemisphere spring"
01:50.03trixterthat is their new not yet available product htey are selling
01:50.04Qwell[]surely you mean the TDM2400P?
01:50.09trixterthe ds3000 (DS3 card) was the other one
01:50.17benjkNo that was 3 years ago or so
01:50.19Qwell[]That's 24 analog ports
01:50.28Qwell[]not 24 T1/E1's :p
01:50.38trixteryeah actually that is what I meant 24 analog..  and afaik that doesnt ship for a couple more weeks
01:50.48mog_homemmm 24t1/e1 board...
01:50.56trixterthe ds3000 is 28 T1
01:50.56trixter:P
01:50.57benjkhow much is the 24 port card?
01:51.00mog_homeits already shipping trixter
01:51.13benjkmog: how much is it?
01:51.18trixterahh, heard that it wasnt yet available from someone @digium last week or something
01:51.21trixterguess they were wrong
01:51.36mog_homewell our resellers and distributors already should have some
01:51.44mog_homeand i think we are selling it on site now?
01:51.47mog_homei could be wrong
01:51.52trixternot everyone there actually is in sales, or whatever, according to cnet's interview with mark they make $10M/year so odds are they have real departments
01:52.02trixteron which site?
01:52.05mog_homevery true trixter
01:52.09mog_homeour site
01:52.14trixterwho is 'our'
01:52.14mog_homeid have to go check
01:52.15trixter?
01:52.19mog_homei try to stay out of sales
01:52.23mog_homei work at digium trixter
01:52.26trixterahh
01:52.59trixterwell then what about the ds3000?
01:53.02benjkmog, just rougly, is it in the 500 USD ballpark, 1000 USD ballpark?
01:53.06trixterthat has been promised for a while ... :P
01:53.21Rawplayerarent there any decent samples of voip?
01:53.29mog_homedepends on the config
01:53.29Rawplayerwith sip and extensions.conf
01:53.31Qwell[]mog_home: Do you know if it's possible to buy a 410, then get the echo can later?
01:53.32trixterits module based iirc 4 ports on a module, so I would push towards the higher end of things
01:53.39mog_homeyes qwell
01:53.45mog_homeolder boards need software upgrade
01:53.48mog_homewhich we do for free
01:53.51Qwell[]ahh, cool
01:53.55mog_homebut any board bought for a while now should work
01:54.05mog_homeits a little cheaper to get em bundled
01:54.05Qwell[]does it cost more to get a 410 then +1 than a 411 does?
01:54.08Qwell[]k
01:54.10mog_homebut not that much more
01:54.20mog_homei think it is 1000 and 800 if you get it with board
01:54.22mog_homebut i dont know
01:54.50trixterI say we go chase the sales guys with torches and pitchforks
01:54.51Qwell[]haha
01:54.54Qwell[]http://www.voipsupply.com/product_info.php?products_id=772
01:55.02Qwell[]They marked the price DOWN from $84 to $89
01:55.17mog_homedid it go up 5
01:55.18mog_homelol
01:55.24mog_homeyay bugs
01:55.35Qwell[]THAT is why I'll NEVER buy from voipsupply
01:55.41benjkIf Mark get the Japanese caller ID working for analog zaptel, we may actually order a few of those 24 port analog boards
01:55.50Qwell[]I saw a TDM400p with 4 FXS for like $150 once
01:55.55mog_homeooh thats hot benjk
01:55.58Qwell[]I called them on it, so they "fixed it", to like $500
01:56.00mog_homethats like a dinner for me
01:56.03Qwell[]then I called them on it again
01:56.09mog_homeyikes
01:56.13benjkBTW, mog, thanks for pulling Mark into this yesterday, that was very helpful
01:56.20mog_homei do what i can
01:56.27mog_homeand i dont like people talking smack ^_^
01:56.31benjkI have been trying to get him to do that for 2 yeas
01:56.33benjk;-)
01:56.44asteriskmonkeywooo fixed the pri odd issue :)
01:56.50Qwell[]somebody got cid working for jp...
01:56.53Qwell[]who was that?
01:56.59mog_homedont yet
01:57.02mog_homemark is working on it
01:57.07Qwell[]I think I recall somebody saying they got it working already
01:57.14mog_homeyeah benjk ^_^
01:57.20mog_homeit works with voicetronix hw
01:57.21Qwell[]no, somebody else...
01:57.26Qwell[]was like...3 weeks ago
01:57.48mog_homei missed it thats for sure
01:57.55Qwell[]maybe it was indications
01:58.37benjkmog, the voicetronix hardware is overpriced (1800 USD for a four port card here in JP) and it doesn't work so well, if it works at all
01:59.11mog_homeeep
01:59.36file[laptop]meeo
02:00.22Qwell[]download the latest firmware!?
02:00.38benjkmog, Voicetronix have signed a deal with a Japanese company to sell their boards in Japan
02:00.51benjkthe Japanese company specialises in Bayonne
02:00.54benjknot Asterisk
02:01.09benjkso all the work for Japan stuff is mostly going into Bayonne
02:01.33mog_homefun stuff
02:01.36benjkand the Japanese company sells those boards with a 300 % profit margin
02:01.45benjkthat's how Japanese companies think
02:01.51mog_homeman i need to make a hard phone
02:01.58mog_homeand charge 100% markup
02:02.03mog_homeand call it misco
02:02.04Qwell[]mog_home: 800%
02:02.08benjksell five boards a month with 300 or 500 or 1000 or 2000 percent markup
02:02.10mog_homenah im not gready
02:02.10Qwell[]call it...
02:02.12Qwell[]Poortel
02:02.13Qwell[]:D
02:02.14mog_homelol
02:02.18mog_homegenius
02:02.31Qwell[]I've got a bunch of names that would violate trademark
02:02.32benjkinstead of selling 100 200 500 cards with 20 25 or 30 percent markup
02:02.34mog_homei think i will have "random" pricing
02:02.36Qwell[]like Qwell Communications
02:03.02mog_homesome one calls in
02:03.04mog_homeits 1 grand
02:03.07mog_homesome one else
02:03.08Qwell[]rofl
02:03.10mog_home50 cents
02:03.21benjkto be fair though, those guys have gone through the trouble of taking out Japanese type approval for the Voicetronix board
02:03.22Qwell[]Don't make it that extreme...people will just call back
02:03.39mog_homeheh thats why i have to get ani information on my t1
02:03.43mog_homecatch people doing that
02:03.45Qwell[]heh
02:03.46mog_homeprice doubles
02:04.09mog_homeand i have to make my hw hard to buy like cisco or polycom
02:04.14Qwell[]ugh
02:04.22Qwell[]Make them sign an ICA
02:04.23mog_hometo own my phone you must have a phd in telephony and go to the misco training course
02:04.31mog_homeand to even talk to me
02:04.31Qwell[]and then have NO mention AT ALL on your site, of HOW to complete the ICE
02:04.33Qwell[]ICA
02:04.37mog_homeill need 3 ndas
02:04.41mog_homeyes
02:04.50mog_homeman id be a billionaire
02:04.50Qwell[]That's what cisco does, heh
02:04.51*** join/#asterisk ThePeopleGA (n=kemtram@rev-204.120.18.37.genesiswireless.us)
02:04.52mog_homeover night
02:04.57mog_homewithout even selling anything
02:05.05mog_homethen i would have to flea the country
02:05.09mog_homego to portugual
02:05.12mog_homeand go by jose
02:05.14Qwell[]because you never actually made any hardware?
02:05.19trixterwhy portugal?
02:05.28mog_homewho would think i would go to portugal
02:05.28Qwell[]no extradition treaty?
02:05.37mog_homeno one would see it comming
02:05.41Qwell[]we would
02:05.44Qwell[]because now we know
02:05.49Qwell[]foiled
02:05.54mog_homeheh thats why i am really going to khazakastan
02:05.54trixterpoprtugal is friends with the US, they are part of nato, odds are they have an extradition treaty to here at least :)
02:05.56mog_homeoops
02:06.14Qwell[]You can come to Qwellsakistan
02:06.21Qwell[]We don't have an extradition treaty
02:06.31Qwell[]no fcc either
02:06.35mog_homeheh qwellsakistan sounds like a fun place to go
02:06.38Qwell[]it is
02:06.40trixterbut would you really want to goto a place where its short name is 'Qwell sak' ?
02:06.42mog_homewhere you from qwell?
02:06.49Qwell[]Qwellsakistan...
02:07.13mog_homewhich is in the vicinity of
02:07.18alephcomHow are the taxes there?
02:07.27trixter200%
02:07.27Qwell[]alephcom: 120% income tax
02:07.32trixterha
02:07.46Qwell[]but, we have free healthcare
02:07.52alephcomLol,  I think he's confused.  He says Qwellsakistan and means Canada. :-)
02:07.54Qwell[]and we're communist
02:08.10mog_home120% glad ill be unemployed
02:08.15Qwell[]mog_home: exactly
02:08.16trixterok french canada :P
02:08.26mog_homewelfare?
02:08.37Qwell[]mog_home: 6000 a week
02:08.55alephcomHmmm,  I think I'll be emigrating.
02:08.59mog_homefreaking sweet
02:09.00Qwell[]which is also taxed
02:09.06mog_homeill have to tell mark i got a better offer
02:09.08alephcomoops.
02:09.10trixterbut 6000 qwellsak dollars is like $0.0000001 you couldnt even get 1 minute of voip time with it
02:09.17Qwell[]trixter: free voip
02:09.31trixteris that free voip as in taxed?
02:09.32alephcomonly if you're unemployed though.
02:09.39Qwell[]trixter: no, truly free
02:09.46trixterwho provides that?
02:10.01Qwell[]we contracted with vonage
02:10.08mog_homeewwww
02:10.09mog_homevonage
02:10.10alephcomrofl
02:10.15Qwell[]past tense
02:10.18Qwell[]they cut us off
02:10.19trixterheh..  should contract with www.trxtel.com :)
02:10.22Qwell[]So now we go through fwd
02:14.00asteriskmonkeycontract with massivetel damn it :)
02:16.17*** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn)
02:17.42asteriskmonkeydamn free calls to all those countries
02:17.44asteriskmonkeythat looks sick
02:17.46asteriskmonkeyis it a joke
02:18.54*** join/#asterisk diego_br (n=diegovie@200-213-122-121-mns.cpe.vivax.com.br)
02:19.42Vcoof course, a good percentage of those taxes also go towards subsidizing the development of beer that doesn't taste like piss
02:20.05Qwell[]Vco: No, we're a vodka drinking people
02:20.13Vcomeh..
02:20.30Vcounless it's got some gin and an olive involved i dont' really touch vodka
02:20.51trixterasteriskmonkey: nope, it should be up soon for all
02:21.08trixterstill testing
02:21.31asteriskmonkeyhow are you doing it, using enum?
02:21.58trixternot entirely
02:22.04trixternot everything is in enum
02:22.04asteriskmonkeyspill
02:22.11asteriskmonkeyah dundi/enum
02:22.22trixtermostly through donated services and equipment
02:22.44trixterright now there is no dundi or enum built in
02:23.03trixterits all straight to a pstn gateway i do plan on adding enum and dundi though
02:23.55asteriskmonkeywell whats the local pstn its on ?
02:24.05asteriskmonkeyand is it a pri ? ds3 oc-12?
02:24.13trixterit uses several
02:24.24asteriskmonkeyseveral what? bits of string?
02:24.32asteriskmonkeywhats it connected with...
02:24.35trixteryes that must be it
02:24.37mog_homepris i imagine
02:24.43Dr-Linuxasteriskgeeks: what will be the sequence of the [ivr] and [default] contexts ?
02:25.00asteriskmonkeydefault first ivr next
02:25.28asteriskmonkeywell trixter where are you located ill trade you some channels on my local pri
02:25.35Dr-Linuxokey let me try
02:26.18trixtertrx telecommunications, inc has multiple facilities depending on what is being terminated..  trx is a real clec in montana but has servers elsewhere for other operations
02:26.21Dr-Linuxasteriskgeeks: i understand and where will be include => default
02:26.32Dr-Linuxas sequence?
02:26.46asteriskmonkeyin the [ivr] context
02:27.12asteriskmonkeyhttp://www.voip-info.org
02:27.17Dr-Linuxokey let me try
02:27.26asteriskmonkeytrixter you own that clec?
02:28.36asteriskmonkeydarn it i run a hackjob of a clec :) would be nice to find some other partners to swap ld with
02:28.56Qwell[]asteriskmonkey: I give you free minute to Qwellsakistan
02:29.16asteriskmonkeylol non one has internet there
02:29.23Qwell[]everyone has internet
02:29.25Qwell[]free wifi
02:29.40*** join/#asterisk coppice (n=chatzill@168.155.17.210.dyn.pacific.net.hk)
02:29.48asteriskmonkeydude you not talking about your house are you?
02:29.48*** join/#asterisk clinthome (n=clinthom@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net)
02:29.54Qwell[]no
02:30.17Qwell[]front and back porch, and surrounding areas too
02:30.17asteriskmonkeylol
02:30.17Vcoheh..heh..
02:30.17Qwell[]I mean...
02:30.18mog_homeooh the back porch
02:30.18Qwell[]inside and outside castle
02:30.28mog_homei get charged like 100 dollars a minute for calls there
02:30.36asteriskmonkeydamn
02:30.40Vcoand around the  car hold
02:30.40nick125wow
02:30.50Qwell[]Vco: No cars in Qwellsakistan
02:31.01*** join/#asterisk kks (n=kks@202.73.8.130)
02:31.01asteriskmonkeyhey i got to play with some kick as wireless sip phones today
02:31.04Qwell[]cars must be outsourced...to parkinglotsakistan
02:31.19mog_homelol
02:31.24asteriskmonkeythey take sim chips so you can use fido then user voip when you pick up a wireless connection
02:32.17trixterasteriskmonkey: yes I do own it
02:32.50asteriskmonkeyso you have a multi hommed t1/pri then or acuallt mult location multi pri
02:33.03Vcotrixter p0wnz
02:34.22Dr-Linuxasteriskmonkey: http://pastebin.ca/31276
02:34.27Dr-Linuxplease check it out
02:34.44Dr-Linuxbcoz its not working, i believe i'm missing/wrong somewhere
02:34.57*** join/#asterisk camonz (n=camonz@200.8.21.123)
02:36.29asteriskmonkeyapart from your overuse of the background function it should work reload your extensions
02:37.33asteriskmonkeyoh wait try putting include => default affter #
02:37.33asteriskmonkeyexten => s,22,Background(to-reach-operator)
02:37.42asteriskmonkeythen reload
02:37.51asteriskmonkeythen foward payments to digium :P
02:37.55Dr-Linuxokey wait
02:38.37Dr-Linuxinclude => default affter # << sir i didnt understand this
02:38.59Qwell[]don't put users in default
02:39.01Qwell[]thats unsafe
02:39.43*** join/#asterisk TheCops (i=nobody@got.securebinary.com)
02:39.52Dr-LinuxQwell[]: i will change the contexts once i understand a few things
02:40.58Dr-Linuxwhere should i put this "#" sign ?
02:41.07Qwell[]nowhere
02:42.01Dr-Linuxbut this doesn't work for me >> http://pastebin.ca/31276
02:42.08Qwell[]Why doesn't it work?
02:42.36Dr-Linuxoohhh
02:42.54Dr-Linuxexten => 7777,1,Goto(ivr,s,1) << this extension should under the [default] context?
02:43.02*** join/#asterisk ntwrknggeek (n=ntwrkngg@pool-70-105-186-165.alt.east.verizon.net)
02:43.16Qwell[]Why would you want to go back to the ivr if somebody dialed an extension?
02:43.48ntwrknggeekI am looking for some help with an asterisk@home install...
02:44.02Qwell[]ntwrknggeek: #asteriskathome
02:44.16Dr-LinuxQwell: yes sir
02:44.24ntwrknggeekthanks qwell
02:44.25Qwell[]*@h only barely resembles asterisk
02:44.36*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
02:44.39ntwrknggeekoh really???
02:44.40Dr-Linuxmy actuall problem is that, i have more then 200 users in [default] context
02:44.51ntwrknggeekwhat do you mean quell
02:45.23m160858benjk?
02:45.30Qwell[]Dr-Linux: okay, so whats the problem?
02:45.51mog_homeasterisk @ home isnt horrid just dont expect help here
02:46.00Qwell[]it is horrid
02:46.11Qwell[]mog_home: You heard the story about my tech writer, right?
02:46.24mog_home?
02:46.30mog_homei am the biggest a@h hater
02:46.33Qwell[]I'm having to teach her all of the options in AMP, so that she can mockup a new GUI for me to write
02:46.36mog_homebut the core is the same
02:46.46mog_homeyour writing a gui qwell?
02:46.51Qwell[]supposed to be
02:46.58Qwell[]It's not gonna suck though
02:47.06Qwell[]besides being written in C#/ASP.NET :D
02:47.23ntwrknggeekWhat is the difference between *@h and *?!?
02:47.33Qwell[]ntwrknggeek: *@h completely butchers the configs
02:47.34mog_homeBLECH
02:47.36mog_homewhat the hell
02:47.46Dr-LinuxQwell: problem is that, caller can listen all greeting etc, but i want caller if he wanna dial any extension from [default] context during greeting he should
02:47.49Qwell[]mog_home: mine isn't gonna butcher said configs. :D
02:47.55Qwell[]Dr-Linux: so let him
02:48.00mog_homewhy not do the right thing qwell
02:48.06mog_homego make rt stuff
02:48.15Qwell[]mog_home: It is going to use realtime
02:48.23Qwell[]and not the BS that AMP does, where it has its own tables
02:48.29mog_homei mean all of configs in rt
02:48.32Qwell[]yeah
02:48.33mog_homedont touch flat files
02:48.37Qwell[]not going to
02:48.40mog_homeif you do its gonna be uggies
02:49.07Qwell[]gonna even do blah.conf => odbc,asterisk,blah.conf
02:49.15Qwell[]so I grab the general sections and such too
02:49.22Dr-LinuxQwell[]: :S
02:49.39mog_homenice
02:49.58Qwell[]mog_home: it'll be decent.  It'll still suck because it's a gui, but...
02:50.31Qwell[]my work is actually putting some money behind writing it.  It'll have documentation, a few fulltime programmers, testing...
02:50.58*** part/#asterisk kimc (n=freenode@pcp04041197pcs.wbrmfd01.mi.comcast.net)
02:50.59mog_homeim fine with guis
02:51.06mog_homejust as long as i dont have to touch em
02:51.08*** join/#asterisk kimc (n=freenode@pcp04041197pcs.wbrmfd01.mi.comcast.net)
02:51.13Qwell[]heh
02:51.23mog_homewhat i want though is a phpmyadmin for ncurses
02:51.26mog_homeso if customer has rt
02:51.32mog_homei can go view mysql tables fast
02:51.34mog_homeas i am lazy
02:51.37mog_homelazy as all
02:51.38Qwell[]we're just not going to be able to admin * for all of the different sites
02:51.44*** join/#asterisk jaristizabal (n=jaristiz@69.79.133.185)
02:51.49Qwell[]so we need something that doesn't completely suck, for them to config stuff on their own
02:51.54Dr-Linuxasteriskmonkey: sir really thanks, you were right, i'm done now it works what i want :)
02:52.10ntwrknggeekQwell[] but for the basic home use and someone who is not all that good with linux *@h is ok?
02:52.17Qwell[]ntwrknggeek: not really
02:52.29Qwell[]You'll never learn * or Linux if you use AMP
02:52.30mog_homedoing asterisk is really easy
02:52.37mog_homeexactly
02:52.48camonzhi, i've a question, since * is OSS isn't the bussiness to provide administration for the service?
02:52.57Dr-Linux<Qwell[]> You'll never learn * or Linux if you use AMP << i gree
02:52.57Qwell[]camonz: huh?
02:53.02*** join/#asterisk wildcard0 (n=generic@S0106006097e16040.vc.shawcable.net)
02:53.07camonzinstalling, configuring the dialplan, etc
02:53.14Qwell[]camonz: What business?
02:53.17camonzat least that's the way it's done over here
02:53.20camonzvoip
02:53.26Qwell[]huh?
02:53.30camonzsetting pbx for enterprises and such
02:53.31wildcard0hey.  im trying to connect asterisk to an emergent system and im getting 'Got SIP response 400 "Bad Request" back from <ip>'
02:53.36wildcard0any ideas on how i could fix this?
02:53.37Qwell[]You saying the provider should setup your pbx?
02:53.41camonznope
02:53.56camonzanother enterprise
02:54.02Qwell[]camonz: well, people aren't going to do it for you for free
02:54.02ntwrknggeekQwell[] Do you recommend a good resource for learning * with AMP and a little bit of linux?
02:54.06camonzof course
02:54.18Qwell[]ntwrknggeek: There are no resources for learning * when using amp.  amp breaks things
02:54.30Qwell[]If you want to actually learn asterisk, ...
02:54.32Qwell[]~docs
02:54.33jbothmm... docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com
02:54.35camonzthat's what i mean, in making a GUI that makes easy to configure things for the end user isn't that going to hurt your bussiness as a consultant
02:54.44benjkI recommend learning about networking first if you have to learn "a little bit of Linux"
02:54.57benjkI was helping this German chap last night
02:55.07benjkhe got all sorts of SIP error messages
02:55.15camonzby example over here there's a company that charges about 9000$ for instalation
02:55.21benjkturns out he diodnt have any DNS
02:55.27wildcard0was one of them '400 "Bad Request"'?
02:55.32camonzplus you have to contract them for maintenance
02:55.37benjkbut he managed to get on to IRC
02:55.47ntwrknggeekThanks Qwell[]
02:55.51benjkthough he couldn';t tell me how he did it
02:55.55camonzi.e. adding mailboxes, configuring the dialplan, and things like that
02:56.07benjkhe had no clue whatsoever about networking
02:56.26benjkif you don't know about networking on Unix, forget about Asterisk
02:56.32mog_homebuy a router, plug things into said router
02:56.41mog_homeset asterisk box as static
02:56.46mog_homeand port forward or dmz
02:56.47mog_homeyour done
02:57.05benjkmog, if you cant' condigure your Asterisk box to connext to the router then the router will do you no good
02:57.30benjkif your Asterisk box cannot resolve any DNS names, you won't get very far
02:57.42mog_homeyou pull a dhcp
02:57.48mog_homeand then you just set it static
02:57.52mog_homenetworking is easy
02:58.00mog_homeif you need help
02:58.03benjkwell, that's why I said, learn about networking on Unix
02:58.03mog_homebuy a linksys
02:58.09mog_home^+^
02:58.22benjkso that you know 1) what you have to do and b) how to do it
02:58.51benjkwhat good does that Linksys do you if your Asterisk box won't talk to it
02:59.04mog_home<PROTECTED>
02:59.06mog_homeoops
02:59.10file[laptop];)
02:59.19mog_home^_^
03:00.34wildcard0so...SIP errors? anyone?  anyone?
03:00.55camonzi mean, the real question is wich bussiness models are you using to make profits with *
03:01.10wildcard0mine's working :)
03:02.31asteriskmonkeywhat make money using * ???
03:02.34asteriskmonkeytell me more
03:03.20wildcard0ya.  unfortunately im not allowed to talk about it
03:04.59camonz:->, here in vzla there are just 1 or 2 bussiness doing the research on how * works so they can begin doing installations
03:05.31camonzso, the concept of voip is kind of new over here
03:05.35Vcoi'd suspect your math may be off
03:05.58Vcosee...you need to actually know of the other ones out there, and count them too
03:06.27camonzsure, maybe more than 1 or 2, but still not many
03:07.22*** join/#asterisk ritesha (n=ritesha@c-24-6-80-22.hsd1.ca.comcast.net)
03:07.33*** part/#asterisk kimc (n=freenode@pcp04041197pcs.wbrmfd01.mi.comcast.net)
03:07.54riteshahello.
03:08.59Dr-Linuxasteriskmonkey: if the caller doesn't press anything in end of greeeting, it says "goodbye" but user is still connected,
03:09.02Dr-Linuxhere
03:09.03Dr-Linuxexten => t,1,Playback(vm-goodbye)
03:09.04Dr-Linuxexten => t,2,Hangup()
03:09.06riteshaI need to understand how asterisk connects to database. Specifically, how to add SIP extenstions to a database  so that asterisk can read it. Do I need to have a standard table name e.g. sipfriends or is there some toher way to specify a table name to asterisk?
03:09.22Dr-Linuxi want the user Hangup
03:10.04Dr-Linuxor start again from first
03:10.32riteshai saw some examples where the sip.conf just specifies the database name, username and passwd. I am curious to to know how does the asterisk know whihc table belongs to sip extensions for bunch of other tables that my database may have?
03:11.11riteshadon't see details description about asterisk + database anywhere include voip-info.org
03:11.15riteshaplease help...
03:14.58Qwell[]ritesha: realtime
03:15.23riteshaQwell: realtime meaning?
03:15.32Qwell[]realtime is what you want to google
03:15.41riteshaoh!! thanks... let me try
03:18.51jaristizabalritesha, maybe help you: http://www.asteriskguru.com/tutorials/realtime_pgsql.html
03:21.03*** join/#asterisk m160858 (n=jsaenz@200.89.12.46)
03:21.09*** join/#asterisk rowter (n=SilverDr@201.135.26.195)
03:21.20riteshathanks Qwell and jaristizabal. Found the information on voip-info.org. Reading through it.
03:21.50jaristizabalok
03:28.09*** join/#asterisk BleedingMe (n=Bleeding@ppp-71-137-216-107.dsl.scrm01.pacbell.net)
03:29.13BleedingMeanyone know an easy way to generate a random number variable in asterisk... like, generate a random number between 1 and 100
03:29.16BleedingMe?
03:29.23mog_homehrmm
03:29.27mog_homewell
03:29.37mog_homeyou can execute system commands
03:29.37Vcoflip some coins?
03:29.48mog_homeand then read them off with readfile
03:29.51mog_homehmm
03:30.00mog_homeoh wait
03:30.01mog_homei got it
03:30.05mog_homethe unique id
03:30.06mog_homeof the call
03:30.08mog_homeits random
03:30.15BleedingMeunique id?
03:30.25Chujiapp_mysql
03:30.37Chujigrab a key
03:31.05bbzWould anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. --
03:31.10mog_homebbz
03:31.12mog_homebegone
03:32.35alephcombbz: I messaged you and you didn't respond.  You mustn't want to be rid of them too badly
03:34.14phsdshftDoes anyone have experience working with Sipura devices behind NAT connecting to an asterisk server that has a public IP?
03:35.08alephcomI little bit.  What's going on?
03:35.18phsdshftwell.. I keep getting one way audio
03:35.42Vcoget a second phone
03:35.44Vcohahahaha
03:35.49phsdshftI can hear the phone menu on the asterisk server (or person off of the asterisk server) but they can't hear me
03:36.06phsdshftme meaning the phone off of the sipura device
03:36.19alephcomWhat kind of firewall?
03:37.04phsdshftits a linksys broadband router
03:37.25phsdshftI have ports 5060, 5061, and 10000 - 20000 forwarded to the sipura
03:38.22phsdshfthowever
03:38.30phsdshftI get one way audio even if its directly off of my cable modem
03:38.59phsdshftthe other side has an access list that drops traffic going to the asterisk server... but it allows ports 10k - 20k, 5060 and 5061
03:39.34phsdshft(udp)
03:44.41phsdshftsoo.. yeah.. heh
03:46.51alephcomHmmm, I'm not sure what the problem is.  If it does the same thing directly off of the cable modem is very interesting.
03:46.53benjkphsdshft: its not as simple as forwarding
03:47.15phsdshftbenjk: Right.. the sipura has to be configured to do nat as well..
03:47.17benjkbecause it matters what's inside the SIP messages
03:47.26phsdshftI specified the ext ip in the sipura as my public ip
03:47.30phsdshftset nat mapping to yes
03:48.04benjkthe TO: FROM: and the CONTACT: fields must match the *endpoint*, not your router
03:48.08phsdshftand set ... handle via received, substitute via addr and everything but STUN to yes
03:48.34phsdshftbenjk: endpoint meaning the private, or public address?
03:48.52*** join/#asterisk nxtw (n=matt@adsl-69-221-114-151.dsl.akrnoh.ameritech.net)
03:48.57benjkthe private address
03:49.28benjkultimately
03:49.48benjkbut it depends on the sip engine how the device handles the traffic
03:50.29phsdshftwell... on the asterisk side I have (changing it to one line to make it not scroll the room):
03:50.41benjkif it gets the traffic looks in the field and discovers that the address isn't its own, then says "hey, that's not meant for me" and drops it, well then you get problems
03:50.49BleedingMeis there a way to play an audio file directly to a channel while on the phone?
03:51.37JunK-YBleedingMe: u mean with the CLI?
03:51.53BleedingMesure
03:51.59phsdshft[2001]; type=friend; username=2001; host=dynamic; context=customersin; allow=all; dtmfmode=inband; nat=yes; qualify=yes;
03:52.09phsdshft(the ; means new line, they arent actually in the config)
03:52.54benjkdidn't you say sipura?
03:53.32phsdshftyes.. thats from the asterisk side
03:53.42phsdshftI have control over both endpoints
03:54.03benjkthe asterisk box is on a public address?
03:54.06phsdshftyes
03:54.22benjkand the sipura is behind NAT
03:54.33phsdshftit has a router with an acl in front of it... thats it.. and that acl allows 10k - 20k, 5060 and 5061 through (udp)
03:54.41phsdshftcorrect
03:55.05benjkthe asterisk is behind a NAT router?
03:55.31benjkforwarding addresses is not the same as having a public address
03:56.00phsdshftno.. it has a public (not private) address, and no nat is being done
03:56.05phsdshftits behind a cisco 7204
03:56.06benjkok
03:56.18phsdshftthe other side is behind a linksys (the side with the sipura)
03:56.25*** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com)
03:56.36benjkdid you configure rtp.conf to make sure it doesn't use any ports outside of your range
03:57.56phsdshftyes.. it says 10k to 20k, which is inside of the range
03:57.56benjkok
03:58.02phsdshftof course, at the moment I'm noticing that after upgrading the sipura's firmware I no longer am receiving RTP packets... heh doh
03:58.04benjkand the sipura registers with the Asterisk server?
03:58.05phsdshft(or sending)
03:58.09phsdshftyes
03:58.23phsdshftbut it didnt work before upgrading the firmware either
03:58.31phsdshftalthough at the moment it has no audio at all
03:59.12benjkhave you done a sip debug on the asterisk to see whats going on?
03:59.30phsdshftyes.. but it doesnt seem to be helpful
04:00.13*** join/#asterisk robl^ (n=robl@dsl093-025-118.hou1.dsl.speakeasy.net)
04:00.23benjkwhat does it say as contact in the incoming SIP messages from the sipura
04:01.29benjkalso, does the sipura user rport ?
04:01.36benjker use rport
04:01.49phsdshftfrom, to or via?
04:02.30benjkwell at first when the connection is established, the important field is "contact:"
04:02.45benjkthat tells the other side where to reply to
04:03.07phsdshftcool now audio works (after restarting the asterisk server.)
04:03.10phsdshftone side
04:03.12phsdshftnow let me check
04:03.12benjkso if that is off, then you have no chance of getting a two way converation going betwen the devices
04:05.24benjkI have had a whole array of problems with SIP over cable here in Japan -- I am being told it's better in the US
04:05.35*** join/#asterisk stkn__ (i=nobody@gentoo/developer/pdpc.active.stkn)
04:05.45phsdshftContact: 2001 <sip:2001@66.189.25.218:212>
04:06.04Qwell[]kram: evening
04:06.06phsdshftok, so it has my public IP
04:06.21benjkanyway, one of the problems I frequently encountered was that the cable guys seem to drop packets or delay them in a way that the NAT entries time out, that's something to watch out for
04:06.39*** join/#asterisk sudhir492 (n=sudhir@pool-71-114-48-29.washdc.dsl-w.verizon.net)
04:06.42sudhir492Hi all
04:06.42Vcobenjk where are you in japan?
04:06.49benjkTokyo
04:07.10sudhir492Anyone here has pap2-sp2k.bin ?
04:07.47benjkphsd: is that the address of the linksys?
04:07.54phsdshftthe public ip of it yes
04:08.40benjkok, so Asterisk knows where to send the packets off and they are likely to reach the linksys (unless there are troubles on the cable network)
04:09.22benjkdo you have a means to see (at the sipura said) if those packets show up at the sipura?
04:09.26phsdshftwell, I'm getting audio from the asterisk server
04:09.29Vcoany idea if you can use fritz isdn card with ntt?
04:09.33phsdshftits audio from the sipura to the asterisk server that doesnt work
04:09.49benjkVco: Most likely not
04:10.01Vcothats what i figured..boo
04:10.04benjkJapanese BRI is different from Euro BRI
04:10.09Vco**nod**
04:10.34benjkVco: Mark Spencer is looking into getting the Japanese Caller ID working for Zaptel
04:10.49Vcowhats the easiest way to get multi line service out there plugged into a server?
04:10.51Vcolike..
04:10.51benjkso you could use a TA and use a Zaptel card
04:10.53phsdshfthmm that depends if this is a switch or a hub on the back of the linksys.. one sec
04:10.55Vco6 or so
04:11.12coppicebenjk: do you have any lines with japanese caller ID?
04:11.28benjkcoppice: yes I do
04:11.47coppicegood. I will want someone to test japanese caller ID in openpbx soon
04:12.00benjkAnd I also have the English documentation (PDF) in which the NTT line interface is documented (in great detail)
04:12.11Vcoo.O
04:12.22benjkcoppice: just let me know, you can log in here remotely
04:12.39krambenjk: we already have almost everything
04:12.45coppiceyuo gave me that a long time ago, and I implemented it in spandsp. I just never got a chance to try it out before.
04:12.52benjkMark, that's terrific!
04:12.59kramwe already have the demodulator
04:13.03kramit already supports the mode proposed
04:13.06benjkcool
04:13.10kramchan_zap already supports using it
04:13.27kramthe only things that are different are going off hook and the semi-parity bit
04:13.34kramnone of which should be so complex
04:13.39benjkVco: you should be looking for a TA and get a Zaptel card ;-)
04:13.59mog_homeman i thought you said semi-party marko
04:14.03mog_homei was gonna head to office
04:14.10benjkfor 6 lines you could probably use Digium's new tdm2400 card
04:14.18benjkpopulate it with 6 FXO modules
04:14.37Qwell[]multiples of 4 I thought
04:14.41Vco6
04:14.46coppicethe japanese caller ID is really weird, mixing parity and CRC in the same block
04:14.47Qwell[]oh, 6 modules...right
04:14.49benjksemi-parity bit
04:14.59benjkgeez that sounds like a paradoxon
04:15.00Vcoi thought it was 6X4 modules
04:15.04krami figure it's easiest to run the demodulator without parity turned on
04:15.09kramthen manually do the parity
04:15.13Qwell[]Vco: 6 modules carrying 4 ports each
04:15.15kramsince it's 7 bits and not 8, it's probably just easier this way
04:15.32kramwe go ahead and run it as an 8-bit modem, and just calculate the 8th bit as parity when it should be and leave it alone when we do the crc-16
04:15.33Vcooh..wait..ya
04:16.03benjksounds like a plan
04:16.04phsdshftno its a switch so I cant see if the rtp packets hit the sipura
04:16.43benjkphsd, how about using some other device on that end first?
04:17.00phsdshftlike a soft phone?
04:17.00benjkfor example a Linux box with a SIP spftphone and Ethereal
04:17.29benjkthen see if you get it working with that since you can see what traffic is coming in and going out
04:17.38benjkthen try the sipura again
04:18.10phsdshftwhats a good sip client for windows (my linux box does not have a microphone, whereas my laptop has one built in)?
04:18.13Vcoknow of anyone doing termination/origination over there then?
04:18.21Qwell[]phsdshft: xlite is decent I hear
04:18.24benjkVco: did you mean virtual phone lines? DIDs?
04:18.54VcoWell, I'm lookign for Osaka DID's for now..
04:19.02Vcowhich hasn't been fun
04:19.11benjkphsd: X-lite has a debug window that shows you the sip messages but it can be a bitch to configure
04:19.27benjkVco: how many do you need?
04:19.38Vconot many for now.
04:19.43Vcolike less than 10
04:19.59Vcomaybe 3 to start and test with
04:20.15benjkThe trick is to get multiple parties who will be able to share a PRI
04:20.27Vcoor 1 that can support multiple channels
04:20.30phsdshftbenjk: I can use windump while using a sip phone probably
04:21.11benjkphsd: wouldn't know about that, I don't use Windoze, but yeah, use whatever you are familiar with
04:22.25benjkVco: if we can find three parties who are interested in Osaka virtual phone numbers or DIDs we can set up a PRI there
04:22.49Vconeed a number or two for inbound dialing to canada, and sounds like some guy that has 11 locations for his shop around osaka that wants to possibly do some
04:22.59Vcobusiness
04:23.09Vcoinlaw are still over there...
04:23.16benjkVco: where are you located?
04:23.20VcoCanada
04:23.26benjkOh I see
04:23.28VcoInlaws are still over there
04:23.38Vcoaparantly pimping my skills to their customers
04:23.40benjkok
04:23.47benjk:-)
04:23.50Vcoheh..heh.
04:24.06benjkwell, if you need somebody to do ground work at location, let me know
04:24.11Vconishnomia city around there..
04:24.15Vcowill do..
04:24.21Vcosaves me a 18hr flight
04:24.25riteshaRealtime mapping for 'queues' found to engine 'mysql', but the engine is not available....anyone knows what's going on?
04:24.25benjkbenjamin at sunrise-tel dot com
04:24.32Vcoahhhhhhhh
04:24.46Qwell[]ritesha: Did you setup the mysql engine?
04:24.47Vcoa familiar domain name
04:24.49Vco:)
04:24.55benjk:-)
04:25.10riteshahmm. what do I need to setup the mysql engine
04:25.20riteshai started the mysqld service
04:25.21Qwell[]ritesha: mysql
04:25.27riteshaisn't that enough?
04:25.57Qwell[]You need to create a database for asterisk, create the tables, create users, setup the configs in * to know all of this information
04:25.57benjkVco: the telco I am working with here in Tokyo has datacentres in Osaka too
04:26.24benjkthey can provide PRI in Osaka and they are higher quality and lower price than NTT
04:26.46riteshai did all the setup. but the engine is not available is something of a trouble.
04:27.21riteshaif it's somethign to do with the * setup, I could look again. Else I could check for the mysql gotchas?
04:28.18benjkVco: still there?
04:28.29Vcoyo
04:28.47Vcois kddi only for international?
04:28.53benjkIn the short term, your client could get a VoIP service from OCN
04:29.13benjkI know how to hook Asterisk up to that directly, without any adapters ;-)
04:29.22Vcoyea, been having fun navigating for info from here...
04:29.23benjkofficially you cant do that
04:29.51Vcoor getting my wife to explain to my father in law to get info from him about what he pays etc
04:30.15benjkkddi used to be internationaly only but I think they now do long distance too
04:30.47Vcoi had a link from something i thought was ntt west, for what sounded like voip kinda stuff..
04:31.31benjkAnyway, if you get ADSL from OCN on a single NTT analog line, then you can have their bundled VoIP service for about 500 yen per month
04:31.43benjkthis gets you a 050 telephone number
04:31.52benjk]which can be called from the PSTN
04:32.17benjkthe cost of calling a 050 number from PSTN lines is 10.5 yen per 3 mins
04:32.19Vcoinlaws have a yahoobb line upsatirs at the house, but down in the salon they have a business line
04:32.30benjkYahooBB sucks
04:32.34Vcotell me about it
04:32.45benjkfalse hangups all the time
04:32.51benjkbad line noise
04:32.56Vcohad 3 modems go tits up
04:32.56benjkbad sound
04:33.01benjkyep
04:33.20benjkand you cannot ever hope to get Asterisk working with that directly
04:33.34benjkalways need to go through the analog port
04:33.37Vcoyea, thats why i was wondering about isdn
04:33.44*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
04:33.44Vcoi think thats what they have downstairs
04:33.53benjkNo, get ADSL from OCN on their business line
04:33.54coppicewhat is it with broadband modems? their reliability makes windows look solid, and all makes seem the same
04:33.54Vcois it net64 or something
04:33.59benjkthen cancel YahooBB
04:34.08Vcoocn eh?? have an english link?
04:34.22Vcomeh, i'll google
04:34.29benjkOCN is the Internet service of NTT COmmunications
04:34.52benjkYour inlaws will know how to sign up for OCN
04:34.58*** join/#asterisk file[laptop] (n=jcolp@mctnnbsa24w-142167049176.nb.aliant.net)
04:35.18benjkjust go to an electroncis store, and they have the form
04:35.45Vcoor any  streetcorn ein umeda on a saturday afternoon in the summer ;)
04:35.52benjkyep
04:35.54Vcostreetcorner in ...
04:35.56Vcoeven
04:36.07benjkOCN is about on par with YahooBB now
04:37.06Vconow..inbound calls are free on landling too?
04:37.11Vcoor just mobiles?
04:37.13benjkmany smaller ISPs who think they are too small to afford their own VoIP service (because they only know NEC and Fujitsu and have never heard of Asterisk) resell OCN service under their own name without making any profit (just so their customers dont churn)
04:37.33Vcoor does it depend on plan/carrier/time of day/prefecture/populatin density in the area
04:37.48Vcoand a million other factors for determining call rates there..
04:37.49benjkinbound calls are alwasy free
04:37.51Vcoinsane
04:38.06benjkunless you have a toll-free number of course
04:38.45benjkin Japan VoIP service is always bundled with your ISP
04:38.59benjkif you want to change, you have to switch your ISP
04:39.15phsdshftcool.. I fixed my voip problem
04:39.27benjkphsd: what was the trouble?
04:39.42Vcowakatta
04:40.56phsdshftbenjk: I changed a bunch of things at once... but... part of it was I had dtmf set to inband instead of rfc.... so, the way I was testing if audio got from me to the asterisk server was flawed.. so it might have worked correctly (except dtmf) after I upgraded the asterisk server code..
04:41.37phsdshftI had upgraded the firmware on the sipura, the version of asterisk on the server side, reset the sipura to defaults and reconfigured it and disabled 4 options in the nat support parameters on the sipura
04:41.45phsdshftso one of those fixed it heh
04:42.00Vcowhat about for solid internet connection there? dsl/fiber?
04:42.07Vcofibre
04:42.16Vcowhatever
04:42.57*** join/#asterisk mcadory (n=mcadory@208-149-64-246.adsl.nexband.com)
04:43.08*** part/#asterisk mcadory (n=mcadory@208-149-64-246.adsl.nexband.com)
04:43.26*** part/#asterisk shmooz (n=shmooz@H142.C72.B0.tor.eicat.ca)
04:43.51benjkVco: FTTH is best of course
04:43.59Vco:)
04:44.08Vcoas far as price/perf
04:44.10benjkand the best value for money you get from USEN
04:44.44benjkabout 50 USD per month for 100Mbit full duplex with a block of 8 ip addresses included
04:44.51Vcoshit
04:44.52benjkwhich is a steal
04:44.55Vcoya
04:45.05Vcolast i heard it was still $85-$100ish
04:45.08benjkbecause in Japan ip addresses are worth more than diamonds
04:45.38benjkon OCN if you get fibre, you pay 350 USD per month just for the IP addresses
04:46.27Vcomaybe when i'm off this contract i 'll need to head back over there instead of going snowboarding..  :/
04:46.33benjkVco: it is still 85-100 with other providers
04:46.48benjkhaha
04:47.13Vcohaven't been over since spring '04
04:48.31benjkBTW, OCN have a business VoIP paclkage
04:48.50benjk4 lines on a single ADSL circuit
04:48.53*** join/#asterisk konfuzed (n=KonfuzeD@H129.C72.B0.tor.eicat.ca)
04:49.02Vcohmm....
04:49.17benjkso you could get four 050 numbers directly into Asterisk
04:49.31Vcointeresting, i have a spave tdm400 card too
04:49.38benjkor a single number with 4 channels
04:49.43Vco**nod**
04:49.46benjkyou don't need it
04:49.57Vcoahh.
04:50.03benjkas I said, I can make Asterisk connect to OCN directly
04:50.05PakiPenguinVco: send it to me :p
04:50.07Vcoyea..
04:50.23benjkofficially you cannot do that
04:50.35benjkbut I have it working here ;-)
04:50.40Vcocool.
04:50.49benjkworks very nicely
04:50.53benjkG711
04:50.58benjkand caller ID
04:51.04benjkcrisp sound
04:51.26benjknot like YahooBB, which sounds like your'e calling somewhere in Africa
04:52.21VcoWife is pretty active in the cultural center here, so she knows a fair number of people from her prefecture, want to get a local inbound number there, since a lot of other people there we also know only use a mobile phone..
04:52.28Vcoso calling here is a pain in the butt
04:52.35h3xwith a bluebox, and a operator line
04:52.35h3xheh
04:53.59Vcoi guess DID's are the only real need tho..
04:54.03phsdshftI'm getting a 404 not found error when doing a sip debug for calls incoming to the sipura (outbound work fine)... and the asterisk server reports 'everyone is busy/congested' ... the line I have in my extensions.conf is: exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@2001,30,r)
04:54.06phsdshftany ideas?
04:55.05Vcocalling there is 3.5/min without looking for the cheapest possible rates around, so i don' tknow if there is a grounbreaking savings by having a local number there
04:55.17Vcoat least on the scale we'd be using it inititally
04:57.14benjkfour inbound you will find that most people now either user a mobile, or they will use a VoIP line
04:57.47benjkif they are on YahooBB, then the call to your 050 OCN number will be 10.5 yen per 3 mins like it is on NTT analog lines
04:57.49phsdshftbenjk: any ideas?
04:58.13Vcowhat about toll free stuff there?
04:58.14benjkif they are on any other VoIP service, then their call to your 050 number will be FREE
04:58.40benjktoll-free is prohibitive
04:59.12benjklast time I checked on toll-free you had to pay for inbound long distance charges
04:59.33phsdshftmy toll free is flat rate per minute ... higher intrastate than interstate
04:59.56Vcoand then there's calling mobiles as well..
05:01.14benjkthey best way to do toll free in Japan, maybe you should get 1x YahooBB and connect via Zaptel to the Yahoo modem/ata, and 1x OCN connected directly over SIP, then you tell your callers to call the 050 number of your YahooBB line if they are on YahooBB (free) and the others you tell to call you on your OCN 050 number (free)
05:01.35Vcoya..
05:02.00Vcobeen dealing with a company, totally insane setup right now..
05:02.03Vcook..
05:02.09benjkunfortunately there is no way to tell by the 050 number if it is YahooBB or OCN (and allied services)
05:02.23Vcothey have offices in Toyko, Hawaii, and Sydney
05:02.46phsdshftI'm getting a 404 not found error when doing a sip debug for calls incoming to the sipura (outbound work fine)... and the asterisk server reports 'everyone is busy/congested' ... the line I have in my extensions.conf is: exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@2001,30,r)
05:04.03Vcomost of the calls in hawaii are to tokyo, they call the customer, but for the customer to call them back is overseas, so they leave the tokyo number, which may or may not still be long distance for the caller, the customer calls back, the receptionist asnwers, ....instant messages the person in Hawaii, and make arrangments for the person in Hawaii to call backteh person in japan right away
05:04.19*** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net)
05:04.39Vcoall i could ask them when they explained this to me was
05:04.43Vco"WHY?"
05:04.45benjkphsd: did you configure inbound on your sipura? those sipuras can be a bitch to get set up right
05:05.10phsdshftI /think/ so...
05:05.51benjkphsd: again, back to the softphone refernece ;-)
05:06.12phsdshftben: I got it working for outbound calls now... so just inbound /calls/ (not audio) are a problem now
05:06.15benjkVco: sounds like a perfectly sound Japanese business process
05:06.23Vcoyup..heh.
05:06.31phsdshftso I have two way audio now... inbound calls dont work at all at the moment and I think its likely the way my asterisk config or something is
05:08.17benjkVco: set up an OCN ADSL/VoIP in their Tokyo office, route that 050 to Hawaii, problem solved
05:08.26Vcothats what i'm looking at
05:09.22Vcothey do a lot of inter-office as well, so be likely doing 2 severs over iax or the like etc
05:09.29Vcoservers
05:09.43Vcogoddammit
05:09.56Vcowhoa..i guess there is a lot crud stuck in my keyboard
05:10.08benjkyes, and perhaps replace their office telephone system along with it ;-)
05:10.17Vcoyup
05:10.51Vcothe kinda of setup they have can't go anywhere but up quite frankly..
05:11.05benjkas Mark is now fixing the Japanese Caller ID, it will be possible to actually deploy some Asterisk servers as PBXes in Japan
05:11.40coppicebenjk: do you still face serious permission to connect issues, or is japan loosing up?
05:11.59phsdshftdoh... the dial line needed to be userid@contextinsip.conf not number@contextinsip.conf
05:13.04benjkcoppice: I have cultivated a relationship with JATE, been there many times for meetings, explained Asterisk to them and the Zaptel stuff
05:13.42benjkI am also familiar with the process and what is needed to fill out the (immense) paperwork
05:14.19benjkI also got a Japanese engineer who I can call upon to help with preparing some of the documents
05:14.51benjkso, I feel confident to take on the task of getting JATE approval, at least for digital
05:15.01coppicejapanese seem to like paperwork. buying a train ticket actually gets you several which machines along the way swallow one by one. weird
05:15.05benjkanalog is more elaborate
05:15.24Vcoit's the same ticket
05:15.41Vcoit spits out the other end
05:16.27benjkcoppice: the type approval does cost a bit of money though
05:16.31coppicenope. I had a two step journey. I got three tickets. I had to put all three through each machine, and less came out each time
05:17.05benjkhaha that's because you took the Shinkansen
05:17.10Vcoheh..heh.
05:17.20benjkone ticket is for the distance (basic fee)
05:17.33Vco"zones"
05:17.34benjkthe other is for the express fare
05:17.45benjka surcharge so to speak
05:17.51*** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV)
05:17.55Vco"luxury tax"
05:17.55benjkand the third one is for the Shinkansen
05:18.03benjkanother surcharge
05:18.10coppicethe two tickets for the stages had prices which didn't add up to the overall price on the third ticket. i never figured that one out
05:18.26benjkand you had to use the shinkansen ticket to get into the shinkansen area of the station
05:18.39Vco*sigh*
05:18.50Vcoi want to go back
05:18.59h3xthem japanese must have been hanging out with the us government to come up with that shit
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05:19.51benjkh3x: trust me, the Japanese need nobody's help to come up with their weird processes
05:20.12h3xso, you guys use T1s, and is it ulaw?
05:21.46benjkwhat in Japan?
05:21.52h3xyeah
05:21.53benjkwe don;t use T1s
05:21.58benjkwe use J1s
05:22.01h3xi thought it was north america,
05:22.02h3xoh
05:22.19h3xbut isnt it electrically the same as t1?
05:22.28benjkthat's why those interface cards (some of them) are branded T1/E1/J1
05:22.44benjkit is a futzed up T1
05:22.48h3xright
05:22.52coppiceits electrically the same, its just twisted for incompatibility
05:23.01h3xbut it is ulaw right?
05:23.16coppicethey got that into Taiwan too, since NEC basically build their early digital networks
05:23.27h3xi thought all that MFC R2 signalling did a good enough job of fucking up compatibility
05:23.42h3xon E1s
05:23.47coppicethey don't use R2
05:24.10h3xyea i know but seeing as ericcson thought it would be a great idea to come up with a different R2 in each country
05:24.25h3xim suprised you had T1 influence there
05:24.30coppicethat was nothing to do with ericsson
05:25.50coppiceactually, just one form of R2 in each country would be an improvement :-)
05:25.57h3xhaha
05:26.13h3xman i cant believe aculab had the balls to sell me a $13k card in 2000
05:26.19h3xknowing i was using sun solaris
05:26.27h3xand they didnt even have any ulaw algorithyms working yet
05:26.33h3xon the DSPs
05:26.35h3xfor like 2 years
05:26.44h3xi think they were counting on me not getting along that far
05:26.47h3xbefore them . hah
05:27.16h3xbut whats even more balsy
05:27.25*** join/#asterisk genmud (n=genmud@ip68-98-82-206.ph.ph.cox.net)
05:27.27Vcobah..
05:27.39coppiceaculab keep far more of their customers happy than the competitors do :-)
05:27.50h3xis their "asterisk driver".  all it does is nails up channels to record and playback
05:27.54h3xlike the dialogic drivers
05:28.04*** join/#asterisk alphaque (n=alphaque@218.208.239.119)
05:28.07h3xaculab is doing terrible in the US
05:28.12h3xhas always been
05:28.20alphaquethey seem to be doing ok in oz
05:28.23alphaqueor so i hear
05:28.43h3xthey actually engineered the stuff well but they have too many parallel development projects that arent compatible with each other
05:30.46h3xlike, i was fortunate enough to start with the TiNG drivers on solaris, which was ported to linux eventually
05:31.00h3xthe previous generation drivers died off on os/2, sco, linux, etc.
05:31.07h3xthen TiNG took over
05:31.22*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
05:32.03h3xThey have TERRIBLE code samples
05:32.11h3xim glad its all ANSI C though
05:32.33h3xbut i guess it dosent matter anymore
05:32.46h3xcoz the only thing it does better than asterisk is fax and asr/tts.
05:32.57h3xbut not for long
05:33.29coppicewhy not for long?
05:33.53h3xwe have faith in your t.38 and spandsp skills :D
05:33.59sudhir492exit
05:34.14coppicethat doesn't address 2 out of the 3 things you listed
05:34.14h3xand i donno, im sure cepstreal has something cooking, ...
05:34.23sudhir492ooops, sorry. wrong window
05:34.43h3xWell, actually you are best off with some type of HMP with nuance or scansoft stuff
05:35.03coppiceTTS is easy to cover. the asr part is the problem
05:35.09h3xyeah
05:35.23h3xi mean, the aculab boards dont do anything for asr/tts besides playback/record
05:35.48h3xthey do have a "free" asr and tts, but the asr connected word recognition cant do that big of a vocabulary at once
05:36.47coppiceI think they do V.34 fax now, which * will not do
05:38.03h3xyeah well they seem to have it togehter for faxes but those cards cost way too much to use just for faxing
05:38.07*** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw)
05:38.23*** join/#asterisk clinthome (n=clinthom@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net)
05:38.31h3xalthough
05:38.39h3xthe new voip board can do 500 transmits on ethernet
05:38.42h3xand its like 13 grand
05:38.50h3xso thats reasonable
05:39.00h3xas g.711 of course
05:39.03coppicethey provide cheap SS7 too
05:39.31h3xit has a lot of limitations
05:39.42h3xtcap is terrible, you may as well start from scratch
05:40.24h3xi dont remember if its still limited to 8 T1/E1s per linkset or not
05:40.30h3xfor ISUP
05:40.32coppicereally? I don't know anyone who has used it, but I thought their SS7 was supposed to be very good.
05:40.57h3xit is, but its really low level.  although ISUP is the same general interface as doing any other CAS/CCS signalling
05:41.32h3xOh, I know.  ANSI ISUP is non-existant
05:41.46h3xthat pretty much brought all my ideas for it to a screeching halt
05:49.28h3xusing sangoma for ss7 is probably cheaper and a smarter thing to do
05:49.49phsdshftanyone doing distinctive ring with the sipuras?
05:49.57phsdshft(and asterisk)
06:17.46bbzWould anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. --
06:25.06Dr_Raywhy not post to the asterisk-biz mailing list
06:34.02*** join/#asterisk mrgoby (n=knoppix@pcp05307400pcs.wanarb01.mi.comcast.net)
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06:41.55h3x<PROTECTED>
06:41.56h3xheh
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06:48.28benjkmy ignore didn't seem to have worked
06:48.38benjkotherwise I wouldn't have seem his post
06:48.43benjk:(
06:49.10h3x<PROTECTED>
06:49.17benjkah
06:49.30benjkthanks
06:49.49Dr_RayI regulated Katty to the ether long ago
06:50.07benjkbbz is Katty?
06:50.11Dr_Rayno
06:51.00benjkstill I think bbz should be banned from freenode
06:51.12Dr_Raywhy?
06:51.30benjkbecause its very bad ettiquette
06:51.47benjkand its automated
06:52.00h3xdude hes been spamming every hour for a week
06:52.03Dr_Rayfreenode/irc would be empty based on ettiyuette
06:52.03h3xits like the guy went on vacation
06:52.23benjkif somebody did this while personally around and participating, well I guess it would still be acceptable
06:52.30*** join/#asterisk diamon (n=diamon@c-66-176-91-189.hsd1.fl.comcast.net)
06:52.54Dr_RayI did not know it was automated
06:53.04Dr_Rayhence my suggestion to take it to the biz mailing list
06:53.06benjkh3x: that's why I said its automated, its a bot, a spam bot
06:53.20benjkspam bots should be banned
06:53.46diamonOk, I've finally got some time (courtesy the holiday!) to get * working.  Is there any suggested distro of linux to use with it?  My two main prefs are Fedora and Slackware.
06:54.01Dr_RayI use fedora
06:54.17benjkwhatever distro you are familiar with works best
06:54.22diamonIs there an rpm for it, or did you build it?
06:54.24benjkusually, that is
06:54.29Dr_RayI build it from cvs
06:54.32Dr_Rayit's ez
06:54.44*** join/#asterisk tessier_ (n=treed@wsip-68-15-4-13.sd.sd.cox.net)
06:54.56Dr_Rayif you search teh wiki for fedora you will find all the install help you need
06:55.09benjkpretty much every serious astmaster builds his asterisk himself
06:55.14diamonDR: Yeah, compiling is easy, I was just trying to get myself used to using RPM's whenever I can.  It makes yum updates less likely to break things by accident.
06:55.19h3xwell it might be a regular irc client
06:55.21h3xwith a timer
06:55.38diamonbenjk: Ideally, I'd build my own rpm for a FC4 system.  :)
06:55.48benjkh3x: in my book that still counts as a spam bot
06:55.55Dr_Rayyou can, but asterisk is a moving target
06:55.55diamonBut anything that works is good enough for me.
06:56.20benjkdiamon: sure, I build my own installer for MacOSX
06:56.50diamonDr: I don't upgrade unless there's a security problem or a bug I'm impacted by, or a feature I feel I must have...  Learned my lesson about arbitrary upgrades on my MythTV system.  Now it actually works, and keeps working even.
06:57.02Dr_Raydiamon - that's how I do it
06:57.17benjksame here, my PBX still runs a version of Asterisk that is more than 1 year old
06:57.25benjkif it aint broken don;t fix it
06:57.28Dr_Raymy asterisk is from june
06:57.35diamonThough now that I remember, I'll have to hit #mythtv-users and ask about disabling the auto-eat-all-my-crap feature...
06:57.39Dr_Raysince it's off the internet, I'm not super concerned
06:57.51benjkbut I have plenty of other asterisks to play with newer stuff
06:58.27Dr_Raythere are asterisk rpm's if you want, but from cvs is just simple
06:59.31diamonOk, so here's what I wanted to do in general...  I've got an X100p Digium card clone which seems to be doing well enough so far.  I want * to answer my POTS, but to do mixed VoIP and POTS answering.  I'd think that's a common setup, but my thoughts and reality often don't mix well.  :)
06:59.55benjkthere is no such thing as a X100P Digium card clone
07:00.07diamonDr: The ones I saw were 1.0 of *, but I'll keep looking for a 1.2.  Seems to be a lot of features and functionality in 1.2 that I'd like to play with eventually.
07:00.12IronHelixyes there is, the x100 is just a rebadged intel voice modem
07:00.30IronHelixif you buy the same kind of voice modem with the right chipset, its a 'x100 clone'
07:00.31benjkDiogium's own cards were rebadged Intel modems
07:00.36diamonIronHelix: Sure of that?  It's an intel modem?
07:00.43Dr_Rayit's not a clone
07:00.46IronHelixyeah, the x100 is
07:00.48mog_homehey now
07:00.55mog_homewe made a small change ^_^
07:00.58IronHelixdiamon- if you do that, i recommend connecting your other pots phones to asterisk too
07:01.06mog_homeand we never claimed they were anything else
07:01.06diamonBwahahaaaa!  I'd been trying for years to get the command set for voice modems to do more-or-less what this card will now do.  Irony...
07:01.15IronHelixhehehe
07:01.17benjkso if you say that some card is a clone of Digium's card than that implies that Digium's card is an original, which it is not
07:01.24IronHelixno definately not bumping your products Mog <3
07:01.30mog_homewell
07:01.33mog_homebenjk
07:01.36mog_homedigium wrote the driver
07:01.46mog_homethanks iron its all good
07:01.46benjkYes, the driver is original
07:01.51diamonIron: I don't have the card to provide ring (FXS?) for my lines, nor an ATA.
07:01.52benjkbut the card is vanilla
07:02.03mog_homewell driver is 99% of the work in this case
07:02.10mog_homeand in most cases regarding zaptel hw
07:02.12IronHelixdiamon- i would recommend getting that, either an ata or some fxs ports
07:02.16IronHelixit will be much more useful
07:02.17mog_homethe genius is in the code not the card
07:02.23IronHelixvery true
07:02.27benjkmog, nobody disputes that
07:02.28Dr_Raythe genius is asterisk
07:02.28IronHelixthe card is just a dumb interface
07:02.32IronHelixand zaptel
07:02.32Dr_Raynot the zaptel driver
07:02.43diamonIs there any kind of cheap FXS card, or should I just suck it up and get an ATA?
07:02.43mog_homei dont dr_ray
07:02.46IronHelixthe problem is a mixed setup diamon, if you pick up another phone it will put you on the line with *
07:02.47mog_homethere is a lot of code in zaptel
07:02.52mog_homethat gets over looked
07:02.56IronHelixdiamon- depends on your definition of cheap
07:03.05benjkdiamon, get the real thing
07:03.06IronHelixdigium tdm400 series is nice
07:03.10IronHelixpretty easy to setup too
07:03.10diamonI just want to do a proof-of-concept type thing for myself, and maybe a client/friend or two of mine who might get good use out of this...
07:03.12benjkget an IP phone
07:03.14mog_homelike asterisk is one of the few pbxes that can generate and understand rotery
07:03.18benjkanalog sucks balls
07:03.42diamonIron: Those looked nice, but aren't they like $400USD?
07:03.43IronHelixdiamon are you trying to make a really cool answering machine or do something more complicated?
07:03.51IronHelix$400 fully loaded with ports
07:03.52benjkproof of concept use phones: Grandstram BT100 for 50$
07:04.07IronHelixyou can get one that has only 1 port for like $200
07:04.17IronHelixyeah benjk might have a better idea
07:04.17diamonIron: More complicated eventually, for now I want an uber-spiff answering system with some PBX-type features.
07:04.21Dr_Raymy proof of concept was the asterisk dev kit (back when it included a x100 card) and a bugetone
07:04.30benjkyou can get 4 Grandstream IP phones for 200  $
07:04.40IronHelixgrandstream bt100 barbietel phones are cheap and they wrok well
07:04.48IronHelixplus if you forward your ports and configure * correctly
07:04.52IronHelixyou can demo it from your friends house
07:04.55*** join/#asterisk rustyPixel (n=rustyPix@cpe-024-162-252-094.nc.res.rr.com)
07:04.56IronHelixjust plug in the phone and go
07:05.15IronHelixwithout bringing the whole server
07:05.17Dr_RayI'd not marry myself to 4 budgetones
07:05.29benjkHowever, if you must do FXS for something like -say- a cordless base, then you may want to get an IAXy
07:06.08Dr_RayI was happy with the budgetone for testing, but not an everyday phone
07:06.10IronHelixor get that new uniden thingy
07:06.12benjkbut FXS ports are just dumb
07:06.20IronHelixi agree, when possible
07:06.22diamonbenjk: I'll probably just get some kind of ATA and hook it to the whole house, I don't have much inside, but I'd prefer not to tear everything up.  :)
07:06.32IronHelixdiamon- some tearing will be useful
07:06.40IronHelixif you tear enough that each phone gets its own fxs port
07:06.45IronHelixthen you can dial from one phone to another
07:06.45benjkunless you have a client who is stubbornly braindead and demands analog phones
07:06.49diamonIron: I'll preserve tearing up for when I try for IVR.  :)
07:07.09benjkor a client with thousands of analog phones that havent been written off yet, don't use FXS
07:07.10IronHelixbenjk- or has no cat5
07:07.21IronHelixfor a small one at least
07:07.42IronHelixid much rather install pots channels than get out the drill and run cat5 through walls
07:08.04benjkIron: that no cat5 thing falls under stubbornly braindead client
07:08.05diamonbenjk: I choose my clients carefully.  If I suggest it, they know to get exactly that or I don't support it at all, and whatever I suggest that they get, I support as best I can at no charge for simple stuff.  My usual rates cover a bit of support, and most of them are bright folks.
07:08.08Dr_Raywe've been very happy with channel banks and zap
07:08.48benjkdiamon: then do yourself and your clients a big favour and get IP phones
07:09.00IronHelixagreed
07:09.03benjkthere is plenty of decent IP phones to choose from
07:09.06diamonThe main client I was thinking of had hell trying to get a Cisco setup in and working, but best I could tell it was the moron (read: nepotism!) that the CFO hired, some friend of his or something.
07:09.07IronHelixdiamon- at the very least get ONE barbietel and play with it
07:09.29mog_homespas are great too
07:09.31diamonIron: You'd suggest a Barbitel over an ATA?
07:09.32mog_homeno barbietel
07:09.50IronHelixdepends on the use.  to play with and learn i'd suggest a barbietel any day of the week
07:10.04diamonOk, I'll consider that.
07:10.04IronHelixif you need to be using it constantly AND you have a good pots phone, then get the ata
07:10.04benjkdiamon: if they want Cisco phones (cause they impress visitors) then give them Cisco phones, Asterisk can handle those
07:10.14Dr_Raywe drive 7960's
07:10.19Dr_Rayvia asterisk
07:10.21mog_homeother than a cisco or polycom, id rather have something like a spa or iaxy hooked into my 900mhz nice phone
07:10.22IronHelixyeah from what i hear asterisk has made a bunch of progress with chan_sccp
07:10.26mog_homeonly thing that sucks is cid
07:10.35mog_homebut i get that from jabber these days anyways
07:10.44diamonbenjk: Yeah, I told them to get the unlock access, so we're good there, AFAICT the guy is a buffoon who has nearly no idea what he's doing.  Even the wiring was screwy...
07:11.16benjkOh BTW, did you see that the Grandstream GXP-2000 has PoE ???
07:11.23benjkI was baffled by that
07:11.43mog_homeisnt that phone a legitament one
07:11.49mog_homenot like the barbies
07:11.52IronHelixhehe i know the type.  i once came on a job where some kid was trying to wire stuff up, he was stripping the cat5 wire before crimping it (i mean like stripping the individual wires).  his first Q to me was why the switch was broken (slow steady flash on all lights = bad cable)
07:11.56benjklegitament?
07:12.01IronHelixyeah the gxp is pretty good
07:12.05IronHelixi use one on my desk
07:12.10IronHelixwith the new firmware its actually pretty cool
07:12.12mog_homeGood
07:12.14mog_homei thought those were quality ones
07:12.19mog_homeunlike the barbie crap
07:12.28IronHelixin another two months or so it will be awesome once they get some more bugs/features in
07:12.28diamonThey laid down a second network just to be totally sure of available bandwidth (they do have brutal usage spikes at times), so that was sensible, but that moron wired it odd, and cross-linked stuff, and didn't set up anything...  Very odd.
07:12.41diamonIron: Arrgh, I hade that kind of stuff...
07:13.22mog_homeman any of yall watch late night tv
07:13.37IronHelixhell no
07:13.39IronHelixget a tivo
07:13.39mog_homethey have the wierdest stuff on
07:13.47mog_homeheh
07:13.49mog_homei dont get cable
07:13.55benjkmog, our late night tv is your early bird morning program
07:13.58mog_homei just was bored this week
07:14.04mog_homeso i flipped it on
07:14.07mog_homeheh
07:14.11IronHelixyeah, once i explained to him that having the wires touch each other inside the thingy is bad, hes like OHHHHHH that explains it, goes and takes a huge scissors and cuts off the last ~50 cable ends that he did
07:14.13mog_homethis guy is selling knives
07:14.24mog_homeand he is acting like it is the most amazing thing ever
07:14.30IronHelixcutco?
07:14.35mog_homeoh no
07:14.40mog_homethis guy is a dealer
07:14.41diamonIron: Though that reminds me...  Once I had a call from a client who had a LED they didn't recognize turning off in flickers on their network switches.  I went up to take a look.  It wasn't a switch, it was an 8-hub stack, daisy-chained together.  The light that was turning off?  COL.
07:14.48mog_homeand he sells all different kinds of knives
07:14.54mog_homei could buy 200 knives
07:15.00mog_homeat 99cents a knife
07:15.03benjkthe guy who messed up the wires is selling knoves on tv?
07:15.05mog_homeand that includes a sword
07:15.35mog_homehrm?
07:15.42mog_homeoh noes
07:15.48mog_homeonly 1 minute to buy the set
07:15.50diamonI was like, "Uh, that's only supposed to flicker on."  They got worried when it was flickering *off*!  I split it up with an 8-port 10/100 switch attached to the rack, one hub per switch.  Wow did that fix a lot of slowness...
07:16.04IronHelixhahahahahahaha
07:16.47benjkmog, in Taiwan, they make famous knives of the steel from Chinese mortar shells that the Chinese fired at them over the years from the mainland across Taiwan straits
07:17.15benjkand they then export those knives to China, they are a big hit in Chine
07:17.19diamonI countead heads after that (new client) and found I had a collision domain of over 300 systems, including all the servers.  Nothing was switched.  I had a lot of work to do for that place for sure.  Good money and they were thrilled with the results, so I was happy.
07:17.28mog_homeheh
07:17.33mog_homeI just dont get it
07:17.36benjkthey even come with a certificate that they are made from steel from China
07:17.38mog_homewho buys knives off tv
07:17.51diamonbenjk: Wow, that's funny!  I'd call it IRONic, but hey...
07:17.51mog_homethis show is on every night 5 nights a week
07:17.53mog_homefor an hour
07:17.55benjkmog, people who are bored
07:18.06mog_homehow can you have 5 hours a week of knives
07:18.20diamonmog: There's a lot of bored people out there...
07:18.25mog_homeyeah
07:18.28mog_homei mean im watching it
07:18.35mog_homebut its a  far cry from buying
07:18.37benjkshame on you
07:19.00benjkwait another two weeks and you'll start buying ;-)
07:19.05diamonIron: So the uber-answering-rig I was asking about is no big deal then overall?
07:19.15mog_homenah i only am watching it as i am frustrated with my code
07:19.20*** join/#asterisk tainted_ (n=identd@adsl-71-129-45-84.dsl.irvnca.pacbell.net)
07:19.24mog_homeand i have given up for the night
07:19.26IronHelixyeah it shouldnt be that bad just make some architecture decisions before you do anything
07:19.29diamonmog: Errr, if you are watching a knife infomercial, do your brain a favor and turn it off, then go get a book...
07:19.34benjkmog, that's how it always starts
07:19.41mog_homeheh i have many a book
07:19.42IronHelixwhat you can always do to start out at least is wire your whole house to a single FXS port or ATA
07:19.53benjkand it ends with buying useless stuff off the TV
07:19.57mog_homei am reading specs right now
07:19.58IronHelixthen split off extensions later or run cat5 cable for IP phones if you decide to
07:20.05IronHelixthus making the decision that you arent going to decide
07:20.05mog_homei will never buy stuff off tv
07:20.07mog_homealthough
07:20.14mog_homemy woman has the ronco oven thing
07:20.27IronHelixi do highly recommend however that you pick up a grandstream budgetone 101 or grandstream gxp2000 and play with it
07:20.32benjkno, if you wire your houise, start with cat5
07:20.35IronHelixwww.voipsupply.com i've had good luck with them
07:20.44diamonIron: That's exactly what I was thinking to do.  Moreso because my phone lines are badly damaged in half the house, and it'll be a PITA to fix, so I'll likely just run cat5 pairs to each outlet, do a network and phone line at once.
07:20.49IronHelixif you RUN wires then run 5x cat5 everywhere
07:20.53benjkIron: how fast do they deliver?
07:21.05IronHelixi never get rush shipping so 3-5 days
07:21.09benjkI was looking at them this week
07:21.10diamon5 strands?!
07:21.25IronHelixthe more the better
07:21.30IronHelixrunning ANYTHIN Is a PITA
07:21.33IronHelixso if you run 5 at once
07:21.35IronHelixthen you're set
07:21.40benjkA customer of mine ordered some ATAs and stuff from Voxilla and it took two months
07:21.43IronHelixleave 4 of them hanging there maybe
07:21.43benjkscary
07:21.49IronHelixwow
07:21.51IronHelixthats horrible
07:22.05IronHelixtry voipsupply, they're reputable, if it makes any difference bandwidth.com uses them as a supplier
07:22.09mog_homei bought some iaxys from digium
07:22.12mog_hometook a week
07:22.16IronHelixlol
07:22.19mog_homeand then i just walked down the hall
07:22.22mog_homeand grabbed em
07:22.26benjkhe called Voxilla every day and therew was always only answering mode
07:22.27mog_home^_^
07:22.32diamonIron: A triple is my max, 5 would be huge, and with 5 outlets in the house to rewire, my main point would have 25 cat5 sets coming to it!
07:22.32IronHelixhehehe
07:22.39benjka week is ok
07:22.43benjkbut 2 months
07:22.52IronHelixdiamon- it doesnt have to be 5 but it should def. be more than one
07:23.05IronHelixrun good cable too, cat6 if you can afford it
07:23.08IronHelixso upgrading is easy
07:23.28IronHelixor better
07:23.32IronHelixthis is HARD
07:23.36IronHelixbut if you run conduit
07:23.41IronHelixall your problems will be solved forever
07:23.48diamonYeah, a pair is my usual, or a triple.  Can't do cat6 yet.  First, it's not a spec; second, the folks who claim to be cat6 are HUGELY expensive per foot.  I'll just use my cat5e.  It'll do.
07:24.13diamonIron: I was toying with doing that, but I think the house's design will prevent me from being able to do a proper conduit.
07:24.13IronHelixi thought it had been nailed down
07:24.24mog_homelol
07:24.36diamonIron: If it was, it's VERY recent, I looked into that just a month ago or so.
07:24.36mog_homethey say if the knife isnt up to the right level of density
07:24.46mog_homeforget name of scale
07:24.50mog_homebut if it doesnt meet it
07:24.53mog_homeyou can send it back
07:25.00diamonIron: Most folks are using the base pre-spec that they think will become the standard.  Might, might not.
07:25.08IronHelixdiamon- yeah, then triple cat5e should be fine then
07:25.11diamonmog: Moh's hardness?
07:25.17mog_homeyeah
07:25.28IronHelixhow the hell are you going to test that
07:25.32diamonWhat rating are they saying?
07:25.39IronHelixi highly doubt that most people happen to have a materials tester laying around  :\
07:25.40mog_homethey didnt say
07:25.45mog_homeits just guaranteed
07:25.46diamonIron: With items higher and lower on the scale, silly! :)
07:26.00mog_homethey sell like 80 knives
07:26.07mog_homeso some are harder than others
07:26.17mog_homebut if its not hard enough they will rma it for ya
07:26.19IronHelixlol
07:26.31mog_homei wonder how do you test it
07:26.35mog_homedo you crush it?
07:26.53mog_homeor put it in water and condense water pressure?
07:27.04mog_homeerr increase not condense
07:27.59diamonNah, scratch it.  Moh's is a hardness scale.  Higher hardness will hold a tighter edge, but is more likely to break.
07:28.22mog_homeoh so go get some constant and try to scratch it
07:28.34mog_homelol, they just went to commercial
07:28.44diamonSo if they claim it should be a 7.2 Moh's metal, you can get something a bit higher to test it, and a bit lower to see if it fails to scratch it...
07:28.46mog_homehow can you sell ads during an ad
07:28.52mog_homeyeah
07:28.55mog_homethat makes sense
07:29.01diamonLike, diamonds are a 10, and talc is a 1 or something...
07:29.15mog_homeyeah i remember 6th grade now
07:30.07*** join/#asterisk L|NUX (i=linux@202.5.131.28)
07:32.12*** join/#asterisk jayk- (i=jayk@lasziv.reprehensible.net)
07:32.39diamonMy poor computer...  It's showing ECC errors in the IDE SMART info.  That's not good...
07:32.58mog_homeeep
07:33.19IronHelixeek
07:33.25IronHelixhope you got backups :)
07:33.34jayk-might just be your ide controller
07:35.29diamonI think it's my cable.  Stupid SATA rev1 cables don't have clips to keep them in place, always wiggling loose.
07:35.50IronHelixblah
07:35.56diamonNo problems yet, I was just running SpinRite to check the drive.
07:36.07diamonI love SpinRite...  It's my friend.
07:37.57diamonSo, assuming I get an ATA for my phone system so I can keep my current cordless, is there any simple way to pass a code to specify usage of VoIP or POTS for dialling outbound?  Like, if I just dial it uses VoIP, and if I do *9 then dial it's VoIP?
07:38.36Dr_Rayyes, in your extensions.conf
07:38.43mog_homeokay ive decided
07:38.46mog_homeenough with voip
07:38.51benjkdiamon: get an IAXy
07:38.51diamonDr: I'll read up on that then.
07:38.52mog_homeim going into the knife business
07:39.03h3xmog, kyocera? hehe
07:39.12mog_homeahhh
07:39.13BladeRunner05IronHelix: Hola
07:39.19IronHelixblade!
07:39.22IronHelixhows it goin
07:39.24Dr_Rayget a cnc machine
07:39.34diamonbenjk: That was the leading one so far in my list, though codec support seems a bit limited...  Can that be extended?
07:39.39mog_homewho has a cnc machine
07:39.42IronHelixdiamon- yeah you can do that
07:39.47benjkDr_Ray: a CNC machine for the cordless? how would that work?
07:39.48mog_homeand how can i borrow it for an hour.....
07:40.00Dr_Rayget a cnc machine for the mogknifefacotry
07:40.06diamonLoL.
07:40.09benjkdiamon: no it cannot
07:40.12mog_homeooh thats a great idea
07:40.25Dr_Rayemachineshop.com
07:40.32benjkdiamon: Atcomm makes IAX ATA's with more codecs
07:40.33mog_homewhy resell someone else's knives when i can make my own
07:40.46Dr_Raybut if you are going to make more than 1 then a cnc machine would pay for itself
07:40.54mog_homeyeah
07:41.01mog_homehow much is a cnc machine
07:41.05diamonbenjk: Foo.  Thought so, but I was hoping.  Can I have the * system change codecs for it?
07:41.06mog_homei know its massively expensive
07:41.06IronHelix$10k
07:41.10Dr_Ray~1000 or so on ebay
07:41.12mog_homeyikes
07:41.13diamonmog: If you have to ask, you can't afford one.
07:41.16mog_homelol
07:41.23mog_homejust cheap
07:41.30Dr_Rayemachineshop.com is an online one
07:41.32mog_homewell not really high
07:41.34Dr_Raywiht cad software
07:41.43Dr_Rayer, cadlike software
07:41.44benjkdiamon: sure
07:41.46mog_homeooh
07:41.58mog_homei saw a project like asterisk for cnc stuff i want to try
07:42.03Dr_Rayemachineshop is big fun
07:42.04IronHelixanyway
07:42.05IronHelixim out
07:42.07IronHelixnite all
07:42.10diamonYeah, emachineshop is nifty.  I was toying with having them carve me a few GPU heatsink water blocks, but someone finally made a non-crappy one.
07:42.11mog_homei dont know what i would make with a machine like that
07:42.15IronHelixdiamon- good luck wiring your pad
07:42.21mog_homei think i would just make cases for my toys
07:42.37diamonIron: Yeah, it's gonna be fun...
07:42.53Dr_RayI've yet to pay for any parts to be made, but I've enjoyed making stuff with the software and then pricing it
07:43.51diamonHeh, good thing it's so automated!  I did have a client who wanted to figure out how to get a custom-made crankshaft for an old-ass motorcycle, I pointed him there.  Worked out nice.
07:44.10Dr_Rayyeah, I figured out how to make some custom payphone parts
07:44.13Dr_Rayit was nifty
07:44.18mog_homecool
07:44.44mog_homei want just want to make cool cases for my projects
07:45.07mog_homehow long would it take a cnc machine to make something as complicated as like a linksys router case
07:45.21Dr_Rayin plastic or in metal?
07:45.43diamonThat's one thing my customers always love.  If they're willing to pay my rates, I'm willing to do most anything.  Had a guy pay me an hour a day to drive to his place and spend 30 mins feeding and playing with his dogs when he had to leave town on no notice.  He paid for the 15-min drive and my 30 mins there, so 1hr/day, and he was happy.  Nice dogs, too.
07:46.41mog_homeplastic
07:46.53diamonmog: For plastic stuff, you might be better served with some kind of vacuum molding or injection system, especially if you want to do small sets of more than one.
07:47.12mog_homewell what about a pc case
07:47.13Dr_Rayemachineshop will do vacume and plastic injection
07:47.24Dr_Raythey have a bunch of tools
07:47.28mog_homehow hard is something like that out of curiosity
07:47.37mog_homei just wonder how fast i could make like 1000 units
07:47.44diamonmog: Heck, with a metal roller and folding sheetstock set you can make one cheap.
07:47.52jayk-we have a cnc machine
07:47.58mog_homeooh your lucky
07:48.00jayk-3 & 5 axis
07:48.01Dr_Raywith injection molding (you need molds) and with vaccum you need a blank to form it on
07:48.16Dr_Rayonce that is created it's a snap
07:48.20jayk-we specialize in vacuum molding too
07:48.25mog_homeso like a few minutes
07:48.28mog_homeor seconds
07:48.44jayk-we are a plastics, tooling, and composites manufacturer
07:48.48diamonmog: It'd probably need more time to cool enough to remove than anything, once the forms exist.
07:48.52Dr_Rayemachine shop is good for 1's and 2's, if you wanted 10k or whatever you'd be better off going overseas
07:49.06mog_homejayk can you make me something ^_^
07:49.10jayk-sure
07:49.15mog_homereally?
07:49.17jayk-yeah
07:49.21mog_homei want a case for my soekris
07:49.24mog_homeand my x100p
07:49.38mog_homethat looks like a linksys router case basically
07:49.49jayk-how many do you want
07:49.53mog_home1....
07:49.59jayk-heh
07:50.02jayk-probably not worth it
07:50.05mog_homeyeah
07:50.10mog_homenot worth it to you
07:50.10diamonmog: Uniques are always wildly pricey.
07:50.12jayk-nope
07:50.13mog_home^_^
07:50.16Dr_Rayemachineshop
07:50.19mog_homeyeah i figured as much
07:50.20Dr_Raycheck em out mog
07:50.21jayk-maybe if you wanted 500
07:50.22jayk-:)
07:50.41mog_homei need to go round up 500 friends
07:50.43*** join/#asterisk mazzanet (n=irc@unaffiliated/mazzanet)
07:50.45mog_homehrm only 300 in channel
07:50.50mazzanetok
07:50.55mazzanetcrazy question of the day
07:51.04jayk-heh
07:51.04jayk-well
07:51.06jayk-maybe 50
07:51.06diamonAnd one less; I'd best go ask my Mythtv questions before it gets too much later...
07:51.08mog_homehow much is it per unit at 500?
07:51.12jayk-beats me
07:51.16Dr_RayI priced making brass payphone keys, and 1 was $200 and 200 was $310
07:51.30mog_homeokies
07:51.30Dr_Raythe first one is the hardest
07:51.31mazzaneti need an app say like hyperterminal, that can work over SIP
07:51.46mog_homeumm
07:51.46diamonDr: Exactly.
07:52.10mog_homehmm
07:52.11diamonMazza: You just sprained my mind...  That's like wanting to do VoIP over SMTP or something...
07:52.16mog_homeyou could do a modem connection over sip
07:52.18mog_homein ulaw
07:52.22mazzanetthats what i meant
07:52.24mazzanetmodem connection
07:52.24Dr_Rayemachineshop is great for prototyping.. then you send them off to jayk after the design is done
07:52.29mog_homebut it will work rarely at best
07:52.39mog_homewell my problem is its just for me kinda stuff
07:52.48mog_homeso at most i would need 10-25 for friends and family
07:52.49Dr_Raythen emachineshop is the way to go
07:53.01mog_homeso it would never make sense for me to get it really done
07:53.05mog_homei just dream
07:53.13Dr_Rayor you could learn to do it yourself
07:53.23Dr_Rayvacumm forming is not difficult
07:53.23mog_homeyeah
07:53.24diamonDr: Oooh, hand-carved!
07:53.30mog_homeheh
07:53.31mazzanetie. the software equivalent of going hyperterminal -> 56k modem -> ATA -> asterisk -> pstn
07:53.35mog_homei did a wood case once
07:53.39diamonmog: Make it from wood!  Give it a niiiice finish.  :)
07:53.42mog_homedont do it mazz
07:53.45mazzaneti just want to go hyperterminal or similar -> asterisk -> pstn
07:53.48mog_homeit over heated
07:53.52mog_homeand wood warped over time
07:54.15mog_homei mostly leave stuff open or in tuperware containers
07:54.18jayk-i got my keyboard wet one time
07:54.22jayk-and set it up on the heat register
07:54.23mog_homeas you can get em in most shapes and sizes
07:54.26benjkmog: is the tdm2400 a 6port card only?
07:54.31jayk-after about 12 hours, it had melted
07:54.34mog_homeno
07:54.34jayk-but very very slowly
07:54.35mog_home24
07:54.36jayk-it looked really weird
07:54.39mog_home6 modules
07:54.40diamonmog: More cooling, and a better-cured wood then...  I did a 50W car amp cover in wood, with two fan ports to keep the MOSFETs cool.
07:54.44*** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net)
07:54.47mog_home4 ports per module
07:54.58mog_homewell i didnt have fans
07:54.59Dr_Raydoes digium have new toys?
07:55.01mog_homethat was the problem
07:55.04shido6yes
07:55.04mog_hometdm2400p
07:55.06shido6check the site
07:55.09diamonjayk: Press the Escher key to continue!
07:55.35benjkmog: you may want to talk to whover is in charge of making the content for Digium;'s website, the tdm2400 page is not really clear on this
07:55.38diamon(Pun on melting-flowy keyboard image I have in my mind)
07:55.42Dr_Ray!
07:55.57mazzanetmog_home: i can use G.729 by the way
07:55.59mog_homehrmm i will pass word along
07:56.00jayk-diamon :)
07:56.02jayk-i should have taken a picture
07:56.03mog_homewhat?
07:56.30diamonjayk: I had a fun one trying to remove a keyboard's melted remains out of a fancy dishwasher.  It went....poorly.
07:56.40mog_homeheh
07:56.44Dr_Raydoes this have IRQ problems?
07:56.44benjkso these are different modules than the ones for the tdm400 then, I see
07:56.52mog_homei have only really messed up one keyboard
07:56.54mog_homeyes
07:56.59mog_homewell slightly
07:57.07mog_homethe fxs modules are more different
07:57.15diamonA customer decided my cleaning method was too slow and dainty, so they just tossed it in the dishwasher.  Heated dry did it in I think, though the sanitizing heat-wash did it no good I'm sure.
07:57.16mog_homethe fxo is very similar
07:57.22mog_homejust more on one board
07:57.26Dr_Raybut you can put multiple of these cards in a box?
07:57.29jayk-did it ruin the dishwasher?
07:57.31mog_homeyes
07:57.36mog_homewe ran 4 in a machine for grins
07:57.38benjkwhat's that strange looging connector sticking out at the back of the card?
07:57.45mog_homedigium wont ever reccomend 2-3 cards in a machine
07:57.53diamonjayk: Yeah, turned out it had jammed the pump impellers.  I wish I could have seen it in action.
07:57.57mog_homethat is an amphenol (sp)
07:58.00mog_homeconnector
07:58.00Dr_Raybenjk tel;co punchdown
07:58.07mog_homeyou plug a punchdown block
07:58.07benjkI can recommend up to 6 cards in a single machine
07:58.11mog_homeor breakout box
07:58.19mog_homeheh we do it in testing lab
07:58.21mog_homeand it works
07:58.25benjkworks flawlessly, no problems
07:58.27jayk-the mesh screening in mine had a gaping hole
07:58.30mog_homebut you wont catch us reccomending it
07:58.33benjkbut it has to be a Mac
07:58.36jayk-it was sickening all the junk that passed through it and clogged up my drain lines
07:58.38benjka vintage Mac
07:58.41benjk9600
07:58.41mog_homei have seen 5 quad cards in a machine
07:58.44mog_homeand seen it just work
07:58.45diamonIf you're going to have that many cards, wouldn't you want to just give up and get a DS/T1 card and a channel bank for it and be done?
07:58.49*** join/#asterisk grimse (n=grimse@p5481CCBF.dip.t-dialin.net)
07:58.50mog_homebut i wouldnt tell peopl to do it
07:58.58Dr_Raywhat's the price of this monster?
07:59.04mog_homewell it is nice to just have one box diamon
07:59.06benjkmog: no problem on the Mac architecture
07:59.16mog_homeyeah
07:59.21diamonjayk: I won't even ask, I was toying with food and want to keep my appetite.
07:59.22mog_homeppc arch is better at irqs
07:59.22benjkunfortunately these days Macs only have 3 PCI slots
07:59.24Dr_Rayif you need to drive fxo lines, then a channel bank stinks
07:59.29mog_homeyup
07:59.35mog_homeyeah that too
07:59.40Dr_Raystinks. well, less than ideal
07:59.43mog_homeits cheaper than channel bank, at least rhinos
07:59.56diamonWouldn't it be better to split the cards between systems to have a bit of physical redundancy anyway?
07:59.58mog_homewell fxo lines arent fun
08:00.01mog_homevia la pri
08:00.04mog_homeYES
08:00.07Dr_Rayso $1000 fully populated?
08:00.13Dr_Rayish?
08:00.15mog_homeermm no
08:00.55mog_homeI think its on our site and voip-supply
08:00.55mog_homeas well as our resellers
08:00.55mog_homei dont deal with the moneys though sorry
08:00.55Dr_RayI don't see it at the asterisk store
08:01.01mog_home:(
08:01.04mog_homesomeone must die
08:01.23benjkso does this card also come with a converter to RJ11?
08:01.39benjkor do you have to make your own adapter cable?
08:01.41Dr_Raybenjk - you use the punch down block
08:01.42mog_homeno not from digium, but we are going to offer breakout boxes
08:01.54mog_homei imagine some people will sell it with it
08:01.58mog_homeits a standard interface
08:02.07Dr_Raygreybar sells them for $20
08:02.09benjkI don't know what a punch down block is, but anyway, does it come with the card?
08:02.23mog_homeyeah greybar rocks
08:02.31mog_homedepends who you get it from
08:02.32diamonAgreed.
08:02.34benjkthats very bad news then
08:02.40mog_homedigium will sell breakout boxes as well
08:02.43Dr_Raygreybar is great to walk in and say I need this, and have them
08:02.51benjkit will increase the cost of the card here in Japan by at least 100 USD if not 150
08:03.03mog_homeeep benjk
08:03.14mog_homeyou have to be able to get breakout boxes in japan
08:03.32benjkyes because if you havbe to order the cable somewhere else then iot is additional FedEx, customs and import duties
08:03.35mog_homeits an age old standard
08:03.40Dr_Rayok, that's awesome
08:03.48mog_home?
08:03.52Dr_Raythis new card
08:03.57benjknothing telco related in Japan is compatible with anywhere else
08:03.59mog_homeyeah its pretty cool
08:04.03Dr_Rayin a 1U case, that's studly
08:04.12mog_homewe rolled some of digium over to it a while back
08:04.15mog_homeit works great
08:04.26mog_homeim sorry ben
08:04.26Dr_RayI'm going to keep using adit 600's but that will have it's uses
08:04.27*** join/#asterisk Joe2000x (n=jo@202.155.89.122)
08:04.45Joe2000xhi
08:04.47Dr_Rayesp, the abilty to reconfigure on the fly
08:04.57Joe2000xis anyone ever use asterisk realtime ?
08:05.01mog_homemake sure you get it bundled or order with your card, they usually arent that expensive
08:05.02mog_homeyup
08:05.22benjkmog, you better get your sales/marketing gurus to ship that card with an adapter cable for overseas customers
08:05.24Joe2000xwhy i dont have the realtime command in my CLI?
08:05.32Dr_RayI was hoping you guys were going to make a 24 port iaxy type device, but this is good enough
08:05.58Dr_RayI don't get the impression that mog is running the joint over there
08:06.01Dr_Ray:)
08:06.05benjkAh, so Digium *will* seel it bundled with some RJ11 solution
08:06.10mog_homelol
08:06.22Dr_Raybenjk - I'm sure if you axed them, they would set you up
08:06.23mog_homewe are gonna have breakout boxes available ben
08:06.28mog_homejust buy one with it
08:06.46mog_homeand i dont run the joint, but we are a family opperation
08:06.47Dr_Raybut that is studly
08:06.58jayk-what do you run, mog
08:07.11mog_homeaway
08:07.12mog_home^_^
08:07.14benjkmog makes the coffee
08:07.19mog_homeheh
08:07.22benjkhence the name mug
08:07.22mog_homewe dont drink coffee
08:07.26mog_homejust redbull
08:07.26Dr_Raymaking the copies
08:07.31benjkah no that s the wrong spelling
08:07.46mog_homei work in dev
08:08.02diamonUgh, Red Bull makes my heart skip after just a few...
08:08.10Dr_Rayso the echo cancelation is one per card?
08:08.15benjkI work in the factory
08:08.19benjkOn the chain?
08:08.29benjkNo, we are allowd to walk around freely
08:08.53*** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net)
08:08.54mog_homeyes dr_ray it has an optional echo can
08:08.57*** part/#asterisk diamon (n=diamon@c-66-176-91-189.hsd1.fl.comcast.net)
08:09.07mog_homelike the echo can for te405/410
08:10.43newmemberI cant seem to dial out on my FX0 interface,   where is a good place to watch the call process for debugging?
08:10.53Dr_Rayasterisk -r
08:10.53mog_homeeasiest way
08:10.56mog_homeis zap barge
08:10.56Dr_Rayset verbose 10
08:11.05mog_homeif you actually want to hear the audio
08:11.08mog_homeotherwise do that
08:11.48newmemberI have my phones registering and they can call each other, but my 'dial 9' does nt pick up a line to call out
08:11.50benjkmaybe its because its an FX0 card
08:12.29benjkthe dialing goes zero
08:12.46benjkif it was an FXO card on the other hand ...
08:12.46newmemberFX0 goes to the PSTN?
08:12.59newmemberright
08:13.02newmemberFXO
08:13.07benjkFXO does, but FX0 I am not so sure ;-)
08:13.14newmemberright
08:13.48newmemberSo it doesnt pick up the FXO port
08:13.59newmemberwhen I dial nine and a number
08:14.26benjkwhat does it say when you do zap show channels
08:14.34benjkdoes it show up?
08:15.04newmemberpseudo            from-internal   en
08:15.04newmember<PROTECTED>
08:15.04newmember<PROTECTED>
08:15.04newmember<PROTECTED>
08:15.04newmember<PROTECTED>
08:15.28newmemberI think thats a goos sign
08:15.33newmembergood sign
08:16.57mog_homewhat does your dial statement look like
08:17.38shido6amp?
08:17.46newmembereither
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08:20.43Dr_Raythanks mog
08:21.35*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
08:22.08mog_homeno problema
08:24.34newmemberI see this when I call out
08:24.36Dr_Rayso is the echo cancelation for the tdm2400 an add on module?
08:24.36newmember<PROTECTED>
08:24.36newmember<PROTECTED>
08:24.36newmember<PROTECTED>
08:24.52newmemberit looks liek it going to the zap trunk
08:25.40newmemberI think my FXO is port 4, I think it should look like this   -- Zap/1-4
08:25.55viraah
08:25.56viraer
08:25.58Dr_RayZap/4-1
08:26.42newmemberwhich is easier to change, the fxo on the card or the zaptel.conf?
08:26.56Dr_Rayzaptel.conf
08:27.08Dr_RayIMO
08:27.45*** join/#asterisk P4C0 (i=1000@201.224.107.47)
08:27.47P4C0hello guys
08:28.00mog_homehi
08:28.12mazzanethow can i put a call on hold?
08:28.42mog_homeflash
08:28.44P4C0does anyones knows a good softphone client that works in linux KDE with artsd?
08:28.47newmemberIntersting, when I ran compiled the zapel I thought it look correct, it looks like it has the fxo and fxs mixed up
08:28.49newmemberfxoks=1
08:28.49newmemberfxoks=2
08:28.49newmemberfxoks=3
08:28.49newmemberfxsks=4
08:28.49mog_homekphone
08:29.06Dr_Raythey are reversed in zaptel.conf
08:29.11mog_homeand in zapata
08:29.50newmemberrevesred on purpose?
08:29.54mog_homeyes
08:30.01mog_homefxo modules speak fxs signalling
08:30.04benjkso mog, what do you work on? asterisk, libpri or drivers?
08:30.06mog_homeand fxs modules speak fxo signalling
08:30.07Dr_Rayreversed by perspective of where you are in the pbx
08:30.27newmemberty
08:30.27mog_homeits intentionally reversed to match standard, not to confuse you
08:30.31Dr_Raythe fact that it is confusing is free
08:30.34mog_homei do internal things at digium, and i work on asterisk code
08:30.39mog_homenice side effect
08:30.42newmembergotta like oss
08:31.20newmemberin the real world you call that the 'network' side of the voice link
08:31.24Dr_Rayextensions you are looking in to the asterisk server, zapata and zaptel, you are looking out, so it will be reversed
08:31.34mog_homeno newmember
08:31.49mog_homei mean its not userfriendly
08:31.56benjkDr_Ray, it's not "reversed"
08:31.59mog_homebut its friendly to anyone that knew telephony pre asterisk
08:32.13newmemberSo I am looking to change from Zap/1-1   to Zap/4-1
08:32.31benjkand FXO *interface* has the purpose to connect to an exchange office, hence the name
08:33.05Dr_Rayand it uses fxs signalling
08:33.19benjkthe fxo interface has to speak FXS signaling to the office because the office only talks to stations
08:33.36benjkits perfectly logic
08:33.44benjklogical
08:33.52Dr_Raymog - is the echo canceling for the tdm2400 built in or is it a drop in module?
08:33.53mog_homeonce its explained...
08:33.58Dr_Rayonce you get it
08:33.59Dr_Ray:)
08:34.00mog_homeits a drop in module
08:34.08mog_homejust like 410 and 411
08:34.15mog_homeyou could upgrade it if you needed it later
08:34.16Dr_Rayso I can buy it later if needbe
08:34.21Dr_Rayperfect
08:34.21mog_homeyup
08:34.31mog_homethe software echo can rocks in asterisk
08:34.47Dr_RayI've been blessed with damn near no echo
08:34.51mog_homepersonally i think echo can stuff is only needed in serious cases, and in places where you need cpu
08:35.01mog_homebut analog lines are more prone to echo than t1s
08:35.11mazzanetbut say i have a phone that doesn't have a hold button and it's plugged into an ATA
08:35.15mazzanetcan i put people on hold?
08:35.22mog_homeif you flash the line
08:35.25Dr_Rayyes, via flashookl
08:35.30mog_homei bet the ata will put that caller on hold
08:35.34benjkmog: in Japan there is always trouble with echo
08:35.41benjkon analog lines
08:35.51mog_homeman ben if japan is that bad, id leave ^_^
08:35.58mog_homeyeah its that way in the south too
08:35.58benjkin some cases it was so bad that we had to back out of the deal
08:36.05mog_homeyikes
08:36.13mazzanetflash the line?
08:36.15benjkmog: I'll consider it
08:36.19benjk;-)
08:36.21mog_homego on and off hook real fast
08:36.23mog_homeheh
08:36.37mog_homeusually there is a button that says flash
08:36.40mog_homethat will do that for you
08:36.44benjkyeah, that's a moron scheme of a caller ID system isn't it?
08:37.11mog_homemeh, i do find it funny how locked down your whole telephony stuff is
08:37.45mazzanetah cool
08:37.46benjkmog it used to be like that pretty much everywhere other than in the US and Finland
08:37.53mazzanetnow how do i add on hold music
08:38.01mog_homeasteirsk will do it for you
08:38.09benjknational monopolies locked everthing down
08:38.13mog_homeyeah
08:38.21newmemberok getting close
08:38.28newmemberI just need it to drop the 9
08:38.30mog_homeyou have to cut the 9 off
08:38.32benjkJapan is just changing its ways only very reluctantly
08:38.38mog_homedo ${EXTEN:1}
08:38.48mog_homeinstead of ${EXTEN} newmeber
08:38.52mog_homethat will trim it
08:39.06mog_homealso you want this www${EXTEN:1}
08:39.19mog_homethat will wait 300 millaseconds so the line waits for dial tone
08:39.46benjkit will dial your extension via the web browser instead of your land line
08:39.47benjk:-)
08:39.55mog_homeheh
08:40.12mog_homeone day ben
08:41.14newmember_9.,1,Macro(dialout-trunk,1,${EXTEN:1})
08:41.25mog_homeyeah that will work
08:41.32newmemberhmmm
08:41.44mog_homei only reccomend the www s as sometimes it takes some time for the line to sieze
08:41.56newmemberrigh t good idea
08:43.03newmember<PROTECTED>
08:43.03newmember<PROTECTED>
08:43.03newmember<PROTECTED>
08:43.12newmemberdidnt drop the 9
08:43.15mog_homedo a noop
08:43.24mog_homei dont think you are doing that macro
08:43.32mog_homeyou are probably doing something else
08:43.49benjkis that AMP?
08:43.56mog_homeyeah it is i feel
08:44.05benjkOh dear
08:44.08mog_homei can feel the amp /A@H
08:44.49benjkprobably some hidden value in the database that sends the dialplan somewhere else
08:44.54P4C0if I want to register into an asterisk server (voip provider) with my asterisk server in which conf file should I put it?
08:44.56mog_homelol
08:45.04mog_homehow are you connected
08:45.06mog_homesip or iax
08:45.09mog_homeor h323
08:45.32P4C0sip
08:45.33newmembersorry just reading the call log
08:45.37mog_homesip.conf
08:45.40mog_homeyeah
08:45.43mog_homedo a noop in there
08:45.44newmemberya I installed aah
08:45.51mog_homeits likely not going through that macro
08:45.56mog_homeor that macro sucks
08:46.01newmemberI anm rading this
08:46.02newmember<PROTECTED>
08:46.06P4C0but isn't sip.conf for the clients?
08:46.10newmemberI am reading that
08:46.12P4C0I mean, local clients
08:46.20mog_homejust do _9.,1,Dial(zap/g1/www${EXTEN:1})
08:46.22mog_homethere you go
08:46.23newmemberI use SIP to my cisco 7960 phones
08:46.25benjkno sip.conf is for SIP
08:46.25mog_homeits for both
08:46.30mog_homeall SIP
08:46.38P4C0mog_home: thanks
08:46.42mazzanetwell when i flash the line, it definately appears i get put on hold
08:46.46P4C0asteriskguru.com will be my friend :p
08:46.52mazzanetbut i get no music
08:47.16mog_homewell then your ata is not relaying the hold
08:47.23mog_homeid need sip debug to know for sure
08:47.49mazzanetnow thats an idea
08:48.07mog_homebut i tend to not debug sip for fun ^_*
08:48.33*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
08:50.27mazzanetwell from sip debug
08:50.31mazzanetwhen i flash the line
08:50.54mazzaneti just get the usual INVITE -> ACK routine
08:51.47*** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at)
08:54.41mazzanethmm
08:54.42Dr_Raythe redphone thing looks neat
08:54.49mazzanettheres an option on my ATA called
08:54.53*** join/#asterisk delox99 (n=delox99@206-248-149-59.dsl.teksavvy.com)
08:54.57mazzanet"Call Hold using c=0.0.0.0 (RFC 2543) in SDP"
08:55.12mog_homeit is a nifty idea dr_ray
08:55.21mog_homei prefer clustering with iax rather than tdmoe
08:55.43Dr_Raywell, I like the magic box aspect of it
08:55.47Dr_Raydrop and walk away
08:56.07Dr_Raybut my asterisk server is not the weak link in the chain
08:56.56mog_homeyeah
08:57.04mog_homewe dont have any hw failover at digium
08:57.26Dr_Raymy asterisk modules don't even autoload
08:57.27Dr_Ray:)
08:57.32Dr_Rayit's just not been a problem
08:57.33mog_homeand we have had more problems with our pri going down then we have had with asterisk
08:57.37mazzanetwell
08:57.50mazzanetwhen i flashhook
08:58.20mazzanetthe receiving phone get some buzz/crackles every now and then
09:00.00mog_homeif you do sip show channels
09:00.05mog_homeor show channels
09:00.10mog_homedoes it say they are on hold
09:00.17mog_homei cant remeber where
09:00.26mog_homebut somewhere asterisk will show you
09:00.34mog_homeits likely you dont have musiconhold setup
09:01.40mazzanet192.168.1.73     ata         jxry0-tth84  00101/00145  ulaw  Yes      Rx: ACK
09:01.48mazzanetthe Yes is for Hold
09:01.53mog_homeyeah
09:02.04mog_homedo you have mpg123-.59-r installed?
09:02.08mazzanetbut the outgoing trunk isn't on hold
09:02.17myke420247moh is funky
09:02.21myke420247you need a timing device
09:02.25myke420247and even then it may not work
09:02.29myke420247it works fine on my grandstreams
09:02.33myke420247but not on x-lite
09:02.36myke420247duno why
09:02.43mog_homeit will always work with timing device
09:02.59benjkhmmm, maybe we should go to Macedonia ...
09:03.00benjkhttp://www.theregister.co.uk/2005/11/24/worlds_biggest_wlan
09:03.06mazzaneti have mpg123
09:03.09mog_homeit just needs timing and that version of mpg123 but not with the newer stuff
09:03.12mog_homedo you have that version
09:03.16mog_homeit has to be exact
09:03.33myke420247it comes bundled with asterisk so you can always build and install that version
09:03.34mog_homeif you go into asterisk
09:03.39mog_homeand type make mpg123
09:03.43mog_homeit will install it for you
09:04.26bbzWould anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. --
09:04.34mog_homebbz DIE
09:04.40mog_homeyou are a bot
09:04.43mazzanetVersion 0.59q (2002/03/23).
09:04.43mog_homeand if not
09:04.46mog_homeyou need to respond
09:04.51mog_homeit wont work with that version
09:05.01mazzanet.....
09:05.03mog_homedo you have a card in the machine
09:05.07mog_homeor ztdummy loaded?
09:05.25mazzaneti'm not using zaptel.
09:05.36mog_homeyou need timing to do musiconhold right
09:05.59mog_homeyou can install ztdummy real easy if its a 2.6 kernel
09:06.26*** join/#asterisk chapeaurouge (n=chap@85.201.80.249)
09:15.19P4C0which is the conf file where I can put logic stuff like if and code?
09:16.32Dr_Rayextensions.conf?
09:16.43Dr_Raydialplan logic?
09:20.56P4C0dialplan logic that one
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09:21.36asterboyhola
09:23.03*** join/#asterisk mmmToop (n=chatzill@196.31.11.194)
09:25.14asterboyAnyone get through to Atacomm.com for orders?
09:30.19*** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com)
09:32.12mazzanetwell i now have on hold music
09:32.21mazzanetbut it sounds really crap.
09:34.26mazzanetthen again
09:35.00mazzanetresampling a 190-320kbps VBR 44.1khz mp3 down to 8khz has gotta hurt
09:35.04*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
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09:39.27mazzanetnow thats just freaking weird
09:39.56mazzanetok i'm going internal phone -> ata -> asterisk -> pstn -> my other phone
09:40.06mazzanetwhen i put [my other phone] on hold
09:40.32mazzanetthe hold music only plays when [my other phone] is 'transmitting' something
09:40.34mazzanetie. making noise
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09:47.27*** part/#asterisk genuix (n=genuix@62-167-18-224.adslpremium.ch)
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09:54.50pooh_coppice: Nin Zou ;-)
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09:56.41shadebobhi
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09:59.31shadebobcan I detect (with zap channel) answer of th callee party? Answeronpolarity is just for hangup ...?
10:03.39pooh_shadebob: ${dialstatus}
10:05.12shadebobpooh_ : my problem is with an digital line (pri, bri) answer and hangup of callee give good cdr informations. With zap channel, cdr start with ring tone and not answer of callee
10:06.16coppiceshadebob: most types of analogue line do not inform the caller when the called party answers
10:08.49shadebobthere a no asterisk apps to detect the end of the ring tone?
10:09.20coppicethe end of ring tone has no specific meaning
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10:14.17shadebobso cdr information will be always false?
10:14.56coppiceyep. nothing reliable you can do to avoid that
10:15.37coppicethank digital exchanges. all analogue lines had proper supervision in the strowger days. A few countries have it now. most do not
10:15.53shadebobhow analog pbx can give good cdr information?
10:16.39coppicesometimes you can pay a higher rental and get supervision. PBX users like that. in many places that is not available, and PBXs cannot produce an accurate CDR
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10:18.42shadebobhow analog public payphone detect callee answer?
10:19.19coppicethey do not use standard lines. they get supervision in one of several ways
10:20.37shadebob2 cases : they use lines with pulse metering. In the other case they use standard line...
10:21.41coppiceoften privately owned payphones do not start charging correctly. telco owned ones have answer supervision
10:22.34shadebobmaybe I can tell telco to have a line with pulse metering
10:22.55shadebobbut I will code apps to detect 12khz or 16khz sign
10:23.25shadeboband I thing digium card have filter to supress such signal...
10:23.44coppiceyou need a card that can detect 12kHz or 16kHz
10:23.50h3xvoip payphones... heh
10:24.13coppicei saw a VoIP payphone on a website somewhere
10:24.20shadebobIt's for a gateway with existing payphones h3x ;)
10:24.38h3xhack an ATA to do that shit
10:25.13coppicehow exactly would an ATA do this?
10:25.18shadebobI install an asterisk box with tdm card
10:26.11h3xwhy couldnt it
10:26.22shadeboblinksys ata have an open firmware I thing.... but electronics componements are not present for this task....
10:26.22coppiceexplain how
10:26.46h3xisnt the only electrical difference, ground start?
10:27.06coppiceif you know of open fimware for any VoIP other than PA1688, do tell
10:27.21h3xwell thats what i mean
10:27.25coppicethey don't usually ground start
10:27.25h3xmake an ata for payphones
10:27.40h3xusing a 1688 would be a good start
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10:29.53coppiceshadebob: it seems the chips on the digium TDM cards can generate 12kHz and 16kHz, but I don't think they can detect those tones.
10:30.44shadebobold tdm card can.... but newer have new si3215 chip that can not generate
10:31.24coppicewell, its detection you need, so it doesn't make a lot of difference
10:31.42shadebobbut it seem si3215 can detect 12 or 16k signal
10:31.56h3xand turn those tones into sip info messages or whatever
10:32.18h3xthe reason im thinking its a good idea is coz i was at a payphone convention here in vegas recently
10:32.24coppiceso that should be OK
10:32.42h3xand a lot of the guys were talking about how they are getting out of the business because they are still getting raped with 1980's prices from the ILECs
10:32.46h3xfor payphone access lines
10:32.55h3xand they dont have as much usage as before because of cell phones
10:33.12h3xbypassing it with wireless, cable modem, whatever they can get their hands on
10:33.13coppiceand getting broadband access lines would be better?
10:33.16h3xhell even DSL is cheaper than a payphone line here
10:33.26h3xthey are paying 60 bucks a month plus taxes !
10:33.35h3xin some regions
10:33.38coppiceI thought most new payphones were cellphones
10:33.53h3xive never seen those here, it would make more sense tho
10:34.34coppicethey are all over the place. often very fancy with big LCD touch screens that show ads when nobody is calling
10:35.21Dr_Raythe idea is to put 4 or 8 phones in a busy area with one dsl line
10:35.28Dr_Raylike at a hospital
10:35.33Dr_Raywhere poor people are
10:36.01coppiceis anyone really too poor to have a cellphone these days?
10:36.05Dr_Rayyes
10:36.10Dr_Rayor cellphone batteries die
10:36.18Dr_Raywe price our long distance at 10 cents a minute
10:36.27Dr_Raybetter than a ghetto cell phone and calling card
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10:37.07Dr_Rayit's also an amenity
10:37.28coppicechina has 1.2B people and 350M cellphone users. if you discount the poor farmers far from coverage, its pretty much 100% penetration in the cities
10:37.51Dr_Rayit's not 100% penetration
10:37.57Dr_Raysome people have multiple phonex
10:37.59Dr_Rays
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10:38.14coppiceit really is 100%
10:38.20Dr_Rayno, it really is not
10:38.33Dr_Raythe poorest people have cell phones that don't work all the time
10:38.36Dr_Rayrun out of minutes
10:38.36Dr_Rayetc
10:38.42mesuthi
10:38.44shadebobok coppice : in fxo module of tdm card the couple of chips si3050/si3018 is present. That chips can detect polarity reversal and billing tones (http://www2.silabs.com/public/documents/tpub_doc/dsheet/Wireline/Silicon_DAA/en/si3050.pdf)
10:38.51coppiceare you a china expert?
10:39.12Dr_Rayare you? just saying it is, does not make it so
10:39.26Dr_Rayour payphones do fairly well when priced fairly
10:39.51trixterit only takes 1 person to not have a cell phone for it to not be 100%
10:39.56coppiceI spend much of my life in china. no janitor cleaning to toilets would be without their cellphone
10:40.00Dr_Rayno, of course it's 100%
10:40.07Dr_Rayhow could we be so stupid to argue with him
10:40.12Dr_Rayit's clearly 100%
10:40.12trixterheh
10:40.16Dr_RayI apologize
10:41.02trixterhow much of china is rural?  cell phone usage would most likely be less in rural areas
10:41.05Dr_RayI like how 350M of 1.2B is 100%
10:41.10trixtermuch like it is in all other rural areas around the world
10:41.25trixteroh I just walked into it is that what was said?
10:41.35coppicethe main places payphone exist these days seem to be tourist spots, where people don't have roaming coverage
10:41.49trixterahh he qualified with in the cities
10:41.54trixterhow many 2 year olds have cell phones?
10:42.01trixterit only takes one for it not to be 100% in the cities :P
10:42.02coppiceDr idiot seems to have a reading problem
10:42.03Dr_Raywithin the cities he's been too
10:42.21Dr_Rayhey, I apologized and said you right, no need for name calling
10:43.37trixterI would argue that the vast majority of china lives in the cities rather than in the rural areas though..  he siad 350M out of 1.2B - exclude the farmers and you are left with the city population (in effect) and I think it might be the other way around
10:43.45coppicein china 350M means cellphones are already spreading well into rural areas, with their number determined to a large extent by the speed with which coverage spreads
10:44.10trixterlook at america as one example - this type of demographic is fairly constant -- the city of sacramento has 600k people (not counting the cities immediatly adjecent to sacramento) and the state of nebraska has 600k people
10:44.38trixterwhy is that?  farms take a lot of space and you cant pile a lot of people in that space.  by definition that is how it works.  as a result the rural areas have far fewer people
10:44.48trixterI live one county away from sacramento and the whole county only has 40k people
10:44.53trixtersame reason, its farm land
10:44.55coppicechina still has a huge farming population. city dwellers are predominantly around the coast
10:45.18trixternow if 'market penetration' was defined as 'anyone who wants one has one' then perhaps I would agree
10:45.50trixterbut I do not know if even that is accurate.  I do know that not everyone who can have one does indeed have one - which is the marketing wet dream of any company
10:46.12coppiceHK now has significantly more cellphone subscribers than people. i guess quite a few places are like that
10:46.14trixteryeah but if you look at *any* farming vs rural situation you will find that the vast majority of people are in the urban areas
10:46.30mog_homegnite
10:46.34trixterthe state of california is 90% rural by geography, but 90% of the 35M people live in urban areas which comprise 10% of the geography
10:46.43trixterby definition that is what makes those areas urban vs rural
10:47.16coppiceabout 50% of the world's population lives in cities. that's nearly 100% in places like US, and something much lower in less developed places
10:47.17trixternight moggy
10:47.38trixterum what is nearly 100% in the US specifically
10:47.49trixterthe worlds population or the US population or ...
10:48.00Dr_Raythe same as 100% of people have cell phones
10:48.11trixterperhaps 100% of the US is urban?
10:48.14trixterit wasnt clear what you meant
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10:48.46coppiceI think a very high percentage of the US population and certainly a high percentage of western europe live in cities. balancing that places like china are way below 50%
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10:49.52trixterthe problem with the majority of people in any given society living in rural areas is that farms (whether dirt farming or livestock) takes a lot of land, which implicitly means that people cannot occupy  the same per sq foot or however its measured density
10:50.43trixterthe majority of the US is rural however, with the clear majority living in urban areas, NY, LA, CHI are the top 3 cities, that comprises over 10% of the US population in just those 3 cities
10:50.49trixteror close to 10% anyway
10:51.15trixterthat same type of thing must occur everywhere there is a rural area, if it does not the area is not rural
10:51.23trixterit is rural because relatively few people live there
10:51.46trixterlook at ireland, 50% live in dublin county, the other 2M people are spread throughout the country, a good chunk of the remaining live in cork county (I think 25% are in cork)
10:51.47coppicewell, if you roam around rural US the homes are far apart. try that in china and homes are never far away. even the rural population is not that thinly spread. there are huge numbers there
10:52.10trixterdiscovery channel disagrees with your depiction of rural china
10:52.23trixterthey have video footage to go on their side we have only your word
10:52.37coppicepartly because the 1 child policy only applies to city dwellers
10:52.51coppicewhat does discovery channel say?
10:53.19trixterunless something changed recently the discovery channel disagrees with that too, becuase they showed rural clinics doing testing and showed some rural farming folk that were having issues cause their daughter ran away to avoid being tested
10:53.24trixterbasically the parents get arrested too
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10:54.00trixterdiscovery channel showed footage of china which showed houses far apart, they showed large fields for crops and live stock
10:54.15Dr_Raybut coppice has been to 100% of china
10:54.30trixterthey showed people plowing the fields with old mostly manual equipment (but did note that tractors and other higher tech stuff was coming in it just wasnt everywhere yet)
10:55.20trixterhttp://home.earthlink.net/~jackgconrad/travels/china/Rural_Rice2.jpg
10:55.22coppicei didn't say no rules applied. in the cities anyone with a little money can pay taxes and have more children (kinda, sorta). in the countryside there is a sorta 1 child policy, but you pull the "I am a farmer" card, and it doesn't apply. young girls who may not be able to claim to be a farmer have a problem
10:55.24trixterlooks quite urban if you ask me
10:56.20trixterthere are waiting periods before you can buy the right to have another kid, if your first was a girl or something, that was covered however even farmers had limits
10:56.28coppiceyou are seeing one small field. notice there is rather more than one farmhouse beyond it
10:57.04frenzyhi
10:57.06trixterhttp://www.lib.utexas.edu/maps/middle_east_and_asia/china_population_83.jpg
10:57.09frenzyI have issue with Asterisk
10:57.21frenzyits hogging CPU
10:57.29frenzyi've already disabled MOH from loading
10:57.33frenzyyet I face it
10:57.50trixterhttp://www.fao.org/DOCREP/005/AC801E/ac801e09.jpg
10:57.53coppicechina is like 2 countries. 2-300M around the coast which have seem massive development. the interior with the rest of the people looking like the land what time forgot.
10:58.39trixterthe fao.org population density map is pretty good I think anyway, looks better visually than the utexas.edu one
10:59.47trixternote the chineese girls with green eyes are on the western part of china, kinda up to the north a little
10:59.59coppiceso, ignore the east, which is desert, and the population show few obvious urban hotspots. it really is very spread out
11:00.08frenzy* is hogging CPU
11:02.09frenzy?
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11:10.41*** join/#asterisk Frk2 (n=faraz@202.5.145.13)
11:10.46Frk2anybody using asterisk 1.2?
11:10.49*** part/#asterisk delox99 (n=delox99@206-248-149-59.dsl.teksavvy.com)
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11:10.57Frk2im having meetme distortion with anything other than Ulaw
11:11.16delox99i have dificulties to load ztdummy
11:11.35delox99it was loading ok before i recompiled my kernel
11:11.55delox99now i just recompiled zaptel and make install
11:12.13Frk2is this a known issue?
11:12.22delox99do i need to do depmod or something?
11:12.36delox99i had the problem a month ago
11:12.51Frk2with meeetme?
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11:16.49Frk2hmm
11:16.52Frk2everybody sleeps :)
11:17.08trixterfrk2: I know that some stuff has been patched for 1.2 in CVS, I dont know if that specifically is one of the things..  I do plan on instlaling it soon, infact I am specing out the next system as I type this
11:17.27trixterI never run software that was 'just released' cause there is always a few things that dont work right :)
11:18.17coppiceFrk2: Someone was heavily reworking codec management in meet me, but i don't know if that got into 1.2 or not
11:18.46coppicetrixter: well, that's your fault. you won't run it and test it :-)
11:19.34Frk2well i need 1.2
11:19.40Frk2the ael stuff is out of this world
11:19.44Frk2and realtime
11:19.56Frk2and the new jitter buffer management
11:19.58Frk2its great
11:20.18Frk2except this screw up with meetme, which is essential for me, everything perfect
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11:20.36coppicethe jitter buffer only works with IAX
11:20.42Frk2thats fine
11:20.45demetriohello
11:20.53Frk2all my server to server comm is IAX anyways
11:21.09coppicesomeone is generalising jitter buffering now
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11:21.42santoshris there a way to patch asterisk1.2 to recognize inband dtmf
11:21.47demetriohow does ${UNIQUEID} work? I see that it's different even from inside the same call
11:21.52trixterif it hasnt happened already app_meetme needs to take a page out of the app_coinference playbook and do transcoding for each destination type once and only once..  the old (current?) way meetme did transcoding was to transcode for each channel even if its the same codec that was used for the previous channels..  very wasteful
11:21.58trixterhopefully that is getting fixed
11:22.28Frk2crack
11:22.35Frk2so i cant use meetme till patched :(
11:22.46coppicethat's just a minor part of meetme's problems. that just affects efficiency, not the results :-)
11:22.46Frk2i got the nightly cvs
11:22.47trixterwell I dont know  the status of it
11:22.59trixterthat is an obvious performance tweak and I do know that 1.2 had performance as one of its core driving forces
11:23.02Frk2tried 1.2 stable as well
11:23.03trixterso it might have happened
11:23.19Frk2but gsm screws up REAL bad, so does ilbc
11:23.28Frk2everything but ulaw screws up
11:23.48coppicewhat about alaw? does it fall apart with anything at all?
11:23.54santoshrbenjk: asterisk 1.2 doest have suppot for inband.. is there a way (a patch ) or somthing tht can be done for asterisk to recognize it...
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11:28.53kaypeehi all .... i am trying to "make" zaptel .... but get the following error:
11:28.55kaypeeYou do not appear to have the sources for the 2.6.11.12-xenU-rimu1 kernel installed.
11:29.27kaypeeis there a way that i can get the kernel source?
11:30.15{zombie}sure, it should be provided by your linux distribution
11:30.19frenzywhat is ADSI ?
11:30.43kaypeecan i d/l the kernel source off the internet ?
11:30.48{zombie}of course
11:30.57coppiceA Dumb Sort of Interface
11:31.01{zombie}just make sure you have the exact version and patch level your kernel was compiled with
11:31.08kaypeecan i tell apt-get to install them for me ?
11:31.19{zombie}what distribution are you running?
11:31.25frenzycoppice:  ?
11:31.31frenzyis it worth the load ?
11:31.40coppiceAnalogue Display Services Interface
11:31.46kaypeeFC3
11:32.15frenzycrypto.so ?
11:32.21frenzyAm slimming down *
11:32.31frenzyfor some reason * hogs CPU after few hrs
11:32.35{zombie}kaypee: you should be able to, but no idea what FC3 calls it.. possibly kernel-source-2.6.11-somethingorother
11:34.20kaypeeam attempting an apt-get .... will tell u how it goes
11:34.30kaypeethanks for ur help zombie
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11:37.34frenzywhat is chan_modem.so used for ?
11:38.16coppicei thought chan_modem had been thrown out of 1.2
11:38.40frenzyi'm using the old conf
11:38.53coppicemaybe not a good idea
11:39.30frenzywhy ?
11:39.45frenzydont think has changes to that
11:40.01coppicebecause chan_modem must be from an old install if it isn't in 1.2
11:40.06frenzyI havent been chan_modem.so
11:40.21frenzyam just messing with the module.conf
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11:45.33mesuterr,a very newbie here so it may be a stupid question,but,ild like to know,if i can use voip without any voip hardware,i mean without ipphones etc...
11:46.02trixteryes
11:46.08{zombie}sure
11:46.15coppiceprobably the commonest way, in fact
11:46.17trixtersoftware  htat acts like a phone is called a softphone.
11:46.17{zombie}there's several softphones around
11:46.29trixterdepending on the platform you are running there are choices even in the free ones..
11:48.36mesutwe have asterisk installed on the server,and started very new,even couldnt make it work behind nat yet
11:48.50mesutwe used linksys products
11:49.10trixterasterisk also can use your soundcard or if you have a bluetooth headset and bluetooth in your asterisk server it can use that as well
11:49.17mesutbut if it can be done in soft way,thatll be better of course,because prolly well need about 5000 of them
11:49.36trixterbut those are typically not newbie installs because its far easier to get a free softphone somewhere and use that for your first time
11:49.59trixterwell the downside of a softphone is that you have to use the soundcard, so you need good quality headsets
11:50.16mesuti see
11:51.06bbzWould anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. --
11:51.06trixtersjlabs.com has sjphone which is pretty good and can be skinned for a particular setup making it trivial to deploy..  xten.com has xpro which is also pretty good (what I use on my pda) there are others but I have not had any experience with them
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11:52.01mesuthttp://www.linksys.com/servlet/Satellite?childpagename=US%2FLayout&packedargs=c%3DL_Product_C2%26cid%3D1117044308483&pagename=Linksys%2FCommon%2FVisitorWrapper
11:52.10mesutthis is what we used in our first tests
11:52.38mesutreplacing that with softphones will really be great
11:54.28{zombie}you'll get better results using an ATA like the PAP2 than using softphones in my experience
11:54.34{zombie}and even better results using "real" IP phones
11:54.46{zombie}esp if you're going to try and support thousands of them
11:55.36frenzyNov 26 06:56:00 WARNING[4259]: file.c:583 ast_readaudio_callback: Failed to write frame
12:00.04trixterfrenzy: I see that if the remote end dies
12:00.59frenzyhmm
12:02.58mesut{zombie}, hmm,we can make it optional i think...yes that sounds good to me
12:03.07mesutg2g thank you guys
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12:15.00Jpc_Lgchi all
12:15.06jaristizabalhi
12:15.26Jpc_Lgci am search a lot of answer at y question ;=)
12:15.38Jpc_Lgci have buy for test my asterisk a Cisco IP Phone 7910
12:16.03Jpc_Lgcanyone know the process for configuring it ?
12:17.29Jpc_Lgcyou sleep ??? hihi ;=))
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13:11.46kernomanim in uk using grandstream handytone 486 and when i make a call and hang up the phone immediately rings me back - why is this?
13:14.11tzafrir_homedo you really hang up?
13:18.47coppiceso, you have a phoney call problem
13:22.33kernomanwell i put the phone back on the hook...
13:22.52kernomanhow do i fix a phoney call problem?
13:23.06tzafrir_homeset verbose 3
13:23.19tzafrir_homewhat exactly do you see in the CLI?
13:26.29kernomanweird, even though I hung up my home phone (the one i rang) kept ringing????
13:27.19kernomanthat didnt read well, what i meant to say is that even though i hung up the phone my home phone (the number i rang) kept ringing
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13:33.04kernomananyone?
13:35.43RaYmAn-Bxsounds like NAT issues
13:36.36coppiceThat Nat Issues guy is a real pain to everyone in the VoIP business :-)
13:36.47backbluedoes asterisk 1.0 to 1.2 have changes with nat and call termination stuff? in 1.0 sometimes it does not stop the call, and i have calls for hours.
13:37.50kernomanif i use xlite to call out it hangs up fine however using an analog phone plugged in to a grandstream handytone 486 it does not???
13:39.43kernomani take it the response from coppice was to me??
13:40.56coppiceit was just as whimsical comment to anyone out there :-)
13:45.52kernomansorry i meant RaYmAn-Bx
13:48.32Joe2000xwhy i dont have the command "realtime" in my CLI? im using asterisk 1.20..
13:48.37Joe2000xcan anybody help?
13:50.29*** join/#asterisk jaristizabal (n=jaristiz@69.79.133.185)
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13:57.34Gourou_fousalut tout le monde
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14:05.02endrehi there
14:05.15Gourou_fou:)
14:09.34Joe2000xcan anybody help me? T_T
14:13.20*** join/#asterisk frk2 (n=frk2@202.5.145.13)
14:13.34frk2man- what is WRONG with meetme in asterisk 1.2???
14:13.45frk2nothing but ulaw works :( anybody having this issue other than me?
14:14.07frk2sorry to make a rude entry :) just wanna know if this is my issue or a known problem
14:14.07*** join/#asterisk endre_ (i=endre@laptop.endre.be)
14:16.53frk2anybody?
14:16.55*** join/#asterisk litage_ (i=litage@CPE-203-45-246-72.qld.bigpond.net.au)
14:17.02litage_if *-server-1 only uses h323 to speak to other * servers, can the ATAs, IP phones, softphones, etc that connect to *-server-1 speak iax or sip, or is it best to use the same protocol all the way through?
14:17.43frk2i just wanna know if SOMEBODY is running 1.2 w/meetme and gsm without issues
14:17.44*** join/#asterisk jaristizabal (n=jaristiz@69.79.133.185)
14:20.05litage_frk2: questions with "anybody" usually don't get answered. try a more direct or specific question
14:21.43frk2litage- the question was specific. I'm running asterisk 1.2 - everything works fine, except meetme.. if all conference people use ulaw- all is well.. but if one uses gsm- exceptional distortion happens
14:22.08frk2i just wanted to know if this is specific to MY setup or not
14:22.23*** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com)
14:37.46bbzWould anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. --
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14:48.02h3xhahahahahahah
14:48.14h3xtheres a John Holmes MD in the phone book
14:48.24h3xthat must be rough
14:49.12litage_h3x: ?
14:50.42h3xyou know the porn star that died from aids
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14:57.56litage_'fraid not..
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15:08.38backblueanyone here use mcc?
15:09.17*** join/#asterisk burtonez (i=mimx@w201.ljudmila.org)
15:11.19litage_backblue: mandrake control centre?
15:12.08wasimmaryleborne cricket club
15:14.09backbluelitage_: mcc = biling system?
15:14.13backbluehttp://www.paskambink.lt/mcc/
15:14.22litage_ah  =P
15:14.49litage_backblue: rather than asking if people are using something, you'll get a better response if you ask a real question
15:14.50*** join/#asterisk Guest^DJ (n=me@60.49.95.211)
15:15.57Guest^DJhi guys, 1 quick question can 1 * box handles 300 analog phones on channel banks
15:18.59ikarusGuest^DJ: depends on what it has to do ?
15:19.00backbluelitage_: but i dont have it. why should i ask "a real question" if i just want to share information and talk.
15:19.03ikaruswhat else
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15:20.09litage_backblue: i'm just trying to be helpful. my experience on irc over the past years is that questions such as "does anyone use xyz?" don't get answered
15:20.40litage_is there a difference between G729 and G729a?
15:20.46Dr_RayGuest - I would do 3 boxes of 96 ports each
15:21.02ikarusGuest^DJ: so, if it is just switching between different analog phones, quite likely it can handle it without difficulty
15:21.05Dr_Raybut that is just my take on it
15:21.07znoG300 phones on one box
15:21.21znoGheh, and here I am thinking it's a bit much to have 20 phones on one box
15:21.22znoG:)
15:21.34Dr_Raywell, it depends, a hotel would not have all 300 in use
15:21.47Dr_RayI get buy on skimpier hardware
15:21.50ikarusGuest^DJ: but partitioning might be a good thing, it would enable you to optimise the setup later if you want to add more
15:21.54Guest^DJikarus: just handling extensions to extensions
15:21.54Dr_Rayer, by
15:22.16Guest^DJDr_Ray: this is a set up for a local hospital
15:22.33Dr_Raythen I would probably put 3 boxes in
15:22.34ikarusGuest^DJ: that should work
15:22.53ikarusGuest^DJ: but I would divide it per 100 just so you have a multi-box setup already and scaling is more easy
15:23.11Guest^DJi know it sounds lazy, but planning to use asterisk@home
15:23.28Dr_Rayasterisk at home is more work than real asterisk
15:23.33Dr_Rayimo
15:23.34ikarusGuest^DJ: don't
15:23.41marvI thought asterisk@home wasn't supported here?
15:23.43ikarusGuest^DJ: Asterisk is easy enough by itself
15:24.00ikarusAnd adding abstraction only makes debugging and modifications harder
15:24.01Guest^DJok
15:24.30Guest^DJhrmm
15:24.31ikarusGuest^DJ: your setup without voicemail, etc would be no work at all to do plain
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15:25.06Dr_Rayi'd worry about that many cards in one box
15:25.49ikarusDr_Ray: channel banks
15:25.58Dr_Rayis there a cheap way to hook up that ds3 card that digium uses
15:26.05Dr_Rayto a channel bank
15:26.26Dr_Ray30 users is still 4 cards
15:26.27ikarusGuest^DJ: an idea might be to use a VoIP (SIP possibly) channelbank
15:26.29Dr_Rayer, 300
15:26.37ikarusDr_Ray: VoIP channel banks
15:27.00Dr_Rayhmm.. teh adit 600 can do that?
15:27.18Guest^DJvoip channel banks ? hmm never heard of it, have to do some research
15:27.33Dr_Raythey have a g.729 card for the adit 600
15:27.57ikarusDr_Ray: I would use mu or alaw (depending on the country)
15:28.06ikarusthe bandwidth on a dedicated network isn't too much
15:28.07Guest^DJheard Rhino is the easiest to setup
15:28.29ikarusAnd using a channel bank avoids most of the hardware problems with things like IRQ's
15:28.50Dr_Raygot a brand name for a voip channel bank?
15:29.10ikarusDr_Ray: there are some mentioned on the wiki
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15:39.52arcyif i want a lot of FXS for analog phones why should i buy an expensive TDM (eg TDM2400P with full modules) instead of of many ATA adapters (2 FXS each)? It's almost 3 times the difference in money for 24 FXS ports.
15:40.03arcyi am sure i am saying something stupid here, but what?
15:40.52ikarusarcy: ATA's might be more tricky to wire up and maintain (but I agree, the cards are way overpriced)
15:41.09arcybut it would work, right?
15:41.15ikarusYep
15:41.29arcythank you ikarus.
15:41.40litage_if your asterisk server uses only h323 to connect to its provider, will asterisk performance and/or voice quality degrade if the ATAs, IP phones, and softphones use sip or iax?
15:42.32Guest^DJikarus: ATA vs channel banks on analog phone, which is easier to configure and maintain
15:42.33ikaruslitage_: if they use the same codec, NO
15:43.36ikarusGuest^DJ: channel banks with VoIP are like HUGE ATA's, so they are easier to configure as they condense alot of configuration issues and ofcourse less parts to go wrong (imagine 150 ATA's each with it's own power brick)
15:43.49litage_ikarus: so as long as everything's using the same codec, it absolutely will _not_ matter if ATA->* is IAX, and *->provider, *->other*, etc are H.323?
15:44.18Guest^DJikarus: you are right
15:44.21ikaruslitage_: for 99.9%, yes (there might be some out of band signalling issues, but those don't degrade quality)
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15:47.04znoGarcy: i'm going to be using 10 x ATAs for 20 extensions, simply cause any TDM card is a rip off
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15:47.15litage_ikarus: in that situation where everything "upstream" from your * box is using H.323, are there any advantages in using IAX instead of SIP or H.323 for ATA/IP phone/softphone->* ? or H.323 instead of the others?
15:47.24znoGif 10 x ATAs were about $300 cheaper than a 24 port channel bank, i'd go the channel bank
15:47.36znoGbut since 10 x ATAs are half the price (or 1/3) of a channel bank, no way!
15:47.47arcyznoG, i plan to do something like that for the same reason. But since i have never setup an asterix box before i am asking to make sure
15:48.58ikaruslitage_: IAX is mainly important for trunks
15:49.09ikaruslitage_: and H.323 is obsolete
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15:49.55ikarusznoG: hmmmmmmm, can you give me the cost of your ATA's ? (I wonder if it would be worth adding ATA support to my little box)
15:50.07Guest^DJznoG: i have approx 300 ports to support
15:50.13arcythe way i am calculating it is that with small ATAs i pay 45 Euros per line while with TDMs or large ATAs i pay 125 Euros per line. So it's a _big_ difference
15:50.59litage_ikarus: i'm unfortunately in a position where everything upstream from me is H.323 . taking that into consideration, should my ATAs, IP phones, etc use SIP, IAX or H.323? i'm wondering if one of those protocols will perform better in this situation, or if it doesn't matter
15:51.37ikaruslitage_: SIP or IAX (depending on price and trunking nature), H.323 is still obsolete, so don't use it unless you have t
15:51.56ikarusarcy: 45 euro per line, k, I should be able to beat that
15:52.38litage_ikarus: why do you say "depending on price and trunking nature" for IAX? does one need a license to use IAX?
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15:53.07ikaruslitage_: no, but devices supporting IAX are more expensive then those supporting just SIP
15:53.08arcyreally? I found Sipura SPA2002 for 90 Euros (2 FXS each).
15:53.19riteshaMySQL RealTime: Failed to connect database server asterisk on 127.0.0.1. Check debug for more info.
15:53.25chapeaurougearcy, where
15:53.27chapeaurouge?
15:53.28ikarusarcy: I am working on a custom channel bank system
15:53.33ikaruschapeaurouge: siptronic.com
15:53.39chapeaurougek
15:53.54riteshai checked my paswd and it looks okay...wonder if anyone has some clues?
15:53.55arcywww.inkshop.gr (sorry, it's Greek )
15:54.09trixtercircuit city has a post black friday sale, toshiba $200 laptop doesnt much matter what it is, its a backup system with battery backup :)
15:54.24litage_ikarus: ah i see. so unless we'll gain some advantage from using IAX-capable ATAs and IP phones, we might as well use SIP, right?
15:54.24znoGGuest^DJ: well, i would definately NOT suggest the multi-ATA solution for your environment
15:54.30ikaruslitage_: yes
15:54.33znoGGuest^DJ: the multi-ATA solution is good maybe for up to 30 extensions
15:54.38litage_thanks for that info, ikarus
15:55.34riteshaMySQL RealTime: Failed to connect database server asterisk on 127.0.0.1. Check debug for more info....please help....any clues?
15:55.46Guest^DJznoG: thanks
15:56.33Guest^DJstill sourcing the type of channel banks, any recommendation ?
15:56.55znoGGuest^DJ: besides, if you have 300 extensions, it's likely you have the resources ($$) to afford a channel bank
15:57.06znoGwith 20 extensions like I'm about to support, budget is far more limited
15:57.24chapeaurougehow are the Granstream handytone ATA in general?
15:57.31znoGi don't like them much
15:57.40znoGrather a PAP2-NA or Sipura any day
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15:57.44chapeaurougeok
15:57.52znoGpersonal preference anyway, don't listen to me :)
15:57.54Guest^DJchapeaurouge: i used Linksys, so far so good
15:57.56ikarusGuest^DJ: Allied Telesyn has one I know of, btw, why don't you upgrade the on desk phones to VoIP, phones come in at under 75 euro, and a decent set of Ethernet equipment should be cheap
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15:58.11ikarusGuest^DJ: or is the wiring of below cat 3 quality ?
15:58.14chapeaurougehello [TK]D-Fender
15:58.20[TK]D-Fendery0
15:59.08[TK]D-FenderI'm on damage recovery today since I screwed my server last night....
15:59.08Guest^DJikarus: upgrade to Voip will involve new wiring, and there are 6 floors
15:59.11ikarusGuest^DJ: but if the current wiring is of cat3 qualit or higher 10Mbit Ethernet would work just fine
15:59.16ikarusor is it really that bad
15:59.52chapeaurougeikarus, everywhere I read, they advise no less than cat5
16:00.05Guest^DJi have not check on the wiring, but i belive those wires are at least 3 yrs old
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16:00.10ikaruschapeaurouge: cat3 is the official design spec for ethernet
16:00.24chapeaurougei know, where getting old.
16:00.28ikarusGuest^DJ: that is not the issue, cat3 or cat5 is pure wiring quality (in use since long ago)
16:00.29chapeaurouges/where/but
16:03.34Guest^DJi will need to check the cabling
16:04.42ikarusGuest^DJ: but what is the alloted amount of money per phone, I am just looking up the data and erm, the pricing is such that it would easily exceed 120 euro per phone if you go analog
16:06.59ikarusAnd most of those are not VoIP channel banks, but export a E1/T1 interface, which still requires an evilly expensive card in a box
16:07.22riteshaMySQL RealTime: Failed to connect database server asterisk on 127.0.0.1. Check debug for more info.
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16:09.13Guest^DJikarus: no set budget yet, boss is comparing asterisk vs a ericsson box
16:09.23*** join/#asterisk jeffik (n=Jeff@CPE0011505c92d3-CM014350000760.cpe.net.cable.rogers.com)
16:09.37Guest^DJi do not know how much ericsson cost
16:09.41_shad_I'm trying to create sms notifications for incoming calls, but when people call with a & in the callerid, bash takes it literally. It doesn't matter if I enclose the string in " or ' . Am I missing something?
16:12.43Guest^DJany one tried zhone channel bank, any good ?
16:16.26ikarusGuest^DJ: using TDM2400p cards (no channel bank) it would be around 2000 euro per 24 phones and you could probably safely use up to 4 (maybe 5) per box
16:17.36chapeaurougeikarus, 4 per box.. better be a _big_ box :)
16:17.48ikaruschapeaurouge: the physical box, yes :)
16:18.02ikaruschapeaurouge: actually I consider it a crappy option
16:18.47ikaruschapeaurouge: I was sure there was a 24, 30, 48 or 60 port ATA for less then that price per port (including the computer, you need a decent box with that card)
16:19.02chapeaurougeyea
16:20.02ikarusBut appaerently there is a market for such equipment, but no such equipment availible *grin*
16:20.42Guest^DJi would need 4 box each with 4 TDM2400p
16:21.31ikarusGuest^DJ: yep, expensive little joke
16:21.43ikarusconsider a good enough box might also be 2000 euro a piece
16:21.49litage_ikarus: from what you were saying earlier, i got the impression that you're not a fan of H.323 . might i enquire as to why that is?
16:22.18ikaruslitage_: because it doesn't allow signalling to travel seperate paths from data, while it's priority is much lower
16:23.07litage_ikarus: what do you mean by "separate paths"?
16:23.12ikarusGuest^DJ: another alternative might be a channel bank, and a few Digium Wildcard E1/T1 cards in a single box, but that might not work
16:23.15ikaruslitage_: SIP
16:23.28Guest^DJikarus: yes, that was my original plan
16:23.40ikaruslitage_: SIP/RTP allowes the RTP to travel a P2P path, seperate from the SIP traffic
16:23.52Guest^DJikarus: maybe i could source some used channel banks from ebay
16:24.02ikarusGuest^DJ: it depends on how well behaved the Wildcard's are
16:24.11litage_ikarus: ah i see
16:24.37Guest^DJikarus: is going to be a night mare
16:24.39ikaruslitage_: this causes lower loads on a SIP proxy, etc
16:24.45ikarusGuest^DJ: that it is
16:25.08Guest^DJbetter off just buy a off the shelf PBX
16:25.28ikarusGuest^DJ: not like
16:25.29ikarusly
16:25.33ikarusGuest^DJ: but consider new channel banks aswell, just calculate a few possibilities (including the expensive allied VoIP channel banks), it will probably still be cheaper then
16:25.40ikarusthe Ericsson box
16:25.52ikarusConsidering how much more rip-off those companies are
16:26.04ikarusfor 16 phones they tried to charge us 5000 euro
16:26.17ikarusI solved it for 1000 using Asterisk
16:27.29Guest^DJRhino cost approx USD 1,400. i would need like 13 of them
16:27.38Guest^DJ4 asterisk box
16:27.59Guest^DJ4-5 wildcard E1/T1 cards
16:28.02litage_why would you want an ATA to have more than 1 ethernet port?
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16:33.09riteshaMySQL RealTime: Failed to connect database server asterisk on 127.0.0.1. Check debug for more info....please help....any clues?
16:33.39litage_ritesha: did you check the logs?
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16:42.53JonR800litage_: maybe it's a port for a pc?
16:43.16riteshahmm..let me check the log but how do I set the port?
16:43.56litage_JonR800: they're for any ethernet device. i'm just wondering why you'd want to attach 2 ethernet devices (whether they're computers, switches, routers, etc) to an ATA
16:44.14litage_ritesha: off the top of my head, /etc/mysql/my.cnf
16:44.38JonR800litage_: i dunno.. i could see one, but two seems sort of odd.
16:45.29litage_JonR800: my thoughts exactly. but i've noticed that MANY manufacturers sell ATAs with 2 ethernet ports
16:46.00RaYmAn-Bxa fair amount of ATA's has some sort of simple QoS built in which means you just stuff the ATA before your computer when connecting to the internet and your phone won't be affected by your inet traffic
16:46.16trixtermany of the cheaper ones have a bit of POS built in
16:46.17trixter:P
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16:48.50netwetrustHi, any have probe the rapid configuration on asterisk of xorcom.com
16:50.02litage_RaYmAn-Bx: ah, so maybe the ATAs with 2 ethernet ports are implied to be "daisy-chained" between your comp and your comp's router/switch/hub..
16:50.46RaYmAn-Bxlitage_: often, yes..I believe SPA-1001 has that
16:51.33litage_RaYmAn-Bx: the SPA-2000, 2002, and 2100 have 2 ethernet ports, but the 1001 has 1
16:51.44RaYmAn-Bxhmm
16:51.54RaYmAn-BxI guess it's 2100 then perhaps :)
16:51.58litage_ack sorry, only the 2100 has 2 ethernet ports. all of the 2* series have 2 FXS ports
16:52.07RaYmAn-Bxyeah
16:52.50litage_do the extra features that the Cisco ATA-186 (or 188) have justify the price? (~$200)
16:56.51riteshahmm...here is the mysql.log.....http://pastebin.ca/31321
16:57.12riteshadont' see any issue. looks like mysql is not even getting the connect request
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17:00.14litage_ritesha: increase msyql debug/logging
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17:21.02jahanihi
17:21.08jahaniis there any asterisk Subscriber Management System ?
17:22.05rkingjahani: forgive my ignorance - what sorts of things would it do?
17:23.56jahaniprepaid card
17:24.01*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
17:24.17rkingjahani: ahh... hrmm...
17:24.26bbzWould anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. --
17:25.00iDunnobbz: please fuck off and die. or at least stop sending the same fucking advert to the same channel.
17:25.29iDunno:)
17:26.23rkingjahani: so, i'm obviously a complete newbie, but it does seem like i've seen someone do something like that with asterisk... i'm now looking through the asterisk/apps/ dir to see if anything rings a bell.
17:26.49jahaniok thank you
17:28.15sm7syxHi, can I play an announcement to the calling party ? In the 'ringing' sequence...
17:30.01rkingsm7syx: how is that different then answering it and playing a message?
17:30.34rkingsm7syx: just that the other end is getting a ring while the message is playing?
17:32.08rkingjahani: http://www.voip-info.org/wiki-Asterisk+Prepaid+Applications # does this help?
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17:47.12rkingthis stuff is really fun.
17:47.34rkingi am going to have the baddest answering machine in town.
17:56.34litage_rking: if only everyone had video phones, then you could REALLY make an interesting message for your answering machine  =P
17:57.11rkinghrmm.. there's ascii art porn... i wonder if i could make .gsm porn.
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17:57.55marvtt-pr0n?
18:02.16*** join/#asterisk Brijn (n=bas@S0106004063c0fa1f.vn.shawcable.net)
18:02.24BrijnGood morning all
18:02.29RawplayerRE
18:04.30BrijnSomething weird happened here. Everything was working fine, but suddenly all voceimail/MoH sounds are very slurred. Played at roughly 1/10th of the speed
18:04.46rkingwhat is a recommended way, under Linux+(alsa/oss), to record top-quality sounds?  i did arecord foo.wav; sox foo.wav foo.gsm; and it's crackletastic
18:04.58BrijnCalls are OK, retrieved latest cvs, removed modules, recompiled, doesn't make a difference
18:05.01h3xBrijn: did you load res_alcohol_vodka.so ?
18:05.19BrijnNo, but I did try asperin.so
18:05.44rking"Hello, my name is *, and I'm an alco.so.holic."
18:05.48h3xasprin isnt a shared library
18:06.42Brijnh3x: ahh, could that be the problem? But * had a good nice, long sleep, that should have helped as well?
18:07.05BrijnFrom just a headache, the sound shouldn't go weird
18:07.25h3xi think you need that anti hangover medication
18:07.26h3xhehe
18:07.38*** join/#asterisk xheliox (n=jeff@user-0c6s3h2.cable.mindspring.com)
18:07.58Brijn:)
18:08.40BrijnIf people get my voicemail now, they will probably think I was drunk when I left the message.. Yooooooouuuuuuuuu haaaaaavvvvveeeeee reaaaaaaaaacheeeeeed
18:09.46BrijnThe voicemails that are left are OK, the mail version is perfectly fine. But played over my Polycom or Softphone, it sounds horrible
18:10.15BrijnAnd it worked perfectly for a while, I don't know what changed :(
18:15.59rkingBrijn: and it sounds the same with VOIP?
18:18.26Brijnrking: How do you mean? Calls over the incoming SIP get the same weird sound
18:18.38*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
18:18.53BrijnLocal (same net) both phone and softphone get the wrong sounds, calls are ok
18:19.25rkingBrijn: weird stuff =\  i wish i knew enough to help you
18:20.22BrijnI'll check in tonight again, maybe one of the supermasters is in/awake/willing to help
18:27.44asterboyAnyone successfully gets order Atacomm?
18:28.42asterboyme english no good tdoay
18:28.58asterboyhung
18:29.02asterboyovt
18:34.24kernomanim using version 1.2 and when i add load => chan_zap.so to my modules.conf so i can use my x100p card i get this error: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_call any ideas why?
18:38.06JunK-Ykernoman: u need res_features too
18:38.26*** join/#asterisk }btorch{ (n=btorch@c-69-180-105-139.hsd1.fl.comcast.net)
18:39.55}btorch{If a PBX PRI line works connected to the PSTN line then shouldn't the PBX PRI work when connected to *
18:40.23}btorch{do I need something special like PBX -> CSU -< *
18:45.31Rezyou need a crossover t1 cable
18:45.45*** part/#asterisk Cresl1n (n=matt@gateway.digium.com)
18:45.51benjkis mog here?
18:48.42*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
18:51.23}btorch{Rez that's it ?
18:52.22}btorch{So I assume that the cable that comes with the legacy pbxs are not crossover
19:00.10*** join/#asterisk Nukemizer (n=Nuke@160.7.249.15)
19:03.26moralecan someone help me with asterisk? i "thought" i had it all working and now my PSTN line is gone.. when i pickup i get a dialtone, if i dial any digit it hangs up right away
19:03.32morale<PROTECTED>
19:03.32morale<PROTECTED>
19:06.45arcyIs this any good? http://www.voip-info.org/wiki/view/AG-468
19:07.08*** join/#asterisk Dr-Linux (n=loyal@202.59.75.58)
19:08.23*** join/#asterisk alephcom (n=Miranda@207.34.97.130)
19:11.56Dr-Linuxhi alephcom
19:12.46*** join/#asterisk m160858 (n=jsaenz@200.89.12.46)
19:13.49*** join/#asterisk [chico] (n=user@p54913ACF.dip0.t-ipconnect.de)
19:14.13alephcomGreeting
19:18.32tzafrir_homemorale, to what context does the line go? look at 'zap show channels'
19:20.26*** join/#asterisk Entegrity (n=Entegrit@c-24-34-120-110.hsd1.ma.comcast.net)
19:20.38*** join/#asterisk clinthome (n=clinthom@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net)
19:20.39tzafrir_homekernoman, are you sure that chan_zap was actually built? look at the date of the file in /usr/lib/asterisk/modules
19:21.19*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
19:22.45*** join/#asterisk figits (n=me@217.240.33.65.cfl.res.rr.com)
19:26.00moraletzafrir_home: 2005-11-26 12:20:40 WARNING[32083]: channel.c:2313 set_format: Unable to find a codec translation path from g729 to gsm
19:26.15moraleany idea what that means? i got outbound calls working, inbound calls don't work though
19:26.22filethat error is pretty self explanitory
19:26.29mog_homedo you have g729 codec installed
19:26.32moraleyes
19:26.36moralei have it licensed too
19:26.44mog_homedo
19:26.49mog_homeshow g729
19:26.57mog_homeor something like that do you have available lic.
19:28.09*** join/#asterisk t0ke (n=toke@201.Red-81-36-121.dynamicIP.rima-tde.net)
19:28.17t0kehi
19:28.21*** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca)
19:28.37moraleooh.. i don't think i have the module installed now since i did a debian pkg upgrade
19:28.51mog_homedndndndndnaaaa
19:28.52TelamonWhat's it called when two SIP clients send the RTP packets directly to each other? reinvite?
19:28.56mog_homethe truth reveals itself
19:29.00mog_homereinvite
19:29.02t0keanyone know if is possible to use asterisk business edition+astbill ?
19:29.24mog_homeyes, i believe so
19:29.42mog_homemay not be very easy to get it to work together
19:29.51mog_homebut be is very similar to 1.2
19:31.06TelamonDamn...  Okay, second question. :)  I'm using siproxd to proxy SIP phones behind NAT and they can call each other when I send them through the zaptel interface, but on direct extension-to-extension calls, I get no voice.  The call connects fine, just no audio.  I have reinvite disabled in the user's contexts, so any ideas what the problem is?
19:31.26morale<PROTECTED>
19:31.36benjkmog are you here?
19:31.41moralewhen i used to start asterisk i would say something about my license
19:32.11benjkyou may want to put entries for the Digium cards on this page ...
19:32.12benjkhttp://www.voip-info.org/wiki/view/PSTN+Interface+Hardware+for+Computer+Systems
19:34.02morale<PROTECTED>
19:34.05morale<PROTECTED>
19:34.08morale<PROTECTED>
19:34.08mog_homemaybe
19:34.10moralethere we go
19:34.54mog_homeyou want to add it for me?
19:35.57moraleyay it works now
19:36.36benjkmog, I have run out of steam today
19:36.55benjkalmost 5 am
19:37.27mog_homeeep
19:37.38mog_homeyet another problem with japan -_-
19:40.45robl^mmmm.. cookies
19:41.52mog_homeman i could go for a cookie
19:41.56mog_homei have 0 food at apt.
19:42.32robl^I have bagele and cream cheese.... :)
19:42.46mog_homei have bread and penut butter
19:42.49mog_homebut no jelly
19:42.50*** join/#asterisk pifiu (n=myassisb@208.205.181.170)
19:42.50robl^bagels, even
19:42.58pifiuheyy
19:43.07*** join/#asterisk Qwell[] (n=chatzill@pool-71-108-28-219.lsanca.dsl-w.verizon.net)
19:43.16mog_homeQWELL!!!!!!!!!!!!!!!!!!
19:43.24robl^QWELL!
19:43.27Qwell[]omg_home!
19:43.28pifiuanyone know how to use macros for the config files so its easier to configure?
19:43.28fileyay Qwell[]
19:55.19moralewhere can i get a 1-800 number?
19:55.38*** join/#asterisk SwK (n=SwK@12-219-151-128.client.mchsi.com)
19:58.50mog_homemmmm jelly
19:59.15benjkmorale: NuFone
19:59.33benjkhttp://www.nufone.net
19:59.46brimstonewill nufone allow you to call your 1-800 number from a payphone?
20:00.11Qwell[]yes
20:01.11robl^as long as teh payphone allows it!  I have seen some non-Bell operated payphones that require you to pay for any call, including toll free (at local call rate)
20:01.34brimstonethiw was a wally world phone
20:01.38brimstonei think bell operated
20:02.53moraleYes, we are seriously working on this mess.
20:02.54moralePatience is a virtue.
20:02.55moraleheh
20:03.16benjkI have seen payphones in the UK which required you to throw money in to make a toll-free call, but they didn't charge you for the call
20:03.48robl^anyone have any good/bad comments on experiences with Snom 320/360 phones?  I am really tempted to buy some for a new project.
20:04.49robl^in the uk, you have to pop in coins every 20 secs of a call it seems..  even  if you call only 2 km away  ;-)
20:05.06CleanerX~docs
20:05.08jboti heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com
20:05.12gambolputtypay phones might as well have card readers
20:05.18CleanerXfollow the third link, robl^
20:05.22benjksome payphones do
20:05.41moralehmm.. i just signed up with nuphone.. do i have to wait to request a DID?
20:05.52Qwell[]morale: didn't used to
20:06.03robl^CleanerX: thanks..  already read the wiki's info on the Snom phones.. ;)
20:06.18alephcommorale: I think you do now.  I looked a few days ago and it did not look like they automatically provisioned them.
20:06.49moralehmm.. do i have to email them to request a 1-800 number?
20:07.04robl^when I got DIDs from NuFone (a VERY VERY long time ago), it took about a day to get the numbers
20:07.31moraleerm.. it only provides a US48 1800 number.
20:07.34moralenot canada?
20:07.46moraleas least i only put 2$ on the account
20:07.53alephcomNo, Canadian toll free rates are high.
20:08.13moraleah
20:09.02*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
20:09.35benjk~seen shido6
20:09.41jbotshido6 is currently on #asterisk (12h 14m 57s).  Has said a total of 3 messages.  Is idling for 11h 52m 3s
20:09.52alephcomI can't think of anybody right now who will automatically provision CDN toll frees.  I have a few, I might even have an extra one but they are worth around $0.06CDN.
20:09.54benjk~seen JerJer
20:09.55jbotjerjer <n=JerJer@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #debian, 15d 12h 12m 24s ago, saying: 'not really a debian specific question, but someone here should know - Can i merge partitions in Linux?  like my / was created way too small and i would like to blow away another partiton and start over, but one issue is I am currently not ...
20:09.57robl^~seen atacomm
20:09.58jbotatacomm <~dan@69.54.45.98> was last seen on IRC in channel #asterisk, 181d 17h 5m 1s ago, saying: 'lol, well that would be the more specific place, but last time i chceked there's alot of talk about bugs in here, lol...'.
20:10.39benjkwell, morale, the NuFone guys are not around, but if you hang out for a while they may show up
20:10.39robl^181 days?!?!?  Gee, I realy have been out of touch
20:11.06bbzWould anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. --
20:11.10moralealephcom: thanks
20:11.35moralehas anyone got postgresql voicemail storage working?
20:11.47alephcombbz: lol, how many times have you posted that?
20:11.58benjkbbz is a spambot
20:12.11benjkand my /ignore still doesn't work :-(
20:12.30robl^mmmmm..  spam... fried with cheese on white bread!  american cuisine!  :)
20:12.59benjkwhite bread, eeek
20:12.59alephcomlol.   Ok, I'll ignore him.
20:14.11robl^at least he's not spamming for Viagra, ink jet refills, or home mortgages
20:14.49benjki'll have sourdough bread instead
20:14.55benjkhome baked
20:15.10benjkwith Guinness sourdough
20:15.14benjkyummy
20:15.40robl^mmmm.. that sounds good...
20:15.50robl^esp the Guinness part
20:16.13benjkYes, you use Guinness instead of water to make your sourdough
20:16.22benjkgets you a very tasty bread
20:16.46benjkhave to let the Guinness go stale and room temperature though
20:17.01robl^I knew you could usae beer.. never thought of guineess for the sour dough though..
20:17.05benjkthe lacto-bacteria don't like the fizz and they don't like it cold either
20:17.18robl^right
20:17.39benjkI have tried many different beers, Guinness makes the best tasting beer bread
20:18.20robl^I've used newcastle before
20:18.42benjkhttp://www.sunrise-tel.com/misc/Guinness-Bread.JPG
20:19.55*** join/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net)
20:20.19gordonjcpalephcom: nine times, by my counting
20:20.29Qwell[]more like about 30
20:22.14alephcomWhat a waste.  Oh, well.  I better go help my wife. :-)
20:22.38benjkwaste?
20:23.39robl^alephcom:  remember when she asks, you reply "No! it doesn't make your bum look big!" ;)
20:25.46moralewhen voicemail is working with psql, if it type 'show voicemail users' it should select the people from the database table right? all i am seeing is people in my voicemail.conf
20:25.58alephcomlol, I learnt that one already. :-)  Later, everyone.
20:26.49p1tst0pexit
20:37.21m160858hi
20:37.26m160858benjk?
20:37.30m160858are 2 there?
20:37.49benjkyes
20:38.34m160858hi
20:39.01benjkhi
20:39.04m160858hey, do yo know some provider of unlimited calls from USA & Canada?
20:39.26benjkUSA, but Canada no
20:39.41m160858others countrys?
20:39.48m160858i used broadvoice
20:40.09m160858but i have some problems, using many broadvoice accounts on my asterisk
20:40.45m160858so
20:40.54m160858now I am looking for new supplier
20:41.02m160858provider
20:41.31benjkmany people seem to have trouble with broadvoice
20:41.45m160858yes?
20:41.50benjkfor DIDs or for outbound calls
20:42.02robl^I dumped broadvoice about 4 months ago...
20:42.09m160858it's my first time using broadvoice on asterisk
20:42.16m160858exactly
20:43.09m160858hello to all !!
20:43.13m160858i'm from peru
20:43.55m160858i'm looking for a new provider
20:44.12m160858for unlimited calls from USA & Canada
20:44.16tzafrir_homeyou're also flooding the channel
20:44.53m160858oops
20:45.38m160858ok, change the question
20:46.08m160858what card digium recommends to me
20:46.16benjkwho is flooding the channel?
20:46.17m160858i've 3 E1
20:46.54benjkyou can use a quad T1/E1 card from Digium or a quad T1/E1 card from Sangoma
20:47.15*** join/#asterisk zeedo (n=zeedo@80.68.92.188)
20:47.33m160858but, they say to me that there are problems with cards of 4 ports
20:47.57m160858that one uses 3 cards of 1 port each
20:48.41*** join/#asterisk Dr-Linux (n=loyal@202.59.75.58)
20:49.40benjkwho says there are problems with 4 port cards?
20:49.42m160858problems with the bus, for example
20:49.57benjkno, you got that upside down I think
20:49.58Dr-Linuxquestion, i have new cisco 7960/7940, so can i direct load 7.4 firmware, or thats important to 1st load older SIP firmware?
20:50.05m160858a friend that works with asterisk
20:50.12benjkthere are problems with more than two cards in one PC
20:50.22m160858sure?
20:50.34m160858what kind of computer, i need?
20:50.40benjkso if you need 4 E1s, then you should have a single 4 port card
20:50.46benjkdepends
20:50.50benjkwhat codecs
20:51.07m160858i have 3 E1's
20:51.21m160858i don't know .. g729
20:52.49benjkg729 is heavy on CPU
20:53.02m160858i know, but
20:53.04benjkyou cant do that with one server
20:53.16m160858here in peru
20:53.16benjkyou need about 4 servers
20:53.36m160858the internet doesn't so good
20:53.40benjkone server can only handle about 80-100 channels with G729
20:53.43m160858sorry by my english
20:53.50benjkno problem
20:54.38benjkand for that you already need a dual Xeon
20:54.46m160858one server with whatever of processor
20:55.41benjkif you want to do 4 E1s with G729 on a single server, then you will have to use IBM OpenPower 720 4-way
20:55.52benjkcosts about 20.000 USD
20:56.14benjkthat may be able to handle it, just about
20:56.15m160858in peru I do not believe that they sell it
20:56.20mog_homenah
20:56.21m160858hahaha
20:56.29benjkyou can order it in the US, mail order
20:56.30mog_homeyou can do 4 E1 with an echo can card
20:56.34mog_homewith dual xeon box
20:56.43mog_home3 ghz
20:56.47m160858that is very expensive
20:56.47benjkmog, with G729 transcoding?
20:56.51mog_homeyeah
20:57.03benjkis the G729 on the hardware?
20:57.06mog_homeno
20:57.10mog_homesoftware
20:57.13mog_homebut you need echo can board
20:57.13m160858and?
20:57.16m160858a compatible computer
20:57.19benjksince when?
20:57.21mog_hometo take off the cpu load
20:57.26mog_homesince rev2 cards
20:57.35benjkhow many concurrent channels?
20:57.45mog_home124 channels can be done
20:57.56mog_homeprobably a few more
20:58.02benjkok
20:58.05mog_homei just know we have driven 4 e1 spans
20:58.06m160858mmmm
20:58.16benjkthat's cool
20:58.16m16085830 i think
20:58.24m160858no, impossible
20:58.43brimstonenothing's impossible
20:58.50brimstoneimpractical maybe, but not impossible
20:58.52benjkmog, you should be doing some testing with the OpenPower 710 and 720
20:58.55m160858haha
20:59.01mog_homesend me one ben
20:59.08benjkI reckon they can handle 400 or 500 channels
20:59.15Dr-Linuxanybody answer my que
20:59.16m160858I am new in this subject
20:59.18Dr-Linuxquestion, i have new cisco 7960/7940, so can i direct load 7.4 firmware, or thats important to 1st load older SIP firmware?
20:59.23benjket least the 720 4 way
20:59.35mog_homegot me ^_^
20:59.41mog_homei know we tried those sgi boxes
20:59.52benjkthe IBM uses POWER5
20:59.56mog_homeand were dissapointed on how many it could drive compared to a standard dell
20:59.59benjksual core
21:00.05brimstonei wonder how many channels my work machine can drive
21:00.11benjker dual core POWER5
21:00.42benjkIBM has those in their BlueGene super computers
21:00.57benjka whole bunch of them in the top ten of the top500 list
21:01.08m160858excuse, what about on my question?
21:01.45benjkm160858: you should try DIgium's new T1/E1 card WITH ECHO CANCELLATION ON BOARD
21:02.11benjkapparently, that can now handle about 120 channels with G729
21:02.14m160858ok, but it's funcionally on compatible computer?
21:02.30benjkandy PCI based system
21:02.47benjker *any* PCI based system
21:03.04benjkDual Xeon 3GHz
21:03.10benjkfor example
21:03.17m160858ok, thanks
21:03.38m160858I am going to eat something, already I return
21:04.08Dr-Linuxbenjk
21:05.20benjkyes
21:06.11Dr-Linuxbenjk: i asked a question
21:06.14Dr-Linuxquestion, i have new cisco 7960/7940, so can i direct load 7.4 firmware, or thats important to 1st load older SIP firmware?
21:06.20benjkI don;t know
21:06.32benjkI would have answered already if I knew
21:07.05benjksorry, can't help you with that
21:07.11Dr-Linuxno probelm sir
21:07.19Dr-Linuxbenjk: can i go with my other question
21:07.30benjk:-)
21:08.19Dr-Linuxi have 2 fxo cards (4 port each) that i already disucussed with you, but i have some doubts
21:08.52Dr-Linuxso for that >> in zaptel.conf will be fxsks=1-8
21:08.57Dr-Linuxis that right?
21:11.03benjkyep
21:11.25moraleSet(CALLERID(all)="R K McConnachie <4036681593>") - anyone know why that would not work?
21:12.07benjkis that new syntax from 1.2?
21:12.12moralei think so
21:12.24benjksorry, dont know about that
21:12.31benjkI use
21:12.41IronHelixSetCallerID("Name" <number>)
21:12.46IronHelixbut thats 1.0
21:12.53benjkSetCallerIDName(Fred Flintstone)
21:13.11benjkand SetCallerIDNum(555-555-5555)
21:13.18benjkor that, yes
21:19.16Dr-Linuxbenjk: and in my case in zapata.conf will be, [channel] >> signalling=fxs_ks, group=1 channel> 1-8
21:19.20Dr-Linuxis that right?
21:20.14benjkyes, but you may want to put a context there too
21:20.30benjklike context=incoming or so
21:20.43Dr-Linuxoo
21:21.10Dr-Linuxbenjk: yeah what i want to know, thats my real question
21:21.21Dr-Linuxif i put there context=ivr
21:21.24Rawplayerhow can i add a testnumber in my config?
21:21.31Rawplayerto test if my softphone dails
21:21.48IronHelix~softphone
21:21.49jbotsomething that should be drug out into the street and shot
21:21.55Dr-Linuxbcoz my context is in extension.conf is >> [ivr]  where i want the caller to redirect
21:22.03benjksure, ivr is fine, as long as you have a context [ivr] in extensions.conf that will handle the incoming calls
21:22.06IronHelixand/or do exten => 1234,1,echotest()
21:22.13IronHelixbrb
21:23.13RawplayerIronHelix: that was to me?:)
21:23.21IronHelixyeah, both of them
21:23.57Rawplayeri see "call not approved"
21:24.03Rawplayeron my softphone
21:24.08benjkok, I really have to get some sleep now. cya
21:24.34IronHelixeh
21:24.42IronHelixexten => 1234,1,echo()
21:24.44IronHelixnot echotest
21:24.45IronHelixmy bad
21:25.15*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-242-4.buckeyecom.net)
21:25.40Rawplayerstill the same
21:25.45IronHelixdid you reload
21:25.50Rawplayerrestart even
21:26.00IronHelixis the softphone in the right context?
21:26.39IronHelix(in sip.conf for the softphone config block there is context=, is that the same as the [context] in extensions.conf that the echo line is in?
21:27.45Rawplayerexten => 1234,1,echo(SIP/kevin1)
21:27.51IronHelixno
21:27.53*** join/#asterisk p1tst0p (n=admin@82-38-106-54.cable.ubr03.donc.blueyonder.co.uk)
21:28.00IronHelixexten => 1234,1,Echo
21:28.01IronHelixjust that
21:28.21p1tst0phi, i am trying to build CVS on ubuntu.. and it keeps failing here, "/usr/bin/ld: cannot find -lssl"
21:28.32p1tst0pany thoughts, on what i could be missing.
21:28.39Rawplayerlibssl
21:28.42p1tst0pah
21:28.43IronHelixp1- install packages openssl and openssl-devel
21:28.53p1tst0pcheers peeps !
21:29.11Rawplayer[kevin1]
21:29.11Rawplayertype=friend
21:29.11Rawplayercontext=kevin1
21:29.11Rawplayer;callerid=kevin1
21:29.11Rawplayercallerid="2045093kok" <101>
21:29.13Rawplayerhost=dynamic
21:29.16Rawplayersecret=test
21:29.18Rawplayerusername=kevin1
21:29.21Rawplayernat=never
21:29.23Rawplayerthats from sip.conf
21:29.32chapeaurouge~pastebin
21:29.34jbotpastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca/
21:29.40chapeaurouge~p2bin
21:29.41jboti heard p2bin is a script to paste to the http://pastebin.ca from the standard input (linux/unix CLI) . Can be fetched from http://www.madpenguin.org/blogs/chapeaurouge/?p=92
21:30.33IronHelixraw- in put your echo line in the kevin1 context in extensions.conf then
21:32.08*** join/#asterisk marc32422 (n=marc3234@206-248-134-171.dsl.teksavvy.com)
21:32.37pifiuwhere is jer jer?
21:32.59*** join/#asterisk ComputerWarm (n=workingg@66.244.235.210)
21:33.31ComputerWarmHello; Question is there anyway to do it so if someone calls in with a blocked number it auto sends them to voicemail
21:34.50*** join/#asterisk volkerli (n=volkerli@port-212-202-0-231.dynamic.qsc.de)
21:36.24ComputerWarmanyone?
21:38.32file[laptop]ComputerWarm: of course there's a way, you have the callerid accessible in the dialplan, you can compare strings, voila - there you go
21:39.48iDunnoor use exten => dialed/callerid,blah,stuff
21:40.22ComputerWarmin the dial plan for example will it handle if-else statements?
21:40.27*** join/#asterisk denon (i=denon@synapse.subneural.net)
21:40.27*** mode/#asterisk [+o denon] by ChanServ
21:41.13file[laptop]try iDunno's way first, then type show application gotoif and read
21:41.22file[laptop]type it on the CLI that is...
21:41.52ComputerWarmok thanks. the reason i was asking there is a number that calls me and i was wondering if in the dial plan, if the caller id matched that number it would hang up on them
21:44.04IronHelixyou want hte anti ex girlfriend filter :)
21:44.45IronHelixtry this http://www.voip-info.org/wiki-Asterisk+cmd+LookupBlacklist
21:44.58IronHelixuses the asterisk database to store a list of numbers that are blacklisted
21:45.12IronHelixyou can also blacklist the current call wtih a star code of some kind
21:45.28ComputerWarmIronHelix oh cool thank you
21:46.10IronHelixand if they are in the blacklist, lookupBlacklist sends them to priority current+101
21:46.25IronHelixand i think *80 is add current call to blacklist
21:46.31file[laptop]slowly am I going crazy
21:46.40IronHelixhttp://www.voip-info.org/wiki-Asterisk+vertical+service+activation+codes
21:47.02IronHelixfile- depens what on you are about talking?
21:47.25IronHelixmight be *60
21:47.31ComputerWarmIronHelix wow didn`t know there was so much added to asterisk.
21:47.41IronHelixoh theres TONS
21:47.45IronHelixpoke around in the voip-info wiki
21:47.54IronHelixasterisk has become very fully featured
21:48.15ComputerWarmya i remember when i first started using it. you where lucky to get the queue`s working lol
21:48.48IronHelixhehe
21:49.59robl^when I first started using Asterisk, I was lucky to get SIP working with my VoIP provider.  everytime chan_sip was changed, it broke something for me.. but that was long before 1.0 was released.  LOL :)
21:50.12file[laptop]those service codes are only applicable to zap analog lines you know... just a little fyi
21:52.21IronHelixmmm  good point
21:52.24ComputerWarmfile[laptop] ya i don`t think i have those services on my phone line
21:52.33IronHelixno they are asterisk services
21:52.37IronHelixif you have zap channels
21:52.43file[laptop]they're written into chan_zap
21:52.43IronHelixlike if you have a zaptel fxs port youc an use them
21:53.00file[laptop]so if you hook a phone up to an FXS port via a channel bank, or the TDM400P card, then they'll work for it...
21:57.39*** join/#asterisk lorinc (n=ang@caracas-2459.adsl.interware.hu)
22:01.37Dr-Linuxif i put there context=ivr in zapata.conf , so doing that caller will be able to redirect in extionsion.conf [ivr] context ?
22:01.40*** join/#asterisk TheCops (i=nobody@got.securebinary.com)
22:04.15Dr-Linuxwhats best webbased monitoring problem for * ?
22:04.39IronHelixFOP (flash operator panel)?
22:05.13IronHelixdrlinux- if you put context=ivr, when a call comes in it will go to extension s (or whatever the user dialled) in the ivr context and any included channels
22:06.28Dr-Linuxok thanks
22:06.32tzafrir_homeIronHelix, dependeing on the number dialed. s is if no number was dialed
22:06.38IronHelixexactly
22:06.41*** join/#asterisk rking (n=rking@ip68-1-234-152.dl.dl.cox.net)
22:07.56Dr-LinuxIronHelix: my other included context is default, and my all users are in default context, so can you give me example pateren, if the user wanna make a call outside via PSTN ?
22:08.04Dr-Linuxi have 2 fxo cards (4 port) each
22:08.31IronHelixwell put all your fxo ports in a group (call it g0)
22:09.12Dr-Linuxoo
22:09.13IronHelixdo exten => _XXXXXXXXXX,1,Dial(Zap/g0/${EXTEN})
22:09.24IronHelixput as many Xs as you expect to have digits
22:09.40Dr-LinuxIronHelix: sir but in zapata.conf i gonna put  >> group=1  and channel > 8
22:09.41IronHelixas i recally you arent in the US so you have different dialing patterns than we do
22:09.46Dr-Linuxis that wrong?
22:09.55IronHelixwell channel 8 would be IN group on
22:09.58IronHelixone
22:10.10IronHelixso if you dialled group 1 then channel 8 might be picked
22:10.37Dr-Linuxok sir but where i put g0 ?
22:10.45Dr-Linuxreally i dont understand that?
22:11.07IronHelixhttp://www.voip-info.org/wiki-Channels+and+Groups  scroll down to zap groups
22:11.16IronHelixif you put group=1 before you define all your fxo channels
22:11.23IronHelixthey will all be in group1
22:11.30Dr-Linuxokey lemme try
22:11.48IronHelixnote- you DONT wanna put fxs channels (if you have any) in that group
22:12.13IronHelixso before any fxs ports put just group= to define no group
22:12.26Dr-Linuxyes but sir what you suggest/recommend what should be in  group=?
22:12.36IronHelixa number, say 1
22:12.44IronHelixfor the fxo ports
22:12.57IronHelixdo you have any fxs?
22:13.07Dr-Linuxi have only fxo cards, i don't wanna use fxs cards
22:13.11Dr-Linuxno sir
22:13.14IronHelixok then you're good
22:13.29IronHelixjust before you define the fxo ports with channel= do group=1
22:13.30Dr-Linuxi don't wanna use them, i have only 2 fxo cards (4 port each)
22:13.47IronHelixso you have like channel=1-8
22:13.58Rawplayercan anyone tell me whats wrong with this? http://www.nomorepasting.com/paste.php?pasteID=53370 i've got the sample from a book
22:14.21Dr-LinuxIronHelix: so sir in thise case what will be the pateren?
22:14.41IronHelixhuh?
22:14.44Dr-Linuxlike you intstruct bfore
22:14.45Dr-Linux<IronHelix> do exten => _XXXXXXXXXX,1,Dial(Zap/g0/${EXTEN})
22:14.48IronHelixahh
22:14.58IronHelixthen Dial(Zap/g1/${EXTEN})
22:15.12Dr-Linuxonly think i didn't understand is g0
22:15.12rkingRawplayer: does asterisk -rcvvvv show anything interesting when you do stuff?
22:15.26IronHelixg0 is group 0
22:15.41IronHelixdial zap/g0/exten means dial the called number in any zap channel from group0
22:15.46IronHelixit will choose a free one and dial
22:16.27Dr-Linuxoo ic and there will totally 8 channels like g1 , g2 g3
22:16.28Dr-Linuxright?
22:16.34IronHelixno
22:16.40IronHelixg is group
22:16.45IronHelixif you want to dial a particular channel
22:16.55IronHelixsay you wanna dial port 5 (first port on the 2nd card)
22:16.56Rawplayernope nothing
22:17.03IronHelixyou'd dial zap/5/whatever
22:17.32IronHelixbut if you dial g1 that means use the first available zap channel thats associated to group 1
22:17.57IronHelixdrlinux- pastebin your zapata.conf file
22:18.06IronHelixi'll show you what you need to do
22:18.22rkingRawplayer: hrm... nothing interesting or nothing at all?
22:18.54Dr-LinuxIronHelix: can i tell me something in pvt, if u don't mind
22:19.27IronHelixsure
22:22.57*** join/#asterisk clive- (n=pirch@ndn-165-129-135.telkomadsl.co.za)
22:23.59kernomanim trying to connect out via pstn (x100p) card however I keep getting the all circuits are busy now message - why is this?
22:24.38file[laptop]kernoman: you know, that generic is very message...
22:24.48file[laptop]er that message is very generic... you should post what you see on the CLI on pastebin.com
22:26.49kernomansorry, ok ive posted it to pastebin.com
22:27.29Rawplayerno nothing rking
22:27.57rkingRawplayer: from that same "asterisk -crvvv" shell, does a "reload" show anything interesting?
22:27.59file[laptop]kernoman: giving us the link would help so we could see it
22:29.11kernomandoh! sorry im totally new to IRC and pastebin...
22:29.23*** part/#asterisk volkerli (n=volkerli@port-212-202-0-231.dynamic.qsc.de)
22:29.30kernomanthink i have it working now though, just going to test to confirm
22:29.32Rawplayera lot of stuff with found
22:29.43Rawplayereverything looks fine
22:29.52Rawplayer<PROTECTED>
22:29.52Rawplayer<PROTECTED>
22:29.57Rawplayerbut still not working
22:30.28Rawplayersetting up sip.conf and extensions.conf is enough right?
22:30.36rkingRawplayer: i think so
22:30.44*** join/#asterisk p1tst0p (n=admin@82-38-106-54.cable.ubr03.donc.blueyonder.co.uk)
22:31.15*** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net)
22:31.40p1tst0phi, i am trying to ring my friends asterisk box, from my own asterisk box ( both with sipgate accounts) but he keeps seeing,  ast_set_read_format: Unable to find a path from g723 to g726
22:31.47p1tst0pand, ast_set_write_format: Unable to find a path from ulaw to g723
22:31.48*** join/#asterisk clinthome (n=clinthom@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net)
22:31.58p1tst0pno matter what codecs we try, it always says this on his side.
22:32.33IronHelixsomewhere, some thing is needing g723 codec
22:32.36IronHelixblah
22:32.42*** join/#asterisk shimi (n=shimi@unaffiliated/shimi)
22:32.49rkingRawplayer: something is very odd that you don't see anything in the -vvvv console when you try to sip in - it should at least say that it's denying the call for some reason
22:33.38shimiis there a good location where I can get PSTN termination for asterisk (IAX2) that is cheap (but with good sound quality) to destinations in USA / Europe / Japan ?
22:33.56p1tst0pIronHelix, if ring his sipgate account from my mobile all is good.. i can get through to him ok.. its just if i ring him from my sipgate, to his sipgate..
22:33.57*** join/#asterisk fugitivo (n=ajf@209.13.241.110)
22:34.33p1tst0pIronHelix, and we both have allow=g723 in our outgoing sipgate configs.
22:34.50IronHelixdont allow it
22:34.54rowterI been getting lots of this messages Nov 26 17:33:47 DEBUG[2833]: chan_sip.c:7251 handle_request: That's odd...  Got a response on a call we dont know about.
22:35.06IronHelixdo disallow=all then allow=ulaw allow=alaw allow=gsm allow=ilbc
22:35.07*** join/#asterisk volkerli (n=volkerli@port-212-202-0-231.dynamic.qsc.de)
22:35.19IronHelixif you cant get it going with ulaw alaw ilbc or gsm you have bigger rpbolems
22:37.12*** join/#asterisk kimc (n=freenode@pcp04041197pcs.wbrmfd01.mi.comcast.net)
22:37.41*** join/#asterisk RaYmAn-Bx (i=rayman@skumler.dk)
22:37.50clive-can anyone help me get a new network card going...?
22:38.14p1tst0pIronHelix, when i do that we get, process_sdp: No compatible codecs!
22:38.41*** join/#asterisk ThePeopleGA (n=kemtram@rev-204.120.18.37.genesiswireless.us)
22:38.50IronHelixugh
22:38.52IronHelixthat is very odd
22:39.28fugitivop1tst0p: type "show codecs" at the cli
22:39.32robl^check the SIP phones being used one each astersik box
22:39.49fugitivop1tst0p: what do you get?
22:39.50robl^I bet one of the phones is stuck on g723
22:40.03p1tst0pfugitivo, about 15 codecs
22:40.27*** join/#asterisk ThePeopleGA (n=kemtram@rev-204.120.18.37.genesiswireless.us)
22:40.30p1tst0pi see, g723, gsm, ulaw, alaw, g726, g729
22:41.02p1tst0pthing is, on the end that im seeing the Unable to find a path from g723 to g726
22:41.05moraleanyone know why i cannot call gov't of canada numbers with freeworldtel?
22:41.06IronHelixswitch the order around
22:41.20IronHelixput ulaw, alaw, gsm, g729 in that order
22:41.22p1tst0phe can successfully ring 10000, which is the sipgate test phone
22:48.51*** join/#asterisk [TK]D-Fender (i=1000@66.11.164.239)
22:48.57[TK]D-Fenderwhee!
22:48.57jahaniasterisk work fine on what distribution ?
22:49.07jahanifedora or centos?
22:49.10[TK]D-Fenderjahani : Just about everything :)
22:49.30[TK]D-FenderYou name it and Asterisk has been there....
22:49.44[TK]D-FenderI prefer Slackware, others RH based, others Debian based...
22:49.52[TK]D-FenderJust use what you're comfortable with
22:49.58[TK]D-Fenderand the rest is just Linux :)
22:50.40Rawplayerrking: what should be in the config to let sip work
22:50.53Rawplayerconfigureing sip.conf and extensions.conf should be enough right?
22:50.54jahaniok
22:50.56jahanithank you
22:51.01Rawplayeri now got the sample from oreilly book
22:51.06Rawplayerand its still not working
22:51.21*** join/#asterisk Qwell[] (n=chatzill@pool-71-108-28-219.lsanca.dsl-w.verizon.net)
22:51.29rkingRawplayer: as far as i can think, yep - i'm in the process of config'ing sip with you (i've had IAX working), so i'll let you know
22:51.43rkingRawplayer: have you been able to dial any other SIP servers with your client?
22:52.05Rawplayernope
22:52.07Rawplayerjust installed it
22:54.16Rawplayerauthentications works
22:55.35rkingRawplayer: authentication to where?
22:55.42Rawplayerto asterisk
22:55.44Rawplayerwith my client
22:55.47*** join/#asterisk netwetrust (n=sminguel@61.Red-80-24-25.staticIP.rima-tde.net)
22:55.50dudesfor iax trunking does the host have to be static or can it be dynamic
22:56.02rkingRawplayer: ahh, awesome - you're more than half-way there... maybe now the problem is just your dialplan?
22:56.35Rawplayer[internal]
22:56.35Rawplayerexten => 100,1,Dial(SIP/john)
22:56.36Rawplayerexten => 611,1,Echo()
22:56.42Rawplayerthis is in my extensions.conf now
22:57.10Rawplayerin sip.conf i have in john's part context=internal
22:57.22Rawplayerso that it refers to internal in extensions.conf
22:57.47bbzWould anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. --
22:58.04rkingRawplayer: start of with something simpler to tell if it passed or not - just like, exten => s,1,Background(tt-monkeysintro.gsm)
22:58.08rkingurr, no .gsm
22:58.19Rawplayerk lets try
22:58.20manydejadejadejavu
22:58.36Rawplayerbut whats the number then?:)
22:58.47[Airwolf]Rawplayer, can you also do "sip debug" on you console and then paste the output here ?
22:59.00Rawplayerit doesnt show anything :\
22:59.04asteriskmonkeyno paste it in pastebin youll flood the channel
22:59.16asteriskmonkeydo sip intense debug
22:59.33[Airwolf]Rawplayer, how did you start asterisk or got back at the console ?
22:59.47Rawplayerhmm it show something now
23:00.54[Airwolf]And do you here something ?
23:00.59[Airwolf]hear
23:02.08JunK-Ybbz: we actually looking for 20 501s.
23:02.41*** join/#asterisk gvag11 (n=g@ppp23-adsl-86.ath.forthnet.gr)
23:03.19gvag11hi all
23:03.22jahanignophone is for what ?
23:03.33gvag11freevoip rocks !!!!!!!1
23:03.46gvag11i just test it and its really fine ...
23:04.04robl^jahani: not much. it is a VERY OLD IAX sofphone.  Its not mained anymore
23:04.18jahaniok
23:04.18robl^not maintained, even
23:04.20jahanithank you
23:04.47JunK-Yrobl^: i prefer polycom :)
23:05.03jahaniand gtkiaxyprov ?
23:05.49*** join/#asterisk _Thor (i=_Thor@user-vc8fl7n.biz.mindspring.com)
23:06.04robl^Junk-Y: I am considering Snom or Polycom...  I am leaning towards Snom.  After 1 year with Cisco and some of their "caveats", I am ready to switch on my home system
23:06.57manysnom is cool. not perfect, but cool
23:07.11_Thorquestion: is there such a thing as a logoff command on the manager interface?
23:07.19manythey are just a bit bitchy when they figure that your supportrequest is a bug of asterisk :)
23:07.20robl^I don't think any phone is perfect. :)
23:08.52*** part/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net)
23:09.08robl^many: at least snom supports Asterisk, firmware is Linux based, and its easy to get firmware and order equipment.  Ordering from Cisco can be a pain.. You have to order the phone, firmware license, support contract, and power supply seperate.
23:09.09JunK-Ypolycom is aint bad.
23:09.21JunK-Ynever tried a snom phone yet.
23:09.33*** part/#asterisk volkerli (n=volkerli@port-212-202-0-231.dynamic.qsc.de)
23:09.45[TK]D-FenderJunK-Y : I've just reinstalled my server and am approaching normalcy :)
23:09.47robl^JunK-Y. I have been playing with their softphone / emulated Snom 360, and I am quite impressed
23:10.38moralewhy does this not work in my extensions.conf - exten => _101,1,VoicemailMain (101)
23:10.41manyrobl: we work with a complete asterisk/snom installation at work
23:10.42[TK]D-Fenderok, got to reboot, back later...
23:10.49moralei want to dial 101 and it transfer me to my voicemail
23:10.58*** part/#asterisk _Thor (i=_Thor@user-vc8fl7n.biz.mindspring.com)
23:11.11robl^many: do you use 320s and 360s?  any major complaints or "gotchas"?
23:12.13many360s, no major complaints.  tiny things which may seem annoying, but does not affect working with them. from the 13 phones ordered, i sent two back since they were... fuzzy? when doing network stuff (id guess defective nic)
23:13.50robl^hrmmm..    I had network trouble with one of 7960s and had to have it replaced..  so that's not really a Snom only problem :)
23:13.51manyfor example the display shows a little snom.com logo, which i wanted replaced or removed which is not possible
23:13.58JunK-Ytk: after 3 days? :P
23:19.59*** join/#asterisk stillbourne (n=stillbou@c-24-9-8-59.hsd1.co.comcast.net)
23:22.24robl^many: ahh..  ok..  but sound quality and reliability is pretty good?
23:22.51manyIMVHO yes.
23:22.59manyatleast when you upgraded apps to 4.3
23:23.19moralethis asterisk voicemail wants me to register my password evertime i call it
23:23.21manywhen you get'em they usually have 3.6 which seems to have memleaks, but 4.3 seems pretty fine
23:25.09*** join/#asterisk Little-L (n=daniel@0x50a471db.svgnxx1.adsl-dhcp.tele.dk)
23:25.52robl^many:  yeah.  firmware upgrade is no big deal for me.  esp since I am only talking about maybe 5-6 phones.  This will be my home system.  I have Ciscos here now.. and I have ciscos on a couple other servers I put into production..  only looking to switch here now..
23:26.09*** join/#asterisk Dr-Linux (n=loyal@202.59.75.58)
23:26.42Dr-LinuxIronHelix: ufff i got dc and i'm was trying to connect since that time , but had no luck
23:26.54IronHelixgah
23:26.59Dr-Linuxhttp://pastebin.com/439135
23:27.13IronHelixok gimme a few
23:27.24Dr-LinuxIronHelix: i don't know how to copy the whole extensions.conf
23:27.43Dr-Linuxthats why i miss somethings there
23:27.49manyrobl: thats easy anyway.  snoms have a http interface, tell them to load a new firmware, reboot them and then you can even config the rest via dhcp // http-download of configfiles.
23:27.53manypretty nice, i think
23:28.20manythe http interface also allows dialing new numbers and also numbers from lists which is nifty.
23:29.06manyits just not so nice with acls, you cant enable it at work without having the cow-orkers play tricks on their cow-orkers. 8)
23:29.14robl^many:  is there a way to config a large number via config files.. like I could with Cisco?  I found no reference to that in the manuals
23:29.26*** join/#asterisk delphiuk (n=Richard@host86-128-157-3.range86-128.btcentralplus.com)
23:29.32manyYou do.  you only have to configure this once for every phone
23:29.48*** part/#asterisk delphiuk (n=Richard@host86-128-157-3.range86-128.btcentralplus.com)
23:30.17robl^many: ahh.  ok.
23:30.18manySince it downloads from http, rather than tftp like cisco, you can even write some php-or-whatever script to do complex operations
23:31.40robl^many: that is what I was thinking...   if they work well, I really will consider them for other production systems..
23:31.44*** join/#asterisk heath__ (n=root@12-215-32-62.client.mchsi.com)
23:31.44manysadly enough, some features seem to work only with latest asterisk + latest bristuff patch
23:32.49robl^many:  I am rebuilding my server as I type.. anything special beyond the lastest 1.2 cvs and bristuff I should add for snom?
23:32.58manyi think not.
23:33.13manythere are quite nice hints pages on voip-info.org
23:33.46robl^I am finally getting off my bum and upgrading everything on this box from 1.0.7 to 1.2
23:33.50rkinghttp://pastebin.com/439169 # i'm trying to dial * using the X-Lite softphone, and this is the result. =\
23:34.08manyi geuss my server wont see 1.2 anymore *hehe*
23:34.32robl^won't see 1.2?  why not?  gone to head?
23:35.59manyno, i guess i wander off to openpbx
23:36.02Qwell[]rking: Why are you setting context=default, but using [sipincontext] in extensions.conf?
23:36.22robl^isn't openpbx just a fork of Asterisk?
23:36.45manyisnt netbsd just a fork of freebsd? :)
23:36.56Qwell[]Isn't Linux just a fork of sco unix?
23:37.02rkingQwell: good question - i had it the other way before - i've tried dozens of iterations of the configs - let me re-set it to that
23:37.23manyqwell :)
23:37.23robl^Qwell: NO NO NO!!  Linux is a stolen from SCO!  Not a fork!
23:38.08robl^many: I just mean I don't understand how OPenPBX and Asterisk differ besides name..
23:38.10rkingQwell: the result is the same
23:38.20Qwell[]and why two [sipin]'s in sip.conf?
23:38.45manyrobl^: They will differ when opo first releases.
23:39.20manyto give you an idea, since the fork opo had ~1100 commits to svn
23:39.32robl^many:  what is the different "goal" for opo?
23:39.35rkingQwell: i wasn't sure where the username piece was supposed to come from, so i made one user "st" and then later added the "2345" without thinking to change that title
23:40.35rkingsame result with the \names set to different values
23:40.36Qwell[]rking: http://www.asteriskguru.com/tutorials/xlite_softphone.html
23:40.40manyrobl: www.openpbx.org/ wiki.openpbx.org lists some, and to quote the japanese lifestyle: competition is good for your own quality. anyway, i guess this should be taken off here. :-)
23:40.59riteshaMySQL RealTime: Failed to connect database server asterisk on 127.0.0.1. Check debug for more info....couldn't resolve yet. any suggestions on how to increase debug dump for mysql
23:41.44riteshaanyone experienced similar problem would care of comment. I really need to fix this asap and any help would be appreciated
23:41.47riteshaThanks a lot.
23:42.08Qwell[]ritesha: pastebin the exact errors you're getting from asterisk
23:42.12Qwell[]and your configs
23:42.15riteshasure.
23:44.03riteshahttp://pastebin.ca/31358
23:44.13riteshawhich other config do you need?
23:44.23Qwell[]lemme see
23:45.06Qwell[]Is mysql listening on the network, or just the socket?
23:45.16riteshajust the socket
23:45.28Qwell[]Then comment out the dbhost line
23:45.31Qwell[]let it use dbsock
23:45.39Qwell[]probably dbport too
23:46.00Qwell[]if it's listening on only the socket, it won't be able to connect to localhost
23:46.59Qwell[]odd, it says if the host is "localhost", it'll connect through the socket...that's not cool at all
23:47.21m160858hi
23:47.49m160858here i'm agaim
23:47.56riteshahmm..this helped a little. I am not getting the connect error. but...  http://pastebin.ca/31359
23:48.03m160858i'm looking for a new international calls provider, for a unlimited calls to USA & Canada
23:48.09m160858do you know someone?
23:48.17riteshaso I am wondering if it actually got to mysql?
23:48.29Qwell[]ritesha: it did
23:48.34riteshaThaks a lot Qwell for helping me on this. really need it going...
23:48.36Qwell[]I'm not seeing any errors there
23:48.54riteshaOh!! Meaning I might have wrong dbase values for it not to get to the queue
23:49.05Qwell[]or the table could be missing, or the table is empty, or...
23:49.14Qwell[]ritesha: paypal is my friend, if you're *really* thankful... ;)
23:49.37riteshasure...IM me and I will be glad to
23:49.52riteshaseriously...no jokes
23:54.25m160858question: the panasonic telephones are compatibility with asterisk?
23:55.14mog_homeprobably not
23:55.27mog_homeare they  "digital"
23:55.38Qwell[]panasonic?  probably analog
23:55.49Qwell[]or perhaps IP
23:56.01m160858i'm confused
23:56.14Qwell[]m160858: Do you have a web address for the phone?
23:56.44m160858they are analog
23:57.08m160858they working actually with a panasonic pbx
23:57.31dudeshas anyone tried t38 on openpbx
23:57.43mog_homethen no
23:57.47dudesI imagine it's suppose isn't any diff than /w *
23:58.02mog_homebut it is probably similar
23:58.09m160858but i want to replace that panasonic PBX by one asterisk
23:58.51m160858that me force to change all the telephones?
23:59.05mog_homeconnect the pasanoic to asterisk
23:59.10mog_homeover t1
23:59.30Qwell[]panasonic>tincan>string>tincan>asterisk
23:59.58m160858in my country doesn't exists T1

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