irclog2html for #asterisk on 20041210

00:00.06bkw_eKo1 i'm just saying you can't call 64k broadband
00:00.09*** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com)
00:00.24eKo1yeah, i call it shitband
00:00.25ctooleybkw_, technically you can, marketingly you can't
00:00.42*** join/#asterisk l-fy (~diana@diana.null.ro)
00:00.43ctooleybroadband:  A method of transmitting data, voice and video using frequency division multiplexing (FDM), such as used with cable TV.
00:00.44l-fyhello
00:00.51l-fythe today subject is TDMoE
00:00.59bkw_why?
00:01.05bkw_just use IAX
00:01.13l-fycan i convince zapeth to emulate 2 E1's?
00:01.25l-fybkw_ > why do you change the problem before hearing it?
00:01.41znoGhere we go...
00:01.45l-fyhi znoG
00:01.46eKo1oh how i long for the nights when my isp f*cked up their bandwidth manager and my bandwidth increased to 100 kilobytes/sec
00:02.00bkw_l-fy what is the goal?
00:02.06l-fybkw_ > dosen't matter
00:02.16bkw_ztd-loc.c is your friend
00:02.35l-fyi just ask if i can have 2 E1 using ztdeth on the same ethernet?
00:02.47*** part/#asterisk eKo1 (~abc@63.245.57.70)
00:02.51bkw_you can do loop back TDM via ztd-loc
00:02.59l-fyi don't want to loop back
00:03.01ctooleySlimey... Sloaris... hmmm synonyms
00:03.11l-fyi have one ethernet and 2 boxes with ztdeth
00:03.19l-fyand i have 1 E1 now, and i need two
00:03.21l-fythat's all
00:03.40l-fyeverything else is secundary and it will not answer my question
00:03.56bkw_then do TDMoE
00:04.01bkw_from box A to box B
00:04.06l-fyztdeth is not TDMoE?
00:04.19l-fythis is what i'm doing now
00:04.25file[laptop]MUFFIN!
00:04.33bkw_http://www.convergence.com.pk/TDMoE-HOWTO
00:04.51l-fybkw_ > that HOWTO is wrong and it dosen't explain my condition
00:05.10l-fy:s/condition/situation
00:05.17bkw_how is it wrong? it shows you how to do TDMoE between two boxes? is that not what you want?
00:05.24l-fyno
00:05.27l-fyi need 2 E1's
00:05.29l-fynot one
00:05.46l-fyand i can't figure out how to have 2 E1's using ztdeth
00:05.51bkw_more channels
00:05.55l-fyno
00:05.59l-fyi need 2 E1's
00:06.01bkw_add another span
00:06.03l-fynot more chanels
00:06.04l-fyhow?
00:06.17l-fythat was the question, how can i add one more span?
00:06.27bkw_you do realize with TDMoE you're not limited to the same as E1/T1's
00:06.33bkw_you can have an E1 with 100 channels and 1 D channel
00:06.38l-fyi agree
00:06.55l-fyi need 1 E1 with 30 channes and 1 D channel
00:06.58l-fyand another 1 E1 with 30 channes and 1 D channel
00:07.04l-fycan i do that with ztdeth?
00:07.13l-fyi think my question is very clear
00:07.14bkw_just double the dynamic line
00:07.48bkw_ok ok I see how
00:08.05bkw_you create a dynamic span with 62 channels
00:08.06bkw_now
00:08.13bkw_define two E1's within those 62 channels
00:08.41*** join/#asterisk mofu (~eric@pcp04232064pcs.plyntv01.mi.comcast.net)
00:08.53bkw_signalling=euroisdn
00:09.00bkw_channel=>1-30
00:09.10bkw_or something then do 32-61
00:09.15bkw_I think you get what i'm saying now
00:10.23gigagodits not letting me simulate making a call
00:11.03blitzragedamn asterisk... always getting me to do programming
00:11.03bkw_blitzrage you can program? since when?
00:11.03bkw_;)
00:11.03l-fywell, bkw_ it seems i can't
00:11.05blitzragebkw_: ;)
00:11.08l-fyman
00:11.13l-fyis all about zaptel.conf
00:11.19l-fynot about chan_zap.conf
00:11.21blitzragedamn you asterisk for being so damn configurable
00:11.23l-fywhatever
00:11.26l-fyzapata.conf
00:11.28file[laptop]sooooooooooooo hungry
00:11.37blitzragebkw_: I've got an idea for a small patch for Asterisk so that I can learn C too :)
00:11.51blitzragebkw_: right now, learning how to use Asterisk::AGI in Perl
00:12.16bkw_blitzrage small patch?
00:12.20bkw_what might that be?
00:12.21*** join/#asterisk brett_ (~brett@domino.librum.org)
00:12.30brett_hi
00:12.30*** join/#asterisk rowter (~Drake@201.133.209.129)
00:12.43_SimonAnyone here familiar with realtime and extensions? when I issue a "realtime load extensions exten 1123" it shows my values frmo the DB so it must be working right? but when I call the extension it says 1123/default not found.  Am I missing something from my dialplan?
00:12.46brett_is there a good way to have multiple lines share a vmb?
00:13.10bkw_switch => Realtime/context@table
00:13.10l-fy1123@default
00:13.21blitzragebkw_: might not be implemented into Asterisk, but I want to add a description=blah into dundi.conf so that I can give them a customizable string to differenciate them when I do a show dundi peers
00:13.23*** part/#asterisk l-fy (~diana@diana.null.ro)
00:13.31file[laptop]blitzrage: that won't be too bad :)
00:13.32bkw_DING DONG THE BITCH IS GONE!!!
00:13.50blitzragefile[laptop]: yah... I have a patch by junky[work] that I'm going to look at that has some of that kind of thing done to give me an idea
00:13.52_Simonlol
00:14.00_Simonbkw_: much thanks. lemme take a try
00:14.12_Simonso close! then I can write my doc howto for asterisk lol
00:14.16file[laptop]666 anyone?
00:14.21HellHoundjep, here!
00:14.22blitzragefile[laptop]: sure
00:14.23brett_bkw_: any thoughts on the above question?
00:14.35rowter_Simon, great!
00:14.37bkw_vmb?
00:14.41file[laptop]voice mailbox
00:14.56bkw_ya
00:14.56bkw_just mailbox=23423@blah
00:14.56bkw_on each sip entry
00:14.58bkw_easy as pie
00:15.01brett_doesn't work
00:15.03*** join/#asterisk Moc (~mochouina@modemcable212.49-80-70.mc.videotron.ca)
00:15.06bkw_ye it does
00:15.12Mochi all
00:15.17bkw_mailbox=box,box,box
00:15.22mikegrbbkw_: does bancfirst stuck?
00:15.22bkw_if it doesn't work then its broken
00:15.24brett_hrm
00:15.26brett_Dec  9 15:58:26 WARNING[2037]: app_voicemail.c:2194 leave_voicemail: No entry in voicemail config file for '7981'
00:15.30bkw_mikegrb yes they do
00:15.32brett_that's what i get, with mailbox=
00:15.38mikegrbbkw_: k, just checking
00:15.40brett_seems to be ignored
00:15.50bkw_brett_ you don't have a mailbox in the voicemail config then
00:15.59brett_do, for the original #
00:16.04bkw_no you don't
00:16.05bkw_use
00:16.09brett_phone has two lines, 7979, 7981
00:16.11bkw_mailbox=box@context
00:16.25brett_and i just want one vmb for both
00:16.37brett_both entries in sip.conf have mailbox=7979
00:16.47brett_and they are in the same context
00:16.59mikegrbmailbox context
00:17.06mikegrbin voicemail.conf
00:17.10brett_they are both in the default
00:17.20*** join/#asterisk diomedes (~Workshop@dsl-202-173-150-229.qld.westnet.com.au)
00:17.20brett_only one vm context.
00:17.44mikegrbI think it is an issue with your drive geometry
00:17.56brett_there you go.
00:17.57bkw_show voicemail users default
00:18.13mikegrbshow bkw nudie pics
00:18.26The_Dukedoes someone know a number in the official DUNDi e164 context that I may try to lookup????
00:18.36The_Dukewith dundi lookup ....
00:18.48brett_no dice bkw
00:19.11brett_7979 shows up, but 7981 (when it xfers to vm) doesn't get redir'd to that box
00:19.32mrunixnice.  I haven't even hung up the phone and the voicemail is sitting in my email.
00:19.41_Simonbkw_: ahh your a genius!
00:19.43brett_7979 shows up happily in show voicemail users for default
00:20.07_Simonbkw_: thank you soo much!!
00:20.16brett_but even with the mailbox explicitly specified in sip.conf, * still looks for a vmb with the called extension
00:21.01_Simonso tommorow I can start writting the doc to submit
00:22.22file[laptop]hrm
00:22.23mikegrbbrett_: pastebin the maco you use for calling extentions
00:22.24brett_so that's not possible to do without hardcoding it in extensions? that seems ridiculous
00:22.35mikegrbbrett_: it only looks for the extention you specify
00:22.39file[laptop]blitzrage: I'm back
00:22.43blitzragefile[laptop]: sorry.. room mate
00:22.44file[laptop]my internet dropped out for a wee bit
00:23.01blitzrageahhh
00:23.08brett_pastebin'd
00:23.48brett_mikegrb: sure, but then what's the point of the mailbox= option?
00:23.53brett_seems like a tease
00:24.08mikegrbno, that is for sending messages waiting info
00:24.16The_Dukeis some here who can help me to get my dundi settings right???
00:24.30brett_oh.
00:24.33brett_well shit.
00:24.33modulus_jbot dundi?
00:24.36mikegrb;)
00:24.39brett_mikegrb: thx
00:24.44mikegrbno problem
00:25.04The_Dukemodulus: what do you mean....
00:25.04brett_mikegrb: so wtf was bkw saying then?
00:25.26mikegrbI think he thought you knew this
00:25.32brett_mikegrb: ah...
00:25.36mikegrband was trying to help solve a problem
00:26.07brett_mikegrb: this is sorta like the shared line appearance problem
00:26.51bkw_ok what did I need?
00:26.54bkw_where?
00:27.23*** join/#asterisk lancey (Shady@support.net1.cc)
00:27.36blitzragefile[laptop]: hehehe
00:27.39lanceyguys, what is the proper way to restrict extension by callerid?
00:27.46blitzrage~dundi
00:27.58lanceyi'm trying exten => _001./0035986811888,1,Dial(bla-bla)
00:28.02lanceyand it doesn't work?
00:28.07lanceywhat's wrong?
00:28.35_Vilemuhahaha
00:28.48mikegrbbla-blah is not a proper dialstring
00:28.56lanceyyeah i know
00:28.58mikegrbthat is your first problem ;)
00:28.59lanceyjust not to write it up here
00:28.59paulclancey: That'll work.. or should do.. only 00359xxx should be able to dial numbers starting with 001
00:29.10lanceyi'm talking generally, mikegrb!
00:29.15lanceyit's correct in the actual file
00:29.17paulcis the caller ID of the caller "00359xxx"
00:29.31lanceypaulc it's exactly the same
00:30.28paulclancey: make your dial priority 2, and make priority 1 Noop(${CALLERIDNUM})
00:30.42lanceyk, let's see what it is :)
00:31.29lanceyhmm
00:31.31lanceynow it works!
00:31.33lanceystrange
00:31.40paulcNEXT!
00:31.40lancey<PROTECTED>
00:31.41lancey<PROTECTED>
00:31.58lanceypaulc : if i don't have noop, why doesn't it work?
00:32.03paulcIt should do
00:32.08lanceyif priority 1 is the dial
00:32.10paulcgo call from a different caller ID now and make sure it doesn't work
00:33.06lanceyok, paulc: i have this:
00:33.07lanceyexten => _001./00359868118888,1,Dial(IAX2/netone@magrathea1/${EXTEN})
00:33.10lanceyand it DOESN'T work
00:33.15lanceyi change it to this:
00:33.26lanceyexten => _001.,1,Noop(${CALLERIDNUM})
00:33.26lanceyexten => _001./00359868118888,2,Dial(IAX2/netone@magrathea1/${EXTEN})
00:33.30lanceyit DOES work!
00:33.32lanceystrange!
00:34.20florzany hint what I should do to be able to make modem connections through asterisk? In as well as out are hfc isdn cards with zaphfc driver ...
00:34.36paulclancey: add the caller ID stuff to your priority 1 too
00:35.42lanceypaulc : doesn't work
00:35.51lanceyhmmz
00:36.14PTG123where the heck could i find app_vmoutcall
00:36.17lanceyit seems it doesn't find the extension, if we have callerid restriction on priority 1?
00:36.44lanceypaulc: is this normal behaviour?
00:37.10paulclancey: Do you have a _001. extension defined without the caller ID match?
00:40.02file[laptop]paulc: 666!
00:40.18paulcfile: once this song's finished
00:42.53lanceypaulc: no
00:43.24paulclancey: make one :-)
00:43.56lanceypaulc : i see now :)) just asking if this is normal?
00:44.54paulclancey: The extension has to exist without caller ID. Always. Then you can add different options that are caller ID based as well.
00:47.41gigagodhere is a new error---->  NOTICE[16613]: chan_iax2.c:3900 register_verify: No registration for peer 'gigagod' (from 192.168.1.104)
00:47.53blitzragefile[laptop]: back
00:48.24file[laptop]front and side to side
00:49.50gigagodAny ideas of where to register the user?
00:50.11ManxPowergigagod: type=user or type=friend
00:50.17ManxPowerwith host=dynamic
00:51.14PTG123anyone in here know of something t hat can place a call, and play a message?
00:51.45tzangerPTG123: uh, Dial()??
00:51.48paulcPTG123: Asterisk, when configured properly?
00:51.49ManxPowerPTG123: You mean like Asterisk can do.
00:51.56ManxPowersample.call in the asterisk source.
00:52.09PTG123i mean i want an asterisk app, that can go through a db table call the numbers, and leave a message on the voicemail
00:52.16PTG123looking for a module i can base them on
00:52.22ManxPowerPTG123: You want sample.call
00:52.33gigagodI don't type it in to the cli> prompt?
00:52.50*** join/#asterisk Legend (~legend@24.244.142.134)
00:52.53PTG123sample.call?
00:52.55ManxPowergigagod: no.  iax.conf
00:53.06gigagodah
00:53.17ManxPowerPTG123: Yes.  You drop a specially formatted file into /var/spool/asterisk/outgoing and asterisk makes the calls.
00:53.31PTG123is there not a better way?
00:53.38PTG123for a couple of calls this may be ok
00:53.42PTG123but for a ton?
00:54.23PTG123app_vmoutcall.c
00:54.25PTG123is suppose to do it
00:54.27PTG123but can't find the module
00:57.22*** join/#asterisk CoderCR (~creyna@ip68-8-11-127.sd.sd.cox.net)
00:57.24CoderCRhey guys
00:57.35CoderCRwhat TFTP server do you guys recommend?
00:58.01mikegrbhttp://www.fuckinggoogleit.com/
00:58.28nestArlol
00:58.38CoderCRstill does not answer the question
00:58.51CoderCRthere are many of them... the question was.. which do you recommend
00:58.53nestArSolarwinds for windows...
00:59.02CoderCRi am not on windows
00:59.06nestArwhatever comes with debian for linux
00:59.07CoderCRlinux only
00:59.09Legendtftpd32 is godlike on windows
00:59.17CoderCRok
00:59.19ManxPowerPTG123: The app you write does the creation of the .call files.
00:59.42cyclotomictftp-hpa-0.36 is what I use
01:00.16PTG123hmm
01:00.21PTG123let me look into it
01:00.27PTG123that just seems too simple
01:00.32Godseyftp://ftp.mamalinux.com/pub/atftp/ I use this
01:01.42CoderCRok cool
01:01.54GodseyI got a couple polycom phones today :)
01:02.04CoderCRi just got a sayson 480i
01:02.06GodseyI got them configured enough to accept a call, but not make one yet :)
01:02.10GodseyI'll leave that for monday
01:02.14CoderCRi hear the polycom phones are nice
01:03.04gigagodSimular problem still ----> NOTICE[16741]: chan_iax2.c:3900 register_verify: No registration for peer 'gigagod' (from 192.168.1.104)
01:04.18CoderCRaftp is in beta
01:06.04Godseywhere do you see that?
01:06.15The_Dukedoes someone here use dundi with the dundi-test context who may allow me to peer with, so I can finally get one single working dundi peering??? please.....
01:07.02CoderCRhttp://freshmeat.net/projects/atftp/
01:08.02Godseyatftp is by far the best
01:08.06Godseybut use whatever you want :)
01:08.12wolfsonthe_duke: http://colo.aginet.com/dundi/dundi-open.php has a form you can fill out to get a peer
01:08.33The_Dukei just did that, you should have got my mail then...
01:08.49The_Dukeerr ... my mail minutes ago...
01:09.10GodseyI enable my tftp server to create files :)
01:09.18The_DukeWolfson: got your reply....
01:09.36wolfsonyah, its done on this end
01:09.55The_DukeWolfson: can you give me a number I may try to lookup, so I know if everything works...
01:10.02gigagodNice... GOT IT- Thanks ManxPower
01:11.15wolfsonthe_duke: 12522613333
01:12.14blitzrageDBGet can't assign a null value "" to a variable?
01:13.01The_Dukewolfson: can you try to lookup 35280012345 ;
01:13.11_Vilemuahah
01:13.18wolfsonno results
01:13.37The_Dukewolfson: hmmm... okkk, i may need some help then....
01:13.45wolfson#dundi is the place for that
01:13.47_Vileds-3 change order's going through.. fuckers charge me $200 a change
01:13.51The_Dukeok..
01:19.49florzWhy could making a modem connection through * fail if both the inbound and the outbound channels are BRI?
01:20.08*** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com)
01:20.15brc_file[laptop],
01:20.45file[laptop]hello
01:21.55blitzragewoh... when I do SetVar(pincheck=pincheck + 1) in Asterisk, it crashes the system :)
01:22.13file[laptop]I'm about as busy as I can be right now so brc... if you have something urgent, spit it out
01:22.25*** join/#asterisk djMax (~djMax@dsl093-190-107.nyc2.dsl.speakeasy.net)
01:23.00djMaxanybody gotten asterisk working on Windows?  I'm having trouble getting coLinux to see the net
01:23.10brc_file[laptop], sorry I'll blab at you later
01:23.19file[laptop]good.
01:23.25brc_was going to ask you a ser question
01:24.26brc_blitzrage, did you init pincheck to 0 first?
01:24.40brc_and shouldn't that be
01:24.55brc_(pincheck=$[${pincheck} + 1])
01:25.05ManxPowerSetVar(pincheck=$[${pincheck} + 1])
01:25.15ManxPowerSee README.variables
01:25.30blitzrageoops
01:25.39brc_BWAHAHAH!
01:25.49florzor let's put it this way: Do you agree that when echo cancellation is disabled, it should be possible to establish a modem connection through asterisk if both the inbound and the outbound channel are BRI?
01:25.55blitzragebut wow... whatever I did took down the entire Asterisk system :)
01:26.14bkw_how funny
01:26.14brc_blitzrage, and you wannta do setvar(pincheck=0) before you try to +1 it
01:26.14bkw_00000030: 1500 6c69 6e75 782e 6269 6e55 5409 0003  ..linux.binUT...
01:26.15blitzrageseems like some sort of errorcheck should catch something like that :)
01:26.20brc_otherwise asterisk complains
01:26.26brc_yes it does doesn't it
01:26.38blitzragebrc_: not just complains... totally kills Asterisk
01:26.42brc_no
01:26.49blitzragebrc_: I just saw it happen
01:27.27brc_if you do SetVar(pincheck=$[${pincheck} + 1])   but pincheck is not set to anything asterisk will throw a parse error or something... if you do SetVar(pincheck=pincheck + 1) then it'll probably crash
01:27.32brc_no?
01:27.55blitzragebrc_: isn't that what I just said? :)
01:27.58brc_no
01:28.04blitzragebrc_: but it shouldn't crash is what I mean
01:28.13file[laptop]brc_: okay I'm done for a few secs, what's your question?
01:28.21blitzragebrc_: complain yes, destroy the process, I'd say no
01:28.22ManxPowerSetVar(pincheck=$[${pincheck} + 1]) when pincheck isn't set will really be SetVar(pincheck=$[ + 1])
01:28.35brc_I said if you don't setvar(pincheck=0) before you try to +1 pincheck it'll complain and you said it'll crash..which it doesn't if your setvar syntax is right
01:28.42ManxPowerblitzrage: I guess it's time to file a bug report.
01:28.46blitzrageManxPower: damnit!
01:28.47brc_ManxPower, eh?
01:28.53brc_oh yeah
01:29.18ManxPowerSometimes I REALLY HATE X-Chat's paste
01:29.33blitzrageManxPower: that's what I use irssi now :)
01:29.36file[laptop]or atleast I wish I could dance, can't until my workstation is back and working though
01:29.37ManxPowerbrc_: Get out of my mind.
01:29.39brc_ManxPower, ahhhh yes
01:29.41bkw_who is in 666
01:29.43brc_that makes sense
01:29.51file[laptop]bkw_: I could be if I dialed the number
01:29.55*** join/#asterisk wolfson` (~hehe@65.174.122.198)
01:30.31blitzragenot I
01:32.10*** join/#asterisk lancey (Shady@support.net1.cc)
01:32.33lanceyguys what does these errors mean while installing zaptel:
01:32.33lanceydepmod: *** Unresolved symbols in /lib/modules/2.4.28-special-edition/misc/wcusb.o
01:32.33lanceydepmod: *** Unresolved symbols in /lib/modules/2.4.28-special-edition/misc/zaptel.o
01:32.57brc_42/666
01:33.04redder86lancey: means you compiled the modules against a version of Linux that you're not running
01:33.43*** join/#asterisk b0r1qu4 (b0r1qu4@equinox.alluvium.com)
01:34.20b0r1qu4does anyone know where I am able to obtain documentation on queue recording?
01:34.35b0r1qu4I'm looking to learn about the format it uses for recording files
01:34.55b0r1qu4better explained, the syntax for files
01:35.10b0r1qu4file names
01:35.17*** part/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com)
01:35.36brc_botnico, it's very simple
01:35.39brc_look at the sample file
01:35.46lanceyredder86 10x, i'll check this
01:35.52lanceybut i don't think this is the issue
01:35.58brc_what do you mean the syntax for files?
01:36.00brc_filenames?
01:36.23b0r1qu4how it decides to name the file
01:36.57florzDoes anyone of you know how much latency end-to-end is acceptable for modems to connect?
01:36.59brc_there's a magic variable you can set
01:37.03brc_its in the docs
01:38.20b0r1qu4docs?  in the source?
01:40.01*** join/#asterisk Doomgaze-MSU (~Doomgaze@cottonmouth.cse.msu.edu)
01:40.03brc_dude
01:40.06brc_it's self documenting
01:40.22wolfson`florz: most modems should self compensate for quite a bit of latency
01:40.23lanceyredder86 the error seems to be because i don't use devfs, as it sux
01:40.37lanceyany one to use zaptel without devfs?
01:40.38*** join/#asterisk Kumbang (~ecvs@167.205.22.54)
01:40.39wolfson`florz: i've seen upwoards of 500ms work ok
01:40.41lancey*any way
01:40.49florzwolfson`: Hmmmm
01:41.23wolfson`florz: though thats with cisco modem's on one end
01:41.27*** join/#asterisk rfb (~rfb@skyscraper.nu)
01:41.59florzwolfson`: Well, dunno what exactly is at the other end, trying to connect to an ISP ...
01:42.34florzwolfson`: Locally, it's an Acer modem
01:42.52wolfson`florz: what error are you getting on connect?
01:43.59*** join/#asterisk denon (denon@c.coders.org.uk)
01:43.59*** mode/#asterisk [+o denon] by ChanServ
01:44.34florzwolfson`: Ohh
01:45.01florzwolfson`: I got a connect after I reduced asterisk's jitterbuffer ...
01:45.09jontowhmm
01:45.13florzwolfson`: Before it was just NO CARRIER
01:45.21jontowDec  9 20:45:02 NOTICE[18761]: chan_iax2.c:5405 socket_read: Rejected connect attempt from 192.168.2.4
01:45.56wolfson`florz: you are trying to connect via asterisk?
01:46.12blitzragebah!  duplicate sound files with different names
01:46.15sixTelyawn
01:46.15florzwolfson`: yep
01:46.44*** join/#asterisk WilliamK (~wkeller@c-24-0-130-60.client.comcast.net)
01:46.53jontowabstracted, i am trying to make a call from a SIP phone on one * to an internal extension (and ultimately other phones) on another *
01:46.56jontowconnected via LAN
01:46.56wolfson`florz: thats not gonna work real well. v.90 was designed for a single analog<-->digital conversion, anything more will cause issues
01:47.51florzwolfson`: (1) 33.6 k would be OK, too and (2) inbound and outbound are both BRI, so there is no AD/DA conversion going on ...
01:49.10*** join/#asterisk Beave (~beave@jack.vistech.org)
01:49.28*** part/#asterisk tzanger (~tzanger@165.154.13.35)
01:49.42florznow, how did I make the modem tell me the real connection speed?
01:49.56*** join/#asterisk tzanger (~tzanger@165.154.13.35)
01:50.00tzangerdammit I did that again
01:51.38wolfson`i've never seem a reliable way except from the actuall NAS
01:52.16*** join/#asterisk Darwin35 (~darwin35@pool-68-162-181-243.pitt.east.verizon.net)
01:52.21Darwin35eveing all
01:52.47Darwin35anyone here mapped dial8 dial 100 in extenions.conf
01:53.00Darwin35for storing and recalling fast dial nmbrs
01:53.06Nuggetwhat does "mapped dial8 dial 100" mean?
01:53.15Darwin35lik *54###
01:53.28Darwin35its a quick 8 or 100 stored nmbrs
01:53.28blitzragewhen using DBGet, should I be able to assign a NULL value to a variable?
01:53.43*** join/#asterisk denon (denon@c.coders.org.uk)
01:53.43*** mode/#asterisk [+o denon] by ChanServ
01:54.09NuggetI have no idea what you're trying to explain, Darwin35.
01:55.56*** join/#asterisk mlh407 (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net)
01:57.24*** join/#asterisk Yellow_Fuzzy (~yellowfuz@c211-30-2-126.wavrl1.nsw.optusnet.com.au)
01:57.26Yellow_Fuzzyhi
01:57.44djMaxif anyone from digium is here, looks like the Win version of asterisk is much happier with winpcap beta 4 over 3
01:58.09Nuggetspiffy
01:59.08*** join/#asterisk mindCrime (~mindCrime@rrcs-24-106-188-6.se.biz.rr.com)
01:59.13tzangerwhat advantage does running * on windows get you?
02:00.18djMaxnone, a disadvantage surely.  But I only have room for so many boxes.
02:00.33djMaxrunning this on a backup domain controller laptop in a closet. :)
02:00.38tzangersure but the perfomance impact would make it suck some hard balls
02:01.09djMaxsupposedly coLinux does a good job of minimizing that.
02:01.28CoderCRboy i am disappointed with the Sayson 480i
02:01.28djMaxno question it will be worse than running on Linux
02:01.45dan2bkw_: ping
02:01.48djMaxthough I suppose a 2ghz Win box will beat a 700MHz Linux box
02:01.56CoderCRi would rather go analog and with sayson 480e's than go VoIP and with Sayson 480i's
02:02.23tzangerdjMax: I suppose
02:03.16djMaxwe'll see how it goes, if it fails I'll just have to get a super small no-fan linux box
02:03.29djMaxworlds smallest * install. :)
02:04.44djMaxCoderCR have you looked at the Polycom's?
02:04.45tzangerheh
02:04.53tzangeror just ditch the BDC
02:04.57tzangerI've never needed one anyway
02:04.58djMaxThey're damn impossible to get these days, but they're cheap and work really really well.
02:05.16djMaxI should finally learn how the hell to recover a DC, but it's always totally freaked me out
02:05.52tzangerheh I've never done that
02:06.43tzangerour PDC is a P90
02:06.44Darwin35basicly its a stored speed dial
02:06.49tzangerwe image it every quarter
02:06.52Darwin35sorry was on phone
02:07.32Darwin35nugget its where you dial *54and a 3 digit nmbr then it looks in the bd and grabbs the nmbr and dials it
02:08.04Darwin35like *54101= 4122381090
02:08.14Darwin35it would then dial that nmbr
02:08.37tzangerDarwin35: piece of cake
02:08.40*** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net)
02:08.41jontowi don't really understand.. damn :)
02:09.05tzangerexten => *54101,1,Dial(4122381090)
02:09.20jpablo<PROTECTED>
02:09.32JamesDotComhrmmm, anyone with much experience with valet parking?
02:09.51JamesDotCommusic on hold dies every time i park a call
02:09.51JamesDotComcant figure out what/why ;(~
02:09.52drumkillaanyone running Fedora in here?
02:09.57mlh407I am
02:10.14drumkillamlh407: you use zaptel?
02:10.33mlh407No
02:10.41mlh407well yes, but not on that box
02:10.56drumkillaok ...
02:11.07The_Dukewolfson: did you get my posts in #dundi???
02:12.15*** join/#asterisk nikko (~Nik@12-218-243-173.client.mchsi.com)
02:12.37Darwin35the dial8 dial100 is something broadvoice has in thier services and I want to map on my box
02:13.21blitzragedrumkilla: should I be able to load a NULL value into a variable with DBGet?
02:16.56drumkillablitzrage: yeah, i think so
02:17.49blitzragedrumkilla: yah, I was looking at the code, I don't think in ast_expr.y it allows a NULL value
02:18.10[1]funkknobAllo
02:18.11blitzragedrumkilla: Dec  9 21:15:53 WARNING[19019]: ast_expr.y:478 ast_yyerror: ast_yyerror(): syntax error: parse error; Input: = ""
02:18.22[1]funkknobAnyone know what this is about? ---
02:18.27[1]funkknobLogChanRx:8172370 H323RTP Payload type mismatch: expected G729, got CiscoCN. Ignoring packet.
02:18.46[1]funkknobI'm having a one-way audio issue
02:18.52blitzragedrumkilla: I was looking around that area... and it only seems that it looks for spaces and quotes "
02:20.43jontowhmm.figured out my problem.. just gotta figure out the rest of this nonsense now, heheh
02:29.56*** join/#asterisk MrEntropy (~entropy@211.27.210.69)
02:29.59MrEntropyyo
02:30.33florzhmmm
02:31.17MrEntropyagronqui: hi, have you done much with ser nat traversal?
02:35.00*** join/#asterisk _mwoodj_ (~MWoodJ@hyper-eye.digium.sponsor.pdpc)
02:36.26file[laptop][1]funkknob: Cisco Comfort Noise, disable Comfort Noise
02:37.26[1]funkknobfile: Can that cause a one-way audio issue?
02:38.08[1]funkknobfile: Disable on the phone, on * or both?
02:38.18grant_AAnybody know about modern day phreaking
02:38.30file[laptop][1]funkknob: asterisk does not support it
02:41.25ChujiTroll alert
02:41.47blitzragegrant_A: this isn't that kind of channel
02:42.11grant_AYou know you want it
02:42.28Beaveyou're to elite.
02:42.37postelTroll Alert (2)
02:42.41[1]funkknobfile: Any way for * to just ignorantly pass the packets?
02:45.06*** join/#asterisk Mike (~mike@201.135.48.52)
02:48.16florznow everything is configured the same as before I began the tests but modem connections through * still seem to work reliably ... GNAHHHHRGRMPF!
02:50.25file[laptop][1]funkknob: no
02:54.53*** join/#asterisk ardor (~ardorgof@ip68-227-38-164.lv.lv.cox.net)
02:56.01ardorAnyone want to trade an FXO Module & 10$ for 2 FXS Modules?
02:56.01[1]funkknobfile: Is CiscoCN described in RFC3389?
02:58.09file[laptop]no
02:58.55MrEntropywhat does UAS and UAC stand for?
02:59.39*** join/#asterisk ScaredyCat (~ScaredyCa@a179019.upc-a.chello.nl)
03:00.27tzangercan anyone tell me if there is anything holding up bug 2989 from being comitted?
03:01.17brc_doesn't look like it
03:01.23brc_post a note
03:01.29tzangerpost a note stating what?
03:02.45ardor'Anything holding up this bug from being comitted'
03:02.53tzangerheh okay
03:03.02blitzrage~uas
03:03.06blitzrage~uac
03:03.10blitzragebah!
03:04.00MrEntropy=(
03:08.46*** join/#asterisk ZX81 (~matt@222-152-94-228.jetstream.xtra.co.nz)
03:09.31MrEntropyanyone know if IP phones based on the commonality PA1688 chip support symmetric rtp?
03:15.00*** join/#asterisk sixTel (sixtel@sixTel.iax.cc)
03:18.25*** join/#asterisk NormAst (NormAst@Ottawa-HSE-ppp4121010.sympatico.ca)
03:27.29tzangerhmm
03:27.34tzangeram I on a split server or something
03:27.52ZX81I keep getting a lock on downloading dir-nomore.gsm
03:28.01ZX81anyone know why?
03:29.19ZX81~ping
03:29.20jbotpong
03:29.24ZX81hmmm k
03:29.35ZX81tzanger: nah don't think so
03:29.45ZX81/who
03:33.47*** join/#asterisk IQ (~IQ@65-103-165-68.omah.qwest.net)
03:35.29*** join/#asterisk Darwin35 (~darwin35@c-24-3-204-71.client.comcast.net)
03:36.24*** join/#asterisk ScaredyCat (~ScaredyCa@a179019.upc-a.chello.nl)
03:38.10IQHi... anyone got their hands on SPA-841 ?
03:40.15brc_it's not released
03:40.24brc_late january
03:40.39IQshoot...
03:41.16Hogiebang bang
03:41.56IQI paid for this #### thing 3 weeks back
03:42.01*** join/#asterisk Brixius (Brixius@c-24-118-4-197.mn.client2.attbi.com)
03:42.14BrixiusHEllo
03:43.01blitzrageif I do a SetVar on a variable, can I do another SetVar on the same veriable to overwrite the value?
03:43.08blitzrageI'm trying, and it doesn't appear to
03:43.13blitzragejust keeps the original value
03:43.46*** join/#asterisk bjohnson (~bjohnson@ip169-165.tor.istop.com)
03:47.13Brixiusseems pretty quiet in here tonight
03:47.20file[laptop]it is,
03:47.21file[laptop]almost too quiet
03:47.33Qwell</crickets>
03:47.37blitzrageit was noisy... then died all of a sudden and never recovered :)
03:47.47file[laptop]scary
03:47.59file[laptop]I should probably shave one of these days, I appear to have a beard
03:49.22blitzragefile[laptop]: you bastard, I can't grow one of those yet :)
03:49.40file[laptop]haha
03:49.44*** join/#asterisk GarryH (~guangyaoc@S0106009027bbc526.ed.shawcable.net)
03:49.50file[laptop]I need to tirm my sideburns and goatee too I suppose
03:49.51BrixiusI've tried, I can get about 1/4 inch, then it drives me crazy
03:50.19file[laptop]I have a performance tomorrow at some sort of retirement home... how fun
03:50.23blitzragemine just doesn't grow in thick enough
03:50.26file[laptop]then the week after is performing for the chinese folks
03:50.35blitzragefile[laptop]: performing?
03:50.41file[laptop]music
03:50.46file[laptop]Christmas music at that
03:50.48blitzragefile[laptop]: ahhhh cool.  HS band?
03:50.51file[laptop]yes
03:51.06file[laptop]and dear god... what a bad legacy I'm leaving behind in my replacements
03:51.11blitzragefile[laptop]: I was a drummer for my HS jazz band.  We got to go to Japan to perform in a music festival... that was damn cool.
03:51.26file[laptop]nifty
03:51.30file[laptop]we just do local things
03:51.44file[laptop]folks want me to join the community band too... that would be interesting
03:52.10file[laptop]that's what I wanted to do, next time I have my spending account... in the positive... buy a flute
03:52.36drumkillablitzrage: drums are cool  :)
03:52.48file[laptop]we have an insane drummer
03:52.57file[laptop]and as we say, "drummers don't read music - they just improvise nicely"
03:53.12drumkillai can read music  :p
03:53.18BrixiusI tried to play drums, mostly I just made noise.  :)
03:53.20QwellI can hear music
03:53.49blitzragefile[laptop]: my mom has a wicked flute (silver) that she wants to sell
03:53.59*** join/#asterisk twisted- (~twisted@twisted.active.supporter.pdpc)
03:53.59*** mode/#asterisk [+o twisted-] by ChanServ
03:54.09blitzragedrumkilla: they sure are
03:54.15blitzragefile[laptop]: I have my grade 2 theory :)
03:54.23file[laptop]blitzrage: sell eh?
03:54.34blitzragefile[laptop]: yah.  Its just at home not doing anything
03:54.42ghotiso ... lemme get this straight... FXO talks to the telco, and FXS talks to ... a phone?
03:54.44blitzragefile[laptop]: she wanted me to sell it on ebay a while ago, but never got around to it
03:54.50blitzrageghoti: yeppers
03:54.52file[laptop]blitzrage: you should get me a pic
03:55.05blitzragefile[laptop]: ok, I'll get you one at Christmas (next weekend)
03:55.13file[laptop]k
03:55.16twistedall i want for christmas is my two front teef
03:55.20ghotialrighty ... so they're basically the opposite sides of the same "protocol", sort of like DTE and DCE?
03:55.44twistedblitzrage, heh.
03:55.46blitzragehehe
03:55.59file[laptop]tomorrow is Friday, is everyone excited?
03:56.04blitzragefile[laptop]: meh
03:56.17file[laptop]I am because my laser printer should arrive, and I get to configure it on Linux!
03:56.27blitzragefile[laptop]: every day is the same amount of work when you're in school....
03:56.30ghotiFriday means we're almost at the weekend, when I get TWO DAYS to work on stuff without being interrupted by idiots!
03:56.31blitzragelol
03:56.39blitzrageghoti: :)
03:56.56file[laptop]ghoti: you're in the #asterisk channel... there's always idiots interrupting er I mean nice people
03:57.05ghotiheh
03:57.15file[laptop]that's my mini rant
03:57.23file[laptop]you don't want to get me on a conf because I will rant worse
03:57.26file[laptop]blitzrage knows
03:57.27ghotia few more questions, and maybe I won't seem like one of those nice people.  ;)
03:57.34blitzragelol
03:57.39ardorAnyone want to trade an FXO Module & 10$ for 2 FXS Modules?
03:58.46*** join/#asterisk wolfson (hehe@cpe-24-196-251-165.man.nc.charter.com)
04:00.46IQardor: how much u want for 2 FXS?
04:01.11modulus_wordc
04:01.47GarryHI have some trouble with Zap_chan. Someone calls me, My sip phone rings and I picks up the call, and then I handup, my sip phone rings again. help!
04:02.08drumkillaanyone have recent experience with zaptel and udev?
04:02.18ghotimonth-to-month, I'm looking at $666 for 4 channels with 30 DIDs, the minimum config.  Plus $1550 install.  :)
04:03.29mrunixso what's the neatest, raddest, most useful web status interface these days?
04:04.05m-00kie666?
04:04.11ghotiya, fun, eh?
04:04.17modulus_mrunix, webalizer?
04:04.29ghotimod: I suspect not... /whois him.  ;)
04:04.38mrunixfor seeing call status, enqueued calls, etc etc
04:04.53m-00kiemine's a little more than that
04:05.12m-00kie$200 / pri + $50 space + $200 install
04:05.27ghotithat's alot cheaper.
04:05.38m-00kie100 DID's
04:05.48m-00kiewell its only 1 PRI
04:06.00m-00kieeach is $200 apiece
04:06.08ghotiso is this.  $4 per DID out-of-contract...
04:06.08mrunixhmm, brad finally nabbed mrunix.org from me
04:06.10mrunixbummer
04:06.29ghotigoes down to $420/month per PRI if I sign a 5-year contract.
04:06.37*** join/#asterisk gambolputty (~gambolput@cblmdm204-118-177-213.buckeye-express.com)
04:06.40ghotiOf course, these are Canadian dollars...
04:06.47m-00kieah canada
04:07.53twistedbkw_, you round?
04:07.56twistedor rotund?
04:08.20file[laptop]twisted: let's replace his Ciscos with Grandstreams and see if he returns
04:08.27twistedfile[laptop], good idea.
04:10.49blitzrageI love Canada
04:11.01BrixiusI'll donate the 2 Grandstream's I have for exchange for 2 of his cisco's HAHA
04:11.21Mocyour funny..
04:11.30MocI'll exchange a 7960 with a Polycom IP 600
04:13.32Nugget2 grandstreams and you'll throw in the two apostrophes for free!  :)
04:16.44twistedMoc, you are SO being paid by polycom
04:16.55brc_~seen kram
04:16.56jbotkram is currently on #asterisk
04:16.56Brixiuswould it be considered rude to do a wget of www.voip-info.org for reading offline?  I'm going on a trip and would like to do some reading on the plane.
04:17.05*** join/#asterisk Elshar (~Elshar@ip206-91.oregonfast.net)
04:17.07twistedBrixius, how would that be rude?
04:17.35Brixiusjust doesn't seem very polite to suck down an entire web site.
04:18.46ghotiDigium sells the dev kit for $195.  But you can get an IAXy for $100, and an FXO clone for $30.
04:18.49ghotiWhat am I missing?
04:18.53*** join/#asterisk charlesIII (~charlesII@67.183.28.230)
04:19.00charlesIIIhi all
04:19.12*** join/#asterisk ardor2 (~ardorgof@ip68-227-38-164.lv.lv.cox.net)
04:19.12twistedghoti, whare ARE you missing?
04:19.23charlesIIIanyone have a suggestion for a good CRM/call center SIP client ?
04:19.53ghotiwell, I'm missing all the hardware.  but I'm also missing why I'd buy a dev kit when those separate pieces work out to less cash.
04:20.27ghotior is digium just hoping folks will go with the brand name?
04:20.57GarryHI can't hangup a call from x100p. why?
04:21.08*** join/#asterisk ardor (~ardorgof@ip68-227-38-164.lv.lv.cox.net)
04:22.00GarryHAfter hangup, phone rings again.
04:22.49*** join/#asterisk luckyali (~lukyali@203.81.196.166)
04:25.42|Vulture|GarryH: could be disconnect supervision
04:25.50|Vulture|I had the same problem at one of my offices
04:26.00ghotigarryh: you must be in an area of high cosmic background radiation.
04:26.09ghotior perhaps it's sunspots.
04:26.27GarryHVulture: How can fix this problem?
04:26.56charlesIIISIP CRM/call center database integration anyone?
04:27.08GarryHI live in Alberta, Canada
04:27.20|Vulture|GarryH: it is only when the remote party hangs up the phone right?
04:28.00GarryHVulture: When I handup. not the remote party
04:28.26|Vulture|oh nvm
04:28.37|Vulture|not the same issue
04:28.48*** join/#asterisk ast_freak|Home (~Jesse_Lan@c68.112.120.152.stc.mn.charter.com)
04:28.52GarryHVulture: Someone calls me, I pickup up the call with my Sip phone.
04:29.11GarryHVulture: Then I handup
04:29.32GarryHVulture: My sip phone rings agian.
04:29.32*** join/#asterisk ctooley (~ctooley@cs70112158-28.austin.rr.com)
04:29.59ast_freak|Home~seen KalD|Work
04:30.00jbotkald|work is currently on #asterisk.  Has said a total of 9 messages.  Is idling for 5h 12m 59s
04:30.36Brixiusoh well looks like voip-info.org isn't going to be to friendly to wget.
04:31.14|Vulture|GarryH: thats strange
04:31.17GarryHBrixius: The site is made by PHP. Can't wget
04:31.20|Vulture|what SIP phone?
04:31.28bjohnsonghoti: what you are missing is the FXO support from Digium who maintain the X100P driver you would be using and the Asterisk PBX you would be using
04:31.45GarryHI have grandstream 101 and 286
04:33.24GarryHVulture: If the other remote party handup first, I get busy tone on my sip phone. No issue.
04:35.28GarryHVulture: My kernel has already been compiled with ppp support. Do I need to install PPP package from asterisk?
04:35.35b0r1qu4is there a way to automatically login agents and ensure they're always logged in?
04:36.36b0r1qu4I don't need the login/logout features of agents, but I want the reporting on call volume and who answers calls
04:36.48*** join/#asterisk twilson (~twilson@CPE-65-26-25-147.kc.rr.com)
04:37.02|Vulture|GarryH: I have never seen something like that :(
04:38.31ghotibjohnson, I've got asterisk running, and I can talk to it with soft phones.  I'm more interested in playing with it myself than purchasing support...  but are you saying that tha $30 "Digium clone" FXO devices won't work?  Or just won't be supported by the company from whom I didn't buy them?
04:38.59GarryHVulture: After the phone rings again, I pick up and then handup, the remote party is disconnected. My phone won't ring again. Strange!
04:39.30*** join/#asterisk mindfox (~mindfox@adsl19-dynamic-gw5.access.acn.gr)
04:41.02|Vulture|what does "sip show channels" show?
04:41.17|Vulture|check your debug, open asterisk -vvvvvvvvvvvr
04:41.25mindfoxThe channels that are created by the SIP protocol
04:43.34MrEntropyanyone know what the SER variable 'status' does?
04:44.44b0r1qu4anyone know much about agents at all?
04:45.30mindfoxwhat do you need to know about agents?
04:45.48luckyaliis there a peering number for FWD to Skype
04:45.49mindfoxI don't know much, but I'll try to assist as much as I can
04:45.57b0r1qu4I want to know if there's a way to make all my agents be logged in by default, rather than logged out
04:46.06b0r1qu4I appreciate it mindfox
04:46.07arrghhmmmm
04:46.34mindfoxb0r1qu4, do you want them to be logged in on a particular time of the day and from a specific device?
04:46.43arrghis there away for a user to set an extension unavailable?
04:47.05b0r1qu4time of day is less important, but cool nevertheless, definitely from a specific device
04:47.09mindfoxarrgh, there is. Let me finish with b0r1qu4 please and I'll try to explain
04:47.13arrgh;)
04:47.32arrghI'm in no hurry.
04:47.43*** join/#asterisk mindCrime (~mindCrime@rrcs-24-106-188-6.se.biz.rr.com)
04:47.55mindfoxb0r1qu4, there is a very simple way to do this; Just schedule an automated call file and put it in a cron job
04:48.10mindfoxthat call, would call an extention that automatically logs all your agents
04:48.18b0r1qu4that's pretty slick
04:48.23mindfoxbut*
04:48.37b0r1qu4I'll need to read up on call files some more, I've only used them very little to date
04:48.41mindfoxthe good thing about agents, is the ability to "follow-me" their extention
04:49.01mindfoxif you do it automatically, you'll loose this feature ;)
04:49.08b0r1qu4understandable, but in our case, our agents are very fixed
04:49.30mindfoxyou have them only to serve a number of queues?
04:49.49b0r1qu4yes
04:49.55mindfoxaha!
04:49.56b0r1qu4I'm looking to get detailed call reporting
04:50.05mindfoxWhat OS do you run on their station machines?
04:50.13b0r1qu4agents provides some nice funtionality for this
04:50.20b0r1qu4they are mostly Windows XP machines
04:50.32mindfoxok.
04:50.43mindfoxthere is a slick trick, but you have to search a little
04:50.55b0r1qu4what's that?
04:51.02mindfoxyou can have them running a login-script that will telnet for example in some port
04:51.22b0r1qu4I see where you're going with that...
04:51.27mindfoxas soon as the telnet finishes (that telnet will be on the asterisk machine, on a script*program waiting)
04:51.44mindfoxas soon as the script receives the telnet, it will automatically register that specific agent to the asterisk
04:51.55b0r1qu4actually we could do it via call files as well since our file server is NFS enabled
04:51.56mindfoxand exactly the same thing when they logoff
04:52.04mindfoxalso that ;)
04:52.21b0r1qu4now that *is* a slick idea
04:52.21b0r1qu4thanks
04:52.26mindfoxmmy pleasure
04:52.34MrEntropyanyone got experience with portaone's rtpproxy?
04:52.45mindfoxhm not me MrEntropy, sorry
04:52.53mindfoxarrgh, your turn ;)
04:53.14arrghwell
04:53.19arrghwatching you and  b0r
04:53.21arrghI had an idea
04:53.33arrghyou could sorta do it if you set extensions as agents
04:53.46mindfoxwell, that's one possibility
04:54.03mindfoxbut agents are pretty much more complicated to manage (of course with lots of features)
04:54.16mindfoxin order to let users set their unavailable extension
04:54.22arrghyea, it seems so, slightly overkill for what I'm thinking
04:54.48mindfoxyou will use the DB that is already in the Asterisk
04:54.57mindfoxthere is an example of exactly doing that in voip-info.org
04:55.03arrghsay no more.
04:55.08arrghI shall go scour again.
04:55.08mindfoxlet me find the url for you. hold on a minute, will you?
04:55.20arrghwill not make somebody work for something i should have found ;)
04:55.39arrghbut if you're feeling generous, I've been digging on voip-info, just haven't hit it yet.
04:55.47*** join/#asterisk Fpl (~Fpl@200.93.39.123)
04:55.58mindfoxarrgh, it must be on hints and tips if I remember correctly. I'm looking right now :)
04:56.07FplHELLO ALL, i have a quick question ...
04:56.14Fplgood night to all
04:56.25mindfoxthat was quick alright :)
04:56.44Brixiusno so much in the form of a question either
04:57.13mindfoxwell, he said he has a quick question, he never said that he wishes to share it with us ;) lol
04:58.00arrghmy first thought was AGI
04:58.07arrghand soembody has done just that with php and their IM status
04:58.07FplQ : lets say i have two routers (linksys) one is the gateway.router for my isp and the other i will connect in the lan ports to have more ports and a separate network. One has ip address 172. and the other 192. Q : here we go : --------> if i put a gateway in 192 will it be able to register with 172 ?
04:58.37mindfoxyou could do it by AGI also, but since there is already invented, why go after the wheel? Just use it and improve it if you like :)
04:58.41arrghthat seems to be a relatively painless way to do it.
04:58.46arrghahh yes, thats why I asked.
04:58.54arrghif the wheel is round..
04:59.12*** join/#asterisk twisted (~twisted@twisted.active.supporter.pdpc)
04:59.12*** mode/#asterisk [+o twisted] by ChanServ
04:59.38Fplq : or will it be better of using a public ip for my server ? will the iaxy work ? if i am trying to have it register to a private ip address ?
05:00.04mindfoxFpl, you can never use private IPs over the Internet
05:00.27mindfoxif you use private IP addressing for your server, you will have to statically NAT your server
05:00.30Fplmindfox : so will it be possible to use the iaxy to register to my ip address
05:01.00mindfoxthat means, you will assign a public IP that will redirect to your internal IP of your server
05:01.01arrghthanks mindfox
05:01.09mindfoxthat you should do on your router that connects with your ISP
05:01.34mindfoxand remember: you can do PAT. That means, you can redirect only specific ports (IAX2 ports) for your Asterisk box
05:01.36*** join/#asterisk GarryH (~guangyaoc@68.149.230.228)
05:01.51VetoSo 3 way calling is a function of the SIP device, right?  What if I want to do 4 way call?
05:01.53Fplmindfox : cause i have tryied forwarding ports but for some reason it does nt work
05:01.57GarryHVulture: Are you still there?
05:02.01Fplok
05:02.06CoderCRdamn it!
05:02.17CoderCRi did a shutdown -r now on my server with a vim open
05:02.19CoderCRLOL
05:02.25Vetolol
05:02.25CoderCR:(
05:02.46Fpli will try again, but since i am on the 172 and my server is on the 192 then for some reason allthough i have forwarded addresses it is not possible
05:02.51CoderCRi hate that...  now i need to go to cancel the vi retrival
05:02.53Fplweird but hey...
05:02.53GarryHVulture?
05:02.54VetoAt least you didn't do what I did the other day AT A CLIENT: #poweroff to save my laptop battery, it was their main NFS server.
05:03.01BrixiusFpl: I'm a little confused as to what you are doing with 2 linksys routers,  can you explain more as to how you are going to connect them.  I'm pictureing  Internet ----- LS1 ---- 192.168.x.x ---LK2 ----- 172.16.x.x is that right?
05:03.19Fplbrixius : correct
05:03.30mindfoxFpl, you must have a way of leting the 2 routers talk to each other, right?
05:03.37Fplright
05:03.41Fplhow ?
05:03.53Fplone gives me access to the internet over the other
05:03.56mindfoxok, let's start from the beginning
05:04.02BrixiusI'd disable nat on the backend linksys, then add a route to 172.16.x.x via the wan port of the backend linksys
05:04.03mindfoxyou have a wireless link, am I right?
05:04.09Fplone has the wan port with the internet
05:04.35Fplout of the 4 ports (LAN)S i have taken one to the other (2nd) linksys router..
05:05.01Brixiusthen add a rule that allows anything to the 172.16 network in the second linksys, controll all nat and internet access via the internet connected one.
05:05.30Fplthat seems correct but i am not sure my (old) linksys will do it.
05:05.47mindfoxVeto, you should use conferences for more than 3 parties
05:05.54Brixiushopefully that made sense
05:05.58Fpli have an old one (2 years or a little bit more) and the other is brand new
05:06.30Fplthe reason i am using this is because i have them in different floors
05:06.32*** part/#asterisk CoderCR (~creyna@ip68-8-11-127.sd.sd.cox.net)
05:06.45Fpland i wanted to have wireless capabilities in both floors
05:06.55Vetomindfox, being the * noob that I am, I assume that's via an an ext...say 9999 or such?
05:06.57Fpland now i want to integrate my asterisk
05:07.15Fplso i will use two asterisk servers
05:07.17Fplone in floor one
05:07.21Fplone in floor two
05:07.22Fplfollow
05:07.24mindfoxVeto, yes. There is enough info on voip-info.org. Do some digging, you'll be amazed of what you'll find
05:07.35Vetowill do, thanks!
05:07.37Fplboth will be interconnected
05:07.52Fplso i need each one to register to the other
05:07.57mindfoxaarg, here is the link for the unavailable thing... http://www.voip-info.org/wiki-Asterisk+call+forwarding
05:08.18Fplcause both a different set of analog phones, follow ?
05:08.53BrixiusFpl: I don't know if I'd use a linksys firewall as a router between floors, you'd be better off just plugging in a second nick to one of your asterisk boxes and enable routing on it.
05:09.18Vetomindfox: MeetMe looks like, about as easy as pie.
05:09.43Fplbrixius, i think you are correct ...but i didnt have a second nick
05:09.54mindfoxVeto, what ever suits you ;)
05:09.54Nuggetyou have a second server but not a second nic?
05:10.13Fpli "WILL" have a second server
05:10.26mindfoxFpl, is there a reason you have them in seperated subnets?
05:10.37mindfoxis the WAN provider, untrusted?
05:10.51Fplis just for security
05:10.58mindfoxok
05:11.04Fplits silly
05:11.10mindfoxanother thing to do is this:
05:11.17mindfoxYou will do double NATing
05:11.25mindfoxthe WAN provider will NAT to your internal router
05:11.38Fplbut when i try to use my iaxy (S100) it wont register
05:11.38mindfoxand your internal router will NAT into your Asterisk box
05:11.38mindfoxbut*
05:11.58Fplit wont register on my second natted ip address
05:12.04mindfoxif your WAN provider, already uses that ports for its own asterisk, then you're in trouble
05:12.13mindfoxbut there is a way :)
05:12.25Fplall ears :-)
05:12.30mindfoxeither you purchase a small subnet from your ISP
05:12.55mindfoxand in that case, you will do a full NAT to your internal router (different IP from your WAN provider that is)
05:13.04mindfoxOR (but that's not recommended)
05:13.11Nuggetyou guys are making no sense at all.
05:13.17mindfoxchange the ports that IAX uses for your box, and forward those custom ports from your router
05:13.37mindfoxbut that means that you should change the ports to your IAXy also (Don't know if that's possible though)
05:13.44Fplmindfox : do you know why is that when i have a pstn line connected to my zap channel FXO board it takes about 4 seconds to take the line ?
05:14.01mindfoxdepends on your dialplan
05:14.07mindfoxspeed of your server
05:14.14Fplahhhh
05:14.25Fplits a pentium ii 400 mhz with 384 memory
05:14.26Nuggetit should never take 4 seconds to go off hook.
05:14.28mindfoxwhen you try to dial, do you watch the asterisk console with full debuging enabled?
05:14.34Nuggetno matter the speed of your server or your dialplan
05:14.55mindfoxNugget, don't be rush :)
05:15.11Nuggetyou're giving bad, incoherent advice.
05:15.12mindfoxif the dialplan has lots of instructions (for LCR or something) it could take forever) ;)
05:15.20BrixiusFpl: also when your recieving a call, it waits for callerid or a second ring
05:15.55Fplwhen i dial the number to reach the outside line, i have to wait close to 3-4 seconds to get the dialtone  and then i dial whatever  number i want. ( i know this is not a good practice ) but i need it like that.
05:16.11Fplit really bugs me having to wait to get the second dial tone
05:16.26mindfoxNugget, I might missunderstood what you are saying, but I really don't like the tone of your typing right now. Do you mind explaining what *exactly* you mean by saying: "You are giving bad, incoherent advice" ?
05:16.52NuggetI mean that you are giving bad and incoherent advice.
05:17.08Fplso any ideas on the speed of the dialtone ?
05:17.19NuggetFpl: what are you doing the dialing with?
05:17.23Fpli can watch the dial command appear on the screen
05:17.35Fpland then 3 seconds - 4 seconds later i get dial tone
05:17.50FplNugget : an analog phone (touch tone)
05:18.03*** join/#asterisk denon (denon@amp.cif.rochester.edu)
05:18.03*** mode/#asterisk [+o denon] by ChanServ
05:18.03Nuggetplugged in to an IAXy?
05:18.09Fplno,
05:18.17Fplthat is another case
05:18.26Fpli mean fxo (TDM) card
05:18.30Fplzaptel board
05:18.37Nuggetthat would be an FXS port, not an FXO port.
05:18.40Nuggetthat's what confused me.
05:18.45Fplook...
05:18.48*** join/#asterisk denon (denon@amp.cif.rochester.edu)
05:18.50*** mode/#asterisk [+o denon] by ChanServ
05:18.52FplFXS port for the phone
05:18.56FplFXO for the line
05:19.08Fplinverse signalling (FXS---FXO, Fxo----FXs)
05:19.20Fpli got that straight
05:19.21Nuggetso you get the dial tone and dial an extension.  what's that extension's dialplan?
05:19.24mindfoxNugget, after you finish, can we go in private please? (if you have the time of course)
05:19.31Nuggetmindfox: no
05:19.46Fplnugget  : yes
05:19.59Nuggetthe "number to reach an outside line".  that's an extension?
05:20.02Fplpretty simple and straight forward
05:20.06mindfoxok. But I thought you said that the dialplan has nothing to do with the 4 secs... Now you're asking for that?
05:20.08*** join/#asterisk PTG123 (trilluser@ip68-106-19-249.ph.ph.cox.net)
05:20.10FplNugget : correct
05:20.15Nuggetwhat's in that dialplan?
05:20.49PTG123howdy ho
05:20.55mindfoxhi
05:20.58Fplin the dialplan i only have the two or three digits to access the zaptel board (channel that has the pstn line)
05:21.07Fplmeaining.
05:21.22Fpllets says you have to dial 55 to get the pstn line
05:21.40Fplso you get dialtone from the phone, dial 55, and get a second dial tone.
05:21.46Fplin the second dialtone
05:21.50Nuggetis there a reason you chose to do it that way?  why go for a second dial tone at all?
05:21.52*** join/#asterisk BoRiS (~boris@S0106004f4e0ffc6a.wp.shawcable.net)
05:21.54Fplyou dial what ever number you want
05:22.07Nuggetyou can skip all that and just dial the number you want to begin with, you know.
05:22.14mindfoxFpl, it's the digit-timeout
05:22.28Fplmindfox ???
05:22.45Fplso, you know what i need to do ? to get the dialtone faster ?
05:22.54Fpli just would like to set it like this
05:23.03mindfoxThere is a timeout setting for the Asterisk, that waits for *** secs, in order to give you enough time to provide the dial-number you wish
05:23.12mindfoxwhen that time expires, it dials the given number
05:23.13Fpldial 9 to get access line and the XXX XXXX for the number that one wants to dial
05:23.26NuggetFpl: there's absolutely no need to get a second dial tone to do that.
05:23.41FplNugget : i see.
05:23.53Fplwhat will be the right way to do it ? then ?
05:24.05Nuggetjust make an extension that's 9NXXNXXXX and then pass it straight to a Zap/# dial command with the 9 stripped off.
05:24.07Fplmaybe tie timeout is what is killimg me
05:24.10mindfoxYou can dial all the number and when you finish, press #
05:25.17mindfoxFpl, is there a specific reason you want to get the second dialtone?
05:25.38Fplmindfox : no, no
05:25.46mindfoxNugget, I expect something in return of the insult you made towards me... Since it proves I was in the right way, don't you think?
05:25.46Fplmindfox : no, not at all
05:25.51mindfoxok Fpl
05:26.06Nuggetexten => _9NXXXXXX,1,Dial(Zap/4/${EXTEN:1})
05:26.08mindfoxjust follow the provided sample configs and you'll be just fine
05:26.10Nugget^ That's all you need
05:26.22mindfoxJust like what Nugget wrote
05:26.41Nuggetwell, and exten => _91NXXNXXXXXX,1,Dial(Zap/4/${EXTEN:1}) for 11-digit dialing too.
05:26.44mindfoxpay attention to the number of X's you are using
05:27.01mindfoxfor each X it's one digit. So you'll have to configure to the full length of the number
05:27.08Fplthis is what i have exten => _55,1,Dial(Zap/1/)
05:27.18mindfoxof use '.' for wildcard of everything without length limit
05:27.18Nuggetyou don't need to do it in two stages like that.
05:27.21mindfox*or
05:27.23*** join/#asterisk alphaque (~Alphaque@218.111.155.110)
05:27.51mindfoxFpl, you must first plan well, what you want to dial and how it will be served by Asterisk
05:27.53Nuggetif you'd rather use 55 to route to the zap channel, just make the extension _55NXXXXXX instead of _9NXXXXXX
05:28.05Nuggetyou're trying to do it too manually.
05:28.11Fplbut i guess i am completely wrong,
05:28.19Nuggetyou're just making it harder on yourself
05:28.27Fplwhat about stripping the 55 ?
05:28.37Fpldo i need the exten:1 in the end ?
05:28.39znoGanyone from .AR here?
05:28.48mindfoxreplace the :1 with :2
05:28.48Nuggetthe :1 tells it to strip off the first 1 characters.
05:28.53Nuggetto strip off 55, make it :2
05:29.12Fplwaoo.
05:29.16Fplcool
05:29.24Nuggetfor instance, I have a dial plan exten => _1512NXXXXXX,1,Dial(Zap/4/${EXTEN:4})
05:29.38Nuggetthat way if someone tries to dial 1-512-number (local for me) it only dials the seven digits.
05:29.45Nuggetsince 1-512 is unneeded here
05:29.50Fplcool..
05:29.57Fplawesome
05:30.06VetoNugget: or dial 1-830 or 1-210 (your Austin, ehh?)
05:30.08Fpli will change it right now.
05:30.16mindfoxso Nugget, was I wrong in the first place?
05:30.47Nuggetyou were just guessing, just like with your ridiculous attempt to sound like you knew about routing.
05:30.58Nuggetguessing is fine, but stop posturing.
05:31.00mindfoxpathetic. pitty :(
05:31.12Nuggetit's not helpful.
05:31.18Brixiusfpl's first question was more about ip routing it sounded to me then * routing
05:31.24*** join/#asterisk Ridgeback (~Ridgeback@ppp56-33.lns1.adl2.internode.on.net)
05:31.28mindfoxYou can never know what is the knowledge of one typing...
05:31.42Nuggetoh yes you can.
05:31.48mindfoxand I never attacked your knowledge you know... I was just trying to understand why you attacked
05:31.59mindfoxbut... hey. That's how some ppl react, right?
05:32.03Ridgebackhello,  quick question about call parking context [parkedcalls]   need an example of what to put in that context...
05:32.11mindfoxif you never did double natting before, that's understandable
05:32.16MocWhat you think of a * Network Protocol ? http://www.voip-info.org/tiki-index.php?page=Asterisk+Network+Protocol
05:32.19Fplthanks.. to all.
05:32.19mindfoxbut that doesn't mean that it can't be done
05:32.29Fpli still need to keep on learning
05:32.38mindfoxall of us do, Fpl
05:32.42mindfoxthere is no end in knowledge :)
05:33.05mindfox*well, some of us at least. There are others who knows it all, right Nugget?
05:33.14Ridgebackanyone have call parking working?
05:33.16BrixiusThe only time you stop learning is when you slow down to push up the daisy's
05:33.41mindfoxah, wise words Brixius :)
05:34.01NuggetI never claimed to know it all.  But you'll notice that when a subject comes up that I'm unfamiliar with, I usually keep my mouth shut.  You might want to try the same approach.
05:34.21mindfoxRidgeback, I haven't use it since it was renamed to features.conf, but ask anyway :)
05:34.23b0r1qu4can anyone suggest a valid asterisk regexp for describing extensions 3800-3999?
05:34.39Nuggettrying to pose like you know more than you do is unhelpful and adds confusion.
05:34.47FplMindfox and Nugget : you guys made and solve my night !!! i really appreciate it
05:34.54Brixius3[89]xx I think will work
05:35.13Ridgebackthe call parking wiki says you need to include => parkedcalls  (i did that)  but you need to put some parked extensions stuff in there.  any ideas?
05:35.14mindfoxJust stop talking Nugget and if you have some proof of concept, provide it now. Please, enough already
05:35.17FplMindfox and Nugget : it works wonderfull + you solved me a great problem
05:35.18Ridgebackthere were no examples per se
05:35.26b0r1qu4Brixius I tried that, didn't seem to like it
05:35.48b0r1qu4I should clarify that I'm trying to do a comparision within GotoIf
05:36.45FplMindfox, what instruction should i put below the exten => _9NXXXXXX,1,Dial(Zap/4/${EXTEN:1}) so that it hangsup the extension as soon as i hangit up so that it doesnt give a congestions message ?
05:37.12NuggetFpl: exten => _9NXXXXXX,2,Hangup()
05:37.14b0r1qu4I also tried: 3[8-9][0-9][0-9]
05:37.30mindfox??? But, if you hang the phone, it should hang also at the Asterisk box...
05:37.37mindfoxdon't know exactly what you are asking
05:39.05Fplmindfox : whenever i hang up, and i pick the phone again to dial again, i get a congestion message ( i think i am faster then the * box ) in terms of dialing
05:39.20Fpli dont have a exten => _9NXXXXXX,2,Hangup() string like this
05:39.51Brixiusb0r1qu4: how about var1 > 3800 | var1 < 3999
05:39.57mindfoxwell, you could put it in your dialplan for the sakes of sanity, but in normal operation, you wouldn't have to
05:40.06Fplok
05:40.07mindfox*put what Nugget wrote
05:40.24mindfoxah, wait Fpl
05:40.32Fplok
05:40.47mindfoxif you hangup quickly and pickit up again, it would be like *flashing*
05:40.55mindfoxthat means that maybe you are putting the call on hold
05:41.03mindfoxso, don't do it very fast
05:41.06Nuggetthat sounds plausible.
05:41.19mindfoxwait a couple of secs before you pick up again
05:41.25Moonwickicky linux.  :)
05:41.39NuggetI've got a set here, depending on how long that download will take, moon.  :)
05:42.01MoonwickI'm in my boxers, I'll just wait for the dl.  but thanks.  :)
05:42.06mindfoxNugget, I'm from Greece and my English are bad. Plausible?
05:42.20Brixiusb0r1qu4: or var1 : 3[89][0-9][0-9]   These are comming from http://www.voip-info.org/wiki-Asterisk+Expressions
05:42.21b0r1qu4Brixius: that works!
05:42.21Nuggetmindfox: it's a good theory
05:42.41mindfoxah, thanks for the explanation. Never heard or saw that word before
05:43.09mindfoxbye Moonwick
05:43.47mindfoxmind if I ask a question also about the SCCP channel?
05:43.56mindfoxanyone familiar with the subject?
05:44.16znoGtotally OT question: anyone have a clue how much a mini DSLAM is worth? (12 port or so)
05:44.51Brixiusb0r1qu4: actually if your going to use this change it to  var1 >= 3800 | var1 <= 3999
05:44.55mindfoxcheck prices on pricegrabber.com. I bet they have for major companies :)
05:47.28*** join/#asterisk d1ng0 (~dingo@130.205.8.67.cfl.rr.com)
05:47.40d1ng0polycom question ...
05:47.58|Vulture|?
05:48.10d1ng0anyone know how to get it to dial ext...passwd by hitting the voice mail button ?
05:48.26|Vulture|ext....passwd?
05:48.27d1ng0and notify me when there is a new voicemail
05:48.30|Vulture|voicemail ext?
05:48.33|Vulture|yea 1s lemme link you
05:49.00|Vulture|http://www.voip-info.org/wiki-Getting+MWI+on+Polycom+Phones+to+work+with+Asterisk
05:49.18MocPolycom rock !! ;)
05:49.27|Vulture|Moc: you know it
05:49.31|Vulture|you get your 600?
05:49.41Mocnope :(
05:49.44|Vulture|:(
05:49.51|Vulture|I want a review
05:49.59Mocthey should receive it the 10.. so 1 or 2 day before I get it I guess
05:50.05BrixiusDoes anyone know, does * support native transfer's between servers?
05:50.29|Vulture|Brixius: if your servers are linked through IAX?
05:50.54MocI should get a page on the wiki with the polycom features
05:51.17|Vulture|Moc: only 1 :P
05:51.23Brixiusok, just if they are linked through iax or do they have to use the "switch" configuration in iax.conf to do it?
05:52.17d1ng0okay wait, when i hit voicemail button asterisk says enter mailbox number, then asks for the password, can i have the polycom just enter them for me ?
05:52.28|Vulture|Brixius: should work if they are just linked via iax and you have the on different local extension banks... say 1** for server1 and 2** for server2 just transfer a call from server1 to ext. 205
05:52.47|Vulture|tell the server to send all 2** requests to IAX channel for server2
05:53.13shmoozfor DTMF which delivers better call quality (SIP)  INFO or RFC2833 ?
05:53.47FplMindbox : good night and thanks, you really alliviated a problem i had
05:53.55Brixiusok, but I guess what I meant to say, maybe native xfer wasn't the right term, if let's say 100 xfers to 200, then 200 xfers to 101, is the call now looped through the other server
05:54.03FplMindfox : good night and thanks..
05:54.12mindfoxbye Fpl
05:54.12mindfoxGlad I could help :)
05:54.24|Vulture|Brixius: oh you want like a blind xfer through servers... I duno if you can do that
05:54.24Fplnow, i am going to be
05:54.25Fplbed
05:54.30Fplit is 2:00am here
05:54.35Fplbut is what worth the time and wait
05:54.44mindfox:)
05:54.48Brixiusok
05:54.55|Vulture|Brixius: unless you have all calls going into 1 server and getting passed to multiple servers
05:55.00Fplnow, i will be able to control my home pbx even more, with the little tricks !
05:55.16Fplthanks, you were right, there was no need for second stage dialing
05:55.16|Vulture|Brixius: because then the main server would handle all the xfers
05:55.18mindfoxJust check the voip-info.org. Tons of info there Fpl
05:55.22Fpl;-)
05:55.49Fpli check it most of the times, but sometimes i feel owerwe....with the amount of information
05:56.01shmoozfor DTMF which delivers better call quality (SIP)  INFO or RFC2833 ?
05:56.09mindfoxFollow your own pace and you'll manage just fine
05:56.48mindfoxshmmooz, don't know if there is a comparison for that. I know that not all devices supports all formats, if that's what you are asking
05:56.49Brixiusthat's kindof what we have now, all calls come into one * box via t1 then they go to 2 others depending on the location.  Sometimes a call could get xfered back and forth a couple of times
05:57.24PenfoldFpl: know how you feel: abou 98.5% of the traffic on this channel goes 'whoosh' clean over my head at present.
05:57.26shmoozwell I have this ata486 which can be set to audio   SIP INFO   or RFC2833
05:57.41|Vulture|Brixius: yea that should be possible.. but Ive never done it so I couldn't help
05:57.57mindfoxata486? don't know that device. It's not Cisco, is it?
05:58.05Brixiusok thanks
05:58.16Brixiusno the ata486 is a barbietone device
05:58.16*** join/#asterisk clive- (~pirch@myw-stp-66-18-80-129.sentechsa.net)
05:58.18shmoozperhaps call quality is the wrong item impacted the Dial Tone Modulation is really how well the phone sets numbers are translated right?
05:58.26mindfoxoh :)
05:58.27shmoozgrandstream
05:58.38mindfoxshmooz, can't you tell from the specs?
05:58.59BrixiusI think you want to set it to SIP INFO
05:59.44shmoozit supports the 3 of them but I having never really used it before I don't know which one works better with asterisk
06:00.22Brixiusshmooz, call quality would be more a function of the codec you are using, not dtmf settings.
06:00.47Moc|Vulture|, your still here ?
06:00.51|Vulture|yea
06:00.53Mochttp://www.voip-info.org/tiki-index.php?page=Polycom+Phones
06:00.55Moccheck
06:01.01shmoozcan any give a nutshell about the practical difference for the caller between SIP INO and RFC2833 ?
06:01.16mindfoxk, time for me to go to work. Bye all and thanks for the company :)
06:01.25MocI added a few of the feature that I think polycom differ from cisco
06:01.26|Vulture|CAVEATS: is new
06:01.42Moccheck at the top
06:01.56|Vulture|oh nice, did you just add that?
06:02.06Mocyes
06:02.23Brixiusshmooz: http://www.voip-info.org/wiki-SIP+DTMF+Signalling
06:02.28|Vulture|I don't see any reason people would want 79xx anymore other than Cisco branding
06:02.33Ridgebackis it possible to change the CODEC per call? ie. if DEST# = xxxx  then CODEC=iLBC??   I tried SIP_CODEC=ilbc  but it still wouldnt work
06:03.06|Vulture|my IP500s have proven themselves, and above that I can use a wide variety of headsets on them whereas 79xx is limited
06:03.30Moc|Vulture|, well here what cisco have more.  Sound can be set louder, the display is bigger( but not as easy to read), the phone look better, angle of phone can be changed
06:03.42*** part/#asterisk Fpl (~Fpl@200.93.39.123)
06:04.13Mocboot faster
06:04.25Mochas a telnet interface
06:04.33Mocforgot to add the upload log feature of polycom
06:05.04|Vulture|but ip's have tftp/ftp and I like ftp over tftp
06:05.20*** join/#asterisk dagrim (0@12-218-72-56.client.mchsi.com)
06:05.24Mocme too
06:05.35dagrimhey all
06:05.35Moci added the log upload feature to the page too
06:05.51dagrimanybody here run shoutcast.. and stream off of the same linux box?
06:05.52|Vulture|oh and Ive confirmed the ability to upgrade h323 to SIP
06:06.08MocI mean that 17 more 'big' feature compare to the cisco..
06:07.02d1ng0okay so how do i reboot an ip500 ?
06:07.03*** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc)
06:07.22Mocdln0, vol - + Hold Message
06:07.43Mocyou need to press those 4 button for a few secondes
06:08.19d1ng0message ???
06:08.35d1ng0i see the vol + - and the hold button
06:08.38Mocvoicemail
06:08.42d1ng0ahhh
06:10.47d1ng0so is the red light flashing cuz there is voicemail ?
06:10.59d1ng0or does it always flash
06:11.14Mocyour on do not disturb no ?
06:11.16grant_AIts the tornado alarm
06:11.24grant_Ameans theres a tornado
06:11.26d1ng0no
06:11.28grant_AGet the fuck outta dodge
06:11.31Mocdonno then
06:12.03Mocgood, finally Ive put it on the wiki, so I dont have to remember the list of why polycom phone are better than cisco hehe
06:12.03d1ng0so Moc when i hit voicemail and it dials 888 can i get it to dial my extension and password also ?
06:12.32Mocwell, normally, you should create a * extention that allow auto login from your callerid
06:12.35Mocthat is what I do
06:13.04Mocmy Caller ID -> Caller Name app work very well... too cool
06:13.25*** join/#asterisk florz (nobody@p508A67A1.dip.t-dialin.net)
06:13.29*** join/#asterisk los415 (~los415@ssf-office.corp.race.com)
06:13.50d1ng0Moc,  a * extention uhhh ???/
06:14.22shmoozok now I see it straighter,    the ATA486 will send DTMF via  Audio   RTP   or  SIP INFO   which is the best mode to use for dtmf??
06:14.40Mocexten => 855,3,VoicemailMain(s${CALLERIDNUM}@tmc)
06:15.04Mocthe (s <--- wont ask me for a password
06:15.22Mocbe sure only your trusted phones can call that extention
06:15.30d1ng0heheh yeah the light lights up when new voicemail arrives :)
06:15.32Mocsomeone on inet could fake a Cid
06:15.58Mocanyway, it sleep time
06:17.26shmoozI do seem to be having trouble with the Number Dialing sounds being passed accurately to the PSTN line (d100P)
06:19.51Brixiusshmooz: I'd use sip info and let * generate the dtmf to your pstn instead of encoding it in audio, if I remember right that's what worked best for me
06:20.21*** join/#asterisk xamodoug (~doug@ssf-office.corp.race.com)
06:20.25xamodoughey wassup
06:22.17shmoozI was just taking another read and I got the sense that RTP would deliver tones from hand set to asterisk more faithfully   but I  do like the idea of * actually doing the tone generation.
06:23.11shmoozExcept that I just recalled there was some difficulty useing the handset on the ATA486 to retrieve voicemail because the number tones were misinterpreted.
06:23.51shmoozthere's dollar store phone wire extensions all over this place it's kinda scarry.
06:24.20shmoozI had no idea until I plugged the pstn line into * server
06:25.28shmoozwell I have a much better understanding of different expectations for the DTMF modes
06:25.58shmoozI'm try ing to compel them to actually pay me to rewire the whoe palce with cat 5
06:31.55bkw_POS INTERNET
06:31.57bkw_FUCK THE INTERNET
06:33.30d1ng0well damn
06:33.36twistedbkw_, take a chill pill man
06:33.44d1ng03 more of these polycoms and the whole house will be wired
06:33.57drumkillaso emotional ...
06:34.18twisteddrumkilla, my net stabalized, best I can tell
06:34.23drumkillagood stuff
06:34.31mishehuthe internets are acting up again?
06:34.37drumkillastupid internets ...
06:34.45drumkillai need to switch internets
06:34.52drumkillamine is bad
06:35.13twistedoh, btw, last night drumkilla, was scheduled maintenance
06:35.16twistedi read it on the wbsite
06:35.17Penfoldinternet--
06:35.34twistedanywho bedtime
06:35.34drumkillai better get a discount for that
06:39.09*** join/#asterisk dustbunnyman (~sable@CPE000625a6a206-CM013349900582.cpe.net.cable.rogers.com)
06:41.49znoGhow much is a channel bank worth? who sells them?
06:42.18clive-znog ebay
06:42.56znoGi was just headed there... :)
06:43.31znoGdamn, more expensive than i thought
06:44.41dustbunnymanI just wanna little asterisk server so 12 people can do voip .... I need an outside IP address and a firewall that will let me through?
06:44.59dustbunnymanthey all say "just use skype!" but I wanna do the real thing
06:45.19drumkilladustbunnyman: good man!
06:45.53dustbunnymanI should have 2 grandstream phones and 4 QuickJack cards (which I can plug and phone into and then connect the computer to the LAN??)
06:45.59dustbunnymanat my disposal
06:46.12dustbunnymanbut no ability to control my firewall :-(
06:46.24dustbunnyman2 week per request to open a port etc.
06:46.30dustbunnymanam I doomed?
06:46.44znoGfor 2 weeks, yes
06:46.53d1ng0bahhhh
06:47.09d1ng0why does my polycom seem to time out and not ring incoming calls
06:47.12*** join/#asterisk ingenius (~syntax@200.73.175.252)
06:47.16ingeniusHi!
06:47.43ingeniusi'm looking for store tu buy SPA-2000 in toronto canada .. somebody know where ?
06:47.52ingeniuss/tu/to
06:48.31clive-ingenius atacomm.com
06:49.32Atacommroflmao, i thought at first you were calling me ingenius
06:50.54ingeniusclive-: is this local store ?
06:51.34Atacommingenius: we are based out of Minnesota
06:51.52ingeniusapps i need local store ... thanks :)
06:55.57blitzragedrumkilla: evening
06:56.24drumkillablitzrage: hey there
06:56.29blitzragedrumkilla: how goes?
06:56.37drumkillait's ok
06:56.48drumkillabeen working on some guy's asterisk server all night
06:56.50drumkillabut it works now
06:57.06drumkillait got a rootkit  :)
06:57.21blitzragecoo, I've been working on a conference announced join script
06:57.24blitzragefinally works
06:57.29Atacomm~seen implicit
06:57.30jbotimplicit <~implicit@dhcp-250111.mobile.uci.edu> was last seen on IRC in channel #asterisk, 1d 3h 22m 17s ago, saying: 'hi file[laptop]'.
06:57.31*** join/#asterisk andrew` (~andrew@adsl-67-119-26-246.dsl.snfc21.pacbell.net)
06:57.36drumkillaannounced join script, huh?
06:57.38blitzragedrumkilla: you got 2 seconds to give it a shot?
06:57.40drumkillasounds like something i was working on
06:57.42drumkillasure
06:57.55blitzrageIAX2/guest@pbx.leifmadsen.com/5050
06:58.10blitzrageconf 5050
06:58.28*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
07:03.07mishehubkw_: ssshhhhh!  people are trying to sleep
07:03.10mishehuheh
07:03.48d1ng0okay guys how do i keep a polycom from timing out
07:04.09d1ng0asterisk shows its registered but it aint ringing all the time
07:04.23bkw_no cogent fucking barfed for like 30 min
07:04.33blitzrageblah
07:05.16d1ng0grrrr retrans_pkt: Maximum retries exceeded on call
07:05.56Atacommlol, bkw, what??
07:07.04d1ng0damn i swear this phone goes to sleep or something
07:07.35d1ng0lets me dial out no problem, but inbound calls never really get to the phone so it doesnt ring
07:08.43d1ng0is anyone alive in here ?  :)
07:10.02*** join/#asterisk jontow (~jontow@eagle.bsd.st)
07:12.03*** join/#asterisk jontow (~jontow@eagle.bsd.st)
07:22.24wiz8291sip*CLI>
07:22.46luckyali~cli
07:22.47jbotcli is, like, a Command Line Interface, the best form of interface around, of course  Call-Level Interface, originally developed by SQL Access Group, but now known as Microsoft's ODBC standard. an x86 assembly instruction  Common Language Infrastructure (See mono or .net)
07:23.36Nuggetd1ng0: are you using qualify on the device?
07:23.54*** join/#asterisk invi_ (~undisclos@dsl-cap-209-5-169-204-cgy.nucleus.com)
07:25.25*** join/#asterisk bilo (nabeelj@69.158.63.195)
07:25.49biloguys, i have a gentoo/asterisk question, if anybody's still around at this hour
07:25.49invi_any canucks here?
07:25.52biloyeah!
07:25.54biloin toronto
07:25.58Nugget~data
07:25.59jbotDon't Ask To Ask. Just ASK
07:26.04bilolol
07:26.14bilook, i have asterisk running on a home gentoo server just fine
07:26.38biloi signed up for a really cheap ded server running gentoo as well
07:26.52invi_canucks: whats the best place to buy sipura from?
07:27.03biloemerge for asterisk-0.9 worked fine but for asterisk-1.1 didn't
07:27.08biloso i CVSed instead
07:27.27biloeverything's great... but when i run asterisk, port 5060 on the machine does not respond
07:27.37Nuggetdid you checkout HEAD or stable?
07:27.45biloumm
07:27.47biloHEAD i think
07:27.49Nuggetand what do you mean by "does not respond"?
07:27.55drumkilla~asterisk 1.0
07:27.56jboti guess asterisk stable is export CVSROOT=:pserver:anoncvs:anoncvs@cvs.digium.com:/usr/cvsroot && cvs checkout -r v1-0 asterisk
07:27.58NuggetHEAD is kind of messy right now.  You should use stable.
07:28.03bilobut the .conf's are the same on both machines, except someone else set up the kernel on the other machine
07:28.20Nuggetdo what jbot says.  :)
07:28.47biloactually, i have a feeling something might prevent port 5060 from responding
07:28.58bilobut there doesn't seem to iptables or firewall running
07:29.00Nuggethow are you determining that port 5060 is not responding?
07:29.06bilotelnet localhost 5060 ;)
07:29.10Nuggetthat won't work.
07:29.14Nuggetit's udp not tcp.
07:29.18biloah
07:29.39biloi have sip debug on and nothing shows up on the ded server when x-lite tries to connect
07:29.41biloNOTHING
07:30.00bilomethinks there's a firewall somewhere but i see nothing set up on the gentoo server
07:30.22Nuggettry a "netstat -a | grep udp" on the box.
07:30.27Nuggetyou should get a line for 5060
07:30.31Nuggetudp        0      0 *:5060                  *:*
07:30.31SlimeyIs there a SIP ping? I should probably integrate asterisk into my monitoring system
07:30.43invi_k, anybody > whats the best place to buy sipura from? voxilla any good?
07:30.55*** join/#asterisk michael12345 (~mick@staff.tsn.cc)
07:30.59biloudp        0      0 *:5060                  *:*
07:31.12bilothat means it's open and running?
07:31.20Nuggetit means that asterisk is working.
07:31.24Slimeybilo: That looks good
07:31.24michael12345is there anyway to give our helpdesk access to a log that shows why people are failing to authenticate
07:31.33biloinvi: i bought it from voipsupply in buffalo
07:31.34michael12345preferably db
07:31.35Nuggetit could still be blocked by a firewall on the box or somewhere in between you and the box.
07:31.56bilonugget: i see no firewall running - what should i be looking for?
07:32.21NuggetI'm totally clueless about linux packet filtering.  it changes so often.  :)
07:32.46Nuggetit was ipfw, then ipchains, then there was a tables thing.  I have no idea what the popular one is these days
07:33.04bilobut if i do ps -a, i should see it running?
07:33.07Nuggetno
07:33.11Nuggetit would be a kernel thing
07:33.14florziptables is current in 2.4 and 2.6
07:33.25bilonugget: ah
07:33.32bilosorry for my linux ignorance
07:33.48biloanyway, where would the config for it be? running a 2.6 kernel
07:34.07biloand would i need to restart anything to get the new settings to take effect
07:34.21florzthat depends on the distro
07:34.26*** join/#asterisk ahqiang (~ahqiang@202.83.101.135)
07:34.27Nuggetif it's iptables, you control it using the iptables command.
07:34.43Nuggetsomewhere in the rc directories there presumably exists a script which might be calling iptables.
07:34.53d1ng0Nugget, not that i know of, where can i check
07:35.08Nuggetdunno, I'm unfamiliar with gentoo.
07:35.08biloi'm using gentoo
07:35.28luckyaliunload iptables to check
07:35.31bilonetmount being called at default runlevel
07:35.45bilono iptables in /etc/init.d/
07:36.56ahqiangto unload ip tables
07:36.59d1ng0here Nugget Nugget Nugget :)
07:37.03Nuggetheh
07:37.12ahqianggo /etc/rc.d/init.d/iptable stop ?
07:37.24d1ng0Nugget, so where can i check for qulaify settings for this polycom
07:37.29bilogentoo keeps em in /etc/init.d
07:37.33ahqiangoh
07:37.36bilobut there is no iptable or iptables
07:37.42Nuggetd1ng0: put a "qualify=100" line in sip.conf for that device.
07:37.43ahqiangipchains?
07:37.50biloip-nothing ;)
07:37.55ahqianglol
07:37.56ahqiang.
07:37.58luckyalirmmod iptables
07:38.04ahqiangi redhat 9
07:38.07ahqiangsorri
07:38.10ahqiangcant be of help
07:38.36bilonothing that even sounds like ip-anything
07:39.21d1ng0Nugget, sip_poke_noanswer: Peer 'scottsip' is now UNREACHABLE!  Last qualify: 0
07:39.29d1ng0Nugget, reboot phone right
07:39.43Nuggetreload asterisk and reboot the phone.
07:39.57Nuggetsounds like asterisk isn't able to reach the phone reliably (or at all)
07:40.00biloany way to check if the port is open thru localhost?
07:40.14luckyalinetcat
07:40.15Nuggetbilo: that's exactly what that netstat command did.
07:40.16biloto see if it's a software firewall or a hardware firewall somehwere
07:40.25d1ng0Nugget, nat/firewalled in front of it
07:40.32bilonugget: so it's not a software firewall for sure?
07:40.40Nuggetd1ng0: you have nat=yes in sip.conf?
07:40.44d1ng0yupp
07:40.47Nugget*nod*
07:40.57blitzrageg'night all!
07:40.59Nuggetare there multiple VoIP devices behind that same nat?
07:41.05Nugget(talking to you)
07:41.23bilonugget: sorry, was that nod for me?
07:41.29Nuggetbilo: no.  :)
07:41.45Nuggetbilo: you haven't ruled out a software firewall.
07:41.48bilook
07:41.54bilolemme go look for #gentoo
07:42.03Nuggetgood plan, sorry we're so worthless.  :)
07:42.11bilothanks
07:42.20Nuggetif you were on freebsd I could *totally* help!
07:43.11luckyalidoes it telnet other than 5060 ?
07:43.22Nuggetyou cannot telnet to a udp port.
07:43.52luckyalinot the udps
07:43.56d1ng0Nugget, nope just one
07:44.20d1ng0Nugget, and sip show peers shows LAGGED (158 ms)
07:44.23Nuggetd1ng0: hrm.  some NAT gets really confused if there are two similar clients.
07:44.28Nuggetd1ng0: that's good!
07:44.35d1ng0its a openbsd pf firewall
07:44.37Nuggetchange the qualify to 200 instead or 100.
07:44.42Nuggets/or/of/
07:44.50luckyalitelnet fwd.pulver.com 5060 Works !
07:45.07Nuggetthat'll probably fix it, d1ng0.  having qualify on will generate enough traffic that the NAT won't drop the route.
07:45.17bilolucky: that's what i tried first
07:45.20Nuggetmost likely the NAT was dropping the connection from inactivity, making calls fail after
07:45.43d1ng0Nugget, OK (158 ms)
07:45.47d1ng0:)
07:45.51Nuggetspiffy
07:45.54d1ng0its ringing now
07:45.59luckyalidoes it telnet on default port ?
07:46.00d1ng0weeeeeeeeeeeeeeeee
07:46.37d1ng0Nugget, im on cable so it set it for 250ms
07:46.41Nugget*nod*
07:48.29d1ng0heheheh 10 calls every one rang
07:49.37*** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc)
07:49.37*** topic/#asterisk is Asterisk: The Open Source PBX || DUNDi (http://www.dundi.com) #dundi || Please "make clean" or do a fresh checkout before reporting any bugs.