irclog2html for #asterisk on 20050307

00:00.24*** join/#asterisk tzafrir (~tzafrir@62.90.10.53) [NETSPLIT VICTIM]
00:00.31*** join/#asterisk aggelos (~aggelos@egate.eleven.de) [NETSPLIT VICTIM]
00:00.40*** join/#asterisk dg1nsw (~schulte@gate.sympat.de) [NETSPLIT VICTIM]
00:02.29tuxinator_linuxbrc__: cool video's
00:02.47brc__PTG123, eh?
00:02.57brc__exten is a channel variable
00:03.03*** join/#asterisk BigCanOfTuna (~chatzilla@dsl-macn-66-18-205-30-cgy.nucleus.com)
00:03.04brc__so it'll stay with the channel wherever it goes
00:04.04BigCanOfTunaIs there an API for asterisk that will allow me to issue commands to it via another programming language such as Python or Ruby?
00:04.16brc__well there's a long answer
00:04.18brc__and a short answer
00:04.57hermie...and the short answer is AGI
00:05.06brc__the long answer is it depends
00:05.17brc__on what you mean by 'issue commands'
00:05.35harryvvhi brc
00:05.36hermieI was thinking the long answer was "pick a better programming language" :-)
00:05.41brc__greets harry
00:05.59brc__ruby rulez
00:06.07mrgobyyeah, like php
00:06.11brc__EWW
00:06.18hermiemrgoby: no, that'd be even worse
00:06.31mrgoby:-)
00:06.52hermieat least python and ruby were meant to be run
00:06.52brc__keep that where it belongs, short web scripts
00:06.52brc__anyway
00:06.52brc__BigCanOfTuna, did you disappear?
00:06.55BigCanOfTunabrc_: sorry, in the background.
00:07.03*** part/#asterisk GaryH (~ghawkins@gromit.garysoft.co.uk)
00:07.08brc__if you'd like to give a short description of what you want to do maybe somebody can answer the question
00:07.28harryvvbrc, I had a bad experaince last night...hard drive was cycling over and over last night..ie failing. It was my asterisk system.
00:07.28brc__my as in home? or production somewhere
00:07.43harryvvit was my test box running our phones.
00:07.49brc__ah
00:07.49BigCanOfTunaI am planning on creating a web interface to issue command to my asterisk server.
00:07.52harryvvluckily it was internal.
00:07.53brc__had backups?
00:08.02tuxinator_linuxharryvv: I had what I thought was a hard drive problem but it was that my TDP400P was sharing and IRQ
00:08.07harryvvyes, have the confs on the windows machine but not current.
00:08.07brc__BigCanOfTuna, gotta be more specific buddy
00:08.27brc__manager will *probably* do what you want
00:08.29mrgobyuuuum
00:08.43BigCanOfTunabrc__: I would like to invoke certain tasks, such as giving it a number, and asterisk calls it.
00:08.44mrgobyany reasy why my asterisk seg faults when i have an mp3 in my mohmp3 dir ?
00:08.44tuxinator_linuxharryvv: What do you mean by cycling?
00:08.54brc__you can list channels, redirect em, create em, etc
00:09.03harryvvtux, no this is possibly a hard drive failure. Got this message "hda :dma timer_expiry dma stats = 0x21 I tried to fsck the drive and did not work.
00:09.09mrgobys/reasy/reason/g
00:09.22tuxinator_linuxharryvv: I got a very simular error
00:09.26BigCanOfTunabrc__: Isn't AGI asterisk calling external scripts?
00:09.34tuxinator_linuxharryvv: except mine was 0X24
00:09.34brc__yes, agi is not what you want
00:09.37mrgobympg123 must be screwed up
00:09.39*** join/#asterisk dcb (~dcb@CPE-60-231-180-246.vic.bigpond.net.au)
00:09.42harryvvtux, as if it was rebooting over and over but the system was running..my wife was on the phone with voip running but the hard drive kept cycling.
00:09.43BigCanOfTunabrc__: I want external scripts to call the asterisk api.
00:09.47brc__there is a bit of info on the manager interface on the wiki, ti's what you want
00:09.48mrgobyi bet they used the default gentoo moh
00:10.06BigCanOfTunabrc__: thanks.
00:10.25brc__well you can't exactly call the asterisk api unfortuantly..
00:10.30*** part/#asterisk dcb (~dcb@CPE-60-231-180-246.vic.bigpond.net.au)
00:10.35mrgobyi saw there is a make target for mpg123 now...  is that recommended ?
00:10.46brc__yes
00:10.57brc__*highly*
00:11.03mrgobydoes it install itself ?
00:11.06brc__yes
05:12.03*** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc)
05:12.03*** topic/#asterisk is Asterisk: The Open Source PBX || 1.0.6 Released || Dev Conf 1PM CST MARCH 10th -> IAX2/guest@66.250.68.194/996 || ClueCon Dev Conf Aug 3rd - 5th
05:12.05brc_like he said, for the fun of it
05:12.19holycowehe :) fair enough
05:12.57Tr0j4NI guess Asterisk doesn't like it when you service zaptel stop while it's running. Is a Kernel panic bad??
05:13.03sivanain a bottle?.... Brilliant!
05:13.41MikeJ[Jayden]so I can say... asterisk runs on windows, some other guy did it, I just was testing... it has some serious issues with fork, cuz of the way cygwin does it
05:13.47rhollanOT: Grr... wife wants me to go get groceries.... She should be grateful I made Drunkard Noodles for dinner.
05:13.48brc_UNPOSSIBLE!
05:13.51MikeJ[Jayden]but you can make calls
05:13.58holycowoh so it doesn't run on windows
05:14.03holycowit runs on cygwin
05:14.04brc_yes it does
05:14.04holycow*nod*
05:14.18brc_that is different from using colinux
05:14.27holycownot by much
05:14.34geekstercan anyone help me out with an extension issue that im having ?
05:14.35MikeJ[Jayden]sure it is.
05:14.45brc_yes it is holycow
05:15.02rhollanCYGWIN is a commpatibility layer over Win32, IIRC, not not a virtual machine emulator.
05:18.28erwinismMikeJ[Jayden] i cant find manual how to add a user on the links you said
05:18.39*** join/#asterisk pimpwell (~pimpwell@ool-44c6ab45.dyn.optonline.net)
05:18.52brc_erwinism, a 'user'?
05:19.07brc_~asterisk docs project
05:19.11brc_~asterisk docs
05:19.13jbotrumour has it, asterisk documentation project is at http://asteriskdocs.org
05:19.22brc_read
05:19.43*** join/#asterisk Damin_Mobile (~pocketirc@252.sub-70-214-4.myvzw.com)
05:19.50erwinismbrc ok
05:19.58pimpwellgood evening, was wondering about the .call file a little bit and in my situation what are my choices.  I place the .call file in the outbound directory through FTP,  anyway to program the .call file to somehow get me a response back on the calls status wether its busy, no one answered or terminated successfully?
05:20.18Damin_MobileThe eagle has landed!
05:20.28pimpwellthe asterisk server is not local to me
05:20.35Damin_MobileWhere is the beer?
05:20.47pimpwellI got 2 22oz of corona in the fridge
05:20.56pimpwelltake em :o
05:20.57Tr0j4Nwhen i run service zaptel start it tells me kobject_register failed for t1xxp (-17)
05:22.41pimpwellcan the .call file do anything special like send an email or something for return report
05:22.42rhollanI've got some Pyramid IPA and Foster's (damn wife.... ugh) in me fridge. :-)
05:23.42MikeJ[Jayden]damin.. you in the bathroom on IRC again :)
05:24.21pimpwellcan I treat the .call file like PERL?
05:24.40pimpwelland add a conditional statement in there or things of that nature?
05:26.31erwinismbrc, im sorry i couldnt find any articles ont he http://asteriskdocs.org related to adding a user. maybe you could direct me to the link.
05:26.31JonR800pimpwell, my suggestion.. use the manager API or use an AGI
05:27.10Damin_MobileMikej; no in a plane
05:27.14JonR800pimpwell: either use the call file to connect the user to the AGI... or user the manager API to initiate the calll
05:27.33pimpwellthe box isnt on my end
05:27.34MikeJ[Jayden]hdhd
05:27.37MikeJ[Jayden]hehe
05:27.37pimpwellit belongs to someone else
05:27.46JonR800k, then use the AGI method
05:27.50pimpwellwhat I need is to FTP my .call file to their box
05:28.10pimpwelland be able to check status on each call
05:28.12scythelxpimpwell: well... you wont get any response from asterisk on the status of the call if you do that
05:28.29pimpwelleverycall gets logged right?
05:28.48pimpwellI figure through CDR
05:28.50scythelxpimpwell: yea.. but you said you dont have access to the box.. unless you wanted to ftp the log files back and parse them
05:29.02pimpwellya, is it 1 log file?
05:29.05pimpwell1 giant one
05:29.10pimpwellor 1 for each call
05:29.17scythelx1 giant one
05:29.27JonR800depends how you set it up..
05:29.32holycowi'll be off for a while, i just dropped by to give a shout out to any asterisk devs that might around
05:29.34scythelxy does it have to be ftp anyways
05:29.38holycowand anyone else in the community
05:29.40*** join/#asterisk The_Ball (~alex@dsl-73.131.240.220.lns02-wick-bne.dsl.comindico.com.au)
05:29.45*** join/#asterisk hemant (hemant@220.226.49.235)
05:29.46holycow:) this is one rocking tool
05:29.49The_Ballwhat port's need to be open for iax2?
05:29.58JerJer[mobile]udp port 4569
05:30.08pimpwellftp because it's not my box, I just make the calls.
05:30.34JonR800what exactly are you trying to accomplish? why do you need a status?
05:30.44pimpwellsimilar to wake up calls
05:30.51pimpwellmy users need to know if the calls went out
05:30.55pimpwelland how many more are left
05:31.04pimpwellthey just cant click a button and be like okay, im done.
05:31.07scythelxwell.. just have the owner of the box give you a manager interface login and your set...
05:31.23pimpwellI am doign this all throuhg php
05:31.26JonR800this is a programming problem.. not an asterisk problem.
05:31.27scythelxso..
05:31.33pimpwelljon:  its both man
05:31.37JonR800not really
05:31.40The_BallJerJer[mobile], ah! udp, no worries
05:31.53scythelxgoogle -> phpagi
05:31.56JonR800the mechanisms to do this are there on the asterisk end.. you just need to tie it together
05:32.13pimpwellI was just thinking the .call file is more then just a file
05:32.15JerJer[mobile]looks like asterisk made slashdot again
05:32.33MikeJ[Jayden]link?
05:32.42scythelxnah it wont parse dynamic arguments as far as i know
05:32.44JonR800pimpwell: nope
05:33.02scythelxyour best bet is the manager interface... seriously
05:33.12JerJer[mobile]www.slashdot.org
05:33.16MikeJ[Jayden]http://it.slashdot.org/it/05/03/06/1945210.shtml?tid=126&tid=218
05:33.22MikeJ[Jayden]thanks....smartass
05:35.35*** join/#asterisk ethzer0 (~ethzer0@d141-238-51.home.cgocable.net)
05:36.17scythelxpimpwell:
05:36.20scythelxoops
05:36.23scythelxpimpwell: http://www.voip-info.org/wiki-Asterisk+tips+Wake-Up+Call+PHP
05:36.36scythelxits alreadly built for you.. just get a manager login... managers.conf
05:38.13JonR800i wrote the ugly perl portion of that
05:38.25pimpwellI know I read that
05:38.30pimpwellbut, I dont have the bandwidth
05:38.33pimpwellI need to use someone elses box
05:38.36The_Ballis the bindaddr= nessesary in iax.conf?
05:38.50scythelxpimpwell: it does.. thru the manager interface... its like a telnet session....
05:38.52pimpwellsecond, I am not doing wake up calls, it's just something my system could do
05:38.56JonR800you don't need the manager interface
05:39.01MikeJ[Jayden]ball, only if you need it
05:39.02JonR800ahh
05:39.26pimpwellI will be using php to run my system, telnetting in wont help me
05:39.37rhollanwell, i gotta go... later ... been fun
05:39.41JonR800the wake up call app uses call files
05:39.45pimpwellI wont be sitting there checking the status of everycall manually
05:39.45*** part/#asterisk hemant (hemant@220.226.49.235)
05:39.49JonR800and what would you need bandwidth for pimpwell?
05:40.01pimpwellfor 6:00am when 100 people want to wake up
05:40.02scythelxJonR800: ah thats right
05:40.06pimpwellat 6:00am
05:40.09joshua_OK thanks guys.
05:40.10*** part/#asterisk joshua_ (joshua@cl-303.ams-04.nl.sixxs.net)
05:40.55*** join/#asterisk j0 (~dan@S010600105a04ed8d.va.shawcable.net)
05:41.00JonR800pimpwell: i don't get it.. why would that use bandwidth.. this box is local to where you're offering the services right?
05:41.07JonR800you just wanted to remotely monitor
05:41.10pimpwellno, it isn't
05:41.13JonR800oh
05:41.19pimpwellit's a remote box, I FTP the .call file to
05:41.28pimpwelldrop it in the outbound.
05:41.28JerJer[mobile]ugg
05:41.36JerJer[mobile]talk about insecure
05:41.38JonR800sorry i thought it was on a customer's premises.
05:41.43brc_SFTP
05:41.44JerJer[mobile]and bloated
05:41.48JerJer[mobile]SCP
05:41.56pimpwelltalk about people trying to rip me off for bandwidth usage
05:42.01brc_ssh+ftp
05:42.05erwinismdo i have to enable the RTP port range on my firewall?
05:42.45pimpwell30 calls simultaneously, 10$ a month  1.2cents a minute
05:42.47JerJer[mobile]depends on your firewall config
05:42.51pimpwellis what Im paying.
05:43.01JerJer[mobile]pimpwell: you are getting cornholed
05:43.04pimpwellahaha
05:43.10pimpwellnot really.
05:43.14erwinismJerJer[mobile] how can i enable it?
05:43.25scythelxpimpwell: yea u are...
05:43.26Damin_MobileJerJer: You arent in san jose.
05:44.13pimpwellhe will also hold my sound files
05:44.32scythelxhe has to if your ftping your .call files in
05:44.36pimpwellso I dont have to collocate for 150 a month or whatever a certain someone in here was going to charge me to throw my box
05:44.38MikeJ[Jayden]pimpwell.. I'll hold your sound files :)
05:44.58JonR800i'll hold something else.
05:45.14pimpwell<enter dick joke here>
05:45.41JonR800i thought i had that covered..
05:45.41MikeJ[Jayden]speaking of moose jokes, where is bkw
05:45.41JonR800doh
05:45.56brc_von
05:46.06JerJer[mobile]Damin_Mobile: no
05:46.07MikeJ[Jayden]y, I know...
05:46.32JerJer[mobile]pimpwell: if you just want to make a phone call, no you don't have to collocate a box
05:46.42pimpwellI dont want to make a phone call
05:46.50pimpwellI have a system, a web based system
05:46.58JerJer[mobile]ok and the problem is?
05:47.15MikeJ[Jayden]well... if you don't want to make phone calls, then this is probably the wrong place to be chatting :)
05:47.17pimpwellusing a remote asterisk box makes checkign the status of each call a pain
05:47.28scythelxhe wants to ftp .call files in and get the status of the call.. but doesnt want to use the manger interfcae because hes concerned about bandwidth?
05:47.50JonR800have a script on the server side monitor it for you
05:48.04pimpwellscythelx:  manager interface requires manually login?
05:48.13*** join/#asterisk potter (~hq28@202.58.252.14)
05:48.34MikeJ[Jayden]you could progamatically communicate w/ manager, it is just text
05:48.36scythelxpimpwell: php script will log in for you, send the dial commands, and return the status
05:48.42scythelxall back to your web based system
05:48.44JonR800MikeJ[Jayden]: bingo.
05:48.44JerJer[mobile]pimpwell: what is the wrong with the manager interface?
05:48.57MikeJ[Jayden]B-I-N-G-O :)
05:49.01pimpwellhmm
05:49.06newpershow many simultaneous phone calls can asterisk handle?  or is that dependent upon the machine specs?
05:49.06scythelxpimp: its like a telnet interface its all text
05:49.08JerJer[mobile]newpers: all of them
05:49.15MikeJ[Jayden]JerJer... many things, but that is a diff story :)
05:49.18harryvvman installing asterisk on a FC3 system is being a pain.
05:49.28harryvvwhen its a 64 bit system
05:49.36*** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net)
05:49.37JerJer[mobile]64 bit system won't help asterisk
05:49.49*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:49.54harryvvjerjer well i know but its there and thats what my system is.
05:50.06JerJer[mobile]get a real system then
05:50.18harryvvjer, its a opteron system as real as it gets.
05:50.21JerJer[mobile]um no
05:50.43harryvvum yes for a graphics workstation it is. My asterisk hd on the other system failed last night.
05:50.43Primeropteron > *, except when your shit's not too 64 bit friendly
05:50.55Primerin which case, just boot a 32 bit kernel and it's still a very fast box
05:51.01scythelxpimpwell: http://lists.digium.com/pipermail/asterisk-users/2003-November/025595.html
05:51.21scythelxpimpwell: example php script to connect, then u send your dial commands
05:51.46harryvvanyone by chance know what rpm restorecon command belongs to.
05:51.53pimpwellall I know is if my business works and asterisk makes my life easier I am donating so fukin much to whoever runs and maintains this beast
05:52.07pimpwellwould that be digium?
05:52.14JonR800that'd be me
05:52.20scythelxme too :)
05:52.30pimpwelljohn you cant hold my money if your already holding my..
05:52.34pimpwell:o
05:52.45pimpwellgot your hands full there
05:52.46JonR800i have two hands
05:52.50pimpwellahah
05:53.00JonR800just leave the money on the dresser then
05:53.27JerJer[mobile]pimpwell: its your system, why should someone else run and mantain it?
05:53.40pimpwellno, I mean the asterisk community
05:53.45pimpwellit has to have a donate button somewhere
05:53.55pimpwellmost open source do
05:54.10JonR800www.digium.com.. you could buy some hardware or support packages
05:54.31pimpwellonce I get enough customers I get my own lines
05:54.36pimpwelland I will need the hardware
05:54.44scythelxpimpwell: buy us all some g729 licenses
05:55.11JonR800buy me a T1
05:55.56mrgobyis there a way to grab a channel sitting on a waitforresponse and connect it to a meetme ?   i want to be able to have someone holding outside a conf basically and transfered in exactly when i want
05:56.04pimpwellprobably need to get one of you to consult me once that happens
05:56.11*** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net)
05:56.25SedoroxDoes asterisk have a problem with SMP boxes?
05:56.30JerJer[mobile]Sedorox: no
05:56.41Sedoroxhmmm
05:56.56SedoroxI have a FBSD5.3 box... SMP... and about 5 days uptime for asterisk
05:56.58JerJer[mobile]mrgoby: waitforresponse?
05:57.01SedoroxI do a "stop now"
05:57.05Sedoroxit locks the entire box
05:57.09JerJer[mobile]run Linux then
05:57.34Sedoroxcan't.. its a colo box... hard to get to it.. and the person who owns the box prefers fbsd
05:57.42JerJer[mobile]then get your own box
05:57.45JerJer[mobile]and run Linux
05:58.05mrgobyhehe
05:58.11Sedoroxhrm
05:58.21*** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com)
05:58.21JonR800im sure that's just what you want to hear Sedorox
05:58.59Sedoroxwell I know fbsd isn't 'supported'.. but I just mainly wanna know if anyone else has run into problems like this..
05:59.08*** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net)
05:59.12MikeJ[Jayden]JerJer.. you know anyone he can colo a linux * box w/ ?
05:59.31SedoroxI know 2 places I can colo cheap at...
05:59.33JerJer[mobile]sure
05:59.36Sedoroxbut that isn't the point
05:59.39MikeJ[Jayden]hehe
05:59.42Sedorox:-p
05:59.44JonR800Sedorox: another fine answer, have the box owner boot it into a non SMP kernel :-P
05:59.51mrgobywill Manager command redirect work for this ?
06:00.00MikeJ[Jayden]ok.. gnight all
06:00.01Sedoroxthe box owner sits 3 hrs away from it :-p
06:00.04Sedoroxnight
06:00.04JerJer[mobile]unlike others, I don't advertize for myself
06:00.15MikeJ[Jayden]I know..
06:00.16mrgobynufone is great !
06:00.21MikeJ[Jayden]hey does tho
06:00.22mrgobythere, i did it for you jerjer
06:00.24MikeJ[Jayden]he does
06:01.13mrgobywill redirect work for this?  can i have someone sitting on a waitforresponse, then redirect to a conference room ?  basically allowing me to put them in the conf whenever i want ?
06:01.20MikeJ[Jayden]JerJer... what building is your southfield data center in?  If it is in the galeria, I may have a customer for you
06:01.38JerJer[mobile]not in galeria
06:01.42MikeJ[Jayden]k
06:01.53MikeJ[Jayden]ok.. nighty time.
06:02.28mrgobysorry, waitforresponse == responsetimeout
06:02.44JerJer[mobile]mrgoby:  write an app
06:03.01mrgobyan app that transfers the call ?
06:03.23mrgobyi wont need to write one if i can do it through the management interface
06:03.50JerJer[mobile]just sittting in a responsetimeout is very hackish
06:04.17mrgobywell, it is for an art project and not a production system :0) so hacks are okay in this case
06:04.33JerJer[mobile]so like are you trying to make certain users wait until a 'conference moderator' joins?
06:04.59erwinismhello, what does this means? NOTICE[11825]: chan_sip.c:7681  handle_request: Registration from 'xlite <sip:111@192.168.2.200>' failed for '192.168.2.105'
06:05.11JerJer[mobile]means registration failed
06:05.17mrgobybasically....   i'm making it to where people move around in contexts, but if one person finds a certain extension, then all others get pulled into the meetme with them
06:05.19erwinismJerJer[mobile] how can i fix this?
06:05.37JerJer[mobile]erwinism: send the proper username and secret
06:05.56JerJer[mobile]mrgoby: mmkay
06:06.12mrgobyso, i need to be able to yank people into the meetme at any given time
06:06.15mrgobybasically
06:06.31erwinismJerJer[mobile] i did already. maybe i missed something on the sip.conf ?
06:06.32JerJer[mobile]nasty
06:06.35mrgobycan i do that with Action: Redirect ?
06:06.45JerJer[mobile]sounds like a job for redrect
06:06.51mrgobycoolness
06:07.04JerJer[mobile]but its not gong to be fun
06:07.11mrgobywhy come ?
06:07.42JerJer[mobile]how do you plan on figuring out which calls to transfer into the meetme?
06:07.51mrgobyall zap channels
06:08.01mrgobyonly 4
06:08.25*** join/#asterisk ryguillian (~ryguillia@c-24-12-96-52.client.comcast.net)
06:08.46harryvvjerjer also this is going to be the backup asterisk system once i get the original up and running again.
06:10.33*** join/#asterisk Damin_Mobile (~pocketirc@26.sub-70-214-24.myvzw.com)
06:10.34JerJer[mobile]i'm sorry
06:10.47Damin_MobileBastards lost my luggage!
06:10.50JerJer[mobile]lol
06:10.55JerJer[mobile]that sucks
06:11.33harryvvdamin, should have cut off your ancle tracker and put it in the luggage :)
06:11.58Damin_MobileThey think it might come in on the eleven fifteen  fligh in an hour or so.
06:12.26JerJer[mobile]famous last words
06:12.37brc_Damin_Mobile!
06:12.40brc_that sucks
06:13.34Damin_MobileWhenever you hear the phrase 'unexpected plane change' you can be sure that your luggage wont make the switch with you.
06:14.11erwinismJerJer[mobile], ok the slite is already logged in, buy how come it says "Your number is: xlite1 " ??
06:14.28The_BallDoes this look correct for iaxtel? exten => _1700NXXXXXX,1,Dial(IAX2/< my username >:< my password >@iaxtel.com/${EXTEN}@iaxtel)
06:15.33The_Balli get a Mar  7 16:08:47 WARNING[20420]: chan_iax2.c:1477 attempt_transmit: Max retries exceeded to host 69.73.19.178 on IAX2/69.73.19.178:4569/3 (type = 6, subclass = 1, ts=3, seqno=0) message
06:15.41JerJer[mobile]erwinism: no idea what your talkin about
06:15.56JerJer[mobile]is iaxtel up?
06:16.05erwinismjerjer, how can i set a user's number? im using SIP
06:16.13The_BallJerJer[mobile], what do you mean?
06:16.31JerJer[mobile]iaxtel has never been reliable
06:17.54JerJer[mobile]erwinism:  set the username and secret in your softphone
06:18.10*** join/#asterisk tecnico (~tecnico@user-24-236-123-31.knology.net)
06:18.51erwinismjerjer yes i did, my username is: xlite1 and secret is: xlite1
06:19.05erwinismand is already logged in
06:19.18JerJer[mobile]then what is the problem?
06:19.36erwinismnow, how can the other softphone call me? how would i know my number?
06:19.42JerJer[mobile]you have to make one
06:19.45JerJer[mobile]in asterisk
06:19.48JerJer[mobile]extensions.conf
06:19.52erwinismoh!
06:20.04erwinismok, i will look for it. thanks :D
06:20.05JerJer[mobile]i think i see a light bulb turn on
06:20.17*** join/#asterisk bsdfreak (ninja@enterthebass.com)
06:21.01erwinismhehe
06:21.43Damin_MobileSo im sitting in this place called Martini Monkey 's drinking beer.
06:21.49JerJer[mobile]hell yeah
06:21.53JerJer[mobile]waiting on ur luggage?
06:22.22*** join/#asterisk witten (~witten@D-128-208-60-207.dhcp4.washington.edu)
06:22.32Damin_MobileSo I ask the bartender for another Sierra  Nevada  pale ale..
06:22.38Beirdoman, setting up a meetme conference was easier than I expected
06:23.11Damin_MobileHe gives me a bud light and a shot of Crown Royal...
06:23.15Damin_MobileWTF?
06:23.48Damin_MobileA  its free..
06:23.49BeirdoI'll take the Crown Royal, but the bud light?  blech
06:23.54Damin_Mobile<PROTECTED>
06:25.11Beirdoif that arsehole calls at 5am again today, I'll be mega-pissy
06:25.42*** part/#asterisk witten (~witten@D-128-208-60-207.dhcp4.washington.edu)
06:25.47Damin_Mobilebkw_ has arrived!
06:26.14Damin_MobileAlright..Gotta sign off now...
06:26.18Damin_MobileLater...
06:27.45*** join/#asterisk mrgoby (~mrgoby@141.211.162.97)
06:29.47JerJer[mobile]yeah smells like its time to signoff as well
06:32.09*** join/#asterisk zoa (zoa@142.131.189.23)
06:32.12zoahi there
06:32.15zoalive from von
06:32.21zoaor at least live from san jose
06:32.38newpersso, for if i had a 1 voip line -> asterisk with three extensions.  I could only have 1 person on an extension at a time?
06:33.54newpersIn otherwords, I couldn't have all three extensions in use at once?
06:36.37erwinismhello, where can i edit my PBX voice prompts?
06:36.43erwinismit gives me sample voice
06:36.46*** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net)
06:36.50erwinismi just installed asterisk
06:37.00mrgoby~docs
06:37.01jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
06:37.16*** join/#asterisk thepdakid (~vtrandal@c-24-8-106-135.client.comcast.net)
06:38.34thepdakidany use packet8
06:39.04newpersor is that only for pots?
06:39.26thepdakidAnyone use Packet8.net for phone service
06:39.29thepdakid?
06:39.46thepdakidIP over cable modem
06:40.12godsmokethepdakid: what do you want to know?
06:40.20harryvvzoa how has von been today
06:40.43harryvvtime to reboot.
06:41.23thepdakiddoes it work as well as vonage?  And if I am going to setup a asterisk server what service should I have?
06:41.51godsmokethepdakid: "work as well as vonage" -- not sure what you're referring to specifically
06:42.32godsmokevonage has a completely different target audience than packet8 -- their services are both voip, but vonage focuses on home-user needs
06:44.14newpersfor this setup:  2 extensions set up on asterisk with one vonage line.  When I'm on extension one and someone calls, there's calls for extension 2, they get a busy signal?
06:45.42*** join/#asterisk geekster (~Klenert@fw.telehouse.com)
06:47.30PTG123anyone know how to make it if the person calls their own number, to not ask them for their password?
06:47.33PTG123when checking voicemail
06:48.01*** join/#asterisk cc (~cc@byte.fedora)
06:49.59thepdakidAnyone use Packet8.net for phone service?  Any trouble with audio dropouts?
06:51.39thepdakid?
06:51.55Inv_arpthepdakid: packet8 works with *?
06:52.24thepdakidI don't know but it's not SIP
06:54.14*** join/#asterisk Trionnis (buffy@12-203-113-15.client.insightBB.com)
06:54.33thepdakidvonage uses SIP and seems to work better than Packet8
06:55.38TheEmperorhi guys, can anyone help with musiconhold?
06:56.04TheEmperorit doesn't seem to play the .mp3 file i put into /var/lib/asterisk/mohmp3
06:56.22TheEmperoreven when i have already edited musiconhold.conf
06:56.54erwinismhahahahahahahaha at last i already installed asterisk!!!
06:57.04*** join/#asterisk shaZwaz (~adnans@203.81.196.167)
06:57.08justinnnnanyone no why txfax/spandsp (any versions) crash asterisk
06:57.11justinnnnafter it sends a fax ?
06:57.21geeksterTheEmperor, do you see this when you run "ps aux"
06:57.22geeksteroot  83227  0.0  0.6  3260 2824  ??  I    12:28AM   0:00.03 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-river.
06:57.23geeksterroot  83226  0.0  1.0  5428 4916  ??  I    12:28AM   0:01.91 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-river.
06:57.42*** part/#asterisk shaZwaz (~adnans@203.81.196.167)
06:57.58*** join/#asterisk shaZwaz (~adnans@203.81.196.167)
06:58.18geeksterhas anyone gotten wakupcall to work /
06:58.40shaZwazhi all
06:58.41TheEmperorgeekster: ?
06:58.51geeksteryes
06:58.55justinnnnanyone txfax ????
06:58.59justinnnnrxfax works cool
06:59.49TheEmperorgeekster: i don't get those error messages
06:59.49shaZwazjustinnnn, I tried it a few weeks ago , gets stuck
07:00.03geeksterthose are not error message, that is the process list
07:00.42geekstermpg123 should be running to play musiconhold.
07:00.43TheEmperorgeekster: yes, i see those there
07:01.39TheEmperorroot      2919  0.0  0.1  4248 1148 tty1     S    14:51   0:00 mpg123 -q -s --mo
07:01.39TheEmperorroot      2922  0.0  0.0  3772  540 tty1     S    14:51   0:00 mpg123 -q -s --mo
07:02.08geeksterwhats your musiconhold.conf file look like ?
07:02.34TheEmperor[classes]
07:02.35TheEmperordefault => mp3:/var/lib/asterisk/mohmp3/
07:02.35TheEmperor;loud => mp3:/var/lib/asterisk/mohmp3
07:02.35TheEmperor;random => quietmp3:/var/lib/asterisk/mohmp3,-z
07:02.35TheEmperorbeat => quietmp3:/var/lib/asterisk/mohmp3/beat
07:02.43*** join/#asterisk Insanity5 (~feaw@204.134.196.33)
07:03.04geekstertry this: default => quietmp3:/usr/local/share/asterisk/mohmp3
07:03.26TheEmperorok..
07:03.36geeksterremove the other crap for now.
07:04.41Insanity5Is there a recommended solution for deploying asterisk in a mission critical environment?  I want, perhaps in an idealistic sense to be able to have the user simply power cycle the machine if something goes wrong.  Points of failure, including fans and hard disks should be minimized.  Outside of using server grade hardware, are there any solutions to minimize points of failure (hard disks, fans, etc).
07:05.26Insanity5And the issue here is not the size of deployment, but rather that the geek (me) who is setting it up will be many miles away.  If I could approach the reliability of say a traditional panasonic / lucent system, that would be ideal.
07:05.45*** join/#asterisk djin (~djin@gridfox.xs4all.nl)
07:07.54TheEmperorgeekster: there is no asterisk directory in /usr/local/share/
07:08.53geekstermake it match your setup, that is where mine is
07:08.57TheEmperoroh ok
07:09.00TheEmperorduh :)
07:09.48justinnnngrr stupid txfax
07:09.51justinnnndoes it actualy work ?
07:09.55TheEmperorgeekster: it works! :)
07:10.02TheEmperorgeekster: thanks!
07:10.18geeksteryup
07:10.19geeksternp
07:10.36*** join/#asterisk Goshen (Goshen@c-67-172-238-57.client.comcast.net)
07:10.41TheEmperorhow come it didn't work the last time?
07:10.47Trionnisanyone aware of outgoing voice issues with broadvoice?
07:10.55Trionnispeople tell me I keep "fading out"
07:11.08TrionnisI hear incoming audio just fine, however
07:12.06Goshensomeone said something about 4 hours ago about problems with broadvoice...don't remember who it was
07:14.05Trionnisk
07:14.06Trionnisthakns
07:14.08Trionniser
07:14.09Trionnisthanks
07:14.10Trionnis:)
07:14.21Insanity5Can you use stanaphone with asterisk or no?
07:14.21Inv_arpInsanity5: set it up like u would any other mission critical system...
07:15.07Inv_arpInsanity5: iyes
07:15.57TheEmperorgeekster: how do i turn the volume down on musiconhold?
07:15.57justinnnnppls
07:16.00justinnnnwat does this mean ?
07:16.00justinnnn#define MAX_BLOCK_SIZE 240
07:16.03justinnnnin app_txfax.c ?
07:16.18geeksterEMP; that i'm not sure about.
07:16.26Inv_arpTheEmperor: resample the music at a lower volume... any music editor
07:16.31geeksterwhat type of phone do you have.
07:17.15TheEmperorjust using normal office phones
07:17.19Insanity5Inv_arp - I know... There's just something about computers tha says it won't be working 15 years from now without someone touching it :)
07:17.28TheEmperorInv_arp: any recomendations?
07:18.02Inv_arpTheEmperor: audacity  even sox can do that  must must be converted to wav first
07:18.09Insanity5Inv_arp - The reason I asked about stanaphone is because I thought there was some special feature that your voip provider must support to produce a working system.  I can't remember anymore, I'm about to set my first one up though :)
07:19.02TheEmperorInv_arp: ok
07:21.20Goshenwho has an enum164.org entry that I can call to test my enum lookup? I tested it working for my entry
07:22.16*** join/#asterisk par (par@trackbugz.antisecurity.net)
07:26.23Mavviegoshen 61293353018
07:26.38GoshenCalling
07:26.51Goshenwait..what kind of number is that?
07:27.01Mavvieit's my number.
07:27.15Goshen6 is the international prefix?
07:27.18Mavvieno, 61
07:27.28parozzie
07:27.36Mavvieoi!
07:27.40*** join/#asterisk Eight (~blake@12-205-155-39.client.mchsi.com)
07:27.45Mavviemy wife would be proud of me.
07:28.20Goshenso I guess I dial 01161293353018
07:28.40Mavvieyes
07:29.02Goshenconfusing my sipura :)
07:29.04MavvieI won't pick up the phone.
07:29.09Mavvieto save you money.
07:29.21Goshenif you have an enum lookup it should go direct...
07:29.54dfunnellHi all, can anyone help with a dial-out problem I am having? Using CAPI * is dialling as soon as a pattern is matched, which makes it difficult with variable length numbers (such as mobile numbers, etc.)  Using exten => _100.,1,Dial,CAPI/470:${EXTEN:1} for example dials out on 00n (where n is the fourth digit dialled after '100').  V desperate, will pay in beer.
07:30.02Goshenconfusing my dialplan too
07:30.08Mavvie:-)
07:33.16PTG123anyone know of a way to know the name of the sip account dialing the extension?
07:34.31jontowAGI might do it (?)
07:34.39GrimStonethere any solution to "got a response on a call we don't know of" problems with Broadvoice outgoing ?
07:36.22*** join/#asterisk jmhunter (~jmhunter@64.77.199.223)
07:36.22*** mode/#asterisk [+o jmhunter] by ChanServ
07:36.29jmhunterwhats up bitches
07:36.37PTG123well i want to do it in the extension line
07:36.40PTG123in extensions.conf
07:36.50geeksterhas anyone gotten wakupcall working ?
07:37.04GoshenMavvie: SIP/0293353018@barnet.com.au  :)
07:37.09jmhunterI NEED A VON PASS
07:37.10jmhunterNOW
07:37.12jmhunterplease
07:37.22GoshenMavvie: == No one is available to answer at this time
07:37.37jmhunter~ seen brc_
07:37.38jbotbrc_ is currently on #asterisk.  Has said a total of 14 messages.  Is idling for 1h 24m 58s
07:37.58brc_~seen jmhunter
07:37.59jbotjmhunter is currently on #asterisk (1m 37s).  Has said a total of 5 messages.  Is idling for 22s
07:38.11jmhunterwhats up bitch
07:38.20brc_your pm window broken?'
07:38.22MavvieGoshen: didn't see a call coming in.
07:39.02*** join/#asterisk Tarox (someone@pD9E7BF13.dip.t-dialin.net)
07:40.09MavvieGoshen: are you sure it calls here?
07:40.13Mavvieto here?
07:40.37GoshenI think so..I pasted all of the messages in query window
07:40.44GoshenI will try again with sip debug on
07:41.02jmhunter~seen kram
07:41.03jbotkram <~mark@kram.digium.sponsor.pdpc> was last seen on IRC in channel #asterisk, 2d 4h 4m 27s ago, saying: 'oh :)'.
07:41.14jmhunterfucking shit
07:41.32jmhunterwhere are you mother fuckers
07:41.37jmhunterill deop myself for that
07:42.48MavvieGoshen: still: I don't see any packets here
07:42.55*** mode/#asterisk [+o brc_] by jmhunter
07:43.02brc_0_0
07:43.21rvhiis there a way to find out how many agents on the phone in ACD?
07:43.34jmhunterkram
07:43.36jmhunterbkw
07:43.38jmhuntertwisted
07:43.40jmhunterFUUUUCK
07:45.03GoshenMavvie: firewall?
07:45.08MavvieGoshen: none here.
07:45.25jmhunterfuuuck where is everyone
07:45.29jmhuntersomeone should deop me
07:45.33jmhunterbrc hit me
07:45.36brc_yeah they should
07:45.41Mavvie--- #asterisk :You need to be a channel operator to do that
07:45.45Mavviecan't tell that I didn't try
07:46.07*** kick/#asterisk [jmhunter!~brian@brc.base.supporter.pdpc] by brc_ (#asterisk /me hands jm some soap)
07:46.11MavvieGoshen: can you ping tardis.barnet.com.au ?
07:46.20*** join/#asterisk jmhunter (~jmhunter@64.77.199.223)
07:46.20*** mode/#asterisk [+o jmhunter] by ChanServ
07:47.02Goshen64 bytes from tardis.barnet.com.au (202.83.176.38): icmp_seq=1 ttl=45 time=189 ms
07:47.03*** mode/#asterisk [+brc +q!*@*] by jmhunter
07:47.19brc_fumble fingers
07:47.23Goshenthat isn't the machine I am calling though
07:47.31*** mode/#asterisk [+brc -q!*@*] by jmhunter
07:47.38brc_dude
07:47.39brc_haha
07:47.45*** mode/#asterisk [+brc +q!*@*] by jmhunter
07:47.45*** join/#asterisk DrRighteous (~DrRighteo@ool-182c867b.dyn.optonline.net)
07:47.49Qwellhe needs practice. :p
07:47.51brc_hey there dr
07:47.54jmhunterno one can here u
07:47.56brc_DrRighteous, at von?
07:48.02*** mode/#asterisk [+brc -q!*@*] by jmhunter
07:48.06GoshenMavvie: 64 bytes from tim.barnet.com.au (202.83.176.33): icmp_seq=4 ttl=45 time=205 ms
07:48.09DrRighteousI wish
07:48.14dfunnellDid I mention I will pay in beer?  Help!
07:48.16DrRighteousBut I sent file to CON
07:48.18DrRighteousVON
07:48.18Qwell/unban +q!*@*, /mode #asterisk -rc
07:48.20MavvieGoshen: "I don't know"
07:48.20Qwellor something, heh
07:48.28brc_no, it's supposed to be +rc
07:48.37*** mode/#asterisk [-b +q!*@*] by brc_
07:48.42jmhunterher Dr Rightheous.. u hook me up with a von pass.. ill hook u up with OPS
07:49.09DrRighteoushaha, file's an employee... and he knows how to beg
07:49.16*** join/#asterisk footnote (~jhicks@67.141.135.121)
07:49.47GrimStonethats so Rightheous..
07:49.53footnote"he's dead jim"
07:50.13jmhunterjacob not jm
07:50.15jmhunterblah
07:50.20jmhunterjim
07:50.24footnotehrm
07:50.36footnotei don't remember ole jake
07:50.50jmhunteri think i was around before u
07:50.57footnotedoubt it :)
07:51.00jmhunterim an old timer
07:51.11footnotemy grandson born today's name is jacob!
07:51.26footnotejacob tyler hicks
07:51.29Mavviecongratulations.
07:51.46footnoteyeah, he's gonna be one heckuva good flyfisherman
07:52.05*** mode/#asterisk [+o DrRighteous] by jmhunter
07:52.56jmhuntero ok.. well i was here mostly during summer
07:53.19GoshenMavvie: phone ring?
07:55.29GoshenMavvie: sip call didn't work so it fell back to dialing with voipuser.org, it looked like your phone rang, did it?
07:55.51MavvieGoshen: aha, but that wasn't via the sip phone :-)
07:56.14*** join/#asterisk kks (~kks@203.115.210.253)
07:56.31Goshenyea, I can't reach your sip phone...can you?
07:56.37Mavvieyes.
07:56.53Goshenanyone else have a enum164.org entry that I can try calling?
07:57.08*** join/#asterisk HitTop (~Miranda@HSE-Toronto-ppp286299.sympatico.ca)
07:58.24footnotehrm
07:59.54*** join/#asterisk Duckbizkit (~DMAN@ip-216-97-163-53.valornet.com)
08:00.02Duckbizkitsweet
08:00.10jmhunterIn search of VON passes
08:00.12jmhunteranyone
08:00.32tuxinator_linuxjmhunter: Nope, but I'll be at Meet *
08:00.51tuxinator_linuxI'm in search of VON passes
08:01.36Insanity5How many people here use voip for their business?  I am kind of weary of switching over from POTS, simply because of the possibly downtime from the provider and possible issues with SBC dsl.
08:01.52tuxinator_linuxI wouldn't do it over DSL
08:02.00tuxinator_linuxfor a business at least
08:02.11tuxinator_linuxunless you only do one call at a time
08:02.15*** mode/#asterisk [-b -q!*@*] by jmhunter
08:02.21Duckbizkiti've got a tough VM question....any takers?
08:02.34tuxinator_linuxspit it out
08:02.48footnoteVM is hard, let's go shopping.
08:03.03tuxinator_linuxVM like in Voice Mail?
08:03.07Qwellfootnote: I'm in.  I need new clothes.
08:03.20GoshenInsanity5: the other problem with dropping your POTS is loosing your phone directory listing
08:03.23*** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net)
08:03.42footnotevlad the impaler works for atmel now
08:03.44footnotekewl
08:03.54Insanity5Goshen - Can't you port it over?
08:03.55tuxinator_linuxPeople that don't use ILECS loose the phone directory listing anyways
08:03.55Duckbizkitif i call VoiceMailMain(s204) , it works fine and goes straight to VM. if i call VoiceMailMain(s6233239933), it prompts for the login/pass
08:04.05Insanity5Goshen - The goones with the phonebook will gladly sell you any overpriced listing you want, I'd assume.
08:04.18Duckbizkitit seems like these 10 digit numbers i'm passing to VM are just messing it up
08:04.23GoshenInsanity5: of course...but how about directory assistance?
08:04.25tuxinator_linuxThat's what I had to do, buy an add in the phone book
08:04.28Duckbizkitbecause everything eles works
08:04.31Duckbizkit*else
08:04.43tuxinator_linuxnever thought about directory assistance
08:04.48tuxinator_linuxdon't they have a phone book?
08:04.54GoshenInanity5: also let me know when you find a VOIP provider that does number portability on US numbers...
08:05.07GoshenNufone does for their local area code only
08:05.29Insanity5Gos - Vonage?
08:05.45GoshenVonage doesn't allow Asterisk so they are out
08:05.49GoshenGos?
08:05.53QwellGoshen: broadvoice supposedly does.  or, will soon
08:06.07GrimStoneBroadVoice worked fine until Saturday
08:06.15GrimStonethen they screwed it up
08:06.16jmhunterbv has as of like june
08:06.25GoshenQwell: too many people in here complaining today about Broadvoice being down this weekend
08:06.27Insanity5Goshen - AT&T VOIP?
08:06.29Qwelldunno, the site said "coming soon"
08:06.41GoshenThink AT&T is going to allow Asterisk?
08:06.52GrimStonethere seems to be some strange issue with asterisk , and BroadVoice presently
08:07.02Insanity5Goshen - Why not?
08:07.08QwellAre there any providers that do everything most people want?  heh
08:07.12Duckbizkitso any ideas?
08:07.26Goshenbecause it is too open
08:07.38GrimStonekeep getting this on outgoing calls - "Got a response on a call we dont know about. "
08:07.50QwellGrimStone: weird nat issue?
08:08.02Insanity5Goshen - What does Asterisk require, that stanaphone and broadvoice supplies, that vonage won't?
08:08.03GrimStonei don't have NAT .. thats the strange thing
08:08.08Qwellie; the response looks like its coming from your router or something?
08:08.24QwellGrimStone: That would be tricky then, yeah.  heh
08:08.33GrimStoneif i set nat=yes , i can hear the voice data , but those messages still come up in asterisk
08:08.34GoshenInsanity5: allow you to connect your Asterisk server to them...
08:08.38*** join/#asterisk iceyp (~icepick@max.unix.co.nz)
08:08.52GoshenI mean you could go VOIP to FXS, then FXO card to Asterisk server...but who wants that
08:08.53QwellGrimStone: Your box has a external IP?
08:08.58GrimStoneand asterisk seems to think the call is still not connected
08:09.03QwellGoshen: With vonage, or at&T?
08:09.12GoshenQwell, right
08:09.15iceypcan someone test 08449865089 from the UK and tell me if it connects u to an IVR plz, my telephone provider and voip providers dont have PSTN in the UK
08:09.24QwellI mean, were you specifying one in particular?
08:09.27GrimStoneQwell: yes , and broadvoice's subnet has access to the right ports
08:09.49QwellGrimStone: odd.  got me
08:09.51Insanity5Goshen - I mean though, what is so special about asterisk as a SIP client compared to any generic SIP device?  How would they know?
08:09.57Goshenno, just in general foe closed voip providers
08:10.15GoshenInsanity5: because their configurations are secret
08:10.18GrimStoneQwell: and with nat=yes when the call gets over it shows up as NO ANSWER
08:10.19QwellInsanity5: You underestimate them
08:10.44ta[i]ntedhow do i delimit SetVar: in a call file?
08:10.50Duckbizkit:)
08:10.51GoshenBecause when you place a call it says ASTERISK in the call ;)
08:11.02ta[i]ntedSetVar: foo=bar pee=poo
08:11.03QwellGoshen: to be fair, that can be changed, right?
08:11.16QwellDuckbizkit: Doesn't voicemailmain take you to the login always?
08:11.20QwellI haven't rtfm'd lately
08:11.22Insanity5Goshen - OLutside of any possible "referred" type value Asterisk passes along, I see no technical reason why it would not work, provided you obtained the SIP login and pass.
08:11.32Duckbizkitnot if you put the s in front of the mailbox
08:11.46Qwellahh
08:11.59QwellDuckbizkit: and the 10 digit one is valid in voicemail.conf?
08:12.03*** join/#asterisk memic (~memic@195.135.160.215)
08:12.29Duckbizkitwell we're running realtime
08:12.35Duckbizkitbut yes it's valid, i have left several VMs
08:12.43GoshenInsanity5: they don't give you your login and pass...and I am not interested in hacking a provider...rather go with one that supports *
08:13.03QwellGoshen++
08:13.36*** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it)
08:13.45Insanity5Goshen - Is it possible to configure a setup where say, if you don't answer your home phone (voip), asterisk forwards it on out another voip channel to your cell?  Or is this just a bad idea all around for bandwidth and latence/delays.  Is this best handeled at the provider level?
08:13.46QwellDuckbizkit: realtime includes voicemail.conf?
08:13.56QwellDuckbizkit: I've never really looked into it.  I think I might sometime.
08:14.06QwellInsanity5: sure its possible
08:14.11Duckbizkityeah you still need voicemail.conf when you run realtime
08:14.15Duckbizkitfor all the other config stuff
08:14.16atmelfootnote, what?
08:14.20Duckbizkitrealtime just moves off the users to the DB
08:14.27QwellInsanity5: I personally do it at home with my fwd account.
08:14.30*** join/#asterisk iceyp (~icepick@max.unix.co.nz)
08:14.31GoshenInsanity5: better at provider, then if not you can at your * box, you can do anything with *
08:14.32iceypmeh
08:14.33atmel;)
08:14.39atmeleu vin sa te mananc
08:14.41iceypanyone try that number for me from the uk?
08:14.42atmelmuahaha
08:14.57QwellDuckbizkit: oh, I see.
08:15.01Gosheniceyp: try #asterisk-uk
08:15.11GoshenI got someone in there to dial mine to test...
08:15.25Insanity5Goshen - I mean, incomign call over voip from broadvoice, forwaded out to stanaphone to your cell making three trips across the country and back, and on a cell none the less.  Does this smell as bad as I think?  Or is it quite bareable?
08:15.52QwellInsanity5: sounds like me connecting to work with the vpn!
08:15.58Goshendepends on your connection
08:16.05Insanity5Qwell - lol
08:16.07iceypGoshen thanks
08:16.15Insanity5Goshen - 200kbit upstream, downstream's nto an issue
08:16.21QwellInsanity5: CA to IA, back to CA (1 city away from me), out to IA, back to CA, then out to MN for files
08:16.29GoshenI am really looking forward to 3-13-05...thats the day my connection goes to 6mb/s down 768/kb/s up
08:16.33moonwickInsanity5: it's worse than you think
08:16.41QwellGoshen: which provider?
08:16.49Insanity5Sounds like speakeasy
08:16.50GoshenComcast + $10 speed upgrade
08:16.51jmhunter~seen pfn
08:16.52jbotpfn <500@adsl-69-107-210-254.dsl.pltn13.pacbell.net> was last seen on IRC in channel #asterisk, 13d 6h 23m 15s ago, saying: 'only with iax'.
08:16.54Qwellahh
08:16.57moonwicklatency is a very real problem with VoIP
08:17.11Insanity5moonwick - Why does it sound bad, upstream bandwdith or flat out latency?
08:17.21Qwelllatency, no doubt
08:17.25Insanity5The cell phone will add it's own fair share.
08:17.42Insanity5moonwick - Now, arguably, does a POT line or voip line have more latency?
08:18.07moonwickthat's not a very good question, in all honesty
08:18.34*** join/#asterisk mithro (~tim@dsl1-83.gw1.adl1.airnet.com.au)
08:18.39Insanity5moonwick - It probably isn't :(
08:18.40moonwickbut depending on the setup, VoIP is going  to be much more prone to latency
08:18.48*** join/#asterisk djin (~djin@62.58.40.196)
08:18.56Insanity5I've found the voice quality is sometimes better on voip though.
08:19.14mithrohi! i'm after some information for FXO/FXS stuff which is compatible with australia
08:19.41Insanity5moonwick - Would it be a lot less noticeable if I found a local voip provider where latency was say 40 ms instead of 80?  I mean, arguably 40+40 = 80, and 80+80 = 160, but who knwos how the provider is going to backhaul it.
08:19.53moonwickInsanity5: yeah, that can make a considerable difference
08:19.57*** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk)
08:20.02Duckbizkitwell i guess i'm just fuckered
08:20.02Duckbizkitheh
08:20.14Insanity5moonwick - I mean 80ms is acceptable for voip, but the double latency from a two directional link can add up.
08:20.23Insanity5moonwick - Phone lines = $36/month out here :(.  It's going.
08:20.25moonwickthe human ear starts noticing latency even as low as 200ms
08:20.41moonwickand when you consider round trip IP latency, and codec latency...
08:20.42moonwickit all adds up
08:20.47mithropeople have said on the internet they have had problems with the normal X100 cards
08:20.51Insanity5moonwick - What is the typical cell phone latency?
08:21.02moonwickdepends on the network
08:21.05Insanity5sprintpcs
08:21.14Insanity5I've noticed the newer phones compress the hell out of the audio more too.
08:21.19Insanity5cdma / 1.9ghz
08:21.32moonwickI dunno, but my GSM phone seems to have pretty negligible latency
08:21.38*** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com)
08:21.40moonwickit's not as easy to measure as it is with voip
08:21.54Insanity5Nope, but I can tell when two cells are talking.
08:22.14Insanity5I ping ~100ms to sip.broadvoice.com, they a bad idea?
08:22.19*** join/#asterisk af_ (~af@ip-148-227.sn1.eutelia.it)
08:22.38moonwick100ms is probably tolerable, but it also depends on what you plan to do with those calls once they hit your PBX
08:22.46ta[i]ntedanyone here use call files?
08:22.55moonwickI've got my asterisk box set up to ring my cell phone whenever I receive a call
08:23.06Duckbizkitta[i]nted, i've used them to annoy my friends
08:23.12Qwellmoonwick: Just ring it, or forward entirely?
08:23.13Goshenmoonwick : can I see that part of your dialplan?
08:23.20ta[i]ntedDuckbizkit u ever use the SetVar in a call file?
08:23.24QwellGoshen: yeah, I was gonna ask the same
08:23.27moonwickQwell: ring it, along with a couple of SIP phones
08:23.27GoshenI was thinking about doing the same thing
08:23.37ta[i]ntedthat's easy
08:23.40moonwicksec
08:23.46Qwelloh, like Dial(Zap/1&SIP/blah) ?
08:23.53moonwickexactly
08:23.54ta[i]ntedexten s,1,Dial(SIP/cellphone&IAX/foo)
08:23.58Qwelloh, o
08:23.59Qwellk
08:24.13Duckbizkitnope never used SetVar
08:24.24Insanity5moonwick - I'd Asterisk to pick it up, say "connecting your call" nextel style, perhaps play some geeky hold music, while it simultaounsly rings my cell and home (SIP connected) phone.  It would also be acceptable to place 3 rings on the cell and then forward it to the cell.
08:24.43moonwickI have problems with that because I can't get SIP to reliably cut my asterisk box out of the path when a call is answered by the cell phone
08:24.46tzafrirgood morning
08:24.49Insanity5The last line has an obvious type-o :)
08:24.52*** join/#asterisk newpers (newpers@ip24-56-8-180.ph.ph.cox.net)
08:24.55Qwellmoonwick: That doesn't actually take up any minutes on the cell, unless you answer, huh?
08:25.02moonwickcorrect
08:25.08Qwellinteresting
08:25.21mithroso anyone know people using Asterisk in australia?
08:25.29Insanity5moonwick - I suppose that would require suffient outgoing lines, or SIP connected phones.
08:25.32Goshenmoonwick: would still like to see that part of your dialplan
08:25.33moonwickit would work well, if I could find a way to reliably get my asterisk box to pull itself out of the media path for calls answered by the cell
08:25.51moonwickGoshen: just look at what quell wrote
08:26.09moonwickdial(SIP/homephone&IAX/provider/<cell#>)
08:26.35Goshenso you are sending it out over voip
08:26.56moonwickyeah
08:27.07QwellThat kinda sucks though, since it basically charges you twice
08:27.12Goshenwhat if you added a Tt on the end?
08:27.14newpersI want to get asterisk up an running on 3 extensions and one line coming from nufone.net's voip service.  I'm trying to figure out exactly what I need without using softphones.  Is this correct:  voip service, asterisk & server, 3 sip phones or 3 iaxy converters and 3 analog phones?
08:27.35moonwickQwell: I use voicepulse for the incoming DID, which is flat rate
08:27.45Qwellahh
08:27.48Goshenwould that allow the call to bridge? or if you specified bridging in your iax.conf?
08:27.49newpersIs there anything else?  And would it be cheaper to use sip phones over iaxy
08:27.51moonwickbut consider that I'm in austin, and voicepulse's server is in NY
08:27.54Insanity5stanaphone is free incoming too, no monthly fee.  Caveat:  New york #
08:28.02Qwellnewpers: you won't need sip phones if you use an iaxy
08:28.08Qwellerm, nevermind!  misread that
08:28.14moonwickif someone from Austin calls my DID and I answer from the cell, it gets to travel to and from NY twice.
08:28.24*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02m-12-87.d4.club-internet.fr)
08:28.24Insanity5Plus cell phone latency...
08:28.26Qwellnewpers: It can vary greatly.  If you want an expensive SIP phone, sure, it'll cost more.
08:28.33newperswhich would you recommend?
08:28.33Qwellbut if you want a cheapy grandstream, it may be cheaper
08:28.38Insanity5moonwick - How does it sound?
08:28.41*** join/#asterisk Damin_Mobile (~pocketirc@64.sub-70-214-30.myvzw.com)
08:28.43moonwicklike crap
08:28.48Qwellnewpers: I would recommend you do a little research, honestly.
08:28.55Qwellfind out which method would be better for you
08:29.04Insanity5moonwick - Better than a cell to cell call?
08:29.05newpersI didn't know there was a difference
08:29.09moonwickSIP is supposedly capable of optimizing extraneous hosts out of the media path, but I've had a lot of trouble getting that to work reliably
08:29.15*** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc)
08:29.31*** join/#asterisk pbxjunkie (~stormtroo@videocomputer.gr)
08:31.24Insanity5moonwick - No problem with using SIP to connect to an office asterisk server via SIP over the internet and "work from home"?
08:31.42ZeeekI do that every day
08:31.44Insanity5And just be say, an extension?
08:32.02GoshenI had to use IAX to get to my office...
08:32.05jmhunter~seen twisted[work]
08:32.06jbottwisted[work] is currently on #asterisk
08:32.11Goshenoffice is double(tripple) nat
08:32.18Insanity5Goshen - Yuck :)
08:32.19Insanity5Hehe.
08:32.21Goshen* server is NAT
08:32.24moonwickInsanity5: depends on how much latency there is between home and the office
08:32.30Insanity5moonwick - 60ms.
08:32.33Goshenso it was just ugly...IAXy made it right through
08:32.48Qwellthats alot of hops!
08:32.56Insanity5moonwick - Only 128kbit available upstream though.
08:32.57moonwickif the office is connected to the PSTN from there, that's probably no big deal
08:33.04Insanity5moonwick - Yes, it's pstn.
08:33.13moonwickyeah, no big deal, I imagine
08:33.38Qwellit sucks, because our firewall at work is far too restrictive
08:33.51Insanity5Qwell - Probably for good cuase :)
08:33.53moonwick128kbps is plenty for VoIP, but you'll need some sort of QoS
08:34.00QwellInsanity5: Just because I work for a bank...
08:34.01Qwell:p
08:34.10Insanity5Qwell - Understandable :)
08:34.19Zeeeksome offices subscribe to http filtering services. If a site talks about subject "a" they arbitrarily put it in the list
08:34.20Qwellwe get 80, and only http
08:34.33shaZwazis there a cracked version of Eyebeam ?
08:34.35Insanity5Qwell - You're lucky for that.
08:34.40Insanity5moonwick - The bset I can do is a hacked-up linksys router firmware w/ qos.
08:34.54Insanity5moonwick - However, there's only websurfing on the other side.  But it could get ugly quick :0
08:35.24Insanity5Is it best to make it so your fax line never has to touch the * box?
08:35.31moonwickonce I got around to setting up an old pentium with openbsd and altq, I became far more satisfied with voip
08:36.50*** join/#asterisk mitcheloc (~mitchel@69-169-28-46.anhmca.adelphia.net)
08:36.59Insanity5Why did they discontinue the X100P cards?
08:37.06Insanity5I know they're $6.99 on ebay, but still
08:37.10Insanity5There's no analog solutions left.
08:37.22mitcheloc$6.99 for a clone no?
08:37.28Qwellfor the clone, yeah
08:37.29mitchelocthe tdm400b cards are analog
08:37.34mitchelocnot the real thing though
08:38.16Insanity5Anything wrong with this?  http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=44940&item=6748462326&rd=1&ssPageName=WDVW
08:38.16footnotehrm. altivec based codecs
08:38.39QwellInsanity5: its a clone
08:38.50Insanity5Does it matter?  you can't get the real thing anymore anyways.
08:38.54Insanity5Is there any quality loss?
08:39.59brc_yes
08:40.07Insanity5brc_ - Yes to which one? :)
08:40.20brc_last
08:40.30Insanity5Just inferior components?
08:40.49*** join/#asterisk NoCAT (NoCAT@c-24-9-32-2.client.comcast.net)
08:41.07NoCATanyone alive in here?
08:41.17Zeeekbarely
08:41.54*** join/#asterisk andi2 (~andi@212.88.172.176)
08:41.56mitchelocnah
08:42.00mitchelocthey are intel pci modems
08:42.04mitchelocwith the same chipset
08:42.31Insanity5Is there anything wrong with using a fleabay card?
08:42.33mitcheloci bought one too a while ago, but never used it, i realized i didn't have a use and jumped straight to a t100p =), now i've figured a use for it though
08:42.46mitchelocyep, your not supporting digium
08:43.02mitcheloci didn't know anything about asterisk then, i would not buy a clone again
08:43.09Zeeekhave you guys ever needed digium support?
08:43.14mitchelocmany times
08:43.20Zeeekhow is it?
08:43.22moonwickonce
08:43.28mitchelocpretty good, they do try and help you
08:43.30Insanity5mitcheloc - They don't offer a single card anymore, and I don't want to fork out for a 4 channel.
08:43.34mitchelocand will call in other people when you have problems
08:43.44mitcheloc* that they can't solve
08:43.56ZeeekI have a very minor but I emailed. We'll see what happens
08:44.48ZeeekIf those $10 cards really work, it'd be a grea deal to set up for soho and home use
08:45.09Zeeekthe ad/listing emphasies asterisk
08:45.23mitchelocugh, /me found a very annoying bug in asterisk
08:46.24Insanity5Are quad T-1's worth anything?  I have an old clarent Quad T-1 gateway that has such cards in pci form.
08:46.52Insanity5I've been meaning to put the whole box up on ebay... I got it for less than 20 at a bankruptcy auction just to play with it :P
08:47.26mitchelocsounds like expensive hardware
08:48.06Insanity5mitcheloc - It was, they paid something like $10,000 according to the invoice. However, I know it's worth pennies now...  It's some massive voip gateway solution for 96 channels off some custom nt4 software.
08:48.19Insanity5Voip wasn't exactly in its heyday in 1995 either :P
08:50.30Insanity5mitcheloc - They look like this, except they have quad T-1 and soem custom hardware codec solution cards that this one doesn't: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=61839&item=5754686552&rd=1&ssPageName=WDVW
08:51.23Insanity5I think I'll sell it... that's the first one I've seen that sold over the last 6 months :P
08:51.43drumkillame being file
08:51.48tuxinator_linux<PROTECTED>
08:52.16NoCATwhat $10 cards?
08:52.56Insanity5NoCAT - http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=44940&item=6748462326&rd=1&ssPageName=WDVW
08:53.33Insanity5NoCAT - Probably clones / firmware hacks / who knows.  If they work, it's what I need.  I can't find a single port card on diginum's website.
08:54.12*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
08:54.12*** mode/#asterisk [+o bkw_] by ChanServ
08:54.16bkw_yo yo yo
08:54.18bkw_wasabi
08:54.24drumkillayoyo yo yo!!!!!!!!!!!
08:54.28drumkillaVON!
08:54.54Qwellbkw_: sushi?
08:55.10*** join/#asterisk file (~file@251.134.218.209.transedge.com)
08:55.11tuxinator_linuxWuzzup bkw_
08:55.27fileomg Becky!
08:55.37drumkillafile!!!!!!
08:55.42filedrumkilla!!!
08:55.49tuxinator_linuxWhat about me ?
08:55.55Qwellfile&drumkilla!
08:56.31tuxinator_linuxfile: look at her butt, its like, sooooo big
08:56.35TheEmperorhi, can anyone tell me why my voicemails don't get sent to me as an email?
08:56.44TheEmperordo i need to install a webserver on my * box?
08:56.52tuxinator_linuxsendmail
08:56.59Insanity5What happens if you connect more than one device at once with a voip provider?  DO they both rings at once?  Does the voip provider freak out?
08:57.17TheEmperortuxinator_linux: sendmail?
08:58.00QwellInsanity5: depends on if they support multiple devices
08:58.17fileomg Becky, what is it?!?
08:58.20bkw_GOD LOVE SQL INJECTIONS
08:58.20bkw_har har har
08:58.20bkw_NEXT!!!
08:58.21Qwell(it would likely ring them both, if they're both registered)
08:58.37tuxinator_linuxTheEmperor: You need a program to send mail out with.  Hold on...
08:58.40NoCAThow does a voice t1 work?  do you have 24 phone numbers or does asterisk work the numbers out?
08:58.45TheEmperortuxinator_linux: oh...
08:58.53Insanity5Qwell - Is the call quality greater for a voip-to-voip call than pots-to-pots?  I know in house cisco stuff sounds nice.
08:58.57QwellNoCAT: DIDs aren't (they can be) associated to a specific port.
08:59.07QwellInsanity5: got me
08:59.12Insanity5NoCAT - Normally, you'll use it as a hunt group.
08:59.23Insanity5NoCAT - T1's give you lots of fleixibility.
08:59.47NoCAThow does the number assigning work? can you have more then 24 phone numbers?
08:59.55QwellNoCAT: Don't think of a port on a T1 as a phone number.  Its simply a line.  You can set it up for incoming or outgoing (or both).  Then you get as many phone numbers as you want.
09:00.04QwellYou can get 1, or you can get 500.  It doesn't matter.
09:00.15NoCATthats amazing..
09:00.19Qwellbut
09:00.26Qwelleach port, can only have ONE active call
09:00.42Qwellie; if you have 1 T1 with 500 phone numbers, you can still only get 24 calls.
09:00.59tuxinator_linuxQwell: You should clarify that is a ISDN PRI over a T1 circuit
09:00.59Qwell(and that DOES include outgoing as well)
09:01.05Qwelltuxinator_linux: umm, ok
09:01.08Insanity5NoCAT - And you can split half with data and voice :)  And other fun stuff.
09:01.09Qwellwhat he said
09:01.18Qwellin other words, rtfw!
09:01.22tuxinator_linuxI was confused for a long time
09:01.44Insanity5Do any voip providers still let you mess with callerid liek you can with a T1/pri? :)
09:01.57*** join/#asterisk soulz- (~soulz@host-137-132-45-54.imcb.nus.edu.sg)
09:01.59soulz-hello all
09:02.04*** part/#asterisk pbxjunkie (~stormtroo@videocomputer.gr)
09:02.21tuxinator_linuxTheEmperor: http://voip-info.org/wiki-Asterisk+config+voicemail.conf
09:02.34NoCATcan you get numbers from your local phone company?  lets say i have 12+ phonelines now, at $30 each.  and i wanted to get a t1, would i be able to keep numbers i currently have?
09:02.36tuxinator_linuxtake a look at "mailcmd"
09:02.48tuxinator_linuxNoCAT: don't see why now
09:02.50QwellNoCAT: not sure.  maybe you can port them
09:02.50tuxinator_linuxnot
09:03.02QwellDIDs are dirt cheap from what I hear, too
09:03.03tuxinator_linuxNoCAT: That is what I plan to do
09:03.03TheEmperortuxinator_linux: ok thanks
09:03.19Qwellie; you sure as hell aren't going to pay $30, or even $3.  Maybe something like $.30
09:03.25tuxinator_linux$2/month for 20 numbers or something like that
09:03.26NoCATqwest is currently my provider
09:03.29Qwellthe T1 is still going to cost you a bit though, of course
09:03.34Insanity5NoCAT - If they're terminated in the safe local c/o, liekly.
09:03.34Qwellits completely seperate
09:03.35NoCATservice provider
09:03.40tuxinator_linuxYou will pay about 550 for the T1
09:03.58Qwellit varies greatly from state to state (and probably city to city)
09:04.02tuxinator_linuxbut that is 23 voice channels and one control (D)
09:04.03Qwelland of course, between providers
09:04.22*** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au)
09:04.46Qwellsometimes the pricepoint is at like 10-15 lines, right?
09:04.48Insanity5What do people do for extra phone lines as I'm sure any * user finds a need for them.  Free voip provider?
09:04.54TheEmperortuxinator_linux: so do i need to install a send out mail program as well?
09:04.55tuxinator_linuxI am getting ready to purchace my first PRI, does anyone recommend any providers?
09:05.08tuxinator_linuxTheEmperor: Yes
09:05.14Insanity5Qwell - There's also logistical, business, and management decisions.  Do you WANT to manage 16 individual analog lines?
09:05.17FaithfulHi all
09:05.17tuxinator_linuxTheEmperor: It looks like it
09:05.18TheEmperortuxinator_linux: what do you recommend?
09:05.22QwellInsanity5: of course
09:05.32tuxinator_linuxTheEmperor: sendmail is fine
09:05.51tuxinator_linuxTheEmperor: Better to start with that as * expects it.
09:05.52TheEmperortuxinator_linux: so i have to configure mailcmd, the line to send out using sendmail?
09:05.55QwellI'd just have the whole thing sent into *, heh
09:06.00Insanity5Qwell - No, I meant 16 individual phone jacks, phone wires, and copper pairs.  a T-1 is one circuit.  Not to mention they usually come with a higher SLA.
09:06.13Qwellyeah, I know what you meant
09:06.19FaithfulGot my USB Bluetooth adapter today... I'm ready for blue tooth extensions ;)
09:06.30TheEmperortuxinator_linux: oh, not really, I think all I have to do is to install sendmail and get it working and that will work right?
09:06.32Insanity5Qwell - A T-1 is digitally controlled, clarity, instant dialing, flexibility, ability to forge caller id to your main number, it's there.
09:06.48soulz-Mar  7 17:00:25 NOTICE[258]: chan_sip.c:7539 handle_response: Failed to authenticate on INVITE to '"Dinesh" <sip:xxxxx@sip.broadvoice.com>;tag=as498b8f00'
09:06.49QwellInsanity5: yeah, I've been well learned lately
09:06.57soulz-hello all, i seem to have get this lately on bv
09:07.06soulz-does anyone else have the same problem?
09:07.07Insanity5Qwell - Now if you had 8 bri circuits, it might be another story.  But still.
09:07.12Qwellhmm, bv really has been having problems this week, eh?
09:07.24soulz-yeah qwell
09:07.27Faithfulsoulz-: have you made the broadvoice patches?
09:07.28QwellInsanity5: Is there a reason to get 8 BRI vs a PRI?
09:07.52soulz-faithful: CVS-HEAD-02/17/05-02:39:03  its a new cvs
09:07.52Qwellbri is generally what, 2 lines?
09:07.55tuxinator_linuxTheEmperor: You will have to configure * to use it
09:07.58FaithfulQwell: if you only have a bri
09:08.14Faithfulsoulz-:  but you still need to patch for bv
09:08.15TheEmperortuxinator_linux: in the voicemail.conf?
09:08.16QwellFaithful: availability you mean?
09:08.35FaithfulPRI & BRI are different services
09:08.36tuxinator_linuxTheEmperor: I have not done it personally, but from looking at the wiki page, yes
09:08.39soulz-faithful: i heard that the patch is not needed for the new ones as i read on the wiki
09:08.39Qwellsomebody mind explaining something to me?  I'm still slightly lacking right here...
09:08.48soulz-it was working perfect until 3 days ago
09:08.49tuxinator_linuxBRI terminates differently
09:08.49footnoteFaithful: They don't have bluetooth here in georgia.
09:08.50mitchelocset canreinvite = no?
09:08.50GMsofthey guys, anyone have a working config to make asterisk authenticate with gnugk using SimplePasswordAuth ?
09:09.15soulz-and i didn't touch anything
09:09.15QwellWhen you get a PRI, what does the telco do for cabling?
09:09.15footnoteWe've got some GREEN teeth
09:09.15footnoteno blue ones
09:09.15QwellDo they actually have to send a new line to you, or?
09:09.15Faithfulsoulz-:  check the source...
09:09.17Insanity5Qwell - Well, it's better than 16 analog lines.  It depends on the cost and availability.
09:09.29FaithfulI had to patch when I set it up recently
09:09.41Insanity5Qwell - BRI = ISDN = digital = 2 lines generally.
09:09.42soulz-where did u do it from faithful
09:10.03Insanity5Qwell - Can also be used for 64k data on demand per channel.
09:10.03Qwellright
09:10.08soulz-http://edvina.net/broadvoice/broadvoicesip.txt ??
09:10.14Insanity5Qwell - It's a digitally controlled line, there's some advantages there.
09:10.17Qwellsee above about my cabling question?
09:10.19tuxinator_linux~BRI
09:10.20jbotextra, extra, read all about it, bri is the Basic Rate Interface , an ISDN access interface type composed of two B-channels each at 64 kbps and one D-channel at 16kbps (2B+D).
09:10.23Faithfulsoulz-: bv's asterisk setup page
09:10.28tuxinator_linux~PRI
09:10.29jbotit has been said that pri is Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, etc.
09:10.35Qwelloh, so BRI still has a D, I see
09:10.38Faithfulsoulz-: I had to hand patch it ...
09:10.42tuxinator_linuxQwell: both do
09:10.50Qwellyeah, I knew a PRI did
09:11.11NoCATcan i resell acess to my t1 with voip legally?
09:11.19footnoteOk, for extra points, why doesn't E1 total 32 slots?
09:11.28mitchelocif it's a business line
09:11.40Insanity5Qwell - D sends dialing/call waiting/etc data to the telco.
09:11.50QwellInsanity5: Yeah, I know. :)
09:11.50Insanity5NoCAT - Yes, if ytou can make money at it :)
09:11.53mitcheloc* i think, not sure, but i look at it like you can if it's business internet, but you can't if it's a residential internet line
09:11.55QwellI'm not a complete newb
09:12.16soulz-faithful: emm, thanks
09:12.21Insanity5ISP's use PRI's for dialin lines.
09:12.47QwellYou know, its funny
09:12.50Insanity5ISDN was supposed to eb in every household at the turn of the 90's... never happened.  It was to be the analog replacement.
09:13.06mitcheloccould someone evaluate this for me? i've been stuck on it since last night
09:13.07mitchelocexten               =>   7145154091,    3,        GotoIf(${ALERT} = "<http://127.0.0.1/Bellcore-dr3>" ? 100 : 4);
09:13.09Qwellseveral months ago, I thought a phone number could only have 1 call at a time.  I always wondered how ISPs worked in that regard, when they give you a local DID
09:13.21footnoteInsanity5: hayes almost went out of business the first time over ISDN not taking off
09:13.21mitcheloci'm trying to make that work...just a simple goto if
09:13.33FaithfulInsanity5: but DSL/VoIP will be...
09:13.35mitchelocbut it don't work...heh
09:13.39Qwellfootnote: So, why doesn't it equal 32?
09:13.41footnoteAnd that was when they OWNED the modem market
09:13.43tuxinator_linuxQwell: Hunt groups are the other option to DID
09:13.55Insanity5footnote - And then they did anyways.
09:14.12footnoteQwell: slot 0 overhead
09:14.19Insanity5Technology is amazing... I mean the POT has been around since before WWII.
09:14.20QwellSo...when you get a PRI, does it go over standard copper, or do they need to run new cabling to you?
09:14.21Qwellfootnote: ahh
09:14.34tuxinator_linuxQwell: T1 circut
09:14.38Insanity5Qwell - Copper.  Sometimes a new wire from the junction box, capacity permitting.
09:14.46QwellWhat is the junction box?
09:14.57Insanity5Qwell - Look outside.  Carefully.  It's close by.
09:14.58footnoteQwell: oops, slot 16
09:14.59Qwellwhere, I should say
09:15.02tuxinator_linuxThe box outsite your house, or any box on the way to the CO
09:15.12Qwelllittle green guy?
09:15.12footnoteit toggles from overhead to CCS i think
09:15.21footnoteheck, i'd have to go look now :)
09:15.45Insanity5Qwell - Typically disquised that way, yes.  Sometimes it can be cable/electric though.
09:15.55Insanity5Qwell - If you're in a very ritzy area, you might have to dig.
09:16.01Insanity5IE:  Underground :)
09:16.09Qwelland who usually pays for that?
09:16.14Qwellnot the telco, I would imagine
09:16.19Insanity5They do, with contract.
09:16.21Insanity5Generally.
09:16.23Qwelloh
09:16.31Qwelllarge contract, I assume?
09:16.37Insanity512 months at least.
09:16.39tuxinator_linuxQwell: When I got my T1 to my house many years back, they had to dig up my yard and put a box in my backyard because there wasn't enough capacity
09:16.40QwellSeems that only 2-3 years wouldn't cover the cost of digging
09:16.42Insanity5Depends on what you want to pay and the company.
09:16.52Duckbizkit<PROTECTED>
09:16.52footnoteDreaded words: "trenching required"
09:16.57footnoteesp PacHell
09:17.04Insanity5Qwell - A) It's an investment in the infracstructure in teh area B) Sooner or later, it will happen.
09:17.05QwellWhat if you have fiber in your area?
09:17.08Duckbizkiti took one drink and about spit it across the room. i added about 1/4 milk to it and it STILL tasted worse than any straight coffee i've ever had.
09:17.11QwellInsanity5: good point
09:17.16Insanity5Qwell - Out of your league.
09:17.23Insanity5Qwell - And if you're in a busienss district, it's probably out there.
09:17.26QwellInsanity5: well no, I mean...
09:17.35Qwellif there is fiber under you, will they use that instead of copper?
09:17.41tuxinator_linuxQwell: nope
09:17.45Insanity5tuxinator_linux - T-1 to residencial areas usually get through into the penalty box.
09:17.47Qwellthere are several areas around here where they laid fiber recently
09:17.53Insanity5Qwell - Can not, will not, no need.
09:17.57Qwelloh, ok
09:18.06footnoteyou can all use my global mesh network
09:18.10Insanity5HEhe.
09:18.20footnotefor a fee
09:18.28tuxinator_linuxInsanity5: It was when I was running a web hosting business out of my room.
09:18.34Insanity5tuxinator_linux - Business house of the house eh?
09:18.37mitcheloci say we make a wireless network from house to house, one fixed antenna on every house in the world, then we can let voip takeover!
09:18.43Insanity5tuxinator_linux - Now it's a $99 box at rackshack.net... hehe
09:18.58tuxinator_linuxYep
09:18.59footnotemitcheloc: call me when you get the routing schema worked out, i'll invest.
09:19.16Insanity5tuxinator_linux - It's amazing how bandwdith dropped in price.
09:19.18Qwellmitcheloc: 172.023 lng, 109.492 lat, 500' > sealevel
09:19.19footnotedon't violate any patents plz k thx
09:19.22Qwellmitcheloc: toss me a link
09:19.22mitchelocheh screw routing, a simpe linksys 4 port will do the trick
09:19.28footnoteuhm
09:19.34footnotenevermind :)
09:19.45*** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu)
09:19.56mitchelocfootnote: i was just kidding ;)
09:19.57tuxinator_linuxInsanity5: Yep, I'm getting a full T1 for $420/month through Netifice
09:20.03Insanity5tuxinator_linux - I remember when I threw up a game server on a T-1 in 1998, it was filled instantly due to demand.  I put one up in 2005 and there's like 50 million servers and 5 million players.
09:20.04footnotenugget must be getting old.
09:20.12Insanity5tuxinator_linux - Still using the T-1?
09:20.15tuxinator_linuxInsanity5: THey have a great SLA
09:20.20tuxinator_linuxto the house?
09:20.23Insanity5yes
09:20.23footnoteI can remember him hax0ring away until 6AM
09:20.30tuxinator_linuxInsanity5: Nope, long gone
09:20.33Insanity5Or for webhosting for any crazy reason.
09:20.44Insanity5webhosting needs to be moved to the bandwidth :P
09:21.16Insanity5Bandwidth is about three things:  Location, location, location.
09:21.20tuxinator_linuxyep
09:21.25mitchelocfull t1s? they are like $289 or something
09:21.29tuxinator_linuxNext to fiber is nice
09:21.29NoCATtuxinator there are still people who will pay to have private access to game servers
09:21.34Insanity5mitcheloc - Minus the pipe.
09:21.38footnotemove next to a fire station or hospital.
09:21.45tuxinator_linuxNoCAT: oh ya?
09:21.53footnoteNOT a police station.
09:21.57mitchelocInsanity5: not sure...i think my work pays only $289 total
09:21.57Qwellfootnote: firestations have bandwidth?
09:22.00Insanity5lol
09:22.04mitcheloc* ex-work =) (just quit)
09:22.08footnotethey might smell the pot...
09:22.12NoCATnot many, but i know some a few groups.
09:22.30Insanity5NoCAT - They do, and thats' why there's a proliferation of game servers.  I remember when it took more than paying $40/month to get a gameserver... namely it was bandwidth that was the issue.
09:22.30footnoteQwell: they're usually connected and have power outage privileges
09:22.32tuxinator_linuxThere are so many cheep T1 providers with crappy sevice
09:22.35tuxinator_linuxlike XO
09:22.39Qwellfootnote: power outage privs?
09:22.48footnoterolling blackout immunity
09:22.48Qwelllike, first back up?
09:22.51Qwellahh
09:22.51NoCATdoes qwest use interleaving on their t1 lines?
09:23.06Insanity5NoCAT - dsl, yes, minimum 60ms ping back.
09:23.13Insanity5There's a lot of half-duplex t-1's on the market today too.
09:23.16Insanity52 wire crap.
09:23.28footnotehuh?
09:23.28tuxinator_linuxeww
09:23.35Qwellfootnote: come to think of it, when I lived at my moms, we never got hit with a rolling blackout
09:23.45Qwellless then a block away from the fire station
09:23.51footnoteQwell: california?
09:23.53Qwellyeah
09:23.58Insanity5Qwell - Sometimes they'll just tag the circuit.
09:24.02footnoteyep, that was it then
09:24.06Qwellhmm
09:24.10Insanity5tuxinator_linux - Eww on what, teh two wire or latency?  hehe
09:24.13Qwellinteresting, never realized that
09:24.16Insanity5tuxinator_linux - Is level3 bandwidth good stuff?
09:24.20footnoteurm
09:24.22tuxinator_linuxhalf duplex
09:24.24QwellI doubt I'm close enough now. ;/
09:24.26footnoteit's not T1
09:24.27NoCATblackouts were fake
09:24.31Insanity5tuxinator_linux - Ya, it's nasty.
09:24.43footnoteNoCAT: yeah, cheney ken lay, et al
09:24.43tuxinator_linuxInsanity5: Level3 looks good
09:24.44mitchelocInsanity5: are you looking for hosting a server?
09:24.53Insanity5Have one hosted on :)
09:24.57Insanity5mitcheloc - Why?
09:24.59Insanity5hehe
09:25.04NoCATcan't forget arnold
09:25.09footnoteah-nald
09:25.11NoCATarnold for president..haha
09:25.12mitchelocoh, well www.calpop.com has the best bandwith, and worst reliability, just in case you were looking =p
09:25.36Qwellmitcheloc: let me guess, ex-work? :p
09:25.36Insanity5Well I can pull 20megabit... that's way more than I need.
09:25.40tuxinator_linuxmitcheloc: like cox, 5M d/l when its working
09:25.42footnoteI've already seceded my apartment.
09:25.52Insanity5I'll push 4gb a month, if I'm lucky.
09:25.58mitcheloci used to volunteer there, but the ex-work was somewhere else
09:26.01Qwellahh
09:26.05footnoteSovereign State of Footnote.
09:26.12Qwellso, ex-volunteer-work
09:26.19tuxinator_linuxKitty just laied a nasty poo, can't breathe
09:26.26Qwellfootnote: How near is that to the Federation of Qwell?
09:26.28*** join/#asterisk Cresl1n_ (~matt@68.159.151.148)
09:26.30mitchelocyea, but hey it was a great learning experience
09:26.32Insanity5tuxinator_linux - Better clean up the mess.
09:26.35Cresl1n_hey b's
09:26.36Cresl1n_:-)
09:26.39footnoteQwell: we can work out a trade agreement.
09:26.40mitcheloci learned so much in one year that it's just crazy
09:26.46tuxinator_linuxInsanity5: Litter maid, have to wait 10 minutes
09:26.58Insanity5Go hit the button!
09:26.59Insanity5lol
09:27.01footnoteQwell: have your people call my people
09:27.01Qwellfootnote: I'm not really one for international treaties
09:27.03Insanity5unplug-plug it in
09:27.04Insanity5hehe
09:27.06tuxinator_linuxInsanity5: No button
09:27.17Insanity5There has got to be a manual function.
09:27.21tuxinator_linuxInsanity5: That would work
09:27.27Cresl1n_exit
09:27.41tuxinator_linuxah man, what did she eat
09:27.45footnoteQwell: Ok, well I'll see you at the economic summit then
09:27.53Insanity5Whatever you feed her / forgot to put away.
09:27.55*** join/#asterisk ptblank (~MURDER1@68-169-176-29.lmdaca.adelphia.net)
09:28.02tuxinator_linuxAny of you going to Meet *?
09:28.09NoCATif you can only have 24 simultaneous calls on a voice t1.  does running voip allow you to recieve more calls?
09:28.19QwellNoCAT: If its not over the T1, sure
09:28.23tuxinator_linuxNoCAT: 23 calls
09:28.29NoCATsorry
09:28.31NoCAT23
09:28.34footnotehehe, back in the IBM mainframe days we called the '*' a 'splat'
09:28.35jmhunterkram
09:28.39Insanity5NoCAT - If they call your voip #
09:28.41Insanity5lol
09:28.53tuxinator_linuxNoCAT: VoIP can compress more calls in
09:29.02soulz-dial_exec_full: Unable to create channel of type 'SIP'
09:29.04mitchelocanyone here use an ibm laptop?, i'm lookin for a new one, and they caught my eye
09:29.06soulz-what does this error mean?
09:29.09Insanity5tuxinator_linux - Ih ave it when backbone providers forget to set a reverse dns, or leave an outdate reverse dns on a route of a company that they existed as many years ago.
09:29.11tuxinator_linuxNoCAT: but is it worth it?
09:29.20Insanity5mitcheloc - They're nice, durable, full features, lightweight, and EXPENSIVE.
09:29.37Insanity5tuxinator_linux - "ih ave it" = "I have it" :)
09:29.39Qwella decent thinkpad can run $2k easy
09:29.40mitcheloctrue, but worth the money?
09:29.43footnotesoulz-: after making a few calls?
09:29.43Insanity5GRRR - hate it :)
09:29.46Insanity5Can't flipped type
09:29.47Insanity5lol
09:29.48*** join/#asterisk Red_6 (~alex@m174.net81-66-29.noos.fr)
09:29.52mitcheloci was looking at the x40
09:29.52soulz-footnote: yeah
09:29.56*** join/#asterisk Delvar (~irc@83.146.53.34)
09:29.58tuxinator_linuxI like my toshiba
09:30.01mitchelocthey are going to have the new one in april come with the fingerpring scanner =)
09:30.05Insanity5t41 ot 42 or whatever it is looks nice.
09:30.07mitcheloc*fingerprint
09:30.14QwellI might like my Dell...if they would have sent me a freaking power cable
09:30.15footnotesoulz-: sounds sorta like something is hanging and not releasing channels?
09:30.16soulz-footnote: i have this error after i updated my cvs and also updated the invite on bv
09:30.26tuxinator_linuxI'm not a fan of Dells
09:30.28NoCATtuxinator what do you mean is it worth it?  i'm not sure how it works exactly.  it can compress more calls but your still limited to the number of calls you can recieve by the amount of availble lines on the t1
09:30.33Insanity5tuxinator_linux - How is time warner b/w?
09:30.34footnotedoesn't sound good
09:30.42soulz-tell me about it:P
09:30.51*** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu)
09:30.58tuxinator_linuxInsanity5: What about Time Warner?
09:31.00footnotei haven't figured out gdb usage yet
09:31.08footnotehaven't tried
09:31.18tuxinator_linuxNoCAT: You have to worry about sound quality and latency
09:31.19QwellNoCAT: If its going from your voip provider, to your * box, and out to internal extensions, it won't hit the T1 at all
09:31.28footnotea nifty set of * gdb macros would be a good thing
09:31.31NoCATtuxinator even on a lan?
09:31.34*** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de)
09:31.40tuxinator_linuxNoCAT: with a PRI, you dont' have those problems
09:31.46NoCATwhat is a pri?
09:31.51Qwell~pri
09:31.52jbotsomebody said pri was Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, etc.
09:32.14Insanity5tuxinator_linux - How is time warners bandwidth?
09:32.16tuxinator_linuxNoCAT: on a lan, you use SIP, AIX, etc
09:32.20footnoteNoCAT: complicated stuff the phone company created to prevent competition
09:32.34tuxinator_linuxInsanity5: Haven't lived in a Timer Warner area
09:32.54Insanity5tuxinator_linux - Sombody has to have a rate the bandwidth provider page... lol
09:33.15QwellInsanity5: like broadbandreports.com?
09:33.21tuxinator_linuxInsanity5: I was thinking about doing a rate a VOICE/INTERNET/ETC provider page
09:33.22mitchelocyep www.webhostingtalk.com
09:33.25Insanity5Qwell - Yes, but for major backbone providers.
09:33.29Qwelloh
09:33.42Insanity5level3/qwest/uunet/etc...
09:33.45QwellInsanity5: simply put, they all suck :p
09:33.47mitchelocwht is where the serious people go and talk about the stuff
09:33.50Qwell^^overgeneralization
09:33.53tuxinator_linuxUUNET network is nice
09:33.59tuxinator_linuxUsed it while using MCI
09:34.16tuxinator_linuxGenuity was bought by Netifice
09:34.27Insanity5genuity godo or bad?
09:34.30tuxinator_linuxgood
09:34.39Insanity5Used to have them before level3
09:34.44Insanity5In the whole transition mess.
09:34.57tuxinator_linuxInsanity5: Who did you have?
09:35.20mitchelocif you want real solid bandiwth though, you have to go straight to a provider, like pacbell, att, something like that
09:35.20tuxinator_linuxI had PSInet to the house
09:35.22mitchelocnot a reseller
09:35.26QwellGenuity ha[sd] good bandwidth, right?
09:35.37Insanity5tuxinator_linux - Genuity direct for b/w in a chicago datacenter, ran our own transport back.
09:35.39tuxinator_linuxmitcheloc: I dont' think that is entirely true
09:35.43Insanity5tuxinator_linux - Then level3./bbnplanet
09:35.57*** join/#asterisk pif (ldm@zenon.apartia.fr)
09:36.07*** part/#asterisk Duckbizkit (~DMAN@ip-216-97-163-53.valornet.com)
09:36.09mitcheloci don't know, i think it's pretty accurate, resellers add one more point of possible failure
09:36.11tuxinator_linuxlevel3 owns a lot of fiber
09:36.21Insanity5Yes.
09:36.29tuxinator_linuxmitcheloc: but they get the ILEC to be more responsive
09:36.29Insanity5I wonder if latency will ever get lower, or if speed of light is the limit.
09:36.33Insanity5Gah :)
09:36.44QwellInsanity5: well, it is fiber. ;]
09:36.48Insanity5hehe
09:36.49tuxinator_linuxmultiplexing light
09:37.03Insanity5Yes...  Never becoming any faster...
09:37.03mitchelocsweet!!! just finished setting up my door, a $2/month broadvoice # lets me in/out of my front door =)
09:37.08tuxinator_linuxvarious bandwidths of light
09:37.17Qwellits also going through a bunch of silicon, isn't it?
09:37.22mitchelocnext stop is my car, i hate keys!
09:37.23Insanity5But chicago > LA over timewarner taking 75 ms is high.  They should have cross-country down to 40-50ms.
09:37.30*** part/#asterisk Red_6 (~alex@m174.net81-66-29.noos.fr)
09:37.33Insanity5mitcheloc - LOL, you geek.
09:37.49tuxinator_linuxAhhh, kitty poo is gone
09:37.51Insanity5mitcheloc - Any authentication, or just call it?  hehe.
09:37.54mitchelocheh, the worst part is that i have 6 roommates and i forced them all into using their cell phones to get into the house =)
09:38.05Qwellwtf
09:38.09Qwellwhat if bv is down? :P
09:38.11Insanity5mitcheloc - Better carry a spare battery.
09:38.14mitcheloccallerid auth, and yes i know it's fakeable, but you have to know the number and the numbers to fake too
09:38.18tuxinator_linuxmitcheloc: weirdo, he he
09:38.23*** join/#asterisk RoyK (~roy@80.239.107.121)
09:38.35mitcheloci have a back up number too that i was using for a while now, it's off a t1 in another city
09:38.46mitchelocbut just switched it to BV cause it answers faster
09:38.46QwellI hate to ask...
09:38.49Insanity5Better still hide a key.
09:38.51Qwellbut what if both are down?
09:38.59mitchelocthen they can't come in the house, too bad
09:39.02tuxinator_linuxHe could just break a window
09:39.04Qwellheh
09:39.10Insanity5Rock? :)
09:39.11Qwellor turn the knob manually
09:39.17tuxinator_linuxQwell: too much work
09:39.22QwellInsanity5: cellphone would already be in hand
09:39.25mitchelocnope, the knob is fixed, i used an allen wrench to make quick work on that
09:39.27Qwellcellphone would be a good weapon
09:39.31Insanity5You should rig it so the door springs open :)
09:39.38mitchelocheh ;)
09:39.42Insanity5Qwell - Waste of money, cmon now, that'd cost more than a new window.
09:39.45mitcheloci don't need no backup!
09:39.46NoCATdoes asterisk do sms over internet to mobile phones?
09:39.49tuxinator_linuxdoes it great you when you open the door
09:39.59tuxinator_linuxNoCAT: maybe
09:40.03Qwell"Hello Tom, breakfast is ready."
09:40.03Insanity5NoCAT - Could probably rig it too.
09:40.07mitchelocyea it goes BEEP then a pause, about 8 seconds then allison says "THANK YOU"
09:40.08mitchelocand hangs up
09:40.11tuxinator_linuxNoCAT: I think there is a wikie for it
09:40.15Insanity5usually @messaging.sprintpcs.com or @yourprovider.com does the trick.
09:40.34Insanity5mitcheloc - Waht hardware did you use?
09:40.42mitchelocmmm i went the expensive route
09:40.45Insanity5Not true sms, but a gateway over.
09:41.06mitcheloca rabbit + simple relay circuite, a little c programming, and a heavy duty strike from smarthome.com
09:41.15mitchelocoh and a lot of time installing the damn thing
09:41.15NoCATyeah i'm sorry i shoudln't ask questions i can easiy research.. been reading alot lately.
09:41.18mitchelocthat takes forever
09:41.25*** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu)
09:41.33Insanity5People here are VERY forgiving :)
09:41.43tuxinator_linux~rtfw
09:41.44jbotmethinks rtfw is Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php
09:41.54Qwellmitcheloc: rabbit with, or without hamster wheel?
09:41.54tuxinator_linuxno forgivness for you
09:42.13tuxinator_linuxjust kidding NoCAT
09:42.19mitchelocrabbit without hamster wheel =), mines a smart rabbit, plug her into the lan and access her from wherever
09:42.25Insanity5Does anybody have a really neat routing system set up they want to showcase? lol
09:42.25Qwellnice
09:42.43QwellInsanity5: voip, or lan?
09:42.49Insanity5Qwell - Any.
09:42.57QwellI have a crappy routing system. :p
09:43.01mitcheloci have a gps, it routes me well
09:43.03tuxinator_linuxhmm, I'm going to go steal some cookies, be back in a minute
09:43.07*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com)
09:43.09Insanity5there's got to be a good reposity of good voice samples (hint: sexy female voice) somewhere :)
09:43.19Qwellsparcstation with a quad sbus ethernet card - 4 different subnets
09:43.22mitchelocheh...and i thought he meant browser cookies for a second
09:43.29moonwickdo what I do.  have your girlfriend record the samples.
09:43.34mitchelocInsanity5: your welcome to pay my gf ;)
09:43.47Insanity5lol
09:43.51Qwellmitcheloc: dating allison, are we?
09:43.54mitchelocInsanity5: she does a good job, and i have a good phone for her to use to record.
09:43.58mitchelocbah!
09:44.02Insanity5Automated systems must have females answering :)
09:44.04mitcheloc** eww no thanks
09:44.16moonwickI've actually found that the perfect microphone for recording sounds for asterisk is just a pair of headphones.
09:44.17QwellI need sleep
09:44.18Insanity5mitcheloc - I figured a headset would be best... or .wav files
09:44.25mitchelocyea, it's the only relief we have from not seeing girls while sitting infront of the computer debugging asterisk heh
09:44.46mitcheloci prefer a cisco 7960 for recording
09:44.57Qwellmoonwick: heh, you just reminded me of a funny story
09:44.59Insanity5Do you have * pick up everything even before it rings (HOLD, CONENCTING CALL), does it get rid of telemarketers, etc?
09:45.20mitcheloctelemarketers are fun to talk to, why wouldn't you want them?
09:45.24Qwellmoonwick: verizon shut me off once, but I still had DSL.  I had to use dialpad.com(back when they were free) to call verizon to get my phone back.
09:45.34Qwellmoonwick: But, I didn't have a mic, and I owned one pair of headphones
09:45.57mitchelocrecord a track that goes "uh huh.......mmm.....yea.... that sounds good..... maybe.....mmm.....sorry...can you repeat that?"
09:46.00QwellI would nod to my friend, and he would switch from mic to speaker port...ugh
09:46.00mitchelocthen transfer them there
09:46.14Insanity5mitcheloc - LOL, do you do that?
09:46.17tuxinator_linux<PROTECTED>
09:46.20mitchelocno, but i should
09:46.23Insanity5mitcheloc - I would love to place them in a satiracle hold loop.
09:46.43moonwickheh
09:47.43tuxinator_linuxmitcheloc: How did it workout?
09:47.49mitchelocwell......if you screw up the asterisk box, you can't connect to it, cause you lose the t1 connection
09:47.56mitchelocother then that it works great
09:48.17tuxinator_linuxI thought about that
09:48.27mitcheloci did that
09:48.33tuxinator_linuxThat is why I am getting two T1 circuits and and PRI
09:48.36mitchelocbig big pain
09:48.40Insanity5mitcheloc - call 1-866-836-0971 - lol  pick an option
09:48.45tuxinator_linuxtwo T1 for BGP
09:48.55mitchelocby integrated i mean half hdlc data and half phone lines
09:49.01*** join/#asterisk Bonbon (~bonbon@83.146.53.34)
09:49.22*** join/#asterisk benno2 (~benno2@host31-15.pool80182.interbusiness.it)
09:49.30tuxinator_linuxmitcheloc: I know, that is what I have right now through XO, and it never stays up
09:49.42Bonbonhas anyone done any tapi integration with xlite?
09:49.45*** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu)
09:49.49footnoteboy, this is a really scary pic of tux
09:49.57footnotehttp://www.satanic.org
09:49.57mitchelocscratchy
09:49.58Zeeekhey drumkilla?
09:50.00Insanity5mitcheloc- I like option 8 :)
09:50.09mitchelocheh
09:50.16Insanity5mitcheloc - Did you call it?
09:50.19tuxinator_linuxI called it to
09:50.24fileha
09:50.28mitchelochey my gf should do your voices, cause that chick hurts my ears
09:50.33footnotemmm brownies
09:50.34mitchelocwhats with the monkeys?
09:50.37Insanity5It's not mine :)
09:50.38Insanity5rofl
09:50.51Insanity5Came from this:  http://216.239.63.104/search?q=cache:7WPkq6zWSqQJ:www.careerbuilder.com/JobSeeker/Jobs/JobDetails.aspx%3FJob_DID%3DJ8F1576JD0Y1PM5Q4F2+poo+careerbuilder&hl=en
09:50.55filemitcheloc: ah yes your bugnote... it's because asterisk does blocking DNS lookups, no way not to at the moment - so make sure you have stable DNS servers
09:50.56mitchelocthats a real good quality call, what do you use?
09:51.07*** join/#asterisk visik7 (~ciao@visik7.user)
09:51.17Insanity5here's their webistE:  http://www.yeknominc.com/
09:51.19Insanity5lol
09:51.41mitchelocfile: oh i'm, i'm sorry, then you knew about it, i definately use more stable dns servers, just didn't know if anyone knew about this
09:51.47*** join/#asterisk TheJudge (JTR@209-203-52-3.network.ods.co.za)
09:51.57mitcheloc* will use
09:51.59Insanity5mitcheloc - It's not mine... I was just remembered it from a while back.
09:52.04benno2question: is it possible to define an extension that if it gets called, the called party (not the caller) first hears a short message and then the call gets connected. that way the called person can know where the call comes from. (distinctive ring is not an option because the extension could be redirect on a cellphone)
09:52.23mitchelocfile: i guess mostly it was just it was hard to figure out what was the source of my problems, perhaps a comment in the notice to point users the right way
09:52.40TheJudgehello all
09:52.48TheJudgewhat sould I set my jitter size to
09:52.49TheJudge?
09:52.53TheJudgeor how does it work
09:52.57Insanity5mitcheloc - Was your comment about the good voice from the yakmov industries joke?
09:53.14fileeveryone else is going to sleep, I should do the same I suppose
09:53.16tuxinator_linuxI hate being on hold
09:53.24mitchelocInsanity5: no, just the quality of the call
09:53.25tuxinator_linuxthe monkeys are interesting though
09:53.25*** join/#asterisk RoyKa (~roy@80.239.107.80)
09:53.37filebkw is dead to the world
09:53.40filetwisted is the same
09:53.43fileonly me left
09:53.45Insanity5tuxinator_linux - lol.  It's almost a game.
09:53.51*** join/#asterisk chaven (~cbalzac@evil.maas-biolabs.net)
09:53.57Insanity5tuxinator_linux - They even have tv ads on their fake website :P
09:54.03mitchelocfile: we can keep you company ;)
09:54.09*** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de)
09:54.12filehehe
09:54.20TheJudgeI am using broadvoice
09:54.28TheJudgehow do you work out jitter
09:54.29TheJudge?
09:54.36Insanity5TUX - I want something like that to transfer a telemarket too though.
09:54.39mitcheloci bet they are resting for von tomorrow
09:54.44footnoteTheJudge: empirically or mathematically?
09:54.54TheJudgelol
09:54.57TheJudgeeither
09:55.01TheJudgewhat does it do ?
09:55.03*** part/#asterisk jmhunter (~jmhunter@64.77.199.223)
09:55.04footnotewell, is it working?
09:55.16TheJudgecall is jittery ?
09:55.22TheJudgelike static on the line ?
09:55.27footnotepops like slips
09:55.36tuxinator_linuxInsanity5: Me too
09:56.15RoyKafootnote: using what? sip?
09:56.15tuxinator_linuxThose videos were on during the superbowl
09:56.35mitchelocwhat was the purpose of the website then?
09:56.54footnotea "slip" happens when your amount of jitter is larger than the buffered amount of time
09:56.59footnoteor something like that
09:57.02TheJudgeok
09:57.12footnoteso, make it twice as big
09:57.16footnotesee what happens
09:57.21Insanity5tuxinator_linux - They also placed job descriptions like "flung poo collector" thorughout their site :P
09:57.29footnotei haven't actually done this for asterisk yet :)
09:57.32TheJudgeI have cisco router, how can I get it to prio sip trafffic ?
09:57.34mitchelocfootnote: whats the link?
09:57.41Insanity5TheJudge - By port?
09:57.54fileoh no bkw is snoring
09:58.00footnoteTheJudge: paypal Insanity5 he's got Cisco scoop
09:58.01*** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu)
09:58.15Insanity5rofl
09:58.24Insanity5Cisco is just a PITA.
09:58.29footnoteInsanity5: i expect a kickback this time dammit
09:58.38Insanity5hehe
09:58.46TheJudgecool, but what port ?
09:58.54TheJudgedoes it not choose a random port ?
09:58.55Qwellfile: Now seems to be the perfect time to mess with him
09:59.07Qwellok, bed...for real this time
09:59.14filelol
09:59.14*** join/#asterisk mithro (~tim@dsl1-83.gw1.adl1.airnet.com.au)
09:59.38tuxinator_linuxQwell: what are going to do with this 'bed'?
09:59.39Insanity5TheJudge - Use it based on server port.  At least with most, you can't due true QoS on the application level, but you can infer that voip SIP traffic will stay on it's port.  Add it to an access list, and give it priority.
09:59.44tuxinator_linux~bed
09:59.45jbotbed is, like, a thing programmers have never heard of, ask me about shower
09:59.55footnotemithro: is it true that vegemite makes good axle grease?
10:00.03mitcheloc~shower with me
10:00.05tuxinator_linuxfootnote: sure do
10:00.07benno2ok found the solution, cmd dial: A(x): Play an announcement (x.gsm) to the called party.
10:00.08benno2:)
10:00.10Insanity5And I'm assuming you're talking about a router.
10:00.11TheJudgetahnks
10:00.14TheJudgeyes
10:00.15mitchelocmmm jbot didn't like that
10:00.17mitcheloc~shower
10:00.18jbotextra, extra, read all about it, shower is man using one hand in a very usefull way
10:00.31mitcheloc~hand
10:00.32TheJudgeso eg port 41000 and give it priority
10:00.41RoyKa~lart mitcheloc
10:00.41mitcheloc~useful?
10:00.53mitcheloc~thank you?
10:00.53jbotpas de quoi, mitcheloc
10:01.05mitcheloc~mitcheloc
10:01.08Insanity5yes, and you'll have to classify "everything else" in another access group.  Cisco is quirky, but you set two numbers and it'll get say, 7 packets of one to 1 packet of another.
10:01.24tuxinator_linuxjbot habla espanol?
10:01.38footnotejbot not messican
10:01.49Insanity5TheJudge - If you enver a newer IOS, try looking at Weighter fair queueing.
10:01.57mitchelocstatus
10:02.01fileI should plug my laptop in
10:02.21tuxinator_linuxfile: Good Idea
10:02.23TheJudgethanks
10:02.29fileand sleep
10:02.32footnoteI'm finally updating my PowerBook to 10.2
10:02.35tuxinator_linux~sleep
10:02.36jbotfrom memory, sleep is overrated, and a poor substitute for caffeine
10:02.45tuxinator_linux~mac
10:02.46jbotmac is, like, the best computer ever
10:02.56tuxinator_linuxhmmm
10:03.05footnotejbot is a genius!
10:03.19Insanity5TheJudge - Start by making two access control lists, everythign else, and the voip traffic and go from there.
10:03.24tuxinator_linuxOne silly button on the mouse??
10:03.24mitchelocjbot is smart
10:03.25*** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu)
10:03.27mitchelocjbot: what is 60 mph in angstroms per femtosecond?
10:03.29jbotmitcheloc: I think you lost me on that one
10:03.31TheJudgeok eg
10:03.32mitcheloc~what is 60 mph in angstroms per femtosecond
10:03.34jbotI think you lost me on that one, mitcheloc
10:03.39TheJudgematch iup all all in one
10:03.41mitcheloche answered me in private!
10:03.42footnotetuxinator_linux: mainly because it's BSD based and not derivative of linux
10:03.43Insanity5TheJudge - Also just enablign "fair-queue" can sometimes help a bit.
10:03.45mitchelocnow he won't do it in public =/
10:03.46TheJudgeand voip in the other
10:04.03mitcheloclets start off simple
10:04.06Insanity5TheJudge - YOu got it... do it by port number or server ip.
10:04.09mitchelocjbot: what is 2+2?
10:04.10jbot4
10:04.13mitchelocgood boy
10:04.21mitchelocjbot: now what is 2*4?
10:04.23jbotmitcheloc: okay
10:04.23TheJudgedo you know what port broadvoice use ?
10:04.30mitchelocjbot: WHAT IS 2*4?
10:04.32jbotmitcheloc: what are you talking about?
10:04.36rikstahahaha
10:04.40mitchelocjbot: 2*4?
10:04.44footnoteuh oh, it's in eliza mode
10:04.50mitchelocdamnit! YOU MECHANICAL USELESS MONSTER!
10:04.57mitchelocjbot: byte me!
10:04.59jbot/me bytes mitcheloc
10:04.59tuxinator_linux~cry
10:05.00jbotACTION cries and sobs until he nearly drownds in his own tears
10:05.00Insanity5TheJudge - It's SIP.  google it.
10:05.05mitchelocnoo!
10:05.18rikstamitcheloc: if it's any help...i think the answer is 8 ;)
10:05.25mitchelocnickometer bkw
10:05.30tuxinator_linux~kill jbot
10:05.33jbotACTION shoots a ionized fluxproton gun at jbot
10:05.42Insanity5TheJudge - You may also just be able to associate the class map with the protocol instead of relying on an access list... but I seriously doubt it.
10:05.46mitchelocriksta: thanks, but i'm tring to teach jbot, he will learn!
10:05.48tuxinator_linuxnot a very smart monster
10:05.56footnoteoh, kill the messenger!
10:05.58footnotehehe
10:06.03Insanity5TheJudge - If you go the WFQ route.
10:06.07TheJudgeyea
10:06.08TheJudge?
10:06.09mitcheloctheres got to be something useful on this page so that we can destroy jbot!!!! https://jbot.dev.java.net/jbot-user-guide.html
10:06.17TheJudgethanks
10:06.19*** part/#asterisk chaven (~cbalzac@evil.maas-biolabs.net)
10:06.26footnotemitcheloc: oh, it's a java bot
10:06.35footnoteit'll self destruct, be patient
10:06.50Insanity5TheJudge - But I don't think the router will allow SIP as a protocl.   I seriously don't know.
10:07.02tuxinator_linux~bullshit
10:07.03jbotfrom memory, bullshit is If you want to speak bullshit, please go to #debian.bullshit.  sdf dflkj Linux sucks sfg yo momma dfg #debian.bullshit
10:07.07mithrofootnote: dunno, i'd rather eat axle grease then vegemite
10:07.12footnotehahahah
10:07.25Insanity5TheJudge - If you have neough, you may jsut want to dedicate enough bandwidth soely to the VOIP cause.  IF not, let the router handle availability.
10:07.30mitchelocinsult footnote
10:07.31footnotesounds like jim dixon must have set jbot up
10:07.33mitcheloc~insult footnote
10:07.39footnotehaha
10:07.50mitcheloc~insult jbot
10:07.53*** join/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com)
10:08.03footnote~rape mitcheloc
10:08.07footnoteeep
10:08.15tuxinator_linux~qos
10:08.16jbotextra, extra, read all about it, qos is Quality of Service, a great source of information is located @ http://www.lartc.org
10:08.16mitcheloc~ebonify footnote
10:08.17footnotehehe
10:08.18*** join/#asterisk The_Ball (~alex@static-100.35.240.220.dsl.comindico.com.au)
10:08.27mitcheloc~ebonify
10:08.36footnoteSlap mah fro!
10:08.44Insanity5TheJudge - create acl -- ie access list permit udp all all range (sip port start here) (sip port end here).  check the syntax of the all all part, it's probaby wrong.
10:08.44tuxinator_linux~ebonify I need to get some sleep pretty soon
10:08.58The_BallHi, can anyone help me with a newbie dialplan question, here is my five line dialplan: http://channels.debian.net/paste/322   how can i make the phone start to ring at the very begining?
10:08.58footnotehrm    jive | festival for asterisk
10:09.02mitcheloci need to a get uh some uh sleep uh soonah
10:09.23mitchelocnevermind, *sleep=sex
10:09.31Insanity5TheJudge - Create class maps that encompass each access list -- class map class1^M Match Access-group 101^M  (^M= enter).
10:09.48tuxinator_linux~time
10:09.49jbotrumour has it, time is 1 dimensional, or everlasting
10:10.08footnote~quantum mechanics
10:10.11jbotmethinks quantum mechanics is the devil's arithmetic
10:10.11mitchelocjbot: time is the 4th dimension stupid!
10:10.13jbot...but time is already something else...
10:10.32mitchelocjbot: NO it's not!
10:10.42mitchelocjbot: time is change
10:10.43jbot...but time is already something else...
10:10.43tuxinator_linuxjbot is a dork
10:10.49footnote~particle physics
10:10.53mitchelocjbot: change is the only constant
10:10.59mitchelocYES IT DOES!
10:11.09tuxinator_linuxjbot, he is right
10:11.11jbot...but he is already something else...
10:11.23Insanity5TheJudge - Easy way?  flat out limit all to less than max b/w, and give voip suffient b/.w - policy map policy1^M (now you have a policy map).  class class1^M (class1 we just made, voip traffic) bandwidth 80000^M (80kbit) Queue-limit 30 (play with this value).  exit.   Repeate for class two.  Pray it works.
10:11.27tuxinator_linuxjobi: you want to fight about it
10:11.45tuxinator_linuxjbot: you want to fight about it?
10:12.07mitchelocjbot: i'm going to sleep, and i shall kill you in my dreams, good bye
10:12.08Insanity5TheJudge - IF you want to get into actually prioritizing and droppign traffic... well good luck it's beyond most of my abilities.  Start reading here:  http://www.cisco.com/univercd/cc/td/doc/product/software/ios120/120newft/120t/120t5/cbwfq.htm
10:12.14tuxinator_linuxInsanity5: What lanauge are you speaking?
10:12.27Insanity5tuxinator_linux - Cisco :0
10:12.29Insanity5IOs
10:12.34tuxinator_linuxEwww
10:12.38mitchelocInsanity5: qos on a linux router?
10:12.45TheJudgeta
10:12.45Insanity5mitcheloc - No... Cisco IOs
10:12.49footnoteI saw the source for IOS when I was at Cisco.
10:13.02mitchelocInsanity5: - No... i mean, USE a linux router for qos
10:13.02footnoteWhich is why I'll never buy a Cisco product.
10:13.06Insanity5footnote - Now it's on your local peer2peer network lol
10:13.11footnotenope
10:13.16Insanity5mitcheloc - What a concept... I know... he wants to use his cisco.
10:13.21Insanity5footnote - An old version is.
10:13.22tuxinator_linuxI have had the best luck with Cisco routers
10:13.33footnoteIt's the ugliest piece of crap you'll ever look at
10:13.43Insanity5footnote - I beleive you, but it sure is stable.
10:13.50Insanity5Very draconian at times.
10:13.51footnotewell
10:14.02footnoteon some products
10:14.23footnoteAvoid powerpc based cisco products
10:14.30mitcheloci bought a dlink router...or was it netgear
10:14.36mitcheloc10/100/1000 16 port switch
10:14.46mitchelocmanaged, with qos =)
10:14.49footnotepowerpc-based==acquisition
10:14.50mitcheloci'll be here thursday
10:14.59Insanity5mitcheloc - Yes, with an easy to use gui, lol
10:15.03mitchelocthats going in my house, overkill? heh!
10:15.07footnotemips base, "native" cisco
10:15.11mitchelocyep, webmanagement =)
10:15.51Insanity5mitcheloc - wrt54g linksys route with hacked firmware here does the job as good as any:  http://wrt54g.thermoman.de/
10:16.33mitchelocyea but it's not a gigabit switch
10:17.20footnotemitcheloc: cmon, you need gigabit for pr0n?
10:17.24mitchelocwhat WOULD be nice mmm..../me drools, can you use the wrt54g as a wireless router and trigger in out pins somehow on the ports? so i can turn on and off relays via a cheap ~$50 single board computer! =)
10:17.26Insanity5mitcheloc - No, it's consumer junk lol.  but it works.
10:17.40rikstamitcheloc: yeah the wrt54g is good
10:17.42mitchelocyes, for porn
10:18.02Insanity5mitcheloc - Take a look at these:  http://www.parallax.com/html_pages/products/basicstamps/basic_stamps.asp
10:18.07Insanity5mitcheloc - Little computer for those jbos.
10:18.08rikstapersonally, i made my own access point
10:18.08Insanity5jobs
10:18.12mitcheloci have to be able to stream porn to every cat5 jack in my house
10:18.15mitcheloc* at the same time
10:18.25Insanity5hehe
10:18.30Insanity5In your VR world.
10:18.34mitchelocnonono not LAN aware!
10:18.36footnoteriksta: really?
10:18.54rikstafootnote: yeah, if you buy a decent wireless card, you can put it in "master" mode :)
10:18.54mitchelocriksta: pci wireless card + linux router
10:18.55djinpbx.c:1280 pbx_extension_helper: No application 'SerVar'
10:18.58footnoteoh
10:19.04djindid I miss something that changed?
10:19.09mitchelocdjin: setvar not servar
10:19.13rikstafootnote: then you just set a wep key, and an IP and run a dhcp server
10:19.19djinhahha
10:19.22footnotegotit
10:19.40footnoteriksta: i thought you laid out your own board or something :)
10:19.42Zeeekdjin don't be so hardon yourself
10:19.43mitchelocInsanity5: what do you know about cheap computers that i can plug into my lan, that aren't computers =)?
10:19.47rikstafootnote: naw
10:20.04Insanity5mitcheloc - Get some flash disk contraption if that's what your'e really after.
10:20.18mitcheloc*cheap*!!
10:20.28Insanity5mitcheloc - You can interface with one of those stamps via serial.  Just jack a couple pair of wire off your ethernet wiring.
10:20.36Insanity5Stamps are cheap.
10:20.39footnoterabbit
10:20.42footnotez80 like
10:20.43footnotetcp
10:20.52mitchelocyea i use the rabbit
10:20.59mitchelocbut i was hoping for another cheaper alternative
10:21.03djinWell Zeeks. Such a stupid mistake, but overlooked because it's in a longer list of Setvar's.
10:21.05footnoteweird c-like thingy eh?
10:21.05Insanity5They'rel ike $20
10:21.06mitchelocrabbits are my friends =)
10:21.07mithroanyone in Australia using a FXO/FXS stuff?
10:21.20djinZeeks = Zeeek, sorry.
10:21.20Insanity5mitcheloc - I have piles of pentium 2 computers.... pay shipping or you haul... $20
10:21.29Insanity5computers are free lol
10:21.33ZeeekThere are so many Zeeeks
10:21.34mitchelocInsanity5: so stamp + wrt54g??? that would be sweet
10:21.41footnotehrm
10:21.45mitcheloc*** let me change that, cheap and SMALL
10:21.46Insanity5Probably could interface by serial :P
10:21.51footnotepic basic
10:21.59Insanity5and they're easy adn painless to code.
10:22.32Insanity5mit - There http://www.rs485.com/pespsx3.html
10:22.47Insanity5ethernet > serial controller thingy.
10:22.51mitchelocooooooh /drool
10:22.59footnoteI wish I could get a little TMS320 on a credit card board
10:23.06footnotecheep
10:23.07mitcheloccost?
10:23.25Insanity5Dont' know, probably a couple hundred.
10:23.30mitcheloc$300!!!
10:23.35footnotesheesh
10:23.43mitchelocNO THANKS! rabbit, i'll keep
10:23.45footnoteyou can get a terminal server cheaer
10:23.50footnoteer cheeper
10:23.55Insanity5mitcheloc - But somebody has to make one somewher.e
10:24.07footnoteit's for legacy retrofits
10:24.13Insanity5It wasj ust the I'm feelin lucky button on google.com with ethernet and serial entered in.
10:24.16Insanity5Anyways goodnight :)
10:24.20footnotelike a DMS100
10:24.25mitchelocgood night
10:24.38mitchelocterminal server cheaper?
10:24.46mitchelocthin client you mean?
10:25.01footnotenaw a junker terminal server on ebay :)
10:25.01Insanity5mitcheloc - http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=11175&item=5756115258&rd=1
10:25.12ta[i]ntedanyone here have DID through BV?
10:25.21Insanity5mitcheloc - better yet, try this one:  http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=11175&item=5757661249&rd=1
10:25.26ZeeekBV has had problems with asterisk recently
10:25.28Insanity5erial To Ethernet Interface - Network Everything!!
10:25.34mitchelocwhat does that do?
10:25.35ta[i]ntedZeeek do you have DID through BV?
10:25.48Zeeekno but a lot of discussion here about recent issues fwiw
10:26.05*** join/#asterisk pratik (~root@202.149.48.214)
10:26.05Insanity5mitcheloc - Ethernet to serial with documentation.  Only $30 BIN to boot.
10:26.09mitcheloci'm going to googlerape "terminal server" tomorrow and learn it all
10:26.11mitcheloc$30 is cheap
10:26.15Zeeeksomeone even asked to put a link the title
10:26.16footnotehrm, is #asterisk logged?
10:26.31mitchelocmy irc client logs it ;)
10:26.37Bonbondoes anyone use the low-end cisco phones with * ?
10:26.40Zeeekfor BV stuff check the mailing list
10:26.41Insanity5footnote - Why?  lol
10:26.45mitcheloci'm sure most peoples do too
10:26.52RoyKahmmmm. what's the cheapest digium shite I can use for a timing source?
10:26.57mitchelocBonbon: lowend =?
10:26.58footnoteInsanity5: for searching
10:27.05Insanity5footnote-  ahh, ok.
10:27.07pratikhello everyone
10:27.09footnotethat's the trouble with IRC
10:27.16footnoteit just goes *poof*
10:27.20RoyKa~lart Zeeek
10:27.29footnotemailing lists are better for important stuff
10:27.30Insanity5Ya, go punch your nick (if unique) in google, it's fun :)
10:27.32Bonbonmitcheloc - 7905 / 7910
10:27.47mitchelocyea, they work perfect
10:27.51mitcheloccrappy thing is they don't have mics
10:28.09*** join/#asterisk RoyK (~roy@80.239.107.80)
10:28.10RoyKhmmmmmm
10:28.11Insanity5bye :)
10:28.12pratiki am constantly getting this error "Mar  7 16:01:06 WARNING[4164]: chan_sip.c:728 retrans_pkt: Maximum retries exceeded on call 376a6eb97378753c1f38252867c4273f@127.0.0.1 for seqno 121 (Critical Request)
10:28.13pratik"
10:28.16Insanity5mitcheloc - It's $30... just play with it
10:28.30rikstapratik: looks like it isn't  registering
10:28.53pratikbut not registering with what
10:29.11RoyKAnyone here doing large numbers of SIP traffic with asterisk?
10:29.13Bonbonmitcheloc - thanks. so  you mean there is no speakerphone?
10:29.25mitchelocthere is a speaker
10:29.27mitchelocbut no microphone
10:29.30BonbonRoyK: what's a "large number"
10:29.38Bonbonmitcheloc: so we can only monitor?
10:29.42mitchelocyes
10:29.45RoyK100 concurrent calls or so?
10:29.47rikstamitcheloc: are you sure about that, i thought there was a mic
10:30.01RoyKover WAN
10:30.46pratikwhat should i do?
10:30.57BonbonRoyk: yes, we do that
10:31.03rikstapratik: do you have a sip provider
10:31.16mitchelocriksta: maybe your right, maybe it's the 7912..., i'll go look under my bed, i've got one there
10:32.17pratikyes, i tried registering with my own server
10:32.39mitchelocyep yep, it all came back to me now, 7905 has one cat5 jack and 7910 has pass through, but same model, and no microphones
10:32.50mitchelocand i don't think there is a 7912..not sure what i was thinking there
10:32.52pratikbut then i removed it from the registration
10:33.11rikstamitcheloc: hehe...i thought so :)
10:33.16pratiki.e i removed it from the sip.conf and extensions.conf
10:33.49footnotehrm
10:33.53footnotem4
10:34.28pratikand when i see the "sip show peers" it says that the status to that server is unmonitored
10:34.33*** join/#asterisk Ubuz (~momo@CBL217-132-84-235.bb.netvision.net.il)
10:35.35mitchelocok wow i'm tired, good night
10:35.57Ubuzhelp needed - I installed the latest stable version as instructed in the web page, installed everything and now asterisk will not load with an error: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/pbx_dundi.so: undefined symbol: ast_config_load
10:38.05pratikriksta:what is the possible solution to this
10:38.19rikstadepends what you are trying to do
10:38.31rikstaif you took out the register, you need to reload
10:38.33The_BallHi, sorry to nag but can anyone help me with a newbie dialplan question, here is my five line dialplan: http://channels.debian.net/paste/322   how can i make the phone start to ring at the very begining?
10:38.57pratikriksta: ya i reloaded it several times
10:39.45pratikif you want i can paste the sip.conf and the extensions.conf in the pastebin
10:40.01*** join/#asterisk TheJudge (JTR@209-203-52-3.network.ods.co.za)
10:40.24TheJudgeanyone here from South Africa ?
10:40.35rikstayeah, im a bit busy, but paste it there, and maybe someone else can have a look at it before i do
10:40.57pratikok i will
10:43.06footnoteTheJudge: No but I just sent some stuff there :)
10:43.31footnotefor MTN
10:44.26pratikya i have pasted it in pastebin.ca/6949
10:44.38TheJudge:0
10:44.46TheJudgethey trying to get voip running ?
10:44.59TheJudgeI hear the are batteling with there edge device etc ?
10:45.03footnoteit wasn't for voip directly but yes
10:45.19pratikif any one can have a look plz let me know what is the error
10:45.31footnoteTheJudge: I'm only involved with the message store :)
10:45.36TheJudgeok
10:45.43footnote"It's not my fault sir I swear!"
10:45.45footnotehehe
10:45.46pratikeven my fwd calls are not going
10:45.53TheJudgewhat do ppl think of broadvoice ?
10:46.03TheJudgeanyone else know of good voip terminatiors ?
10:46.18pratikwhere as when i do "sip show peers" it shows that it is registered
10:47.48footnotevoip is proof customers will tolerate "one nines" :)
10:48.36TheJudgehuh ?
10:48.49footnote9x.xxx% uptime :)
10:49.11TheJudgelol
10:49.12TheJudge:)
10:49.25TheJudgemore like 1000000
10:49.34TheJudgevoip bring new meaning to uptime ?
10:49.38TheJudgein South Africa
10:49.43footnotehehe
10:49.49rikstafootnote: i don't get it
10:50.02Delvarits ileagel isnt it?
10:50.21TheJudgewhat voip in South Africa ? it was legalised 2 feb 2005
10:50.28pratikriksta:did u go through it
10:50.38DelvarTheJudge: ah didnt know that
10:50.46footnoteriksta: well, five nines is what they pound old school telecom developers with
10:50.58footnoteso testing is a BIG deal
10:51.01rikstafootnote: oh, is that an american thing?
10:51.06footnoteyeah
10:51.10rikstaahh, im UK
10:51.10footnoteit is pure suck
10:51.17footnoteit's mostly marketing
10:51.27footnotewith a lot of red tape for developers
10:51.37footnoteit's impossible to get updates out, etc
10:51.57pratiki have pastd it in www.pastebin.ca/6949
10:52.46TheJudgetrying to setup voip in sa is such a mess
10:53.02TheJudgethere is not enough bandwidth ! !
10:53.22Zeeekpratik I can't find your question: what is the problem?
10:53.22footnoteTheJudge: you surf?
10:53.25TheJudgeyea
10:53.31footnotesharks
10:53.36TheJudgenot a problem
10:53.43TheJudgealso dive instructure
10:53.50TheJudgelove diving with sharks
10:53.52footnoteWhere are you in SA?
10:53.59TheJudgelive in JHB
10:54.06pratiki am continuosly getting an error"Mar  7 16:27:04 WARNING[4164]: chan_sip.c:728 retrans_pkt: Maximum retries exceeded on call 376a6eb97378753c1f38252867c4273f@127.0.0.1 for seqno 198 (Critical Request)
10:54.07pratik"
10:54.09TheJudgebut got house in Capetwon and Durban
10:54.19footnoteso summer is winding down there
10:54.30Zeeekthat means that a sip peer is unreachable
10:54.37Zeeekusually
10:54.56TheJudgeslowly yea, but in winter it is still warm in durban temp is around 23 c and water about 19 c
10:55.01Zeeekone way to check is to remove all the register = statements and put them back one by one
10:55.02TheJudgeso its all good !
10:55.12footnoteTheJudge: I used to live in Huntington Beach, CA USA, we had a lot of folks from SA there
10:55.16TheJudgewhere you from
10:55.26pratikand the when i did "sip show peers" so to that particular server it said request sent
10:55.30footnoteI'm in Dawsonville Georgia now.
10:55.39footnote(home of Bill Elliot!)
10:55.41footnotehehe
10:56.03footnoteTheJudge: Bill Elliot == redneck race car driver
10:56.18Zeeekpratik nothing to with anything, but decide whetheer you want nat=yes and leave one line commented or not
10:56.28pratikso froom the sip.conf and extensions.conf i removed the particular data regarding
10:56.39footnoteok, time for bed before the sun comes up
10:57.01Zeeekpratik can you call fwd ?
10:57.11pratikyes i need nat=yes because our network is behind NAT
10:57.22pratikno but it shows that it is registered
10:57.29Zeeekyes but you have it two times, once commented out - get rid of one
10:57.43pratikok i'll do it
10:57.50pratikjust a minute
10:57.55Zeeekcomment out the fwd register and see if the sip message stops
10:58.20Zeeekcomment this line out of sip.conf : register => 607191:mypassword@fwd.pulver.com/607191
10:59.50Zeeekpratik also you do NOT want dtmfmode=inband
11:00.45*** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com)
11:01.06Zeeekpratik this is all I have in sip.conf: http://pastebin.ca/6950
11:01.27mesipratik/zeeek: My FWD authentication fails all the time :-(
11:01.45ZeeekSIP or IAX?
11:02.06mesiSIP
11:02.22*** join/#asterisk meppl (~mephisto@pD95421C1.dip.t-dialin.net)
11:02.22pratikok zeeek , thanks for that i'll change it but why are the fwd calls not going
11:02.22mesiI just changed my pass and now authentication fails all the time. :-(
11:02.35mepplguten morgen
11:02.42Zeeekmesi: change it back!
11:02.52mesimeppl: Morning. I think it is english here, not german ;-)
11:03.04mesizeeek: I can't! Somebody stole it!
11:03.11mepplgood morning mesi
11:03.12Zeeekmesi get a new account
11:03.21mesiZeeek: But I love my no, 434240!
11:03.30mepplmesi, i did "/amsg guten morgen" ;)
11:03.31mesiOk, I'll try to change it back.
11:03.34ZeeekI can get you a 2XXXXX number
11:03.36mepplmesi, thats the problem ;)
11:04.19mesimeppl: ah, ok. So you are from germany?
11:05.08mepplyes
11:05.12TheJudgedoes any one know of another service provide another the broad voice who support diffrent codecs ? such as 729a or 711
11:05.13TheJudge?
11:05.20mesiThis really sucks! I have got several password problems. With FWD, ENUM and the like :-( Seems to be a bad time!
11:05.30mesimeppl: Me too.
11:06.07*** join/#asterisk ptblank (~MURDER1@68-169-176-29.lmdaca.adelphia.net)
11:06.18tuxinator_linuxYou ever get a mood where you just don't wanna code?
11:06.22tuxinator_linuxI'm in that mod
11:06.24tuxinator_linuxmood
11:06.41mesituxinator: I know that.
11:06.54tuxinator_linuxAnd I love coding
11:07.18tuxinator_linuxbut I need to get it done.  I have added about 10 lines in the last 5 hours
11:07.43mesiTuxinator: That sounds bad.
11:08.06mepplmesi, german speaking people are in germany, austria, switzerland, luxemburg and some parts of africa
11:08.07tuxinator_linuxstarting to pick up, but I will need to sleep soon
11:08.13mepplmesi, oh, okay
11:10.01geeksteranyone using asterisk to peer with sipphone.com ?
11:10.06mesiOh dear! this is really bad! I can only use the stolen FWD password with asterisk, not a newly set one ;-(
11:10.29mesigeekster: I would like to.
11:10.43pratikzeeek: i have checked that, can you paste your extensions.conf
11:10.57geekstermesi: Its not working for me to well
11:11.06mesimeppl: nevertheless, you are from germany :-)
11:11.07tuxinator_linuxmesi: sad part is that it only took 5 minutes to do what I wanted to do
11:11.17shido6pastebin.ca
11:11.24Zeeekpratik I did that above
11:11.58mesituxinator: And why would you have to get this done? Is it for your Job?
11:12.35pratikZeek: ya but that was only the sip.conf , i want to see your extensions.conf
11:12.46soulz-hello all
11:12.49pratikZeeek: if you dont mind
11:13.05soulz-Mar  7 19:06:38 NOTICE[868]: app_dial.c:936 dial_exec_full: Unable to create channel of type 'SIP' (cause 3)
11:13.05geeksterhey there soulz
11:13.09soulz-i am getting this error message
11:13.10geeksterhows it goin ?
11:13.13soulz-can anyone help?
11:13.16soulz-hey geekster
11:14.01geekstersoulz-: I've never seen this error, anyone else want to take a stab ?
11:14.20tuxinator_linuxmesi: I own a couple companies and I am working a few large projects for one of them, including, writing the whole patient records and managed system and *
11:14.23soulz-weird ehh? i never seen as well
11:15.01geekstersoulz-: what actions are you doing that it causes tihs error ?
11:15.02tuxinator_linuxmesi: I stay busy
11:15.17pratikzeeek: i want to know where am i going wrong
11:15.25soulz-geekster: i am using bv, and they just implemented this new thing, called invite auth
11:15.48soulz-and i patched it to the current cvs, and this error comes up
11:15.56Zeeekpratik - I posted my config once and I'm not on pastebin anymore - look above and find it
11:16.17tih'cause 3' is "no route to destination".
11:16.45mesituxinator:  so, than it is really really bad!
11:16.46soulz-tih: exten => _2.,1,Dial(SIP/broadvoice/${EXTEN:1})
11:16.55Zeeekhello tih
11:16.59soulz-tih: this is what causes the cause 3 error
11:17.00tihsoulz-: it doesn't know how to reach broadvoice.
11:17.18pratikzeeek but on 6950 u have only pasted your sip.conf
11:17.19soulz-tih: emmm, okay
11:17.33mesiHas anybody ever changed the fwd password and still registered with asterisk by changing sip.conf? It doesn't work for me here.
11:17.35tihCheck your definition for broadvoice -- it's probably a 'peer' entry in your sip.conf, right? What host are you telling it it's on?
11:17.57tuxinator_linuxmesi: I never did like working for people.  I have fun, and isn't that what your suppose to do in life?
11:18.27Zeeekyou like working in general?
11:18.37tuxinator_linuxyes
11:18.44tuxinator_linuxI enjoy creating
11:18.45Zeeekthat's a step, then :)
11:18.49mesituxinator: Yes, that's what life is all about!!
11:18.54Zeeekah, reating. An ahhtist
11:18.59tuxinator_linuxwriting programs is a power trip
11:19.01soulz-tih: i just use type = friend
11:19.22tihsoulz-: ok, and host = ?
11:19.33mesiDoes Asterisk store fwd passwords somewhere else than in sip.conf where I configured it?
11:19.33soulz-i changed to peer also same thing
11:19.34Zeeekfor BV issues, check the mailing list - there have been problems lately specific to asterisk
11:19.49Zeeekseveral people were discussing this yesterday
11:19.50soulz-host = sip.broadvoice.com
11:19.59soulz-zeek: i did try it
11:20.01Zeeekeveryone is having problems
11:20.06soulz-but my error is not the same
11:20.19Zeeekdod you see the wiki?
11:20.20tihsoulz-: and can you ping sip.broadvoice.com from your machine?
11:20.39soulz-yup
11:20.41soulz-can do so
11:20.53Zeeek[there's been so much chatter about Broadvoice in about the last week]
11:21.16mesiZeek: so broadvoice is not working so well?
11:21.52soulz-zeek: tell me about it, they just implemented something
11:21.58tihHm. Then probably what Zeeek says - or you might want to debug a bit, as in: study the actual SIP packets being sent and received.
11:21.58soulz-without letting the users know
11:22.43soulz-Mar  7 19:22:22 NOTICE[969]: app_dial.c:936 dial_exec_full: Unable to create channel of type 'SIP' (cause 3)
11:22.48soulz-this is bugging me
11:23.06*** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au)
11:23.24Zeeekwe use a customer's weather service which had a procedure for ftp stats. They just changed it with no notice and our cron job went on  blithely downloading the same file every day until someone noticed the temperature never changed :)
11:23.38*** join/#asterisk PCadach (~paul@www.east.telecom.kz)
11:24.35pratik<PROTECTED>
11:25.05pratiklinksys(PAP2) is same as sipura
11:25.38mesiI get it now! FWD password for web interface and sip aren't the same :-(
11:26.03Mavviewhy is there a SIPCALLID, a SIPUSERAGENT and a SIPDOMAIN, but no SIPUSERNAME?
11:26.27Zeeekmesi, uh ya!
11:26.29*** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com)
11:27.04mesiZeek: So how am I expected to change this stupid SIP pass?! ;-)
11:28.47Zeeekthere is a page somewhere
11:28.55Zeeekbut why not use IAX while you're at it?
11:29.08Zeeekooops gotta go
11:29.11Zeeekbye all
11:29.20*** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc)
11:29.33pratik<PROTECTED>
11:29.59mesiZeek: You know, I use IAX with iaxtel. I want to do SIP, too. :-)
11:30.08mesiOh, gone.
11:30.22soulz-pratik: is voip legal in india?
11:30.56*** join/#asterisk pranav (pranav@202-149-48-214.broadband.isp.exatt.net)
11:31.16pratiksoulz:yes it is valid but only for outgoing calls
11:31.25pranavhi
11:32.03mesiHi pranav.
11:32.16pranavhi
11:32.26soulz-pratik: ok
11:32.34*** join/#asterisk asteriskforuk (~vircuser@i-195-137-59-254.freedom2surf.net)
11:32.48asteriskforukhello
11:32.54asteriskforukne1 awake?
11:32.58benno2could it be that with some 1.4.x budgetone firmware versions the call often gets hung up after 40-60secs ?
11:33.26pratiksoulz: can yv tell me i was using sipura device with my asterisk , can i replace it by sipura router
11:33.30asteriskforukne1 help with setting up agents?
11:33.42asteriskforuki've got a call queue and it works ok.. now want to try agent logins etc
11:33.54*** join/#asterisk MuppetMaster (~muppetmas@a82-92-73-185.adsl.xs4all.nl)
11:35.40soulz-pratik: i have never used sipura sorry dude
11:37.02MuppetMasterHellol
11:37.05MuppetMasterWhat is the question on Sipura?
11:37.13pratikok fine, see i have paswted my sip.conf and extensions.conf in the pastebin.ca/6949
11:37.38pratikif you have time just go through it, my fwd calls are not going
11:38.09*** join/#asterisk TheEmperor (TheEmperor@218.111.51.46)
11:38.55shaZwazanyone know where I can get a G723 license ?
11:39.47tuxinator_linuxwhat about G729?
11:40.03soulz-shazwar: u don't need a licence i think
11:40.11*** join/#asterisk Ubuz (~momo@CBL217-132-84-235.bb.netvision.net.il)
11:40.17shaZwazI wnat more compression
11:40.21soulz-just use allow=g723
11:40.37shaZwazto save my bandwidth
11:40.47shaZwazanyone tried speex ?
11:40.52soulz-use gsm or ilbc
11:40.59Ubuzhelp needed - I installed a stable version of asterisk over an existing vesion, and now asterisk won't load any modules. any idea why?
11:41.05shaZwazI am already using 729
11:41.13soulz-works great on a dialup:)
11:41.47shaZwazUbuz, what does the CLI read ?
11:41.49*** join/#asterisk Lethargicclown (~chatzilla@ool-18bee80e.dyn.optonline.net)
11:41.55pratikcan anyone tell me why my fwd calls are not going
11:42.05soulz-iax2 debug
11:42.12RoyKarg
11:42.13soulz-adios
11:42.14RoyKMar  7 13:09:14 WARNING[3243]: chan_zap.c:9615 setup_zap: Ignoring switchtype
11:42.14RoyKMar  7 13:09:14 ERROR[3243]: chan_zap.c:9436 setup_zap: Unknown signalling method 'pri_cpe'
11:42.15RoyKwtf?
11:42.22UbuzshaZwaz: I cannot load the CLI, it stops with an error
11:42.35shaZwazwhat does that say ?
11:43.00shaZwazRoyK, is that 0.6 ?
11:43.18Ubuzloader.c:258 ast_load_resource: /usr/lib/asterisk/modules/pbx_dundi.so: undefined symbol: ast_config_load
11:43.29RoyKshaZwaz: huh? 1.0.6
11:43.30Ubuzloader.c:440 load_modules: Loading module pbx_dundi.so failed!
11:43.53RoyKer
11:43.54RoyKfuck
11:43.56RoyK1.0.3
11:44.21shaZwazUbuz, go to modules.conf and put noload = pbx_dundi.so
11:44.47UbuzshzWaz: If I do that the error appears on the next module
11:44.49ta[i]ntedgod damn it
11:44.52ta[i]ntedi'm addicted to *
11:44.54pratikcan anyone tell me why my fwd calls are not going
11:44.57ta[i]nted:(
11:45.09shaZwazUbuz, may be u didn't compile it right
11:46.43*** join/#asterisk Inferna (~sasha@194.158.51.171)
11:46.47Infernahello
11:47.01shaZwazdo a make clean ; make install on all src directories
11:47.10RoyKhttp://pastebin.ca/6951
11:47.14Infernacan somebody help me with one strange thing?
11:47.14RoyKcan someone help me here?
11:47.21shaZwazand see if there are any compilation errors
11:47.29RoyKshaZwaz: that's 1.0.6, and it just won't load
11:47.35RoyKasterisk 1.0.6 as well
11:47.36UbuzshaZwaz: did it several times, no errors
11:47.43shaZwazRoyK, do a rmmod on the cards
11:47.45Inferna<PROTECTED>
11:47.46RoyKthe config files are identical to those of another server
11:47.52RoyKshaZwaz: tried that as well
11:48.03Infernaand then i have 2 dublicate calls from h323
11:48.08RoyKztcfg -vvvvvv show everything's fine
11:48.24asteriskforukIAX ... ne1 know what ports to open on a firewall? and which iax phone works well?
11:48.26RoyKtrying a rebooot...
11:48.29shaZwazcheck u zapata.conf then
11:48.33shaZwazzaptel.conf
11:48.52RoyKshaZwaz: both of them have been copied from a production server with the same te410p card installed, same hardware
11:48.58RoyKsame suse 9.1 distro
11:49.07RoyKa little older kernel
11:49.30shaZwaz2.4 !
11:49.42*** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode)
11:49.45RoyKproduction server runs 2.6.8.1. test server 2.6.11
11:50.07shaZwazdid u compile with make2.6
11:50.26shaZwazzaptel that is
11:50.28RoyKmake linux26
11:50.33shaZwazyup
11:50.44RoyKshaZwaz: otherwise the modules wouldn't load, obviously......
11:50.49Infernais any guru here?
11:50.53RoyK~lart shaZwaz
11:50.56Infernai have the problem with queue
11:50.59Infernaplz help me
11:51.01ta[i]ntedBV sucks balls
11:51.18RoyKInferna: noone can help you unless you say WHAT is wrong
11:51.19LethargicclownShould I coninue looking to see if asterisk works on cygwin, Or should I give up?
11:51.29pratikok fine, see i have paswted my sip.conf and extensions.conf in the pastebin.ca/6949
11:51.31shaZwazmay be the configs are bad
11:51.37RoyKLethargicclown: it might work, but it'll suck
11:51.44gambolputtyUbuz:  Did you delete the modules of the prior * version?
11:51.49pratikcan anyone check why my fwd calls are not going
11:51.50RoyKshaZwaz: they work on another server
11:51.52shaZwazu must have missed something
11:51.53RoyKsame fscking files
11:52.37shaZwazotherwise use the same source too
11:53.04Inferna<PROTECTED>
11:53.16pratikgambolputty:can you help me?
11:53.32gambolputtywith?
11:53.42pratikmy fwd calls are not going
11:53.56gambolputtyyou use iax or sip to connect to fwd?
11:54.01pratiksip
11:54.04Ubuzgambolputty: How can I delete the modules of the older version?
11:54.46gambolputtyUbuz:  The first thing I do is stop the current instance of *.
11:54.55gambolputtyThen unload zaptel and ztdummy
11:55.08gambolputtyDelete modules in /usr/lib/asterisk
11:55.23gambolputtythen recompile another * version as needed
11:55.36gambolputtymodprobe ztdummy
11:55.44gambolputtyand make sure * works with asterisk -vvvvgc
11:55.45Ubuzgambolputty: I deleted the modules, removed everything under /var/lib/asterisk
11:55.54pratikgambolputty: ihave pasted my sip.conf and extensions.conf in thge pastebin.ca/6949
11:55.55Ubuzthen I recompiled and reinstalled the modules
11:56.03Ubuzthere's not problem with zaptel, it works fine
11:56.12gambolputtyok
11:56.19RoyKfuck
11:56.27gambolputtypratik:  Is your * box behind a firewall?
11:57.46gambolputtyUbuz:  What distro of Linux are you using?
11:58.11RoyKshaZwaz: anyway, asterisk should not fscking tell me pri_cpe is unknown signalling meth
11:58.26Ubuzgambolputty: SuSE 9.1
11:58.46pratikgambolputty:yes
11:59.07pratikgambolputty: does that affect
11:59.08gambolputtypratik:  Use IAX instead to connect to fwd.
11:59.55pratikok IAX i am not very much familiar with, i'll have to read the documents
11:59.58mesigambolputty: I use iax now. But anyway I'd like to change my stolen SIP password!
12:00.12mesipratik: http://www.freeworlddialup.com/content/view/full/1501
12:00.35gambolputtyUbuz:  I use Fedora Core 2
12:00.47gambolputtyI have to be careful when kernel versions change
12:00.49mesiQuite easy. And it uses the password you use for the FWD webpage. Anyway, it seems you cannot change the SIP password :-(
12:00.49pratikgambolputty: but see tell me to make a call i am using a sipura phone, will that worki with the IAX
12:01.07gambolputtyA Sipura device does with with *
12:01.16gambolputtyand then * connects to fwd using IAX
12:01.25pratikgambolputty:and sipura phone is a sip phone
12:01.28mesipratik: is your phone connected to asterisk? Then it will work. Asterisk can do sip to the phone, but be connected via IAX to FWD.
12:01.28gambolputtyAll Sipura devices speak Sip
12:01.38shaZwazRoyK,  pridialplan=unknown
12:01.38shaZwaz<PROTECTED>
12:01.38shaZwaz<PROTECTED>
12:01.45Ubuzgambolputty: Everything worked, but I saw some bugs in asterisk, so I decided to install the latest stable version, and since then it won't work
12:02.11gambolputtyMaybe try cvs head version?
12:02.32pratikmesi:yes my sipura phone and the asterisk are connectedd in the same network
12:02.45shaZwazRoyK, u sure its pri_cpe .. I mean spelled it correct ?
12:03.07*** join/#asterisk jks (~jks@0x503e4c12.arcnxx4.adsl-dhcp.tele.dk)
12:03.09mesipratik: So look at the webpage I pasted. Your sippura will work with asterisk being connected to FWD via IAX.
12:03.24mithroanyone know if the X100P cards work with caller ID in australia?
12:03.26pratikok ya i'll go through it
12:04.15shaZwazWhy is it ignoring the switchtype ...something wrong there may be try switchtype=euroisdn
12:04.55jkshow do I set the mapped address on a SNOM IP-Phone? (I've got 1:1 NAT)
12:05.59Ubuzgambolputty: I got the latest CVS version and it doesn't work too. You think the problem is with the kernel?
12:06.28shaZwazUbuz, revert to older to verify
12:07.13UbuzshaZwaz: Revert to older what and how do I get it?
12:07.54shaZwazu overwrote it didn't u !
12:08.21shaZwazcheck cvs.digium.com
12:08.44The_Ballhow can i make my asterisk server accept "calls" from any anonymous users on any ip?
12:08.51UbuzI am copying a working version from another computer
12:09.08mesiAnybody for a call? My FWD is 434240 :-)
12:10.26mesiWho called? :-)
12:10.31mesiThat was only one ring. :-\
12:10.42shaZwazme
12:10.45shaZwaz:)
12:10.48mesiShaZwaz ;-)
12:11.11mesiEven my hardphones ring, though they aren't internetphones. I connected them to asterisk ;-)
12:11.23mesiThat's funny.
12:11.52shaZwazu must be ringing them in the dialplan
12:12.01shaZwazwhere the call is landing
12:14.07mesiI do. The phone rang, but you obviously hung up immediatly :-\
12:14.29*** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au)
12:15.06FaithXHey guys, where can I get some usefull info on bristuff?
12:15.37RoyK~lart shaZwaz for not listening
12:17.45*** join/#asterisk Jer1326 (~Jer@rdu57-251-152.nc.rr.com)
12:18.23jksanyone got a snom IP phone?
12:19.33Jer1326can someone offer me a sugestion as to why atxfer doesnt work for me?
12:22.52mesiAre all theses sip-providers interconnected? Like sipphone and FWD?
12:23.16Jer1326some are some arent
12:24.04sambalJer1326: which phone?
12:24.15mesiJer: So I have to register my asterisk to several of them to be able to call every member, right?
12:25.21Jer1326yep
12:26.00gambolputtyjks:  I have a snom 190 ip phone
12:26.06Ubuzok, the latest version in the CVS doesn't work well for me. I installed CVS-HEAD-03/07/05 and it works
12:26.16jksgambolputty: have you tried using it with 1:1 NAT?
12:26.17Jer1326sambal: my cell.... calling in  using IAX via teliax
12:26.28jksgambolputty: it keeps using its internal IP in the SIP messages, no matter what I configure :-/
12:26.41gambolputtyWhat do you mean by 1:1 nat?
12:26.57Jer1326even if i set atxfer => *4 or whatever it still says transfer when i hit # :/
12:27.07jksgambolputty: The internal IP corresponds directly to one external IP
12:27.10MakenshiMorning gambolputty :> how are you these days?
12:27.17gambolputtyfine
12:28.25mesiIs it advisable to register with sipphone.com?
12:28.25gambolputtymy snom phone connects with *
12:28.25gambolputtyare you trying to make a sip call to somewhere?
12:28.29jksgambolputty: I'm just trying to get it to connect to my asterisk server
12:28.39mesiMe? Well, I just want to be able to call many people and they should be able to call me :-)
12:28.55jksgambolputty: from the SIP trace I can see that it puts its 10.x.x.x address in the SIP messages, instead of its external IP... that's what I want to change
12:29.06gambolputtyjks:  post your sip.conf on pastebin.com
12:29.33jksgambolputty: it's the phone's settings I want to change?
12:29.40gambolputtynot sure
12:29.47jksit is :-)
12:29.47gambolputtylets make the phone register with * first
12:30.02jksgambolputty: well, the reason it doesn't register is because the phone sends off it's internal IP
12:30.19jksgambolputty: my sip.conf is pretty standard... I have host=dynamic and nat=yes
12:30.26RoyK~seen zoa
12:31.07jbotzoa <zoa@142.131.189.23> was last seen on IRC in channel #asterisk, 5h 58m 46s ago, saying: 'or at least live from san jose'.
12:31.07gambolputtythat's probably what it should do when it registers with *
12:31.07jksgambolputty: (and it works perfectly if I put the phone on the external IP directly)
12:31.07gambolputtyis * behind nat?
12:31.08jksgambolputty: I want the phone to send it's external IP now that it got one
12:32.42gambolputtywhy you want to have the snom phone send and external ip?
12:32.50shaZwazRoyK, what happend ?
12:33.00jksgambolputty: my asterisk server is outside my nat
12:33.05gambolputtyok
12:33.07jksgambolputty: it can contact the phone only on the external ip
12:33.11jksgambolputty: so that's why
12:33.23jksgambolputty: I'm using 1:1 nat, so the phone is the only thing on that external ip
12:33.24pimpwellanyone have a manager intterface I can check out
12:33.25shaZwaz~wakeup RoyK
12:34.28jbotACTION silently aproaches RoyK, who's sleeping (zZzZZZzzZZ, Ronc !!! ronc!), gets off his pants and shoots a noisy fart ... PUBFBFBFBBBFFF!!!
12:34.28pimpwellso I can just get a look at what it's like
12:34.28gambolputtynot sure, both my * box and snom are behind nat on the same subnet
12:34.28gambolputtysip seems to work best if two devices are on the same subnet
12:34.28RoyKshaZwaz: I had forgot to compile libpri before compiling asterisk......
12:34.29RoyKrecompiling asterisk worked
12:34.29shaZwaz:)
12:34.33shaZwazI told u must be missing somthing
12:35.00RoyK~lart shaZwaz
12:35.10shaZwaz~kill RoyK
12:36.09jbotACTION shoots a hyper-charged  neutrino gun at RoyK
12:36.09RoyKshaZwaz: you were just babbling general idiocity
12:36.09jksgambolputty: yes, but I haven't got them on the same subnet... it's not possible for me.
12:36.09shaZwazI told u  u were missing something
12:36.09gambolputtysip and nat have problems getting along
12:36.09shaZwazpersonally I never used te400p cards
12:36.16jksgambolputty: well, I'm not using "normal" nat
12:36.28gambolputtyok
12:36.32jksgambolputty: this should be very easy to get working, if I just could get the phone to send out the right IP
12:36.34shaZwazsince everything looked fine with the config ....
12:36.46jksgambolputty: you don't know how to tell the phone which IP it should think it has?
12:37.00gambolputtynot in that manner
12:37.04gambolputtytry different settings
12:37.04jksgambolputty: okay
12:37.06shaZwazI was sure u missed some step and there u were ...missed compiling libpri
12:37.08jksgambolputty: I have.
12:37.09gambolputtyuse the internal sip trace function
12:37.23jksgambolputty: it's a normal setting on every phone I've come across... I just can't find it on this phone.
12:37.28jksgambolputty: I am using the internal sip trace function....
12:37.32gambolputtyok
12:37.49shaZwazwell anyway I accept that I am not a * guru
12:39.03Infernahow to make ring call progress tone insteed of default music in queue?
12:40.23Delvaranyone know of a free outbound proxy server software?
12:41.05shaZwazsoftware ?
12:41.13moonwicksquid?
12:41.45shaZwazwhat u mean by outbound proxy server software ?
12:44.30shaZwaz~ping
12:45.16jbotpong
12:45.16Delvarsorry
12:45.16Delvarfor SIP/RTP
12:45.16RoyKDelvar: SER?
12:45.16DelvarSer does rtp?
12:45.16RoyKno
12:45.16Delvari thought it was SIP only
12:45.16RoyKa proxy doesn't do RTP
12:45.18RoyKproxy does SIP and not RTP
12:45.23shaZwazwell SIP can't so anything without it ?
12:45.30RoyKDelvar: a gateway does RTP
12:46.01DelvarRoyK: sorry my bad terminolayg, i want a free gateway then :) ... not asterisk
12:46.17shaZwazto use for ur own ?
12:46.22Delvaryes
12:46.39shaZwazover to u RoyK
12:47.24RoyKDelvar: aefirion, sipx, yate
12:47.25gambolputtyDelvar:  What are you trying to do with a SIP proxy?
12:47.32RoyKDelvar: or, of course, asterisk
12:47.52shaZwazRoyK, u ever tried Aferion ?
12:48.07RoyKit's basically the same as asterisk atm
12:48.09RoyKit's a fork
12:48.21RoyKbut it'll change quite dramatically over the next two months or so
12:48.30shaZwazwhat about its SIP capabilities ?
12:48.32DelvarRoyK: i need an outbound proxy to proxy RTP trafic to get arround nat issues without STUN/port forwarding...
12:48.57RoyKshaZwaz: right now, same as asterisk, but it'll use OPAL for sip/h323 instead of the crap in *
12:49.07DelvarRoyK: preferably used alongside SER
12:49.36shaZwazguess have to wait then
12:49.39RoyKDelvar: we use plain asterisk with SIP with the server on an official IP, and it works for thousands of SIP clients
12:50.29shaZwazRoyK, anyother choice for saving bandwidth other than using 729 ?
12:50.34*** join/#asterisk wizard2000 (~dang@fubar.arcbox.com)
12:50.42RoyKshaZwaz: speex? ilbc?
12:50.51wizard2000hi guys, having a few problems with zaptel and FC3, anyone got a sec to help?
12:50.54RoyKg.723.1....
12:50.58shaZwazspeex is proprietry ?
12:51.00DelvarRoyK: so thats a no then?..
12:51.06RoyKshaZwaz: speex is very, very open
12:51.20shaZwazgime the link will u
12:51.25RoyKDelvar: I only say it works for us with asterisk
12:51.33RoyK~google speex
12:51.35RoyK~speex
12:51.47RoyK~lart shaZwaz
12:52.57shaZwazRoyK, have u tested it on a productions server ?
12:53.03DelvarRoyK: i know but asterisk doesnt scale too well, especialy when its getting a lot of registration requests a the same time. and to be honest id refer to use something more dedicated like SER to proxy the rtp.
12:54.18*** join/#asterisk pilif (~pilif@mail.sensational.ch)
12:54.19RoyKshaZwaz: no, but a lot of people do
12:54.24shaZwazI mean any issue with that?
12:54.51RoyKDelvar: then just use SER.....
12:54.56pilifI'm looking for a good, understandable tutorial on how to configure asterisk
12:55.03RoyKSER gets an INVITE, and tells client 'you send that RTP there'
12:55.07pilifare there any current web-resources?
12:55.14RoyK~docs
12:55.24jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
12:55.57Jer1326ugh i just broke my *...recompiled to lastest CVS and not i get a bus error ...arrgh
12:56.09Jer1326it wont even start :(
12:56.12DelvarRoyK: SER does NOT do RTP :)... hence why i was asking
12:56.28ta[i]ntedi see a bad moon risin~
12:57.25Jer1326can someone help with a simple compile question...?
12:57.27RoyKDelvar: SER proxies RTP, but never gateways it
12:57.39RoyKDelvar: read 'asterisk at large' on the wiki
12:57.49RoyKhttp://www.voip-info.org/wiki-Asterisk+at+large
12:58.13DelvarRoyK: ok thanks, ill read about it, i was jsut told SER wouldnt do it, maybe it will.
12:58.16*** join/#asterisk markak (~markak@ndn-165-143-245.telkomadsl.co.za)
12:58.47wizard2000Mar  7 12:58:37 sip wait_for_sysfs[2908]: either wait_for_sysfs (udev 039) needs an update to handle the device '/class/zaptel/zap1' properly (no device symlink) or the sysfs-support of your device's driver needs to be fixed, please report to <linux-hotplug-devel@lists.sourceforge.net>
12:58.51wizard2000any ideas people?
13:02.06markakhi all quick question. if one wanted to for example allow extensions .conf to pull phone number from a mysql database and use it for the dial line is it easy to do. i.e we deal with 1200 + branches accross the country. each branch has a unique branch number. could i place a rule that when our operators dial eg 8 followed by the branch number let say 832 asterisk could query the mysqldb for the phone number and create the dialplan exten =>
13:05.12markakanyone have an idea ?
13:05.39MuppetMastermarkak:  Yes, this is fairly straightforward.
13:06.03MuppetMastermarkak: You could either use an AGI to launch an external app to do the lookup, or you could use the built in MySQL lookup function.
13:06.42MuppetMastermarkak:  http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MYSQL#comments
13:06.43Darwin35anyone know if kram made it to ca
13:07.06markakthanks guys
13:07.09*** part/#asterisk markak (~markak@ndn-165-143-245.telkomadsl.co.za)
13:07.33Darwin35ok now the fun of festival
13:07.38Darwin35and sphinx
13:08.05Jer1326does sphinx realllly work?
13:08.10pilifcan I use the Fritz! USB-Adaptor for my first experiments? I've such a device lying around here and don't want to buy anything just to learn if it'd work
13:08.58Darwin35dont know
13:09.10Darwin35but going to test and see
13:09.52LethargicclownAny special instructions for compiling this in cygwin?
13:09.59Darwin35?
13:10.53LethargicclownJust hoping i can test this in cygwin before I dedicate a machine to it
13:11.03Darwin35ok I am using 300 megs of a 500 meg cf for fbsd and asterisk thats nt bad
13:11.30Darwin35nt/not
13:12.05Darwin35just have to get festival and a few agi apps
13:12.05Darwin35on it  now
13:12.17Jer1326i'm intrested to see how well it works
13:13.12Darwin35so far it works execpt for the g729 issue I ave to take up with kram
13:13.23Darwin35my tdm card works
13:13.31Darwin354 port fxs
13:13.36Jer1326ahh
13:13.46Jer1326can someone help me with a segfault?
13:14.25Darwin35cut  abd paste in pastebin and show it
13:14.39Darwin35see what can be done
13:14.55Jer1326you want the output of gdb?
13:15.31Darwin35yeah pu tit in there also
13:16.44Darwin35oops I said tit
13:16.46Darwin35hheehe
13:17.09Jer1326lol
13:17.50Darwin35man I wish my new credit card would get here
13:18.14Darwin35I need a new sip phone
13:19.41*** join/#asterisk zotz (~zotz@24.231.32.191)
13:20.00Jer1326http://pastebin.ca/6955
13:21.38Darwin35ok looking now
13:23.22Darwin35this linux or ffbsd
13:23.27Jer1326fbsd
13:23.36Darwin35I know that the mailing list had a workaround in it
13:23.41Darwin35hhold on
13:23.48Jer1326alright
13:23.55The_Ballis there any catchall extention? like [incoming] \ exten => *,1,Answer()   and so on?
13:24.08Jer1326isnt it s?
13:24.18The_Balloh, is it
13:24.26Inferna<PROTECTED>
13:24.56The_Balli got an error when a iax user tried to dial in to a non existant extention
13:25.00The_Ballin that context
13:25.12Darwin35jer check pvt window
13:25.15Infernathe_ball: check context
13:25.17The_BallMar  7 22:43:59 NOTICE[8677]: chan_iax2.c:5461 socket_read: Rejected connect attempt from 203.49.132.59, request '80001234@incoming' does not exist
13:26.34MuppetMasterThe_Ball:  Yes, it is the 's' extension exten => s,1,Dial....
13:26.56Darwin35it means yo have not setup the user inthe iax.conf and it cant point to a context it cant find
13:27.27The_Ballhmm, ok
13:27.54Infernaanybody had problem with freezed calls in the queue?
13:28.56*** join/#asterisk theogeor (theogeor@212.118.246.50)
13:29.09*** join/#asterisk ToyMan (~stuq@204-8-82-238.webjogger.net)
13:29.47theogeorHi everybody.... I am looking for help on configuring * and connect it with an MD110 and a TLU76/1 card
13:30.04theogeoranybody who would like to help :)
13:30.36Darwin35its a gothic scottish morning
13:31.16*** join/#asterisk seong (~seong@219.95.129.15)
13:31.24theogeorI am using an * box with TE410 card
13:31.28theogeorany ideas ?
13:31.37Infernasmall question, can X-server be the problem of bad music on hold playing?
13:33.19theogeorAnybody alive in this channel ?
13:33.23theogeor:(
13:35.30theogeorHello anybody here ?
13:37.27Grooby....zzzzZZZZ
13:41.02Darwin35no that mpg123
13:41.37*** join/#asterisk hellop (~LeeHarvey@cpe-70-93-41-67.hawaii.res.rr.com)
13:41.54hellophi
13:42.09hellopHey I just saw the recent slashdot post about Asterisk PBX.
13:42.20hellopgjgj
13:42.55hellopSo..  I have a question:  How much for a commercial, non-asterisk, pbx?  10-20, 50-100, 100-200 lines.
13:43.24hellopI'm doing an English paper on Asterisk...
13:43.26hellopfor scool
13:43.42hellopscewl
13:44.51theogeorHello anybody who has connected * on an MD110 PBX through TLU76/1 card ?
13:44.53hellopMan, I've done so many google searches trying to find some prices..  $2499 for some hardware.   What about $60000 for 100 line system?
13:49.14*** join/#asterisk boch (~as24@200.59.172.98)
13:49.21InfernaDarwin35: how can i fix this mopg123?
13:49.59Darwin35mpg123 is junk now days look at res_mp3 in the addons
13:50.44Darwin35or you can deinstall the ver of mpg123 you have and build the one that * supports
13:50.47InfernaDarwin35: look let me describe you one thing
13:50.56Darwin35in the asterisk src dir type make mpg123
13:51.01InfernaInferna: for SIP calls muscoc is ok
13:51.01hellopSo..  I have a question:  How much for a commercial, non-asterisk, pbx?  10-20, 50-100, 100-200 lines.
13:51.09hellopHow about just a guess?
13:51.10InfernaInferna: i get this problem only for incoming h.323 calls
13:51.30Darwin35I doont deal with h.323
13:51.44Inferna<Darwin35> it's not silence supresion problem
13:51.56InfernaDarwin: becausel voice promts are very good
13:52.05InfernaDarwin35: only music played by mpg123
13:52.09Infernais suxx
13:52.12Darwin35yes
13:52.24CleanerXcan someone do we a favour and do a ns lookup for me?
13:52.42Darwin35mpg123 -50r is a pain
13:52.51InfernaDarwin: i am using the latest one
13:53.01Darwin35you will have problems
13:53.26Darwin35thats why there is res_mp3 in the asterisk_addons
13:53.42InfernaDarwin: so i should try res_mp3?
13:54.25Darwin35that or uniinstall the mpg123 you have installed
13:54.40Darwin35and cd to your asterisk src dir and type make mpg123
13:54.57Darwin35it will pull the last working ver of mpg123 that worked with asterisk
13:55.03Darwin35and build and install it
13:55.16CleanerXcan someone do we a favour and do a ns lookup for me?
13:55.36hellopDoesn't anyone have any idea how much a traditional PBX costs?
13:55.38Darwin35I hope we either fix mpg123 soon or drop it
13:56.38InfernaDdarwin: i don't see res_mp3 in asterisk-addons
13:56.52*** join/#asterisk fac_ (faceoff@devel.acdbddh.eu.org)
13:56.55fac_hello
13:57.50Darwin35it might be res_mpg123
13:58.33Jer1326CleanerX what do you need looked up
13:59.30*** join/#asterisk santiago (~santiago@63.245.86.95)
13:59.39theogeorHello anybody who has connected * on an MD110 PBX through TLU76/1 card ?
14:02.24InfernaDarwin: format_mp3, res_perl and mysql
14:02.30Infernathat's what i can see there
14:02.34Infernamaybe format_mp3?
14:03.34Darwin35think thats it
14:04.03Darwin35but if you want to use mpg123  install the one asterisk uses
14:04.11Darwin35in your asterisk src dir
14:05.07*** part/#asterisk mogorman (~mogorman@146.229.176.173)
14:05.40*** join/#asterisk kant (~bernd@207.42.191.67)
14:09.55Darwin35ok 1.0.6 is about patchd for ports
14:10.07*** part/#asterisk pilif (~pilif@mail.sensational.ch)
14:10.12wizard2000when i pick up my handset (plugged in to a TDM) i do not get a dialtone but constant beeping... any ideas?
14:10.15*** join/#asterisk MikeJ[Jayden] (~ircatjerr@65.170.43.34)
14:10.21wizard2000i can dial that phone from a sip extenstion mind
14:10.31Darwin35you have vm
14:10.45Darwin35its called a studer tne
14:10.55hellopThis is a PBX channel.  Can anyone tell me how much 100 line PBX systems cost?
14:10.58Darwin35but if your not getting tone at all
14:11.01hellopDarwin, you know.  come on
14:11.26wizard2000Darwin35: was that directed at me?
14:11.44Darwin35hellop  depends on function and hardware
14:11.48Darwin35wizard yes
14:11.56Darwin35wizard pvt me
14:12.27hellopDarwin35  standard function, basic.
14:13.07*** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au)
14:13.33mikegrbhellop: ten million dollars
14:14.03hellopReally, I'm just looking for some examples of some traditional PBX systems...
14:14.18Jer1326hellop: why?
14:14.43hellopHundreds of business get started each month, they need PBX, what's the estimated cost?
14:14.57Jer1326<1000
14:15.06Darwin35well nec start at 10 grand and go up
14:15.10Darwin35thats without vm
14:15.12hellopJer1326, Doing a writing assignment for my English 209 Business writing class.
14:15.28Darwin35and thats for 50 extensions
14:15.32kanthellop: With *, you should need to worry about hardware.
14:16.18hellopkant, so, I'm trying to do a comparisan between Asterisk and non-asterisk.  So, I'm trying to find high-priced real world example.
14:17.09Darwin35<PROTECTED>
14:17.34hellopDarwin35, I do searches for stuff like "nex pbx price" or "100 line pbx price"  and I find only stuff like $795 two-line pbx.
14:17.53hellopHow about 100 lines pbx price?
14:18.12Darwin35I  have never priced a 100 unit system
14:18.28Darwin35let look in my pricing book for units I use to sell
14:18.31kantCall Nortel and ask them for a quote on their Norstar system.
14:18.48hellopDarwin35, You said $10,000 for 50 extensions.  That is just the unit, add 10,000 for 50 Cisco 7940's right?
14:18.49drspermIf you are in the US, check with intertel
14:18.53Darwin35Merlin 75 line with  out VM start at 9 grand
14:19.30Darwin35I have not delt with cisco phones yet
14:19.50drspermYou could always check out Cisco...that is what is in peoples face pretty heavy now.
14:20.04*** join/#asterisk inspired (mikael@omicron.wwis.net)
14:20.05kantThe phones will cost you more than the PBX in the end...
14:20.05drspermCisco = $$
14:20.06Darwin35merlin 75 line with vm starts at 15,000
14:20.20hellopok, Nortel, Intertel, Merlin.. but you guys are not talking total cost right?  Just the unit, then you have to pay consultation fee for setup, right?
14:20.38*** join/#asterisk hajekd (~hajekd@mail.idoox.com)
14:20.45kantAnd maintainance fees...
14:20.52dfunnellHi all, can anyone help with a dial-out problem I am having? Using CAPI * is dialling as soon as a pattern is matched, which makes it difficult with variable length numbers (such as mobile numbers, etc.)  Using exten => _100.,1,Dial,CAPI/470:${EXTEN:1} for example dials out on 00n (where n is the fourth digit dialled).  V desperate, will pay in beer.
14:21.04drspermI know that the hourly rate for Intertel is $225/hr...
14:21.08*** join/#asterisk hajekd (~hajekd@mail.idoox.com)
14:21.17drspermMost linux guys charge between $125 - $180/hr
14:21.18hellopI'm saying, you are building $50,000,000 manufacturing plant.  You need 100 phones.  Whats the total cost to install?
14:21.30Darwin35unlike asterisk most pbx systems dont come wiith vm built in
14:21.37Darwin35so its not a fair pricing
14:21.39hellopvm?
14:21.42hellopvoicemail
14:21.45Darwin35voicemail
14:21.47kantyes
14:21.59hellopWell, I mean vm included.
14:22.00Darwin35vm is a addon
14:22.04hellopTotal phone system.
14:22.46drspermWhenever you get numbers from people like us on line, remember to ask where we are located....
14:22.48hellopI worked for a company, they built the thing in a couple months... 100 line phone system installed turnkey.  I didn't even notice.  We never had to call anyone for help.
14:22.52drspermI am in SA, TX, USA
14:23.08hellopGave us a stack of books... how to add extensions, voicemail...  must have cost alot.
14:24.16hellop"The thing"  built the huge manufacturing plant in a few months..
14:24.25drspermdfunnell: sorry I didn't reply...I have no clue yet...
14:24.33drspermlinux = veteran
14:24.38drspermasterisk = newbie
14:24.56hellopchannels?
14:25.11hellophuh?
14:25.33FaithXhey the docs on bristuff are pretty sparse
14:25.36dfunnelldrsperm:  no worries, think it has everyone stumped - can't seem to find a resolution for this one anywhere.
14:26.07dfunnellShould have seen asterisk specialist who came out and had a look yesterday - went home a broken man ;-)
14:26.09hellopok, but thanks for the company names..  I guess I'll call them for a quote.
14:26.15drspermdfunnell: good luck.  catch me in a month, maybe I can help.
14:26.55FaithXdrsperm, do you run a bank or something?
14:27.08drspermWhy would you say that?
14:27.34dfunnellAny other takers?  Did I mention I will pay in beer?
14:27.46drspermoh...sorry I was slow to catch on...
14:27.53MikeJ[Jayden]what's the issue?
14:28.41hellopdfunnell, many it sounds like your problem is a simple regex problem, could be fixed in a couple keystrokes.
14:28.47dfunnellMikeJ[Jayden]:  Can't stop * using CAPI from dialling as soon as pattern is matched.
14:28.49tzangerdammit linux1394.org has been down for a while now
14:29.01dfunnellhellop:  Please go on!
14:29.03hellopmany=man  sorry
14:29.25MikeJ[Jayden]ummm... it's suposed to dial as soon as patern is matched.. although I have no idea on capi
14:29.31hellopdfunnell, oh..   so  Yeah, you need to find in the source code, where it is matching the thing, and change it.
14:30.03djindfunnell, is your example the actual line in extensions?
14:30.35*** join/#asterisk RoyK (~roy@80.239.107.80)
14:30.44hellopdfunnell, I have no idea what you are talking about, but if you tell me what you are trying to match, I could make you a regular expression to match it.
14:30.55dfunnellMikeJ[Jayden]:  Yeah, but using exten => _100.,1,Dial,CAPI/470:${EXTEN:1} it starts dialling on the fourth digit and fails miserably
14:31.30*** join/#asterisk eipi (~eipi@100-172-114-200.fibertel.com.ar)
14:31.53djindfunnel, why not use _100XXXXXXXXX.,1,Dial for example?
14:31.59RoyKhm
14:32.04dfunnellMeans that dialling a US number, for example 100195551234, is impossible, as * tries to dial 001 (leading 1 dropped).
14:32.12RoyKwhen do the digium guys wake up?
14:32.57*** join/#asterisk mesi (~player@dsl-082-083-143-045.arcor-ip.net)
14:33.15*** join/#asterisk oej (~oej@63.83.135.35)
14:33.20mesiHi oej
14:33.21MikeJ[Jayden]welll the leading 1 is dropped because you have :1 after ext.
14:33.23*** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc)
14:33.26RoyKhej....
14:33.34dfunnelldjin:  Problem is numbers are of variable length.  Example - mobile numbers in NZ can be 6 or 7 digits long.  Using _102XXXXXXX works fine for 6 digit numbers (021 and 6 digits, for ex.), but not with 7 digits.
14:33.35ZeeekHej RoyK
14:33.40djindfunnell, did you read my remark?
14:33.51djinHi Zeeek.
14:34.04RoyKhow can I debug a te410/pri?
14:34.07djinoops, dfunnel, sorry :(
14:34.14oejHej - hello from San Jose
14:34.26ZeeekI have a server in San José
14:34.34dfunnelldjin:  No worries, keep the ideas coming
14:34.36dfunnell!
14:34.37Zeeeksay hi
14:34.37MikeJ[Jayden]hey oej..
14:34.41mesiMy asterisk is registered with sipphone.com, but whenever I call it I just get an unavailable message. The message I speak then, is mailed to me. Why am I unavailable on sipphone.com?
14:34.53djindfunnel, add a _102XXXXXXXX (extra X) then?
14:34.53FaithXdfunnell, just chuck a few more rules in...
14:34.58Zeeekmesi did you get FWD working?
14:35.00MikeJ[Jayden]and is this only on capi calls, or is it no matter what you point it too?
14:35.30MikeJ[Jayden]if you make it NoOp(${EXTEN}) it does the same thing, right?
14:35.36dfunnellhellop:  deeply uneasy about messing with the source code.  Not sure I'd even know what I am looking for, let alone how to change it.
14:35.46drspermI have 2 questions for all:
14:35.46drsperm1.  What sip providers would you reccomend for use in Texas,USA
14:35.46drsperm2.  Is there any way to pull my vonage numbers into asterisk?  I read a posting that it was possible and found a config...I was wondering that everyone thought?
14:35.54mesiZeeek: Yes! Thanks. The point was, that it doesn't change this stupid Sip-password. IAX works fine and the password can even be changed.
14:35.54djinyou could create more lines matching variable lengths.
14:36.09dfunnellMikeJ[Jayden]:  Only a problem using CAPI, but this is all I am using for dialling out, so am not using pattern matching for anything else.
14:36.10*** join/#asterisk Darkar (~alex@m174.net81-66-29.noos.fr)
14:36.12Zeeekmesi aha. So IRC rules, yes?
14:36.13MikeJ[Jayden]vonage softphone works with asterisk, the ATA's no
14:36.19Darkar› System Statistics: OS: Windows XP 5.01.2600Service Pack 2 on AMD Athlon(tm) XP 2400+ (AuthenticAMD) Ram: 511MB total, 297MB in use (58%) GPU: 128MB WinFast A340 Uptime: 5hrs 38mins 13secs Hdd's (4): 35.3GB/335GB
14:36.31MikeJ[Jayden]dfullel see ^^ about noop
14:36.34mesiZeeek: IRC rules anywayz. Since about 5 years :-)
14:37.06drspermMikeJ[Jayden]: ok...softphone...ie:  eliminate my ata and go straight to the asterisk box?
14:37.08*** join/#asterisk faccione (~as@host54-54.pool212171.interbusiness.it)
14:37.09EssobiHeh.
14:37.17faccionehiya
14:37.23dfunnellMikeJ[Jayden]:  Not sure about NoOp(${EXTEN}).  Sorry for the stupid question, but what is ^^?
14:37.26EssobiI feel ashamed thinking about how many years I've been logged on to IRC.
14:37.46LethargicclownYou're saying if I have Vonage then I can only use soft phones?
14:37.53Essobidfunnell http://www.voip-info.org
14:38.15Jer1326Lethargicclown: * works with vonage if you have a softphone acct
14:38.20faccionedoes anybody speaks italian here ?
14:38.25Moc^^ mean look at the previous lines
14:38.46drspermok..so I take it there is a difference between a normal vonage account and a softphone account?
14:38.49dfunnelldjin:  Tried that (different lines with different lengths of 'XXX's', but * tries to dial as soon as it reaches the shortest match (so only works if number is as long as shortest match).
14:39.00Jer1326no softphone is an ADDON to a normal account
14:39.07EssobiOr /whois Jer1326
14:39.09Essobilol
14:39.11*** join/#asterisk fugitivo (~ajf@201.255.107.93)
14:39.19drspermJer1326: ah.
14:39.24EssobiJerjer hiding out, or do we got a new jeremy? :)
14:39.27LethargicclownWhat VOIP provider should I use for *?
14:39.37Essobi;)
14:39.43EssobiI wouldn't admit to it either.
14:39.48Jer1326lol
14:39.49dfunnellMoc: oh, of course ;-)
14:39.53EssobiLethargicclown Use Jerjers.
14:39.54*** join/#asterisk ACiDV (~joel@iteckGW.infoteck.qc.ca)
14:40.16Essobiwhat's the name of that?  Umm. damnit it's too early.. I need coffeeeee.
14:40.24Jer1326nuphone?
14:40.30EssobiRoger, roger.
14:40.32Zeeekis everyoing on the planet having trouble logging in to free SIP accounts today or what?
14:40.35hellopdfunnell: don't be scared of messing with the source... you compile asterisk from the source.  You might break it, you might fix it... you might learn alot.
14:40.38EssobiLethargicclown Use nuphone.
14:40.40FaithXdfunnell, Is this something that just started happening or did you upgrade?
14:40.45faccioneGentlemen, question, could be stupid, but i didn't found a solution around. How can I set asterisk to dial ONLY some international prefix and reject all the other ?
14:40.46dfunnellMikeJ[Jayden]:  Are you suggesting  NoOp(${EXTEN}) before the dial?
14:40.51Essobihellop RTFS is my motto.
14:41.01hellopZeeek, maybe the telcos have blocked the ports.
14:41.04ACiDVI have 2 servers linked with IAX, if I set delayreject=1, I cannot receive call from  only 1 side. if I set delayreject=0 all work on both sides, it's normal ? :D
14:41.16Essobifaccione http://www.voip-info.org
14:41.23EssobiSearch for extension matching
14:41.26Jer1326faccione just edit the dialplan
14:41.27Essobiin the dial plan
14:41.37ZeeekI have a new application I'm releasing the first day of next month
14:41.38faccioneEssobi i didn't found a solution there.
14:41.41Essobi_01001234....
14:41.43dfunnellEssobi:  have spent many hours trawling voip-info.org without success.  Def. appreciate any specific URL's you can recommend.
14:41.45EssobiYou didn't look hard enough.
14:41.55Essobidfunnell use the google search
14:41.58faccionedoes not work
14:41.59Jer1326Zeeek what does it do?
14:42.00ZeeekIr's called EetMe() and doesn't take ANY argunments
14:42.08Zeeekmuhahahah
14:42.20Essobianything you see in a dial plan... use "cmd dial" to lookup dial for instance.
14:42.24hellopEssobi, RTFS
14:42.29Essobi:)
14:42.36EssobiRTFS baby.. RTFS
14:42.42dfunnellFaithX: New box, so new problem.
14:42.42EssobiI do it everyday.
14:42.47Jer1326whats RTFS?
14:42.55EssobiREAD THE FUCKING SOURCE
14:42.58EssobiMaha
14:43.00Jer1326hahahahahahahahahahahahahahahaahah
14:43.08ZeeekRoll to Floor Slowly
14:43.16*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com)
14:43.18EssobiUse the Source, Luke.
14:43.28ZeeekMay the SOurce...
14:43.28dfunnellEssobi:  If I bothered looking properly then how could I get the chance to dazzle you all with my blazing ignorance?
14:43.33EssobiHey Zeeek ... You going to that cook out?
14:43.45faccionethanks
14:43.45Zeeek~cookout
14:43.52Essobidfunnell Write that EatMe app zeek was talking about to he couldn't use that joke anymore.
14:43.57Zeeek~jbot cookout
14:44.06EssobiMehe.. I'll leave that joke along.
14:44.12elricdoes asterisk support 64 bit processors?
14:44.16EssobiSure.
14:44.36Essobialone even
14:45.09Zeeekwhat to do what to do?
14:45.29ZeeekI just got a heart rending letter from a Dr in Nigeria who wants to give me all his money
14:45.40drspermZeeek: count me in.
14:45.40Infernacan anybody help me with format_mp3, i loaded it, made proper musiconhold.conf configuration, but cannot hear any music on hold in queue
14:45.46roamer32364 bit processor - 4 terabyte of dialing plan - individual permit/deny to every dialable number in the universe - yeah ! :-)
14:46.05epochZeeek: wow, sounds like a worthy cause... think he could use an extra couple grand?
14:46.12FaithXInferna, you need mpg123
14:46.23FaithXnot mpg321
14:46.25dfunnellhellop:  Sure, may learn a lot, but need to get this puppy installed v soon (like sometime last week!), so a bit worried I'll spend fruitless hours trawling through source I don't understand.  Surely someone has done this before without messing with the code.
14:46.29roamer323a reload takes nine days on a 16 processor SMP system.
14:46.39*** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com)
14:46.46ManxPower~docs
14:46.47jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
14:46.49tzangermorn ManxPower
14:46.54drspermok...so vonage is out...so which provider should I look at for businesses (without the $$ of a DS1)?
14:46.57ManxPower'morning tzanger
14:46.58Makenshiroamer323, surely you would use realtime for that
14:47.04*** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu)
14:47.05tzangerdrsperm: in or out
14:47.17Inferna<FaithX>  i am using format_mp3
14:47.18drspermof what?
14:47.20ManxPowerdrsperm: I don't believe ANY VoIP company is good enough for a business enviroment.
14:47.27tzangerManxPower: I disagree
14:47.38drspermManxPower: sorry...mainly for testing....
14:47.39tzangerwith a good network connection it's no different than traditional POTS
14:47.55tzangerdrsperm: are you looking for long distance termination or someone to route calls to you?
14:47.58*** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com)
14:47.58ManxPowerEven if the VoIP company is good enough, you still have the Big Bad Internet to cause problems.
14:48.02FaithXInferna, do you have mpg123 installed?
14:48.18drspermhere is my topology:  * box resides behind 200MB ethernet uplink to internet backbone...
14:48.20roamer323Makenshi - sure, and a cluster of mySQL servers chucking away continously :-)
14:48.23tzangerManxPower: agreed, but it's really not *that* bad, and use a second cheap connection from another provider for redundancy
14:48.29drspermphones reside at business class cable modems...
14:48.33drspermpots or sip?
14:48.43tzangerdrsperm: I don't care about your topology, are you looking for origination or termination service?
14:48.48drspermboth.
14:48.51tzangerok
14:49.07drspermI can do pots..but I heard I can pick up an echo....
14:49.10tzangertermination I use nufone with great success.  No, they're not the absolute cheapest but then again they've never gone down for me either
14:49.13drsperm...possibly from a bad source...
14:49.17tzangerdrsperm: you can pick up echo anywhere
14:49.27drspermk
14:49.42tzangerfor origination it seems much more fractured, there are lots of providers who will provide DIDs but I have not heard of rock-solid reliabiliyt from ANY of them
14:49.57tzangerI mean just look at the livevoip/broadvoice/sixtel posts on -users
14:50.02tzangerthose are probably the big 3
14:50.11tzangerI don't use origination myself though
14:50.12ManxPowertzafrir: teliax too.
14:50.36tzangerManxPower: I haven't heard anything about teliax... they might be worth investigating since nobody's openly bashing them (maybe their shit works?)
14:50.37drspermok..so what you are saying is that for getting started and testing, use an existing pots...then if possible move to a pri or such....
14:50.45drsperm^^if afforadable...
14:50.47Jer1326i use telax
14:50.49Jer1326teliax
14:50.49tzangerdrsperm: you can still get echo on a PRI (I had it)
14:50.55drspermk...thanks...
14:50.57tzangerJer1326: how is it for origination
14:51.15Jer1326its great i love it...but i cant reach them when i need them
14:51.19*** join/#asterisk gabb0 (~gabb0@indo1.indosoft.unb.ca)
14:51.35tzangerJer1326: ever have your DID stop working?  How many minutes a month do you originate from them
14:51.35gabb0hello
14:51.36LethargicclownAnybody have any information on broadvoice and *?
14:51.48tzangerLethargicclown: it's all over -users and the wiki.  I don't recommend them
14:51.53Jer1326i do close to 5000min/month and no more DID has never died yet
14:51.55hellopecho is the mind killer
14:51.58`SauronLEthargicclown: If you follow the voip-info.org pages, it works great.
14:51.59tzangerJer1326: very nice
14:52.08tzangerI will have to check them out
14:52.15gabb0Just wondering if anyone has any experience using the tdd agi application or any other method for tty
14:52.19tzangerwe do about 5kmin/mo termination and 10kmin/mo origination
14:52.20hellopno serious, my customers won't tolerate echo...
14:52.38tzangerhellop: yes echo sucks
14:52.46Jer1326they offered me extra channels free too :)
14:52.47hellopdrsperm, why is vonage out?
14:52.50tzangerhellop: if it's THAT much of a problem, spend the $1000 and get a hardware T1 echo can, jeez
14:52.56theogeorHello anybody who has connected * on an MD110 PBX through TLU76/1 card ?
14:53.11tzangerhellop: personally zap's builitn ecoh can seems to work well so long as you really tweak it
14:53.40Lethargicclownwhat's this about echo?
14:53.47tzangerwhat's this about echo?  :-)
14:53.49ManxPowerNNNNOOOOOO!!!!!  Capital One Bank is going to aquire the local bank I opened an account with THREE DAYS ago.
14:53.57drspermhellop: from what I understand softphone only...which is ok..but 500 min is out...
14:53.57tzangerManxPower: that sucks
14:54.13helloptzanger, I'm in hawaii.. hope it's feasible.
14:54.22tzangerhellop: hope what is feasable
14:54.33ManxPowertzanger: I picked that specific bank because they are a regional bank with a good rep for services and decent customer service.
14:54.37*** join/#asterisk channan (~channan9@66.180.121.185)
14:54.46tzangerManxPower: well close it out and tell them why
14:55.21ManxPowertzanger: I'm rather tempted.
14:55.28helloptzanger, I hope that when I setup this Bugetone phone, I can make vonage calls over the Pacific Ocean with a quality that is marketable.
14:55.49tzangertempted, shmempted, tell them that they've got 6 months to prove that their customer service wont' take a bath with the takeover
14:56.14mesiCan somebody try to call me on SipPhone? No 17476014869
14:56.19Jer1326hellop you cant resell vonage service....
14:56.21ManxPowertzanger: Honestly, I wasn't planning on having the account for more than 12 months anyway.
14:56.23channanhello - anyone uses Grandstream Budgetone 100 series IP phone model? I bought the 100 model, everything's working ok except the speak phone volume really bad. Would the 101 or 102 be any better? thanks
14:56.28drspermso is Broadvoice cool?
14:56.28mesiI am afraid when calling myself, the line is busy and thus I am unavailable :-(
14:56.39`Saurondrsperm: Haven't had any problems here
14:56.42hellopJer1326, oh.  No ersellers contracts?
14:56.47Zeeekchannan - return the phone if the volume is bad
14:56.49Lethargicclownmesi: it's busy
14:56.50drsperm`Sauron: thanks...
14:57.01hellopJer1326, sides, not reselling, installing their service for a client.
14:57.05mesilethargicclown: That's strangge! :-(
14:57.08drspermThis channel is too active for a Monday morning...
14:57.15`SauronOnly thing I'm waiting for is for them to port my number..
14:57.16channandrsperm- broadvoice is cool. I used it since Christmas and had no problem, calling everywhere in US and Europe
14:57.18`Saurondum di dum
14:57.19`Sauronyawn
14:57.23ManxPowerdrsperm: We blame Slashdot.
14:57.33`SauronOh, right.
14:57.38`Sauronthe * article
14:57.42Zeeekjoin #asterix
14:57.46`SauronI saw it, and bookmarked the page
14:57.46LethargicclownYep, slashdot
14:57.46*** join/#asterisk mrgoby (~mrgoby@defactowireless.org)
14:57.49`Sauronfigured I'll read it later
14:58.01channanthe only thing I don't really like is it's hard to get tek support (even though I did not need it, just want to see how responsive its support is)
14:58.09tzangerhellop: you're using the cheapest of cheap -- you might want to ebay yourself a nice cisco 7960 or polycom just to compare
14:58.21tzangerdrsperm: you should see it when bkw's in his element
14:58.31drspermah...
14:58.31`Sauronchannan: with BV? You just have to wait a while. ONce you get a guy on the line, they're good
14:58.37ZeeekPlease note that when Newsweek or International Herald Tribune does a big story on voIP they talk about Jeff Pulver and Vonage, never mention asterisk.
14:58.45dfunnellMikeJ[Jayden]:  Tried NoOp(${EXTEN}) before Dial, but * also does NoOp as soon as pattern is matched and then moves to next line (i.e. then tries to dial).  Wondering if this isn't a CAPI problem after all?
14:58.46tzangerZeeek: mindshare
14:58.52ZeeekI guess it's because of the media machines these companies have
14:59.00Zeeekmindshare?
14:59.02helloptzafrir, 7940's on the way
14:59.10drspermtzanger: you just brought up something...I just ordered some Polycom 500's...r they good (vs. Cisco)
14:59.17parits all about the benjamins
14:59.27Zeeekyou have to have a mind to share and that lets me out
14:59.33tzangerZeeek: exactly -- asterisk is much less noticeable becuase they're not spewing their name everwhere and coining phrases like "I got voiped"
14:59.41tzangerdrsperm: they are supposed to be very nice
14:59.46ManxPowerCisco: You don't get SIP firmware with it, you don't get a power supply with it.  Polycom: You get SIP firmware with it, you get a power supply with it.  You pick.
14:59.47drspermcool...
14:59.49tzangerI don't do SIP so I cant' say from experience
14:59.55ZeeekDuuuude... I got Ass terisked
15:00.00tzangerhahahaha
15:00.03Jer1326hahaha
15:00.03helloptzanger, I was just wondering if the long distance to Hawaii, would make the VOIP unusable due to echo...
15:00.05tzangerdude, your ass got tricked
15:00.07drspermThat is what I heard...Cisco is very $$ too..
15:00.12Zeeekkiss my ....
15:00.27ManxPowerhellop: Location doesn't matter.  Network latency and jitter matter.
15:00.38tzangerhellop: only if your hybrids are shit or the handset is so fucked up (mechanically) that all your earpiece audio gets received by the mic
15:00.52channanZeeek-I bought in ebay and can't return (did not notice the prob until I needed :( ). are you saying that you don't have that problem? Perhaps mine is just a defective unit?
15:00.55ZeeekThe Internat'l Herald Trib has "email this article... AIM this article... IM this article"
15:01.08tzangerwhat's the difference between AIM and IM
15:01.08tzanger?
15:01.16dfunnellEssobi:  Searched again for resolution, including looking at cmd Dial, haven't found anything new to help.
15:01.26Zeeekchannan I'd say so because [inspite of all the guys that diss the GS line] the phone works GREAT! (for $75)
15:01.30*** join/#asterisk stevekstevek (~stevekste@slim-eth0.horizonlive.net)
15:01.35MuppetMasterAIM is specific to AOL while IM is a broader term for AOL, MSN, Yahoo, Jabber, SIMPLE, etc.
15:01.36tzangeryes for the money it is not a bad phone
15:01.43Zeeekit looks a little wacky and has clown buttons
15:01.50Zeeekbut the speaker ROCKS!
15:01.53tzangerthe white ones look like ass but the lback ones are nice
15:01.58drspermanyone heard of a telco called "Topaz"...I hear they have good $$ on site to site DS1's
15:02.03Zeeeksomehow I could onlmy order black ones
15:02.08tzangerZeeek: if you think the gs speakerphone rocks you have never been on a good conference call with a speakerphone
15:02.11ZeeekI have a 101 and a 102
15:02.29tzangerthe polycom (non voip) speakerphones are bar none, hands down the *best*
15:02.31Zeeekwell I don't need a $2000 sound point, thas fo sho'
15:02.38LethargicclownNothing's better then my half duplex speaker phone!
15:02.45tzangerLethargicclown: yeah, no echo there!  :-)
15:02.46Lethargicclownfor only $120
15:02.48Zeeekbut I am getting an ip500 so I'll see what that's up to next
15:03.02parwho is the cheapest CLEC offering T1?
15:03.28drspermpar: site to site?
15:03.32Zeeekalso ordered on of them IAX chinses ones "just to see"
15:03.32paryepo
15:03.42drspermpar: State? City?
15:04.06parno sorry instead.. not site to site
15:04.20drspermk...where?
15:04.25partexas
15:04.27drsperm4 voice i take it...
15:04.30drspermcity?
15:04.35parno not a PRI
15:04.42drspermk...city?
15:04.43parintegrated T1
15:04.46parSan Antonio
15:04.52channandrsperm-cisco7960 quality is really good. I owned a few. the only thing is it's pretty hard to set up the first time. My friend said it sounds a lot better than caling thru my analog home phone. the price is steep though
15:04.56ManxPowerpar: nobody can answer that.
15:05.03ManxPowerpar: You need to do the research for your location.
15:05.21drspermpar: I can...I live in SA....and own an ISP in sa...  :)
15:05.30ZeeekIt's Katty!!!! http://www.mindsay.com/network/kat1587
15:07.33*** join/#asterisk wazquis (~akv@lnxbx.dk)
15:07.58wazquisain't it possible to run asterisk without a soundcard? i cannot start asterisk up on my laptop
15:08.04Zeeeksure it is
15:08.16Zeeekdon need no stinking sound card
15:08.23wazquisi tried noload => alsa.conf and oss.conf
15:08.34Zeeekwhat error do you get?
15:08.37wazquis.......Ouch ... error while writing audio data: : Broken pipe
15:08.40ManxPowerwazquis: You mean noload => chan_alsa.so and chan_oss.so, right?
15:08.43Zeeekbefore that
15:08.54ManxPowerwazquis: That error is from mpg123
15:08.56MuppetMasterwazquis:  It is possible to run Asterisk with a soundcard.
15:09.03Zeeekor not.
15:09.08wazquisManxPower, oh...
15:09.11wazquisi'll check
15:09.25kantWhen using the 'switch =>...' statement to forward a call to another * server, will CDRs be written to the forwarded * server?
15:09.41wazquisManxPower, yes... chan_alsa.so and chan_oss.so
15:09.45ManxPowerkant: Only for the portion of the call that the remote server handles.
15:10.37helloptks all
15:10.42ZeeekQuestion: A call comes in on ZAP with CID. You do NOT answer it. Is it then possible to call back on the same line if the caller hung up? Without a .call file ?
15:11.02ManxPowerZeeek: Yes.
15:11.06kantSo if I forward a call to an * server with all the PSTN hookups, that * will have the CDRs corresponding to the calls made through it?
15:11.22ManxPowerkant: Yes.
15:11.29wazquisnoload=>chan_alsa.so
15:11.34wazquis........Mar  7 16:10:33 WARNING[1077138752]: chan_skinny.c:2584 reload_config: Unable to get our IP address, Skinny disabled
15:11.35ZeeekManx I tried it in the h extension but no matter what I did it wouldn't work
15:11.38wazquisthe only error i get now..
15:11.45Zeeekdo I need to lmiberate the channel, destroy it?
15:11.49wazquis(and the mpg123 thingie..)
15:11.59kantAnd the originating * will have CDRs corresponding to the 'switch =>'?
15:12.04ManxPowerwazquis: Unless you are using the Skinny protocol, that is a harmless message.
15:12.18Zeeeknot an error but a WARNING
15:12.28ManxPowerZeeek: You mean AUTOMATICALLY call them back?  No.
15:12.36Zeeekheh, amusing
15:12.54Zeeekya, I can call them back on my cell too if I write the number down :)
15:13.09wazquisManxPower, okay... but the asterisk ain't starting up...the last message is the one you said was from mpg123
15:13.12ManxPowerWhy not use the call back function of your phone?
15:13.24*** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it)
15:13.30ZeeekManx what I did was create an app that waits 2 seconds and then copies a .call file to outgouing
15:14.42*** join/#asterisk linagee (~linagee@netblock-66-245-227-114.dslextreme.com)
15:14.52linageecool. they have a user friendly asterisk iso now. :)
15:15.17dfunnellEssobi:  Come on, Essobi, you talk big about not searching hard enough, but it doesn't sound like you have any more idea than I do re. finding an answer to this one!
15:17.15ZeeekManx - it's not a phone that's being called it's a DID from the USA
15:17.24*** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com)
15:17.45Zeeekso the .call file works fine, it was just a thought about changing the channel dynamically
15:18.04RoyKanyone that knows what to do if the te410p doesn't get/put any interrupts?
15:18.14ZeeekDIDcomingn in to asterisk I mean, needing in fact to call out SIP to a phone elsewhere
15:18.46ManxPowerRoyK, Put it in a system without the incompatable Intel chipset.
15:18.49*** part/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net)
15:19.06RoyKManxPower: well, I have another box with bloody identical hardware and there it workks
15:19.11RoyKs/kks/ks
15:19.59ManxPowerRoyK, It's a problem with a specific Intel Chipset.
15:20.14techieAnother day in the world of packetized voice.
15:21.12*** join/#asterisk MasterYoda (~mnicholso@dhcp-155.digium.com)
15:21.21ManxPowerRoyK, see the mailing list archive.  I can't find the page on digium's web site that talks about it.
15:22.22*** join/#asterisk tull (~danka@wwwcache2.livjm.ac.uk)
15:22.28tullhello
15:22.41tullis anyone using sipura 2000 or spa3000?
15:23.00ast_freaksipura 2000
15:23.07tullmay I pvt?
15:23.33ast_freaksure
15:23.38*** join/#asterisk viLeR (1000@ip-33-7.telesat.com.co)
15:24.02wazquisManxPower, the asterisk doesn't start up... i if it those two aint problems..i cannot see what is going wrong
15:24.45*** join/#asterisk hemant (hemant@220.226.25.97)
15:24.47ManxPowerwazquis, noload => chan_skinny.so in /etc/asterisk/modules.conf and rename /etc/asterisk/musiconhold.conf to some other name.
15:26.41Zeeekwazquis what you should do is pastebin the console output, several lines before it stops
15:26.47Zeeek~pastebin
15:26.49jbotrumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
15:26.59kantFinally I get multi-homed connection.
15:27.04RoyKManxPower: you sure? http://www.voip-info.org/wiki-Asterisk+TE410p+No+Interrupts
15:27.19ManxPowerRoyK, Therre it is.
15:27.36*** join/#asterisk sambal (~sambal@gateway.office.flatbox.nl)
15:27.43ZeeekHow common is use of IAX (not IAX2) today?
15:27.56ManxPowerRoyK, That's not the page I was thinking of.
15:28.00RoyKManxPower: ok
15:28.04ManxPowerZeeek, nobody uses IAX
15:28.08tzangerhahhaa
15:28.35ZeeekI'm asking because of the zillion pages on the web that show the wrong port
15:28.53Zeeek[general]
15:28.53Zeeekport=5036 ; What port to use
15:29.13ManxPowerZeeek, Asterisk will ignore that in modern versions.
15:29.28Zeeek[general]If shit like that is NOT true, we should all get rid of it everywhere we can
15:29.43Zeeekit's BAD to even show stuff like that
15:29.51Zeeekneedless waste (a little like my posts here)
15:30.47RoyKManxPower: how come?
15:30.52Catalyst4ChangeReform Asterisk Social Security now
15:31.15wazquisManxPower, ok...i get no errors now... but it just stops after initializing... http://pastebin.com/250517
15:31.31*** join/#asterisk jalsot_ (~tamas@abacus.eworldcom.hu)
15:31.44MasterYodaZeeek: what about port = 5036? for iax?
15:31.58Zeeekit's obviously not right for IAX2
15:32.07MasterYodaZeeek: well iax is 4569
15:32.56mesiWhat can be the reason that I am unavailable on sipphone.com if I am registered and can make calls and the firewall is open?
15:32.57*** join/#asterisk file (~file@251.134.218.209.transedge.com)
15:32.58RoyKManxPower: but then how do you explain that the card works in identical box?
15:33.34tzangerRoyK: easy, the identical box isn't
15:33.37ManxPowerRoyK, I can't.
15:33.41tzangerRoyK: or the card got damaged in transit
15:33.45dfunnellAll - pretty sure that my problem is * related and not CAPI related.
15:33.46tzangerRoyK: or the motherboard has a bad port
15:33.47dfunnell* tries to do whatever (run macro, dial, etc.) as soon as it hits '.' in _102. or similar.  Is this normal?  Any way of getting it to wait for user to dial whole number?
15:33.48ManxPowertzanger, Well that is the OBVIOUS answer.
15:33.50tzangerRoyK: any number of reasons
15:34.04tzangerRoyK: or the BIOS isn't set to identical settings (I have personally seen this)
15:34.07ManxPowerdfunnell, no.
15:34.23MasterYodadfunnell: dial faster....
15:34.31tzangerMasterYoda: :-)
15:34.35ManxPowerdfunnell, unless you do NOT know the number of digits.  Of course if you are using a SIP phone this doesn't apply.
15:35.08dfunnellMasterYoda:  ;-)
15:35.21*** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com)
15:35.21*** mode/#asterisk [+o anthm] by ChanServ
15:35.46dfunnellManxPower:  Bugger.  That is the whole problem - don't know number of digits that make up number.
15:37.18RoyKManxPower: http://karlsbakk.net/asdf.txt
15:37.25RoyKManxPower: that's a simple lspci
15:37.37*** join/#asterisk Jas_Williams (~Jason@host81-155-66-178.range81-155.btcentralplus.com)
15:37.43dfunnellManxPower:  Are you saying that, for all of it's features, * doesn't provide a (relatively straightforward) way to support variable-length numbers?
15:39.02*** join/#asterisk xorol (~x@213.219.182.88.fixedpower.by.edpnet.be)
15:39.04xorolello
15:39.11ManxPowerdfunnell, If you don't know the number of digits (and only if you don't know the number of digits) then you have to use "." in your pattern.  Of course, if you are using VoIP then this does not apply.
15:39.15xoroli got a little question ..
15:39.23jontowi have a feeling we're gonna see that '@home' nonsense an awful lot in the next couple days ;)
15:39.35linageejontow: lol
15:39.38ManxPowerdfunnell, no, I'm saying that most people that think they need variable length numbers don't actually ned them.
15:39.48linageejontow: naw. i was in here prior to @home stuff.
15:40.07xoroli'm trying to transfer a call by using dial(channel,30,t)  but i can't transfer the call afterwards (im using WELLTECH 1501 (fxs gateway) as hardware)
15:40.16ManxPowerjontow, Thank Dog that I'll be online much less over the next few days.
15:40.21dfunnellManxPower:  Please go on.  I'm only having problems dialling out, where I don't have the ability to control number length.
15:40.21jontowhahah
15:40.21xorolit won't react by sending the # ...
15:40.34linageejontow: personally i don't like @home stuff. ;-)  make users buy phones, not backend systems! :)
15:40.40*** join/#asterisk Smythe (~Smythe@spock.cbcag.edu)
15:40.42ManxPowerPinhole, And your AGI is run twice for each call?
15:40.48dfunnellManxPower:  When I use '.' it tries to dial as soon as it reaches the '.' (i.e. doesn't wait for the whole number)
15:41.09ManxPowerdfunnell, You never told me what device you arre using to dial.
15:41.14jontowpersonally .. i like ramps for learning curves, but eh.. honestly sometimes you need to just climb the ladder :)
15:41.22PinholeManxPower, yup.  sometimes more than that.
15:41.33ManxPowerPinhole, THAT is the problem with _.
15:41.44dfunnellManxPower: example dial string:  exten => _102.,1,Dial,CAPI/470,${CALLERIDNUM},${EXTEN:1}
15:42.12ManxPowerdfunnell, Are you going to make me start saying naughty words to get you to tell me what phone you are using?
15:42.28*** join/#asterisk Ubuz (~momo@DSL212-235-37-117.bb.netvision.net.il)
15:42.31dfunnellManxPower:  Doesn't only seem to be problem with CAPI, if I try and run macro of function (such as NoOp) it happens straight away too.
15:42.45dfunnellManxPower:  Sorry, SIP phones - Granstream
15:42.51dfunnellBudgetone 100
15:43.09ManxPowerdfunnell, Your phone collects the whole number, then sends the entire dialed number to Asterisk.
15:43.19linageemuch moreso than traditional four lettered words
15:43.37dfunnellManxPower:  Give me a couple of secs to try something out.  Dont' go anywhere!
15:43.46ManxPowerdfunnell, So look at the phone.  Specifically Early Dial does not work on those phones.
15:44.11linageeManxPower: unless of course you tell your phone to send numbers as soon as they're dialed or something weird and unstandard. (grandstream phones can do that at least)
15:44.14*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
15:44.14*** mode/#asterisk [+o bkw_] by ChanServ
15:44.34ManxPowerlinagee, Yes, but that feature does not work in any phone I've heard of.
15:44.41linageeManxPower: probably not. ;-)
15:44.54*** join/#asterisk Bentley (~rbc@S01060080c8135e6a.cg.shawcable.net)
15:45.06*** join/#asterisk jalsot_ (~tamas@abacus.eworldcom.hu)
15:45.19dfunnellManxPower:  So you think I need to turn early dial off?  If so then do you know any way of avoiding a delay between user dialling internal extension and it actually doing anything?
15:45.22RoyKManxPower: see http://karlsbakk.net/te410p/. the difference between the two lspci -vvv output is two lines
15:45.42jontowdfunnell; with the grandstreams, you can 'tune' the timeout period for dialing..
15:45.45jontowits in the web GUI :)
15:45.48ManxPowerdfunnell, You use a non-piece-of-crap phone
15:46.01jontowthats about all you've got for options though..
15:46.46ManxPowerdfunnell, mid and high end phones let you put a dialplan in the phone.
15:46.56dfunnelljontow:  Ok, thanks.  Also have 4 x analogue phones plugged into Zap's, am I going to have same problem with those?
15:47.04jontowno clue
15:47.11jontowi've yet to use ATAs in any way shape or form..
15:47.12ManxPowerdfunnell, But if you use . in your dialplan you will have to wait until DigitTimeout before the call will happen.
15:47.13jontow<PROTECTED>
15:47.14dfunnellManxPower: I see.  Thanks for your help.  Trying now.
15:47.16mesiEverything on sipphone works for me except that I can't be called on my number.
15:47.25jontowmesi; so setup your extension..
15:47.32*** join/#asterisk trimi` (Pharrel@62.162.232.119)
15:47.35ManxPowerdfunnell, zap devices send each digit as it is dialed.
15:47.49dfunnellManxPower:  Thanks ManxPower.  Do you mean DigitTimeout in phone?
15:47.52*** join/#asterisk JerJer[mobile] (~jj@feth100-fw.fament.net)
15:47.57jontowno, DigitTimeout in *
15:48.06ManxPowerdfunnell, for zap, digittimeout in *
15:48.20ManxPowerfor sip, the digit timeout on the phone
15:48.25MikeJ[Jayden]Manx, you should get paid for this...
15:48.34MikeJ[Jayden]somone paypal Manx some cash
15:48.34dfunnellManxPower:  Oh, I see!
15:48.36ManxPowerMikeJ[Jayden], too bad NOBODY else feels that way.
15:48.57dfunnellManxPower:  I've already offered to pay in beer... when are you next in NZ?
15:49.15MikeJ[Jayden]who here apretiates all of ManxPower's help?
15:49.59mesijontow: I set it up. It doesn't seem to work. In sip.conf I say context=extern-sip, so that this context would be the default for incoming sip calls. And the register line names /1747 at the end. So this extension should be called, right?
15:50.16jontowmanxpower; i've always felt that way.. but i can't stand paypal :(
15:50.27jontowyou helped me repeatedly in subtle ways when i was first picking this stuff up ;)
15:50.30dfunnellMikeJ[Jayden]:  Let me answer that once I see if it is of any use ;)
15:50.38ManxPowersee my /away message
15:51.33MikeJ[Jayden]so, you wana go to europe?
15:51.34ManxPowerjontow, I'm alwways most proud of my insults. 8-)
15:51.39ManxPowerMikeJ[Jayden], Yes.
15:51.39jontowheheh
15:51.51ZeeekGUYS This is it ! the breakthough!
15:51.54jontowi wouldn't mind europe.. :)
15:51.59jontowi think it'd be a hell of a good time
15:52.01ManxPowerMikeJ[Jayden], I would prefer Benelux, but most any place there would be OK
15:52.04ZeeekThe KILLER VOIP APP not kidding
15:52.04jontowat least until i get arrested for something stupid :o
15:52.04visik7european software patents has been approved
15:52.15BuckRogers<PROTECTED>
15:52.44ManxPowerTo everyone that I've helped: Find me a job in Europe.
15:52.51jontowmesi; thats a little out of context.. pun intended..
15:53.08ZeeekAll voIP providers need to arrange credit accounts. When we want to pay someone that helped us, we just dial a number at the provider and Allison will arrange the cross billing! What do you think?
15:53.17jontowmanx; i'd be looking.. but im looking for a new job for me.. my employer(s) are a joke :/
15:53.33ZeeekManx want to do PHP, C and web design in Paris?
15:53.35mesijontow: What is pun?
15:53.36ManxPowerjontow, My entire COUNTRY is becoiming a joke. 8-)
15:53.46ManxPowerZeeek, I am not a programmer.
15:53.48jontow;) you're in the USA like me, hey?
15:53.57ManxPowerjontow, Yup.
15:54.08jontowyeah.. agreed in full.
15:54.18ZeeekManxpower there are a lot of startupasterisk based businesses here in Paris
15:54.36ManxPowerjontow, Not just That Bush Thing.  It all started in the 1980's
15:54.40jontowif you find that position in europe.. want an apprentice?  my gf too :P
15:54.41techieParis, nice.
15:54.58jontowi know .. i was born in the 80s, of course it was ajoke :)  what with the spandex, big hair, and showboat-rock-singers
15:55.04ZeeekManxPower, is the word "cheese-eatin surender mnkeys" in your vocabulary? If it is, forget it!
15:55.16*** part/#asterisk kant (~bernd@207.42.191.67)
15:55.16ManxPowerZeeek, Unfortunatly, I'm a VERY VERY bad programmer.  I can do simple scripts in perl, PHP, C, but I just can't do big projects.
15:55.29*** join/#asterisk Rival (~rival@66.177.249.219)
15:55.36ZeeekAsterisk based starups Manx, I'll arrange it for you
15:55.45ManxPowerZeeek, I'm not all that fond of what little French culture I've seen/heard of, but I don't hate the french.
15:55.53Zeeekok, thazt's good
15:56.10jontowI like the French.. but I'll be honest.. I'm just a sucker for the wine.
15:56.22Zeeekbecause Amer'ican culture isn't much to be crazy about either :) To each his Zone
15:56.27jontowthe language is fun too :)  especially once you're all hopped up on the wine.
15:56.30ManxPowerI have to major issues with working in Europe.  1) I'm not a citizen of the EU, so there's lots of paperwork 2) Like most americans, I only speak English.
15:56.53ZeeekBoth can be fixed Manx - why do you want to work in Europe?
15:56.59xorolmanx : belgian speak the BEST english in the EU (besides UK ofcourse .. try me )
15:57.06ManxPowerZeeek, I want to get out of the USA on (maybe) a perm basis.
15:57.17ZeeekCanada isn"t good enuf?
15:57.20ManxPowerxorol, Well, my preference would be a job somewhere in Benelux.
15:57.33ManxPowerZeeek, Canada would just be a stepping stone to Europe.
15:57.34xorolmanxpower : we are hiring :)
15:57.52xorolmanxpower : we are even located in brussels
15:57.56ManxPowerxorol, Please tell me you are hireing for something other than a programmer.
15:57.58Rivalis there a problem with http://asteriskathome.sourceforge.net/
15:58.00ZeeekManx - the language is not a problem in this business and you surely can network here and in other places to find a job
15:58.02Rivali cant connect
15:58.04ManxPowerxorol, I may be in Burssels this summer.
15:58.17xorolManx:: just pass by .. we'll have a lunch in the Hilton
15:58.18Rivalis there a secondary link anyone would be kind enough to msg me with?
15:58.27ManxPowerxorol, your e-mail address?
15:58.49ManxPowerxorol, If I get a job offer, I won't wait until summer. 8-)
15:58.59ManxPower~docs
15:59.00jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
15:59.12*** join/#asterisk kswtch (~killswitc@213.146.107.241)
15:59.16ZeeekManxPower where can we find a recent CV of you ?
15:59.19ManxPowerI want to go to Europe this summer for VON and Astricon.
15:59.31ManxPowerZeeek, I have to finish making one.
15:59.50ManxPowerxorol and zeek, /msg me your e-mail addresses.
16:00.15RoyKManxPower: did you see the lspci outputs?
16:00.35ManxPowerRoyK, No.  I can't really help with hardware issues at that level.
16:00.59RoyKManxPower: the point is, they're equal
16:01.24ZeeekManxPower send one to me here: ManxPower-RMRAsterisk@sneakemail.com
16:01.32ManxPowerRoyK, The only thing I know about no-interrupts is what I read on the mailing list.
16:01.39ManxPowerZeeek, Does that address expire?
16:01.48nestArwtf
16:01.51Zeeekno but I'll have to kill it if I start getting crap
16:01.58nestArPolycom says you can't reprogram the softkeys
16:02.00ManxPowerZeeek, Of course.
16:02.00ZeeekSneakemail rulez!
16:02.23Zeeekshould be fine for a few weeks minimum
16:02.52nestArargh
16:02.53ZeeekAlso Max, take a look at the guru.com site and see if stuff is open in the areas that interest you (both geo and work wise)
16:03.02ZeeekMax/Manx/
16:03.05nestAri'm starting to regret buying these phones.
16:03.16Zeeekwhich phones nestAr ?
16:03.21nestArPolycom's
16:03.29ZeeekOH?
16:03.36nestArwhat's the point of having soft buttons
16:03.40nestArif you can't change them
16:03.40ZeeekI haven't ordered yet, what's wrong?
16:03.57Zeeeknone of the docs mention this?
16:04.12nestArthere's a section about reprogramming a whole key
16:04.23nestArdoesn't exactly make a lot of sense to me
16:04.24dfunnellManxPower:  Thanks Manx, it all works, can't believe it was that easy!
16:04.32nestArso i inquire with tech support
16:04.35*** join/#asterisk gdb (~cbell@circe.inetdb.com)
16:04.36nestAri get a one line reply..
16:04.41nestAr"Voip engineering has not allowed the softkeys to be programmed.
16:04.41nestAr"
16:04.56ManxPowernestAr, Polycom does NOT do end user support.
16:05.02ManxPowernestAr, they lie.
16:05.27dfunnellManxPower:  Where do I send the beer?
16:05.38ManxPowerI managed to program a softkey on a Polycom about 6 months ago.  Decided that it was more trouble than I had time to deal with at that time, since it was a "cool thing" and not a "required thing"
16:05.40*** join/#asterisk waszi (waszi@vorlon.icpnet.pl)
16:05.51ManxPowerdfunnell, Can you paypal it to me?  *grin*
16:06.22nestArfucking cocksucking bastards is what they are.
16:06.31ZeeekPolycom guys, what docs do you have? I noticed about 5 different PDF available in open download?
16:06.39ManxPowernestAr, I talked to at least one other person that did it at the time, the FUCTION of the key is set in one place, the LABEL for the key is done in the language localication stuff.
16:07.05*** join/#asterisk eKo1 (~bernd@207.42.191.67)
16:07.15ManxPowerZeeek, I use the Polycom Admin Soundpoint IP SIP document.
16:07.38Zeeekand that has enuf tech data to program if you have the time?
16:08.02ManxPowerZeeek, That and a LOT of work.  It's a reference guide for people that already know how to do it.
16:08.30Zeeekso in short, when we get this PayManx() app ruinning, you'll be able to do it?
16:08.44ManxPowerLOL!
16:09.09Zeeekdid you see what I said above? Providers like nufone could give us a number to call and this would credit your account (with them or for cach)
16:09.12MuppetMasterAnyone on here having problems with MOH after upgrading to v1.0.6?  http://voxilla.com/forum-viewtopic-t-2586.html
16:09.22Zeeektotally doable by and for the asterisk community
16:09.36ZeeekMuppet ya, it's broken
16:09.49ManxPowerMuppetMaster, Yes, 1.0.6 has a MoH bug.  get CVS -r v1-0 or wait for 1.0.7
16:09.53*** join/#asterisk jalsot_ (~tamas@abacus.eworldcom.hu)
16:09.53MuppetMasterOkay, is there already a bug report.
16:09.53Zeeekhttp://willypick.mindsay.com/?entry=16
16:10.09Zeeekans a patch
16:10.26mesiCan somebody please call 17476014869 again and tell me what happens?
16:10.31mesion sipphone.
16:11.07ZeeekYou you call a number like Pay-Manx on nufone and nufone moves credit from your account to the payee's - ya see?
16:11.14Zeeekthis is what the Japanese do with cellphones
16:11.24Zeeekbypass PayPal
16:11.46Zeeekinstant notification - the payee can then help you
16:11.54Zeeekit isn't as dumb as it sounds
16:12.17Zeeekall stays in the community
16:12.23Zeeekanyone out there?
16:12.47ManxPowerZeeek, It's a cool idea, seems like it would require a LOT of work not related to Asterisk.
16:12.48JerJer[mobile]nope
16:12.51JerJer[mobile]just us bots
16:12.51kswtch"We're sorry, the user bla bla has not setup voicemail bla bla "
16:12.53kswtch@ Zeeek
16:13.36Zeeekjust as I convinced my partner to use the SIP unlimited channel, ever rpovider is UNREACHABLE
16:13.43MuppetMasterZeeek:  Where may I find the bug report for MOH that contains details on the patch?
16:13.49kswtchoh it was mesi who requested the call...
16:13.56MuppetMasterZeek:  Can't seem to find it on Mantis with MOH/music on hold/etc.
16:14.12Zeeekjust a sec I was there yesterday
16:14.43Zeeek<PROTECTED>
16:15.01Zeeekin fact download THIS: http://bugs.digium.com/file_download.php?file_id=5032&type=bug
16:15.53ZeeekManx a lot of shit requires a lot of work not related to Asterisk
16:16.10MuppetMasterThanks!
16:16.10Zeeekone thing is sure - PayPal, as good an idea as it is, SUCKS
16:16.23jontowso does adobe acrobat reader for unix
16:16.27Zeeekthey fucking spam me every time I touch anything PayPal related
16:16.55ZeeekAdobe in genral sucks and thaey also are world class spammers and home callers - as good as Real Networks
16:17.04nestAri don't get any paypal spam
16:17.06nestAr:dunno:
16:17.24`SauronI think paypal has slowly crept up far enough in my bayes filter that they're getting filtered out
16:17.25eKo1I get paypal spamn.
16:17.36*** join/#asterisk xpasha (~pavel@217.30.252.68)
16:17.43`Sauronwhich has the added side effect of also filtering legit paypal email (payment confirmations, etc)
16:17.47xpashahi
16:17.51eKo1Don't know why since I never use it.
16:18.02xpashaanybody used zaphfc from bristuff?
16:18.14ZeeekThey keep sending me stuff about how my life would be better if I got Verified(tm)
16:18.26`Sauronhum
16:18.30`Sauronyeah
16:18.32`Sauronthey do that to me too
16:18.40`Sauronit's cuz they want access to your bank account
16:18.42ZeeekEvery purchase
16:18.43shaZwazverified ?
16:18.56ZeeekI did that once - too scaray
16:18.57shaZwazoh
16:19.01xpashaMar  7 17:18:16 localhost kernel: zaphfc: bchan rx fifo not enough bytes to receive! (z1=528, z2=527, wanted 8 got 2), probably a buffer overrun.
16:19.03`Sauronyeah
16:19.05xpashawhat is the shit?
16:19.14`Sauronthey want you to associate your paypal account with a bank account
16:19.21`Sauronso they can do ACH payments/withdrawals
16:19.24ZeeekPLUS I tried to pay using my normal debit card at two sites yesterday and the payment was refused
16:19.54`SauronI did finally break down and associate one of my low-limit credit cards with my paypal account
16:19.55*** join/#asterisk alerque (~alerque@bear.ouraynet.com)
16:20.04`Sauronthat way they can't screw me out of too much money
16:21.10*** join/#asterisk beta3 (~dan@dan2.active.supporter.pdpc)
16:21.27eKo1No such switch 'IAX' <--- I always get this when doing a switch => IAX/...
16:21.33beta3how do I do sutter tone based on channels
16:21.40beta3every time I set it seems to affect globally
16:22.00*** join/#asterisk Trionnis (buffy@12-203-113-15.client.insightBB.com)
16:22.01Rivaldidnt ebay buy out paypal?
16:22.04eKo1It's as if IAX doesn't exist...
16:22.06`Sauronyeah
16:22.22Trionnisheh
16:22.29Trionnisthose guys are rich mofo's too
16:23.28beta3???
16:23.38Trionnisthe guys that started paypal
16:23.51Trionnisstarted it here at UIUC Beckman
16:24.03Trionnisnow they're rolling around town in H2's
16:24.10Trionnislol
16:24.20Rivalah they earned it
16:24.24Rivalpaypal was a good idea
16:24.33Trionnisyes, it was
16:24.47jontowpotentially bad implementation though :)
16:24.59Trionniscan't say I'm real keen on thier business methods
16:25.00nestAri replied to polycom that I believed they were lying to me..
16:25.10nestAri feel better now
16:25.29Rivaltrue but same can be said for any uber succesfull business
16:25.30wolfsonother than the fact that regulators in multiple states are after them for being an unauthoirzed bank
16:26.07ManxPowerI only run reasonably small amount of money thru Paypal.
16:26.14Zeeekwhich is why my idea of paying thru our trusted providers si a good one :)
16:26.18beta3PayPal is fraudulent
16:26.32nestAri love paypal
16:26.34Zeeektrusted being the key word here
16:26.35nestAri use it all the time
16:26.42ZeeekPooPal
16:26.49NuggetI use paypal, but I sure don't trust it.
16:26.59beta3ManxPower: do you have any idea on how to get stutter tone for differing voicemail boxes on different channels
16:26.59Nuggetand it's a very expensive service
16:27.11Zeeekthe results are in, we all use it but onlt one person trusts it
16:27.13BuckRogerspaypal is used for the tranfer of money that are directed at websites that promote islamic extremism
16:27.17jontowi don't use it :)
16:27.18jontowi never have
16:27.32BuckRogersdonations to them are accepted by paypal
16:27.55nestAri have no reason to not trust it
16:27.55ManxPowerbeta3, mailbox=mailbosnumber@thecontextinvoicemailconf
16:27.55nestArnever had a problem with their service
16:27.55ZeeekBuck so are Citibanks and every other important financial institution
16:27.58beta3ManxPower: thats not the issue, its the problem of getting different mailboxes for different zaptel channels
16:27.59ManxPoweryou can have multiple mailbosnumber@thecontextinvoicemailconf seperated by a ,
16:28.00jontowI had an account once.. only because they said they'd give me $10 for free if i did.. and all i had to do to get it was request that they mail me a check.. i requested said check, it never came, i never actually used the account due to that :)
16:28.00BuckRogersi have never had a problem with their service either
16:28.17BuckRogersyes i know zeeek
16:28.18jontowi figured if they wouldn't come through on such a simple offer, then handing them my money wasn't going to be a beneficial thing for me.
16:28.19Nuggetthe plural of "anecdote" is not "data".
16:28.23jontowthat was .. years ago
16:28.28BuckRogerswe have a major problem
16:28.44BuckRogersim not trying to single them out
16:28.51ManxPowerbeta3, I don't really understand your issue.
16:28.54Nuggetthere are numerous stories of paypal freezing accounts due to misunderstanding and never returning the money.
16:29.14`SauronNugget: Which is why I have never verified my account
16:29.18Nuggetpaypal is perfectly reliable until they decide you're doing something bad, and once that happens you're screwed.
16:29.19BuckRogersyeah and where does that money go when frozen
16:29.29beta3ManxPower: everytime I set mailbox= in zapata.conf, it sets it globally, I want to set it for each fxs
16:29.30Nuggeteven if you're not doing something bad
16:29.52cbachmanNugget... that's perfectly true.  My understanding is that there is one (1!) individual who decides what is bad
16:29.59BuckRogersi got realtime astrisk woking very good right now
16:30.02ManxPowerbeta3, you need to define each channel seperatly, then define the mailbox= right above the channel => for that channel
16:30.12beta3ManxPower: I've alreayd done that
16:30.13BuckRogersmuch better then my english
16:30.23ManxPowerbeta3, *shrug*  It works for everyone else.
16:30.32ManxPowerbeta3, put your zapata.conf on pastebin.ca
16:31.09dfunnellFellow geeks - general question this time.  (Sorry about newbie question, but CAPI documentation is wafer thin).
16:31.11dfunnellI've got 4 x BRI in my * machine (and am using CAPI) and I want * to try dialling each available CAPI channel when dialling out (i.e. move from one to the next if the first one is busy).  Anyone know the correct syntax for doing so?  Can you group CAPI channels like you can group Zaps?
16:31.53ManxPowerdfunnell, I don't know for sure, but using group= like in zaptel is the best way, and I would ASSUME chan_capi would support group=
16:32.11BuckRogershas any one experimented with php and mysql for web configuation of user accounts
16:32.21ManxPowerdfunnell, You should ask on the mailing list if you don't find the answer here
16:32.56BuckRogerssuch as turning on and off features like anonomus call rejection
16:33.00BuckRogersfrom a website
16:33.17Trionnishow else would you do web configuration?
16:33.21Trionnis;)
16:33.31JerJer[mobile]vi index.html
16:33.36ManxPowerBuckRogers, it seems that most people that have done that consider it an advantage in competing and so don't release the scripts.
16:33.52BuckRogersTrionnis: i have no idea other then what i stated do you?
16:34.00dfunnellManxPower:  Thanks.  Do you mean CAPI mailing list?  Also having problem where, if a trunk is in use, incoming calls ring and ring, but * doesn't seem to recognise and answer ringing line.
16:34.08Trionnisthat was a smartass crack about your last line of text
16:34.15*** join/#asterisk Sedorox (brandon@Neptune.client.wlgrv.pa.sed6.net)
16:34.24TrionnisI really wouldn't know... I'm just here for lame comic relief
16:34.27BuckRogersright on bro
16:34.30ManxPowerdfunnell, Unfortunatly I've never used CAPI, so I can only give you very general advice about it.
16:35.06Trionnisalthough I agree with ManxPower... it's likely that you won't find one "prefabbed" really
16:35.30TrionnisI'm willing to help ya write one if you GPL it ;)
16:35.38BuckRogerswe are working on it now
16:35.55*** join/#asterisk algorithmn (~na@ool-18bce89c.dyn.optonline.net)
16:35.56beta3ManxPower: its too big
16:36.08BuckRogersi think we will keep it private also
16:36.15Trionnisof course you will
16:36.16BuckRogersand sell it to the highest bidder
16:36.17Nuggetewwww.  GPL is ucky.  :)
16:36.18BuckRogersj/k
16:36.25BuckRogersno one will pay for that
16:36.28ManxPowerThe way I PLAN on doing this when the time comes is use PHP to provide the interface, MySQL as the backend (since something more powerful is not needed) and then AGI in the dialplan to query the settings and set the channel variables, then let the dialplan do what needs to be done via the channel variables.  Who knows what I'll ACTUALLY do when the time comes.
16:36.32Trionnis;)
16:36.44JerJer[mobile]agi is not desired or needed
16:37.05JerJer[mobile]but you will figure this out once you write one
16:37.09NuggetThat sounds like a sane approach, ManxPower.
16:37.30Nugget(well, except for the mysql part -- naturally :)
16:37.36Trionnislol
16:37.40ManxPowerNugget, As JerJer points out AGI could be an issue, but since I'll never have more than 100 users, I dont think it will be an issue.
16:37.40BuckRogersjer jer what do u use instead of agi
16:37.41BuckRogers?
16:38.13ManxPowerNugget, It's a perfect app for MySQL.  i.e. lots of reads, not many writes.
16:38.18algorithmnwhats the deal my buck of rogers?
16:38.22NuggetI know, i was just trolling.
16:38.38BuckRogerscold chilling, holding it down my man
16:38.40*** part/#asterisk Inferna (~sasha@194.158.51.171)
16:38.51Nuggetalthough I'd always suggest postgresql over mysql I don't honestly mean to suggest that mysql is a /bad/ choice.  it's just not the best choice.
16:38.55algorithmnword... thanks for the wiskey
16:39.05BuckRogersno problem
16:39.16ManxPowerNugget, People forget "the right tool for the job" maxim.
16:39.16algorithmnit really put some punch in the mornin coffee
16:39.18eKo1How well do you think AGI scales when you have over 100 users?
16:39.24Nuggetmysql has a host of problems that come with it -- but anyone willing to deal with those shortcomings is clearly free to choose mysql.
16:39.31tzangerI'll come right out and say it's a bad choice
16:39.36*** join/#asterisk zapa (zapa@200.77.110.182)
16:39.36ManxPowereKo1, AGI has quite a bit of overhead to start up.
16:39.47tzangerif you only need somewhere to throw your data, use sqlite or db2/3/4 or even a flat file...
16:39.55BuckRogersnugget what type of problems that woudl affect asterisk
16:39.55BuckRogers?
16:40.02tzangeror if you use php, by all means use mysql :-)
16:40.03eKo1ManxPower: So building a module would be the 'faster' choice?
16:40.09ManxPowertzafrir, sqlite is also something I've considered.
16:40.13tzangerI really like sqlite
16:40.19tzangerit's small, portable and works well
16:40.20ManxPowereKo1, by orders of magnitude
16:40.27tzangerI just wish it had some network connectivity :-)
16:40.51NuggetBuckRogers: using mysql means never being certain that the data you're reading is what you wrote.
16:40.58Nuggetit's fast, though
16:41.05tzangerNugget: correct, but it's not THAT fast
16:41.11eKo1But making AGIs are orders of magnitude easier than modules.
16:41.14tzangercompare it to pg for realworld use, they're neck and neck
16:41.19tzafrirManxPower, it is even something I have considered. My latest * packages have sqlite support
16:41.24tzangerit's like saying your P4/3.02 is faster than your P4/3.00
16:41.31tzangersure it's faster but it's not noticeable
16:41.39BuckRogersNugget i dont understand?
16:41.46tzangerPostgres < 7.x did suck for speed
16:41.52tzangerbut 7.4.3 and the 8.x betas are insane
16:41.56BuckRogerswhat i read is not what i wrote? Mysql injection?
16:42.08BuckRogersthats a hacker tool right
16:42.15Nuggetmysql is designed to silently change the data you insert if it doesn't fit.
16:42.16tzangerBuckRogers: MySQL will take it upon itself to make your data fit, instead of tleling you the data won't fit
16:42.37tzangerit's one of *the* most *assinine* "features" for a DB I've ever ever run across
16:42.57BuckRogersso make sure your inserts are of a certian size?
16:43.04algorithmnbuckrogers:  i woulnd;t worry about injection.  i'll help u put in some detection into the php scripts w/admin e-mail notification
16:43.04tzangerBuckRogers: no it's worse than that
16:43.10tzangerBuckRogers: create a table with an integer column
16:43.17tzangernow insert "mary had a little lamb" into that column
16:43.22tzangermysql will take it and not even blink.
16:43.41algorithmndoes it drop it or convert ascii->int
16:43.41BuckRogersok thats good though right
16:43.45tzanger??!
16:43.50tzangerBuckRogers: step AWAY from the keyboard
16:43.50Nuggetno, that's terrible.
16:43.54*** join/#asterisk Damin_Mobile (~pocketirc@72.sub-70-214-30.myvzw.com)
16:44.05BuckRogersahh letters where numbers should be
16:44.07ManxPowertzanger, Some people would say that you need to validate your data before sending it to your database.
16:44.07BuckRogersgot cha
16:44.19*** join/#asterisk skrusty (muad@217.79.111.73)
16:44.24skrustyafternoon
16:44.27ManxPowerI don't disagree with that.
16:44.29BuckRogersgood point ManxPower
16:44.30NuggetManxPower: most people would agree that the database is the sanest place to do that validation.
16:44.33tzangerManxPower: fuck that -- the DB is there to store the data I give it.  If it can't, it should throw back an error, not silently mangle it
16:44.38algorithmnprogramming convention can help alliviate that,  just more time wated i guess
16:44.39Nuggetbut mysql makes such validation impossible.
16:44.39tzangerManxPower: besides
16:44.52tzangerManxPower: if the fucking db has CONSTRAINTS and it doesn't listen to them, why does it have them in the first place?!
16:44.56ManxPowerBut honestly, the choice of database can be changed pretty easily, especially if you do it before you start coding.
16:45.20tzangerManxPower: if you code correctly (SQL92/95 compliant queries) it sure helps
16:45.30ManxPowertzanger, The Sweeds have a twisted sense of humor?
16:45.33sivanawhat's a good ATA that has dual ethernet (computer, connection)?
16:45.35eKo1and/or use odbc
16:45.52BuckRogerssivana : grandstream.com
16:46.02BuckRogersnice equipment
16:46.06tzangereKo1: ODBC is *A* solution but it's not necessarily a good one :-)
16:46.15tzangerBuckRogers: CHEAP equipment, not necessarily good :-)
16:46.16*** join/#asterisk Makenshi (~makenshi@2001:630:1c0:2001:280:c8ff:fee2:921f)
16:46.25Nuggetimagine if asterisk just dialed a random number if you tried to call a number which was not specified in the dialplan.  would the solution be to just "always make sure you dial the right number?"
16:46.40sivanajust until the MixBox is done mfg :)
16:46.40BuckRogersreally i had better luck with them then sipura
16:46.55tzangersivana: shhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhaddap...  :-0
16:46.57tzangerer :-)
16:47.04BuckRogerssipura gsm does not work correctly i emailed support over 3 weeks ago with no respones
16:47.21Nuggetwould the solution involve modifying every possible soft and hard phone out there to validate that the dialed number was one that the dialplan covered?
16:47.21sivana<PROTECTED>
16:47.30tzangerNugget: asterisk isn't actually a good comparison...  I can say "pollywaddledoodle=yes" in iax.conf and * will happily ignore it
16:47.39sivanaAnyone used the SPA-2100?
16:47.42Trionnisanyone know if there's some docs floating around about preconfiguring xlite before installed?
16:47.53BuckRogersyes we have one
16:47.59TrionnisI'd like to set up a buddy of mine on my * server, but I'm *not* walking him through setting it up
16:48.00Nuggettrue enough.  I was more using the dialplan specifically as the comparison.
16:48.05sivanayou like?
16:48.05BuckRogersi would rater have the 2000
16:48.06tzanger~google preconfigure xlite asterisk
16:48.14tzanger... odd
16:48.17sivanaBuckRogers: oh... why?
16:48.20Trionnisalready did that :)
16:48.20*** join/#asterisk [Outcast] (~knoppix@h0006259a2649.ne.client2.attbi.com)
16:48.24tzangerNugget: :-)
16:48.41BuckRogersjust poor firm ware
16:49.08BuckRogersi think they try to promis more then they give
16:49.17*** part/#asterisk Smythe (~Smythe@spock.cbcag.edu)
16:51.00mesiI just cannot configure this stupid sipphone thingy. Everything's just like with sip to fwd, but still receiving calls from sipphone doesn't work.
16:51.54shido6mesi
16:51.57shido6dont freak out
16:51.59mesishido: HI!
16:52.04shido6whats up?
16:52.21mesishido: I try to receive calls form siphphone, but it wouldn't work.
16:52.37shido6ok
16:52.40shido6how are you trying?
16:52.42shido6show me what u got
16:52.47mesishido: though everything is fine, I am registered, the web page sipphone.com knows my ip address, I can make calls...
16:52.47shido6at pastebin.ca (sip.conf)
16:52.59shido6making calls is easy
16:53.06shido6receiving calls ...esp if ur nat'd can be a problem
16:53.09shido6but you can fix it
16:53.22mesiI don't do nat with fwd and still it works fine!
16:53.32shido6ok show me what u got
16:53.34Trionnis~google preconfigure xlite
16:54.02Trionniserf
16:54.06mesishido: never used pastebin, but I'll try...
16:54.42BuckRogersyou might as well program in pastel
16:54.45*** join/#asterisk crash3m (crash3m@crash3m.user)
16:54.55MikeJ[Jayden]~pastebin
16:54.56jboti guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
16:54.58ManxPowertzanger, Once I get the coppice bounty out of the way I may issue a bounty to generate an error/warning for invalid .conf entries.
16:55.12shido6hey ManxPower
16:55.19ManxPowerhello shaZwaz
16:55.19tzangerwhat's coppice's bounty
16:55.23ManxPower..er..hello shido6
16:55.26shido6someone asked me about AS5300s and ppp dialup with the Quad span cards, you ever set that up?
16:55.40*** part/#asterisk crash3m (crash3m@crash3m.user)
16:55.41ManxPowertzanger, The stuck channels problem that he fixed, but he's STILL not collected it.
16:55.42shido6i've set up hdlc and data/voice T's
16:55.44shido6but never ppp
16:55.48shaZwazhowdy ManxPower
16:55.55Trionnis~google preconfigure sip phone
16:56.04Trionniserf
16:56.06ManxPowershido6, no, since that requires an ISDN DATA connection and not a modem connection.
16:56.24mesishido: I have pasted my sip.conf there.
16:56.26tzangerManxPower: oh yeah uh... I fixed that... yeah...
16:56.35Trionnisok, enough google spamming.. guess I'll open a browser
16:56.37Trionnis:)
16:56.37mesishidO: into "mesi: sip.conf"
16:56.45ManxPowertzanger, Hmm?
16:56.59ManxPowertzanger, I thought you helped with the E&M/Wink problem?
16:57.17mesishido: My extensions.conf is quite huge!
16:57.19ManxPowerthe stuck channels problem was a different issue 8-)
16:57.58*** join/#asterisk marshall (~test@S0106000f66563988.wp.shawcable.net)
16:58.28Trionniseek!
16:58.38*** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com)
16:58.38Trionnisno suppression?
16:59.23shido6just show me the extension u have given the sip provider access to
16:59.24*** part/#asterisk oej (~oej@63.83.135.35)
16:59.45Darwin35they had surpression but the damage is done
16:59.52Darwin35it was in a back corner
16:59.55Trionnis:(
17:00.01tzangerManxPower: I'm joking, I'm just trying to collect his bounty
17:00.28Trionnissounds like it wasn't properly installed
17:00.49ManxPowertzanger, LOL!
17:00.49TrionnisI'd think that there should be 100% coverage of machine areas
17:01.04Trionnissorry to hear that man... that really sucks
17:01.16Darwin35its a old building refurbished
17:01.29Darwin35built in 18 96
17:01.37Trionniseek
17:01.38shido6mesi, whats the pastebin.ca url they gave you?
17:01.38Darwin35new wiring
17:01.44Darwin35new lights
17:01.50Trionnisdoesn't sound like the best place for a DC
17:01.58Darwin35but the server room was never properly setup
17:01.59*** join/#asterisk Blackvel (~blackvel@dsl-213-023-034-235.arcor-ip.net)
17:02.20Darwin35well now I get to have my way with the offic
17:02.27Darwin35eand redo the server room
17:02.31Trionnisany clues as to what started it?
17:02.47Darwin35not yet they wont let us in the area yet
17:02.51inspiredwhat the hell? suddenly one of my peers have stopped appearing in "iax2 show peers", however it is still in iax.conf. nothing was changed in any config file. does anyone know what's wrong? the system is in production
17:03.06Darwin35I have a feeling a non grounded power strip
17:03.10tzangerinspired: interesting
17:03.14mesishido:  http://pastebin.ca/6965 is my extensions.conf
17:03.20tzangerpastebin their entry and mangle hte password/username
17:03.28mesi<PROTECTED>
17:03.29Trionnisnon.... grounded?
17:03.34Trionnisin a server room??
17:03.35inspiredok tzanger
17:03.56Trionniswow
17:03.58Darwin35there were to stripps missing the grounding pins
17:04.00Trionnisthat's.....
17:04.10Trionnisincredibly farking stupid
17:04.11Blackvelso who did fix that broadvoice problem for incoming voicemail instead of asterisk call yet?
17:04.12Darwin35I had put dont use on them but have a feeling they got used
17:04.13Trionnis=)
17:04.16*** join/#asterisk Tarox (someone@pD9E79ED7.dip.t-dialin.net)
17:04.24inspiredhttp://pastebin.ca/6966
17:04.30inspiredcheck it tzanger
17:04.36Trionnisshoulda pitched 'em
17:04.50Darwin351 is the broadvoice issue
17:05.00tzangerinspired: and sip show peers doesn't show briiz?
17:05.03Darwin352 is the mpg123 issues with *
17:05.06Trionnisthere's a broadvoice issue?
17:05.17Sedoroxwhats wrong with mpg123?
17:05.20Blackveldarkskiez: i am tspeaking to someone
17:05.24TrionnisI had some screwy stuff last night, but I've not noticed anything so far today
17:05.24ManxPowerDarwin35, A fire usually gets management to listen to MIS 8-)
17:05.25Darwin35mpg123 sucks
17:05.28inspiredno, it's not a sip peer
17:05.28Sedoroxlol
17:05.31Blackveland he says, your 1.0.5 fix does not fix that problems :)
17:05.38Trionnishahaah, very true ManxPower
17:05.58*** join/#asterisk Juggie (agony@CPE00c049d9f271-CM014270110981.cpe.net.cable.rogers.com)
17:06.02Darwin35I am looking into the 1.0.6 issue with broadvoice and *
17:06.09BlackvelDarwin35 i mean
17:06.13Trionnisahh, I haven't upgraded yet
17:06.15Darwin35give me time to figure it out
17:06.17Blackvelbut he now got 1.0.5 :)
17:06.19Trionnisseems like it's good I didn't
17:06.21Trionnis;)
17:06.37*** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au)
17:06.39inspiredtzanger: this is scary. users are trying to call out but our termination partner is gone from iax2 show peers
17:06.44Blackvelseems to work from time to time, but not always, so broadvoice sends callers instead to asterisk to BV voicemail :)
17:06.48tzangeruh
17:06.53Darwin35I have to put everything on the back burner and go to a meeting about this fire
17:06.55tzangerinspired: you showed me a sip peer did you not?
17:06.58Darwin35I will return
17:07.03Trionnisok
17:07.10Trionnisgood luck Darwin
17:07.17Blackveli am not sure if I could understand your issues very well darwin and if was the problem you where working on
17:07.18Trionnishopefully they'll listen
17:07.19Blackvelfire?
17:07.19ManxPowerDarwin35, TURN OFF THE BURNER FIRST! 8-)
17:07.20Blackveluff
17:07.25Darwin35hahaha
17:07.27Trionnisoooh
17:07.30Trionnispwn!
17:07.31Trionnislol
17:07.47inspiredtzanger: no, it's an iax2 peer
17:07.57tzangerinspired: ahh
17:08.02inspiredand it's been working fine for a long time
17:08.07tzangerodd
17:08.08Blackvelwhat?
17:08.08inspiredbut now it disappeared
17:08.20tzangerinspired: there's not a space in front of it or anything
17:08.24Trionnisuh, stupid question, but have you made sure the service didn't go down?
17:08.24*** join/#asterisk Dibbler (~Dibbler@snaddy.plus.com)
17:08.38Trionnissometimes it's the simple things ;)
17:09.08inspiredtzanger: in front of it?
17:09.08*** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com)
17:09.08Darwin35Manx take over for me for awhile
17:09.08Darwin35back in 30   min
17:09.08inspiredno, everything is aligned to the left
17:09.09tzangeri..e  "[briiz]" not " [briiz]"
17:09.28tzangerinspired: just for shits and giggles, add a "qualify=500" to it and reload
17:09.32*** join/#asterisk sob0l (~peter@uo166.internetdsl.tpnet.pl)
17:09.52*** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com)
17:10.10inspiredok
17:11.50inspiredand another thing. asterisk has started having SHITTY responsiveness when it comes to issuing commands on the CLI. a simple "reload" takes 15-30 seconds before anything happens
17:11.57inspireda stop gracefully has the same problem
17:12.26shaZwazjust intalled speex ..looks fine thou
17:13.19*** join/#asterisk alerque (~alerque@onyx.ouraynet.com)
17:13.20*** part/#asterisk alerque (~alerque@onyx.ouraynet.com)
17:13.21jontowinspired; look at 'top' .. is mpg123 going nuts?
17:13.26*** join/#asterisk alerque (~alerque@onyx.ouraynet.com)
17:13.29inspiredmpg123 is not even installed
17:13.34inspiredload 0.02
17:14.16*** join/#asterisk blitzrage (~blitzrage@blitzrage.documenter.extraordinaire.of.asterisk)
17:14.19blitzrageok... fuck this guy
17:14.21blitzragehttp://techdatapros.com/asterisk/
17:14.46Zeeekhey Blitz
17:14.48inspiredjust restarted asterisk. takes fucking one minute before it even loads any modules
17:15.12blitzrage"Digium X100P FXO card which can be purchased off eBay for $6.95"
17:15.23blitzrageobviously a clone, and we'll probably have a million newbs asking how to get it to work
17:15.31JerJer[mobile]Caveat Emptor
17:15.46blitzrageanyways... thats my rant, I'm going back to the productive IRC channels
17:15.54blitzrageZeeek: btw - yo!
17:15.54*** part/#asterisk blitzrage (~blitzrage@blitzrage.documenter.extraordinaire.of.asterisk)
17:15.56Nuggetheh
17:15.56Trionniswould help if they weren't advertised that way
17:16.43Trionniscould just be a case of simple ignorance :)
17:20.08inspiredtzanger: qualify=500 didn't help at all
17:22.26inspiredgoing to test it
17:22.37inspireduhm, just received a fresh iax2 phone.
17:22.42inspiredwill make a review
17:22.51inspiredat least it looks good :D
17:23.11Sedoroxis it in production?
17:23.35*** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc)
17:23.35inspiredyep, it's an ATCOM something
17:23.39inspiredgot it from iaxtalk.com
17:23.57inspiredthe lcd is even flippable!
17:23.58inspired:D
17:24.03Sedoroxhmmm
17:24.15Zeeekinspired how you like phone ?
17:24.42inspiredhaven't tried it yet.
17:24.50Sedoroxhttp://www.iaxtalk.com/product_info.php?cPath=1&products_id=36&osCsid=d9391e67327e4075a742ca43ea1f92db
17:24.51Sedorox??
17:24.54shmaltz<PROTECTED>
17:25.03inspiredATCOM
17:25.15inspiredwill volume import these to EU if it works good
17:25.33inspiredit certainly doesn't look like a grandstream
17:25.35Blackvelwho got iaxtel?
17:25.36inspiredand it's even cheaper!
17:25.45Blackveland is open minded for a call test?
17:26.01inspiredSedorox: yep, except I have the two port one
17:26.10Sedoroxinspired: was gonna say that.. cheaper then a grandstream.. and better looking
17:26.11Sedoroxah ok
17:26.12Zeeekinspired $30-$50 shipping
17:26.19Zeeekadd ^^^^^
17:26.21*** part/#asterisk alerque (~alerque@onyx.ouraynet.com)
17:26.21Zeeekfrom China
17:26.28*** join/#asterisk jalsot_ (~tamas@abacus.eworldcom.hu)
17:26.33inspiredstill cheaper than buying a grandstream in EU
17:26.36*** join/#asterisk harryvv (~plato@S010600055d210201.vs.shawcable.net)
17:26.47Zeeekbuying anything here is way more expensive
17:26.47inspiredplus if I import many of them, I will have cheaper shipping
17:26.52Zeeekyes
17:27.00Zeeekand you could get a lot of us to group an order
17:27.10harryvvzeek you at von now
17:27.16inspiredyep
17:27.17Zeeekexcept I just  ordered one from the States
17:27.22*** join/#asterisk X-Gen (~x-gen@rrba-146-120-223.telkomadsl.co.za)
17:27.25BuckRogerswhat is the codec support for the iaxtel?
17:27.27inspiredthe same model?
17:27.32Zeeekno I'm at Von's doing the grocery shopping
17:28.09harryvvzeeek :) how many voip products do you have in your grocery cart now?
17:28.09SedoroxI wanna know how those ATA's from that company work...
17:28.42BuckRogersAT-320EE VoIP phone, what codecs does it support?
17:28.57inspiredBuckRogers: Audio codec G.711,G.723,G.729,GSM
17:29.10inspiredSedorox: I also bought the ATA. will test that too
17:29.12BuckRogersInspired thats soilid
17:29.24BuckRogersI luv my gsm
17:29.31inspiredyep, gsm rocks
17:29.39BuckRogersnice and tight and compact
17:29.46BuckRogerssounds great
17:29.52BuckRogerslittle overhead
17:29.52Sedoroxinspired: please let me know about that!!!
17:29.58inspiredfucking shit. I can't add new users to iax.conf
17:30.05Sedorox?
17:30.07inspiredwhat the fuck happened?!
17:30.19inspiredone peer disappeared and I can't even add new ones
17:30.36inspiredexit
17:30.41inspiredoops. hehe
17:31.01harryvvjust a little oversight
17:33.05jontowwoohoo.. getting my pair of T100P cards back today :P
17:33.11*** join/#asterisk j0 (dan@S010600095b00a5a9.vc.shawcable.net)
17:36.36*** join/#asterisk Gh0sty (~Ghosty@ip-81-11-221-15.dsl.scarlet.be)
17:37.08*** join/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com)
17:38.23inspirednow that kicks ASS. the phone has a 100 mbit switch
17:38.26inspiredand is 69$
17:38.32inspiredwill upgrade to iax firmware now
17:38.43Sedoroxlol
17:38.51Sedoroxoh.. doesn't come with the iax firmware, huh?
17:39.13inspiredno, sadly :(
17:39.21Sedoroxbut they include it on a CD or something?
17:39.57inspiredon the web
17:40.43Sedoroxhmmm
17:40.43Sedoroxok
17:41.21tzangerinspired: odd
17:41.29tzangerasterisk is otherwise working fine??
17:41.49Sedoroxbbiab
17:42.02*** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net)
17:43.25*** join/#asterisk Damin_Mobile (~pocketirc@90.sub-70-214-0.myvzw.com)
17:44.45dontmsgmenn
17:44.52opus___hello
17:44.57opus___is there a way to encrypt sip calls
17:44.57dontmsgmeHi
17:45.18opus___hello dont
17:46.02*** join/#asterisk Conductor (~thomas@62.8.240.132)
17:46.21Conductorhi all. i have some problems with the webinterface of meetme2...
17:46.41Darwin35ok I have a neww as*hole and so does the whole office
17:46.44Conductoreverything is ok, but when i click on Listen or Kick nothing seems to happen
17:46.54*** join/#asterisk beta3 (~dan@dan2.active.supporter.pdpc)
17:47.06Trionnisdamn Darwin35... :(
17:47.07beta3I'm having major echo issues on calls coming from the fxo
17:47.08opus___where is the web interface code for meetme2 , curious
17:47.11beta3any suggestions
17:47.30Trionnisisn't it great when people choose to piss and moan instead of addressing the issue?
17:47.43Trionnis(yes, that's sarcasm)
17:47.46Conductoropus___, http://www.areski.net/asterisk-meetme/about.php
17:47.49opus___beta3 have you contacted the manufacture, is the cable to long?
17:48.01beta3opus___: this is a digium card
17:48.07opus___conductor - thanks
17:48.07Darwin35yes
17:48.17Darwin35so now they gave me my own project
17:48.21beta3opus___: I hooked it up at the dmark of the house, so its shorter than the phone system originally for the home
17:48.28Trionnishopefully things will come together properly now?
17:48.35Darwin35my 17 boxes are to be moved to my office
17:48.40Damin_MobileDarwin: What happened?
17:48.50Darwin35we had a small fire
17:48.58Trionnisah
17:48.58Darwin35lost 11 servers
17:49.02Trionnisso you're getting space heaters
17:49.04Conductorhasn't anyone ever tried meetme2?
17:49.06Trionnisthat's nice of them
17:49.26Darwin35most where old old 21264 dec alpha servers
17:49.39opus___condcutor - i'll check it out later tonight if you see me again i'll let you know if i had the same problem.. what was the problem you are having again?
17:49.40Trionniswow... people still use those?
17:49.43Trionnis;)
17:49.53Damin_MobileInsurance shoulld help
17:50.13beta3does anybody know how to get rid of echo on the fxo interfaces
17:50.26Conductoropus___,  everything is ok, but when i click on Listen or Kick nothing seems to happen
17:50.40Conductoropus___, you need to apply a patch to the source code..
17:51.11Damin_Mobilebeta3: Enable echo cancellation. look at the wiki.
17:51.39opus___hmmm
17:51.43beta3Damin_Mobile: thats already done
17:51.48Darwin35yeah insurance will cover
17:51.55BuckRogersConductor that website looks nice very promising how does it work, good so/so bad?
17:52.14harryvvdarwin was it the ps in one of the servers that caught fire
17:52.16*** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com)
17:52.27Damin_MobileDarwin; prolly get a decent payout.
17:53.41Darwin35I wish I was getting money out of it
17:53.45Trionnisyeah, make sure you point out how hard it is to replace those "highly specialized" servers
17:53.48Trionnis;)
17:54.14harryvvDo most standard PBX units have a onboard ups mounted in the same plastic enclosure?
17:54.37epochharryvv: some of them have batteries yeah
17:54.39Darwin35but now my project will get a boost
17:54.42harryvvWas looking for a nortel networks unit and wondered because of its large size
17:55.03Trionnishi harry :)
17:55.09Trionnishow ya doin?
17:55.12Damin_MobileNew servers!
17:55.24harryvvepoch, whats the typical Ah rating of those batteries?
17:55.34TrionnisDamin_Mobile: that's like xmas to an IT geek ;)
17:55.41harryvvhi Trionnis
17:55.41Darwin35I just got new servers for this project of setting up asterisk pbx systems
17:55.46epochharryvv: hell if I know :)
17:56.23Darwin35I need someone to go over my new extensions.conf file
17:56.23ConductorBuckRogers, doesnt work very well. must apply patch first, cant use postgres and even then it doesnt work correctly
17:56.24harryvvI guess the best way is to ask a ex pbx technican here.
17:56.30*** join/#asterisk Tarox (someone@pD9E7BAF5.dip.t-dialin.net)
17:56.33Conductorcu
17:56.36Conductor\quit
17:57.11Trionnisharryvv: some do, some don't
17:57.38harryvvTrionnis, is that at the request of the customer ?
17:57.43TrionnisI ran an NEC ivs-2000 a while back that used a golf cart battery
17:57.47Trionnisseparate
17:57.51harryvvk
17:57.59Trionnisbut I've also seen panasonics with small ones built in
17:58.00*** join/#asterisk SexyKen (~sexyken@c-67-161-5-149.client.comcast.net)
17:58.12SexyKenHey guys -- does Asterisk support Extension Mobility?
17:58.27TrionnisI think some of the nortels, et al have space for one, but you have to get the "kit" to use it
17:58.29Darwin35I need someone to go over my new extensions.conf file and tell me what is wrong
17:58.31harryvvI would feel more comfortable selling a unit with a ups onboard.
17:58.40Trionnislikely cheaper and easier to just get a second ups
17:58.43Darwin35I think for the most part its fine
17:58.44Trionniser
17:58.48Trionnisaddon ups, that is
17:59.03harryvvdarwin whats the problem
17:59.16*** join/#asterisk cool4ever2 (~craeck@mail.innovate-it.ch)
17:59.18Darwin35not all the functions seem to get called
17:59.31harryvvare thay all using sip
17:59.32Darwin35so I just went over it
17:59.37Darwin35yes
17:59.42ManxPower~docs
17:59.43jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
17:59.46harryvvturn on sip debug
17:59.55trimi`any1 have used calling card application in asterist? if yes which is the best? any help ?
17:59.56harryvvif it comes to that
17:59.57harryvv:)
18:00.01Darwin35i have its not showing enough info
18:00.28harryvvwell then it sounds like its a misconfig or not enough configuration information.
18:00.52trimi`any1 have used calling card application in asterist? if yes which is the best? any help ?
18:00.54Darwin35thats why I want some one to read the file and see if they see what I am missing
18:01.01Darwin35yes
18:01.04Darwin35BKW does
18:01.15TrionnisDarwin35: stick it on pastebin.ca
18:01.28Darwin35ok
18:01.29Trionnismany eyes will catch things quicker ;)
18:01.42SexyKenDoes anyone here know what Extension Mobility is?
18:01.58shaZwazok guys see u later
18:02.13harryvvnever heard of it unless its just a extension thats directed to a cell phone number which I have done.
18:02.32SexyKenhttp://www.voip-info.org/tiki-index.php?page=PBX%20Extension%20Mobility
18:02.36SexyKenI just can't find any other info on it.
18:02.55trimi`any1 have used calling card application in asterist? if yes which is the best? any help ?
18:03.56harryvvsexy, just sign up for a voip service and make a extention that logs into that service and configure the line to dial your cell phone.
18:04.19SexyKen•harryvv• You're not at all close to what I need done.
18:04.32SexyKenharry - Apparently you use Asterisk on a personal level.
18:04.33Darwin35http://pastebin.ca/6970
18:04.37harryvvokay then your needs are different
18:05.39Darwin35I know there ar emore functions out there
18:05.42Zeeekcan you transfer an incoming call to meetme and hangup?
18:05.47Darwin35but this is what I have
18:05.56Darwin35yes
18:06.02Trionnislooking now Darwin35
18:06.44machinehdWith a 7960 can you set it up so it can see if other extensions are currently in use? Or are the 6 lines completely dedicated to that phone?
18:07.09ManxPowermachinehd, I believe the Cisco SIP does not support shared call appearances.
18:07.17ManxPowerUse Flash Operator Panel if you need that.
18:07.52*** part/#asterisk Bentley (~rbc@S01060080c8135e6a.cg.shawcable.net)
18:07.55*** join/#asterisk denon (denon@66.207.128.103)
18:07.55*** mode/#asterisk [+o denon] by ChanServ
18:08.20machinehdManxPower, thanks, I've been using FOP. Do you know of any phone that can do that? Secretaries seem to like the phones where they can see what lines are in use
18:08.26trimi`any1 have used calling card application in asterisk? if yes which is the best? any help ?
18:08.30ZeeekIf I tranfer to meetme and hang up, both channels are hung up
18:08.48ManxPowermachinehd, see the mailing list archives.
18:08.51ManxPower~mailinglist
18:08.52jbotrumour has it, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.
18:09.01Trionnisya know trimi`.... if you keep that up, you're gonna get people honked off at you
18:09.23ManxPowermachinehd, At least the SNOMs, maybe others support it.  Asterisk's shared line appearance is not all that good and there are like 4 standards for doing that feature.
18:09.48Darwin35SNOM can lick my crack those thieves
18:10.02Darwin35sorry sore spot
18:10.21ManxPowerTrionnis, Yes they will.  Personally I don't usually answer questons from people using "words" like "any1", "r", "u", etc.
18:10.44Trionnisheh
18:10.51modulus_manx, y not?
18:11.03*** join/#asterisk neko2 (~neko@212.200.132.7)
18:11.39Darwin35Tri the biggest part in that is the calling card function
18:12.21Darwin35the only part any one gave me
18:12.24Trionnisok, I just got to that part
18:12.28Trionnislemme look
18:12.35Trionnisso far it's nicely done
18:12.38Trionniscommented well
18:12.39Darwin35I wrote 98 % my self
18:12.39Trionnis;)
18:12.53Darwin35I am a stickler for that
18:13.07Trionnisgood habit to have :)
18:13.25ManxPowermodulus_, Because if they are too damn lazy to type real words, then I'm too damn lazy to help them.
18:13.34modulus_manx, i c
18:13.40Darwin35when this is done I am going to post it in the wiki pages as a very full robust extensions file
18:13.45Darwin35maybe
18:14.06Trionnishahah
18:14.14Trionnisnicely played modulus_
18:14.15Trionnis;)
18:14.42*** join/#asterisk iceyp (~icepick@max.unix.co.nz)
18:14.52iceypanyone from nufone.net here?
18:15.06iceypshido6 ?
18:15.38Darwin35rollover I have not yet tested I wrote that today
18:16.04*** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no)
18:17.10opus___is there a reason I see a million " chan_sip.c:901 __sip_ack: Stopping retransmission on '5f614b1a1193a64e2a41f50c0acfc4b6@192.168.2.168' of Request 102: Found
18:17.15Trionnisso far that calling care stuff looks decent
18:17.22opus___lines when my 3+ SJphone implementation?
18:17.33opus___is that a bad message
18:17.34TrionnisI wouldn't sign off on it without testing of course, but it *looks* right
18:18.01ManxPoweropus___, stop running in debug mode.
18:18.01Darwin35brb hold a min
18:18.06*** join/#asterisk boch (~as24@200.59.172.98)
18:18.29Trionnisk
18:18.34shido6aroo?
18:18.37iceypopus___ I get them for missing UDP packets all the time
18:18.37shido6whats up?
18:18.51iceypor invalid DTMF tone
18:18.53iceypshido6 umm
18:19.10Darwin35back
18:19.13iceypany reason I can dial this number +448449865089 from nufone?
18:19.19Darwin35Tri PVt me if you have time
18:19.44ManxPowericeyp, 011448449865089
18:20.00ManxPowerAnd you have to make sure you are authorized by Nufone to make international calls
18:20.02iceypyeah, i have a setting which makes me use 00 for intl
18:20.14iceypallows me to make calls to other countrys
18:20.25ManxPowericeyp, no, in the USA you need 011 as the international prefix
18:20.27iceypjust having problems with a few ranges
18:20.44iceypManxPower i have a dialplan
18:20.45iceypmkay
18:20.51ManxPowericeyp, Are you sure they are not Premium numbers?
18:21.04iceypwhat u mean
18:21.09iceypsome countries can dial them
18:21.15tzanger"premium numbers" ??
18:21.23RoyKpremium arsehole?
18:21.33ManxPowertzanger, like 900 and 976 numbers in the USa.
18:21.36tzangerPremium Poontang
18:21.41develis the sqlite cdr support "complete"?
18:21.41RoyKpremium wannabee :)
18:21.43tzangerManxPower: ahh okay
18:21.46Trionnisonly $4.95 a minute!
18:21.48Zeeekthe 900 PayManx number?
18:21.49Trionnis:)
18:21.54iceypnah they not
18:22.01Zeeekhey that's it - I got it now
18:22.03iceyp3-5c per min
18:22.07iceypcant member it
18:22.09Zeeeksupport DID
18:22.10*** join/#asterisk Bentley (~rbc@S01060080c8135e6a.cg.shawcable.net)
18:22.11iceyp3-5p even
18:22.17*** join/#asterisk RaYmAn-Bx (user@213.237.12.147.adsl.vby.tiscali.dk)
18:22.19iceypits like the 0870 number
18:22.25Trionniswow iceyp... that went *way* over your head, didn't it?
18:22.30Trionnis=)
18:22.47Zeeekshido6 in fact I had a problem today dialing an 800 number
18:23.08Zeeekit went through with other provierds but not nf
18:23.56shido6u have a callerid set you're sending?
18:23.57Zeeekanyone use nufone that can try this 800?
18:23.57tzangerZeeek: what was your outgoing callerid set to
18:23.57shido6err
18:24.01shido6u have to send a callerid
18:24.08tzangerZeeek: give me the 800#
18:24.09*** join/#asterisk Goshen (~Goshen@70-57-80-147.slkc.qwest.net)
18:24.19ZeeekI went thriough this last time with JerJer: callerid is 866nnnnn
18:24.37tzangerZeeek: some 800#s will NOT accept calls from 800#s
18:24.41ZeeekWells Fargo bank: (auto ans) 800 742 4932
18:25.03Darwin35ok time to setup a 976 sex line with asterisk
18:25.07ZeeekI seem to rememnber that it worked at one time
18:25.08Goshenthats odd...I am trying to call this number 18887713493, when it goes over FWD(IAX), command line shows it connecting, but I don't hear anything, when I change it to dial out over ENUM it goes over a sip server
18:25.10tzangerI found this out -- it's not a nufone problem, it's the far-end problem, if I set my outoging callerid to my 800# I couldn't get through, but dialing with a different (non-tf) 800# went fine
18:25.13Darwin35setup a bunch of conf rooms
18:25.16Goshenand its the same problem, I don't hear any audio
18:25.25Goshenother toll free numbers have been working fine today
18:25.42Zeeektzanger did you try it thru nufone?
18:25.42SexyKenDoes asterisk support fucking extension mobility or extension roaming
18:25.52tzangerGoshen: FWD's a free service... make sure you're getting your money's worth
18:25.54Zeeekfucking extensions are banned
18:25.58tzangerZeeek: hold
18:25.59Blackvelwho uses iaxtel here?
18:26.10SexyKenI use iaxtel
18:26.12stevekstevekfucking extensions -- setting up a 900# service?
18:26.20Zeeekeggs actly
18:26.22SexyKenoh wait
18:26.23SexyKenno i dont
18:26.25SexyKenI use tel iax
18:26.38Goshentzanger: the point is that it isn't working with two providers...
18:26.43Goshenguess I will try nufone now
18:26.47SexyKenNO!
18:26.49SexyKenDont use nufone
18:26.51SexyKenThey suck.
18:27.03*** join/#asterisk harryvv (~plato@S010600055d210201.vs.shawcable.net)
18:27.08mishehusucking can be good.  depends on in what context
18:27.15GoshenSexy: keep your sucking and fu***ing to yourself ;)
18:27.23SexyKenBend over bitch.
18:27.27SexyKenI'll show you what it's all about.
18:27.31Zeeektzanger it just kept ringing for me
18:27.33TrionnisI'd imagine the "fucking extensions" might have a counterpart that would involve sucking
18:27.36Darwin35ok  back to topic
18:27.45tzangerZeeek: works just fine
18:27.46Darwin35take the sextalk to #moosepenis
18:27.49jontowneat.. i have 5 * PBX's connected via IAX2 to a 'central switch' running freebsd/*
18:27.52Zeeekactually it makes sense that a bank would be leery of 866 numbers
18:27.56jontowthe 'switch' is routing calls between the PBXs :)
18:27.57Zeeekoh?
18:28.01Zeeeklet me try again...
18:28.05tzangerIAX2/myusername@nufone-1/18007424932||g
18:28.22tzangerfunky guitar music and slow sexy voice "welcome to wells fargo"
18:28.54SexyKen•jontow• I'm interested in talking to you regarding some developmen.t
18:29.15opus___there was a number that if you called on a mpx200 smartphone it would emit some tone that caused the phone not able to input dtmf
18:29.15Juggieare you paying? :P
18:29.23Zeeektzanger: calling @NuFone/18007424932
18:29.36Zeeekmy CID 866nnnnnnn
18:29.38tzangerZeeek: I call the same number with a SetCIDNum(8005062688) in front of it and I get congestion
18:30.16Zeeekso 866 would give that too?
18:30.19tzangerZeeek: as I said, it depends entirely on what the far-end 800# is set to accept
18:30.22ZeeekI don't get congestion, it just rings and rings
18:30.29*** join/#asterisk viLeR (1000@ip-33-7.telesat.com.co)
18:30.41Zeeekdrops thru to voipjet and connects right away
18:30.48tzangerZeeek: change your callerid and see what it does
18:30.55Zeeekk
18:31.02*** join/#asterisk mixi (~mixi@pD9EE1CBE.dip.t-dialin.net)
18:32.50Zeeektzanger - that's hilarious - ok, understood
18:33.07tzangerZeeek: I ran into this with a customer
18:33.08tzangersame issue
18:33.18tzangersolution was to set the outgoing callerdi to a non-800# when dialing 800#s
18:36.06ZeeekI thought that was impossible since
18:36.15Trionnis~amp
18:36.16jbot[amp] an Audio MPEG Player.  [non-free]
18:36.19Trionniserf
18:36.22Trionnis~AMP
18:36.22jbot[amp] an Audio MPEG Player.  [non-free]
18:36.26Trionnisgrrr
18:36.43ManxPowerTrionnis, Do it again!  Maybe jbot will change it's answer!
18:36.43Trionnis~google asterisk management portal
18:36.53Trionnislol
18:36.54Trionnis:P
18:37.42*** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com)
18:38.03Zeeek~seen Sideways
18:38.04jbotZeeek: i haven't seen 'sideways'
18:38.04GoshenWhat is this?
18:38.05GoshenMar  7 11:33:13 WARNING[21547]: chan_zap.c:5653 handle_init_event: Detected alar
18:38.05Goshenm on channel 1: Red Alarm
18:38.14Zeeekyou should, it won all the oscars
18:38.32ManxPowerGoshen, that means the telco line went away
18:38.49Goshenodd
18:38.57GoshenManxPower: thanks
18:39.40Trionnisthe POTS line died?? but according to that commercial I saw from SBC, that doesn't happen!!
18:39.47Trionnis./sarcasm
18:41.10*** join/#asterisk Defraz (~t0tal@sonicwall.dcdi.net)
18:46.46thepdakidSexyKen, please explain what you mean by extension mobility.
18:46.49Goshenhmm, perhaps my generic x100p is acting up
18:47.27Zeeekgive Digium support a call
18:47.38Zeeeknot
18:47.41Goshenlol, not
18:49.42opus___hey
18:51.27Zeeekso with all theis testing of free providers maybe someone has come up with a way to route a call from FWD to sipgate to gossiptel to CallUk to sipphone to mytcom.it to like2phone and back?
18:51.48Zeeeklag about 1800ms
18:52.13jontowthats a lot of call processign.. :)
18:52.17Trionniswow
18:52.31Trionnisreminds me of my ... er "a friend's" blueboxing days
18:54.06modulus_k3wl! 4n 31337 ph0n3 h4x0r!
18:54.15Zeeekso getting back to my meetme transfer problem: I'd like to be able to start a conference from a customer I call, send him to meetme and then go there my self on a lowly Grandstream. Is that possible (GS doesn't do actualtransfers)
18:54.15Trionnislaf
18:54.18Trionnis;)
18:54.27modulus_:P
18:54.35*** join/#asterisk spackle (~spackle@209.234.83.19)
18:54.38Trionnisit was fun to call through all that, and hear like 3-5 seconds of lag in the audio
18:55.04ZeeekI got that on FWD in the old days
18:55.14ZeeekIt was like being on the moon
18:55.22*** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net)
18:55.37Trionnisyup
18:56.02spackleTrionnis: just joined, & I'm curous about what you were calling though.
18:56.22spackleer, through.
18:56.47spacklePlaying with tin cans and string again?
18:57.02Zeeekold POTS blueboxing
18:57.06*** join/#asterisk iceyp (~icepick@firewall.unix.co.nz)
18:57.22iceyphey guys, how do you run asterisk in the most possible debug mode... asterisk -vvvvvvc ?
18:57.45iceypI want more than that, i.e. a call that hits the pabx but doesnt have right context, or extension dont exist etc
18:57.56Trionnisadd a few more v's
18:58.20Ashiceyp: just enable debugging from the cli
18:58.25iceypwhen someone calls my UK number, i was getting a 'sip-inbound' doesn't exist in my logs, so i added the context and the extension, but now i get no logs
18:58.36Ashdoes asterisk have a way of piping out debug output elsewhere?
18:58.46iceypso i dont know if i ment to do 004487 or 4487 or 87.. as the extension
18:58.51Ashother than, say, running it behind 'script'
18:58.59*** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no)
19:00.35opus___whoah i've got 1008ms lag to broadvoice
19:00.39Delvarit can spit out most info to log files...
19:02.32opus___why is asterisk continously 'auto destroying call' and 'stopping retransmission' for sip clients
19:03.25modulus_for every sprinkle i find, i shall kill you. - stewie
19:03.42harryvvopus, does it destroy the call when you hang up
19:03.54Zeeekhow can I put a caller into a conference room?
19:03.57Blackvelopus___: brvoice?
19:04.00Blackveldoes it work again?
19:04.06modulus_broadvoice sucks
19:04.09modulus_broadvoice sucks donkey balls
19:04.10Blackvelwhy
19:04.12opus___harryvv - good question
19:04.13Blackveltell me
19:04.17harryvv:)
19:04.17opus___modulus - who do you recommend
19:04.32modulus_i recommend getting your own pstn<-> voip gateway
19:04.44opus___Uhh, I think I'll stick with $7 dollars a month
19:04.53modulus_bastard
19:04.54Blackvelopus: $7? where?
19:05.00Trionnishahaha
19:05.04Trionnis7?
19:05.07modulus_7 bob
19:05.07opus___broadvoice.com
19:05.10Blackvelmodulus_: you didnt answer the question :)
19:05.14modulus_i sure did
19:05.19Blackvelit sucks
19:05.20Blackvelbut why?
19:05.22modulus_[07-Mar:11:04 modulus_] i recommend getting your own pstn<-> voip gateway
19:05.27Trionnisyou can get a level3 DID from sipmedia with unlim incoming and 60 outbound for 4.95
19:05.31Blackvel(not for business use)
19:05.31modulus_oh b/c _ALL_ voip providers suck
19:05.40Blackvelhmm
19:05.41Blackveli see
19:05.47Blackvelyou are against everything
19:05.47modulus_trionnis, how's sipmedia?
19:05.47Blackveleheh
19:05.53Trionnisso far no complaints
19:05.55opus___trionnis -- how is there uptime / latency
19:05.57Trionnisno outages
19:06.01modulus_quality?
19:06.08Blackvelopus: do you have BV working right now with *?
19:06.11opus___broadvoice seems to go down all the time although my configuration is kinda shaky
19:06.11Trionnislatency is low, but I'm coming from a coloed box in a Savvis DC
19:06.15opus___blackvel yes
19:06.20Trionnisquality is fine
19:06.26Trionnisulaw only tho :-/
19:06.28modulus_trionnis, do they support iax?
19:06.30modulus_ulaw is fine
19:06.31Trionnisnot yet
19:06.34modulus_b/w is not an issue for me
19:06.39Blackvelopus___: let me call you on your BV pstn
19:06.47TrionnisI talked to their engineer, and he says it's coming in the next month or so
19:06.50Blackveland I want to see if BV sends me to your voicemail
19:06.57Trionnisthat seems to be the usual BS line, so I'm leery
19:07.05opus___blackvel - no it goes to my asterisk voice tree right now
19:07.16Blackvelcan you show me?
19:07.27harryvvWhat would cause this when starting *  Unable to open '/dev/zap/channel': No such file or dir.
19:07.27Blackvela well known contact of me has a big problem with it rightnow
19:07.41BlackvelBV doesnt work as expected, does silly stuff like voicemail instead of ringing asterisk
19:07.47Blackvelhow come it does work for you?
19:07.51Trionnisharryvv: "modprobe zaptel"
19:07.55modulus_trionnis, hah!
19:08.03harryvvtri, all the modules are installed.
19:08.11modulus_it's _always_ "coming soon"
19:08.18Trionnisyep
19:08.24opus___shit this irc script sucks
19:08.39Trionnisalthough I have to admit they've been good about the other stuff they've added
19:08.46Trionnislike I said, time will tell
19:08.52modulus_60 outbound min. cost 4.95?
19:08.56Trionnisya
19:08.57harryvvTri, the directory does not exist. I had this resolved in the past but my asterisk hd failed and dont recall the steps to resolve it. I do suspect udev has to be linkes to /dev/zap/channel?
19:09.00Trionnis2.5c after
19:09.13TrionnisI think, lemme look
19:09.14modulus_nationwide?
19:09.16Trionnisya
19:09.22modulus_interesting
19:09.40*** join/#asterisk mbranca (~matteo@host-84-222-6-8.cust-adsl.tiscali.it)
19:09.45harryvvTri do a ls -il on /dev/zap/channel
19:09.48*** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net)
19:10.04Trionnishttps://www.myphonecompany.com/download/MyDevice%20Form.PDF
19:10.13Trionnisthat's the signup form for their "MyDevice" plan
19:10.23Trionnisthe plans themselves aren't listed on the site
19:10.27Darwin35http://pastebin.ca/6970
19:10.34Trionnis1 sec harryvv
19:10.37harryvvk
19:10.53*** part/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net)
19:10.58harryvvI think udev is where the config information is being hardlinked to.
19:11.05harryvvor from
19:11.20RoyKAnyone that knows if Chris Hozian is here?
19:11.42ZeeekWho wants to know?
19:11.45Nuggetanyone!
19:11.59*** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net)
19:12.02ZeeekI don't know anyone
19:12.13RoyKdigium support guy
19:12.53*** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net)
19:13.00ZeeekSo, in order to call my client and transfer him to a conference, I really need to call the conference on my phone and then call him using a call file that dumps him in the conference, yes?
19:13.27ZeeekThats' the only way I coan think of that is technology and phone independent
19:13.53RoyKZeeek: unplug the cable and plug it into the 'conference' switch
19:13.55RoyKsimple
19:14.00RoyK~lart Zeeek
19:14.05Zeeek~lart RoyK
19:14.19Zeeekewww
19:14.24Zeeekgreen blood
19:14.31RoyK:)
19:14.32RoyKblue!
19:14.43Zeeeklots of incest in those countries, huh?
19:15.27tzangeroh dear
19:15.47*** join/#asterisk gi0ffe (~giofe@200.121.60.8)
19:16.00RoyKZeeek: blue blood is a phrase used for royal people
19:16.10Zeeekso if I dial say *90123456789 and generate a call file in an app right before dialing Meetme i'm cool, huh? huh?
19:16.26ZeeekAnd we all know that royals are all related
19:16.39Zeeekor didn't you know that RoyK? Check out the UK Royals
19:16.42RoyKZeeek: you'll prolly need to do more than that to be cool
19:16.48ZeeekLook at those ears for example
19:16.49RoyK:)
19:16.50RoyKhehehehehe
19:17.02Zeeekno I think I WILL be cool
19:17.12Zeeekat least in the context of calling a conference
19:17.14RoyKI thought perhaps you'd missed the phrase
19:17.24Zeeek???
19:17.30*** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net)
19:17.46KalD|WorkIs there anything out there that supports dynamic conference w/ expire times for asterisk?
19:17.49Zeeekphase modulation? Far superior to Amplitude
19:17.49*** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au)
19:17.57ZeeekWhoa dude...
19:18.32ZeeekI think  there is a flag like LAST_PERSON_TURN_OUT_LIGHTS or something
19:18.46*** join/#asterisk Luhiwu (~marsosa@200.63.89.248)
19:18.53KalD|Worklol that'd be cool
19:19.17PTG123hey
19:19.20PTG123can someone test a sip account for me?
19:19.28PTG123i am not getting audio, and want to see if its me locally
19:19.29PTG123or my server
19:20.15*** join/#asterisk topping (~topping@dhcp024-210-082-196.columbus.rr.com)
19:20.40modulus_trionnis, myphonecompany.com ?
19:21.08Zeeekwhy not call FWD echo or something?
19:21.15ZeeekPTG
19:21.21Luhiwuhi all, anyone knows where can I find the expected telephone number lenght for each country?
19:21.25Zeeekthat's what those things are for
19:21.32PTG123zeek: well its def not working fo rme
19:21.39PTG123zeek: i can't even hear voicemail prompts
19:21.45ZeeekLuhiwu - there isn't an easy way to do that
19:21.55Zeeekbecause some countries have variable lenghs
19:22.12ZeeekPTG123 are you using NAT and what phone?
19:22.23LuhiwuZeeek: ok, but is there any list for the fixed lenghs countries?
19:22.28opus___is a Pentium III 700mhz with 128 megs of ram to slow for asterisk?
19:22.32PTG123zeek: yes and xpro
19:22.35PTG123tried from several computers
19:22.45toppingopus___: no
19:22.48Zeeekopus_ not enuf RAM, no CPU problem tho
19:22.56opus___awesome
19:22.59*** join/#asterisk Damin_Mobile (~pocketirc@158.sub-70-214-16.myvzw.com)
19:23.02sambaldoes someone know why i get this error when i try to register on a sip account?
19:23.02sambalGot SIP response 403 "This domain is not served here" back from
19:23.10sambalwhat can be wrong?
19:23.16opus___whats your fromdoain=
19:23.18ZeeekPTG did you turn ON transmit silence in X-Lite/Por
19:23.27sambalopus___: nothing ;)
19:23.30Damin_Mobilesambal: you sip domain is wrong
19:23.30opus___hehe
19:23.42PTG123zeek yah
19:23.42jksanybody using siproxd?
19:23.42PTG123i hear nothing
19:23.45opus___try fromdomain= same as host
19:23.53Zeeekwhat's your NAT forwarding situation?
19:24.19*** join/#asterisk adorah (~jack@80.179.34.21.forward.012.net.il)
19:25.08adorahI have sip audio problems with remote users: anyboday can help?
19:26.34Moc____hi all
19:26.42adorahhi hi
19:26.43tzangerafternoon Moc____
19:26.53adorahI have sip audio problems with remote users: anyboday can help?
19:27.07tzangeradorah: two things.
19:27.18spackleMoc, are you working on the Nortel protocol?
19:27.20tzangeradorah: first, not getting an answer within 60 seconds is not a good reason to repeat
19:27.20adorahlisten..
19:27.21PTG123adorah: join the club :)
19:27.31tzangeradorah: second, you need to ask better questions
19:27.35adorah:)
19:27.39adorahsuach as?
19:27.42tzanger~google how to ask smart questions
19:27.44Moc____spackle: not rightnow, I know 2 person does
19:27.53tzangerthat first link is perfect
19:28.15*** part/#asterisk gi0ffe (~giofe@200.121.60.8)
19:28.20spackleMoc, Just going to offer some encouragement if you were.
19:28.31Moc____;)
19:28.31adorahThx..I hope it is not a link for how to ask smart questions:)
19:28.42tzangeradorah: it is exactly that
19:28.47tzangerwe can't help you unless you ask good questions
19:28.51adorahLOL
19:28.55Damin_MobileDude... this conference ROCKS!
19:29.00tzangerand "sip audio problems with remote users" is *not* a smart question
19:29.07tzangerDamin_Mobile: which conf is that
19:29.20tzangertopping: who me?
19:29.22tzangerI am not trolling
19:29.24PTG123you know
19:29.30PTG123i think i found my problem, its not choosing a codec
19:29.36toppingtzanger: definitely not you
19:29.36tzangerhe wants help, and he needs to be able to communicate effectively to receive it
19:29.38PTG123the question is, why isn't asterisk logging errors about it
19:29.50modulus_isn't there a def. codec?
19:29.51Damin_MobileVon...
19:29.55ZeeekPTG usually you see that on CLI
19:30.05opus___damin - what type of conference setup do you have
19:30.05adorahInded I get no error msg at all..
19:30.06tzangerDamin_Mobile: ahhhh  I thought you were listening to a conf :-)
19:30.07ZeeekFWD is ulaw only IIRC
19:30.13PTG123Zeeek: you would htink :)
19:30.18modulus_hi zeeeeeeeeeeeeeeeeek
19:30.19Damin_Mobileopus; the works..
19:30.24PTG123<PROTECTED>
19:30.25PTG123<PROTECTED>
19:30.25PTG123<PROTECTED>
19:30.25PTG123<PROTECTED>
19:30.25PTG123<PROTECTED>
19:30.25PTG123<PROTECTED>
19:30.26PTG123<PROTECTED>
19:30.28PTG123<PROTECTED>
19:30.29Trionnismodulus_: I'm here... bumped the scroll wheel
19:30.30PTG123<PROTECTED>
19:30.31TrionnisACK
19:30.32PTG123<PROTECTED>
19:30.33TrionnisSTOP
19:30.34PTG123<PROTECTED>
19:30.36PTG123<PROTECTED>
19:30.38PTG123<PROTECTED>
19:30.40PTG123<PROTECTED>
19:30.41Trionnis...
19:30.42PTG123all i see
19:30.44PTG123you see any codec probs?
19:30.45modulus_someone's got a slow connection
19:30.46toppingPTG123: http://rafb.net/paste
19:30.49nestArlol
19:30.49Trionnis~lart PTG123
19:31.05tzangerthat's not good enough jbot
19:31.09tzanger~lart PTG123
19:31.15Trionnismuch better
19:31.19PTG123blah blah
19:31.21Trionnissheesh
19:31.22modulus_jbot halberd?
19:31.24PTG123byt anyhow see any issues?
19:31.25PTG123am i blind? :)
19:31.29tzangerPTG123: it seelected ulaw
19:31.40Trionnismodulus_: yes, that's sipmedia's "consumer" brand
19:31.43PTG123no it selected ulaw between my box and teliax
19:31.46opus___does sip show peers show that its reachable?
19:31.57Trionnissorry, bumped my scroll wheel
19:32.01Trionnisdidn't see your question
19:32.11modulus_trionnis, as opposed to non-consumer?
19:32.23Trionnisyeah, they do bulk accounts
19:32.31modulus_sipmedia is bulk?
19:32.36Trionnisdon't know much more than that really, just know it exists
19:32.43Blackveldoes anyone use iaxtel? i want to the the out/in dailing
19:33.50opus___shesh
19:33.58opus___$30 activation free plus +$10 shipping
19:34.23*** part/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc)
19:34.29Trionnisopus___: ?
19:34.37opus___myphonecompany .. is that sipmedia?
19:34.41Trionnisyes
19:34.50Trionnislook in the upper right hand corner of the main page
19:34.59roamer323hey - anyone from VON online?
19:34.59modulus_trionnis, you do any calling card stuffs?
19:35.00PTG123yes and they suck :)
19:35.05opus___I don't immediately see the $5 dollar plan, is this a special deal
19:35.05PTG123myphonecompany sucks
19:35.10TrionnisI pasted the link
19:35.20Trionnishttps://www.myphonecompany.com/download/MyDevice%20Form.PDF
19:35.24Trionnisthat's the signup form
19:35.28Trionnisfor the "MyDevice" plan
19:35.32opus___thanks a bunch
19:35.38Trionniswelcome
19:35.48Trionnisit's only $10 activ.
19:35.51opus___Wait, if myphonecompany.com sucks and broadvoice.com sucks, who else has a deal $<7 dollars that doesn't suck? :)
19:36.01modulus_opus___, enter the voip
19:36.03adorahanyone can explain the meaning of that msg:
19:36.07adorahMar  7 21:05:02 WARNING[1804]: chan_sip.c:755 retrans_pkt: Maximum retries exceeded on call 6ed142327c9d4c7733c7e4710ad452c8@192.168.2.102 for seqno 102 (Critical Request)
19:36.25Trionnispacket loss would be the first guess
19:36.28anthmmeans network error
19:36.29opus___firewall
19:36.34*** join/#asterisk asjoyner (~asjoyner@dargo.trilug.org)
19:36.35anthmcant ping the other end
19:36.44PTG123opus: all i know is they don't respond to most of their support requests, you can't get through to them on the phone, etc
19:36.48adorahno firewall along the way but router/nat..
19:36.51PTG123i ported my #s from them
19:36.53RaYmAn-BxI get a bunch of those as well..without any problems
19:37.05opus___adorah try nat=yes combinations ?
19:37.06modulus_who wants to look at some agi code?
19:37.07TrionnisI'll refute that
19:37.10asjoynerSomeone correct me if I'm wrong, but you can't do traditional PBX "line presences" with Cisco 7960 phones w/ SIP image talking to Asterisk
19:37.14adorahI did NAT=yes
19:37.19roamer323it is impossible to compare voip provider unless everyone is calling the same number, from the same geographic location :-(
19:37.25modulus_asterisk::agi is kinda sucky
19:37.27TrionnisI just talked to the engineer about a porting situation less than an hour ago
19:37.33Trionnisanswered on the 3rd ring
19:37.33opus___adorah - for [general] ?
19:37.39opus___adorah - do you have externip ?
19:38.01adorahnope havn't done for [general] wilco..thx
19:38.27adorahwhat the hack is externip?
19:38.38modulus_opposite of internip?
19:38.46modulus_i'm a fuckin' genius
19:38.49*** join/#asterisk marshall (~test@S0106000f66563988.wp.shawcable.net)
19:38.56Trionnisironically enough, your typo is accurate
19:39.04opus___sets the outgoing address in NAT
19:39.05adorahoh.yes I've read smthg about it..will try..
19:39.09Trionnisit's a hack to IPv4 to make NAT traversal work
19:39.11Trionnis;)
19:39.36modulus_trionnis what do you use * for?
19:39.42Trionnislots of stuff
19:39.47Trionniscompany incoming calls
19:39.50opus___i got a system to work that was double natted while still on an external IP for the internet with nat=yes and externip=localarea_ip_number
19:39.55modulus_trionnis, office stuffs?
19:39.57Trionnisbluetooth presence to do auto transfer to my cell
19:39.59Trionnisyeah
19:40.10modulus_trionnis, anything prepaid?
19:40.26Trionnisalso have a shoutcast radio show that connects to a meetme conference for a site I manage
19:40.34Trionnisprepaid?
19:40.36Trionnisdon't follow
19:40.42opus___couldn't get mp3player to play shoutcast yet...
19:40.50Trionniswell, icecast, actually
19:40.57Trionnis.ogg stream
19:40.59opus___icecast? is that a program
19:41.04modulus_trionnis, calling card apps?
19:41.06Trionniswww.icecast.org
19:41.07BoRiSmarshall!!!!!
19:41.07Trionnisah
19:41.08Trionnisno
19:41.10TrionnisI don't
19:41.24modulus_i wrote a perl agi script to deduct from a balance per outgoing call
19:41.31modulus_the problem is agi kinda sucks
19:41.32modulus_it's slow
19:41.39modulus_so is perl
19:41.48BoRiSmodulus_: What did you expect? :-p
19:41.49modulus_and asterisk::agi sucks
19:41.50Trionnisphp > all
19:41.52Trionnis;)
19:42.14modulus_asterisk::agi == more unneccessary layers of abstraction
19:42.18adorahpoint is I can dial from remote sip to Zap or sip and it ring the other hand but can't extablish voice both was
19:42.36adorahboth ways..
19:42.49opus___adorah -- sounds like a firewall is blocking RTP .. just a guess
19:42.50Trionnisfirewall issue, most likely adorah
19:42.54modulus_anthm, it's not blasphemey
19:42.56*** join/#asterisk harryvv (~plato@S010600055d210201.vs.shawcable.net)
19:43.02modulus_anthm, perl has it's many uses
19:43.03Trionnismake sure you have the ports forwarded properly
19:43.05opus___unblock udp ports 1024-50000 ?
19:43.07adorahI disabled firewalls both sides
19:43.11modulus_anthm, not when performance is an issue though
19:43.23modulus_don't get me wrong i'm a perler
19:43.49opus___adorah - i think there is a bug also in asterisk with nat=yes where you have to actually 'stop now' and restart the whole process
19:44.03*** join/#asterisk Goshen (Goshen@c-67-172-238-57.client.comcast.net)
19:44.17adorahgive me dime for every time I've done it..:)
19:44.25Goshenwhen I got that alarm 1 red from my zap line...
19:44.34opus___sip show peers shows the user as reachable?
19:44.35GoshenI picked up the PTSN phone, and sure enough it was dead
19:44.40adorahyup
19:44.46opus___it rings, no sound
19:44.49opus___on either end?
19:44.50Goshenso I went down the street to the interface box, and two guys were there working on it
19:44.53adorahindeed
19:44.54Goshenso they fixed it
19:45.05adorahring no sound
19:45.24opus___i'd run a tcpdump on it
19:45.32opus___see if rtp packets are going by successfully
19:45.44adorahand if mot?
19:45.44opus___and if so, i would experiment with turning off all codecs and enabling them one by one
19:46.24adorahI use xlite btw..may be there is the problem?
19:46.25opus___adorah run that by again, mot?
19:46.34*** join/#asterisk gr8nash (~basketoju@mamabear.si-forest.com)
19:46.35opus___dunno, never used it. I use sjphone
19:46.43opus___.. heh, try sjphone
19:46.55spackleI helped somebody today who had a problem with xlite.
19:47.04opus___for some reason i hear xlite is evil
19:47.07BuckRogersi got xlite, its nice
19:47.08adorahthe dtmdtf is inband in sjphone?
19:47.13spackleSIP implemenation is off a little in xlite.
19:47.17BuckRogerswhy its free great for testing
19:47.26*** join/#asterisk [Outcast] (~knoppix@h0006259a2649.ne.client2.attbi.com)
19:47.41opus___adorah - dunno
19:47.49BuckRogersspackle dont forget they want you to buy the full version
19:48.00adorahsjphone is free too..here ip phones r very expensive yet
19:48.22spackleBuckRogers: I've never used it.  I used firefly for softphone.
19:48.22Nuggeteveryone has differing ideas on what "very expensive" means.
19:48.25scrubbcan anyone tell me whatis  the biggest cause of:  WARNING[1102212400]: PRI: !! Got reject for frame 4, retransmitting frame 6 now, updating n_are!
19:48.32*** join/#asterisk nani707 (~nene@nat-183.sjc1.globix.net)
19:48.35opus___does anybody use an IP phone that they swear by... I support about 60 polycom IP500 and have to RMA about one a week...
19:48.36nani707hi everybody
19:48.39adorahI think over 200$ is very expensive
19:48.58*** join/#asterisk lucca (~lucca@export.accela.net)
19:49.27nani707anybody can tell how two trunk  SIP  outgoing lines ,  as we  do with Zap
19:49.37opus___oh yeah, if you want to buy a cheap IP500 the company I rma'em to is selling them on ebay for $80/pop
19:49.46Nuggetheh
19:49.47Juggieopus___, i've had no problems with the mitel5055
19:49.52Nuggetpre-abused phones.
19:49.54spackleopus__: that's scary, are they phone abusers?
19:49.55Juggiebut the 5220's are out now
19:49.56opus___give me the MAC and i'll tell you why I rma'd it :)
19:50.01Nuggethaha
19:50.19adorahI know how it is on ebay..just that import is not easy..local regulations..
19:50.21opus___juggie - thanks, I'll check it out
19:50.23Nuggetwe should set up a carfax-style phonefax site for phone MACs.
19:50.29Juggiecisco's have problems withg the pins in the handset
19:50.30adorahtoo much trouble and paperwork
19:50.38Juggiethey allways come loose so you cant hear anything on the headset
19:51.07opus___polycom has serious problems with their headset port as well, weird.
19:51.22Juggiethe 5055/5220 i've had no problems with. other then the 5055 locks up if you dont send keep alive packets from the sip server... so set qualify=yes
19:51.27opus___last week one person blew up 4 phones in one day
19:51.28adorahDid anyone try the IAX sf from Sokol?
19:53.02opus___mitel 5055 is around $350?
19:53.47Juggiei think we get them for like 250$ cdn
19:53.52opus___oh
19:53.56spackleopus__: was he feeding them coffee?
19:54.28modulus_app_dial.c:510 wait_for_answer: Unable to forward frame
19:54.30opus___nope, I think it was either bad power or the guy was turning the headset amp up _so_ high that it blew a circuit
19:54.34modulus_almost as useful as BSOD
19:54.54nani707hi, i need to trunk two sip outbound lines as single trunk (if possible load balance),  there are instructions for zap channels , how to do it with sip
19:54.56*** join/#asterisk newpers (newpers@ip24-56-8-180.ph.ph.cox.net)
19:55.38Juggienani707, multiple providers?
19:55.52SexyKenHey guys I need a web app that will show me queue info
19:55.54nani707same provider juggie
19:55.57SexyKenIs this hard to develop?
19:56.12Juggienani707, then it shoudn't really be an issue, unless you are using two different accounts?
19:56.22nani707yes  two  different  accounts
19:56.32spackleUm, SexyKen, have you looked at Flash operator panel?
19:56.38nani707sip/1001    sip/1002  outbound lines  for example
19:56.38Juggiewrite some code in your dialplan to count the number of active calls per account.
19:56.43Juggieuse global variables.
19:56.58Juggiewhich ever account is lower, dial with that
19:57.17adorahWell enough stupid questions 4 1 day. Thx opus..will try..
19:57.25nani707i have  used  availchan  ,  this worked but  if i have 50 lines i  am afraid it takes too much time
19:57.27Juggieor you can just use random to select the sip to use
19:57.40*** part/#asterisk adorah (~jack@80.179.34.21.forward.012.net.il)
19:57.43modulus_Mar  7 12:05:23 WARNING[16904]: Unable to forward frame
19:57.47modulus_wtf IS that?
19:58.05Juggienani707, sip has no knowledge of number of lines, its up to your provider to determine how many calls it wants to host for you
19:58.06*** join/#asterisk j0 (dan@S010600095b00a5a9.vc.shawcable.net)
19:58.44nani707yes,  i am using  broadvoice , he provides only one per  sip
19:59.11nani707i read somewhere like   zap/g2  which can  roundrobin within a  t1  , can we  do same with  sip
19:59.47Juggienani707, then that is easy
20:00.02Juggiejust track of the sip line is in use or ont
20:00.03nani707but  zap is  for  analog lines only  right
20:00.10nani707oh!
20:00.14Juggiezap is for pstn, not necessairly analog
20:00.20Juggie*of=if
20:00.22nani707what if i have 100 lines
20:00.38Juggiewell if you have 100 broadvoice accounts that would be rather painful no?
20:00.39nani707can i use  availchan  still,  it may take time right
20:01.03Juggiei'm not famaliar with availchan & sip
20:01.17nani707may be not  broadvoice,  but  some other providers
20:01.24modulus_wow that agi was pretty damn fast
20:01.29Juggiealso, you can just try the providers in order
20:01.35nani707ok
20:01.39Nuggetchanisavail() can take multiple targets -- it's not like you'd need to make 100 calls to that application.
20:01.43Juggiei mean, if you have an active call, it will reject the second one?
20:01.53Nuggetbut I fear there is probably a more fundamental misunderstanding at work here
20:01.55Juggiethen you just move to trying the next provider
20:02.11nani707got  it  juggie
20:02.40nani707may be some kind of macro may do it
20:02.50Juggieyou will just have to do a Dial and trap for the error that means call rejected....
20:02.57nani707yeah
20:03.09newpersdo you all se the cisco 7920 wireless ip phone ever dropping it's price from $750 to ~$200?
20:03.32Juggiethat would be easiest.... thats what i do when people dial 4 digit numbers on my sip phones.,
20:03.34Juggiei try it via sip
20:03.38Juggieif it fails, i dial pri
20:03.49Juggiei'm on an internal pri so i can call 4 digits on it.
20:03.50nani707ok
20:03.57nani707ok
20:04.12nani707thanks  juggie and  nugget
20:04.19*** join/#asterisk Damin_Mobile (~pocketirc@99.sub-70-214-6.myvzw.com)
20:04.33*** join/#asterisk hajekd (~hajekd@21.208.65.212.contactel.net)
20:05.01nani707this asterisk is  really  interesting,  works  the way it is supposed to
20:05.13modulus_no it isn't
20:05.14modulus_voip sucdks
20:05.17modulus_sucks*
20:05.28Damin_MobileYeah... Unlike Windows!
20:05.32opus___is there any way to seemlessly crypto sip calls...
20:05.34Nuggetall software sucks.
20:05.40QwellNugget: people too
20:05.48nani707there is no software  which  has no  bugs
20:05.55Qwellexcept mine
20:05.59nani707factor of  good better  best
20:06.08Nuggetno software is best.
20:06.25*** join/#asterisk nel (~oeo@199.75.106.11)
20:06.28modulus_my prepaid agi software is the best.
20:06.33nelhello
20:06.35nani707and what is  it?
20:06.39PTG123my software is usually pretty bug free :)
20:06.50modulus_my software follows the unix philosophy
20:06.54modulus_[07-Mar:12:06] *boch!~as24@200.59.172.98* hey
20:06.57nani707don  wory  ptg,  when u put load on  it  u will see bugs popping up
20:07.03nelI'm having problems uploading sip.dl file on Polycom Soundpoint 500 anybody using those type of phones?
20:07.09modulus_hi boch!
20:07.15PTG123nani: i am use to writing software that does millions of requests per day :) load is my specialty
20:07.16bochmodulus_ hi
20:07.17nani707i saw this   areskiCC  , it';s  cool
20:07.28nani707wow  PTG
20:07.31PTG123right now my big problem is fighting asterisk bugs
20:07.52nani707may be  write  your own Asterisk
20:07.52bochmodulus_ is your agi in perl?
20:07.57modulus_boch, it sure is
20:08.17PTG123nani707: thought about it, but alot of work :)
20:08.32PTG123nani707: i think i am gonna start by redoing chan_sip.c
20:08.39opus___nel - got it to work, didn't make any changes to the polycom files
20:08.43nani707see!!!,  the one  who  wrote Core  is  not  a  idiot
20:08.58bochmodulus_ can u giveme an example of comunicating with * in perl?
20:09.05opus___nel -used the web interface to configure the phone for SIP and for REGISTRATION, make sure you have the IP in both
20:09.13modulus_boch, sure hold on
20:09.24opus___nel - i used the 1.3.1 rom updated from ftp
20:09.25bochmodulus_ :D
20:09.38modulus_ummm i'm gonna go smoke
20:09.41modulus_i'll be right back
20:09.44nani707k  mod
20:10.05modulus_then i'll show you my perl
20:10.16NuggetI quit smoking 7 years, 10 months, 23 hours, 10 minutes, and 16 seconds ago.  During that time, I would have smoked 62,922 cigarettes. (That's like smoking a 2.98 mile-long cigarette)  By quitting, I've saved $11,011.35!  I've avoided inhaling 1.64 kg of tar, 100 grams of nicotine, and 1.01 kg of carbon monoxide.
20:10.29Nugget^ my perl  :)
20:10.31Trionnisoh boy
20:10.37Trionniser
20:10.48opus___PTG123 -  what type of bugs
20:12.30PTG123opus: my latest one is a codec pproblem :)
20:12.33shido6boink
20:12.45bochmodulus_ ok, np
20:12.47nani707codec  ,  is it possible
20:13.00PTG123opus: calls just weren't sneding audio, yet there were no errors in the cli.. finally i realized they weren't selecting any codec
20:13.29PTG123it seems the allow/deny codec stuff is flakey
20:13.35PTG123it worked before i went to bed which is sad :)
20:13.39PTG123i woke up to no calls working
20:14.13nani707oh  no
20:14.31PTG123disallow all
20:14.34PTG123allow g729
20:14.37PTG123didn't work :)
20:14.43Pinholelots of stuff works when you go to bed with *.  I find that if asterisk restarts when I go to bed, more stuff works in the morning.
20:14.48opus___seems very straight forward, like, you would have to put effort into messing up that functionality:)
20:14.49nani707sometimes i  see unknown  in  show sip channels,  also  internal addresses on  show sip channels
20:15.03opus___pinhole - i have that same problem:)
20:15.04PTG123well allow all
20:15.06PTG123fixed it
20:15.06PTG123heh
20:15.09PTG123with nothing else
20:15.16PTG123i don't care enough to restrict codecs anyhow
20:17.38opus___i think thats to get around buggy sip implemenations anyway
20:17.40*** join/#asterisk tigger42 (~tigger42@p54A2536F.dip.t-dialin.net)
20:17.46opus___disallow all
20:17.52opus___allow the_one_that_works
20:18.02PTG123opus___: i think chan_sip just needs to be redone.. i am gonna put that on my list
20:18.09PTG123althought i hear someone else may be doing that
20:18.35opus___this ircII script seriously sucks
20:18.44PTG123i already replaced all the sip user and peer stuff with my own code
20:18.52opus___really
20:19.09PTG123yes works great :)
20:19.22PTG123its more intelligent too, voicemail is now automatic, etc
20:19.22opus___what were the problems with the origional stack?
20:19.32opus___you just wanted extra features?
20:19.33PTG123they were slow pigs :)
20:19.45PTG123well i didn't like all the string queries and link list passing of realtime
20:19.46opus___i notice that
20:19.48eKo1You mean the code is ugly and slow?
20:19.48PTG123and i needed a realtime db
20:19.52PTG123hah yes
20:20.07eKo1Hmm...I'll have to take a look at it.
20:20.08PTG123mine is all hash based in memory, synched to disk.. with a remote server for adding accounts
20:20.25opus___do you use odbc?
20:20.37PTG123no i use my own db.. sort of like berkeleydb
20:20.45PTG123and my own client/server sofware for ifacing with it
20:21.26tigger42hi all!, before i start to ask silly questions: is there a tutorial to read for following setup: isdn line, isdn capi card, isdn incoming/outgoing calls gated to/from PCs via SIP, no outgoing VoIP Traffic?
20:21.48opus___have you searched voip-info.org?
20:21.55opus___cd /usr/src/
20:22.22tigger42opos: thanks, will look there.
20:22.23MikeJ[Jayden]anyone here using head and using IAX trunking?
20:22.41develso where are the sqlite database schema creation scripts then?
20:22.45opus___./channels/chan_sip.c
20:24.10develah, i see it "auto creates" the table (not sure i care for that behaviour)
20:24.31opus___uhhh
20:24.37opus___<PROTECTED>
20:24.37opus___nds FROM sipfriends WHERE ipaddr=\"%s\" AND port=\"%d\"", ast_inet_ntoa(iabuf, sizeof(iabuf), sin->sin_addr), ntohs(s
20:24.37opus___in->sin_port));
20:24.43opus___mysql functions?
20:25.02opus___where ipaddr=\""; drop database master;
20:25.02opus___hehe
20:26.08*** join/#asterisk Skysky (~Miranda@host6614613596.biz.tor.fcibroadband.com)
20:26.09*** join/#asterisk marc324 (~marc32344@64-34-29-65.dsl.teksavvy.com)
20:26.32Skyskyhi, can someone suggest me a cheap 1800 provider pls?
20:26.47*** join/#asterisk harryvv (~plato@S010600055d210201.vs.shawcable.net)
20:27.06nestAriax.cc
20:27.55*** join/#asterisk topping (~topping@dhcp024-210-082-196.columbus.rr.com)
20:28.01harryvvwhat process during the asterisk install makes /dev/zap/channel ? is it make install26 while in /usr/src/zaptel ?
20:28.19harryvvlinux26 I mean
20:28.52tzangerharryvv: it doesn't do that in 2.6
20:28.55tzangerit uses udev
20:28.57tzangerread README.udev
20:29.03hajekdhow many calls can asterisk handle on pIII/800 MHz?
20:29.05Trionnisnot completely true
20:29.07SkyskynestAr: im from canada. i wonder if iax.cc do service for canada
20:29.15TrionnisI'm running a CentOS 3 box that doesn't use udev
20:29.18tzangerSkysky: no
20:29.21MikeJ[Jayden]hajekd, depends
20:29.28aminorexhajekd: about 20 or 30
20:29.35nestArmy suggestion is to move.
20:29.38nestArbut that's just me
20:29.41Trionnis2.6.10-grsec kernel
20:30.32Juggiedoes anyone still have a copy of x-web (from xten) they dont offer it as a download anymore.
20:30.48Skyskytzanger: by not serving
20:30.57tzangerSkysky: huh?
20:31.11Skyskysry.. i hit enter by mistake
20:31.32harryvvtzanger yea i did and modified the permission in /etc/udev/permissions.f/50-udev.permissions and does not seem to resolve the permission issue. if thats what the problem is.
20:31.45tzangerharryvv: are you running as non-root?
20:31.46Skyskyi was just asking . by saying not serving, does it mean that ppl from canada won't be able to reach me by dialing my specific 1800 #
20:31.53harryvvno running as su -
20:32.11tzangerSkysky: by not serviing I mean that they will have to subject you to something on the line of a 10c/min tarriff (international 800#)
20:32.20tzangerSkysky: it's not iax.cc who imposes that, it is the gov.
20:32.26tzangerharryvv: that's not hte issue then
20:32.36Skyskyoic~
20:32.37Skyskythx
20:32.40tzangerdid you add the lines to the udev.rules that *create* the dev structure for it?
20:32.54*** join/#asterisk Corydon-w (tan@vcchgate.vcch01.springfield.tn.us.vcch.net)
20:33.23harryvvtzanger you mean 50-udev.permissions ?
20:33.36Darwin35re
20:33.36tzangerno
20:33.38tzangerthat's the permissions
20:33.45tzangerI mean the udev rules themseelves
20:33.46Darwin35the t-3 in my rack is stp
20:33.54Darwin35boxes back online
20:34.05tzangerharryvv:
20:34.06tzangerI mean
20:34.07MikeJ[Jayden]harryvv, there is a doc on the wiki for centos4, the udev.rules lines you need are in there
20:34.08tzangerKERNEL="zapctl",        NAME="zap/ctl"
20:34.08tzangerKERNEL="zaptimer",      NAME="zap/timer"
20:34.08tzangerKERNEL="zapchannel",    NAME="zap/channel"
20:34.10tzangerKERNEL="zappseudo",     NAME="zap/pseudo"
20:34.13tzangerKERNEL="zap[0-9]*",     NAME="zap/%n"
20:34.15tzangerin the udev.rules file
20:34.16Darwin35everyone here is fired go homre
20:34.21MikeJ[Jayden]or you could just copy those :)
20:34.26NuggetI am home.
20:34.32harryvvi dont see one but do have a 50-udev.rules
20:34.38tzangerharryvv: add 'em there then
20:34.41harryvvk
20:34.59*** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no)
20:35.28Juggieanyone have x-web still?
20:35.44harryvvtzanger, looks like it was already taken care of.
20:35.46MikeJ[Jayden]Juggie, is ther ean echo in here?
20:36.25tzangerharryvv: ok
20:36.37tzangerharryvv: and /dev/zap/ctl (and timer and channel and so on) were all created?
20:36.49harryvvno
20:36.58Moc____what voip provider do you people recommend ?
20:37.00tzangerthen it wasn't already done :-)
20:37.01eKo1Registration timeouts are usually a signed of network problems right?
20:37.07tzangerharryvv: or you're not running udev
20:37.11Moc____maybe there is one I havent tryed ;)
20:37.13tzangerharryvv: or you haven't loaded the zaptel drivres
20:37.16wolfsonis it normal for out of state calls to just get a CIDNAME of "VA CALL", etc... I assume the telco is trying to save money on lookups
20:37.19MikeJ[Jayden]MOC!
20:37.19tzangerMoc____: nufone
20:37.27harryvvsorry some of them were. Channel was not created
20:37.39tzangerharryvv: your udev rules file must be fucked up then
20:37.46harryvvzaptel drivers are loading on boot and have been verified with ztcfg
20:38.03tzangerharryvv: then your udev rules aren't right
20:38.09tzangeror udev is br0ked
20:38.32*** join/#asterisk andrew_un (~andrew@h-67-102-251-250.nycmny83.covad.net)
20:39.02harryvvtzanger,  here is whats at the bottom of 50-rules.d KERNEL="zapctl", NAME="zap/ctl"
20:39.02harryvvKERNEL="zaptimer",NAME="zap/timer"
20:39.02harryvvKERNEL="zapchannel",NAME="zap/pseudo"
20:39.02harryvvKERNEL="zap[0-9]*",NAME="zap/%n"
20:39.30harryvvmm
20:40.33Moc____tzanger: Im gaving problem with nufone ..
20:40.39eKo1I just know I have network problems somewhere. I'm getting a lot of: Maximum retries exceeded on call...
20:41.15tzangerharryvv: read README.udev and follow it precisely
20:41.16Moc____hi MikeJ
20:41.19tzangerMoc____: what problems?
20:41.35harryvvalready read but will look again.
20:41.48Moc____audio aint good, I get echo, I also get call that audio work only in 1 direction
20:42.18tzangerMoc____: odd, I do 5kmin/mo with them without *any* issue
20:42.38Moc____everything my routing drop on nufone, Im sure to receive a complain
20:42.48tzangerMoc____: huh?
20:43.02MikeJ[Jayden]tzanger, where are you?
20:43.12tzangerMikeJ[Jayden]: midwestern ontario, canada
20:43.21Moc____everytime my routing make me use nufone, I get a call sayign qualify aint good
20:43.24opus___anyone use scandsp?
20:43.25MikeJ[Jayden]o... your one of those :)
20:44.00Moc____I get the same bad qualify (exact same problem) using Voiceconduits
20:44.01MikeJ[Jayden]opus, no, no one uses scandsp, some use spandsp
20:44.15Moc____asterlink seem alot better
20:46.48Luhiwuanyone knows who can sell local numbers in USA & Europe?
20:47.11jksanyone knows of an easy way to get conference call working? (without timing hardware)
20:47.24SexyKenHey guys
20:47.39Luhiwujks, i've read there is a module for zaptel timing based on usb
20:47.46tzangerMoc____: that is strange, what's your traceroute to switch-1 look like
20:47.46SexyKenI keep getting this problem in Asterisk. Agent 1304 is logged into the queue, yet he's not getting the calls. Instead Asterisk CLI shows this:
20:47.47SexyKen<PROTECTED>
20:47.47SexyKen<PROTECTED>
20:47.47SexyKen<PROTECTED>
20:47.51jksLuhiwu: do you know it's name?
20:48.01jksLuhiwu: I've tried app_conference, but it's very alpha quality
20:48.02Luhiwujks: looking for it
20:48.23*** join/#asterisk iceyp (~icepick@max.unix.co.nz)
20:48.33andrew_unHi All! any one can help with tons of "Uhhuh. NMI received for unknown reason 21..." stuff?
20:48.49nelI'm having problems uploading sip.dl file on Polycom Soundpoint 500 anybody using those type of phones?
20:48.57Luhiwujks: http://www.voip-info.org/wiki-Asterisk+timer
20:49.03*** join/#asterisk santiago (~santiago@63.245.86.95)
20:49.06iceyphey guys, creating a meeting room or conf room without a password, is it required to have zaptel drivers or are their other ways
20:49.06harryvvvonage is really pushing there marketing up here in bc canada.
20:49.20*** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com)
20:49.47jksiceyp: try app_conference
20:49.54*** join/#asterisk Televoip (~Televoip@HSE-Ottawa-ppp3493161.sympatico.ca)
20:49.55harryvvsee it on tv on the radio in newspaper flyers.
20:50.49*** part/#asterisk Televoip (~Televoip@HSE-Ottawa-ppp3493161.sympatico.ca)
20:52.35*** join/#asterisk Spacebar (~stingray@stingr.net)
20:52.54*** join/#asterisk nomercious (~nomerciou@HSE-Ottawa-ppp3493161.sympatico.ca)
20:53.28opus___do you need to have the asterisk zaptel dummy timer installed for pure VOIP?
20:53.37Jer1326<PROTECTED>
20:53.40*** join/#asterisk Sedorox (brandon@Neptune.client.wlgrv.pa.sed6.net)
20:54.14nelI'm having problems uploading sip.dl file on Polycom Soundpoint 500 anybody using those type of phones?
20:54.17RoyKopus___: depends what you do
20:54.29RoyKopus___: for SIP, non-trunking IAX etc, no need
20:54.46*** join/#asterisk ckruetze (~ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com)
20:56.45iceypjks to transfer a call it also requires zaptel?
20:56.58jksiceyp: no
20:57.02PTG123what is the diff between the cisco 7960 and 7960G?
20:57.18iceypwhere is the cheapest location to get a 7960?
20:57.31PTG123i'll tell you what if you tell me my answer :)
20:57.43iceyplol i dont know :(
20:57.47iceypi can ask google tho
20:57.58PTG123you want new or used?
20:58.07iceypeither
20:58.14iceypebay too expesnive and difficult to get em
20:58.15Nuggetthe 7960G has pictures instead of words on the buttons.
20:58.26RoyK~seen dsfr
20:58.39jbotdsfr is currently on #asterisk
20:58.43RoyK~seen beer
20:58.44jbotbeer <ExUser@h-67-101-28-114.nycmny83.dynamic.covad.net> was last seen on IRC in channel #kde, 20d 18h 13m 26s ago, saying: 'to work from my home pc'.
20:59.01tuxinator_linuxPTG123: the G model is 1000M
20:59.38Nuggetwhat does "1000M" mean?
20:59.48tuxinator_linuxGig Ethernet
21:01.03tuxinator_linuxPTG123: Sorry I a was wrong
21:01.28fac_any idea to generate some uniqueid
21:01.34fac_maybe sessionid? on webservice?
21:01.43fac_this is for temporaly filename
21:02.00shido6isnt there a timestamp.agi
21:02.05shido6or something out there somewhere
21:03.06Nuggettuxinator_linux: I don't believe that is correct.
21:03.21tuxinator_linuxNugget: tuxinator_linux: PTG123: Sorry I a was wrong
21:03.32iceypfsck i didnt find out where to get the phone from
21:03.39Nuggetafaik, the only two differences between the 7960 and 7960G are the pictures/words on the buttons and the wiring needed to do PoE.
21:03.42*** join/#asterisk denon (denon@synapse.subneural.net)
21:03.42*** mode/#asterisk [+o denon] by ChanServ
21:03.47Sedoroxiceyp: voipsupply.com?
21:04.01iceypSedorox  they the cheapest?
21:04.10Nuggetthe "G" certainly has no correlation to gigabit.
21:04.15Sedoroxdunno
21:04.16Nugget(if that's what you were guessing)
21:04.16Luhiwuanyone knows who can sell bulk local numbers in USA & Europe?
21:04.23Sedoroxbrb
21:04.41Nuggetthe "G" in 7960G is for "Global"
21:05.23tuxinator_linuxNugget: I got mixed up with the '-GE' on some models
21:07.11tuxinator_linuxMan, I overslept, but like 4 hours
21:07.34tuxinator_linuxoh well
21:08.42*** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl)
21:08.55*** join/#asterisk davidleib (reik@85-250-7-229.bb.netvision.net.il)
21:10.03*** join/#asterisk BigCanOfTuna (~chatzilla@dsl-macn-66-18-205-30-cgy.nucleus.com)
21:12.32BigCanOfTunaI would like to have my asterisk server call a PSTN number and present a dialtone to the user who picks up. Anyone have an idea how this might be done through the manger api?
21:12.48opus___so, whats the best asterisk/database setup (stable) around now?  any good instructions?
21:13.21SedoroxBigCanOfTuna: there is a wiki about how to have asterisk do callbacks
21:13.32Sedoroxprobably what you would want...
21:13.39BigCanOfTunaSedorox: great, I'll have a look, thanks.
21:13.55Sedoroxyup.. I'm not sure where exactly it is.. but I've seen it thrown in there somewhere
21:13.56Sedoroxlol
21:14.42*** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk)
21:15.34Goshendoes anyone here have personal experice faxing with VOIP?
21:16.06opus___goshen - i was looking into building scandsp but in the current cvs head it fails
21:16.07nelanybody can give me a hand configuring polycom ip soundpoint 500 for asterisk?
21:16.33eKo1Hmm....Having * behind a multihomed nat router is trouble I tell you.
21:16.35opus___nel - sure
21:16.48nelopus___: thanks
21:16.51opus___eKo1 - yup i got it to work
21:17.02opus___nel - use the 1.3.1 rom
21:17.04nelI'm having problems uploading the sip.ld file from the asterisk wki
21:17.08nel1.3.1 rom
21:17.10nelok
21:17.16neljust that file?
21:17.20opus___Umm,
21:17.31eKo1Somehow, the load balancing is fucking up my sip registrations.
21:18.00opus___nel - on your ftp server have this archive unziped http://www.freedomphones.net/polycom/files/SoundPointIP_SIP_1_3_1.zip
21:18.05Goshenopus: I tried but mine failed too..I posted a bug report on it, and he said you have to edit the makefile or something like that
21:18.07opus___don't make any changes to any of the files
21:18.25opus___nel - then, have an entry in your sip.conf
21:18.54iceypok found somewhere cheaper... http://ecoustics.pricegrabber.com/search_getprod.php/masterid=559911
21:19.03iceyp$296 for 7960G
21:19.25nelopus___: let me try that, thanks.,..I tried with some of roms on that page and I have problems uploading the sip.ld file
21:19.25opus___nat=yes \ type=friend \ host=dynamic \ dtmfmode=inband \ progressinband=no \ progressinband=never \ qualify=yes
21:19.27nellet me try again
21:19.34nelthanks!!
21:19.36tzangerMoc____: nufone does all their own TDM termination -- if you place a call to nufone it is terminated to TDM by them, not handed off ot a third party VOIP provider
21:19.40opus___nel - the trick is to first get it working using the web interface
21:19.49nelI see
21:19.54nelI have an old firmware installed
21:19.56nelfrom 2002
21:20.00opus___then, making changes:)
21:20.07nelI don't know if that's not letting me upload the new rom
21:20.14*** join/#asterisk e-guy (~e-guy@host162-48.pool8252.interbusiness.it)
21:20.16opus___what ftp server are you running?
21:20.21nelwu-ftpd
21:20.28opus___thats your problem
21:20.36nel:O
21:20.38opus___the word is wu-ftpd doesn't work
21:20.43opus___vsftpd does
21:20.52nelthanks I'm gonna try that
21:20.56nelthank opus!!!!!!!!!
21:20.57nel:D:D:D
21:21.24e-guyany coder with experience in iax out there?
21:22.12*** join/#asterisk SleepyCow (SleepyCow@wnpgmb09dc1-89-6.dynamic.mts.net)
21:22.17nelbrb
21:22.18nelbye
21:24.28*** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no)
21:24.30SleepyCowHello all. I am the technical director of a small, non-profit radio station, and would like to implement VOIP telphony. Would anyone be able to take the time to answer a few questions regarding a odd configuration that we would have to implement?
21:24.39*** part/#asterisk santiago (~santiago@63.245.86.95)
21:24.46opus___sleepycow sure
21:25.01shido6shoot, SleepyCow
21:25.04SleepyCowFantastic
21:25.22SleepyCowCurrently, the radio station has a total of eight phone lines, seven of which I would like to replace with VOIP
21:25.30RoyK~seen dsfr
21:25.32jbotdsfr is currently on #asterisk
21:25.42SleepyCowThe eighth is a leased line to our transmitter site, so it has to stay with the telco
21:25.45tuxinator_linux~seen me
21:25.47jbotme <~daniel@tc26.chem.vu.nl> was last seen on IRC in channel #debian, 5d 3h 43m 54s ago, saying: 'can anyone help with getting an external monitor to run?'.
21:25.55RoyK--- [dsfr] idle 46:33:30, signon: Fri Mar  4 17:55:30 .......... OMG
21:26.21tuxinator_linux~seen tuxinator_linux
21:26.22jbottuxinator_linux is currently on #asterisk.  Has said a total of 155 messages.  Is idling for 1s
21:26.29RoyK~lart jbot
21:26.32Sedorox~seen Sedorox
21:26.33jbotsedorox is currently on #asterisk (32m 53s).  Has said a total of 8 messages.  Is idling for 1s
21:26.37SleepyCowWe have plenty of bandwidth, and have an all purpouse box we use as a router, file server, DB server, etc. It is a Sempron 2600+ with 256ram, 2x 80gb RAID 1, dual ethernet
21:26.43RoyK~lart Sedorox
21:26.49SleepyCowAnd it is currently running debian, 2.4 kernal i think
21:26.54SleepyCowAnyhow,
21:27.03Sedoroxfunny.. as I hate my access cladd
21:27.21SleepyCowThree of the phone lines are used for the office. They currently have a 474 Prefix and opperate as follows:
21:27.53SleepyCow1 Line is a rotary (7027) that if 7027 is busy it rings the second line (6588) and if thats busy dumps straight to voice mail
21:28.09SleepyCowthe third line is (6518), not a rotary, if busy staright to voice mail
21:28.26SleepyCowtwo phone in the office, one with 7027 & 6588, one with 6518 & 7027
21:28.36SleepyCowwe also have 4 POTS lines, hooked to our radio equipment
21:28.39shido65 phones so far, right?
21:28.45shido64 pots lines from the telco
21:28.56SleepyCow2 phones on 3 lines, all from a PBX, prefix 474
21:29.08SleepyCow0 phones (radio equipment) on 4 pots lines, prefix 269
21:29.47SleepyCowthe POTS lines opperate as follows: The main line (8636) is a rotary for two more lines. If all three are busy they just busy out.
21:30.07SleepyCowThere is a fouth line we use as a VIP line for outgoing calls only, generally (8929 i think)
21:30.16SleepyCowso here is what I need to know:
21:30.34*** join/#asterisk zyke (~zakforeve@84.45.132.117)
21:30.51SleepyCow#1: Can Asterisk (or any VOIP solution, for that matter) handle 'rotary' configurations where one phone line will drop through to another of busy
21:31.00shido6yes
21:31.05opus___is rotary an euroopean word for queue?
21:31.08SleepyCow#2: How can i interface 4 pots devices to VOIP wihtout buying 4 seperate SIP to POTS converters
21:31.11shido6rotary, analog
21:31.13SleepyCowno no
21:31.18SleepyCowrotary means, say you have 2 lines
21:31.19shido6pulse phones
21:31.21tuxinator_linuxpulse
21:31.24SleepyCowNOT PULSE
21:31.26shido6ok
21:31.29Blackvelhehe
21:31.31SleepyCow555-1111 and 555-1112
21:31.35SleepyCowi am talking on 555-1111
21:31.41tuxinator_linuxcall hunt?
21:31.48SleepyCowyou call 555-1111. It is buys, but instead of giving you busy it ring 1112
21:31.51shido6yes
21:31.52*** join/#asterisk jterrero (~jterrero@mcse-irc.isys-networks.com)
21:31.52opus___you mean a work queue, workgroup, rotary to us means grandma's phone
21:31.53SleepyCowcall hunt might be a word for it yes
21:31.57shido6u can do that in Asterisk
21:32.00shido6yes you can, SleepyCow
21:32.02Juggiethats called a huntgroup
21:32.03Juggievery simple
21:32.09tuxinator_linuxIn US is is called 'call hunting'
21:32.16SleepyCowsorry thats what my telco calls it belive iit or not (MTS Alstream)
21:32.27SleepyCowANYHow...
21:32.27opus___oh yeah, call hunting != queue. what am i smoking.
21:32.32terrapenanyone using a rotary phone with an IAXy
21:32.33Juggienone the less, yes you can call all the phones in order
21:32.53SleepyCowOK, i figured the 'call hunting' wouldnt be a problem
21:32.59terrapeni'm wondering if the IAXy translates pulse to tone
21:33.06SleepyCowto do it, I need a VOIP account / phone number for each, right?
21:33.08terrapenso i can use a rotary phone to navigate IVRs
21:33.14*** join/#asterisk jedaustin (~chatzilla@140.198.4.225)
21:33.25SleepyCowterrapen: Best best is to build/buy a pulse to DTMF converter
21:33.28opus___terrapen - yeah, i really would like to do that
21:34.09SleepyCowAnyhow, so now the radio station stuff is all POTS only (The office phones can be replaced with SIP phones)
21:34.20SleepyCowhow do I interface 4 pots lines to VOIP without buying 4 seperate converters?
21:34.28SleepyCowis there a reasonably priced PCI card that will do it?
21:35.15*** join/#asterisk yaaar (~chatzilla@lifebook.tranquility.net)
21:35.20yaaarword
21:35.34*** join/#asterisk file[Digium] (~jcolp@66.199.241.90)
21:35.36file[Digium]hello everyone
21:35.45SleepyCowGreetings & Saultations
21:35.59sivanablinking green/amber on a Cisco switch means what?
21:36.00Sedoroxyou sucj
21:36.02Sedoroxsuck
21:36.07opus___sleepycow - you could outsource all the lines to a voip provider fairly cheap.  I don't have any experience with the hardware part
21:36.14yaaaranybody using asterisk@home? And perhaps know why after installation it sits at "GRUB loading stage 2" for like 5-10 minutes before just giving up and leaving me at a "grub>" prompt?
21:36.31Sedoroxsomething is messed up on the filesystem
21:36.35opus___yaaar - wrong cpu for kernel
21:36.45SleepyCowOkay. Another question. here in canada, are there any VOIP providers that work with asterix, that are cheap, and can get local 204 area code?
21:36.56opus___sleepycow chances are yes
21:36.57SleepyCowmy research says no (vonage doesnt work with asterisk right?)
21:36.57Sedoroxgrub has nothing to do with kernel and CPU
21:37.02yaaaropus___: i don't know...it's a bog-standard PII-550
21:37.08yaaarwait...make that III
21:37.17Blackveltried already to reinstall grub into MBR?
21:37.27opus___sedorox - the kernel won't load after stage 2 if its for pentium III and you run it on pentium II
21:37.53Sedoroxummmm... ookkk.. never heard of that
21:37.55opus___yaaar dunno then man maybe sedorox is right about the filesystem
21:38.01yaaarBlackvel: not yet. It has only done it once, and I just ctrl-alt-del and it's still sitting a "GRUB loading stage 2" still
21:38.14*** join/#asterisk JerJer[mobile] (~jj@feth100-fw.fament.net)
21:38.19SedoroxSleepyCow: in the wiki there are voip providers for North Amaerica, and have canada and US
21:38.29SleepyCowBeen there allready
21:38.32CoaxDIs there an option in voicemail.conf to ONLY email a NOTIFICATION of voicemail to the email address specified?  (i.e. not the attachment too.)  If not, I'll hack the source..
21:38.38Sedoroxthen how can you say no?
21:38.39yaaaropus___: like I said, the box is P-III ....the doc says you need at least a pII-300
21:38.43file[Digium]hey JerJEr
21:38.58file[Digium]Russell is scaring me, he wants to demonstrate festival
21:39.11SleepyCowAlso, seeing as I dont need any extensions or anything. Just straight voice mail on a few lines, is asterisk the right solution? perhaps we can use somehitng simpler or??
21:39.19CoaxDreason being, I wanna specify the email address of a cellphone so I get paged when there's voicemail..  But obviously, the attachment is gonna be extraneous..
21:39.32*** join/#asterisk Skysky (~Miranda@host6614613596.biz.tor.fcibroadband.com)
21:39.37file[Digium]bbl
21:39.39file[Digium]gotta setup stuff
21:39.46Blackveln8
21:40.11CoaxDduh. pager-email.  nevermind
21:40.19opus___sleepycow - yes esp if you are non-profit and don't want to buy a $5k system
21:43.47SleepyCowwhy would we spend $5k
21:43.58SleepyCowi figure all i need are 2 sip phones and perhaps some pots adapters
21:44.47SleepyCowor am I missing something
21:44.48*** join/#asterisk Darwin35 (~Darin@c-24-3-241-22.client.comcast.net)
21:45.11gr8nashhi all.. can anyone recomend a good VOIP company . ie broadvoice or nuphone. etc?
21:45.20gr8nashim in the US
21:46.22eKo1at&t
21:46.37*** join/#asterisk alegh (~ag11@OL12-112.fibertel.com.ar)
21:46.37jedaustinI was going to ask the same question.. which VOIP provider is the best solution for asterisk
21:47.08SleepyCowi would like to find any VOIP provider that will work with Asterisk that has (204) area code
21:47.09*** join/#asterisk zotz (~zotz@24.231.32.191)
21:47.10*** join/#asterisk harryvv (~plato@S010600055d210201.vs.shawcable.net)
21:47.12*** join/#asterisk miguellinux (~miguellin@200.47.223.190)
21:47.25eKo1Check the wiki people.
21:47.39SleepyCowI did
21:47.41*** join/#asterisk nel (~oeo@199.75.106.64)
21:47.55opus___I wonder if I could sit here all day and broker sip connections:)
21:48.13SleepyCowhey opus if you think you can sell me service be my guest :)
21:48.19Darwin35well broadvoice I am having issues with
21:48.26nelopus___:Hi again
21:48.41jedaustinI've heard that Nuphone offers IAX termination, but but other providers offer unlimited options
21:48.43opus___sleepycow - I use broadvoice.  Setup an account for $7 dollars and experiment
21:48.49opus___nel any success? :)
21:48.50gr8nashDarwin35 what problems?
21:48.52Darwin353 weeks now and no one from thier tech suppor has responded to my request for a phone call
21:49.00gr8nashahh
21:49.01nelI downloaded 1.3.1
21:49.03xkevcan I check out from cvs code from a certain point in time?  I started modifying crap without making a copy
21:49.03jedaustinDarwin35: Thats not good
21:49.03Darwin35calls not rining threw
21:49.07neland used vsftpd
21:49.11nelbut still having the same problem
21:49.14Darwin35dialing out and getting nothing but dead air
21:49.17nel"error saving application sip.ld
21:49.45nel0307082947|so   |4|01|---------- Initial log entry ----------
21:49.46nel0307082947|cfg  |4|01|Initial log entry.
21:49.46nel0307082947|cdp  |4|01|Initial log entry
21:49.46nel0307082947|cdp  |5|01|CDP is DISABLED. CDP not detected at boot.
21:49.46nel0307082947|cdp  |5|01|802.1Q/VLAN tagging is DISABLED.
21:49.46nel0307082947|so   |3|01|Target: Orion, Board=2345-11500-001 Rev=A, IP=172.16.1.21
21:49.48nel0307082947|so   |3|01|Target: Build=2.0.2 30-Apr-02 16:33
21:49.50nel0307082947|so   |3|01|Target: BootBlock=1.0.1 15-Jan-02 17:14
21:49.52nel0307082947|so   |3|01|Serial: Part number=2345-11500-001;Revision=1;
21:49.54nel0307082947|app1 |4|01|Initial log entry.
21:49.57Darwin35caller id not working  call waiting not working
21:49.58nel0307082947|app1 |4|01|Could not initialize resolver library with server 0.0.0.0 and domain .
21:50.00nel0307082948|so   |3|01|Link status is Net up, PC down.
21:50.02nel0307083003|app1 |4|01|Could not load time from (0.0.0.0) repeatedly.  Bad sign.
21:50.04nel0307083003|app1 |3|01|Bootline: eim(0,0)bootHost:flash e=172.16.1.21:ffffff00 h=172.16.1.6 g=172.16.1.1 u=polycom pw=password tn=CircaIP
21:50.06Darwin35nel
21:50.08nel0307083021|cfg  |3|01|Attempting to use 000000000000.cfg.
21:50.09nel0307083022|cfg  |4|01|File is 6089998, which is bigger than file system.!!
21:50.11nel0307083022|app1 |6|01|Error in saving application.
21:50.11jedaustinnel: I read theres a patch on http://www.broadvoice.com/support_install_asterisk.html
21:50.11Darwin35~pastebin
21:50.13jbotrumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
21:50.15jontowyikes man
21:50.20opus___nel - oh shit
21:50.26spacklemake it stop
21:50.43*** join/#asterisk MikeJ[Jayden] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net)
21:50.44opus___nel -- did you wipe out the entire directory in the ftp server before extracting 1.3.1
21:50.45opus___?
21:50.56nelmmm
21:51.00nelyes
21:51.06nelwell I have a backup folder
21:51.10nellet me wipe everything
21:51.11nelexcept
21:51.17nelthe files from the SIP 1.3.1
21:51.18SexyKenAnyone know why the Flash Operator Panel would show everyone as available when we're really on calls?
21:51.28opus___does the phone come up eventually?
21:51.30jontownot updating?
21:51.47SexyKen•jontow• But when I shut down op_server.pl it knows.
21:51.49Darwin35yes
21:51.54neljedaustin: I will try the patch
21:51.55tigger42could someone point me to a tutorial on how to setup asterisk as an isdn gateway using a isdn4linux card so all PCs with SIP software can call out using ISDN lines?
21:51.56jontowsexyken; look at the CLI and see if a user is logging into the manager interface when you restart op_server
21:51.59terrapenim not sure that i want to use a pulse-to-tone converter
21:52.01zykeany one using the port sipura ?   i can't make concurrent calls with it .. does it need particular setting in asterisk?
21:52.03terrapenbetween the phone and the IAXy
21:52.09zyke2 port sipura?
21:52.10terrapenbut i guess i won't have much of a choice
21:52.26SexyKenjontow - nope I dont
21:52.45Darwin35broadvoice is pissing me off.
21:52.50jontowok, thats a potential problem.. look at op_server.cfg
21:53.07SexyKenAh there it goes.
21:53.16*** join/#asterisk Skysky (~Miranda@host6614613596.biz.tor.fcibroadband.com)
21:53.20neljedaustin: I think the patch is not related to Polycom phones?
21:53.32SexyKenjontow - Do you know what the little arrow above my queues means?
21:53.36nelis there anyway to erase everything on the polycom phone?
21:53.36opus___nel - i didn't need a patch .. hmm
21:53.37jontowdouble click on it
21:53.40jontowits your queue data
21:53.41jontow:)
21:53.50nelmaybe there is not enough free space
21:53.52jontowyour phones should get it too once there is call history availble
21:54.02jontowas well as conferences and zaptel channels
21:54.21opus___nel - on the wiki they talk about the problem you are having.. but its about version 2.5 or something, thats why i asked if you wiped the directory
21:54.28jedaustinRight.. patch related to asterisk/Broadvoice
21:54.30MikeJ[Jayden]anyone doing call waiting?
21:55.01SexyKenjontow - Interesting I must have done something wrong then...cuz it shows no one in the queues.
21:55.07SexyKenAnd still no one on the phones
21:55.11SexyKenBut the manager is logged in.
21:55.14jontowweird :)
21:55.21opus___nel -- when I had that problem I tried 1.3.1 and it worked.  Eventually ( I think 10 minutes) it just booted.. also the polycom IP500 phone I have is actually a rebranded one from Shoreline called the sp500 I dunno if it has extra memory
21:55.34MikeJ[Jayden]I have not done anything with call waiting but think I want to with the analog phones at the house, just not sure quite how it works with the devices Ihave
21:55.37nelI see
21:55.41nel:(
21:55.54*** part/#asterisk nomercious (~nomerciou@HSE-Ottawa-ppp3493161.sympatico.ca)
21:55.57opus___umm
21:56.15opus___If you know vxworks you can telnet into the phone and initiate a reflash
21:56.19opus___:) dunno about that
21:56.29opus___same OS that fucked up the mars rover:)
21:56.39Darwin35http://pastebin.ca/6978
21:57.22nelI will try to telnet the phone
21:57.31wolfsonthat was a design issue not an problem in vxworks, they actually would likely have not fixed it, if it was not running vxworks
21:57.46Beirdoopus___: it wasn't the OS that fucked the rover
21:57.54wolfsonbad things happen when you run out of of hd and swap space
21:57.56Beirdoit was crappy software design
21:58.02Darwin35that extensions.conf should kepp you all happy major functionality
21:58.03Beirdopriority inversion
21:58.05jedaustinCan asterisk be set to call 3 numbers simultaneously and transfer an inbound call to the first that picks up?
21:58.08*** join/#asterisk wizhippo (~wizhippo@Quebec-HSE-ppp233869.qc.sympatico.ca)
21:58.22terrapenwouldn't it be cool if Asterisk could do Pulse ---> DTMF translation internally?
21:58.28terrapenwithout the use of special hardware?
21:58.35MikeJ[Jayden]jedaustin, yes, just use dial, and seperate with ,
21:58.39Darwin35it can I thought
21:58.46Darwin35I thought they fixed it
21:58.53terrapenhow?
21:59.08Darwin35should be in the wiki
21:59.16terrapenhttp://www.sandman.com/images/superdial.jpg
21:59.24terrapenthat damned thing is 50$US
21:59.27Darwin35it might need a tdm card
21:59.35terrapenhttp://www.voip-info.org/wiki-Dial+Pulse+to+Touchtone+DTMF+Converters
22:00.17MikeJ[Jayden]not , .... & sorry
22:00.33SexyKenjontow you here?
22:00.39terrapenit does dialing on a ZAP channel
22:00.54terrapenbut I don't think it can take an IAX2 channel and convert pulse to DTMF
22:00.58terrapenbut maybe i am wrong
22:00.59denonso what the heck still uses pulse anyway?
22:01.03terrapeni don't understand how it would work
22:01.07*** join/#asterisk xorol (~x@d51532D42.access.telenet.be)
22:01.08xorolello
22:01.09terrapendenon, me and my crazy-ass phones
22:01.14*** join/#asterisk ScythelX (Fleb@pc-24-181-176-72.sbi.ct.charter.com)
22:01.15xorolanyone using welltech 1501 or 3502 ?
22:01.18terrapendenon, i want to put a rotary payphone in my room
22:01.18denonold antiques or somethin?
22:01.28denonah yeah, figured that had to be the case
22:01.30Darwin35lol
22:01.37terrapeni like 1970s phones
22:01.41terrapenesp. Western Electric
22:01.54eKo1Pulse is cool. I remember dialing 911 using nothing but the hook button.
22:02.06Darwin35I have a batphone when I  pick itup it dial only 1 extension
22:02.24*** join/#asterisk Damin_Mobile (~pocketirc@3.sub-70-214-7.myvzw.com)
22:02.30ScythelXhello all I'm setting up voicemail to read messages from the database in blob format - I have the table setup now do I edit extconfig.conf to point asterisk to start using the database to store its information?
22:02.35nestArlol @ batphone
22:02.53nestArdo you keep a glass globe over it?
22:02.57Darwin35they have them for sale on ebay
22:03.13Darwin35its the redphone with a sticker that says batfone
22:03.18nestArlol
22:03.32terrapenhttp://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&rd=1&item=6158793968&ssPageName=STRK:MEWA:IT
22:03.37terrapenthinking about that for my bedroom
22:03.43terrapenbut that guy's shipping is a rip-off
22:03.50terrapenso i may go with a full-on payphone
22:04.15ScythelXthats sweet
22:04.23terrapenisn't it.
22:04.25xorolanyone using welltech 1501 or 3502 ? i got transfer problems
22:04.33Darwin35I am working on a touchscreenn phone with video
22:04.44terrapenand that jackass wants to sell you the mounting bracket, too
22:04.50terrapenwhich came with the damned phone
22:04.51ScythelXisnt the cisco 7970 touch screen?
22:05.21terrapeni *have* to have a bell stand for this
22:05.29Darwin35those blue phones are for callingcard and creditcard and collect calls only
22:05.37terrapenhttp://www.payphoneoutlet.com/9833.html
22:05.44terrapenthat's what i want in my room
22:05.51terrapendarwin: or Asterisk :)
22:05.52SexyKenAnyone here use Flash OPerator Panel?
22:06.04Darwin35yeah
22:06.23Darwin35I want a old British Pay Fone
22:06.33SexyKenWhen someone is on a call, shoulnd't it show that via the panel?
22:06.35terrapenthat damned pedestal is $268
22:06.37Darwin35and the call box they caame in
22:06.43KalD|Workanyone ever have an IAXY box just lock up after an hour?
22:06.45terrapenso, add $100 for the phone
22:07.05terrapenkald: i hope not. haven't tested mine that much yet
22:07.45Darwin351950's Princess Retro-style Telephone
22:07.45Darwin35Regular price: $89.95Sale price: $59.49
22:08.12KalD|Workterrapen, I was in a conference and the box just went dead - I pulled the network cable and the two lights on the cat5 jack were solid - and the off-hook like was on even after I unplugged the phone.  I had to pull power for 30 seconds to get it to come back
22:08.56terrapendamn.
22:10.03*** join/#asterisk TheBear (~brif8@lazyjtrainingcenter.com)
22:10.03xorolanyone using welltech 1501 or 3502 ?
22:10.09Darwin35ok thats it I am going to develop a sip pay phone
22:10.09xorolsrry
22:10.41Darwin35that can use creditcards calling cards   and cash
22:10.45TheBearok trying to install asterisk on gentoo kernel 2.6. On 'modprobe zaptel' I get in /var/log/messages  "zaptel: Unknown symbol crc_ccitt_table"  any ideas ?
22:10.52*** part/#asterisk davidl (reik@85-250-7-229.bb.netvision.net.il)
22:13.30terrapenwhich would be cooler in an apartment:
22:13.35terrapena regular payphone
22:13.39terrapenor a coinless payphone
22:13.40SleepyCowQuick Q: If three poeple on three phones want to make three seperate outgoing calls, all form the same location/network, I need 3 voip accounts, or just one?
22:13.41*** join/#asterisk Beave (~beave@vistech.org)
22:13.50SleepyCowterrapen: Regualr pay phone
22:14.02SleepyCowterrapen: Preferable rotary dial 3 slot
22:14.10terrapeni dont like the 3-slots
22:14.19SleepyCowDamn young whipersnapper
22:14.19terrapeni'm more of a fan of 1970s era
22:14.22SleepyCow;)
22:14.25Darwin35http://www.payphoneoutlet.com/19paclmopain.html
22:14.51terrapenhttp://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=985&item=6156299182&rd=1&ssPageName=WDVW
22:14.56eKo1terrapen: So you have an afro?
22:14.56*** join/#asterisk Martz (Martz_test@62.3.201.11)
22:14.57terrapensome lucky bastard go that
22:15.00SleepyCowSo anyone have an awesome answer to my basic question?
22:15.29terrapenWOW
22:15.30terrapenhttp://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=38038&item=6511839802&rd=1&ssPageName=WDVW
22:15.32terrapenlook at that
22:15.35terrapenlucky bastard
22:15.46Darwin35question whats your question you old fart of a sleepy cow
22:15.47SleepyCowthats a POS
22:15.50spackleSleepyCow, if you have IAX trunking you would only need one account.
22:16.09SleepyCowspackle, elaborate please - can I get that (cheaply) from a VOIP provider?
22:16.20*** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au)
22:16.20SleepyCowand run Asterisk at my end on our multipurpose linux box?
22:16.26ScythelXhello all I'm setting up voicemail to read messages from the database in blob format - I have the table setup now do I edit extconfig.conf to point asterisk to start using the database to store its information?  odbcstorage => odbc,mysql1,voicemessages isnt workin
22:17.17eKo1Are you using head?
22:17.21ScythelXyeah
22:17.38spackleSleepyCow, I use, Ahem, Nufone for ouot bound calls, and they allow trunking.  I can make multiple outbound calls on the same account.  I don't think it is very expensive, but that is subjective.
22:18.59jedaustinOther than nuphone, what voip providers allow trunking?
22:19.39SexyKenAnyone know why agent stuff wont work with flash operator panel?
22:20.11SleepyCowspackle: How about multiple inbound on the same number. Possible?
22:21.13spackleSleepycow, I think that depends on your provider.  I haven't gotten that far myself yet.  ;-)
22:21.18SleepyCowanyone know if voip.net is useable with Asterisk?
22:22.32nelthis polycom phone is frustrating me
22:22.42nelI have tried with all the SIP.zip files in http://www.freedomphones.net/polycom/files/
22:22.45nelnone of them works
22:23.56Darwin35asterisk did this to me
22:23.58Darwin35asterisk did this to me
22:24.19jedaustinDarwin35: download asterisk@home.. start over
22:24.31SexyKenAsterisk wont show agents logging into the queue what's up with that
22:25.48Darwin35asterisk@home doees not like my mini-itx board
22:25.55*** join/#asterisk MatsK (~NNSCRIPT@8.80-202-60.nextgentel.com)
22:26.01jedaustinIt's likely Linux
22:26.13gr8nashjedaustin i just installed @home and my x-lite client says "Discovered Port Restricted Cone NAT Firewall"
22:26.39gr8nashi dont have a firewall on my box.. and there is none between me and @home.. unless @home has one?
22:26.46jedaustingr8nash: are you NATing behind a router?
22:27.10gr8nashim behind one at work..but both boxes are on the same side of the firewall
22:27.18Darwin35I am just working on the fbsd port of * updateing the patches and the ports
22:28.22Darwin35and I have to go salvage a bad kernel now
22:28.35gr8nashNAT: 65.75.199.122  its grabbing my external address when all i want to do is connect to the local astrix box.. heh.. =/
22:29.00ScythelXfreebsd doesnt support zaptel yet does it?
22:29.03SexyKenDoesn't anyone know a damn thing about flash operator panel?
22:29.06Jer1326yep it do3es
22:29.16nelopus__: I used SoundPoint_IP_bootrom_2.3.0_w-SIP.zip
22:29.16ScythelXreally
22:29.20neland it seems it worked
22:29.22ScythelXeven ztdummy?
22:29.23nelis that enough???
22:29.25jedaustinwhat's the web site address for nuphone?
22:29.25Jer1326yes
22:29.31ScythelXthats awesome
22:29.36Jer1326i use ztdummy all the time
22:29.50ScythelXhow stable is it on freebsd
22:30.06ScythelXbecause I wanted to run it on freebsd a while back and couldnt because of the use of zaptel wasnt there
22:30.06*** join/#asterisk ruied (~a@85.138.10.212)
22:30.09Jer1326havent had it crash my box yet in 2 mths
22:30.20Darwin35jer1326 greetings
22:30.22ScythelXwell thats just awesome
22:30.24opus___nel - really
22:30.29Jer1326hiya darwin :)
22:30.32ScythelXdefinately switching it to my freebsd box
22:30.48Darwin35give me a day or 2 before you do
22:30.56Darwin35I am working on the ports tree ver
22:30.56Jer1326why?
22:31.07Darwin35updating to 1.0.6 and a few new patches
22:31.08Jer1326please dont make it havew a millon deps
22:31.17Jer1326but i need CVS :)
22:31.30*** part/#asterisk wizhippo (~wizhippo@Quebec-HSE-ppp233869.qc.sympatico.ca)
22:31.32ScythelXJer1326: are you running CVS head as well on the bsd box
22:31.33Darwin35why
22:31.39Jer1326yep
22:31.47Jer1326atxfer anit in 1.0.6
22:31.52ScythelXJer1326: that just made my day
22:32.01Darwin35then your installing like a linux box and not how we do on fbsd
22:32.02Jer1326i'm glad i can make someone happy :)
22:32.26Jer1326no i modded my make files to fit
22:32.34ScythelXhopefully it will run ok in a production environment
22:32.48spackleSexyKen, go read the FOP docs in the configs and on the web page, what you want to know is there.  getting upset isn't a good way to ask for help.
22:33.02Jer1326well this server handles 25 concun chans no problem
22:33.04Darwin35well I will make a head ver of the port
22:33.11Jer1326THANKYOU!!!! :))
22:33.15Darwin35but keeping it patched will be hard
22:33.20Jer1326yea...
22:33.45Darwin35do you use the patches we have in the asterisk port now
22:34.04gr8nashanyone know if @home comes installed with a firewall?
22:34.10Jer1326me? no. i roll my own
22:34.23Darwin35they need to also make asterisk-sounds a part of *
22:34.29*** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com)
22:34.44Darwin35tired of manualing moving files
22:34.48Darwin35but thats me
22:34.57xorolanyone using welltech 1501 or 3502 ? i got transfer problems
22:36.32Darwin35http://pastebin.ca/6978
22:36.45Darwin35there you go  jer go read
22:37.37*** part/#asterisk nel (~oeo@199.75.106.64)
22:39.05modulus_hi
22:41.07Darwin35they need the asterisk to make a asterisk-head.tar.gz
22:41.18Darwin35every night at midnight
22:41.24Darwin35and 6 am
22:41.29Darwin35and 12pm
22:41.38Darwin35and6 pm
22:41.56Darwin35that way you get the best head at that tiime asterisk has
22:42.05Darwin35sorry that sound s bad
22:42.21MikeJ[Jayden]darwin, do you have bandwidth?
22:42.34MikeJ[Jayden]set it up and post it.
22:42.36Darwin35no I am on dsl
22:43.06Darwin35or atleast once a day
22:43.19Darwin35its hard to make a pport pull cvs
22:46.29*** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu)
22:55.00*** join/#asterisk [Paul] (~paul@80.100.33.108)
22:55.16[Paul]wow
22:56.18*** join/#asterisk ruiner (ruiner@ruiner.netslacking.net)
22:58.32ruinerok, i'm a real newb here and don't want to sound dumb, but i'm going to anyway.  i've got an asterisk box setup, have a pots line connected to an fxo (i think) card in a cisco 3640, which is routing all calls to my asterisk box via sip.  i can call the number and asterisk picks up with the demo sound, but dialing numbers does nothing, i can't do the echo test by typing 600 and can't figure out why.  can someone point me in the right direction?
23:00.00*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
23:00.00*** mode/#asterisk [+o bkw_] by ChanServ
23:00.20[Paul]is your phone set to send DTMF tones?
23:00.21jontowwoi.. you've got a full blown cisco 3640 for a single POTS line/ :)
23:00.41jontowwhy not just a single $75 FXO card in the asterisk box?
23:01.01ruinerwork for an isp, we're just using hardware we have laying around to mess around with this stuff
23:01.04jontowaha :)
23:01.07jontowcool, then
23:01.11ruineras far as dtmf tones, as far as i know it is
23:01.13ruinerit's not pulse dialing
23:01.14jontowlearning experiences are good
23:01.27jontowso is dinner.. which is where i think i'm headed :D
23:01.30ruineryeah, we want to eventually roll out VoIP for broadband customers
23:01.49ruinerflat fee to dial any city we provide to
23:02.13ruinerheh, ok
23:05.17ruineris it possible my router is blocking those tones somehow?
23:06.39[Paul]i have no idea
23:06.43[Paul]newbe myself :)
23:07.26[Paul]i'm stuck on transfering calls
23:08.04Jer1326what is the problem with it?
23:08.10Jer1326paul:
23:08.52[Paul]it does work with internal extensions, but does not with an external isdn number
23:09.11jedaustinis nupone a us voip provider?
23:09.16jedaustiner nuphone
23:09.18[Paul]that's for unsupervised calls
23:09.31[Paul]- transfers
23:09.38*** join/#asterisk stustu (~stustu@fluffy.fatburen.org)
23:09.49[Paul]don't know how to initiate a supervised transfer at all
23:09.57Jer1326you need to be running cvs for that
23:10.02Nugget<-- requires supervision
23:10.21mikegrbYou so do.
23:10.34jedaustinWhat's supervised call transfer
23:10.45[Paul]i'm kind of a linux-newbe
23:10.45mikegrbwhen your boss is listening in
23:11.10[Paul]i've installed cvs with DSELECT
23:11.24Jer1326put t in your dialpan
23:11.28KalD|Workanyone have the proper sox arguments to convert .wav to .gsm?
23:11.28Jer1326dialplan
23:11.44Jer1326sox -r 8000 -c 1 file.wav file.gsm
23:11.59KalD|WorkJer1326, thx
23:12.30[Paul]and i have this annoying echo with SIP
23:12.37*** join/#asterisk justinnnn (~dsf@solid.mpa.net.au)
23:12.38justinnnnhey
23:12.44justinnnnanyone wana make some money helping me with txfax :) ?
23:12.56[Paul]but apparently there's nothing to do about that
23:13.35Jer1326what do you nede justinnnn
23:14.19[Paul]however i've hot asterisk version 1.0.5. that includes the supervised transfers, doesn't it?
23:14.33Jer1326no
23:14.41[Paul]:(
23:16.57*** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net)
23:17.18*** join/#asterisk redder86 (~lee@gateway.howardsilvan.com)
23:17.37*** join/#asterisk impressmenicely (impressmen@host-66-81-63-177.rev.o1.com)
23:18.18redder86Thanks for all the fish?
23:18.37mikegrbSo long.
23:18.47*** join/#asterisk techie (gus@asterisk.horizonte.us)
23:19.00*** part/#asterisk eKo1 (~bernd@207.42.191.67)
23:19.34Darwin35fish I will take to big mouth bass and a side of troout with lobster tail to finish it off
23:19.36[Paul]i'm doing a MAKE UPDATE
23:19.42[Paul]that's the idea?
23:20.00stustuA question about CVS and Asterisk versions: Are all the 1.0.x releases limited to bug fixes, and all new functionality in CVS?  Are there any know plans as for when a 1.1 or 2.0 release is up with more new stuff included?  How stable is the CVS verison?
23:20.07mikegrbTodays fish is trout a la crem.
23:20.30KalD|WorkJer1326, what can i do if it plays back at half the speed?
23:20.30Darwin35ok
23:20.58Jer1326KalD|Work let me check my syntax hang on
23:21.05develhey, is there any way to "clear" a SetGroup?  i did a 'SetGroup(${CALLERIDNUM})', and the first call works as expected, but just keeps incrementing, so fails every call after that (even after hangup)
23:21.58redder86stustu: in theory the 1.0.x releases are limited to bug fixes only
23:21.59KalD|WorkJer1326, I got it -
23:22.06Jer1326good
23:22.11KalD|WorkJer1326, put your args inbetween the file names
23:22.42redder86stustu: but "bugfix" is usually a very vague term around here.
23:22.48Jer1326that works too
23:22.53redder86stustu: one person's bug is another's feature
23:22.53*** join/#asterisk gezick (gezick@sartre.ispvip.biz)
23:23.01*** part/#asterisk gezick (gezick@sartre.ispvip.biz)
23:23.14*** join/#asterisk gezick (gezick@sartre.ispvip.biz)
23:23.25[Paul]i get the following message: cvs [update aborted]: connect to cvs.digium.com(66.225.202.81):2401 failed: Connection timed out
23:23.32redder86stustu: some people run CVS HEAD in production environments... so it's gotta be somewhat stable, although I'm sure you have to be more cautious than using CVS 1_0
23:23.57gezickcould someone point me to an explanation of sip uri's?
23:24.04gezicktrying to figure out what Registration from '<sip:192.168.0.61@192.168.0.61:5060>' failed for '192.168.0.79' means
23:24.04stusturedder86: Looking at the CVS logs, it seems that the branch was created about half a year ago...  I'm running FreeBSD, so I started out with the port (1.0.5 now).
23:25.37stustuThe FreeBSD port contains quite a few patches... But maybe the CVS HEAD does not need them any more?
23:26.06Darwin35I am working on that right now
23:26.13dwC-does anyone know what the best way is to go about having a inbound call from my pstn get a prompt to enter a CIDnum then another prompt to dialout to a IAX channel with that CIDnum set?
23:26.25Darwin35it neeeds a few of the patche like the patch:Makefile
23:26.35Darwin35to set the paths right
23:26.40Jer1326privacymgr
23:26.44Darwin35give me time guys
23:26.54Darwin35what abot it
23:27.06Darwin35jbot sex now
23:27.22Darwin35jbot is  gone
23:27.28Darwin35booo whooo
23:27.35sivana~seen normast
23:27.37jbotnormast <HydraIRC@Ottawa-HSE-ppp4119108.sympatico.ca> was last seen on IRC in channel #asterisk, 4d 17h 40m 28s ago, saying: 'juggie: Thanks keep it in mind.'.
23:27.50Darwin35~jobot sex
23:28.01Darwin35~jbot sex
23:28.03jbotI'm pregnant
23:28.07Darwin35hehhee
23:28.08mikegrbDarwin35: stop playing with the box
23:28.09redder86lovely
23:28.10Jer1326lol
23:28.45gezickwhat does chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 488a7d7843dccfd23f547998336f1644@192.168.0.61 for seqno 102 (Non-critical Request) mean?
23:28.51Darwin35patching head is not easy
23:29.16Darwin35where is kram
23:29.20tuxinator_linuxVON
23:29.29redder86gezick: asterisk gave up trying to place that call?
23:29.31Darwin35we need a daily tar.gz of cvs-head
23:29.42dwC-hrmm
23:29.45Darwin35so this port I am making will work
23:29.48stustuI tried typing "make" on a FreeBSD 5.3 system right now.  At least it did compile, but linking complains about "__use_ast_pthread_create_instead__" missing...
23:29.59gezickredder86: why is it trying to call it... is it because i put a defaultip entry in that extension?
23:30.05Darwin35there is a patch
23:30.11Darwin35its ion the mailing list
23:30.12*** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com)
23:30.18JerJer[mobile]don't use freebsd
23:30.25opus___gazick - i get that message like every second when no activity is going on as well, can't figure it out
23:30.28redder86gezick: I don't know
23:30.36Darwin35I am going to add the patch
23:30.41*** part/#asterisk newpers (newpers@ip24-56-8-180.ph.ph.cox.net)
23:30.50gezicki'm also getting Mar  7 17:30:06 NOTICE[10937]: chan_sip.c:7681 handle_request: Registration from '<sip:192.168.0.61@192.168.0.61:5060>' failed for '192.168.0.79'
23:30.52gezickall the time
23:30.54redder86gezick: I don't think that the IP address should worry you.
23:31.08Darwin35stop putting fbsd down I do alot to keep asterisk ported
23:31.12gezickwhere 192.168.0.61 is my asterisk server, and 192.168.0.79 is my polycom phone
23:31.25Jer1326i had no problem with compiling while it was patched
23:31.29Jer1326once
23:31.29redder86gezick: well, that's just saying that you have a context messed up in your sip.conf or the SIP client is misconfigured
23:31.45justinnnnany txfax ppls ?
23:31.53redder86txfax: ewe
23:32.05justinnnnwat do u use ?
23:32.10redder86hylafax
23:32.11gezickis the context before or after the @?
23:32.23redder86gezick: in the [brackets]
23:32.51opus___i got this message continously for ever
23:32.53opus___Mar  7 16:41:08 DEBUG[3529]: chan_sip.c:832 __sip_autodestruct: Auto destroying call 'AE6616BF-0097-4261-B58E-C4403B3CEEF5@192.168.1.101'
23:32.54ruinerah, I think I found my problem.  In case anyone else is interested, if your Cisco router is taking your calls and passing it to your asterisk box, you need to have a fairly recent version of IOS for DTMF tones to be sent through SIP
23:33.18Darwin35http://pastebin.ca/6983 there is the utils.c patch
23:33.23*** part/#asterisk redder86 (~lee@gateway.howardsilvan.com)
23:33.58*** join/#asterisk Dibbler (~Dibbler@snaddy.plus.com)
23:34.47opus___anybody have an idea what that means?
23:35.23Darwin35for fbsd
23:35.43opus___i'm on lamex
23:36.08`Sauronopus: If you think it's so lame, then quit using it.
23:36.11`SauronDuh
23:36.19opus___sometimes being lame is cool
23:37.49modulus_so you're like
23:37.52modulus_really REALLY cool?
23:38.04ruinerroffle
23:38.35opus___yeah, my refigurator runs the latest 2.6.11.1 kernel
23:38.46*** join/#asterisk laloo (~laloo@042.142-60-66.FTTH-SWI.surewest.net)
23:39.25stustuDarwin35: Thanks!  I got the latest CVS through the compiler now...  I probably still need to remake with a patched Makefile in order to have the paths right...  Would you know where to find a patch for that as well?
23:40.17*** join/#asterisk ManxPower (~eric@dsl-209-205-172-111.i-55.com)
23:40.23ManxPower~docs
23:40.24jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
23:40.52lalooGuys. Can someone help me? I just upgraded asterisk to version 1.0.6. I was running some old dev version earlier. Now, when I try to run asterisk, it comes back with an error, "/usr/lib/asterisk/modules/pbx_dundi.so: undefined symbol: pbx_substitute_variables_varshead"
23:40.59KalD|Workhow do you conditionally strip the last digit in a number?  i.e. if it is # discard it
23:41.21ManxPowerEvery day I hate MS Windows more.
23:41.35tuxinator_linuxManxPower: You too?
23:42.07ManxPowertuxinator_linux, Silly me, I thought I could save my roaming profile to a USB drive.  Unfortunatly my "roaming profile" is almost 200 megs.
23:42.09Nuggetthat's how I feel about linux.
23:42.23opus___anyone know of a cheap supplier of heatsets (mic/headphone jack based)?
23:42.25ManxPowerNugget, Heritic
23:42.43NuggetI'd probably feel that way about windows, too, but thankfully my job doesn't require me to use it.
23:42.48ManxPowerAnd I may have gotten my first spyware infestation
23:42.52tuxinator_linuxManxPower: Unneccisarily huge
23:42.52Nuggetoof
23:42.55KalD|WorkManxPower, but as MS would tell you - drive space is cheap - buy a bigger usb drive..  losers!
23:43.19Nuggetall I do with Windows is play Day of Defeat.
23:43.22Nuggetand run Quicken
23:43.24KalD|WorkManxPower, they dont get the point - just because you have 1TB doesn't mean fill it
23:43.35tuxinator_linuxKalD|Work: Cheap is more expensive than free
23:43.36ManxPowerI run windows on my laptop, that's all.
23:43.52tuxinator_linuxI am moving my laptops to linux
23:43.59tuxinator_linuxwell dual boot
23:44.13ManxPowertuxinator_linux, I could do that, but I'd get mad all the time because my /home dir was not sync'd.
23:44.32KalD|Worktuxinator_linux, exactly why I run linux =)   I can get 'cheap' copies of MS software but I still pay for it in terms of time and just pure screaming at the top of my lungs about how much I hate it over and over... or I can just run linux and be happy =)
23:45.23tuxinator_linuxI just think it is silly that a program with so many bugs can cost so much and be so popular
23:45.53tuxinator_linuxI do spend more time maintaining Windows machines that my linux ones
23:45.55Nuggetit's popular because it does what people want, unlike linux which does what developers want.
23:46.08tuxinator_linuxNugget: true...ist
23:46.11tuxinator_linuxish
23:46.40Nuggetmy beef with linux is that it suffers from trying to do everything -- it ends up doing all things in mediocrity and no things well.
23:46.44tuxinator_linuxYou should hear my wife yell at Word when she has to use it at school
23:46.45ruinerlinux can be a pain as well...especially when installing new hardware.  with windows (usually) it just works
23:46.49Nuggetit just isn't very good at things.
23:47.01ruinerbut i do prefer linux over windows because i have a lot more control over things
23:47.02Nuggetit's just adequate at doing just about everything
23:47.14ruinerbut i use windows so i can play games
23:47.26Darwin35?
23:47.28tuxinator_linuxNugget: good at being stable, and being a server, but you're right not a perfect desktop OS yet
23:47.33NuggetI disagree.
23:47.42Nuggetfor stability or servers there are much better solutions.
23:47.45*** part/#asterisk laloo (~laloo@042.142-60-66.FTTH-SWI.surewest.net)
23:48.07tuxinator_linuxruiner: I have not had any problews with installing hardware on linux, kudzu is nice
23:48.07Darwin35/usr/ports/net/asterisk/files
23:48.15Nuggetand much of the push to make linux more appropriate for desktop use are coming at the direct expense of stability and server tasks.
23:48.23Darwin35thats all the current fbsd patches
23:48.35ManxPowertuxinator_linux, You kow who P. T. Barnum was, right?
23:48.48tuxinator_linuxManxPower: sorry
23:49.11ManxPowertuxinator_linux, he was am american showman.  He is the one that said "A sucker is born every minute."
23:49.28tuxinator_linuxManxPower: Sounds like a cool guy
23:49.33dwC-is there any way to use DISA but without a dialtone, just dead air while waiting for user input?
23:50.09stustuDarwin35: Ok, I guess it will be possible to apply patch-Makefile to the CVS head Makefile.  I thought that maybe a diff relative to a newer Makefile is floating around somewhere...
23:50.14ManxPowerHe is also the guy that opened a "freak show" type of traveling museum.  It was so popular that people were not leaving quickly enough, so he put up a sign "This Way To The Egress", people followed the sign, found themselves outside and had to pay again to get back in.
23:50.32tuxinator_linuxthat's funny
23:51.31ManxPowertuxinator_linux, *nod*  He's one of my minor heros.
23:51.35Darwin35I just use th eports and now help maintain it
23:51.53Darwin35it all works execpt for I have to update bri now
23:53.33*** part/#asterisk impressmenicely (impressmen@host-66-81-63-177.rev.o1.com)
23:53.46ruinerlater gents and gentettes
23:55.17stustuI really have a hard time finding out what's in the bristuff patch.  Can I read about that somewhere (except examining the patch itself?)
23:55.24tuxinator_linuxManxPower: http://home.nycap.rr.com/useless/barnum/
23:56.25gr8nashanyone have a page for VOIP providers.. i must have missed it on the wikki
23:56.30gr8nashim in the US
23:57.01ManxPowertuxinator_linux, Cool.
23:57.21Lethargicclowngr8nash, broadvoice looks good to me, but i would wait for an answer from someone more experienced then myself
23:57.30SleepyCowHey guys. Does one get better sound quality from a PCI FXS card (Like the wild 4 port) than from an ethernet to pots bridge?
23:57.51mishehuSleepyCow: Yes.
23:58.00mishehuSleepyCow: No.
23:58.05gr8nashLethargicclown they looked ok to me to..but alot of people have not-so-nice of things to say about them
23:58.07SleepyCowmu?
23:58.10mishehuSleepyCow: I guess the answer is "maybe"
23:58.16mishehuthe kow says "mu"
23:58.23SleepyCowcan you elaboratE?
23:58.31SleepyCowCows say moo, kow says 'mu'
23:58.40opus___yes
23:58.41opus___no
23:58.42opus___:)
23:58.51modulus_jbot dogcow?
23:58.52jbotMOOOFF!!
23:59.19opus___"Yeah, we're actually ... we don't have any passengers on board, so we decided to have a little fun and come up here,"
23:59.22opus___great idea
23:59.24SleepyCowSeriously, is there any advantage to using the 4 port fxs card right in the asterix box versus external pots to ethernet bridges?
23:59.39JerJer[mobile]yes
23:59.42mishehujbot mog
23:59.43SleepyCow?

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