irclog2html for #asterisk on 20051228

00:00.07[av]baniand they never heard of ramdisks either
00:01.24warthogas far as solid state goes, what do you guys think of this http://linuxdevices.com/articles/AT8596095318.html
00:01.26ManxPowersvn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2 would get 1.2.1 + patches that will go into 1.2.2?
00:02.05[av]baniwtf, $150?
00:02.34warthoglooks interesting eh!
00:02.56[av]bani$150 for a $8 ide-cf adaptor with asterisk and debian woody...
00:03.09_Sam--i would pay 150 if it would load a full debian
00:03.18_Sam--and let you add packages using apt-get and stuff
00:03.20[av]bani64mb flash, heh
00:03.25[av]banithats a nice profit
00:03.51warthogthat article was from quite some time ago, I am sure it is bigger now
00:03.52_Sam--yeah if they put like 5gb on there it would probably sell
00:05.08_Sam--they should make an external USB debian that you can just go buy at like best buy...buy a 1gb USB memory stick, through it in your USB port, and run debian when you want
00:05.41meredyddbeen done, _Sam--
00:05.50_Sam--i know you can do it...they should SELL one
00:05.55meredyddCan't find the link, tho...think it got /.ed
00:07.07_Sam--or better yet..someone should package debian and asterisk on a USB stick and sell it as a package
00:08.03warthogmeredydd, with all your kernel debuging experience, you must know something about asterisk manager sockets
00:08.06rikstahey there. If i have a very simple dial plan with 5 numbers, each from one zap span, connecting to a single AGI, and logging to mysql cdr db, would i likely encounter anything to change on upgrading from 1.0.9 to 1.2.1? thanks a lot
00:08.26Strom_Criksta: probably not
00:08.39_Sam--change your modules (upgrade asterisk-addons) for the cdr mysql
00:08.40rikstaStrom_C, do you know of a list of changes anywhere
00:08.45juanjocDoes anyone here have any knowledge of how the MixMonitor app works?
00:08.59riksta_Sam--, yeah, would that be all?
00:09.05Strom_Criksta: asterisk.org?
00:09.14rikstaStrom_C, i cant find anything
00:09.25_Sam--maybe some changes in extensions.conf for set digit timeouts and stuff
00:09.31_Sam--the console will complain and tell you
00:09.47rikstaok thanks, i'm just a bit scared cuz it's a live system :P
00:10.06_Sam--i just did it recently...was kind of scared as well, but went without any troubles
00:10.22_Sam--make/install zap,libpri,asterisk
00:10.25riksta_Sam--, i don't always have such luck :P
00:10.43_Sam--just make sure to backup everything (like your current working binary and configs)
00:11.28rikstayeah i can probably do that easy with gentoo
00:11.34_Sam--the only other thing...dont know what your AGI does/contains...but there were some caller ID changes that we had to change around some of our AGIs
00:12.11rikstaim using asterisk-java and that is supporting both versions
00:13.07*** join/#asterisk Kyreeth (n=ashley@aquila.feathers.net)
00:13.11_Sam--what do you do with asterisk-java?  you write apps in java that interact with *?
00:13.17rikstayep
00:13.22rikstalike IVRs
00:13.26_Sam--why in java?
00:13.48rikstabecause i started writing ADM : http://adm.hamnett.org that i wanted x-platform
00:14.07_Sam--like give me an example of something that you do in java that interacts with *
00:14.35riksta_Sam--, anything that any other AGI script can do
00:14.50_Sam--so instead of doing it in say, php, you use java
00:15.10rikstayeah, i can do what i like can't I :P
00:15.25_Sam--hell yeah i m just trying to understand!
00:15.33_Sam--i thought maybe you had a cool web based java client for * :)
00:15.42*** join/#asterisk TheCops (i=nobody@got.securebinary.com)
00:15.43rikstai do, kinda
00:15.45rikstaand a GUI
00:15.47rikstalook at the link
00:15.51_Sam--looks pretty
00:15.59rikstaits much nicer now need to update screenshots
00:16.13_Sam--i think you could do it in 1/2 the code in php :P
00:16.19rikstaanyway so yeah thats why i use asterisk-java, because it has a nice API into the manager interface of asterisk
00:16.25riksta_Sam--, i bet you can't
00:16.54_Sam--so i understand...you are using that java frontend to input/retrieve values from an SQL table?
00:17.03NewSolecan some one tell me whats going on here...
00:17.06NewSole<PROTECTED>
00:17.06NewSoleDec 27 19:09:48 NOTICE[3891]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
00:17.20riksta_Sam--, no :P
00:17.42asteriskmonkeyverything that could possibly go wrong
00:17.42asteriskmonkey<[av]bani> drivers, irqs, echo, not answering
00:17.42asteriskmonkey<[av]bani> crashes
00:17.42asteriskmonkey<[av]bani> hangs
00:17.51asteriskmonkeydoh :P
00:17.53riksta_Sam--, for this perticular box i use the java api to manage an IVR that is linked into another legacy system to provide automated information via IVR
00:18.19asteriskmonkeyDoes anyone had any issues using iax connections where a number dials and rings then drops and goes busy?
00:18.23_Sam--alot of the screenshots are of a java softclient?
00:18.36riksta_Sam--, yes the other stuff is private work
00:18.39Strom_Casteriskmonkey: busy signal or reorder tone?>
00:18.51rikstathey both use the java api
00:18.56*** join/#asterisk jeffik (n=jeffik@CPE0050babf4cd5-CM014350000760.cpe.net.cable.rogers.com)
00:19.00asteriskmonkeyStrom_C: busy signal
00:19.11asteriskmonkeyi am using link2voip as a long distance terminator :P
00:19.28warthogrisksta, have you made socket connections to the manager from java, if so have you figured out how to watch for Event: Newchannel.* type information as you can with perls telnet?
00:19.31_Sam--riskta:  so what you're working on would be like a combination of AMP plus a soft client?
00:19.39*** join/#asterisk ryan___ (n=ryan@londonderry-cuda2-68-171-162-161.lndnnh.adelphia.net)
00:19.50asteriskmonkeyi always seem to get a dialing then at the point something is suppost to go to answering machine it craps out to a busy signal
00:20.05rikstawarthog, asterisk-java.org it's all documented, it makes it very easy to look for new channel events look at the sample code
00:20.26Strom_Casteriskmonkey: are you sure it's a busy signal and not a reorder tone?  (reorder tone is sometimes called "fast busy")
00:20.40rikstawarthog, i also have done that stuff in perl code parsing the telnet output, but it's very messy
00:20.47warthogcool, I am trying to figure out how to do that in perl with sockets, but have not found documentation yet, perhaps I can get clues from other languages.
00:20.48[av]banihm, only one review on the cg-410 and it wasnt good
00:20.58rikstawarthog, it's simple man
00:21.01asteriskmonkeyStrom_C: yes its a vust signal.. sometimes no busy signal just cuts right off to a hangup
00:21.10rikstawarthog, just use something like Net::Telnet
00:21.29Strom_Cweird
00:21.36Strom_Cdoes the same thing happen with a different IAX carrier?
00:21.43rikstawarthog, the api is on voip-info.org for the commands you need to log into asterisk manager api
00:21.52rikstaand turn on event watching etc
00:22.14warthogis it no specific to telnet, the watchfor command?
00:22.19rikstagod knows why you would do all this manually rather than using an api like asterisk-java or whatever other APIs there are in different langs, it's all been done before
00:23.03warthogI am just getting into this so I am not sure of all my options, is perl sockets not a good way to go?
00:23.16rikstano, net::telnet
00:23.20argusassuming the permissions are correct on the /var/run directoies i should be able to 'asterisk -r' from any user in the the asterisk group correct?
00:23.38warthogok, I steer that way then.
00:23.44warthogthanks
00:23.52rikstawarthog, why are you doing this, there is already a perl asterisk api
00:24.01rikstahttp://www.voip-info.org/wiki/view/Asterisk+perl+library
00:24.21asteriskmonkeyStrom_C: havnt tried any other 3rd party carriers yet
00:24.46warthogcause I am initiating the connection from a cron, I have been told that AGI and perl will only work if asterisk initiated the script, is that true?
00:25.08rikstano, you can use the manager interface of asterisk
00:25.17rikstato watch for events
00:25.27rikstago to the link i just sent you
00:25.37warthogrisksta, cool reading now...
00:26.02[av]banihmm
00:26.03[av]banihttp://cgi.ebay.com/6-Ports-FXO-VOIP-SIP-PSTN-Gateway-Asterisk-IP-PBX_W0QQitemZ5845021296QQcategoryZ61839QQssPageNameZWDVWQQrdZ1QQcmdZViewItem
00:26.06[av]banilooks risky
00:28.01warthogrisksta, a point of confusion on my part, I have tried to initiate dial commands with this before and nothing happens, I assumed this was cause asterisk did not call my script, I did from the command line and there seems to be no way to authenticate to asterisk using Asterisk::AGI, I am in left field here?
00:29.00NewSolecan anyone help with sip problem
00:29.21argusNewSole: most likley
00:30.09NewSoleI am getting Rejected from a host because that dam asterisk is trying to do a sip keep alive
00:30.09TheCopsSomeone is using page feature and Snom 320 phone ?
00:30.21warthogrisksta, I guess that is what Originate is for then, hmmm
00:31.00Strom_Cdo any of you know offhand if there's an extensions.conf function that sends a progress message down the ISDN d-channel?
00:32.01Kattyhi lads!
00:33.06argusNewSole: have you disabled the keep alive
00:33.29*** part/#asterisk Utah_Dave (n=boucha@0-2pool130-155.nas28.salt-lake-city1.ut.us.da.qwest.net)
00:33.49Kattywell don't say hi all at once ;)
00:33.50trixterGoogle Sued Over VoIP  http://news.tmcnet.com/news/-google-voip-rates-technology-/2005/dec/1242631.htm
00:34.03NewSoleyes
00:34.40NewSolekeep getting
00:34.41NewSolemcitrunk                   209.47.231.68        N      5060     UNREACHABLE
00:35.31*** join/#asterisk robl^ (n=robl@dsl093-025-118.hou1.dsl.speakeasy.net)
00:36.45*** join/#asterisk pigpen2 (n=mark@fw.seamans.cc)
00:37.40*** join/#asterisk _DAW-LAPTOP (n=DAW-LAPT@adsl-156-94-42.msy.bellsouth.net)
00:37.50argusNewSole: i'm not that good w/ external sip gateway stuff, but it almost looks like its being cut due to a NAT/Routing problem
00:38.37warthoghi katty
00:39.14*** join/#asterisk perlmonkee (i=oginvu@c-24-22-125-15.hsd1.or.comcast.net)
00:40.00perlmonkeeCan someone tell me how I allow SIP registration from hosts outside my LAN?
00:40.54perlmonkee(allow device [whatever] to register with username/pass from *any* host)
00:41.04NewSoleno I was told by provider system was blocking me cause it was sending data with out calls
00:41.07argusperlmonkee: host=dynamic ; you might have to do other routing stuff
00:41.25perlmonkeeI have host=dynamic - it still wont let me register.
00:41.39argusperlmonkee: whats the error
00:42.30[av]banihttp://www.voiptalk.org/products/Mediatrix+1204+FXO+Gateway_review_104.html
00:42.37[av]banimediatrix 1204 seems out too :(
00:42.42MeHas anyone had a problem where you can dial an extension from a local client but cannot dial that extension from a DID?
00:42.55robl^perlmonkee: then its prolly firewall / NAT relatd
00:43.12MeI am having a problem where I cannot dial ext 300 when I call in on my DID but I can dial it on a local client
00:43.15perlmonkeeargus: I get no output from Asterisk directly - even with sip debug on
00:43.28TheCopsSomeone is using page feature and Snom phone ?
00:43.41perlmonkeeI can register from the current host I'm at to SipPhone.com just fine (I'm using the SJPhone client)
00:43.42perlmonkeebut I can
00:43.46perlmonkeet register to my Asterisk box.
00:43.53perlmonkeewhich is not behind a firewall at all.
00:44.19perlmonkee(Here, where I can connect to SipPhone.com - SJPhone says I am behind a "restricted cone nat")
00:45.14*** join/#asterisk Seggy (i=rbutler@tsss.org)
00:45.26[av]banihttp://www.opensubscriber.com/message/asterisk-users@lists.digium.com/2857831.html
00:45.29[av]banibleahhhh
00:45.30argusperlmonkee: if your not getting a registration error ( or anything from sip debug ) it's not getting to the asterisk box
00:46.06[av]baniman, it almost seems less hassle to buy a pile of spa-3000 for fxo :)
00:46.36perlmonkeeargus: Can you think of any reason I would be able to connect to SipPhone.com, but not my Asterisk box?
00:46.42_DAW-LAPTOPMe - pastebin your conf
00:46.49perlmonkeewhen I have no firewall in front of the asterisk box.
00:47.30perlmonkeeare there other settings I can change to make sure I'm getting all debugging information?
00:47.46Meexten =>300,1,Answer()
00:47.46Meexten =>300,2,Dial(SIP/011747*69889806090139214@proxy01.sipphone.com,20,r)
00:48.07argusperlmonkee: not really, i'd run tcpdump on that host and see if your getting anything
00:48.14MeDAW is that enough?
00:48.36Methat is under [default]
00:51.59perlmonkeeokay - I'm getting data through
00:52.04perlmonkeeI was reading some of this wrong before.
00:52.25perlmonkeeI can now see which parts from sip debug have something to do with this connection attempt.
00:53.58perlmonkeeThe problem appears to be with this host telling Asterisk its INTERNAL network IP where I am now.
00:54.34argusperlmonkee: do you have nat=yes on both ends
00:56.15_DAW-LAPTOPMe - check your private message..
00:56.23MeI did
00:56.30MeDid you not get my responses?
00:56.42_DAW-LAPTOPnope
00:56.44Meweird
00:57.11*** join/#asterisk _DAW-LAPTOP (n=DAW-LAPT@adsl-156-94-42.msy.bellsouth.net)
00:57.22perlmonkeeargus: D'Oh!
00:57.33MeI just tried to send you one now
00:57.35perlmonkeea very sheepish thankyou.
00:57.38Meafter your restart
00:57.41*** join/#asterisk javar (n=javar@69.79.133.185)
00:57.44perlmonkeeI hate it when I make stupid mistakes like that.
00:57.52argusperlmonkee: no prob, i have done it before :)
00:58.33MeDAW - do you use Yahoo Instant Messenger?
01:00.30_DAW-LAPTOPnah.. did you register with nickserv? If not I dont think you can pm...
01:00.46Meok
01:01.10*** join/#asterisk Me (n=icechat5@user-0ce2dhc.cable.mindspring.com)
01:01.27[av]baniblah. every external 4port sip fxo i've found has got scathingly bad reviews
01:01.38*** part/#asterisk javar (n=javar@69.79.133.185)
01:01.52*** join/#asterisk javar (n=javar@69.79.133.185)
01:01.55MeDAW > you get my pm that time?
01:02.04*** join/#asterisk sac|h0p (n=h0p@S01060002b3eb8fa7.ok.shawcable.net)
01:02.13_DAW-LAPTOPno
01:02.47argusassuming the permissions are correct on the /var/run directoies i should be able to 'asterisk -r' from any user in the the asterisk group correct?
01:02.59MeDo you use another IM program?
01:03.05MeI have to get running.
01:03.11MeI have to leave now
01:03.16Mebut can I contact you later DAW?
01:03.35MeI thank you for trying.  I will have to register with the nickserver when I have the chance in a couple hours
01:03.42MeI have a meeting to get to
01:03.51_DAW-LAPTOPok
01:04.19MeYour help is much appreciated
01:04.33*** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com)
01:11.55*** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com)
01:13.49TheCopsI receiv this error: -- Incoming call: Got SIP response 489 "Bad Event" back from 192.168.50.103 when I added a Destination in on my snom phone, someone know why ?
01:26.10*** join/#asterisk warthog (n=nvadekar@66.55.112.240.ppp.northrock.bm)
01:26.43warthoganyone here really good at callfiles, when I copy my x.call into the outgoing dir, is disappears as it should but nothing happens in the asterisk console at all, no errors, no dialing, nothing, yet it I telnet to localhost 5038 and action a call, it works fine.
01:26.59warthogahem, I am doing a move, not a copy....
01:29.07trixtertry increasing your logging..  in logger.conf look for the 'full' entry..  that will let you see at least if there is anything else going on
01:29.18trixterit may be a permission issue where it cant read the file, it may be that the file contains invalid data
01:29.33warthogthanks, checking...
01:30.42warthoghmmm, full is already enabled.
01:32.47warthognothing in both syslog or asterisk -rvvvvv
01:33.07trixterfull should be logging into /var/log/asterisk/full or whever asterisk normally logs to
01:33.19trixterand you see nothing?  what are the perms of the file that gets moved?
01:33.32trixterand do you move it on the same partition?
01:33.45trixtera mv across different partitions is the same thing as cp file; rm file
01:33.46*** join/#asterisk MasterObi-WanK (i=MasterOb@200.122.159.228)
01:34.19Krillhi
01:34.29MasterObi-WanKhello
01:34.38MasterObi-WanKanyonw knows how to use Asterisk At Home 2.2 ?
01:35.07SkramXWhat about it, " MasterObi-WanK "
01:35.36MasterObi-WanKshido6, hello
01:35.37loudis that som ething you boot from a cd
01:36.29MasterObi-WanKSkramX, I have a digium tdm400 with 4 fxo , but if the incoming call comes in the zap1 it will ring good, but if it comes on the zap2 channel, it will ring both zap channels
01:36.30*** join/#asterisk Zach^^ (n=Zachary@65.121.244.130)
01:36.38harryvvSeems more and more companies go with some commercial voip system then asterisk.
01:36.49Zach^^how do i setup askerisk to display Name and Number on inbound caller id?
01:36.51harryvvat least around here.
01:37.39[hC]Anyone used 'shared line appearance' on any sipura/linksys phone? most notably the spa941?
01:37.40*** join/#asterisk licued (i=licucude@ool-44c784a0.dyn.optonline.net)
01:37.45[hC]Im trying to figure out what it actually means
01:38.58warthogtrixter, I assume you need to be asterisk:asterisk, it is a perm issue, of course when I move it, it changes to root, oops, I guess I have to move it as asterisk?
01:39.04[hC]and i didnt think asterisk handled shared line appearances
01:39.05Luke-JrWhy doesn't Asterisk run with realtime priority?
01:40.12MasterObi-WanKI have a digium tdm400 with 4 fxo , but if the incoming call comes in the zap1 it will ring good, but if it comes on the zap2 channel, it will ring both zap channels
01:40.26trixterwarthog: are you moving it across different partitions?
01:40.37trixtermv should preserve ownership and permissions by default
01:41.07_Sam--hmmm this is exactly what i was working on...someone beat me to it...NEW VERSION Asterisk Live CD is Released. AstBill Live CD contains AstBill-0.9.0.14, Asterisk 1.2 and MySQL 5.0.16 and is based on DSL and Knoppix.
01:41.18_Sam--No installation. Just boot from the CD.
01:41.28trixterthere are a few live CDs for asterisk
01:41.48harryvvsam, so what are you trying to say
01:41.48_Sam--trixter:  would that cd run entirely from cd, or it needs an HD?
01:41.49trixterastlinux.org has their own (27M is all it needs), knopsterisk.org is a knoppix based live CD
01:41.51trixteretc
01:42.05trixterI dont know about astbill but the others I have seen run just off the CD
01:42.28_Sam--nice i guess that saves me alot of work
01:42.31trixterin janurary I am doing a presentation on using astlinux so I have been working with that since the goal is to have everyone install something and configure it
01:42.43trixterastlinux also has a vmware image for the free vmware player
01:42.53trixterso you can play, although performance will be an issue since the timing clock will drift
01:43.10harryvvtrixster presentation to students?
01:43.15trixterbut it lets you show it off and see if its something you want..  the vmware player with astlinux wants 64M of ram so its not that bad
01:43.29trixterharryvv: sacramento asterisk users group
01:43.53trixterwant to get everyone to a certain level so we can have contests and give away the prizes that were donated by digium and thevoipconnection.com
01:44.15trixterwe have 5 grandstream ATAs, 3 gxp2000s and 1 tdm410p for prizes -- not bad :)
01:44.17MasterObi-WanKWhats the best ISO for an SOHO office , with digium 4 fxo, and several VOIP 800 numbers in iax and sip ?
01:44.20Luke-JrWhy doesn't Asterisk run with realtime priority? I have a DV capture program that does this and it's *very* effective in preventing dropped frames.
01:44.47_Sam--trixter:  do you have any opinion about which live cds may be better than others?
01:45.11trixterI have only used A@H for 5 minutes and astlinux for much longer
01:45.15trixterbetween the two I prefer astlinux
01:45.26ManxPowerLuke-Jr, it's supposed to if you run safe_asterism
01:45.27MasterObi-WanKwhats the url ?
01:45.31ManxPowersafe_asterisk
01:45.34trixterthere is a debian liveCD I read about but never used that I would prefer to A@H
01:45.50[av]baniluke, i have a dv capture program and i never used realtime priority, never dropped frames
01:45.54MasterObi-WanKtrixter: whats the url of astlinux ?
01:45.54Luke-JrManxPower: what's that?
01:46.03Luke-Jr[av]bani: sounds like a non-busy system ;)
01:46.05MasterObi-WanKDebian cd is Xorcom
01:46.09[av]banino it's plenty busy
01:46.15ManxPowerLuke-Jr, the script that you should use to start asterisk.
01:46.19Luke-Jr[av]bani: my system never has 100% idle
01:46.21[av]baniwatching vids while capturing, compiling in background
01:46.30Luke-JrManxPower: I use initscripts o.o
01:46.32[av]banino problem, video didnt skip either
01:46.57ManxPowerit's usually called by "service asterisk start" or whatever your linux uses to start and stop services, when you run "make config"
01:47.12Luke-Jr[av]bani: ok, let me explain... there are often times my system completely freezes for a period of seconds/minutes
01:47.40[av]baniwhat is it, a p200mmx or something?
01:47.58ManxPowerLuke-Jr, that is not common.
01:48.10[av]banionly other reason would be swapping, due to inadequate ram
01:48.13Luke-JrManxPower: so what part of the script is supposed to do it?
01:48.20Luke-JrOS: GNU/Linux 2.6.14-gentoo-r2-ljr/x86_64 - CPU: 1 x AMD Athlon(tm) 64 Processor 3200+ (2202.918 MHz) - Processes: 336 - Uptime: 22d 20h 38m - Users: 62 - Load Average: 0.96 - Memory Usage: 3387MB/1003MB (337%)
01:48.33ManxPowerLuke-Jr, it's some command line parameter, "man asterisk"
01:48.37warthogtrixter, thanks, that was it
01:48.43[av]baniif your system completely freezes for seconds/minutes, then your system is broken somehow
01:49.01Luke-Jr[av]bani: likely the swapping
01:49.03Zach^^anyone know how i can get the name shown on the callerid for inbound VOIP calls??
01:49.15trixterwarthog: what?  cross partition move?
01:49.21[av]banirealtime wont save you for swapping, since that causes io contention
01:49.52Luke-Jr[av]bani: no, but it helps
01:50.06[av]banisince realtime can't override kernel
01:50.12[av]banionly other userspace apps
01:50.25[av]baniyou're better off adding another couple gigs of ram imo
01:50.34Luke-Jrcan't afford it
01:50.44[av]banioh well :)
01:50.51Luke-Jr:(
01:50.56[av]banidont run so much shit?
01:50.59warthogtrixter, I never expected asterisk to be able to delete a file that had the wrong ownership, so when the file was deleted, I never even though to look at that and stupidly I was looking at syslog and forgot to check the full log.   it was just wrong ownership, not cross partition
01:51.16Luke-Jr[av]bani: can't afford a seperate desktop box (= so much)
01:51.39warthogtrixter, since I have never done callfiles before, I was not sure were it could be, now it seems so obvious
01:51.56trixterahh..  yeah my stuff will either run as the asterisk user or chown asterisk.asterisk prior to the mv so it can be dealt with..  if asterisk can read it generally it will process it and just warn that it cant set utime on the file
01:52.05[av]banigentoo.. why am i not suprised :)
01:53.49warthogtrixter, in terms of utime, is that a method of call scheduling?
01:53.56_Sam--trixter:  if you rn astlinux  entirely from rom (no hd)...how does it get its initial config, or know what hosts to use for SQL realtime?
01:54.23trixterwarthog: its a file attribute
01:54.25[av]banifloppy!
01:54.42_Sam--for example, if you burn the astlinux cdrom, and boot and run from the cdrom...how does it get any config info?
01:54.53trixter_Sam--: it sees what hardware you have and all, it will use a default config set unless you save that stuff to some other media
01:55.02trixterit has provisions to save and restore the configuration
01:55.15ManxPower_Sam--, I assume from a USB key or floppy
01:55.22Luke-Jrhrm... asterisk's realtime is only -11? why not -99? =p
01:55.33trixterits web gui isnt that bad either, one its https not just http so that is good, two it lets you edit the files directly rather than obfuscating everything so it does let people learn easier
01:55.35trixterI think anyway
01:55.52[av]banibecause once you go realtime, rt priority is only vs other rt priority apps
01:55.56Luke-Jrpfft guis...
01:56.05_Sam--i know which files i need to edit...ive edited them enough times.
01:56.05[av]baniyou're already scheduled on top of everything else
01:56.08trixterits basically a text editor via the web
01:56.35trixterphpconfig style, its fairly basic and lets people navigate something they already know how to use, that way you dont have to teach vi or something in addition to whatever else
01:56.38Luke-Jr[av]bani: any way to verify it's RT other than -99?
01:56.44trixterfrom a classroom perspective its a good thing
01:57.46[av]banithere might be a version of top or ps which shows rt scheduling
01:57.51[av]banimight be in proc too...
01:59.03*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)
01:59.39[av]banikpm shows scheduling policy
02:09.14warthogtrixter, is there a way to get asterisk in the dialplan priorities to wait for a specific tone, then jump to the next priority, I.E. call a pager, wait for the call to be answered, wait for a the tone indicating you can now leave a numeric message, then jump to the next priority and leave that dmtf message?
02:12.52trixteryes but it requires coding on your part
02:12.57trixterafaik no one has done that
02:13.01trixterbleh
02:16.35SkramX<PROTECTED>
02:16.42SkramXwhats up?
02:20.49Kattymew!
02:21.37*** join/#asterisk Atebo1 (n=ugob@modemcable002.152-81-70.mc.videotron.ca)
02:23.14Atebo1Hi guys,  I've got an * box here, a Sipura SPA-3000 and a 841... I manage to make the voicemail work and calls between extensions as well... I just don't really know where to start to use the spa-3000 for a PSTN gateway...
02:24.04Atebo1in fact someone helped me and created a sip.conf and extensions.conf files, but they don't really work as expected
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02:28.10AteboyHi guys,  I've got an * box here, a Sipura SPA-3000 and a 841... I manage to make the voicemail work and calls between extensions as well... I just don't really know where to start to use the spa-3000 for a PSTN gateway..
02:28.36QwellAteboy: ask every 5 minutes...great way to get help
02:28.48Ateboyoups....  sorry
02:29.04AteboyI tought it didn't get through...
02:29.58coppicethe asking every 10 seconds approach is even more effective
02:30.47Ateboysorry, sorry, sorry...
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02:32.53thazzaGreat guys.. Now AteBoy is never going to ask anyone another question.. Good way of help.. =p
02:34.54Kattybored.
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02:36.17argusyea, i remmeber when this channel used to never stop ( i just started loggin back in to free node )
02:36.54coppiceAteboy: the spa-3000 is not a PSTN gateway
02:38.12coppiceAteboy: sorry. i'm mixing up the models. the 3000 does have a gateway port
02:38.26Ateboyyes, one fxs and one fxo
02:38.34coppiceso, what's the problem with it being a gateway?
02:38.46Ateboyin fact someone already helped me with the config files, but it still doesn't work
02:39.32AteboyAlone, what I achieved to get the voicemail working and the internal calls
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02:39.51Ateboywhat I'd like to do now is being able to use my pstn line for outoing calls
02:40.16_DAW-LAPTOPAteboy: Have you configured the SPA correctly?
02:40.25AteboyI think so...
02:41.02AteboyIn fact it is working at least partly, since there is a regular phone connected to it and I can call the 841 from it
02:41.27_DAW-LAPTOPThats the fxs port.
02:41.43Ateboyyes... so the lan and fxs seems to be ok
02:41.54Ateboynow I'd like to configure the fxo port
02:42.07ManxPowergetting FXOs to work in a way that makes sense with Asterisk is not trivial
02:42.45AteboyI'd also really appreciate to understand what the person who helped me wrote in the config file he wrote...
02:43.02Ateboyespecially extensions.conf
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02:56.46pr0mhow do i specify the record directory for meetme?
02:57.24Qwellpr0m: it defaults to /var/lib/asterisk/sounds/
02:57.30QwellYou can put it in a subdir of that
02:59.53pr0mbut how do i configure in /etc/asterisk/* ?
03:00.14pr0mi don't want to change default sound directory.
03:00.16asteriskmonkeyanyone know why an iaxy would sound crakly ? even the rining noise is crakly...
03:00.21asteriskmonkeypacket loss?
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03:00.26pr0mjust want to change the record directory for meetme.
03:00.29Qwellpr0m: I don't know you can
03:00.46pr0mright now all the records are mixed with the other sounds.
03:00.54pr0mhmm
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03:01.29PMantisWhat kind of bandwidth is required for 40 simultaneous calls. SIP vs IAX2 ?
03:01.38pr0mok.  next question...
03:01.45QwellPMantis: SIP and IAX2 will take about the same bandwidth
03:01.50pr0mhow do i upload ringtones to my polycom 601?
03:02.04PMantisQwell: Really? I heard it's a 90K vs 13K packet size.
03:02.11Qwellpr0m: Why not just make a subdir?
03:02.14QwellPMantis: depending on codec
03:02.15pr0mi've followed directions on configuring sip.cfg and ipmid.cfg on the tftp/ftp server.
03:02.19Qwellthey both use the same codecs
03:02.31pr0mquell: but where do i tell * to save the meetme records?
03:02.48pr0mnot a codec thing.
03:03.01PMantisQwell: Ok, any idea if a T1 is sufficient for 40 calls then?
03:03.07Qwellpr0m: SET(MEETME_RECORDINGFILE=subdir/filename.wav)
03:03.13QwellPMantis: for which codec?
03:03.14pr0mah!
03:03.18Qwelland, I'd say no, regardless
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03:03.46[TK]D-Fenderipmid is depreciated... thats pre 1.5 standard
03:03.49pr0mhmmm
03:03.51Qwellactually
03:03.54pr0mreally?
03:03.55Qwellpr0m: remove the .wav at the end
03:03.57[TK]D-Fenderyup
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03:04.05PMantisQwell: ok.. that's where I was heading. I need to recommend a minimum bandwidth requirement to a client. They need 40 calls with good quality. I was thinking ulaw
03:04.07[TK]D-FenderI run an all-Polycom shop at work
03:04.13QwellYou use MEETME_RECORDINGFORMAT to change that
03:04.16pr0mcan i just set(MEETME_RECORDINGDIR=subdir) ?
03:04.20QwellPMantis: at least 3 T1s
03:04.31Qwellprobably more like 5
03:04.41Qwellpr0m: no, but you can do like
03:04.56Qwellpr0m: SET(MEETME_RECORDINGFILE=subdir/meetme-conf-rec-${CONFNO}-${UNIQUEID})
03:05.03Qwellor something
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03:05.13Qwellpr0m: show application meetme
03:05.20asteriskmonkeywhat causes crakles and pops on a voip line?
03:05.20PMantisQwell: So.. like 7Mbit *min*. How confident are you?
03:05.33QwellPMantis: confident enough
03:05.40QwellPMantis: ulaw is about 85k/s per call
03:05.56Qwellgive or take 5-10k
03:06.02Qwell(mostly give)
03:06.12warthogis backgrounddetect the only cmd that can detect talking?
03:06.30Qwellwarthog: voicemail detects silence
03:06.35QwellDoes that count?
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03:06.58PMantisQwell: That's 3,400 k/s, so perhaps 3 T1's?
03:07.05pr0myou're right.  no MEETME_RECORDDIR in app_meetme.c
03:07.19warthogwell,  ok, but I need to call a pager, detect talking then their tone, then send them a number to be paged and this it harder than I expected.
03:07.47QwellPMantis: at least
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03:08.16PMantiswarthog: Does the paging system stop playing audio and listed to DTMF if you press digits while its playing?
03:08.32rob0asteriskmonkey: rice krispies? (Does it snap too?)
03:09.09asteriskmonkeyfunny.. trying to figure out why my voip lines pops and crakles
03:09.12asteriskmonkey:P
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03:10.04pr0mok color me stupid... but what does -> mean in C?
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03:11.15{zombie}asteriskmonkey: packet loss, or the latency is jittering too much
03:11.58rob0~weather kmgm
03:11.59asteriskmonkeyi look at iax2 show channels and i see no jitter buffer :P
03:12.32warthogwhen a callfile initiates a call, it starts down the priorities way before the call in answered, is there a way to get asterisk to pause until the call is answered, or a voice is heard or a tone is heard before executing the remaining priorities?
03:12.59SkramXwarthog: it should start executing once the line is picked up
03:13.52warthogskramx, that is what I thought, but I am quite sure it is executing as soon as zap/1 is picked up, I can see my test sound files playing way before I answer the phone
03:14.28SkramXhmm
03:14.37SkramXi have never worked with a zap card.
03:15.40warthogskramx, this is a real problem for me cause I need to put an numeric page into a pager, I have to wait till they answer, say their welcome message, then a tone, then play the digits to them, much harder than I expected.
03:16.04MasterObi-WanKhow can I forward these udp packets from an public ip to and internal ip ? 4569, 5004-5082, and 10000-20000 ?
03:16.18SkramXwarthog: just text message?
03:16.29warthogI don't seem to have a reliable method of listening for their pick up of the line or for their tone before I try and play the dialtones to them.   no, it has to be a numeric page.
03:16.57warthogthe pager company is in the stone ages and I HAVE to make this work.
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03:17.45SkramXuse toneloc?
03:17.46SkramXLOL
03:17.53PMantisQwell: Thank you. I hope it's accurate enough in practice! :)
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03:21.58warthogscramx, I keep getting kicked off, did you say anything after I said I was working with a stone ages pager company?
03:22.13MasterObi-WanKhow can I forward these udp packets from an public ip to and internal ip ? 4569, 5004-5082, and 10000-20000 ?
03:23.01*** part/#asterisk asteriskmonkey (n=phil@69.158.154.22)
03:23.11*** part/#asterisk jebba (n=jebba@200.115.209.178)
03:23.27thazza<-- warthog has left this server. (Read error: 104 (Connection reset by peer))
03:23.28thazza[14:17] <SkramX> use toneloc?
03:23.28thazza[14:17] <SkramX> LOL
03:23.51MasterObi-WanKQwell, Hey man whats up ?
03:23.56iCEBrkrtoneloc? That's some old skewl wardialer
03:24.13QwellMasterObi-WanK: not a lot
03:25.06iCEBrkr..and I can't beleive warthog is STILL working on this...
03:25.39*** join/#asterisk _Soul_ (n=Soul@87-196-2-198.net.novis.pt)
03:25.44_Soul_greetings
03:25.56_Soul_a trivial question for you guys:
03:26.10_Soul_we had asterisk 0.9 running fine, and upgraded to 1.2.1
03:26.11*** join/#asterisk argus_ (n=ryan@londonderry-cuda2-68-170-153-62.lndnnh.adelphia.net)
03:26.41_Soul_we're seeing some syntax problems, hope you can help me nail them. here0s the relevant part of the log
03:26.49trixterwarthog: before you got dropped earlier I was gonna suggest that you look at app_nv_backgrounddetect - while it most likely wont work stock it might give you a frame of reference to detect a tone, hwoever I question the reliability
03:27.03_Soul_Dec 28 03:00:39 VERBOSE[29673] logger.c:     -- Executing GotoIf("SIP/sergio-b785", "0?20") in new stack
03:27.04_Soul_Dec 28 03:00:39 WARNING[29673] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected TOK_EQ, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKE
03:27.04_Soul_N; Input:
03:27.04_Soul_<PROTECTED>
03:27.04_Soul_<PROTECTED>
03:27.05_Soul_Dec 28 03:00:39 WARNING[29673] ast_expr2.fl: If you have questions, please refer to doc/README.variables in the asterisk source.
03:27.06trixtergah didnt see he is gone again
03:27.08Qwell~pb
03:27.20iCEBrkr_Soul_: Don't put a space between the values
03:27.34_Soul_iCEBrkr, let me try that. thanks 4 the help
03:27.48iCEBrkrtrixter: the cheapo x100p cards will detect if the line is answered.
03:28.02trixterhe asked about detecting a tone after answer
03:28.07iCEBrkroh
03:28.21trixterits not a modem/fax tone so you would have to alter what is looked for but ...
03:28.45_Soul_iCEBrkr, is that restriction new with 1.2.1 ? i mean, everything went fine with 0.9..
03:29.06iCEBrkr_Soul_: Something changed in the parser.  I had that issue with all my gotoIf()'s
03:29.07trixterthe problem of course is that systems may use different frequencies, and you dont know if its a single tone or a mixed tone or what so detecting it vs other noise may be hard, but possible generally
03:29.25_Soul_iCEBrkr, here too with the gotoif
03:29.38iCEBrkr_Soul_:  I removed the spaces around my ='s and it fixed it
03:29.48trixterI have seen that if a variable is null
03:30.04iCEBrkrtrixter: yea, well that's when you just prepend an 'X'
03:30.20ManxPower_Soul_, that restriction happened in 1.0 I believe
03:30.24trixteryeah been doing that in shell scripts for like a decade
03:30.24_Soul_iCEBrkr, inside the gotoif's, or in the declaration part ?
03:30.40iCEBrkr_Soul_: inside.
03:30.46_Soul_iCEBrkr, thanks
03:31.08iCEBrkrGotoIf($[${VAL1}=${VAL2}]...)
03:31.35trixterdont forget your x's :P
03:31.39iCEBrkrhehe
03:31.42Qwellreal men use foo's
03:31.50iCEBrkryeah if there's a chance the one value is null or blank...
03:33.42*** join/#asterisk warthog (n=warthog@66.55.112.240.ppp.northrock.bm)
03:34.04warthoganyone else getting kicked of all the time today?
03:34.12Qwellwarthog: just you
03:35.10warthogI am trying xchat instead of gaim, we will see....
03:35.27trixterwarthog: before you got dropped earlier I was gonna suggest that you look at app_nv_backgrounddetect - while it most likely wont work stock it might give you a frame of reference to detect a tone, hwoever I question the reliability
03:35.37_Soul_iCEBrkr, ManxPower, nope, i still got:
03:35.38_Soul_Dec 28 03:34:50 VERBOSE[30312] logger.c:     -- Executing GotoIf("SIP/sergio-8264", "0?20") in new stack
03:35.38_Soul_Dec 28 03:34:50 WARNING[30312] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected TOK_EQ, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:
03:35.38_Soul_=81.92.197.134
03:35.40_Soul_^
03:35.42_Soul_Dec 28 03:34:50 WARNING[30312] ast_expr2.fl: If you have questions, please refer to doc/README.variables in the asterisk source.
03:35.45Qwell~pastebin
03:35.46jbotfrom memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
03:35.49iCEBrkr_Soul_: dude.. pastebin.ca
03:35.57ManxPower_Soul_, Use pastebin.
03:35.59_Soul_iCEBrkr, sorry
03:36.07iCEBrkr_Soul_: then the left value is blank or null
03:36.09trixterit would take a little coding on your part to detect the beep but it can be done..  however I dont think the beep frequency is standard so you have to use a little fuzzy logic or similar to detect it
03:36.16ManxPowerPaste the ONE line from extensions.conf that causes that message.
03:36.18Qwelljbot: pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
03:36.20jbot...but pb is already something else...
03:36.24Qwellwtf
03:36.41_Soul_ManxPower, i got lots of those lines
03:36.41trixterpeanut butter
03:36.47iCEBrkr_Soul_: So make it someting like X${VAL1}=X${VAL2}
03:36.49Qwellhe won't tell me what it is
03:36.56ManxPowerThen paste them to pastebin.ca
03:36.58trixter~pb
03:37.01iCEBrkrQwell: he don't like you
03:37.07Qwellhe doesn't like ~pb
03:37.09Qwelljbot: forget pb
03:37.18Qwelljbot: no, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
03:37.20jbotQwell: okay
03:37.20trixterwonder if jbot has definitions on a per channel basis or not
03:37.20ManxPowerjbot can be prissy sometimes
03:37.28Qwellbetter
03:37.29Qwell~pb
03:37.30jbot[pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
03:37.36*** join/#asterisk smirl (n=owned@68-169-204-147.agstme.adelphia.net)
03:37.41trixterpb is a coder for familiar (phil blundel) ...  htey use jbot on here too if it doesnt seperate channels then ...
03:37.54Qwelloh...oh well :P
03:37.57smirlanyone got NvFaxDetect ?
03:38.11trixtersmril: I used the 1.0.6 ver with 1.2.1 noi problem
03:38.19trixteralong with nvbackgrounddetect
03:38.22smirli just need the download
03:38.28trixternewumantelecom
03:38.36smirli emailed em, no response for 24 hours
03:38.49trixterhttp://www.newmantelecom.com/download/asterisk/faxdetect/1.0.6/app_nv_backgrounddetect.c
03:38.56warthogtrixter, well, I know exactly how long the pager co talks, so If I get talking for about that time then silence I know I am good to go, the problem I did not expect is that asterisk 1.2.1 is not waiting to get a answer before it executes it priorities on this extention and I did not know that backgrounddetect HAS to go to the talk extention ONLY when it detects talking, I wish it could go to a priority.
03:38.57smirltyvm
03:38.59trixtergo up a little higher for the o thers
03:39.06_Soul_iCEBrkr, ManxPower: http://pastebin.com/481105
03:39.33trixterwarthog: why cant talk goto the extension you want?
03:39.42trixtermaybe use channel variables to customize its goto ability a bit more
03:40.23smirltrixter, is there a readme for installation ?
03:40.33ManxPower_Soul_, NO, the lines from extensions.conf not from the console.
03:41.13_Soul_how do i know what are the relevant lines ? the logs dont point me to a specific line number
03:41.29ManxPowerThey would be prolly GotoIfs
03:41.55warthogtrixter, ok, I am working off the list at http://www.voip-info.org/wiki/index.php?page=Asterisk+-+documentation+of+application+commands, which I suspect is woofully incomplete, am I right, there must be many apps I don't know about then.
03:42.11ManxPoweror at least anything that has a $[ in it
03:42.14warthogthis could be really good news!
03:42.15_Soul_ManxPower, iCEBrkr: http://pastebin.com/481114
03:42.35ManxPowerwarthog, "show applications" in the Asterisk console
03:42.47trixterwarthog: there are possibly apps you dont know about - I dont know what you know so I cant say
03:43.03iCEBrkr_Soul_: You need to prepend your values with an X or something incase they're null or blank
03:43.13trixtersmirl: um not afaik there might be, its trivial to edit the make file in asterisk/apps to make it build
03:43.28_Soul_iCEBrkr, please provide with an example, im not following your explanation
03:43.47iCEBrkrs,1,GotoIf($[X${ARG1}="X"]?10)
03:43.54iCEBrkroh and that'll break
03:43.55iCEBrkrhang
03:44.00iCEBrkrs,1,GotoIf($["X${ARG1}"="X"]?10)
03:44.03iCEBrkrthats better
03:44.20warthoginteresting, talk is not listed in show applications
03:45.05trixterwhy would it be?
03:45.05ManxPoweriCEBrkr, X is not needed in 1.2.x,  IIRC
03:45.08trixtertalk is an extension it jumps to if it detects talking
03:45.18trixterlike fax is an extension it goes to if it gets the initial modem tone
03:45.33trixterManxPower: it is if you dont want that warning
03:45.41trixterif you just ignore it like everyone else then its not needed
03:45.44ManxPowerBut if you don't use X then you need quotes
03:45.45iCEBrkrhaha
03:46.02iCEBrkrManxPower: I quote everything and toss an X in there.. I have no problems.
03:46.19warthogok, I see, you mean the extetion of backgrounddetect,   I can explain...
03:46.32ManxPowerexten => s,3,GotoIf($[${LEN(${FAX_DEST})} = 0]?9:4)
03:46.42ManxPowerThere's how *I* handle it.
03:47.05smirltrixter, found the instructions on the wiki page i already had open lol. I'm retarded
03:47.07smirlthanks
03:47.11ManxPowerLEN will always return a number so you don't have to worry about a null value
03:47.36iCEBrkrManxPower: So you do your LEN() compare before the actually compare?  Much overhead.
03:47.53ManxPowerHere's one to hurt your brain.  Don't try this at home kids.  exten => _XXXX,5,GotoIf($[${LEN(${DIAL_TIMEOUT[${INDEX}]})} = 0]?6:7)
03:48.08iCEBrkrie, don't wanna type that much!
03:48.10ManxPoweriCEBrkr, if it's 0 I jump over a bunch of stuff.
03:48.35iCEBrkrYeah, but when you're actually looking for matching values...
03:49.30*** join/#asterisk mrdigital-laptop (n=erob@pool-68-236-61-51.phil.east.verizon.net)
03:50.03ManxPowerAh, you mean like this: exten => _XXXX,8,GotoIf($["${DIALSTATUS}" = "BUSY" | "${DIALSTATUS}" = "CHANUNAVAIL" | "${DIALSTATUS}" = "CONGESTION"]?12:9)
03:50.26De_MonI wish asterisk's parameter expansion was more bash-like
03:50.27iCEBrkryeah
03:50.53QwellDe_Mon: what, like allowing ${SOMEVAR:1} ?
03:51.24QwellManxPower: wanna see a painful one?
03:51.29Qwell~striplastdigit
03:51.32jboti heard striplastdigit is ${EXTEN:0:$[${LEN(${EXTEN})} - 1]} , will remove the last digit from EXTEN, making 5551212 become 555121.  Change the "1" to remove more digits.
03:51.32*** join/#asterisk bmg505 (n=leon@dsl-146-54-145.telkomadsl.co.za)
03:51.33ManxPowerThere's AEL, JavaScript, and Perl apps for doing dialplan PROGRAMMING.
03:51.57De_MonQwell # ## % %% :index:len etc etc
03:52.04root__can anyone help me on asterisk 1.2
03:52.30iCEBrkrroot__: not when you IRC as root
03:52.59QwellDe_Mon: It can do the len/offset stuff
03:53.33warthogtrixter, If i use the talk extention, it would have to work like this, callfile initiates call to pager co and connects that to astrisk extention for a specific user, we then do a backgrounddetect, jump to talk extention, I would then need to know the extention I jumped from which I could use to deterine the tones I need to send to the pager co.  how do you detect what extention you jumped from to get to talk extention (forgive if that
03:53.33warthog<PROTECTED>
03:53.53smirl~weather KMWN
03:54.03_Soul_Dec 28 03:52:48 ERROR[30597] pbx.c: Function LEN not registered
03:54.27_Soul_ManxPower, was trying to do it your way
03:54.44root__can i know the etenstion of any support Engineer of IAX2
03:55.05De_MonQwell youre right, i got carried away
03:55.12trixterwathog: sounds like a problem that can be solved with a channel variable
03:56.03iCEBrkrAlmost sounds like a answering machine detection stuff
03:56.10trixterbasically it is
03:56.11iCEBrkrWhich is a great anal pain..
03:56.15iCEBrkrPAIN! I SAY!
03:56.20trixterjust a pager company instead of a tape recorder at someones how
03:56.21trixterhouse
03:56.27iCEBrkryeah
03:56.34trixternewmantelecom is alledgly working on answering machine detect ubt its a tricky problem
03:56.47iCEBrkrI had to come up with a clever way to use the app_machinedetect and backgrounddetect to get it accurate
03:56.52trixterthere may or may not be leading/trailing silence
03:56.55Qwellthere is answering machine detection code up on the wiki
03:57.00Qwellerm
03:57.02Qwellbug tracker
03:57.13iCEBrkrYea, app_machinedetect
03:57.14trixterthere may or may not be a beep, the beep may or may not be a single or multi-frequency tone, the duration of the beep may or may not be fixed length
03:57.14iCEBrkrIt works
03:57.24trixterthey have it?  interesting
03:57.38trixterdo you have a url for it?  or is that not newmantelecom's?
03:57.39ManxPoweriCEBrkr, does it detect BEFORE the answering machine message is finished?
03:57.56iCEBrkrManxPower: It blocks when it hears 'noise
03:58.08De_Monim on voip-info.org looking for info on dialplans in perl, not seeing anything can I get a link?
03:58.09iCEBrkrYou have to do some dialplan logic from there.
03:58.29trixteriCEBrkr: link?
03:58.32iCEBrkrYou have some threshholds you have to set and if there's X amount of noise versus N amount of silence.. etc. etc.
03:58.36iCEBrkrIt's on the Wiki somewhere
03:58.37warthog<PROTECTED>
03:58.50iCEBrkrI got it working pretty good
03:58.54trixterthink I found it what version do you have?
03:58.55ManxPoweriCEBrkr, so I could use it to avoid leaving messages on people's cell phones telling them they have voicemail and to press # to access their voicemail (which doesn't work since it's a recording of the system calling)
03:59.01trixterhttp://www.thenetbrain.com/files/app_machinedetect.c
03:59.14iCEBrkrBut like I said, I had to use BackgroundDetect to actually trigger the answering machine detection code.. and you have to make multiple calls
03:59.22iCEBrkrManxPower: For sure
03:59.36iCEBrkrManxPower: It works with my T-Mobile voicemail.
03:59.44iCEBrkrtrixter: yup, thats it
03:59.45mrdigital-laptopis DAW still here.
03:59.46ManxPowernifty
04:00.01iCEBrkrYou'll have to hack it in to 1.2.x tho.. It won't compile without some minor mods
04:00.06iCEBrkrJust have to shift some code around
04:00.13trixterdo you have anything newer than feburary 2005?  seems old for the low version number would have anticipated some bugs to show themselves
04:00.24iCEBrkrIt's pretty basic really.
04:00.35iCEBrkrThe code works, it's all about how you use it in your dial plan
04:00.38trixterwell it doesnt look that long ...  I will download it and play with it
04:00.40iCEBrkrand getting the damn thresholds working
04:00.52warthogthis code is based on "wait for silence" that sound like it is right up my alley, after I detect talking....
04:01.04iCEBrkrYeah
04:01.05trixterwonder if you can use it in reverse, detect telemarketing calls that way
04:01.12trixterI am sick of getting calls from arnold
04:01.15iCEBrkrhaha
04:01.39iCEBrkrThat's what the ZapATeller stuff is for :)
04:02.09trixterwill have to look at it later though, ...  wow it just occured to me all of the tasks that were piled on are finally getting done..  nearly all of em
04:02.20trixteralmost time to start finding more work
04:02.33trixterzapateller afaik only plays SIT tones which isnt that effective anymore
04:02.51iCEBrkrZapateller + Privacy Manager work
04:02.55trixterbecause of the popularity of the radio shack branded device many companies turned that off
04:03.29warthogany of you guys know why when I dial out on zap/1 say on priority 1, it starts executing priority 2 before the call is even answered?
04:03.47iCEBrkrYou have callprogress enabled?
04:04.03warthogwill check
04:05.26trixterdont say that!  it will get you into trouble
04:05.27iCEBrkrI haven't upgraded to 1.2.1  I'm still on 1.2.0
04:05.40iCEBrkrtrixter: Yea, don't you work on a lot of 1.2.x code :P
04:06.12trixterI dont release anything I do work on becuase contributing to gpl projects is against my religion
04:06.20iCEBrkrhaha
04:06.20trixterhas been for about 8 years
04:07.26coppicethe worshipful company of parasites?
04:08.14_Soul_i know this sounds stupid, but how do i load the LEN function ?
04:08.36_Soul_searching on voip-info turned up nothing
04:08.49iCEBrkr_Soul_: ALmost sounds like you don't have a clean build
04:09.52trixteryeah give your build a bath
04:09.55trixterbe sure to use soap
04:09.56_Soul_iCEBrkr: Dec 28 03:52:48 ERROR[30597] pbx.c: Function LEN not registered
04:10.04trixterif you dont have a clean build soap is really important
04:10.08iCEBrkrhar har har , soap
04:10.12trixter:D
04:10.31*** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn)
04:11.14warthogicsbrkr, just trying callprogress=yes now, they say it is experimental, have you tried it yourself?
04:11.27iCEBrkrsur ehave
04:11.29iCEBrkrand use it
04:11.41warthogcool, I guess that means it works!!!!!!!!!!
04:12.07trixterno it means that for him on his box with his setup it works, doesnt mean it works everywhere :)
04:12.21iCEBrkrMileage may very :D
04:12.24warthogok, I have not been kicked off yet so I guess GAIM is NOT a good irc client!
04:12.33warthogxchat keeps me on no problem
04:15.09*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
04:17.57shmaltzso quiet
04:19.57*** join/#asterisk aless (n=fruribe@245-76-246-201.adsl.terra.cl)
04:20.56alesshi, im trying to call to another extension from a sipura 841 but i get a voice "that number is not in the speeddial system ..."
04:21.24loudthats not an asterisk problem.
04:21.34shmaltzaless, sounds like a sipura dp problem
04:21.35warthogicdbrkr, weird, with callpresence=on, now I don't see any verbose output on what file is playing in the asterisk console as I did before
04:21.52alessoh, thanks
04:21.59Primerhrmm, someone is telling me there's a skype->asterisk gateway
04:22.04Primeranyone hear of this?
04:22.29*** join/#asterisk bkw__ (n=brian@ppp-69-155-251-101.dsl.tulsok.swbell.net)
04:22.35shmaltzPrimer, any details? or is that person using an ATA?
04:22.46Qwellbkw_: y0
04:23.30Primershmaltz: person now tells me it's some commercial windows only app
04:23.56warthogthat will win lots of converts, nope...
04:24.06shmaltzPrimer, http://www.voip-info.org/wiki-Skype%20Gateways
04:24.44*** part/#asterisk rozo (n=rozo@c-24-17-192-196.hsd1.wa.comcast.net)
04:25.42shmaltzhere is another one:
04:25.43shmaltzhttp://mybroadband.co.za/vb/showthread.php?p=394474
04:27.05harryvvThat would be cool if there was a skype to asterisk gateway.
04:27.27trixteralledgly ebay-paypal will be at etel in janurary, which means they are prolly gonna showcase their newest purchase - skype.  I will try to ask them if they plan on making libraries so others can interface with it
04:27.48iCEBrkrwarthog: You sure you didn't set verbose lower?
04:28.07iCEBrkrHow eBay is gonna use Skype is beyond me.
04:28.16iCEBrkrTHey gonna bring back the old auctioneers?
04:28.19warthogI just double checked and definately not.
04:28.20trixterthey have to do something cause skype as it is now is not suited for businesses and ebay-paypal wants businesses to use their service, they make more with them
04:28.20Primershmaltz: yeah, dissapointing...
04:28.33trixteriCEBrkr: click2call prolly - contact a merchant
04:28.50thazzaupay4it
04:28.52iCEBrkrCuz yea, *I'm* gonna answer a 'call'
04:29.09trixterwell look at how it is now, lets say you are a smaller business only 5 employees
04:29.15*** join/#asterisk dominicand (n=ni@pcp958687pcs.bechgr01.in.comcast.net)
04:29.18dominicandhello
04:29.23trixterwith skype its 1 account largely you cant use it in a business like that
04:29.24dominicandanybody in?
04:29.59dominicandI have a 4 port ISA Dialogic Corp card
04:30.02dominicandanybody interested
04:30.09iCEBrkrLOL
04:30.19iCEBrkrdominicand: Sure I'll take a sledge hammer to it for ya
04:30.23dominicandhey
04:30.25dominicandu never know
04:30.30iCEBrkr$1 a swing?
04:30.31dominicandi would have done it
04:30.39dominicandbut maybe somebody needs it
04:30.56iCEBrkrYea, cuz people need a dialogic card like they need a hole in their head.
04:31.15trixterand an isa one at that
04:31.24shmaltzdominicand, ppl don't need Dialogic cards here, try #windowz
04:31.27trixterhow many people have isa at all anymore?  most boards dont have isa connectors
04:31.41trixterwell you never know someone may want it
04:31.42warthogdahdodading, you can almost hear the intel commercial in the background!
04:31.45iCEBrkrDialogic == Headache
04:31.46dominicandasterisk is the os form pbx
04:31.47dominicandright?
04:31.52*** join/#asterisk xtr (n=01928375@S0106000c41ed11e1.vf.shawcable.net)
04:31.54iCEBrkrOS?
04:31.55iCEBrkrNo
04:31.59iCEBrkrIt's PBX software
04:32.07dominicandohok
04:32.12dominicandthat is what i meant
04:32.26dominicandwell if anyboyd woulod need an Dialogic board, isa or not, it would be here
04:32.34iCEBrkrehhh, not really.
04:32.45shmaltzdominicand, try the asterisk-biz list
04:33.16*** join/#asterisk ryansc_ (n=ryansc@h139-055-149-047.adsl.navix.net)
04:33.18dominicandwell i will keep asking, i am sure somebody wants it
04:33.29iCEBrkrGood luck with that one
04:33.36dominicandhehe
04:34.03dominicandiCEBrkr: what do u you Asterisk for?
04:34.21iCEBrkrAt home I use it for a glorified answering machine.
04:34.28iCEBrkrAt work, I've been doing some IVR stuff with it.
04:34.32iCEBrkrDB integration etc.
04:34.49mrdigitaliCEBrkr: mysql db?
04:34.50iCEBrkr...and I suppose I use VoIP here at home too.
04:34.59iCEBrkrmrdigital: MS-SQL and MySQL
04:35.12mrdigitaliCEBrkr: PM?
04:35.18*** part/#asterisk PMantis (n=sswitzer@cpe-66-66-115-197.rochester.res.rr.com)
04:35.27*** join/#asterisk JunK-Y_ (n=junky@67.71.110.21)
04:35.46iCEBrkrPlaying World of Warcraft at the moment, responses will be delayed :)
04:36.14mrdigitalthats fine
04:36.31warthogicsbrkr, I just confirmed, with callpresence=yes, not only does verbose not work, the sound files did NOT actually play, I just removed the line and restart zaptel + asterisk and it works as before, soundfiles play, but they start BEFORE the call is actually anwered, this is darn frusterating!!!!!!!!
04:37.17iCEBrkrThen somehow it doesn't know the call is being answered
04:37.52iCEBrkr<PROTECTED>
04:37.52iCEBrkr<PROTECTED>
04:37.55iCEBrkrWTF is this shit?
04:38.04iCEBrkrI swear FreeNode is tehgay
04:38.07bkw_OPERATION IMPENDING DOOM II
04:38.19mrdigitaliCEBrkr: register yoru nick
04:38.21iCEBrkrum
04:38.22iCEBrkrIt is?
04:38.32mrdigitaldid you try mesging me back?
04:40.46*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)
04:41.13warthoglooks like I will have to deal with app_waitforsilence as my only hope of getting this, if you have any further ideas on how to get asterisk to not execute until the caller actually does answer, keep me in mind.
04:41.55bkw_warthog, you want the holy grail of apps
04:42.00bkw_you don't want much do ya?
04:42.01bkw_:P
04:42.14benjkwarthog, go digital
04:42.43coppicebkw_ which form of impending doom won't wipe us all out this time?
04:42.49warthoglater all, thanks for the ideas icebrkr and trixter, the rest of ya, the client gets what they are willing to demand (and pay for....)
04:42.50*** join/#asterisk J4k3 (i=j4k3@dhcp-12-197-128-55.intrastar.net)
04:42.58bkw_coppice, you watch Invader Zim?
04:43.15coppicei have no idea what invader zim is
04:43.26bkw_ok thats where I got that
04:43.39bkw_http://images.google.com/images?q=Invader+Zim&hl=en&lr=&client=safari&rls=en&sa=N&tab=ii&oi=imagest
04:44.18benjkanybody got a Hitachi WIP-5000 ?
04:45.13benjkwell, as it stands right now, there isn;t much reason to wish for one of those
04:45.25bkw_"Why would you do all of that?" .... "Because its cool!"
04:45.41iCEBrkrGeek factor, duh
04:45.42trixterbenjk: you still asking about that? :P
04:45.46bkw_Invader Spooge? wtf
04:45.53bkw_thats WRONG
04:46.06bkw_they actually called one of their invaders.. "Invader Spooge"
04:46.12benjktrixter: I have managed to find the secret unlock code that makes the phone accept input
04:46.37trixterahh
04:46.40iCEBrkrI need the secret unlock code for chicks....
04:46.42trixterwas it *31337#?
04:46.50trixtercause that would just be funny
04:46.52benjkof course it rejects without explanation/notification if it isn't unlocked and there is no unofficial unlock menu
04:46.52bkw_iCEBrkr, act gay
04:47.01iCEBrkrbkw_: Tried that, all it got me were dudes..
04:47.04Qwells/act/be/
04:47.09bkw_iCEBrkr, you did it wrong then
04:47.12iCEBrkrSHIT
04:47.14coppiceiCEBrkr: the secret is not to attach the chastity belt in the first place
04:47.17benjkso you have no clue why it doesn't accept any input when it does reject
04:47.20*** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net)
04:47.24trixterheh
04:47.37trixterdoesnt sound that out of the ordinary for rushed to market consumer goods
04:47.40iCEBrkrI was flattered. I hear gay guys have good taste :P
04:47.46benjkthe secret unlock code is 000000 instead of the user password in the user password menu
04:47.57bkw_I would have tried that first
04:48.09bkw_0000,00000,000000,1234,12345,123456
04:48.21dudesSometimes paperclips can be your friend too
04:48.21bkw_ok ok third
04:48.32asterboyanyone know a good url to describe SIP client server relationships?
04:48.34dudesAt least /w Cisco ATA's
04:48.35benjkwell, in fact I did try that, but it wasn't clear that it changed anything
04:48.49benjkbecause the phone gives you no feedback really
04:48.50bkw_asterboy, the 17,000+ pages of RFC
04:48.55asterboylol
04:49.01asterboyya, that is harsh reading
04:49.09bkw_but learn it
04:49.11bkw_love it
04:49.23asterboywas hoping for illustrations
04:49.30bkw_they exists
04:50.02dudesDr Suese books have cool illustrations =0
04:50.19asterboythey do.
04:50.37benjkanyway, now that I have configured all the settings -- took me about 20 minutes to enter all the data on that tiny clumsy keypad -- the phone seems to ignore those settings anyway
04:50.50asterboy~sip
04:50.51jbotrumour has it, sip is http://www.cs.columbia.edu/sip/  X11 PPP dialer interface written in gtk+. URL: http://www.geocities.com/SiliconValley/Campus/3104/sip/  Session Initiation Protocol (see RFC 3261)
04:51.10coppicedudes: but reading those cat in the hat stories to my kids is a pita
04:51.50bkw_I'm going to Plano, Tx friday to watch Brokeback Mountain.. muhahahah
04:51.51bkw_I can't wait
04:52.28*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
04:52.42dudescoppice - Never actually read cat in the hat
04:52.47bkw_WHAT?
04:52.50bkw_are you mad?
04:52.57dudesI just remember "The cat in the Hat"
04:53.08bkw_your parents should be shamed
04:53.08dudesI read green eggs and ham though
04:53.26dudesI was busy playing on the old tandy as a kid
04:54.10asterboyjbot's sip page is no longer valid.
04:54.27bkw_asterboy, you haven't heard of google?
04:54.42bkw_google can find ANYTHING
04:54.48asterboyI hate google
04:54.48bkw_except a "Moose Penis"
04:54.51bkw_WHAT
04:54.54bkw_get the fuck out!
04:54.59bkw_you're fird
04:55.00bkw_er fired
04:55.01asterboyclusty.com
04:55.03bkw_get out
04:55.07dudesI just lost my breathe
04:55.07asterboyyahoo.ca
04:55.13dudesHATE google, but how
04:55.15joelsolankidudes: hey
04:55.23bkw_you use an inferior search engine
04:55.28bkw_no wonder you can't find anything out about SIP
04:55.32asterboylol
04:55.37bkw_GOOGLE can find all
04:55.50dudesGoogle is long for god of search engines
04:55.55dudeserr short
04:56.00bkw_haha
04:56.10bkw_I <3 Google
04:56.19joelsolankidudes: want to know how much asterisk can handle simul calls on p4 with 1 gb ram ?
04:56.22dudesexcept their liberal I like them
04:56.37bkw_joelsolanki, what speed p4?
04:56.41_Soul_ok, those pesky gotoif's are fixed, still got a problem with sip tapi + asterisk 1.2.1 + mcc2, but i need some sleep
04:56.51joelsolankibkw: 2.4 G
04:56.59bkw_try about 96 - 128
04:57.01_Soul_iCEBrkr, ManxPower, thanks 4 all the help
04:57.06iCEBrkrnp
04:57.07bkw_and it depends on codec
04:57.21dudesI can't imagine that doing more than 20-30 g729's
04:57.30dudestranscodes of course
04:57.35joelsolankibtkw: i have g729 codec.
04:57.35bkw_<- 5500 sip sessions on a 3 GHZ HT box
04:57.40bkw_you'll get 10-12
04:57.54joelsolankiohhhhhhhh only 10-12 simul calls ?
04:57.59_Soul_just a one more question, im curious, i've been searching for a way to implement click 2 call on a browser
04:58.04*** join/#asterisk Shakh (n=shakhruz@83.221.168.141)
04:58.06bkw_g729 is a CPU W H O R E
04:58.19dudesI know my XP 2800 Mobile can handle around 20 or so g729's
04:58.23Corydon76-homebut then again, so am I...
04:58.29*** join/#asterisk nico2 (n=tecnico@user-24-236-120-2.knology.net)
04:58.31bkw_haha
04:58.35coppicebkw_ anything decent uses CPU. G.729 is actually better than most
04:58.35bkw_yo Corydon76-home  ltns
04:58.40Corydon76-homeYep
04:58.42bkw_coppice, true
04:58.53bkw_you hit a limit in asterisk before you can really push it hard
04:58.59joelsolankibkw: which hardware can handle 80 to 100 calls simul ?
04:59.01_Soul_i'd like to click on a callto: url, and shoot a tapi event. sip tapi would to the rest.. do you guys know any app that turns firefox or internet explorer tapi aware ?
04:59.01dudesI'm suprised I was able to get a slin to g729 on a 850 Athlon
04:59.11bkw_locking retention builds as you load the box till all you're doing is waiting on locks or deadlock
04:59.32bkw_locking === bad
04:59.45Corydon76-homeLocks are just to control access to shared resources
04:59.55bkw_if you need that many locks you designed it wrong
05:00.08bkw_you can get by with much less locking if done correctly
05:00.18Corydon76-homeNo, if you're continuously WAITING on locks, you have a problem
05:00.24bkw_yep
05:00.28nico2Hi. Any hints on what I need to do to keep DISA from hanging up too early ? the default timeout is too short for me to type all the numbers. I alread have digit=10 and response=15 but didn't seem to make a difference
05:00.39Corydon76-homeYou can add more resources to wait less on each
05:00.51nico2me ?
05:00.52bkw_asterisk doesn't scale to the levels I want
05:00.55bkw_doubt it ever will
05:01.02trixterjames bond in poonraker is on tv now
05:01.04trixterer I meant moonraker
05:01.05Corydon76-homeFor example, instead of using a single ODBC connection, you can pool multiple connections
05:01.07trixterstupid keyboard
05:01.24bkw_I want Quad DS3 cards with hardware echo cancel.  PCI-X
05:01.32bkw_I want to put two of those per box
05:01.42bkw_<- Don't expect nor want much do I?
05:01.55Corydon76-homeThe PCI bus isn't capable of handling that much data at the interrupt speed that you need
05:02.01bkw_yes it is
05:02.02Qwellbkw_: only 4 DS3s?  sissy
05:02.04Qwell:P
05:02.06joelsolankibkw: i want to have 80 to 100 calls simul ..which hardware will be suitalbe?
05:02.10bkw_you don't have tu run 1000/sec interrupts
05:02.25Corydon76-homeActually, you should have 8000/sec interrupts
05:02.29bkw_no you don't
05:02.33Corydon76-home1000 is just the closest we can get
05:02.34bkw_we don't do that with sangoma
05:02.48bkw_we slice it off in 10ms, 20ms, 30ms or what ever we want
05:02.51bkw_it works just as good
05:02.58bkw_lowering the interrupt requirements
05:03.08Corydon76-homeHowever, you're adding latency
05:03.14bkw_you have that anyway
05:03.22bkw_if you're going voip
05:03.27bkw_20ms is perfect for most tasks
05:03.28Corydon76-homeYes, but it's better to reduce the latency to the best you can get
05:03.35dudesjoelsolanki - I've seen 5 TE410P's doing zap to sip calls (granted ulaw to slin) on a Dual Xeon 3Ghz /w 1GB of ram
05:03.46bkw_thats what the hardware echo cancel is for
05:03.48Corydon76-homeFaxing, for example, requires low latency
05:03.50joelsolankiok
05:04.03*** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-68-73-192-129.dsl.sfldmi.ameritech.net)
05:04.28bkw_20ms is plenty to do faxing
05:04.30dudesI don't think they had more than 380 active zap channels though
05:04.44bkw_why do 1ms chunks filling up a 20ms buffer then sending it when you can suck 20ms off the wire and send it without waiting
05:04.52Corydon76-homeSome fax machines are more tolerant than others
05:04.58bkw_you  have 20ms either way you go
05:05.08Corydon76-homeSome fax machines will cancel immediately if you have 20ms latency
05:05.11bkw_Corydon76-home, 90% of the fax machines are broken
05:05.35bkw_and no 20ms will not kill it.. its if you loose 20ms and don't have that data to pass on you're fucked
05:05.36Corydon76-homeThat's a nice stat, but people expect their fax machines to work
05:05.52bkw_i'm telling you in our testing 20ms doesn't have a single issue with faxes
05:05.55Corydon76-homeespecially if they work on the PSTN
05:06.15asterboyman, even http://www.cs.columbia.edu/sip/ has 404 pages!
05:06.31asterboythere has to be a good url
05:06.41dudes3 hops and a 3ms ping to a VOIP provider gives excellent results
05:06.56bkw_I can fix via IAX over 60ms better than I can over the PSTN
05:06.59bkw_its funny.. but true.
05:07.34Corydon76-homeThen the fax you're using and the fax everybody else is using are not compatible
05:07.35asterboymmusic?
05:07.44bkw_and you know this?
05:07.51bkw_we have done it via a TDM card to IAX too
05:07.57bkw_between that and our hylafax PRI
05:07.59nico2Anyone knows how to adjust DISA's timeout ? It's hanging up at the middle of me dialing a sequence of digits.
05:08.16Corydon76-homebkw_: you've tested every fax machine?
05:08.29bkw_You can't test them all but every one we have so far works fine.
05:08.35bkw_you cant not have 100% fax compatibility
05:08.37bkw_its impossible
05:08.43asterboyIs the latest RFC 3265?
05:08.44bkw_you can work around most of it
05:08.50Corydon76-homeWe have customers for whom their fax machines will not work across a voip link
05:09.01bkw_yes it happens
05:09.09bkw_we have t.38 too that works fine also
05:09.13trixterI have customers who have problems using faxes across pstn links
05:09.18dudescanons tend to be troublesome
05:09.22bkw_trixter, so do we
05:09.24bkw_dudes, yep
05:09.34asterboysetting the fax speed to 9600 usually helps
05:09.37bkw_yep
05:09.42bkw_thats how we work around 90% of our issues
05:09.45bkw_limit the speed
05:09.52dudesover VOIP that's normally what you'll get
05:10.15bkw_we are actually waiting on spandsp to get t.38 then we can redo chan_fax
05:10.29bkw_coppice, any word on that one :P
05:10.47dudesSpandsp works pretty good /w t30, heh
05:11.06bkw_some fax machines still have drama with it.. but I think coppice and redder86 have been working hard on those
05:11.30dudesI'm eager to see 0.0.3 working right
05:11.44dudesFrom my testing it tended to seg *
05:11.48*** join/#asterisk xtr (n=01928375@S0106000c41ed11e1.vf.shawcable.net)
05:13.08*** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-68-73-192-129.dsl.sfldmi.ameritech.net)
05:13.09bkw_Asterisk has a nick name around here.. Disasterisk :P
05:13.18asterboychrist!, click on FAQ on the main SIP site and it says moved but does not tell you where!
05:13.46asterboymust be reading it wrong.
05:13.55bkw_asterboy, use the "cache" feature of google
05:14.03bkw_to view the cached version of the page
05:14.06MikeJ[Laptop]hello all
05:14.09asterboygood suggestion for now.
05:14.17bkw_MikeJ[Laptop], you see what I commited to SVN tonight?
05:14.30MikeJ[Laptop]where?
05:14.34asterboythats crazy though...you would think SIP would be an important FAQ.
05:14.35bkw_our svn
05:14.37bkw_go update boi
05:14.41MikeJ[Laptop]ummmm
05:15.15MikeJ[Laptop]private me what you are talking about, cuz I just did and I don't see it
05:15.45Corydon76-homeHe renamed another set of functions...
05:16.31file[laptop]he's talking about something completely different
05:16.40implicithi
05:16.55MikeJ[Laptop]and now for somthing completely different
05:17.19file[laptop]ugh what a day
05:17.26MikeJ[Laptop]yes indeed
05:17.39MikeJ[Laptop]heh
05:17.43MikeJ[Laptop]good luck with that...
05:17.58dudesif it ain't busy you ain't working
05:18.24trixterI guess that fits if you are tech support
05:18.57bkw_file oh file did you see what was done today?
05:19.04asterboyhttp://www.voip-info.org/wiki-SIP
05:19.15file[laptop]bkw_: no, I'm almost afraid to ask
05:19.22bkw_look on screen 5
05:19.25file[laptop]who did what and what did they break
05:19.30asterboyRFC 3261 is the latest SIP
05:19.31bkw_tony and nothing
05:19.38asterboyhow do we update jbot?
05:20.10file[laptop]oh that
05:20.15file[laptop]yes
05:20.54file[laptop]I'm mindless right now
05:20.59file[laptop]mmm
05:20.59bkw_I hear ya
05:21.00file[laptop]brains!
05:21.00dudesthat's what bv/cokes can do to ya
05:21.02bkw_did you know what I fixed today?
05:21.06*** join/#asterisk nswint (n=nswint@c-24-98-129-84.hsd1.ga.comcast.net)
05:21.09file[laptop]bkw_: no, I know nothing about today
05:21.28bkw_look on screen 6
05:21.30file[laptop]except I did cleanup before I passed out, and then did something small for Angela when I woke up
05:21.37*** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-68-73-192-129.dsl.sfldmi.ameritech.net)
05:21.56file[laptop]oh that
05:21.59file[laptop]yes, I know about that too
05:22.09file[laptop]odd - I know about stuff I didn't know I knew about
05:22.54nswintanyone know where I can find some Home Automation AGI's?
05:22.54bkw_no more vpn drama
05:22.58bkw_I said ENOUGH today
05:23.12*** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-68-73-192-129.dsl.sfldmi.ameritech.net)
05:23.14bkw_oh oh I gotta tell this story
05:23.16bkw_its funny
05:23.26*** join/#asterisk BugKham (n=lamer@202.8.86.170)
05:23.41BugKhamanyone using A400P?
05:23.50trixterbkw_: isnt that what col klink used to say?  HO-GENT!
05:23.50MikeJ[Laptop]damn inet connection, and damn softphones..
05:23.52bkw_I email cogent to update their route filter so I can announce our /23 to work around their stupid uunet/cogent drama... the guy emails back saying "you're not on the contact list"
05:24.05MikeJ[Laptop]gunna have to go compile somthing to make a call... grrr
05:24.11BugKhamI am having some silly questions
05:24.13bkw_I called up they had assumed I was with apple computer becuase of my mac.com address
05:24.15trixtercogent has drama with uunet now?
05:24.18trixterin november it was level3
05:24.31bkw_no the cogent/uunet interconnect in IAD is maxed out
05:24.46benjkbkw: that's called the reverse halo effect
05:24.51coppiceBugKham: what is an A400P?
05:24.58trixterahh the level3 stuff was cause level3 wanted cogent to pay the overages on bandwidth peering (access charges) and cogent didnt wanna
05:25.03benjkcoppice an original TDM400
05:25.08bkw_coppice, sounds like the OpenVox clone hardware
05:25.24BugKhamcoppice: TDM400 clone from openvox
05:26.35BugKhamcoppice: it's my first time using it =)
05:27.05dominicandI have a 4 port ISA Dialogic Corp card, if anybody is interested let me know...
05:27.16bkw_dominicand, makes great target practice
05:27.28dominicandlo
05:27.28coppicegood for landfill too
05:27.29dominicandlol
05:27.39dominicandmaybe somebody here collects them
05:27.52benjkdominicand: how much do you pay me for taking it off your hands?
05:28.05bkw_BugKham, you didn't buy from Digium?
05:28.08dominicandlol it is free
05:28.33benjkwell, here where I live we have to pay for garbage disposal
05:28.34asterboyok I have memorized every word of RFC 3261
05:28.49bkw_*gasp and swoon*
05:28.50dominicandi was going to trash it, then i saw this channel, so myabe i though someobody might wanted it
05:29.04asterboy:P
05:29.26bkw_Clutch the pearls, I'm shocked!
05:29.37file[laptop]I should get to bed
05:29.53bkw_file you have a new task
05:29.57bkw_you and I have to work on this one
05:30.00bkw_muhahahahahah
05:30.07file[laptop]what is it now
05:30.08bkw_this one is fun
05:30.16benjkdominicand: maybe eBay will be more useful
05:30.18bkw_the bot backend processing scrypt
05:30.22bkw_er scrypt
05:30.24bkw_doh
05:30.25bkw_script
05:30.27bkw_fuck I can't type
05:30.32file[laptop]oh, yeah
05:30.33bkw_tired and brain is warped
05:30.48benjkor ask on the GNU Telephony mailing list, I think they have drivers for that thing
05:30.54bkw_new check box on contacts "Please msg my IM when I get a call"
05:31.01MikeJ[Laptop]brain permanantly warped..
05:31.04dominicandwho knows myabe on ebay they have to pay me instead of my paying them
05:31.16dominicandlol
05:31.21benjk:)
05:31.22file[laptop]bkw_: any web stuff hand it my way, I find that part enjoyable :P
05:32.00MikeJ[Laptop]heh
05:32.03MikeJ[Laptop]well...
05:32.10MikeJ[Laptop]I need to fix this build...
05:32.18bkw_MikeJ[Laptop], did we break it?
05:32.21bkw_we might have ;)
05:32.22MikeJ[Laptop]no
05:32.27MikeJ[Laptop]same issue as before ..
05:32.36bkw_we found one strange issue when we compiled it on OpenBSD
05:32.48MikeJ[Laptop]the const** vs const* blah[0] crap
05:32.53bkw_the main binary was static to the lib.. and the .so was dynamic to the lib
05:32.58bkw_so each had their own namespace
05:33.07bkw_didn't show up on Linux, Mac OS X or Win323
05:33.09bkw_er Win32
05:33.12MikeJ[Laptop]welll..
05:33.17MikeJ[Laptop]not an issue on win32...
05:33.27file[laptop]yeah, I'm gone... I'll be back tomorrow
05:33.29bkw_and apparently not on Linux or OS X
05:33.30MikeJ[Laptop]totally diff build system there
05:33.40MikeJ[Laptop]the other unixes are the weird part..
05:33.41file[laptop]tomorrow... today...
05:33.44file[laptop]somewhere around there
05:33.45MikeJ[Laptop]but the problem makes sense...
05:33.53bkw_hard one to find
05:34.11MikeJ[Laptop]well.. the const ** is the right way to do it...
05:34.12MikeJ[Laptop]BUT
05:34.19MikeJ[Laptop]I am getting some other BS error now.
05:39.18*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
05:40.21asterboybenjk, my neighbors are from Japan and confirm that like CallerID, everything else is Japan only also.
05:40.42asterboystrange world of its own.
05:41.27SwKasterboy: anything and everything Telcom in Japan is just for Japan
05:42.05benjkwell, Japan was totally shut off from the rest of the world for 4 centuries until the US sent warships to Yokohama in 1868
05:42.09asterboyyes, it is fascinating...there is a reason for it, although I doubt I'll ever understand it.
05:42.11SwKthey even modified T1 to be a J1... 99% of J1 is the same as T1.. just enuff different to make it where its not compat
05:42.28asterboylol, T1 to J1
05:42.54coppiceit was called T1M (M foe modified) for a long time
05:43.32benjkin the 15th century the Portuguese bought the exclusive rights to Japan from the Pope in Rome
05:43.36asterboyJapan also has natural resources...although not exported.
05:43.38*** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-68-73-192-129.dsl.sfldmi.ameritech.net)
05:44.15asterboythat is interesting history....bought the rights to another country from a religious icon.
05:44.18asterboycrazy
05:44.24benjkat the time they had clergy in Japan who esteblished themselves as influential advisers to the Japanese leadership
05:44.38*** join/#asterisk jake1932 (n=jake1932@pool-68-236-10-151.phil.east.verizon.net)
05:44.41benjkthe Dutch had just won their independence from Spain
05:45.36benjkand they wanted to do trade with Japan -- silk trading was a very lucrative business at that time
05:45.36asterboywhat is the status of that now? Are they still under Portuguese influence?
05:45.41benjkbut the Japanese said "you people are not to be trusted cause you disobeyed your king, the king of Spain"
05:46.17benjkthe Dutch said "this ain't our king, we were made a colony, just like you are being made a colony of the king of Portugal now"
05:46.29benjkthe Japanese said "prove it!"
05:47.00benjkso the Dutch captured a papal ship with the documents that proved it -- right with the seal of the pope
05:47.28benjkthen they gave the documents to the Japanese and the Japanese knew they were in trouble
05:47.48benjkso they said "what can we do? they have these modern weapons and ships"
05:48.24benjkso the Dutch said "here's the deal: we help you fight off the Portuguese in return for a trade agreement between Japan and the Netherlands"
05:48.34benjkand that's what they did
05:48.49benjkbut the Japanese didn't trust those European devils
05:49.04asterboyfantastic history lesson
05:49.19benjkthey thought that the Dutch would sooner or later aspire to become colonial masters themselves
05:49.32benjkso they built a little artificial island in the bay of Nagasaki
05:49.35*** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au)
05:49.42nico2Anyone knows how to adjust DISA's timeout ? It's hanging up at the middle of me dialing a sequence of digits.
05:49.43benjkcalled Dejima (out-island)
05:49.57benjkthis became the Dutch trading post in Japan
05:50.09*** join/#asterisk dokhench (n=dochench@adsl-156-26-67.bna.bellsouth.net)
05:50.14benjkthe Chinese had their out-island only a few hundred yards away
05:50.56benjkother than those two trading posts on two artificial islands in Nagasaki bay, no foreigner was allowed in Japan and no Japanese was allowed to leave the country
05:51.40MikeJ[Laptop]ok
05:51.42benjkthis remained so until 1868 when COmmodore Perry was sent by the US government to force Japan to open
05:51.43MikeJ[Laptop]all better now.
05:51.46MikeJ[Laptop]yay!
05:52.07benjkand this isolation has formed a Japanese mentality that persists to this day
05:52.31asterboyYou really have to respect the Japanese for how they conduct themselves.
05:52.45asterboyeven if it looks wack on the outside.
05:52.46benjksometimes yes, sometimes no
05:53.13mrdigitalbenjk: can you use * to set your cid ?
05:53.18asterboyno doubt their laws create some friction and frustration.
05:53.23coppicebenjk: nothing too strange about the japanese insular mentality. what is strange is the british insular mentality, when their history is of going out and conquering half the planet
05:53.28benjkmrdigital: on ISDN yes
05:53.33mrdigitallike if you called 111-222-3333 can you make their cid say 203-222-3232 ?
05:53.44mrdigitaleven tho ur # is not that
05:53.59mrdigitalthhats a random #
05:55.17benjkcoppice: most of the time isolationist policies of countries like Japan, China, Burma etc etc are an outcome of Western colonialism
05:55.41coppicenot in china's case
05:55.44benjkcan we blame the Chinese for not trusting us with all the shit we have unloaded on them?
05:56.48coppicebenjk: the shit went both ways.
05:57.00benjkwe wanted their tea and had nothing we could have paid them with
05:57.22benjksame trouble many third world countries are in today
05:57.25aminorexopium from afghanistan
05:57.48benjkaminorex: precisely
05:57.54aminorexcia learned a trick or two there
05:58.06benjkso the Chinese did just what we do today: fight a war on drugs
05:58.40aminorexfrom bhutan to north korea, asia is a land of hermit kingdoms
05:58.41benjkonly that the result was foreign armies invading
05:59.04coppicebenjk: garbage. they had a high value commodity - tea - and the emporer would only allow the traders to accept gold and silver in exchange, because he got most of it. up in the north he couldn't give have given a damn what was happening in places like fujian, as long as he got high gold
05:59.09benjkhow about Colombia was to invade the US for not wanting to buy cocain?
06:00.18benjkcoppice: fair enough, but dumping highly addictive drugs on China which caused even more misery isn't exactly something I would call a reasonable response
06:00.57coppicethe british found they could circumvent the emporer'r rules by trading opium - as legitimate a commodity to trade in those days, as tobacco today
06:01.36coppicein those days opium was the preferred pastime of the average affluent british lady
06:01.40*** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-68-73-192-129.dsl.sfldmi.ameritech.net)
06:01.57benjkin the beginning yes, but then the Chinese experiencing trouble with that commodity, tried to stop it just like we have illegal narcotics laws today after we didn't have them 100 years back
06:02.14benjkand the response to that was invasion
06:03.38*** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-68-73-192-129.dsl.sfldmi.ameritech.net)
06:03.38benjkits like the US passes laws to forbid wine and France, Italy, Spain, South Africa, Australia and New Zealand invade the US to force them to allow wine back in
06:04.05benjkwith the difference that wine is less addictive than opium
06:08.29thazzai do wonder what opium and other drugs have to do with asterisk and VoIP...
06:08.40benjka lot
06:08.49coppicewell, it seems many current chinese historians lay the blame firmly at the feet of those controlling a system where the provincial leaders in the south were so afraid to tell bad news to the guys in beijing, they were paralysed and unable to seek help when things got a bit rough
06:08.52loudaddiction.
06:09.52coppicethazza: if you look at the design of most VoIP equipment its clear the designers were on something :-)
06:10.03asterboylol, yes addiction is certainly a common thread.
06:11.33asterboyif you drink coffee, tea or eat chocolate or smoke anything...your on something.
06:12.39coppiceI love this thread on the mailing list about don't buy barbietones because of handset echo. The writers don't seem to have noticed that most phone have the same problem, and many to a greater extent. Somehow Grandstream always seems to get the worst of people's complaints
06:13.11coppiceasterboy: at least I don't take chocolate intravenously
06:13.38asterboylol
06:14.03loudgs does not care that much about complains, they do what they can and make millions, millions a year.
06:14.46thazzaasterboy: Shame i don't drink coffee tea or smoke anything, never have never will..
06:15.05thazzaAnd it is very rare that i eat chocolate.
06:15.26*** join/#asterisk SERGEUS_ (n=s@195.112.98.13)
06:15.34asterboyhow about the tons of drugs that make it into your drinking water?
06:16.13asterboywhere does all that expired Viagra go?
06:16.40thazzadrinking water?
06:17.07asterboyA lot of expired drugs return to the water table.
06:17.21loudwtf, he drinks water from the swiss alps.
06:17.31thazzalol
06:17.34coppiceand all the ones passing through people's bodies
06:17.35asterboylol...dam he's got great water!
06:18.04asterboythink about how many TONS of Viagra are created every year.
06:18.18asterboyit has to go somewhere...although obviously diluted.
06:18.36coppicemore worrying is PCBs
06:18.37thazzais that why i can't get it down?
06:19.33thazzato much drinking water viagra?
06:20.00coppicethe PCBs are making you sterile, so I guess the viagra helps counteract that
06:21.09thazzavery good.. i don't want kids.. they get in the way of my goal.
06:21.34thazzaearning money for myself. lol
06:22.05thazzai am still wondering  why this is a topic in a asterisk channel.. lol
06:22.07*** join/#asterisk aless (n=fruribe@245-76-246-201.adsl.terra.cl)
06:22.17asterboylol
06:22.31coppicePCBs have been heavily used in the telecoms industry
06:22.56nico2Anyone knows how to adjust DISA's timeout ? It's hanging up at the middle of me dialing a sequence of digits.
06:22.57coppiceprobably still are in areas with poor controls
06:23.15asterboywell, we went from Japan's CallerID and how they do everything different based on history of trade
06:25.26*** join/#asterisk tengulre (n=tengulre@61.185.224.66)
06:31.28benjkasterboy: today, it's probably more to do with the fact that the traditional way to isolate has turned into an easy to maintain and proof, thus convenient trade barrier to keep foreign competition out
06:32.09benjksometimes things done differently in Japan make good sense
06:34.05benjkfor example if you ride a bus in the country side, you will find that it is the exact opposite of how we do it elsewhere on the planet. You enter the bus at the backdoor and take a ticket from a ticket dispenser. Then you leave the bus at the front door paying the driver according to the number on the ticket. The dispenser increases the number at every stop.
06:34.18*** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye)
06:34.56*** join/#asterisk aless_ (n=fruribe@42-79-246-201.adsl.terra.cl)
06:35.10benjkand a display in the bus shows you the fare for each number, so you know exactly how much you are to pay if you leave at the next stop.
06:35.31benjkthis is far more efficient and transparent than any other system I have come across
06:36.21benjkunfortunately, things like that are all too easily abandoned by the Japanese in favour of "more modern" western systems
06:37.14benjkwhile at the same time they have to fiddle with things like T1, adding no value, serving no purpose other than to be just incompatible enough to make it difficult for foreign companies to sell their equipment in Japan
06:39.58coppicebenjk: when T1M was developed I think the japanese market was still rather vulnerable from a development point of view. however, they had a telco monopoly, so outsiders were locked out anyway. NEC pushing T1M into Taiwan made more sense. Taiwan was locked into japanese only kit for years
06:40.28benjkpoor Taiwanese
06:41.51benjkthe Hitachi Cable guys, the ones with that WIP-5000 WiFi phone, told me that they have 14 different versions of their firmware
06:42.02benjk13 different versions for Japan
06:42.08benjk1 version for the rest of the planet
06:43.07benjkthis is in order to accommodate the differences in SIP proxy/servers by Japanese vendors
06:43.22benjkSIP is a god sent for Japan
06:43.41benjkits a Japanese engineer's dream come true
06:43.55coppiceStrange they never used R2 :-)
06:43.56benjkfinally a standard that allows messing it up to total incompatibility
06:44.53coppicebenjk: SIP is no dream. its too easy to fix. they need things that require volumne commitment to fix, as T1M did for a long time
06:45.11benjkwell, so far it works pretty well for them
06:46.59coppicethe use of english by large japanese companies still amuses me. shiseido's current advertising campaign seems to use slogans generated by a random word generator :-)
06:48.05*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
06:48.30benjkI am running my * box so that it talks directly to my ISP's SIP service and it took about 2 weeks of work to make this work. I am confident that the total number of people in all of Japan who bypass the official black box is a dozen or two, but not more
06:48.33coppice"The advent of the beauty climax". Well, beautiful women make men climax, but I don't that's what they mean - assuming they mean anything at all
06:49.11benjkJapanese names mean nothing either
06:49.30benjkits like those Chinese names that have something "Lucky" in them
06:50.26benjkin the period before WWII just about every new Japanese company was called Great Pacific Foobar
06:50.40coppicenames don't generally mean anything. those chinese names with lucky in them are direct translations, and the chinese is more meaningful than most english names
06:50.45benjkin the period before WWI they were called First Industrial Foobar
06:52.46asterboyso many dimensions to Japanese culture.
06:53.00benjkRecently some creativity has shown up
06:53.04benjkTomato Bank
06:53.38benjkso renamed by his president "because he likes to eat tomatoes"
06:54.47coppicenames are not intended to illuminate, but to divert attention - e.g. MicroSoft
06:56.00coppiceIntel - the company which desperately wanted to be in telecoms, failed, and still ended up twice the size of the next semicondutor company
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06:58.29coppiceI've seen various things about how startups which reach IPO tend to be doing something different from what they started doing. The other day I saw the figure of 90% for VC funded companies. so much for founder's vision :-)
07:00.58asterboyya, look at Google.
07:01.54asterboynow they just want to spread like a virus to every web page and control every aspect of human essence.
07:02.19asterboysure wish they would just make their search engine better.
07:02.24benjkwhy is everybody so anti-Google these days but nobody cares about Microsoft and Skype
07:03.07asterboygood point...guess its just that Google has become another 1000lb Gorilla like Microsoft and Skype.
07:03.21coppicei think its google's rate of change. its good for MS, though - its diverting attention
07:03.39*** join/#asterisk douthat (n=john@ip68-105-159-32.br.no.cox.net)
07:03.57benjkyes but I see their potential for impact on all aspects of life far less than that of MSFT or Skype
07:04.42benjkin the end, Google may control all of the worlds advertising and advertising cost may skyrocket
07:05.15coppicedunno. a directory monopoly is exactly what MS would like to be, and have abysmally failed at. if you control the world's directories, every other company is at your knees
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07:06.03asterboydiversity in nature has been proven scientifically to be the best adapt at handling change and dealing with hardship...Google, MS, Skype...et al...just want to control EVERYTHING.
07:06.09benjkthat's hardly as big a problem as MSFT's stranglehold on all data processing and the way we work, or Skype's goal to become the worlds sole telephony provider and charge more than we have ever paid before for telephone calls
07:06.10coppicevisa has been pretty benign, but its a similar gatekeeper function they perform
07:07.17tengulrehi,all
07:07.19coppicethere is potentially more money is being the gatekeeper than there will ever be in data processing
07:07.25tengulreI M COMING!!
07:07.45benjkbut MSFT is a gatekeeper by controling the door handle
07:08.01asterboytengulre....ewwww
07:08.24benjktenguire: please not here and not in public
07:08.36tengulre:(
07:08.58benjkBTW, to give a further insight into the Japanese being different theme ...
07:09.00tengulreon one like me ? :(
07:09.15tengulreWuWu....
07:09.16benjkthe Japanese say "I am going" when they mean "I am coming"
07:09.31asterboythats confusing.
07:09.39benjknot for the Japanese, no
07:10.02tengulreI don't like japanese! :(
07:10.10wasimhow do they differentiate between this go, and the other go
07:10.25benjkwasim: they are masters of context
07:10.57benjkalso, black and white is black and white in most languages I have come across
07:10.59asterboy"masters of context"...should be a title for a book
07:11.04benjkit's white and black in Japanese
07:11.04tengulreanybody know where have IAX client application, and have source code and under Windows ?
07:11.04coppicewasim: that could be said of the various comes too
07:11.11benjkinside out is outside in
07:11.21benjkupside down is downside up
07:11.37coppice"masters of context" sounds like an advanced OOP book
07:11.44benjkLOL
07:12.17benjkand Japanese grammar is RPN based
07:12.33tengulreI come from CHINA, english is not my country language, so .... It have many errors of syntax!
07:13.05benjkcoppice: explain to tenguire what "I am coming" means ;)
07:13.15tengulre:(
07:13.23tengulredon't discuss this, OK?
07:13.47benjkin C-antonese or Pythonghua
07:13.51asterboyIn the news...Google buys SKype
07:13.55benjkor C-antonese++
07:13.56asterboy:P
07:14.10coppicewifey used to say "it comes". that seems so impersonal :-\
07:14.16asterboylol c++
07:14.19tengulrenobody discuss asterisk ??
07:14.27wasimtengulre: not here, god forbid
07:14.28asterboywhats that?
07:14.37smirltrixter, still her?
07:14.40smirlhere*
07:14.43asterboyoh, shift 8
07:14.46benjkasterboy: how can Google by Skype? EBay already bought them a while ago
07:15.10tengulreNI MEN DAO DI ZAI SHUO SHEN ME??
07:15.11asterboyjust kidding around...didn't know ebay bought them.
07:15.18asterboyNOoooooooo.
07:15.33asterboyso eBay is evil also.
07:15.35asterboydam
07:15.43benjktenguire can you write that in Hanzi please
07:15.51wasimmajor controversy in dams here
07:15.55benjkotherwise I don't have a chance to figure it out
07:15.56tengulrebenjk: HAO!
07:16.12tengulre?????????
07:16.15smirlanyone else compiled nvFaxDetect before that would like to login to my server to compile it for me. I'm getting weird compile errors.
07:16.41tengulreSHUI NENG HUI DA WO DE WEN TI ?
07:17.31coppicetengulre: that's doesn't look too much like hanzi :-)
07:17.42benjklet me guess: Shui is 水
07:17.45wasimi don't think thay have T1 in .cn, you probably want WEN E1
07:17.52smirlCREA MOF SUM YUNG UY!
07:18.46wasimsmirl: thats thai
07:19.33smirlAPPS+=app_nv_faxdetect.so
07:19.34benjkand hao is 好
07:19.51dudesand this is English
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07:19.56coppice????我的文題
07:20.11tengulre11my networking is bad!
07:20.13tengulre11:(
07:20.14coppicewhoops
07:20.25tengulre11coppice: what's that??
07:20.34benjkC-antonese
07:20.46benjkas opposed to Pythonghua
07:20.47[av]banijapan, the land of two mains frequencies
07:20.51[av]bani50hz and 60hz!
07:21.23coppice????我的問題
07:21.42benjkcoppice: the first four chars show up as ???? here
07:22.17coppicei don't know what they should be
07:22.37benjkmaybe you have to use Pythonghua then
07:22.57smirlhttp://pastebin.ca/35044
07:23.15smirlcompile errors ^^^
07:23.17coppiceyue gwoh kui yung gwong dung wah ping yam, ngoh wooi ming
07:23.22benjkalthough I have both traditional and simplified
07:24.35benjkI meant simplified fonts, not pinyin :)
07:26.23smirlIn file included from app_nv_faxdetect.c:26:
07:26.23smirl../include/asterisk/file.h:27:2: #error You must include stdio.h before file.h!
07:26.23smirlIn file included from app_nv_faxdetect.c:26:
07:26.24smirl../include/asterisk/file.h:56: error: syntax error before '*' token
07:26.24smirl../include/asterisk/file.h:56: warning: function declaration isn't a prototype
07:26.24smirl../include/asterisk/file.h:57: error: syntax error before '*' token
07:26.26smirl../include/asterisk/file.h:57: warning: function declaration isn't a prototype
07:26.28smirlapp_nv_faxdetect.c: In function `nv_detectfax_exec':
07:26.30smirlapp_nv_faxdetect.c:123: warning: implicit declaration of function `sscanf'
07:26.32smirlapp_nv_faxdetect.c:269: warning: implicit declaration of function `sprintf'
07:26.43wasimthwap thwap thwap
07:27.22smirlthis is the asterisk channel isn't it?
07:27.42smirli can't believe 276 people have no clue what i should do.
07:27.54benjksmilr: what's that off-topic stuff you're posting there?
07:28.02smirlFUCK
07:28.09benjkhey, just kidding
07:28.20wasimtsk tsk ... touchy
07:28.44benjkgeez
07:28.55benjkhe's gone offline
07:29.12dudesDoes NVFax even work /w head anymore
07:29.13coppicei'm sure we can find 276 people who don't know the answer, if that is what you need
07:29.57benjkcoppice: he's gone offline
07:30.48benjkanyway, for the record ... it looks to me as if the first error message is what's causing all the other errors/warnings
07:33.19wasimnow that we've helped one person, we're free till '06
07:33.38dudesfree to get drunk ...
07:34.06benjkdudes: what is it going to be today? Guinness?
07:34.26dudeswhat's what going to be
07:34.39dudesI'm writing a bitch and moan on T1 costs where I live
07:34.41dudesfor fun
07:35.07coppice99 bugs in a line of code
07:35.07benjkno I meant, what kind of booze to get drunk on ;)
07:35.09coppice99 bugs to go
07:35.10coppice99 bugs in a line of code.........
07:35.11coppice100 bugs to go
07:35.30dudesbenjk - bv/coke
07:35.39dudesthat's my friend through new years
07:35.44benjkBV?
07:35.50dudesBlack Velvet
07:36.03benjkwhazzat? Whiskey?
07:36.06dudesJust wish a litter would make through 2 nights
07:36.10dudeserr liter
07:36.27benjka litter of whiskey dogs, haha
07:39.07dudeswhiskey is good
07:39.23dudesBut it can be your enemy
07:39.41wasimbut single malts remain friends for life
07:40.01dudesI learned quick why I should not take shots ...
07:40.31benjkI prefer Armagnacs or Gognacs
07:40.38benjkor a nice Italian Grappa
07:40.53dudesI don't even know what that is
07:41.00benjkGrappa?
07:41.04dudesRed Label is good
07:41.11dudesNo clue dude
07:41.33wasimdudes: Blue Label is good, Black is passable, Red is not
07:41.54benjkIt's made from the remains of the grapes and stuff after pressing it for making wine
07:42.00benjkthen distilled
07:42.15dudesRed Label is good IMO
07:42.19benjkdepending on where you are, it will have a different name
07:42.22dudesWine?
07:42.31benjklike Marc in French speaking countries
07:42.58benjkin Italy its called Grappa and the Balkan countries have their own name (escapes me now)
07:43.40benjknot Wine, the grapes, skins, seeds which are pressed to make wine, left over from the pressing
07:44.12benjkthen distilled into a strong clear liqour
07:45.11benjkhttp://www.clearcreekdistillery.com/Grappa.htm
07:46.04benjkhttp://en.wikipedia.org/wiki/Grappa
07:46.23benjkhey, there is even some Java software that's named after it
07:47.08dudeshow many DS3's make up a gig-e
07:47.19wasimdudes: red is like $15, black is like $25, blue is like $150, try blue or black sometime, you'll like it
07:47.39dudesred label is only 15 for a liter?
07:47.42benjkhttp://www.cs.unm.edu/~moret/GRAPPA/
07:47.56dudesIt's like $30 USD here
07:48.18wasimwow ... prices have risen since '91
07:48.34benjkwasim: its called inflation ;)
07:48.36dudes<PROTECTED>
07:48.59benjkthe best stuff is that which has handwritten labels
07:49.07wasimdudes: try Famous Grouse if you get the chance, its the best blended scotch in the price range
07:49.27wasimdudes: normally, most scotches are 70% blended, 30% single malt, Grouse is reverse
07:49.49dudesJ&B ... didn't like that
07:50.08dudestoo bland of taste
07:50.12wasimdudes: a DS3 is 672 DS0 (64kbps)
07:50.25dudes28 T1's =)
07:51.08wasimso 24 DS3 == GigE
07:51.32dudesI thought a oc48 was 2.5 giges
07:51.48dudesFuck if I know, hell a f'n T1 is 2,300/mth here
07:52.05dudesand that's if you get one of those lenghy contracts and pay X mths
07:52.10coppicecorrect. OC48 is 2.5G
07:52.19benjkis that in Honk Kong dollars or Turkish Lira?
07:52.28dudesUSD
07:53.00benjkyou must be pretty far out in the desert or mountains I presume
07:53.25*** join/#asterisk Entegrity (n=Entegrit@c-65-96-116-121.hsd1.ma.comcast.net)
07:53.32dudes20 miles from here you can get a T1 for 500-600
07:53.41nico2Has anyone had the problem with WaitExten not waiting the number of seconds requested _ If I put (1) it's almost instant return, but anything over 1 (I tried 1.5 2 3 and 5) and the wait is arround 1 minute.  (v 1.2)
07:53.45coppiceyou can wrap an OC48 right round the planet for 2.5G US
07:53.59benjkso how much for 20 miles of 4 wire copper?
07:54.01wasimdudes: you might consider getting your own wireless last mile and carry it across
07:54.11wasimlast 20 mile, i.e.
07:54.18dudesa 1.92m/bit sdsl is 192.00
07:54.29coppiceuse wireless to reach the cheap wiremore
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07:54.34dudesI wouldn't get a T1 here cause it's too expensive.
07:54.41dudesCable is decent though
07:54.52dudesDownside, when the power goes out it doesn't work
07:55.15benjkUPS
07:55.27dudesWe had power for computers and shit
07:55.35dudesbut the town had no power and the internet didn't work
07:55.53benjkah
07:56.04dudesI didn't have enough power to run the heat though
07:56.08benjktell the mayor he should pay the bills in time
07:56.11dudesso that sucked, but it was only 2 days
07:56.39dudesstayed above freezing and upstairs it was 50-60ish where I like it
07:57.07benjk50-60? you must be well done by now
07:57.28dudesI like it around there
07:57.37coppicesounds like sahara weather
07:57.37benjkbraised human
07:57.52dudescold is the way to go
07:58.01coppiceI've been in the desert at 55. really nice. for a little while :-)
07:58.08dudesThough had I know the power would be out for two days I'd have turned the heat upto 70 or something
07:58.15dudeshuh
07:58.17benjkhow little? 20 ms ?
07:58.18dudes55F
07:58.35dudesor 55C
07:58.37benjkhow about an old fashioned oven
07:58.43benjkheater
07:58.47benjkstove
07:58.51dudesI wasn't worried about it
07:58.53*** join/#asterisk lme (n=lme@gw-r-ifc.infoclip.fr)
07:58.59benjkcamp fire
07:58.59coppicea few minutes. until the dehydration starts taking effect. the low humidity actually makes it quite comforting
07:59.02lmehi guys !
07:59.05dudeswe had 7 people over and we sat around drinking beer all night
07:59.33dudesI'm talking 50F not 50C (that'd be insane) and I'd just end myself t hen if it got that hot.
07:59.52dudesHell the days it was 100ish was hell.  Couldn't keep the house under 78
07:59.54dudesF
07:59.59dudeswhich f'n sucked
08:00.05benjkah roasted human
08:00.43dudessure
08:01.01dudesit's 30F outside now
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08:01.51dudesWhen the power was out it was -13F I think
08:02.29nico2IS there another option to allow a user to type an extension without using WaitExten ???
08:02.54benjkbackground(silence/2)
08:03.06benjkwith John Todd's recordings
08:03.11benjkwww.loligo.com
08:03.46dudessometimes ||s|m is needed for dtmf to get through
08:03.52nico2how would I capture what they type ? ${EXTEN}  ?
08:04.01nico2benjk: ?
08:04.15dudesumm, background handles that
08:05.14dudes5526k/bytes <--- look right for a DS3
08:05.40wasim672*64/8
08:06.19dudesthat comes to 5376
08:06.57dudes5.25M/bytes
08:08.44benjknico2: see private message
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08:16.53nico2tnx. benjk   I was trying to reply but my msg. to you is getting blocked
08:19.32benjkthat's weird
08:19.50nico2'cause I'm not registered
08:20.07nico2prv. msg. are blocked from non registered users
08:20.18benjkI didn't know that
08:20.20nico2just learned that
08:20.23*** join/#asterisk psk (n=psk@golia.caltanet.it)
08:20.59nico2Here is what I was trying to ask you.. in case you can give me a hint.  I was origintally trying to use exten => 2089,n,WaitExten(5) ,  but no matter what number I put, the wait is always arround 1 1/2 minutes... Do you notice anything wrong in my syntax or am I missing something ?
08:21.14nico2using v.1.2
08:21.17atif_hello there, a quick question regarding cdr_pgsql.c
08:21.17atif_PGresult is defined globally, and PQclear is not called for the result
08:21.17atif_where as, documentation says every result should be cleared after use, otherwise it will result memory leak in application
08:21.17atif_can anyone please shed some light on it
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08:27.54hypnoticxAnyone from Canada?
08:27.54nswint/msg NickServ test
08:29.57moralehypnoticx: i am.
08:30.28nswint<PROTECTED>
08:30.57hypnoticxsweet, where in canada are u from?
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08:35.28dudesLooking to score with some hot sluts?  <--- the vulgarity
08:36.16*** part/#asterisk hypnoticx (n=idunno_f@ip68-96-173-149.lv.lv.cox.net)
08:36.32dudeshttp://www.dailyfunnyshit.com/article.php/20051218231247798 <--- in the event you cared to know where that came from
08:36.39dudesthe beer and men joke is funny too
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08:41.51benjknswint: Andy Powell is the home automation guru
08:42.02benjkwww.automated.it
08:42.07benjkor something like that
08:43.29dudesmade that drink a bit strong
08:43.31dudesopps
08:44.07nswintbenjk: thanks
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08:52.44camonzhi
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08:53.04camonzi'm having a problem trying to do playback of mp3 files with *
08:55.08camonzon the CLI console i'm getting a app_mp3.c:108 timed_read : Poll timed out/errored out with 0
08:56.23camonzi didn't install the mpg123 package at first, but i've cleaned the prior build, make mpg123, then make && make install
08:56.31davnanyone using oztell here?
08:56.33camonzand i'm still getting the same error from the CLI console
08:57.40camonzany ideas on why this might be happening, or how to completely uninstall asterisk and reinstall it with the mpg123 library
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09:26.42BugKhamWhat's the disconnect supervision used in TDM400?
09:27.26BugKhamtone detection or reversed polarity of battery?
09:28.15BugKhamit doesn't seem to detect a hangup from the calls generated from my Ericsson BP250
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09:35.21wasimBugKham: both, depending on ls or ks
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09:39.02benjkfalse hangups or undetected hangups have been a problem with all the analog cards for years
09:39.17benjkits not exact science
09:39.24wasimfunny how pbx people do it well
09:39.28benjkmore to do with woodoo
09:39.45benjksorcery
09:40.04benjkPBX people use the local flavour of pagan sorcery
09:40.12benjkAsterisk people use woodoo
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09:40.44kuku5IS there any way that I can do some test on a line ( digium t1 card )
09:41.00wasimloop it
09:41.16kuku5ok.... and then what
09:41.22wasimwatch the far end
09:41.27benjkmake a call to yourself
09:41.31kuku5I wont be able to do anything
09:41.37kuku5but that checks the card
09:41.40kuku5I want to check the line
09:42.15benjkdo you mean electrically testing the line?
09:42.23kuku5no - like crc
09:42.24benjkas in the cable
09:42.28kuku5check for crc errors
09:42.51benjkturn CRC on in the driver (if its not already on)
09:42.59kuku5where is that?
09:43.06wasimzaptel.conf
09:43.51benjknow if you had a Sangoma card ... you could just run the diagnostics tool
09:43.53kuku5how do I set it
09:44.06kuku5Sangoma is better than digium ?
09:44.21benjkthat depends on who you ask
09:44.25kuku5:)
09:44.32benjkbut Sangoma's software comes with a diagnostics tool
09:44.34kuku5So where od I set this crc thing - and where do I check
09:44.40wasim,crc4
09:44.58BugKhamwasim: ls is for tone detection?
09:45.12benjkls = loopstart
09:45.33wasimBugKham: not really, ks is for reverse battery check, ls won't do that
09:45.33kuku5Do I need to reset the card somehow after that?
09:45.43wasimkuku5: ztcfg -vvvvv
09:45.47benjkks = kewlstart
09:46.01kuku5How do I check for crc now ?
09:46.34benjkpri debug or pri intense debug should show you any errors
09:46.41benjkearthquake
09:47.03kuku5where
09:47.04benjka long one
09:47.06benjkhere
09:47.14kuku5where you at
09:47.17benjkTokyo
09:47.24kuku5long or big?
09:47.26wasimkuku5: zttool
09:47.29benjklong
09:47.41benjkmaybe a mag 3
09:47.53benjkdepending on how far away the epicentre is
09:48.04benjknow it's finished
09:48.11kuku5heh
09:48.14kuku5cool stuff
09:48.23wasimno, not cool
09:48.23kuku5wasim: ok - what would it show me if there was a problem?
09:48.35benjkcool as long as they are small
09:48.46benjknot so cool if there's a big one
09:48.58benjkor if it's on the ocean floor
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09:50.14vlrki want to upload some of my code changes to the asteirsk cvs code how can i do that ?
09:50.29wasimvlrk: file a bug
09:50.45wasimvlrk: then pray
09:50.53kuku5wasim: Do I need to check for something with zttool ?
09:50.55vlrkwasim: this is a new feature
09:51.12wasimvlrk: bugs are features as well, depending on who you ask
09:51.15vlrkwhich i want to add
09:52.31benjkvlrk: you will also have to sign over your code to Digium
09:52.51benjkotherwise it won't be included in the CVS
09:53.33benjkbut it may perhaps be allowed to go into add-ons
09:58.04vlrkwhen i gone through the voip-info.org they mentioned to send one disclaimer
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10:03.12benjkthe quake was magnitude 4, about 100kms from here
10:05.04wasimvlrk: if you can't disclaim your code, it can either go in add-ons or openpbx.org
10:05.41*** join/#asterisk lorinc (n=ang@caracas-3352.adsl.interware.hu)
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10:11.11BugKhambenjk: is * well known in japan?
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10:13.13bussswhat's asterik ? I get "later." on the page and thats it
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10:15.42fulgasio
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10:39.32meredyddHey
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10:40.27meredyddAnyone know whether *'s SIP/RTP implementation is ~ide novo~i , or does it use an existing library?
10:40.31jahani2how to configure incoming calls?
10:40.46meredyddincoming calls from where, jahani2?
10:41.01jahani2in asterisk
10:41.22jahani2i have PSTN lines connected to a gateway
10:41.30jahani2i can make outgoing calls
10:41.39jahani2but i can not accept incoming
10:41.48*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no)
10:41.58meredyddwhich gateway?
10:42.15jahani2FXO gateway
10:42.50meredyddokaay...not familiar, but looking
10:43.29jahani2ok
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10:44.41jahani2when i call to the number connected to asterisk
10:44.52jahani2i hear girl voice saytsomethink
10:44.58jahani2and the line cut after that
10:44.59meredyddsaying what?
10:45.12jahani2not understand what she say
10:45.13meredyddIs that voice from your phone company, or from asterisk?
10:45.21jahani2from asterisk
10:45.25meredyddwell...
10:45.40meredyddI'm presuming you've got your gateway hooked up to a particualr context
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10:47.23jahani2http://www.micronet.info/Products/voip/SP5054.asp
10:47.26jahani2i use this gateway
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11:10.48saftsackhi
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11:19.17saftsackhowto realize redirection to a handy so that my asterisk server has to pay the telephone costs and not the caller?
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11:44.29fugitivomorning
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11:46.33saftsackhi
11:46.38saftsacksome voicemail experts here?
11:46.50fugitivojust ask
11:47.42saftsackok ... i spoke my own unavailable message into the telphone and it works
11:48.09saftsackbut after the message is spoken out a voice in the telephone says please leave your message after the .....
11:48.25fugitivohow are you calling your voicemail?
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11:48.46saftsackexten => s,2,VoiceMail(u11@default)
11:49.08fugitivosaftsack: show application voicemail in your cli
11:49.12fugitivosee the Options
11:49.36saftsackshow not found :(
11:49.47fugitivo??
11:49.50saftsackoh false puty ^
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11:50.16saftsack<PROTECTED>
11:50.27saftsackyes and this works but after the greeting asterisk says something
11:50.43saftsackoh
11:50.44saftsack:)
11:50.45saftsacks
11:50.57saftsackthanks
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11:59.40saftsackis the telephone responsable for redialing or does asterisk do this?
12:01.16fugitivoyou can use asterisk for redialing too
12:02.24moralehow do i limit call lengths?
12:02.34moralemy wife talks too long on the phone
12:02.41cpmheh
12:03.11morale0.02 cents a minute, 120 minutes minimum a day, thats like 60$ a month for voip
12:03.19saftsackfugitivo, i want redialing to be done by the telephones
12:03.39cpmmorale, seems reasonable.
12:03.58mutilatormorale: grab the phone from her hand?
12:04.19moralemy pstn line i was only paying like 30$ a month, although long-distance was much more expensive
12:04.24saftsackanother question does the budge tel 101 provide g711?
12:05.10fugitivomorale: hit her
12:05.27cpmheh
12:05.52moralefugitivo: only if i catch her in the act, kinda like a puppy pissing on your floor.. gotta pick it up and throw it outside so it knows where to go
12:06.09*** join/#asterisk pengyong (n=lala@218.93.103.120)
12:06.49fugitivoright, hehe
12:07.35mutilatorunplug all the phones but the cordless and then stick it in ya pocket
12:07.39mutilatorO_o
12:07.55mutilatorwhen she says "honey the phones are dead"
12:08.05moralehaha. i guess i could just do a 'stop now' to the asterisk server periodically with cron
12:08.22fugitivosetup asterisk to send you an email when she's using the phone, then go where she is, and hit her
12:08.29mutilatoryes!
12:08.48moralehaha
12:08.59moraleviolence can solve anything.
12:09.12moraleif your wife won't clean - hit her
12:09.17moraleif your wife won't shutup - hit her
12:09.23moraleif your wife won't cook - hit her
12:09.26zoa2if she cleans, hit her anyway
12:09.38fugitivoyes, if not, she'll forget the punishment
12:09.44moralehaha
12:11.27moraleand women think they have it rough
12:11.37saftsackif i call for example the extension 025245232 how can i detect whats the first digit. _0,1,Dial(...... doesnt work in this case
12:11.46cpmwhat do you tell a woman with two black eyes?
12:12.30moralenothing, shes already learned her lesson?
12:13.09cpmNothing, there's no point, you've already told her twice!
12:13.57*** join/#asterisk lubomier (n=lubomier@sunteq.sk)
12:15.37moraletime for some more south park
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12:17.59saftsackwhat was the letter for the digit 0?
12:18.02saftsackN?
12:20.07saftsackexten => _0.,1,Dial(misdn/g:TEports/${EXTEN:1},120) this line doesnt work if i call to fast :(
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12:20.44saftsackso that the telephone sends 0124 and not 0 .. 1 .. 2 .. 4
12:23.56saftsackno one who can help?
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12:53.45mutilatorBOO!
12:53.53mutilatori keep falling asleep
12:53.56mutilator=\
12:54.10mutilatornot good when ya at work
12:54.16mutilatoro_O
12:54.54moraletwo more hours for me too then i can sleep
12:55.34mutilatori think i might end up goin home at noon
12:55.47mutilatoror earlier maybe
12:56.03mutilatori can hardly stay away and i've already had 3 cups of coffee
12:56.05mutilatorstrong coffee
12:56.13mutilatoraway = awake
12:57.23saftsack^
12:57.37saftsackhere in germany it is 13:57 now :)
12:57.38fugitivocoffee is bad
12:57.41fugitivodrink energy drinks
12:57.47mutilatorbad?
12:57.49fugitivoyes
12:58.05mutilatorit costs me like 25 cents a day for a coffee
12:58.15mutilatorand an energy drink is like $1.15
12:58.20mutilatorfor a single can
12:58.42fugitivoyou had 3 cups of coffee
12:58.49mutilatorya
12:58.49fugitivothat's 0.75
12:58.53mutilatornot 25 cents a cup
12:58.58mutilatorlike a day
12:59.01fugitivooh, a day
12:59.03fugitivothat's cheap
12:59.22saftsackif i dial my number and if i press on green then asterisk fails because it takes the whole number as extension
12:59.30mutilatori don't buy bat shit coffee or anything, i get whatever theres a deal on
12:59.37*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-121-27.buckeyecom.net)
12:59.41fugitivosaftsack: what??
13:00.00saftsack^^
13:00.03mutilatormaybe he's playing twister?
13:00.06saftsackif i do redialing for example
13:00.19saftsackthe telephone sends the number as a whole block to asterisk
13:00.38fugitivosaftsack: be more specific, what are you dialing, from where to where, dialplan, etc
13:00.44mutilatoranyone screw up a number port?
13:00.47mutilatori sure did yesterday
13:01.01mutilatori ported a guys home phone number instead of his fax number
13:01.05saftsacki have an isdn telephone on port 4 and if i dial a zero the extension for dialing out starts
13:01.13mutilatormy LOA and tac case i wrote had the correct number
13:01.19saftsackexten => _0.,1,Dial(misdn/g:TEports/${EXTEN:1},120)
13:01.21mutilatorbut the order with the clec had the wrong number
13:01.25mutilator:P
13:01.41mutilatorgot fixed in like an hour tho so i was happy
13:02.04saftsackso but if i do redialing the telephone sends the number as a whole unit to the asterisk server, like 0123455
13:02.10saftsackfugitivo, did you get me? ^^
13:02.22fugitivowhat's wrong with that?
13:02.32fugitivoit should work
13:02.41saftsackthat asterisk doesnt jump into my extension exten => _0.,1,Dial(misdn/g:TEports/${EXTEN:1},120)
13:02.53fugitivowhy not? is the phone sending the 0?
13:03.02saftsackyes
13:03.07mutilatorwhy not use a normal number like 9?
13:03.13saftsackSpawn extension (raus, 0157925, 1) exited non-zero on 'mISDN/4-u3'
13:03.37fugitivodo you have an extension 0157925?
13:03.45saftsackno
13:04.22fugitivoare you sure the phone is sending the right number?
13:04.39saftsackyes it shows me the right number on the display
13:04.47saftsackthe number which i called just before
13:04.59saftsackdo you think it is a bug in the misdn driver?
13:05.44fugitivono...
13:06.06fugitivopastebin what the CLI shows when you redial
13:06.17saftsackok
13:09.22saftsackfugitivo, ok it dials correctly :)
13:09.32saftsackbut i cannot here something in the telephone
13:10.13saftsackdo you want to see the output?
13:11.12fugitivoyou can't what?
13:11.42saftsacksry
13:11.43saftsackhttp://pastebin.com/481455
13:12.00saftsacki substituted the last 4 digits with a X
13:12.35fugitivopastebin your dialplan
13:13.19saftsackok
13:14.18saftsackfugitivo, http://pastebin.com/481458
13:14.23saftsackthe part for outgoing calls
13:14.41fugitivois that german?
13:14.55saftsackyes ^^
13:15.07fugitivoit looks difficult :)
13:15.18saftsackwhat? ^^ german or my dialplan?
13:15.22fugitivogerman
13:15.23fugitivolol
13:16.15fugitivowhat is TEports?
13:16.30saftsackoutgoing ports connected with the isdn line
13:16.58fugitivoExecuting GotoIf("mISDN/1-2", "1?anrufbeantworter|s|1") in new stack
13:17.03fugitivoi don't see that in your dialplan
13:17.23saftsackyes because i posted the outgoing section
13:17.35saftsackthe gotoifline is in the incoming
13:17.47fugitivocan't help you if i don't have all the dialplan
13:18.53*** join/#asterisk javar (n=javar@69.79.133.185)
13:19.17saftsackyes ok but a short moment i have to delete parts of the numbers
13:22.01saftsackhttp://pastebin.com/481465
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13:23.44fugitivoare you calling your own number?
13:24.13saftsackyes i thought the same
13:24.15saftsacksooooo yes
13:24.18saftsackthats the false
13:24.35saftsackbut my incomingsection is not included in the raus section
13:24.42saftsackwheres the fault then?
13:25.08saftsackbut thats my own number so its not important
13:25.14saftsackthanks for that help :)
13:25.17saftsackthanks for the time
13:28.04fugitivoif you're calling your own number then it's ok
13:28.06*** join/#asterisk asteriskmonkey (n=phil@69.156.197.242)
13:28.55saftsackyes in every other case it works :)
13:33.08lubomierplease, dontt u have idea why that capi **, doesn't work? http://pastebin.com/481414
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14:02.51P4C0hello
14:03.01*** join/#asterisk newl (n=newlook@203-59-61-62.dyn.iinet.net.au)
14:03.07NewSolefound a bug in asterisk last night needs to be fixed.....
14:04.23P4C0I'm having problems registering into my sip provider with asterisk and making outgoing calls throw that... can somebody help me?
14:04.27iCEBrkrNewSole: Bug, eh?
14:04.32NewSoleif it is tring to qualify a peer and does not qualify..... it will not create channel
14:04.55iCEBrkrNewSole: Probably cuz it thinks it's unreachable.
14:05.04NewSolebut problem is the peer I was connecting to did not accept qualify
14:05.14*** join/#asterisk jahani2 (n=k@adsl196-206-241-217-196.adsl196-16.iam.net.ma)
14:05.29*** join/#asterisk burton (i=mimx@w201.ljudmila.org)
14:05.32iCEBrkrNewSole: So disable qualify in your sip.conf for that device
14:05.46NewSoleso even if its unreachable it still should create channel
14:05.52iCEBrkrum, no.
14:06.37*** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com)
14:06.41TheCopsHi
14:06.44iCEBrkrThat's like saying even if I can't connect to a web page, a page should still load for me.
14:07.08NewSoleyes but thats another problem.... it tries to qualify via sip using asterisk@XXX.XXX.XXX.XXX and our peer will take a form of qualify but u need to state user not "asterisk"
14:07.42P4C0this is the error I'm getting when dialing outside: http://pastebin.com/481498
14:08.03NewSolethere needs to be a field in sip.conf to state alternitive
14:09.06*** join/#asterisk RoyK (n=roy@g-001.osl255.netcom.no)
14:09.08iCEBrkrP4C0: ummm, your Dial() doesn't look right
14:09.12*** join/#asterisk littleball (n=littleba@cm240.epsilon174.maxonline.com.sg)
14:09.13P4C0this is my sip: http://pastebin.com/481499
14:09.44iCEBrkrP4C0: How about ya pastebin your extensions.conf--- or at least the part where you're issuing Dial()
14:10.05P4C0this is my extensions: http://pastebin.com/481500
14:10.14TheCopsI did a debug for the subscription feature for my Snom phone and I've got this: http://pastebin.ca/35059
14:10.20TheCopsand its not working
14:10.27littleballhello, anyone can suggest me some system which is based on asterisk and J2EE architecture?
14:10.55P4C0I followed this: http://www.voip-info.org/wiki-Asterisk+config+sip.conf
14:11.02[TK]D-FenderTheCops : Thats presence, right?
14:11.07TheCopsyeah
14:11.08TheCopssorry
14:11.20moraleP4C0: make sure you change your password now you pasted it to the channel so no one uses your account.
14:11.30[TK]D-FenderI think you're missing your "HINT" pririties...
14:11.41*** join/#asterisk jaike (n=a@203.131.137.76)
14:11.46TheCops[TK]D-Fender I have hint for each extension
14:11.50Kattyhi lads.
14:11.57P4C0morale, my sip provider have ip based filters... no problem with that
14:11.58[TK]D-FenderKatty : mew.
14:12.00iCEBrkrP4C0: You don't have a sipprovider-out context
14:12.00TheCopsin the same context of the Dial Macro for the extension
14:12.08*** join/#asterisk yxa (i=empty@cm121.gamma228.maxonline.com.sg)
14:12.12jaikewhere can i find info on the voicemail bug with 1.2.1..want to check if itll affect us if we upgrade
14:12.13*** join/#asterisk amir (n=amir@hacker-173-236.congress.ccc.de)
14:12.18P4C0iCEBrkr, in sip.conf?
14:12.29iCEBrkrP4C0: extensions.conf
14:12.32[TK]D-FenderTheCops : Hmmmm
14:12.49iCEBrkrmorale: I think were adult enought not to yoink someones account.. But then again, you never know :)
14:13.07P4C0iCEBrkr, http://www.voip-info.org/wiki-Asterisk+config+sip.conf don't say nothing about that in extensions
14:13.35iCEBrkrP4C0: you just want to dial out VoIP via your 'provider', correct?
14:13.45P4C0iCEBrkr, I do have that context in sip.conf
14:13.57moraleiCEBrkr: when i was learning asterisk i wondered why i was getting charged so much for calls to places i don't know anyone :)
14:14.12P4C0iCEBrkr, yes, can you point me a good example?
14:15.13P4C0iCEBrkr, do I need to put the same mysipprovider-out context that I have in sip.conf in extensions.conf?
14:16.27iCEBrkrP4C0: well, if you're trying to dial OUT, you don't createa a [sipprovider-out] in sip.conf.  You need a register => line
14:16.50littleballhello, is it possible to store all extension configurations in database in 1.2.1? If i modify the database, will it take effects immediatelY?
14:16.52TheCops[TK]D-Fender no idea ?! :) Your buddy watch list on Polycom is working great ?!
14:17.10iCEBrkrlittleball: Yea, it's called 'Realtime'
14:17.18P4C0iCEBrkr, why all the docs that I follow seems to be outdated!? :'(
14:17.36iCEBrkrP4C0: quite possible.
14:17.52P4C0iCEBrkr, I do have a register line in sip.conf, how will a sipprovider-out context looks like in extensions.conf?
14:17.55[TK]D-FenderTheCops : Yeah it works fine (up to the bug of a limit of 7 people watched)
14:18.07TheCopsduh it is a bug ?
14:18.19[TK]D-FenderTheCops : Pastebin your sip config for the phone & its context in extensions.conf
14:18.34iCEBrkrP4C0: Read the part "Asterisk as a SIP client"
14:18.34[TK]D-FenderTheCops : On the Polycom, but you are talking about SNOM
14:18.43iCEBrkrP4C0: Your asterisk box is a 'client' of your provider.
14:18.56P4C0iCEBrkr, that what I followed: http://www.voip-info.org/wiki-Asterisk+config+sip.conf
14:19.05iCEBrkrP4C0: Right
14:19.31P4C0iCEBrkr, ... that the doc that I'm follow...
14:20.06iCEBrkrP4C0: ok, I see you have a register line in there.
14:20.58P4C0iCEBrkr, yes, I do follow the docs, the only that I may be wrong and put the sipprovider-out in sip.conf instead of extensions.conf?
14:21.39TheCops[TK]D-Fender, http://pastebin.ca/35060
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14:22.42iCEBrkrP4C0: rename sipprovider-out to sipprovider-in
14:22.49iCEBrkrP4C0: in your sip.conf
14:23.23P4C0iCEBrkr, ok, I will
14:23.54P4C0iCEBrkr, but the in calls works fine...
14:25.21iCEBrkrYa know, I don't even have a sip 'provider' setup.
14:25.47iCEBrkrI guess I have FWD, but I've never used it and from what I can tell, it doesn't look like it'd ever work.
14:26.21TheCops[TK]D-Fender weird, eh ?! :)
14:26.53P4C0iCEBrkr, now what should I do?
14:27.01*** join/#asterisk Entegrity (n=Entegrit@c-65-96-116-121.hsd1.ma.comcast.net)
14:28.14*** part/#asterisk jaike (n=a@203.131.137.76)
14:32.00P4C0iCEBrkr, are you there?
14:32.08*** join/#asterisk slappingt (n=randygre@pcp03933849pcs.sthind01.mo.comcast.net)
14:33.38*** join/#asterisk kimosabe (n=kimosabe@dsl-201-129-75-8.prod-infinitum.com.mx)
14:33.51*** join/#asterisk cfh (n=luca@82.193.23.6)
14:33.59kimosabedoes any one know any thing about the xorcom boot cd with asterisk
14:35.13*** join/#asterisk _fan_ (n=allan@static-64-222-189-188.man.east.verizon.net)
14:35.15*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-121-27.buckeyecom.net)
14:35.42znoGhrm, i swear i was able to dial Voicemail(su${EXT}) before and it worked, without having to specify the context the user is in
14:35.45[TK]D-FenderTheCops : you don't need subscribe context for HINTS....
14:36.00znoGis this a new Asterisk "feature" that you have to tell it which context the user is in to find the right entry in voicemail.conf?
14:36.05TheCops[TK]D-Fender, I tried without
14:36.08TheCopsI tried with
14:36.11TheCopsnothing work at all
14:36.13*** join/#asterisk synthetiq (n=roger@64.201.13.50)
14:36.25[TK]D-FenderznoG : you don't need the context if they're in general.
14:36.32znoGoh, they're not :)
14:36.38[TK]D-Fender:p
14:36.40znoGbut, i think i'll stick 'em in there
14:38.24*** join/#asterisk _DAW (n=bob@adsl-156-94-42.msy.bellsouth.net)
14:38.40znoG[TK]D-Fender: you sure? i added them to "general" and can't see them still. Sure it's general and not default?
14:39.35*** join/#asterisk antonios (n=anton@VPN.accessdevices.co.uk)
14:39.42lubomierplease, don't u have some resource describes the new syntax in chan_capi 0.4.0? the old chan_capie 0.3.5 does not work... ;/
14:40.09_DAWhello all
14:41.35*** join/#asterisk seele_ (n=seele@200.124.172.72)
14:42.21antonioshello, I upgraded my asterisk to 1.2.1 and I get  No D-channels available!  Using Primary channel 16 as D-channel anyway! every second or so, any ideas?
14:44.51seele_How do i pick up incoming calls from extensions different of my own\
14:45.10iCEBrkrKatty: nerd.
14:45.43seele_hello.. need help with that
14:45.53Kattyseele_: see features.conf
14:46.01_fan_seele_: look at features.conf
14:46.02Kattyseele_: i think it's *8 or something.
14:46.03seele_yes and
14:46.06*** join/#asterisk amir|22c3 (n=amir@hacker-173-236.congress.ccc.de)
14:46.06Kattyseele_: also show features at the cli
14:46.15seele_thanks katty
14:47.53seele_Whats the command in the CLI Katty?
14:48.11Kattyshow features
14:48.11*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-121-27.buckeyecom.net)
14:48.58seele_Is that for incoming trunk calls or just extensions calls?
14:49.31Kattyneither.
14:49.39Kattyoh.
14:49.50seele_what do u mean?
14:49.51Kattyboth
14:49.57seele_AH! ok
14:50.08*** join/#asterisk amir (n=amir@hacker-173-236.congress.ccc.de)
14:50.31iDunnoyay! an anti-cry cookie - now if someone would fix Java the whole world would be good again :)
14:50.34seele_for instance i make one call from extension 200 to 201 and in ext 205 i can pickm it up by dialing  *8
14:50.54KattyiDunno: k, all better.
14:50.58iDunno:)
14:51.07seele_right?
14:51.09Kattyseele_: *8 is for any ringing line in the building.
14:51.22Kattyseele_: whether it's incoming, or transfer.
14:51.26seele_Ok that's what i wanted to know
14:51.29Kattyk
14:51.31seele_thanks
14:51.50*** join/#asterisk basta (n=basta@213-156-52-98.fastres.net)
14:52.23*** join/#asterisk Jammy (n=jammy@24.244.182.192)
14:52.50*** join/#asterisk lunk (n=lunk@negative-influence.com)
14:53.35KattyiDunno: do you have any breakfast recipes?
14:54.00seele_Do i have to config the Outgoing dial rules, so i can make cellphone calls
14:54.03iDunnoKatty: they normally just involve kettle, mug, instant coffee, sugar and milk...
14:54.13KattyiDunno: horror.
14:54.24seele_In my area cellphone calls start with 033XXXXXXXXX
14:54.29iDunnoKatty: it's what comes of playing the "fall out of bed, head to work" game.
14:54.39KattyiDunno: silly rabbit.
14:55.43KattyiDunno: see, i have company this saturday.
14:55.50KattyiDunno: and, they must be spoiled see.
14:56.04iDunnoKatty: ahhhh! you should cheat - get them to cook breakfast ;)
14:56.13KattyiDunno: he'd probably do it too
14:56.41seele_hello
14:57.08seele_Do i have to config the Outgoing dial rules, so i can make cellphone calls
14:57.19shido6allow them to makew conference calls
14:57.28*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
14:57.30shido6then set them up with pin numbers on your aseterisk box
14:57.37shido6happy holidays chaching
14:57.40*** part/#asterisk Cresl1n (n=matt@gateway.digium.com)
14:58.10*** join/#asterisk jcwunder (n=chris@ppp-82-135-79-235.mnet-online.de)
14:58.19P4C0iCEBrkr, are u there?
14:58.23*** part/#asterisk _fan_ (n=allan@static-64-222-189-188.man.east.verizon.net)
14:59.13TheCops[TK]D-Fender the rest of my config are right ? I already make it work on 1.0.9, since I upgraded the hint dont work
15:01.01*** join/#asterisk cjmoya (n=cjmoya@208.195.223.50)
15:01.33cjmoyahello all
15:02.03Kattymister fender.
15:02.13Katty[TK]D-Fender: what's your favorite breakfast thing.
15:02.34seele_Katty i got some trouble here. I dial *8 when trying to pick up a call and it sounds busy
15:02.46*** join/#asterisk mkrufky (n=mk@68.160.103.77)
15:02.55*** join/#asterisk file (n=joshnet@mctnnbsa24w-142167049176.nb.aliant.net)
15:03.06Kattyfile: there you are.
15:03.15Kattyfile: i know /you'll/ help me
15:03.17cjmoyamister D-fender
15:03.44filewhat am I helping with?
15:03.48seele_Does anybody else knows how to take incoming calls from any extensions??
15:03.57Kattyfile: i require breakfast recipe.
15:04.02Kattyfile: or at least suggestion.
15:04.18fileI fear I have none
15:04.19iDunnocornflakes and milk? full english fry up?
15:04.30fileI wake up at lunch time every day, and go straight to lunch
15:04.33fileno morning for me!
15:04.45cjmoyai installed the te110p card and up the wcte11xp module, also config the /etc/zaptel.conf and /etc/asterisk/zapata.conf
15:04.56Kattyfile: horror.
15:05.05KattyiDunno: full english fry up?
15:05.13KattyiDunno: have you insaned? vegans don't eat such things.
15:05.18seele_Does anybody else knows how to take incoming calls from any extensions??
15:05.20cjmoyathe E1 ISDN up, but the channels in E1 is block...
15:05.21MeHi all, I have been having a problem where I dial ext 300 from my DID and it will sound like its ringing but it does not actually ring the remote extension.  I can dial 300 from another internal extension and it goes through fine.  Also, I can dial any other extension from my DID and it works fine.   Does anyone have any ideas as to what might be wrong?
15:05.26iDunnoahh - full english fry up with no food, then? ;)
15:05.36fileseele_: you can use the directed pickup app...
15:05.37iDunnoit's a plate and some bread and butter, innit?
15:05.48benjkKatty: how about cereals with soymilk?
15:05.53seele_hows that works file
15:06.31Kattybenjk: too easy. i'm spoiling someone.
15:06.40benjkah
15:06.53Kattyusing crescent rolls.
15:06.53fileseele_: you give it an extension and (if you want) context and it'll try to pickup any calls that are ringing to that extension
15:06.57Kattyand soy sausage, veggies, etc.
15:07.05Kattyshredded hashbrowns too, possibly.
15:07.08Kattyand strawberry compote!
15:07.22seele_file, yes but that's a permanent option
15:08.06benjktofu with maple sirup
15:08.13fileseele_: uh... what do you mean?
15:08.23seele_i'd like to know what to dial when there is an incoming call and im far from the extension it's ringing
15:08.33benjkrun
15:08.36fileexactly...
15:08.40fileexample:
15:08.46fileexten => 145,1,Dial(SIP/jcolp)
15:08.55fileexten => _7X.,1,Pickup(${EXTEN})
15:09.13fileif I dial 7145, it'll pick up any calls that are ringing on that extension
15:09.26filedo you mean as a group?
15:09.38cjmoyawho know a softphone sip free for windows?
15:09.46filewithout knowing the extension perhaps?
15:09.59Mecjmoya  xlite
15:09.59benjksip free as in IAX?
15:10.08Mewww.xten.com/index.php?menu=download
15:10.32cjmoyathanks...
15:10.40Meany time
15:11.22*** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com)
15:11.25Nivexooooh... gotta remember that trick.  thanks file.
15:12.34Kattyfile: i have a suitable recipe. cresent rolls as pizza dough, soy sausage,maple syrup, mushrooms, onions, green peppers, shredded hashbrowns, and oregano as top. plus strawberry smoothy.
15:12.57Mehow close does soy sausage taste to the real thing?
15:13.17KattyMe: quite close.
15:13.26KattyMe: especially if you mix it with maple syrup.
15:13.29benjkdepends on what you mean by real thing
15:14.13*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivflkj.dialup.mindspring.com)
15:14.13benjkcause there are so many different sausages
15:14.27*** part/#asterisk mhnoyes (n=mhnoyes@user-2ivflkj.dialup.mindspring.com)
15:14.32Mejust which ever kind.
15:14.40MeI was being purposefully vague
15:15.07KattyMe: perhaps you should try it and find out for yourself.
15:15.17MeI probably will at some point
15:15.30seele_file, that example how should i apply it, in what .conf file?
15:15.36MeI have been drastically reducing the amount of meat that I eat over the last few weeks
15:15.57benjkI used to buy soy based meat-substitute cubes for making stew
15:16.05benjkback in Europe
15:16.05fileseele_: it's for dialplan logic, extensions.conf
15:16.25benjkthis was pretty close to beef stew
15:16.33Meintersting
15:16.36*** join/#asterisk CleanerX_idle (n=jens@nat-ph3-wh.rz.uni-karlsruhe.de)
15:16.55MeHow about nutritional yeast, you guys eat any of it?
15:17.16seele_file, the way i see it i have to config that dialplant for every extension i make,
15:17.23benjkif you make a nice wine based stew sauce or an Irish stew with Guinness, its very nice
15:17.30fileseele_: if you use pattern matching, no
15:17.43seele_file, and that is?
15:17.49fileexten => _7X.,1,Pickup(${EXTEN})
15:17.54seele_file, i'm a newbie at this
15:17.57fileprefix the extension with 7 and it'll try to pick it up
15:18.03file718005558355, 7145, 75000
15:18.05benjkI already asked Katty that, but ever got any answer
15:18.05filewhateva
15:18.07KattyMe: yep.
15:18.12fileand that should be ${EXTEN:1}
15:18.13filesilly me
15:18.18fileI'm not awake yet today
15:18.27Kattyfile: you're never awake dear.
15:18.53benjkKatty what about the lacto-bacteria in dough, and especially in sour-dough?
15:19.05tzangeroh lord.  this again
15:19.08Kattybenjk: bacteria is not in the animal kingdom.
15:19.15benjkwell, you tell me
15:19.21Kattybenjk: i just did.
15:19.56benjkI don't know where to draw the line, I even accept that plants are living beings
15:19.57ManxPowerCOOL!  The power company is here to install another telephone/power pole!
15:20.02cjmoyaMr me, one question...
15:20.06fileManxPower: yay
15:20.07benjkbut I wouldn't stop eating because of that
15:20.17tzangerManxPower: ?
15:20.19Kattybenjk: of course they are.
15:20.32benjkthere is a thing called the baxter effect
15:20.41Kattyand a thing called the katty effect.
15:20.47ManxPowertzanger, they needed to install another telephone/power pole to get me service.
15:20.53benjkit establishes that plants have a sense of consciousness
15:21.27benjkfor example if you cut a tree down, the tree is aware that it is going to die
15:21.36fileManxPower: are you still considering covering the mountain with wifi? :D
15:21.48ManxPowerfile, considering.
15:21.54tzangerthat katty effect is when she bitch-slaps you across the back of the head for putting her on the spot again :-)
15:22.11cjmoyaMr Me, the xlite softphone can use with asterisk?
15:22.37*** join/#asterisk rikstah (n=rick@87.113.11.91.bbplus.pte-ag1.dyn.plus.net)
15:22.38benjkso maybe the only thing we can eat without harming anything is things like fruits which don't harm the plant if you pick them
15:22.41*** join/#asterisk ast_freak (n=jesse@68-112-134-195.dhcp.stcl.mn.charter.com)
15:22.46tzangerManxPower: that's remote!  wow
15:22.52Kattybenjk: fruitarian.
15:23.03benjkbut no salad, no cereals etc
15:23.07Meyes, look for the book Asterisk TFOT chapter 4
15:23.11Kattytzanger: i love you ;)
15:23.12ManxPowertzanger, there was only 2 pair coming into the location from the telco.
15:23.14MeI will send a link in a second
15:23.23tzangerKatty: ?
15:23.28Methis damned adobe is locking up
15:23.35Kattytzanger: your katty effect comment.
15:23.40Kattytzanger: &heart;
15:23.45tzangerKatty: haha
15:23.56tzangerManxPower: and they couldn't use that two pair to bring you in a nice hDSL T1?
15:24.10ManxPowertzanger, I'm not bringing a T-1 for a while.
15:24.14ManxPower$500/month at least.
15:24.27tzangerManxPower: ok, and the tower is so laden now that it cant' support a 50pr trunk?  :-)
15:24.37ManxPowerI think they are bringing 25 pair for the campground.
15:24.39P4C0guys, does anyone know a good doc that shows how to use asterisk as a sip client to register with a sip provider and send and receive calls??
15:25.01MeP4C0: look up Asterisk TFOT
15:25.13MeIt describes how to do it pretty well
15:25.19P4C0Me, TFOT?
15:25.24benjkI don't mean to say I don't respect your choice of not eating any animal products, but the point is that you cant really claim to be non-violent against living creatures because of it
15:25.25P4C0Me, where?
15:25.29Medo a search on google for it
15:25.36P4C0Me, ok thanks
15:25.36Memy browser is locked up
15:25.41Meso I can't send you a link
15:25.48Mejust now
15:26.01cjmoyaMr Me, where find TFOT?
15:26.06*** join/#asterisk marv[work] (n=timr@64.89.118.139)
15:26.11Medo a search on google for it
15:26.20Kattyyay for google!
15:26.28tzangerwow that rooster sauce is hot
15:26.30tzangerbut GOOD
15:26.39Kattytzanger: is it made from roosters?
15:26.40Mecjmoya: http://www.google.com/search?hl=en&q=asterisk+tfot+xlite&btnG=Google+Search
15:26.45*** join/#asterisk ruza (n=ruza@holly.cervenytrpaslik.cz)
15:27.44Mecjmoya: http://www.google.com/search?hl=en&lr=&safe=off&q=asterisk+tfot+sip+configuration&btnG=Search
15:28.03tzangerKatty: lord I hope not
15:28.16benjktzanger: do you mean peri peri sauce?
15:28.18Me<P4C0>  that last link was meant for you, I can't cut and paste
15:28.28tzangerbenjk: it's a korean chili sauce
15:28.40Mewhy you not eat meat Katty?
15:28.40benjkah, sambal oelek
15:28.49MeEthical reasons?
15:28.53MeMedical Reasons?
15:29.09P4C0Me, thanks
15:29.11MeI have recently found that meat is just not appealing to me
15:29.17MeP4CO Anytime
15:29.17Kattytzanger: ^_^
15:29.24KattyMe: i don't want to eat animals.
15:29.25MeP4CO look at that first link there on google
15:29.49[TK]D-FenderI like my filet mignon blue & seared :D
15:29.50benjkKatty: but why no milk then?
15:30.01Kattybenjk: milk is commercial explotation.
15:30.13MeKatty, would you drink raw milk from a family farm?
15:30.18KattyMe: yes.
15:30.23KattyMe: however that is illegal.
15:30.33MeNot in all states
15:30.34[TK]D-FenderKatty : pfftt!
15:30.37benjkso if you got your milk from a subsistence farmer who's got a single cow, would that make a difference
15:30.39MeHere it is
15:30.40KattyMe: it's illegal in the usa.
15:30.41[TK]D-Fender*sigh* mortals.......
15:30.47Mein NC it is illegal
15:30.47KattyMe: you cannot buy milk that has not been through processing.
15:30.59KattyMe: FDA requirement.
15:31.01MeYou can buy it in CA, WA
15:31.12Katty[TK]D-Fender: you know you love us.
15:31.25fugitivomilk is bad
15:31.47Kattyfugitivo: those who can drink it are genetic mutants.
15:31.47benjkI could perhaps do without meat
15:31.52fugitivowww.notmilk.com
15:31.53benjkbut without cheese?
15:31.55benjknever
15:32.04implicitKatty, you always come and spout off bullshit
15:32.09Kattyimplicit: yay!
15:32.18implicitglad to see me?
15:32.26Kattyobviously.
15:32.29fugitivopeople shouldn't drink or eat milk based food
15:32.31implicitok
15:32.41benjkwhy is it bullshit? it;s her choice
15:32.43antoniosmy only problem not drinking milk is the coffee. It doesnt taste as nice with soy milk
15:32.46benjkpeople are free to choose
15:32.50MeI hate milk, I would never drink it.  But I will use it in some food
15:32.56Kattybenjk: and people are free to call others idiots too.
15:33.04implicitMe: you hate the *IDEA* of milk
15:33.06Kattybenjk: if it makes him feel better to say i'm spouting off bullshit, let him ;)
15:33.15benjkyeah, well, but that's not exactly good etiquette
15:33.19implicitbenjk, and it does make me feel better
15:33.19Kattyhahah
15:33.23Kattygood etiquette
15:33.24Kattyin here?
15:33.26Kattythat'll be the day.
15:33.31implicitbenjk, seriously
15:33.36implicitbenjk, go screw yourself :)
15:33.39*** join/#asterisk jmolenski (n=jjones@216.147.224.254)
15:33.39*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
15:33.39*** mode/#asterisk [+o anthm] by ChanServ
15:33.41fugitivookey, there're cientific reasons for not drinking milk, just people don't know that
15:33.47tzangeryou can buy 'raw' milk here so long as you sign a disclaimer.  I've been meaning to do that with a neighbour of mine
15:33.51iDunnowtf?! no dairy products?! christ, no meat, no dairy products, can't eat vegetables because they're obviously evil...
15:33.59anthmhi
15:34.03KattyiDunno: veggies will eat you ALIVE
15:34.07benjkfugitivo: I don't care about the milk, but what about the cheese?
15:34.11anthmyou teasing ppl again?
15:34.17Kattyanthm: i'm always teasing people.
15:34.18fugitivobenjk: cheese is milk based
15:34.21MeWhat state are you in Katty?
15:34.24benjkyeah I know that
15:34.26KattyMe: alive.
15:34.28antoniosanimal farming is very bad for the envirnment, a big % of the global warming gasses are being farted by cows
15:34.28implicitvegtables and meat are what animals eat to stay alive
15:34.30fugitivobenjk: it's bad too
15:34.32KattyMe: also On.
15:34.32iDunnodrunken? *grin*
15:34.33tzangeranthm: yeha she's a shit disturber
15:34.33benjkalso cake
15:34.34implicitso i don't think either one is good to eat
15:34.37tzangerBURN HER!  BURN HER!!
15:34.40Kattytzanger: :>
15:34.40implicittheir are scientific reasons for starving yourself
15:34.41tzangerer I mean BAN HER!! BAN HER!  :-)
15:34.44Kattytzanger: at least it will be warm!
15:34.54implicitpeople just don't know it yet
15:35.14iDunnoimplicit: is it to cut population levels? ;)
15:35.14Mefunny Katty
15:35.18implicitfugitivo, see how retarded your logic is now?
15:35.18fugitivowww.notmilk.com
15:35.24implicitiDunno, just making fun of fugitivo
15:35.27fugitivoimplicit: it's not my logic
15:35.35implicitfugitivo, it is your dick, small as hell
15:35.41Kattyi've started another arguement.
15:35.42Kattygo me!
15:35.44fugitivooh well
15:35.49tzangerwho removed the +r?  THANK YOU THANK YOU THANK YOU THANK YOU THANK YOU THANK YOU THANK YOU
15:35.52benjkeverything is bad if consumed in overdoese
15:35.56benjkoverdoses
15:36.13fugitivobenjk: we're not talking about "overdoses"
15:36.16Nugget/sajoin tzanger #asterisk-unregistered-ghetto
15:36.17Nuggeterp
15:36.18*** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm)
15:36.20CleanerX_idleyeah even ideoligical wars...
15:36.20iDunnobenjk: nah - binge drinking is very good for your health ;)
15:36.23benjkbut I do
15:36.23MeKatty: http://www.realmilk.com/milk-laws-1.html
15:36.58*** join/#asterisk dalabera (n=dalabera@pmr.pmrtechnologies.com)
15:38.10benjkit is true that milk drinking is often overemphasised, but there is nothing wrong with cheeses or using milk in pastries, unless of course you have ethical objections, but that's a different story
15:38.35fugitivobenjk: there's a good reason
15:38.53implicithahaha, this notmilk shit is funny
15:38.55iDunnoit's good to drink milk before it goes off.
15:39.12implicit'drinking whole milk 3 or more times daily had a 2-fold increase in lung cancer risk compared to those never drinking whole milk'
15:39.15iDunnodrinking it after it's gone off I'd highly discourage.
15:39.21CleanerX_idle<PROTECTED>
15:39.23implicitwho the FUCK drinks WHOLE MILK more than 3 TIMES DAILY!
15:39.26fugitivobenjk: a human baby, needs months to duplicate his weight drinking human milk
15:39.38iDunnoimplicit: how much is "one time"?
15:39.45fugitivobenjk: a "baby cow" needs 2 weeks to duplicate his weigth
15:40.12implicitiDunno, it doesn't say anything about 'one time', but the funny thing is that it doesn't say much about 'milk'
15:40.22fugitivobenjk: the reason of that, is the quantity of calories and fat that cow milk has
15:40.23benjkyeah, well, if you eat cheese the bacteria in the milk have entirely transformed the milk into a different product
15:40.32fugitivobenjk: that's not normal for a human
15:40.36implicitbenjk, screw the bacteria
15:40.46benjkbacteria are good
15:40.54implicitbenjk, i know
15:40.58benjkespecially lacto-bacteria
15:40.58implicitbenjk, scrwe them anyway
15:41.03implicitok i am going to go eat
15:41.07implicitsick of this bullshit milk stuff
15:41.09tzangerI should get some sourdough mmmm
15:41.16MeMaybe they need to change the name of this room from #asterisk to #milk ?
15:41.17benjkyeah, well that's why I eat 'em :)
15:41.19coppicebenjk: sounds a bit like letting someone else maintain your software :-)
15:41.39benjk#lacto-asterisk
15:41.46iDunno(did I mention yet today how much I hate Java?)
15:42.02impliciti am going to go drink some milk right now
15:42.24implicitand lick nipples for fun also
15:42.29antoniosmilk is number envirnmental hazard :)
15:42.43antoniosnumber one that is
15:43.13benjktzanger: sourdough has lacto-bacteria in it, so if you are in the anti-milk camp, you may want to reconsider :)
15:43.13coppiceiDunno: do you hate your Java more with milk or black?
15:43.32fugitivobenjk: cow milk permit a fast grow of good and bad cells, that's why it's associated with cancer too
15:43.35tzangerbenjk: I have no problem eating any of God's creatures
15:43.36jmolenskiam i in the right place?  i thought i was going to talk about the open-source pbx...
15:43.40tzangerwell, not the tasty ones anyway
15:44.05tzangerjmolenski: no that's not this channel
15:44.10Me<jmolenski> : You need to go to #asterisk, you are now in #lacto-asterisk
15:44.22*** part/#asterisk jmolenski (n=jjones@216.147.224.254)
15:44.30iCEBrkrLOL
15:44.36MeI guess he believed it
15:44.38*** join/#asterisk frenzy (n=frenzy@80.255.63.30)
15:44.41coppiceliving in asia, I've eaten a pretty wide range of the earth's creatures. god didn't obviously seem to be claiming ownership of any of them, so what the heck
15:44.45iDunnocoppice: now that Java I like - possibly with a bit of milk and maybe some sugar :)
15:44.46benjkI guess I should register a trademark on that
15:44.59Megood idea ben
15:45.05benjkNow, sugar ... that is really bad stuff
15:45.20Mewhat about raw sugar?
15:45.21iCEBrkrScrew you hippies
15:45.32Meare you cute ICEBrkr?
15:45.37fugitivosugar is not bad
15:45.42*** join/#asterisk jmolenski (n=jjones@216.147.224.254)
15:45.43fugitivoit comes from a plant
15:45.51iCEBrkrMe: Butt ugly.
15:45.54Merefined sugar is a poison
15:46.02Methen no thanks iCEBrkr
15:46.08benjkjmolenski: welcome to #lacto-asterisk
15:46.13coppiceI know some bad people who seem to be vegetables
15:46.49jmolenskimethinks i've been hood-winked... oh well...
15:46.51jmolenski:)
15:47.04frenzyhey all
15:47.20frenzyis there a channel for ms-excel geeks ?
15:47.39frenzyunless we got some excel geeks in here
15:47.40frenzy:)
15:47.42benjkif it's got anything to do with milk, it's ok to discuss it here
15:47.51fugitivofrenzy: openoffice calc is the same?
15:47.56*** join/#asterisk steff (n=steff@80.125.254.220)
15:48.09benjklacto-office
15:48.24benjklacto-calc
15:48.44rikstamOOcalc
15:48.49rikstamine's better ;)
15:49.06fugitivoc0wlc
15:49.12frenzywhats with all the miking
15:49.13frenzy:)
15:49.20frenzymilking.... *
15:49.32coppiceif you have a moral problem with cow's milk, stick to human milk
15:49.42fugitivoyeah, human milk is good
15:49.50frenzyfugitivo: ewww
15:49.52fugitivobut hard to find...
15:50.09*** join/#asterisk P4C0 (n=paco@200.124.22.34)
15:50.10riksta"BITTY" anyone in the UK will get that ;)
15:50.19coppicea bit like creamy carror juice
15:50.27benjkgoats milk is nice
15:50.28coppices/carror/carrot
15:50.34benjkespecially as a base for cheese
15:50.57rikstahmm decisions, decisions, do i dare to upgrade my production server to 1.2.1 in the middle of the night tonight?
15:50.59tzangerI've heard that goatmilk cheese is very good
15:51.06fugitivoriksta: yes, do it
15:51.10rikstatzanger, it is indeed
15:51.13benjktzanger it is
15:51.18rikstafugitivo, easy for you to say :P
15:51.28coppicegoat's milk yoghurt is good too
15:51.28benjkevery had Roquefort?
15:51.47rikstayeah man roquefort cheese on a steak
15:51.54jmolenskiBitty, Mommy....
15:51.58rikstabitty!
15:51.59_Sam--im testing the "AstBill Live CD"...does anyone know how to get it to run entirely from CD and not use the HD?
15:52.30riksta_Sam--, what does it use the HDD for
15:52.39jmolenskii assume we're talking about the same video?  what a riot...
15:52.41_Sam--i dont know...but it keeps trying to access it when it boots
15:52.56trixtergiven the fact that astbill will do a chmod 777 on the database files in their regular install script if it doesnt by defaul;t it doesnt at all
15:52.59_Sam--and the HD in the box is dead
15:53.06_Sam--i tried some boot options
15:53.18*** part/#asterisk frenzy (n=frenzy@80.255.63.30)
15:53.19rikstajmolenski, it's from a comedy series in the UK called little britain, where a middle age man asks for "bitty" from his mum in the middle of a resteraunt n stuff
15:53.35*** join/#asterisk krstone (n=krstone@ool-4573f3dc.dyn.optonline.net)
15:53.36riksta_Sam--, unplug it then?
15:53.43jmolenskiyep, must be what i saw a clip from...
15:53.52trixterhmm, so it only tries to access when it boots..  well the bios will prolly scan the drive the linux kernel will ...
15:54.07*** join/#asterisk umay (n=chris@65-37-2-236.nrp2.roc.ny.frontiernet.net)
15:54.08trixternothing more3 than a 'who are you' type thing by the kernel
15:54.16rikstathe kernel will be looking, just disable the IDE stuff in the bios
15:54.20rikstafor the primary chan or something
15:54.23_Sam--just did....trying now
15:55.42*** part/#asterisk krstone (n=krstone@ool-4573f3dc.dyn.optonline.net)
15:56.21*** part/#asterisk cfh (n=luca@82.193.23.6)
15:58.05*** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek)
15:58.25P4C0how many Answers do I have to put?? I mean if I go throw a lot of menus and submenus do I have to answer each time I do a GoTo? (new context?)
15:58.42fileP4C0: no.
15:58.53P4C0file, no? and waits?
15:58.58fileAnswering once is fine...
15:59.04*** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com)
15:59.04nfi|ermeshow can i let the music play till the extrension ring ???
15:59.11TheCopsSomeone have Snom phone and have some difficulty with presences support ? I upgraded to 1.2.1 from 1.0.9 and now it's not working
15:59.29fileP4C0: answering multiple times in the dialplan won't hurt though
15:59.31ManxPowerI should have known they would shut off the power for a few mins
15:59.41*** join/#asterisk zishanov (n=mail@d57-249-149.home.cgocable.net)
15:59.49filenfi|ermes: read the available options for the Dial application - ie: type show application dial at the asterisk console
16:00.03*** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net)
16:00.49zishanovI have problem with MoH. On CLI, it says Started Music on Hold,... and then immediately says Stopped music on hold. Why is it doing this
16:01.12filezishanov: are you using native files, or mpg123? have the right mpg123 version if so?
16:01.30*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
16:01.36zishanovits whatever comes with Asterisk 1.2.0, I didn't install any new software
16:01.38trixterohh today I get all the goodies from www.thevoipconnection.com - prizes for the sacramento AUG contests
16:02.09filezishanov: then you need to if you want MOH to work, there's instructions online - I suggest reading them...
16:02.10file~docs
16:02.12jbotextra, extra, read all about it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
16:02.13P4C0file, thanks
16:02.40trixterhttp://os.newsforge.com/article.pl?sid=05/12/27/1715239  GNU TelephonyStack Announced
16:02.49zishanovI have all the mpeg files in /var/lib/asterisk/mohmp3
16:03.20fileDavid Sugar... I met him!
16:03.37zishanovfile: where are the online instructions. The ones on wiki I already followed, there is not much there, didn't help
16:03.45filezishanov: you need to install mpg123 0.59r
16:03.57filezishanov: this can be done by typing "make mpg123" in the asterisk source directory, and then "make install"
16:04.14zishanovok, let me try that
16:04.36zishanovP4C0, how are you, did your asterisk work, did it receive calls?
16:05.43MeHmmmmmmm, I wonder if someone who has managed to find this irc room has probably read the * documentation?
16:05.59filethis channel is mentioned in documentation... I'll say that
16:06.09fileI highly recommend the O'Reilly book though
16:06.17cjmoyaMr D-fender...
16:06.17P4C0zishanov, yes :)
16:06.30P4C0zishanov, and I can call also :)
16:06.38Meonly Sending out generic links to the main doc files is rude
16:06.41zishanovP4C0: so what was the problem, did you figure it out
16:06.47Mein a case like that
16:06.51P4C0zishanov, yep
16:07.14zishanovP4C0: I'd like to know what was it, something in sip.conf
16:07.20fileMe: it's 12PM, and I'm working - I'm not going to Google for the exact document
16:07.26filecatch me when I'm off work and I might have :P
16:07.32malverian[work]Is there an easy way to subscribe to DTMF events from a channel with the manager API?
16:07.32P4C0zishanov, I missed the canreinvite=no and fromuser and fromdomain in mysipprovider-out in sip.conf
16:07.36Meits ok, its a pet peeve of mine
16:07.47*** part/#asterisk cjmoya (n=cjmoya@208.195.223.50)
16:07.48P4C0now I'm having problems with the voicemail...
16:07.51*** join/#asterisk tengulre (n=tengulre@219.145.57.171)
16:07.58jmolenskii gots a question... and i've looked pretty hard for an answer, so please be nice :)  what audio format should i save files in for use in my auto attendant in amp?
16:08.02zishanovfile: I didn't make mpg123, but doing make install will reinstall all the asterisk, right
16:08.08tengulreHi,all
16:08.15tengulreI M BACKING!
16:08.18Metelling someone to google for a set of keywords is good, but I hate going into a support forum after reading the docs and someone saying "read the docs"
16:08.21slappingtare there any good forums for running Asterisk on a Mac?
16:08.23filezishanov: you need make mpg123, it builds mpg123 for install, and make install reinstalls the headers/modules/everything except configuration files
16:08.37zishanovP4C0, I'd recently setup up asterisk right from scratch, everything works, except now for MoH
16:08.59zishanovso maybe I can help you with voicemail, for me its working perect, its fun
16:09.07P4C0zishanov, MoH? do you use the voicebox?
16:09.12MeWhat *nix distribution you running Z?
16:09.17tengulreanybody use Speakfreely??
16:09.17P4C0zishanov, sure :)
16:09.18slappingti have been reading astmasters site
16:09.25*** join/#asterisk cjmoya (n=cjmoya@208.195.223.50)
16:09.26zishanovmaybe you should read the book Asterisk, Futuer of the Telephony, it has all the step by step instructions.
16:09.31*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)
16:09.31tengulreI m point under Asterisk!
16:09.45zishanovFirst you setup the voicemail accounts in voicemai.conf
16:09.51zishanovvoicemail.conf
16:10.01P4C0zishanov, let me pass you my voicemail.conf...
16:10.09*** join/#asterisk wt (n=wt@adsl-070-145-131-253.sip.mem.bellsouth.net)
16:10.10zishanovok
16:10.10Zeeekslappingt read "scotts place"
16:10.27slappingtthanks Zeeek
16:10.42Medoes anyone know why debian's apt-get isntalls mpg321 when you try to install mpg123?
16:10.49Zeeekslappingt : http://scottstuff.net/blog/articles/category/Asterisk/?page=4
16:10.55Zeeekbacktrack from there
16:10.59MeI screwed with that for about 45 minutes
16:11.01*** join/#asterisk a1fa (n=a1fa@207.210.210.202)
16:11.01P4C0zishanov, http://pastebin.com/481645
16:11.02Methe other day
16:11.04malverian[work]Anyone?
16:11.07a1fahey.. i am trying to setup speed-dial
16:11.20a1faexten => 300,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
16:11.30tengulreanybody use SpeakFreely in asterisk??
16:11.36a1faexten => 300,1,dial(SIP/13002003427@sip.broadvoice.com,30) ?
16:11.51zishanovP4C0, it is ok, so what happens when you try to send the caller on voicemail?
16:11.52a1fais that how you setup speed dial?
16:11.55MeZeek, what you wanting to connect to?
16:12.11malverian[work]a1fa, If you want "300" to dial 13002003427 then yes.
16:12.13MeI scerwd that up sorry Zeek
16:12.14tengulreor anybody know which website can download SIP client application and it have source code!
16:12.20Zeeekthat was a link about asterisk + mac
16:12.29MeWhat you wanting to connect to <tengulre> ?
16:12.30slappingtThanks Zeek,  I can spend the day reading while I wait for my Sipura 3000 to arrive tomorrow.
16:12.34a1famalverian[work] : cool
16:12.35P4C0zishanov, I have this for calling the mailbox: exten => 1717,1,VoiceMailMain(${CALLERIDNUM})
16:12.41a1famalverian[work] : do i need anything else?
16:12.54a1falike congestion and busy?
16:13.09Zeeekslappingt I don't have a mac, but that site has some interesting stuff on it including spandsp/faxing instructions
16:13.09*** join/#asterisk Connor_ (n=Connor@198-144-174-5.knx.tn.nxs.net)
16:13.11P4C0zishanov, the voicebox answers but don't give me options, when I press 1 it ask for password, then loging incorrect console shows app_voicemail.c:4947 vm_authenticate: Couldn't read username
16:13.30Connor_anyone tried porting a number away from Level 3 ?
16:13.36zishanovcan you show me the extensions.conf
16:13.43P4C0zishanov, sure
16:13.49tengulreMe: I want to use a SIP client + Asterisk+Digum card to building a voip platform!
16:13.53cjmoyawho know about howto document in spanish for installing asterisk
16:14.23Meten: what you need speekfreely for?
16:14.35tengulreMe: but I can found a free SIP client and source code under windows.
16:14.39Zeeekpc40 does this happen when you dial 1234# ?
16:14.41Meyeah xlite?
16:14.44malverian[work]a1fa, Wouldn't hurt.
16:15.00tengulreMe: It have source code?
16:15.00benjkcjmoya: there is a spanish asterisk website
16:15.06a1famalverian[work] : what benefits?
16:15.08P4C0zishanov, http://pastebin.com/481646
16:15.19malverian[work]a1fa, Will return busy to the SIP channel you dialed from.
16:15.34malverian[work]a1fa, Though it may do that automatically now.. not sure.
16:16.01*** join/#asterisk theNOTO (n=biggs@69-165-25-59.clvdoh.adelphia.net)
16:16.03a1faok
16:16.42a1faexten => ,2,congestion() ; No answer, nothing
16:16.55a1faor i need anything else?
16:17.38zishanovP4C0: It seems all good
16:17.38a1fai wonder if i need a number
16:17.40a1fabefore i
16:18.14zishanovwhen you enter 1717, it should say Comedian Mail, Mail Box...
16:18.22a1falol
16:18.23a1fa:P
16:18.23zishanovthen you enter the mail box number
16:18.28zishanovdoes it do that?
16:18.39P4C0zishanov, humm I think that I need to specify the name not the numbers... cause the call is for example: Executing VoiceMailMain("SIP/ruben-7d9d", "ruben") but theres no "ruben" on voicemail.conf
16:18.44*** join/#asterisk cianhughes (n=cian@87.192.36.98)
16:18.46tengulreMe: do it have source code, I m point xlite?
16:19.08P4C0zishanov, and I'm getting Incorrect password '782' for user '1' user 1?? wtf!
16:19.27P4C0zishanov, can you pastebin all your extensions.conf sip.conf and voicemailc.onf!? :p
16:19.58zishanovI've never used pastepin, i don't know how to use it
16:20.10iDunno~pb
16:20.12jbotpb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
16:20.48zishanovtry to remove ${CALLERIDNUM}, just make it (), this is what I have from my book
16:21.01*** join/#asterisk Utah_Dave (n=boucha@0-1pool139-113.nas28.salt-lake-city1.ut.us.da.qwest.net)
16:21.14*** join/#asterisk _mistral (i=mistral@jstevenson.plus.com)
16:21.37zishanovthen in your voicemail.conf, try to change password to 1111, i.e. 722 => 722,Paco,manuel.arguelles@wirelesszt.com
16:21.42P4C0zishanov, just go and paste ;) con then ok and then copy past the url in the channel or pm it to me
16:21.43_Sam--im trying to learn the ASTBill Live CD....when it boots, how do you configure your asterisk stuff?
16:22.14P4C0zishanov, wait
16:22.36zishanovi.e. 722 => 1111,Paco,manuel.arguelles@wirelesszt.com
16:22.50zishanovdo you have 722 defined in your SIP
16:23.13zishanovalso exten => 1717,1,VoiceMailMain()
16:23.21P4C0zishanov, ${CALLERIDNUM}
16:23.21zishanovtry this, this is all what I have
16:23.35P4C0why 1111?
16:23.56P4C0the problem is that voicemail can't read the username
16:24.26zishanovdo you have a 722 user extension configured in sip.conf
16:25.09zishanovI don't see any user 722 in your extensions.conf, which means you don't have any user 722
16:25.12*** join/#asterisk Twister (n=jason@216.30.232.106)
16:25.14ZeeekUse Voicemail Reference : http://www.bluelavasoftware.com/BLWeb/pub/BLWeb/ResourceLibrary/vm_ug.pdf
16:25.30*** join/#asterisk mistral (i=mistral@jstevenson.plus.com)
16:25.33zishanovactually sorry, it is there
16:25.52P4C0zishanov, yes 722 is my extension
16:25.56Twisterhey all, can somoene direct me to a good site for integrating Asterisk and Avaya Partner ACS
16:26.04Zeeekoops wrong link!
16:26.07Twisteri have r3
16:26.41P4C0zishanov, [paco] type=frien username=paco secret=paco host=dynamic context=office mailbox=722@voiceboxes <-- sip.conf
16:28.18*** join/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net)
16:29.42P4C0Zeeek, thanks for the link, but I first need to get into that menu ;)
16:29.49zishanovI think there is missing '@voiceboxes'
16:29.56jake1932Twister: did you check the wiki?
16:29.58zishanovit should be exten => 722,n,Voicemail(u${EXTEN}@voiceboxes)
16:30.06a1fai love my MOH
16:30.23zishanovtry this and see if it works now
16:30.37P4C0zishanov, right now I just one the user to dial into the mailbox extension (1717) so he can configure his/her box
16:30.39a1faDec 28 16:23:12 WARNING[25873] pbx.c: Timeout, but no rule 't' in context 'myphones'
16:30.44ZeeekPC40 : yeah wrong link. Try http://www.asteriskguru.com/tutorials/asterisk_voicemail.html
16:30.45a1fai hate this
16:30.54P4C0Zeeek, :)
16:30.57Zeeekstill not the one I was thinking of, but it might help
16:31.02*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
16:31.18ZeeekThere used to be a page that had everything there is to know about the subject but I can't find it now
16:31.19a1fathe timeout is in macro
16:32.11Zeeeknees to e in myphones apparently
16:32.28zishanovP4C0, add @voiceboxes after ${EXTEN} for all the extensions, like exten => 722,n,Voicemail(u${EXTEN}@voiceboxes)
16:32.28Zeeekyou can't argue with an error message
16:32.39a1fa:P
16:32.42P4C0zishanov, oks
16:33.16zishanovfile: how can I check which mpg123 version I am using?
16:34.10P4C0zishanov, done
16:34.33ManxPowermpg123 -v
16:34.44zishanovManxPower, thanks
16:35.42zishanovManxPower, if it says command not found, does this mean the mpg123 is not installed?
16:35.53zishanovI tried it both on Asterisk CLI and Linux CLI
16:36.36zishanovP4C0, how is it now
16:36.51P4C0zishanov, wait
16:36.59P4C0I'm trying to call in to test
16:37.11a1fai want to give a busy signal, do i ned to enter extension or i can exten => ,2,busy()?
16:37.19benjkits usually /usr/bin/mpg123
16:38.18P4C0zishanov, app_voicemail.c:2384 leave_voicemail: No entry in voicemail config file for ''
16:39.04ManxPowerP4C0, We can't help you unless you paste the line from extensions.conf that runs Voicemail
16:39.11zishanovP4C0, did you restart your asterisk?
16:39.12benjkif you are on a distro that's rpm based try this: rpm -q mpg123
16:39.14P4C0ManxPower, ok
16:39.17P4C0zishanov, sure
16:39.50P4C0ManxPower, http://pastebin.com/481646 here u go
16:40.00a1fayo anybody?
16:40.10zishanovalso you don't need to call in all the time to test it. Instead make an extension like exten => 7777,1,Goto(incoming,s,1)
16:40.25zishanovreplace incoming with whatever context you're using for incoming calls
16:40.28P4C0zishanov, I think that u${EXTEN} is not working....
16:41.26zishanovwhen you call in, after waiting for 10 second is your call redirected to voicemail with unavailable voice prompt?
16:42.20P4C0zishanov, no, it hangup and in the logs I get: app_voicemail.c:2384 leave_voicemail: No entry in voicemail config file for ''
16:42.29a1fai get 403 forbiden on my speed dial, wtf?
16:43.20zishanovI don't see you added what I'd told you to add, i.e. '@voiceboxes' after all the '${EXTEN}'
16:43.25*** join/#asterisk PoWeRKiLL (n=PoWeRKiL@bzq-84-109-161-65.red.bezeqint.net)
16:43.39ManxPowerWell @voicemailcontext at least
16:44.00zishanovit should be like:
16:44.01zishanovexten => 722,1,Dial(SIP/paco,10)
16:44.01zishanovexten => 722,n,Voicemail(u${EXTEN}@voiceboxes)
16:44.01zishanovexten => 722,n,HangUp()
16:44.15zishanovbecause your voicemail context is [voiceboxes]
16:44.17P4C0zishanov, that's the way I have it
16:44.29zishanovso then change it
16:44.41zishanovI mean, I don't see it in the pastebin
16:44.54*** join/#asterisk RoadRunnR (n=MrRoadRu@213.187.82.17)
16:45.05P4C0zishanov, http://pastebin.com/481687
16:45.07ManxPowerP4C0, when you reuse the same pastebin browser and proxies will cache the info and not allow people to see the changed infor
16:45.29P4C0ManxPower, yep, http://pastebin.com/481687 the new one
16:45.48a1fai get 403 when my call times out.. what is up with that
16:45.51ManxPowerP4C0, do a reload.  If it still doesn't work the paste the ONE line that shows up on the CLI with Voicemal in it.
16:46.18RoadRunnRhi all, what is the best place to find support information for chan_misdn? beronet seems to have only the bugtracker, and the maillinglist archives don't have much infos about it as well
16:46.27a1faexten => ,2,congestion() ; No answer, nothing
16:46.30P4C0ManxPower, ok
16:46.31a1fais this allowed?
16:46.48tzangerdo I still need to use ${} when using functions?
16:46.52trixteryou dont have an extension
16:46.57ManxPowera1fa, NO, because you never sopecify an extensions
16:47.04a1faok.. what should i do
16:47.07a1faput a .
16:47.09a1fain there?
16:47.10tzangeri.e. Dial(Zap/user@foo/CALLERID(number) or ${CALLERID(number)}
16:47.37ManxPowera1fa, how about putting in the extension
16:47.39*** join/#asterisk bweschke (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net)
16:47.57a1faexten => 30.,2,congestion() ; No answer, nothing
16:47.58*** join/#asterisk BugKham (n=lamer@203.130.150.139)
16:48.00a1fai dont want it
16:48.01seele_is posible configurate gnugk in aah?
16:48.03a1faits my speed dial
16:48.09*** join/#asterisk amir_ (n=amir@hacker-217-147.congress.ccc.de)
16:48.11a1faextensions 300-309 is my speed dial
16:48.18a1faso would 30. be ok?
16:48.20P4C0ManxPower, http://pastebin.com/481690
16:49.16ManxPower<PROTECTED>
16:49.27ManxPowerit's running voicemail out of exten => s
16:49.41*** join/#asterisk RoyK (n=roy@80.239.107.70)
16:49.43a1fafuck.. still 403 when the call times out
16:49.45a1fahow can i fix this
16:49.53a1fai get 403 Forbiden when the call times out
16:50.02ManxPowera1fa, you need to do some reading
16:50.28a1falol
16:50.29a1fai did
16:50.36PoWeRKiLLHi
16:50.45PoWeRKiLLSomeone use phpagi with asterisk 1.2.1 ?
16:50.47a1fat is for timeout, i know
16:51.13ManxPowera1fa, not if you tried to use exten => ,2,congestion() you didn't
16:51.14PoWeRKiLLI can't get DIALSTATUS variable anymore one * 1.2.1 and it was working on 1.2.0 any idea ?
16:51.22P4C0ok, here's the think
16:51.45a1faManxPower : i didnt know if i should use 30. or t
16:51.58*** join/#asterisk _fan_ (n=allan@static-64-222-189-188.man.east.verizon.net)
16:52.08a1fahm
16:52.08ManxPowera1fa, How about a PATTERN or SOMETHING TO MATCH
16:52.20a1fa30.
16:52.34a1faor
16:52.37a1fa30[0-9]
16:53.13zishanovP4C0, I think there is something else needed to be changed
16:53.35benjk[0-9] == X
16:53.42P4C0zishanov, wait a second
16:54.07zishanovfor the sake of clarity, changed ${EXTEN} with the extension numbers and see if if still works
16:54.17a1fabenjk : so if my speed dial is set for 300, i can have
16:54.18ManxPowera1fa, Well, what do you want to match?  300-309 or any number beginning with 30 of any length.
16:54.19a1faexten => 30X,2,congestion() ; No answer, nothing
16:54.29zishanovexten => 722,n,Voicemail(u${EXTEN}@voiceboxes)
16:54.31a1fa300-309
16:54.43*** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at)
16:54.43zishanovi.e. exten => 722,n,Voicemail(u722@voiceboxes)
16:54.44ManxPowera1fa, Read the damn docs.  ALWAYS use an _ to tell asterisk it's a mattern match
16:54.57_Sam--what is the best solution to get 8 fxo ports to asterisk?  tdm2400?
16:54.57benjk30X would match anything from 300 to 309 yes
16:55.22P4C0zishanov, how can each user configure it's voicemail?
16:55.34ManxPowerzishanov, based on his paste, the call isn't hitting any of those extension, it's hitting exten => s and so Voicemail(u${EXTEN}) is evaluating as Voicemail(us)
16:55.35a1fai am paste bin my speeddial plan
16:55.46ManxPower~docs
16:55.47jbotit has been said that docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
16:55.50a1fahttp://pastebin.ca/35065
16:55.56a1faCan someone look at my speed dial?
16:56.07ManxPowera1fa, I really can't spend a day or two teaching you about dialplans
16:56.18a1faManxPower : hm
16:56.32badboyzis it possible for each asterisk extension to record its own voicemail recording easily/
16:56.42ManxPowerIf you don't understand such basic things as creating a valid extension or pattern matches....
16:56.45a1fajust take a peak, please
16:56.51*** part/#asterisk BugKham (n=lamer@203.130.150.139)
16:57.06*** join/#asterisk fiber0pti (n=John@pcp01876618pcs.sandia01.nm.comcast.net)
16:57.16fiber0ptiis there a way to not have a meetme conference ask for anything and just put the user in?
16:57.32*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
16:57.34badboyzfiber0pti: ask for something?
16:57.36a1fais 503 Service Unavailable, a busy signal?
16:57.46ManxPowera1fa, It's been a long time since I've seen such a wrong dialplan
16:57.56fiber0ptibadboyz: like right now it asks for the user to state their name and then press the pound key
16:58.01ManxPowerdid you read the new asterisk book, available online for free??????????????
16:58.06benjkthere are plenty of things wrong with your dialplan, a1fa
16:58.07a1faabout dialplans?
16:58.12ManxPowerDid you even try to read the parts of the Wiki?
16:58.16badboyzfiber0pti: edit your meetme.conf file, whats in there?
16:58.17a1fayes man
16:58.24benjkfor starters, what do you expect 011XXXXXX@sip.broadvoice.com to do for you?
16:58.25zishanovP4C0, any progress
16:58.34zishanovfor the sake of debugging, try this
16:58.34ManxPowera1fa, So you know that all wildcard patters start with an _
16:58.35zishanovexten => 722,1,Dial(SIP/paco,10)
16:58.35zishanovexten => 722,n,NoOp(${EXTEN})
16:58.35zishanovexten => 722,n,Voicemail(u${EXTEN}@voiceboxes)
16:58.35zishanovexten => 722,n,NoOp(${EXTEN})
16:58.35zishanovexten => 722,n,HangUp()
16:58.39fiber0ptijust "conf => 3456,1111" for that conference room
16:58.47benjkis 011XXXXXX@sip.broadvoice.com a valid SIP URI of somebody you want to call?
16:58.55P4C0zishanov, what is NoOp?
16:58.57a1faits blank
16:59.00zishanovthis will show if you have the right digits in ${EXTEN}
16:59.00a1fai put XXX in there
16:59.00ManxPowerP4C0, REMOVE and lines that start with exten => s
16:59.05a1fai didnt want to put my number in there
16:59.22a1fabenjk : its called ---- censored
16:59.23zishanovNoOp is No Operation, and after that in brackets is what you want to display on the screen
16:59.26P4C0ManxPower, yes done, now how can I set an extension where users can call and check modify their voicebox settings
16:59.32a1fabejk: i removed my phone number
16:59.35a1falol
16:59.58benjkok
17:00.03zishanovFirst make the voicemail work, about that you'll worry later
17:00.09ManxPowera1fa, If your phone number is secret then how can anyone call you?
17:00.12benjkbut your extensions are wrong too
17:00.30a1faManxPower this is only for paste bin
17:00.35benjkit should be 30X,1,...
17:00.37a1fai dont want my phone number out there
17:00.41benjk30X,2,...
17:00.48benjk30X,3,...
17:00.50benjketc
17:00.51a1fabenjk : 300 is the speed dial
17:00.53a1falol
17:00.56*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
17:00.57ManxPowerbenjk, Din't you mean _30X
17:00.57a1faman you are not understanding
17:01.03fiber0ptibadyboyz: any ideas?
17:01.09shmaltzjbot sleep
17:01.10jbot[sleep] overrated, and a poor substitute for caffeine.
17:01.14*** part/#asterisk lubomier (n=lubomier@sunteq.sk)
17:01.15wtwould asterisk detect a ringing FXO port if the Ring and Tip are reversed?
17:01.17a1fai dial 300 -> it dials 011SOME-OTHER-NUMBER-AND-NOTHING-TO-DO-WITH-X
17:01.18badboyzfiber0pti: no idea, what version of * you using?
17:01.19benjkindeed you need an underscore too
17:01.27a1fai dial 301 -> it dials 011-SOME-OTHER
17:01.30fiber0ptithe latest stable
17:01.38a1fathe dialplan is fine
17:01.42fiber0pti1.2
17:01.44badboyzfiber0pti: hmm, might be some new option =/
17:01.55badboyzchecked hte wiki?
17:01.56benjkbut anyway you can't start off with 300,1, ... and then continue with _30X,2, ...
17:02.09Qwellbenjk: you can
17:02.23benjksince when?
17:02.28Qwellsince always?
17:02.29a1falol
17:02.33a1faagain
17:02.36a1fayou are not understanding
17:02.44benjknot on my box
17:03.04a1fahttp://pastebin.ca/35066
17:03.08a1fahere. check it again
17:03.14ManxPowerbenjk, you can, but WEIRD things happen
17:03.23a1faManxPower : this is a valid and working dial plan
17:03.25a1fahttp://pastebin.ca/35066
17:03.26ManxPowerI consider it a BUG
17:03.31a1fathere is nothing wrong with it
17:03.31*** join/#asterisk trymwork (n=trym@c213-158-252-242.sdsl.no)
17:03.34*** part/#asterisk trymwork (n=trym@c213-158-252-242.sdsl.no)
17:03.37badboyzalfa: whats your question?
17:03.42*** join/#asterisk trym (n=trym@c213-158-252-242.sdsl.no)
17:03.52ManxPowera1fa, I cannot help you further.
17:03.57a1fabadboyz : i get 503 service unavailable.. i was wondering if that == busy()
17:04.02benjkyou'll need the underscore, too
17:04.19benjkunless you have a bug that lets you omit that one too
17:04.20a1fain 300?
17:04.27badboyzalfa: are you dialing a broadvoice #, or is broadvoice your voip provider?
17:04.29benjkno wherever it is a pattern
17:04.36benjk300 is not a pattern
17:04.39benjk30X is
17:04.42a1faprovider
17:04.43a1fano
17:04.47ManxPowerbenjk, I've told him that like 4 times.
17:04.54a1fabenjk : so this is fine then
17:04.56badboyzalfa: put _300
17:05.06a1fawhy?
17:05.13badboyztells * that its trying to match on the 300
17:05.15benjkno its not fine
17:05.17a1faif i want to dial 300 -> it dials 5555
17:05.19a1faok
17:05.19ManxPowerbadboyz, no.
17:05.22a1falet me put that in there now
17:05.32ManxPowerbadboyz, you only need a leading _ when it's a WILDCARD PATTERN MATCH.
17:05.40ManxPowerWhat is this, newbie day in #asterisk
17:05.48a1faManxPower : LOL. so > 300 is fine?
17:05.54QwellManxPower: day...week...month...
17:05.56ManxPowerI give up.  I'll come back when people do some basic reading first.
17:06.01*** part/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com)
17:06.02a1fahm
17:06.02benjkif you won't listen and make fun of people who know, then I guess you don't bother asking in the first place
17:06.05a1fafag :P
17:06.17a1fabenjk : i am trying to understand
17:06.24twisted[asteria]a1fa, watch yourself
17:06.27a1fa300 is a valid extension. there is no need for matching
17:06.27*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
17:06.27*** mode/#asterisk [+o anthm] by ChanServ
17:06.49a1fa_30X in the other hand, for everything else, is busy()
17:06.53Qwelltwisted[asteria]: !
17:06.55benjkfirst I suggest you apologise for "a1fa: fag "
17:07.01a1falol
17:07.03a1fai am sorry
17:07.05twisted[asteria]Qwell, !
17:07.17*** join/#asterisk slappingt (n=randygre@pcp03933849pcs.sthind01.mo.comcast.net)
17:07.18badboyz300 doesnt have to be a valid extension, 300 is what * is matching as user input
17:07.34a1faanyway
17:07.42a1fabadboyz http://pastebin.ca/35066
17:07.45badboyzyou dial 300 on your phone, it executes --> dial(SIP/555555@sip.broadvoice.com,30)
17:07.45a1faso this is fine then?
17:07.49a1faright
17:07.57badboyzso you are getting a 503?
17:08.01benjkno its not, because the 30X is a pattern
17:08.04a1fawhen it rings out
17:08.19benjkand if you use a pattern, you must precede it with an underscore
17:08.24badboyz503 is normally when you arent registering properly w/ broadvoice
17:08.28badboyzhave you did a sip debug from your CLI ?
17:08.43a1fawell, it rings fine and everything
17:08.52badboyzringing means nothing
17:08.57a1faand when the phone rings-out, it gives me 503
17:09.06twisted[asteria]badboyz, 503 is actually a "Service Unavailable" message, and is not specific to broadvoice
17:09.09a1fabadboyz : it rings my phone that i dialed
17:09.21a1fathe phone is next to me
17:09.34benjkno space between the underscore and the pattern
17:09.35a1faso i hear it ringing, and i can answer it and it works fine
17:09.36badboyztwisted: in his instance, since im assuming he is only using broadvoice, his 503 IS specific to broadvoice
17:09.37Qwelltwisted[asteria]: I'd say broadvoice gives 503 much more than other providers though. ;]
17:09.45zishanovP4C0, I tried your settings on my Asterisk and they work fine, no problems at all. There is then something wrong in your extensions.conf. {EXTEN} is passing the wrong value to voiceboxes
17:09.54twisted[asteria]Qwell, but that's irrelevant.  i'm trying to cut out some confusion
17:09.58Qwell:p
17:09.59a1faif i dont pick it up, it gives me 503. so i was thinking its w/ my dialplan
17:10.09benjk_30X,2,..., _30X,3,... etc
17:10.34seele_does anybody have any experience with gnugk in aasterisk@home?
17:10.35badboyzcan you slide your underscore up next to your X
17:10.36badboyzer
17:10.46a1fai did
17:10.47badboyzexten => _30X
17:10.58badboyznot sure what =_ is ..
17:11.18badboyzi see a > missing
17:11.37a1faok
17:11.45*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
17:11.59badboyzseele_: gnugk ?
17:12.01*** part/#asterisk cjmoya (n=cjmoya@208.195.223.50)
17:12.14seele_badboyz, yes
17:12.18a1faok
17:12.23a1fai still get 503 instead of BUSY
17:12.25badboyzseele_: whats that an abbreviation of?
17:12.31seele_gnu gatekeeper ... h323
17:12.48badboyzalfa: repaste
17:12.52a1fastupid broadvoice
17:13.00badboyzyea broadvoice is stupid.. ditch it
17:13.03seele_badboyz, I need to make a h323 gatekeeper  with asterisk
17:13.10*** mode/#asterisk [+o Cresl1n] by twisted[asteria]
17:13.19badboyzseele_: havent did that =/ sorry mate
17:13.41*** part/#asterisk _fan_ (n=allan@static-64-222-189-188.man.east.verizon.net)
17:13.44a1fahttp://pastebin.ca/35068
17:13.47*** join/#asterisk BugKham (n=lamer@203.130.150.139)
17:14.09seele_badboyz, ok
17:14.43a1fai guess i dont even need this busy crap and congestion
17:15.05badboyzalfa: id suggest first, to try 300,1,dial --- 300,102,Busy()
17:15.09*** join/#asterisk cjmoya (n=cjmoya@208.195.223.50)
17:15.11badboyzdrop out the pattern matching
17:15.14badboyzsee if you get the same results
17:15.33a1fai know what the problem is
17:15.38a1fathe timeout is set to 30
17:15.46a1fabut broadvoice only goes for 25s or so
17:15.58*** join/#asterisk Uther_P (n=uther_p@66.180.120.82)
17:16.03a1faso since there is nothing to ring any more (my cellphone hangsup after X rings)
17:16.07a1fait gives it 503
17:16.17a1faits fine
17:16.28a1faget a webcam
17:16.35twisted[asteria]nah
17:17.29zishanovdoesn't asterisk 1.2 come installed with mpg123?
17:19.07a1fahehe
17:19.13Uther_Pcurious... if dial is called to a voip phone... if that call is dropped do to a lost ip route, is there any way for an agi to detect that being the reason for the lost call in order to handle the remaining party accordingly?
17:19.23a1faits funny, you know, I ommit my number and get flamed for bad syntax :p
17:20.22badboyzalfa: welcome to IRC dude
17:20.24Uther_P...for example to reconnect the call via some other transport
17:20.32benjkyou didn't get flamed
17:20.37BugKhamwhat's equivalent to  Dial(Zap/1/${EXTEN}) on a E1 Card
17:20.38*** join/#asterisk locid (n=locid@206-248-157-129.dsl.teksavvy.com)
17:20.52a1fabenjk : hehe, i got attacked
17:20.57benjkI said I am not sure what you intended it to mean
17:21.01a1faoh :P
17:21.06a1famaxpower or whutever
17:21.11a1fahe was outraged
17:21.16Uther_Pheh
17:21.30a1faanybody selling 1-800 w/ unlimited inbound minutes?
17:21.44badboyzyour dreaming :)
17:21.56Uther_Pyou sure?  demeanor is easily misconstrued via irc :P
17:22.06a1faheh. i have to pay 0.03 a minute for my 1-800
17:22.15*** join/#asterisk Hakan (i=EmeL@213.186.176.142)
17:22.17benjkyou got criticised for not being all too polite
17:22.25a1faas always
17:22.32a1fano speako englesko?
17:22.32trixterjoy its armageddon week on the history channel
17:22.43trixterwhich means the kooks are gonna start preaching the end of the world soon
17:23.10twisted[asteria]soon?
17:23.12twisted[asteria]hahah
17:23.20twisted[asteria]my grandma already started at christmas
17:23.28*** join/#asterisk grimse (n=grimse@p5481DB77.dip.t-dialin.net)
17:23.37trixterall but the hardcore ones die down most of the year
17:23.38benjktrixter: wasnt that five or six years ago?
17:23.41a1facrazy bible belt people
17:24.18Uther_Pif someone really believed the end of the world was comming... why would they bother preaching it?  ...as if it would somehow make something different if they convinced enough people to believe them
17:24.20*** join/#asterisk Luke-Jr (n=luke-jr@user-0c938qu.cable.mindspring.com)
17:24.34trixterpeople do preach it
17:24.39trixterlook at heavensgate
17:24.39a1faUther_P : those people need to be put to prison and shot to death
17:24.43*** join/#asterisk chrisvarns (n=chris@ACD58A64.ipt.aol.com)
17:24.44a1faand beaten with wooden battons
17:24.47zishanovI made MoH work at last. Now I have almost every feature of Asterisk working. Now I'll be working on queue
17:24.50trixterbarbituate laced applesauce with vodka
17:24.55twisted[asteria]a1fa, how open minded of you
17:24.59a1fayeah
17:25.02a1fai am a communist ;P
17:25.05zishanovcan anybody help on setting up live streaming from some radio etc. for MoH
17:25.13trixterlook at some church that got a TON of donations in 1998 saying the end of the world would be october 1998 and then they just vanished
17:25.24benjkzishanov: app_ices
17:25.30a1fahehhee
17:25.30*** join/#asterisk chrisvarns (n=chris@ACD58A64.ipt.aol.com)
17:25.31trixterfor well over 1000 years there have been those that preach end of the world..
17:25.38a1fathe end of the world is here.. give me all yor money
17:25.51a1fasure, the end of the world, is only 20 milion years away
17:25.54a1fabut who cares
17:26.20Uther_Pa1fa: end of the world... or end of human kind?
17:26.20trixterwell statistically speaking its likely that we are in the first or last 2.5% of human existance than the 95% other
17:26.37trixtera professor at princeton like 15 years ago came up with that one - and its provable under his limited conditions
17:26.38a1fasame difference ;P
17:26.39Uther_Plast
17:26.42Uther_Pdefinatly last
17:27.03a1fai think people are sutpid
17:27.12Uther_Pwe think you're sutpid
17:27.15Uther_Phaha
17:27.17a1fasure
17:27.26benjkits true as in "today is the first day of the rest of your life"
17:27.26a1fawhat comes around, goes around, i guess
17:28.26Uther_PI'm all for human-bashing, but if you're gonna bash on irc... you should always make sure you are doing it gramatically correct. :P
17:29.59Uther_Pthe only provable truths would be ones which have no reliance on the 'known' universe
17:30.36Uther_Pfor its all just neuro-sciences
17:30.55a1fai want to transmit silence
17:31.03a1fawhat option should i use
17:31.11a1fai hate when there is no noise on the phone
17:31.12a1faits scarry
17:31.15lesouvageI have a script to modify and copy a callfile to /var/spool/asterisk/outgoing . The last inch doesn''t work (passing the phonenumber into the callfile). Can some of you please look at  http://pastebin.ca/35069 .
17:31.23*** join/#asterisk leenuxg33k (n=bpeck@71-10-248-241.dhcp.oxfr.ma.charter.com)
17:31.24a1fai like to hear that humming noise
17:31.31a1fawhat is that called?
17:31.40meredyddComfort noise, a1fa.
17:31.47a1fayeah, Comfort noise
17:31.51leenuxg33kquestion on the seeting FROMUSER
17:31.53a1faCan * do that?
17:31.53leenuxg33kFrom: "asterisk" <sip:asterisk@216.143.130.36>;tag=as0baa4a22
17:31.56meredyddThere's even an RTP extension for it, though I don't know how to tweak it on *.
17:32.05meredyddgoogle it
17:32.17leenuxg33kIf I have FROMUSER set to something other than asterisk..  shouldn't it show that username here instead of asterisk
17:32.20leenuxg33k?
17:32.39Uther_Pa1fa: its called the buzzing in your head from basking under the florecent lights :P
17:32.43twisted[asteria]fromuser= in sip.conf in the peer entry you're sending the call to
17:33.05*** part/#asterisk BugKham (n=lamer@203.130.150.139)
17:33.14leenuxg33ktwisted[asteria]: I am..  seems like its ignoring it
17:33.23twisted[asteria]leenuxg33k, paste relevant config
17:33.28twisted[asteria]http://www.pastebin.ca
17:33.39a1faUther_P : that too
17:33.47a1faanybody workerd with CNG?
17:34.07twisted[asteria]just blow into the phone
17:34.17Uther_Phehh
17:34.17a1fano dude
17:34.24a1fai'll blow you in the ass
17:34.29twisted[asteria]hah.
17:34.32Uther_Pyou sick bastard
17:34.58Uther_Pcarefull though, that the ass doth blow in your face instead
17:35.04twisted[asteria]that was another comment that was uncalled for
17:35.11*** join/#asterisk diclophis (n=diclophi@adsl-69-238-124-226.dsl.pltn13.pacbell.net)
17:35.16diclophishello all
17:35.18leenuxg33k[galaxyvoice]
17:35.19leenuxg33kport=5060
17:35.19leenuxg33kusername=RXXXXX
17:35.19leenuxg33kauthname=RXXXXX type=friend secret=XXXXX
17:35.19leenuxg33kreinvite=no
17:35.19leenuxg33kqualify=1000
17:35.20leenuxg33knat=never insecure=very
17:35.22leenuxg33khost=216.143.130.36 fromuser=RXXXXX
17:35.24twisted[asteria]leenuxg33k, that's why i said pastebin.
17:35.24leenuxg33k;fromdomain=216.143.130.36
17:35.26leenuxg33kfromdomain=sip.gis.net
17:35.28*** mode/#asterisk [+b %leenuxg33k!*@*] by twisted[asteria]
17:35.30diclophisgah... pastebin...?
17:35.32Uther_Pleenuxg33k: you must have missed the post about http://pastebin.ca
17:35.49diclophisso..., I have my PRIs setup and they are working fine with my dialplan
17:35.51*** mode/#asterisk [-b %leenuxg33k!*@*] by twisted[asteria]
17:35.52twisted[asteria]okay
17:35.54Uther_Pheh,  where the hell did it go...
17:35.59diclophisthe only problem I have is that the last 4 digits are sent only
17:36.08leenuxg33khttp://pastebin.ca/35070
17:36.12leenuxg33kdoh!
17:36.12trixter~pb
17:36.14jbotrumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
17:36.14diclophis... how can I confirm that this is a problem on my side, compared to a problem with my provider?
17:36.14twisted[asteria]leenuxg33k, yay :)
17:36.14*** join/#asterisk svenna_ (n=svenna@p548D3ED1.dip0.t-ipconnect.de)
17:36.25twisted[asteria]okay well
17:36.32leenuxg33ksome linefeeds got screwed up
17:36.36twisted[asteria]you need to separate each statement into it's own line.
17:36.37leenuxg33kbut otherwise I think its right
17:37.05a1fanice
17:37.12a1faX-PRO has this Transmit SIlence options
17:37.20a1fai wonder if my PAP2-NA has that
17:37.29leenuxg33ktwisted[asteria]: they are seperated..  weird cut and paste error
17:37.48twisted[asteria]strange, i've never seen cut/paste do that.... ever.
17:37.59badboyzleenuxg33k: so whats your question?
17:38.23twisted[asteria]leenuxg33k, just make absolutely sure that there are newlines at the end of each line in your config
17:38.24leenuxg33kbadboyz: my tcpdump shows the FROM: being from asterisk.. not the fromuser I set in the peer
17:38.32twisted[asteria]that would be a reason it's being ignored
17:38.38badboyzi agree, absolutely
17:39.08badboyzthat paste is so irregular, w/ auth / type / secrets on same line, as well as host / fromuser that its very fish
17:39.10badboyz+y
17:39.10*** join/#asterisk _fan_ (n=allan@static-64-222-189-188.man.east.verizon.net)
17:39.45twisted[asteria]leenuxg33k, if nothing else, just add linefeeds at the end of each variable/value pair, then go back into the cli and issue a "reload chan_sip.so" without the quotes.
17:39.56twisted[asteria]i've gotta run off for a sec
17:42.24Uther_Pleenuxg33k:  is galaxyvoice the context the call is being made TO or FROM?
17:43.48*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
17:44.09a1faok
17:44.14a1faRally's time
17:44.22leenuxg33kUther_P: not sure what you mean.  [galaxyvoice] is my voip provider.. I seem to register ok..   but when I do sip show peers, it shows UNREACHABLE.
17:44.40a1fag33k firewall?
17:44.47leenuxg33kI redid the end of line returns..  and restarted asterisk
17:44.53leenuxg33kI got OK (42 ms)
17:45.01badboyzexcellent
17:45.03Uther_Peh.. you are debugging a problem with the username the packets show.... and you can't even get to your provider?
17:45.06leenuxg33kwas able to make one call and now it says UNREACHABLE again
17:45.14leenuxg33kI'm running siproxd on my linksys
17:45.21leenuxg33kso no nat settings
17:45.26badboyzrecheck your sip.conf file, see if that line return issue came back
17:45.29a1faanybody like Rally's?
17:45.32*** join/#asterisk lurking1 (i=tall-guy@207-195-103-110.regn.hssx.sasknet.sk.ca)
17:46.16leenuxg33kUther_P: its sparadic
17:46.45leenuxg33kI don't know if galaxyvoice is really having an issue or now.. I've emailed there tech support
17:46.49asterboyAnyone suggest a good search engine besides google and yahoo...was using yahoo but now they have some stupid flash content that slows everything down...even typing in the search window.
17:46.57Uther_Psounds like your linksys isn't doing what it's supposed to
17:47.12Uther_Pasterboy: whats wrong with google?
17:47.26asterboyjust don't want to use them anymore
17:47.31asterboyneed to gte off their tit
17:47.42implicitor touch it
17:47.57lurking1not using google is like boycotting oxygen isn't it?
17:48.04implicit?
17:48.05asterboyand AOL
17:48.07Uther_Pasterboy: haha.. ok... well... there is  lycos and altavista, off hand
17:48.11implicitdont be dumb as fuck lurking1
17:48.16asterboyand corporate greed
17:48.16leenuxg33kUther_P: I tried just using ip forwarding without siproxd and get the same thing
17:48.29asterboyaltavista!
17:48.33asterboyforgot about them.
17:48.40implicitaltavista also is greedy
17:48.43Uther_Pasterboy: but they suck
17:48.46implicitaltavista is in it for the money
17:48.47RoyK~nickometer leenuxg33k
17:48.47lurking1implicit: 'scuse me?
17:48.49asterboydam
17:48.56implicitand they never find anything
17:49.01implicit~nickometer lurking1
17:49.09asterboywish there was a good search engine like google was in the good ol days.
17:49.18*** part/#asterisk diclophis (n=diclophi@adsl-69-238-124-226.dsl.pltn13.pacbell.net)
17:49.28asterboynow all they want to do is control every aspect of my life.
17:49.35implicitasterboy, how did the engine itself deteriorate?
17:49.47implicitasterboy, it improved if anything
17:49.49asterboybecause they have not improved it.
17:49.49badboyzuse the search engine mamma.com ;)
17:49.50Uther_Pleenuxg33k:  set the 'externip' option to your public address
17:49.57asterboythey are concentrating on every other service.
17:49.58*** join/#asterisk shadebob (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma)
17:50.00asterboyadding new services
17:50.11asterboythe search technology is still so inadequate
17:50.27Uther_Pasterboy: sounds like you simply don't know how to formulate a good google query
17:50.42implicitasterboy, what would you do in their situation? hire 3 girls for each programmer at google to suck one ball & dick each
17:50.44implicit?
17:50.47asterboywell I have used all the advanced options.
17:50.50Tall-guyany search engine that can find a post I did 19 years ago from a Bitnet account is adequate in my opinion
17:50.58implicitasterboy, you dont know any advanced options
17:51.13implicitasterboy, advanced options are to not think the internet has everything on it
17:51.14Uther_Pasterboy:  + for require, - for not,  quotes for literals
17:51.25shadebobhi, I have a problem with musiconhold on an asterisk trunk version... Music come and stop, come and stop... I have no digium card installed in my PC... and I use mode=files
17:51.26asterboyyes yes...done that
17:51.31implicitthe internet fucking sucks anyway
17:51.35Uther_Phaha
17:51.38asterboylol implicit
17:51.44Uther_Pwhy are you using it then?
17:51.47implicitit doesn't have everything on it
17:52.00implicit99% of the content is complete BULL
17:52.04asterboytrue
17:52.08implicitand the other 1% is wrong
17:52.11asterboylook at wikipiedia
17:52.25implicitI know, wikipedia has so much incorrect information it is unbeleivable
17:52.26Tall-guyimplicit: are you the 1% or 99%? :)
17:52.32Uther_Pimplicit: which of those do you speak from?  the 99% bull or the 1% wrong?
17:52.35Uther_Phaha
17:52.36Uther_P:P
17:52.39Tall-guy:)
17:52.45implicitboth, i have experienced it all
17:52.46implicitand it all sucks
17:52.53Uther_Phehe, right over your head
17:52.55impliciti speak from experience
17:52.58Uther_Pthats alright though
17:53.00asterboygreat resource, but when you have the founder editing the history to show the incorrect version of events...makes you wonder what else is bogus.
17:53.17implicitasterboy, i know
17:53.28implicitasterboy, and also, look at www.com
17:53.44asterboyyes google and wiki are the lesser of all the evils...but why can't google focus on making search technology better?
17:53.46Tall-guyI miss archie and gopher :)
17:53.49implicitthe web starts there
17:53.53asterboyarchie!
17:53.55Uther_Phow silly to complain about a free service... don't like it?  don't use it... to bitch about it is pretty damn pointless
17:53.56implicitand it is all advertisements and shit
17:53.56asterboygopher!
17:54.07asterboygoogle is NOT free
17:54.09Uther_Pasterboy: google's search is excelent
17:54.13implicitUther_P, no it's not
17:54.13Uther_Pasterboy: it is to you
17:54.26implicitUther_P, free services that decieve and misguide you are better not to exist
17:54.31implicitUther_P, they WASTE your motherfucking time
17:54.34asterboynot when I surf web pages that are infected with countless google ads
17:54.39implicitUther_P, just like VoIP, it is useless as fuck
17:54.42Uther_Phaha, google didn't do that... people like you that don't know what the hell you're looking for did that
17:54.58implicitUther_P, but wikipedia did do that
17:55.01implicitand it is also a free service
17:55.05Uther_Pgoogle's scoring is based largly on who clicks on what when they search for something
17:55.10implicitthink before you say stupid shit
17:55.25implicitUther_P, ???, they don't know who ckicks on what
17:55.40Uther_Phaha, dude, you're fuckin stupid
17:55.47implicitUther_P, you are fucking stupid
17:55.53asterboynow now
17:55.56Uther_P:)
17:56.01Tall-guy*sigh*....and here I was hoping to update my asterisk knowledge today
17:56.02implicitlinks are like this
17:56.03implicithttp://www.hello.com/
17:56.06asterboystupid is what stupid does
17:56.06implicitoff google website
17:56.17implicitthey dont link to another one of the google sites to forward
17:56.20Uther_PTall-guy: ask forth your question, just ignore the dribble
17:56.32Tall-guyuther: i learn more by listening, I'm questionless today
17:56.42implicitoh the other hand
17:56.45implicitaltavist does know where you click
17:56.48implicithttp://av.rds.yahoo.com/_ylt=A9ibyKvL0bJDInMA4oVrCqMX;_ylu=X3oDMTBvdmM3bGlxBHBndANhdl93ZWJfcmVzdWx0BHNlYwNzcg--/SIG=11olgb2of/EXP=1135878987/**http%3a//katyharclerodes.blogspot.com/
17:56.55implicitcause they have links like this that forward you
17:57.07*** join/#asterisk fjean (n=fjean@201009209056.user.veloxzone.com.br)
17:57.13*** join/#asterisk leenuxg33k (n=bpeck@71-10-248-241.dhcp.oxfr.ma.charter.com)
17:57.23implicitUther_P, go suck a nut, and then understand the extent of your fuckin stupidity
17:57.27benjkhow about cookies
17:57.30Uther_Phaha
17:57.33implicitbenjk, how about no
17:57.38*** join/#asterisk davidw (n=davidw@apache/committer/davidw)
17:58.17davidwhey.... I am fooling around with txfax - it says that it returns -1 on failure.  How do I make that jive with the priorities (i.e. make it do something on failure)
17:58.20asterboyhttp://www.theregister.co.uk/2005/01/11/open_source_google_scraper/
17:59.19seele_SOMEONE WHO HELPS ME CONFIGURING A POLYCOM SOUNDPOINT 501 PLEASE!
17:59.21trixterisnt the google api easier to use than a scraper?
17:59.37badboyzseele_: whats wrong?
17:59.48dudesdavidw - it can be done
17:59.53trixterI think his caps lock button is broke, at least that is wrong :P
18:00.17seele_badboyz, in the phone's display i get: "Url call disabled"
18:00.20leenuxg33kLAGGED (2879 ms)
18:00.32leenuxg33kwould that be the VOIP provider?  and not me?
18:00.43seele_badboyz, so, i can recieve calls to that polycom, but it cant make them
18:00.46bkw_drumkilla,
18:01.04drumkillabkw_:
18:01.10seele_badboyz, Any clues?
18:01.30*** part/#asterisk _fan_ (n=allan@static-64-222-189-188.man.east.verizon.net)
18:01.45*** join/#asterisk J4k3 (i=j4k3@dhcp-12-197-128-55.intrastar.net)
18:02.28badboyzseele_: check /msg
18:03.42badboyzwe've been using telasip with great success
18:03.58leenuxg33kbadboyz: do you have a url for them?
18:04.07badboyznot sure if im allowed to post that
18:04.10davidwdudes, well... my question is really more basic.  "what's the -1 mean in terms of the dial plan?"
18:04.12badboyza quick google will dig it up
18:04.20leenuxg33kbadboyz: the fact that I can register to fwd.net fine via sip makes me think its galaxyvoice with the problem
18:04.28leenuxg33kI get errors 483
18:04.31leenuxg33ktoo many hops
18:04.58leenuxg33kand 408 Request Timeout
18:05.03leenuxg33kbadboyz: thanks
18:05.04benjkhttp://www.google-watch.org/cgi-bin/cookie.htm
18:05.07badboyznp
18:05.32dudesdavidw - been awhile since I dug into the code
18:05.41leenuxg33kbadboyz: and you have a working asterisk config for them?
18:05.44dudesI'll pull it up in a sec
18:05.47leenuxg33kbadboyz: do they support asterisk?
18:05.51badboyzleenuxg33k: definately
18:05.53badboyzusing them right now
18:06.29leenuxg33kbadboyz: would you mind messaging me your peer config minus your login info?
18:06.32davidwdudes, you think it would involve C hacking?
18:06.34badboyzsure, sec
18:06.35seele_hi there, i need to make outside calls (PSTN), but i cant get the nuber 033XXXXXX out... why??
18:07.15seele_someone tell me the correct dial rules for making a call to this number 03315608XXXX
18:07.29twisted[asteria]wheeeeee
18:07.53dudesdavidw - not really.  When we added stuff it was only a couple lines
18:10.18shido6because you screwed something up.. what do you have in the dialplan to dial that , seele?
18:12.09dudesdavidw - it was more than 1 line.  like 10 in txfax to update the return stuff for *
18:12.53fileseele_: we can't because we don't know how your system works, how you are sending calls out, what context, etc
18:13.32dudesdavidw - and -1 means failure otherwise it throws 0 (if I"m ready it right, but i'm doing 10 things at once) ... which it's pretty easy to improve upon -1 or 0
18:14.17*** join/#asterisk Connor_ (n=billyhud@198-144-174-5.knx.tn.nxs.net)
18:16.32*** join/#asterisk mistral (i=mistral@jstevenson.plus.com)
18:18.30davidwdamn...my keyboard freaked out
18:18.35fjeanhello all, anybody has successfully connected a axg800 tenor (or any quintum) to asterisk using SIP ?
18:18.49davidwweird
18:19.26davidwdudes, ok, so the return value... how does that interact with the dial plan... how is it visible to asterisk?
18:22.15drumkilladavidw: you can't do anything with it.
18:22.24drumkillaas a matter of fact, we removed that from all of the application descriptions
18:22.44drumkillathe only significance it is, is that if it returns negative, the call is ended
18:22.48davidwah
18:23.07drumkillazero means the call continues at the next priority
18:23.10davidwso... mmmm this is all with the idea of providing some sort of report to people attempting to send faxes
18:23.14Cresl1nmmm....
18:23.28davidwif the fax doesn't go through for some reason
18:23.56drumkillathe way other applications are doing it, is they are setting a channel variable that you can check
18:24.05drumkillafor example, the Dial application sets DIALSTATUS
18:24.10drumkillaI don't know about the fax apps
18:24.50davidwdon't seem to do much.  Hrm.  is there a guide to the asterisk C API (in other words, how do I set the DIALSTATUS?:-)
18:25.32drumkillathe most comprehensive guide is here: http://www.asterisk.org/doxygen/
18:25.43drumkillaalso linked from the developers page on asterisk.org
18:26.15*** part/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.hssx.sasknet.sk.ca)
18:26.44drumkillathe doxygen documentation on asterisk.org is automatically updated nightly
18:28.09*** join/#asterisk L|NUX (i=linux@203.101.168.28)
18:28.29*** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk)
18:28.59davidwaha: pbx_builtin_setvar_helper
18:29.23drumkillayup
18:30.06a1fawhats a good codec for dialup
18:30.13a1fathat i can force for my bro?
18:30.29*** join/#asterisk uther (n=uther_p@66.180.120.82)
18:30.39a1faG729 good for dialup
18:30.40a1fa?
18:30.43rayvdSmall bears!
18:30.58jmolenskianyone know what specs i should use when saving my wav filess for my ivr?  freq, bits, etc?
18:31.19drumkillajmolenski: 8000 kHz mono
18:31.30rayvdwhat's g729?  ~20Kbps?
18:31.36jmolenski16bit, 8bit?
18:31.42*** join/#asterisk lesouvage (n=lesouvag@82.74.11.143)
18:31.45a1fa8bit
18:31.50jmolenskicool, thanks
18:31.53a1fai need something very small
18:32.37a1fawhat can i use for dialup
18:33.26rayvdoh, g729 is 8Kbps
18:33.28rayvdi would think that would be your best bet
18:33.33a1faok
18:35.37znoGhow does one interpret the info on "show codecs" ?
18:36.01znoGi'm trying to find the right codec to use between 2 asterisk servers to consume the least amount of bandwidth while maintaining an acceptable quality of sound
18:36.12RoyKznoG: read about the codecs
18:36.17RoyKznoG: on voip-infop
18:36.27znoGgood point.
18:36.29znoG:)
18:36.29RoyKshow codecs only shows bs from a user's point of view
18:36.45RoyKspeex is low bandwidth, quite good quality and eats lots of cpu
18:36.54a1fawhen i force g729
18:36.57a1fai get this error msg
18:36.57a1faDec 28 18:36:35 WARNING[26397]: app_dial.c:1553 dial_exec_full: Had to drop call because I couldn't make SIP/235113-ca3e compatible with SIP/235114-e987
18:36.59znoGyea, CPU is another factor. Not a whole lot of bw available
18:37.10znoGerr CPU
18:37.16a1fawe are both using XPRO
18:37.41a1fawtf
18:38.14iCEBrkrblah blah blah
18:38.23*** join/#asterisk Zach^^ (i=chaos@dialup-4.225.2.170.Dial1.Cincinnati1.Level3.net)
18:38.27davidwdudes, drumkilla - thanks for the help...
18:38.35davidwwe'll see what the txfax guy has to say
18:38.41davidwI think I can hack this if I need to...
18:38.55rayvda1fa, do both your asterisk and the ata support g729?
18:38.59Zach^^i have the fax setup... to anwser on incomming calls via "system" and have the mail setup... howcome it is not accepting faxes?
18:40.02*** join/#asterisk TheCops (i=nobody@got.securebinary.com)
18:40.11a1fai guess you cant conference yourself in
18:40.41a1fawith ulaw
18:40.49a1fai tried to ulaw -> g279 dial g279
18:41.02shido6got a license?
18:41.17badboyzanyone have an idea where you can override the default naming scheme for the MeetMe recordings?
18:41.21a1fafor?
18:41.50RoyKa1fa: have you bought the g.729 codec from digium?
18:42.01a1fait come swith asterisk?
18:42.03a1fadoes it?
18:42.10a1fais speex free?
18:42.18RoyKspeex is free
18:42.21a1faok
18:42.28TheCopsa1fa more CPU load
18:42.31TheCopswith speex
18:42.50iCEBrkrbadboyz: Do your own recording routines?
18:42.53RoyKbut less cost per codec......
18:42.53znoGi think i'll go with gsm
18:43.07iCEBrkrHrrm, I guess that won't work cuz Meetme doesn't have any 'tear-down'
18:43.17TheCopsyup
18:43.37TheCopsznoG, g729 is a really good for bandwidth issue
18:43.47znoGTheCops: yep, its also commercial ;)
18:43.59badboyziCEBrkr: define recording routines?
18:44.11iCEBrkrbadboyz: you're able to call Monitor() or Record() manually.
18:44.56badboyzif i issue the monitor command before the meetme(), will it record everything that happens in that context?
18:45.07a1faif i dial with ulaw
18:45.13a1fai cant conf my self in with speex
18:45.14badboyzbecause id assume everything that joins the meetme, it would reissue the monitor command
18:45.19iCEBrkrbadboyz: Why wouldn't it?
18:45.34iCEBrkrbadboyz: yea, that would happen too.
18:45.38badboyzyea, thats bad ;)
18:45.43TheCopsznoG yeah :) but 10$ per channel, not very expensive
18:45.51qzxcdcya
18:46.13a1fathis sucks
18:46.29iCEBrkrbadboyz: Well you could hack in a DBPut()/DBGet() to toggle the status flag of the recording.
18:46.35a1fai establish an ulaw via broadvoice, conference myself in, and it wont work, bcos they have speex
18:46.41a1fai thought asterix eliminates that problem
18:46.46a1faand lets codecs work together
18:46.50iCEBrkrbcos? LOL
18:47.05TheCopsa1fa, does your SIP client (or whatever you are using) support speex ?
18:47.10a1fayup
18:47.13a1faX-PRO
18:47.16badboyzheh, alfa must come from playing counterstrike or something ;)
18:47.28iCEBrkrbadboyz: yea, really
18:47.47*** join/#asterisk Druken (n=blowme@static.abss.ca)
18:47.50iCEBrkrbadboyz: you should see the crap on World of Warcraft, I swear it's loaded with 12yr olds
18:48.15*** join/#asterisk Gh0sty (i=ghosty@kiekeboe.x-plose.be)
18:48.24badboyziCEBrkr: yea ive been playing for over a year now sadly :(
18:48.32badboyzbarrens chat FTW
18:48.33Gh0styhello all
18:48.37badboyzok -- thats all im saying heh
18:48.57iCEBrkrbadboyz: Oh? Barrens chat is like that across ALL servers?!
18:48.57Gh0styi've a small question: when i try to conference and i dial 8200
18:48.59iCEBrkrgeeesh
18:49.06badboyziCEBrkr: yes, definately, lol
18:49.10Gh0styi get: this is not a valid conference number
18:49.20iCEBrkrI've only been playing for about 2weeks. I finally gave in. :(
18:49.21Gh0styanyone ideas what could cause this?
18:49.28badboyziCEBrkr: hehe, right on, server?
18:49.34DrukenGh0sty: did you make the confrence?
18:49.40iCEBrkrbadboyz: Hyjal mainly
18:49.45Drukenor "meetme" :)
18:49.48Gh0styDruken: what do you mean? :s
18:50.09Gh0styi have another box where it works perfectly out of the box (its asterisk@home)
18:50.21Gh0stybut this one always gives me the error :/
18:50.38Drukenk... i know nothing about a@h...
18:50.41iCEBrkrbadboyz: 14 Pally, 9 Priest, 6 Hunter, 8 Druid.  across 3 servers. :)
18:51.07iCEBrkrGh0sty: Well that's your problem right there.. Asterisk@Home :P~~~~
18:51.10badboyziCEBrkr: hyjal is pve, do yourself a favor, and get on a pvp server
18:51.20iCEBrkrbadboyz: I'm on one.
18:51.23Gh0styiCEBrkr: sure :p
18:51.35Drukenuhg...
18:51.38Gh0styDruken: i've set in meetme.conf the next
18:51.41badboyziCEBrkr: stick to it, you will kick yourself later if you build on a pve
18:51.58badboyzDruken: whats your a@h questions, ive used it
18:52.01iCEBrkrbadboyz: I got friends on 3 different servers. So I play wherever.
18:52.08Gh0styDruken: conf => 8200
18:52.09a1faso, guys
18:52.13Gh0stylike most manuals say
18:52.20a1fado both clients need to be on same codec to call eachother?
18:52.23Gh0stybut still says invalid conference number
18:52.28Drukenbadboyz: i dun have any questions... i wouldn't use a@h if you paid me... ok well... MAYBE if you paid me....
18:52.37badboyzoh -- who had the a@h question then?
18:52.45DrukenGh0sty did
18:52.52*** join/#asterisk Defraz (i=t0tal@72.24.26.215)
18:52.57iCEBrkrGh0sty: I'm thinking you have to reload to get the meatme.conf changes to take
18:53.04Drukena1fa: no
18:53.07Gh0styhas been reloaded
18:53.16iCEBrkrGh0sty: and you still get invalid conf number?
18:53.20Drukenrestart not reload
18:53.20*** part/#asterisk pr0m (n=pr0m@24-75-196-70.chvlva.adelphia.net)
18:53.23Gh0styyes
18:53.43badboyzGh0sty: in a@h it creates a meetme_additional.conf file, that has the conf =>8(extension) created automatically
18:53.54a1faDruken : so i was using X-PRO, ulaw to my home phone, trying to conference my bro via sip speex, and it didnt work
18:54.22Drukenwell, "it didn't work" is a bit vaige
18:54.28a1fa403 Forbidden
18:54.30Gh0stybadboyz: yes i can see that so its a redundant entry even :)
18:54.40Drukenthen it's an access issue
18:55.04iCEBrkr"Access issue? I don't even have MS-Access installed!!"
18:55.10badboyzGh0sty: ehh, it shouldnt be redundant, it just defines a conference room for each extension
18:55.17Druken:)
18:55.19iCEBrkrhehe
18:55.31twisted[asteria]lol
18:55.33Drukendon't make me get the ice pick
18:55.34twisted[asteria]he said meatme.conf
18:55.41iCEBrkrhey hey hey, woah now..
18:55.52iCEBrkreasy their fella.
18:55.57Gh0stybadboyz: ok i deleted the entry so there is only one in meetme_additional.conf
18:56.03badboyziCEBrkr: paly , priest, hunter, drood -- only good choice was the priest ;)
18:56.08Gh0stybut still gives not a valid conference number
18:56.22twisted[asteria]do you have a timing device?
18:56.26Drukendoesn't a@h have their own support channel ??
18:56.36badboyzGh0sty: so you got conf => 8200 right in the meetme_additional.conf file right?
18:56.48twisted[asteria]meetme requires a zaptel timer
18:56.56badboyzDruken: i spose its all one in the same, no?
18:57.01Gh0stybadboyz: yes
18:57.04twisted[asteria]badboyz, no
18:57.09iCEBrkrbadboyz: Depends on who I'm playing with, one group of my friends needed a Priest.  The other needed a fighter type.  So I picked Paladin.  The hunter is my own doing going old skewl EQ and recreating a character
18:57.09badboyzGh0sty: where are you dialing from?
18:57.11Drukentwisted[asteria]: shouldn't need that to see them tho... just for it to work
18:57.18Gh0stybadboyz: local network
18:57.23twisted[asteria]Druken, yes, you do
18:57.40P4C0how can I make a extension for my users to check their voicebox?
18:57.49iCEBrkrDruken: I'm with you.. A@H shouldn't be 'supported' here :D
18:57.49Drukenwell, there ya go.. ya learn something new everyday :)
18:57.54badboyziCEBrkr: paladin is a tankish role, fighter type = rogue/mage (dps)
18:58.16iCEBrkrbadboyz: I'm attempting to play types I haven't played in the past.
18:58.16twisted[asteria]Gh0sty, do you have a zaptel timer?
18:58.21*** join/#asterisk _fan_ (n=allan@static-64-222-189-188.man.east.verizon.net)
18:58.24iCEBrkrDruken: Sweeet! I get a badge!
18:59.06Drukeni have a beaver police badge...
18:59.08Gh0stytwisted[asteria]: i read something about that trough google some module called ztdummy
18:59.23Gh0stytwisted[asteria]: but i don't see this module loaded on my other box either
18:59.23twisted[asteria]Gh0sty, right, and do you have that installed?
18:59.24a1fawill 128 kbps be enough of upload for ulaw?
18:59.29iCEBrkro/~ badger badger badgerbadgerbadgerbadger badger o/~
18:59.33a1fai know it takes 86 kbps..
18:59.35iCEBrkrSNAAAAAAAAAAKE!
18:59.36badboyziCEBrkr: my viewpoint comes from how discouraging you want the game to become for you -- if you want acceptance, warrior/priest/mage are the 3 that are always necessary -- the other classes get the shaft alot
18:59.37twisted[asteria]for meetme to get past the "invalide conference" you need a timer
18:59.42twisted[asteria]-e on invalid
18:59.43Gh0stytwisted[asteria]: installed? :s
18:59.45Gh0styhow?
18:59.53Dandanmodprobe ztdummy
18:59.54twisted[asteria]insmod?
19:00.05Dandanbut you have a proper usb module inserted
19:00.12iCEBrkrbadboyz: I don't believe the hype.  I'm not like most gamers when it comes to classes.. I don't power game and I'm not in it to get the best of the best gear
19:00.13[TK]D-Fender[13:59] <iCEBrkr> o/~ badger badger badgerbadgerbadgerbadger badger o/~ <- YOU ARE AN EVIL, evil, PERSON....
19:00.19iCEBrkrfelipex: :D
19:00.26iCEBrkrerr
19:00.29Dandan:>
19:00.31iCEBrkr[TK]D-Fender:
19:00.37Dandannickname completion?
19:00.37twisted[asteria]oh noes!11!!11!!11!oneone! not the badger!
19:00.47badboyziCEBrkr: i know your kind all too well -- good luck holding true to that in WoW ;)
19:00.50iCEBrkrfenlander: Change your nick to something other than 'fen'
19:01.01Gh0stytwisted[asteria]: when i look at my working box: lsmod |grep ztdummy gives me nothing
19:01.15twisted[asteria]Gh0sty, k, then there's your problem
19:01.17[TK]D-FenderI still haven't let go of my old clan tag...
19:01.22Gh0stywhen i look at the troubled box: insmod ztdummy failed
19:01.27P4C0is that possible?
19:01.28iCEBrkrbadboyz: I played EQ for 4yrs, DAoC for a year, SWG for 2yrs, CoH for 2yrs.. Nothing impresses me anymore.
19:01.58badboyziCEBrkr: not even 5 mill current subscribers to WoW ?
19:02.11iCEBrkrbadboyz: No, cuz it's all 12yr olds who don't know any better.
19:02.15badboyznaaa
19:02.25iCEBrkrbadboyz: just like Halo was game of the year or whatever, That came BLOES
19:02.25_fan_I'm trying to move a configuration from a working Asterisk CVS-v1-0-11/13 to a 1.2.1 - I've made all the configuration changes to eliminate all errors and warnings.  Internal dial plan and incoming calls all work fine.  On outgoing calls I am receiving 503 - Service Unavailable.... any ideas?
19:02.27iCEBrkrBLOWS TOO
19:02.27badboyzyou will find some good apples in there
19:02.50Gh0stytwisted[asteria]: no it works on box #1 which has no ztdummy module loaded and box #2 no either (and there it doesn't work), also in order to get ztdummy working you need usb_uhci loaded and both boxes have usb_ohci loaded
19:02.56badboyziCEBrkr: only advice i can give you, is get the mod that allows your ignore list to hold unlimited names -- then the game is perfect :)
19:02.57iCEBrkr_fan_: PSTN? VoIP?
19:03.02_fan_VOIP
19:03.05P4C0Incorrect password '782' for user '782' (context = default) <--- why it sais context default if I'm telling him to use my context
19:03.15iCEBrkr_fan_: Something didn't register??
19:03.21iCEBrkrbadboyz: haha
19:03.27iCEBrkrbadboyz: Titan bar is evil.
19:03.43iCEBrkrTime to next level: 15mins
19:03.45_fan_copy of register statement from working installation
19:03.54twisted[asteria]P4C0, you have to specify the context if it's not default
19:04.05P4C0twisted, I did
19:04.25P4C0twisted[asteria], exten => 1717,1,VoiceMailMain(u782@voiceboxes) the context is voiceboxes
19:04.26iCEBrkr_fan_: sip show registry and see if they're ok
19:04.52_fan_state = registered
19:05.15iCEBrkr_fan_: anything on the console/CLI?
19:05.24*** part/#asterisk jmolenski (n=jjones@216.147.224.254)
19:05.37znoGis there a wav or mp3 to GSM converter?
19:05.45iCEBrkrznoG: sox
19:05.46_fan_no warnings or errors
19:05.53_fan_just debug info
19:05.54*** join/#asterisk alephcom (n=alephcom@openbsd.hagenhomes.net)
19:06.01iCEBrkr_fan_: do you even see the Dial() statement?
19:07.12_fan_the dial also responds with -- Got SIP response 400 "Bad Request" back
19:07.48P4C0_fan_, are you having problems with sip registry? I do, I can registry but after a while I got wrong password
19:08.55_fan_wouldn't a sip show registry show unregistered
19:09.05P4C0its ${CIDNum} or ${CALLERIDNUM} !??
19:09.19RoyK~lart himself
19:09.39RoyKP4C0: the latter
19:09.47RoyK~nickometer P4C0
19:09.53[TK]D-FenderP4C0 : what version of * are you using?
19:10.11P4C0[TK]D-Fender, 1.2.1
19:10.18RoyK[TK]D-Fender: it's CALLERIDNUM anyway
19:10.26[TK]D-FenderP4C0 : then it should be ${CALLERID(num)}
19:10.27RoyK[TK]D-Fender: but 1.2 supports macros as well
19:10.40RoyK[TK]D-Fender: no, ${CALLERID(number)}
19:10.42P4C0[TK]D-Fender, ok, cause calleridnum always returns 1
19:10.47iCEBrkrhaha
19:10.50_fan_also - if it wasn't registering correctly.. I shouldn't be able to receive incoming calls
19:10.57RoyKP4C0: with caps or lowercase?
19:11.05P4C0RoyK, caps
19:11.06[TK]D-FenderRoyK : Not unless the WIKI page on functions is b0rked...
19:11.22[TK]D-Fenderor there is a 2nd form that is valid...
19:11.25iCEBrkr_fan_: ok, so then it's something in the [sip-provider-inbound] section of sip.conf
19:11.30P4C0ok, in * 1.2.1 how can I get the number of the person how is calling?
19:11.44P4C0s/how/who
19:11.46RoyK[TK]D-Fender: hm. seems you're right
19:11.49RoyKsorry
19:11.53*** join/#asterisk trym (n=trym@c213-158-252-242.sdsl.no)
19:11.54iCEBrkrP4C0: Dude, do you READ the wiki?
19:12.20DrukenP4C0: what do you think caller id is for?
19:12.21P4C0iCEBrkr, yes, but I have found like 521 different ways... and no one seems to work
19:12.23[TK]D-FenderWIKIwikiWIKIwikiWIKIwikiWIKIwikiWIKIwikiWIKIwikiWIKIwikiWIKIwikiWIKIwiki
19:12.29*** join/#asterisk N9URK (n=icechat5@user-0ce2dhc.cable.mindspring.com)
19:12.30iCEBrkrP4C0: Set(cid=${CALLERID(number)}) or something
19:12.47_fan_iCEBrkr: has anything changed in that section between prior to 1.0 and 1.2.1 that you can think of that would cause that problem?
19:12.48P4C0but ${CALLERID(number)? what number?? I need to get it...
19:12.50RoyKis there a way to make asterisk allow multiple SIP registrations from multiple hosts at the same time?
19:12.55*** join/#asterisk IOscanner (n=IOscanne@38.114.50.130)
19:13.14Vijayhello everyone
19:13.15iCEBrkrP4C0: um, yea. CALLERID() is a function that returns the number
19:13.22P4C0so setCIDNum(3800735) dosen't work?
19:13.34Vijayi am configuring an asterisk server on a gentoo linux system
19:13.36iCEBrkrP4C0: you said you wanna GET the CallerID not SET it.
19:13.36Drukenit SETS the CIDNUM
19:13.43IOscannerI am having a problem with my 4 port FXO lines.  I have 2 lines going to a 2 ports of a TDM 4 port FXO card
19:13.43P4C0iCEBrkr, yep, thanks
19:14.05RoyKP4C0: use the function
19:14.09P4C0so exten => 1717,1,VoiceMailMain(${CALLERID()}@voiceboxes) this is correct?
19:14.11Vijaythe hardware i want to use for caling is a 8 port fxs device, audiocode MP108
19:14.13DrukenIOscanner: let me guess... echo ?
19:14.13RoyKSet(CALLERID(num)=1234)
19:14.14IOscannerI can call out then it seem the card is hanging and not releasing the call.  I then have to wait a bit to get the line to release
19:14.26iCEBrkrEVERYONE LEARNING ASTERISK BOOKMARK THIS FUCKING PAGE
19:14.27iCEBrkrhttp://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands
19:14.30iCEBrkrNOW
19:14.30RoyKP4C0: NONONO. You need to specify what number.....
19:14.35P4C0RoyK, Set(CALLERID(num)=1234 <-- num?
19:14.36IOscannerI don't get any echo that I know of
19:14.37RoyK~docs
19:14.38jbotdocs is probably probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
19:14.40IOscannercall seems good
19:14.41Vijayany idea what's the specific configuration required for tis box in sip.conf & extensions.conf
19:14.41[TK]D-FenderP4C0 : When you want to set the callerid, use the reverse syntax of the function : Set(CALLERID(num)=1234567)
19:14.42RoyKP4C0: yes
19:14.49iCEBrkrhttp://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands
19:14.53RoyK[TK]D-Fender: i just told him :P
19:14.56Vijayi am configuring an asterisk server on a gentoo linux system
19:14.57iCEBrkrHAS YOUR ANSWERS
19:14.58Vijaythe hardware i want to use for caling is a 8 port fxs device, audiocode MP108
19:14.58DrukenIOscanner: callprogress=yes in zapata.conf
19:15.03IOscannerjust doesn't seem to release the line.  So then I get busy for a bit.
19:15.06iCEBrkrGeesus
19:15.07Vijayany idea what's the specific configuration required for tis box in sip.conf & extensions.conf
19:15.15[TK]D-FenderRoyK : be it wouldn't be called "ganging up on" if I didn't join in! ;)
19:15.20iCEBrkr>: |
19:15.24P4C0ok I'll read the wiki that iCEBrkr sent, thanks
19:15.30IOscannerit was commented
19:15.42rayvdShepherds kick ass!
19:15.46iCEBrkrDruken: Ooooooo
19:16.00azziewasn't Wiki deprecated in favor of this IRC channel ?
19:16.01Vijayhi
19:16.07RoyK[TK]D-Fender: :)
19:16.07rayvdhaha :)
19:16.08Drukencool ya down a lil :)
19:16.18iCEBrkrazzie: hehehe
19:16.27RoyKazzie: increase your medication, please
19:16.29rayvdwiki was deprecated in favor of direct emails to developers ;)
19:16.34IOscannerWould that cause the lines not to disconnect?
19:16.38P4C0ok: SetCallerID: Set CallerID. Deprecated in favor of CALLERID. Not yet, SetCallerID is in Asterisk 1.2.x, please check this if I am wrong. !?? so.. back to the begining
19:16.57RoyKP4C0: use the function
19:17.01*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
19:17.04P4C0RoyK, ok, thanks
19:17.13iCEBrkrDruken: I've said it a bunch of times.  But I'll say it again.  I'm not the brighest person in the world and if *I* was able to configure/program and tweak Asterisk via the Wiki, ANYONE can.
19:17.28[TK]D-FenderI'm with iCEBrkr on this one...
19:17.33rayvdiCEBrkr: my mother would be unable to :(
19:17.35iCEBrkr[TK]D-Fender: Which part? :)
19:17.37DrukeniCEBrkr: i couldn't have said it better myself
19:17.47rayvdshe can make some yummy waffles though
19:17.53iCEBrkrrayvd: mmmm waffles
19:17.55Drukenif i were a lightbulb, i'd be a 20watt
19:17.56[TK]D-FenderIf you understand programming flow at all, the only thing you need is a syntax list and Google :)
19:17.58P4C0RoyK, where is the documentation of the function... I'm not sure that I'm looking at the correct wiki... this seems like a joke for me
19:18.01Gh0stytwisted[asteria]: i checked both boxes and the #1 (meetme working) zttest works, no probs; #2 (meetme not working) zttest doesn't work, gives error: Unable to open zap interface: No such device or address
19:18.14[TK]D-FenderiCEBrkr : on using the WIKI for pretty much everything.
19:18.18iCEBrkr90% of the questions people ask in here are found on the Wiki.
19:18.27Vijayanyone knowing the ocnfiguration of sip.conf and extension.conf for audiocode mp108 fxs
19:18.31[TK]D-FenderiCEBrkr : I might push that to 95%
19:18.33RoyKP4C0: show function CALLERID
19:18.33*** part/#asterisk trym (n=trym@c213-158-252-242.sdsl.no)
19:18.34iCEBrkr[TK]D-Fender: Oh good, I was hoping you weren't agreeing with me about the 'brightest person' part :)
19:18.50[TK]D-FenderiCEBrkr : Never said I didn't ;)
19:18.54iCEBrkrLOL
19:18.54RoyKP4C0: and see http://pastebin.com/481904
19:18.59*** part/#asterisk _fan_ (n=allan@static-64-222-189-188.man.east.verizon.net)
19:18.59[TK]D-Fender:p
19:19.15iCEBrkrNow I understand that GotoIF() and some of the newer function stuff can be tricky..
19:19.24Vijayanyone knowing the ocnfiguration of sip.conf and extension.conf for audiocode mp108 fxs
19:19.29Drukengotoif is my friend
19:19.32[TK]D-FenderRoyK : lol
19:19.43RoyK[TK]D-Fender: :)
19:19.53iCEBrkrROFL
19:19.56P4C0RoyK, :p, ok you said callerid(num) and the fm says number.. so which one is it?
19:20.16[TK]D-Fenderfm?
19:20.28iCEBrkrFM minus the RT
19:20.30DrukenRTFM == "Read the FUCKING MANUAL!!!"
19:20.41[TK]D-FenderJKLhadlskjdhlkj&*A^23ug4ljhkgsaD
19:20.41Druken:)
19:20.42IOscannerif I set callprogress=yes I can't start *
19:20.59[TK]D-FenderIOscanner : Callprogress = trouble.  Don't use it.
19:21.13iCEBrkr[TK]D-Fender: eh? Callprogress works just fine for me!
19:21.13N9URKHI Guys, is there a way to pipe the output of "sip debug"? asterisk -rx "sip debug" > debug.txt only puts "SIP Debugging Enabled" into debug.txt.
19:21.17*** join/#asterisk mog_work (n=mogorman@gateway.digium.com)
19:21.25IOscannerWhat do you think is the problem?
19:21.27Druken[TK]D-Fender: he's having hanging zap lines... callprogress works for me... :)
19:21.30[TK]D-FenderiCEBrkr : On something non-T1 based?
19:21.37iCEBrkr[TK]D-Fender: For sure.
19:22.03IOscannerI can call out then it seems the TDM card is not hanging up the zap lines.
19:22.04_Sam--is there a big gain to getting the digium 2400/ w echo cancellation?  it costs alot than without echo canc...
19:22.05P4C0I will love to see a FM of asterisk as it, instead of having to cut little pieces from here and there in order to build it, so there's no asterisk's FM to read at!!!
19:22.14[TK]D-FenderN9URK : thats because it doesn't time the output and event are logged as they arrive.  as there is no set end point it quits after the message
19:22.18IOscannerI wait a bit then I can call again but until then I get fast busy
19:22.22mog_workpraise be
19:22.26IOscannerI can call in and it answers
19:22.29iCEBrkr_Sam--: From what I hear ( pun pun pun ) it's best to go with the EC card.
19:22.35DrukenIOscanner: do you have the hangup when busy on ?
19:22.41_Sam--ty for the advice
19:22.43IOscannerjust outbound seems to be the problem
19:22.58N9URKTk:  Thanks.  Is there some context that will let me log the output?  I didn't see anything in the docs
19:23.04iCEBrkrP4C0: Hey, a little common sense and it all fits together.. You think there's going to be a manual to document every crazy shit'n thing you people come up with??
19:23.28[TK]D-FenderN9URK : Nothing with a set point.  Couldn't begin to guess how you'd do it..
19:23.33IOscannerWhere would that be zapata.conf?
19:23.45N9URKTK: Thanks for the reply
19:23.50[TK]D-FenderiCEBrkr : Should I show him MY setup? ;)
19:24.05P4C0iCEBrkr, I don't think that making an extension so everyon cheks their voicemail is a crazy shit....
19:24.24[TK]D-FenderiCEBrkr : around 50 lines in my stdexten :)
19:24.44DrukenP4C0: i have one of those... *98, but it works with accountcodes, NOT CID
19:24.46[TK]D-FenderN9URK : just wish I had a better idea for you.
19:24.49DrukenCID can be falsafied
19:25.02_Sam--iCEBrkr:  if i bought a 2400 w/ echo canc. and 4 fxo...if i added more fxo later, i dont need anything additional related to the echo canc later?
19:25.18_Sam--just add the fxo modules?
19:25.24Drukennot only, i allow multipul numbers per account, with a changing CID
19:25.25[TK]D-FenderDruken : true, but you wouldn't give non-trusted calls access to the function.
19:25.43znoGthis is .. strange. Asterisk shows:     -- Playing 'digits/1' (language 'es')
19:25.45[TK]D-Fender* only uses CID from a phone if its not set in sip.conf (or similar file)
19:25.49P4C0Druken, ok I need to pass the externsion number (mailbox) of the person that's calling: exten => 1717,1,VoiceMailMain(${CALLERID(number)}@voiceboxes) can callerid(number) is getting the name not the number...
19:25.50znoGyet it plays the English "1"
19:25.59znoGand /var/lib/asterisk/sounds/es/digits/1.gsm exists
19:26.01Druken[TK]D-Fender: very true, but hey... i'm one of those untrusting make it hard for them kinda people
19:26.27[TK]D-FenderP4C0 : exten => 1717,1,VoiceMailMain(${CALLERID(num)}@voiceboxes) <---------------------------------
19:26.36Druken[TK]D-Fender: i set the CID in the dialplan.... i don't care what the customer wants for CID
19:27.00*** join/#asterisk trym (n=trym@c213-158-252-242.sdsl.no)
19:27.09iCEBrkr[TK]D-Fender: Geesh, hand him the answer. :)
19:27.12[TK]D-FenderDruken : My level of trust is in proportion to the control placed on a given thing being judged.
19:27.33[TK]D-FenderiCEBrkr : I do believe I did!  And RoyK also gave it to him and commented as such!
19:27.39iCEBrkr:)
19:27.50iCEBrkrYou know the wiki does state 'number' not 'num'
19:28.13iCEBrkrLART LART LART
19:28.13P4C0[TK]D-Fender, nop, it keesp asking for the mailbox, think that dosen't happend when I put ther extension number isted of ${CALLERID(num)}
19:28.30iCEBrkrP4C0: you can see a lot at the CLI and what CALLERID is evalutating to
19:28.38iCEBrkrevaluating even
19:28.57[TK]D-FenderP4C0 : aCTUALLY JUST PASTEBIN THE EVAL'D LINE
19:29.11P4C0iCEBrkr, it's getting the name no the number
19:29.12Gh0stywhat exactly is a "zap interface" ?
19:29.39P4C0[TK]D-Fender, what eval line?
19:30.13[TK]D-FenderP4C0 : the line wher you see the vmmain being called
19:30.39P4C0[TK]D-Fender, Executing VoiceMailMain("SIP/ruben-7682", "ruben@voiceboxes") in new stack
19:31.00[TK]D-FenderiCEBrkr : from the WIKI -> CALLERID(datatype) - Gets or sets Caller*ID data on the channel. The allowable datatypes are "all", "name", "num", "ANI", "DNID", "RDNIS".
19:31.07[TK]D-FenderiCEBrkr : not number!
19:31.10[TK]D-Fendernum!
19:31.36[TK]D-FenderP4C0 : pastebin your extensions.conf live for that exten.
19:31.36P4C0[TK]D-Fender, http://www.voip-info.org/wiki/index.php?page=Asterisk+func+callerid <-- this say number instead of num... and lower case for the last 3
19:31.49[TK]D-Fenderpfft!
19:31.55[TK]D-FenderMine works for me!
19:32.08P4C0[TK]D-Fender, exten => 1717,1,VoiceMailMain(${CALLERID(num)}@voiceboxes)
19:32.18iCEBrkrF pastebin for a second
19:32.20iCEBrkrdatatype may be one of the following:
19:32.20iCEBrkr<PROTECTED>
19:32.20iCEBrkr<PROTECTED>
19:32.20iCEBrkr<PROTECTED>
19:32.23iCEBrkr<PROTECTED>
19:32.25[TK]D-FenderP4C0 : now pastebin the setup info for that phone
19:32.25iCEBrkr<PROTECTED>
19:32.28iCEBrkr<PROTECTED>
19:32.44P4C0[TK]D-Fender, setup info? sip.conf?
19:32.44[TK]D-FenderiCEBrkr : I took mine from : http://www.voip-info.org/wiki/view/Functions
19:32.47iCEBrkr-- Executing VoiceMailMain("SIP/2101-5385", "s2101") in new stack
19:32.54[TK]D-FenderP4C0 : sip.conf if thats the origin
19:32.59iCEBrkryou see how ${CALLERID(numner)} eval'd to 2101
19:33.13iCEBrkrerr number even :)
19:33.39P4C0[TK]D-Fender, http://pastebin.com/481942
19:34.02iCEBrkr${CALLERID(num)} returns 1
19:34.07iCEBrkrSo you gotta spell out number.
19:34.15iCEBrkroh shit, wait.
19:34.19iCEBrkrBoth seems to work
19:34.26iCEBrkrI read the wrong line on the CLI
19:34.48IOscannerso any other ideas what would cause outbound fxo line to hang for a bit
19:35.00iCEBrkrIOscanner: Hang for a bit?
19:35.11iCEBrkrIOscanner: It takes awhile for it to place a call?
19:35.27IOscanner3-5 min
19:35.31iCEBrkrIOscanner: That's your dialplan definition.....
19:35.35iCEBrkr3-5 min?
19:35.36iCEBrkrgeesh
19:35.38IOscannerI have never see that before
19:35.51IOscannersame plan I am using on other systems
19:35.54iCEBrkrIf it were 10 seconds I could see it being the dialplan definition in your sip device.
19:35.59zoa2hey ho
19:36.01iCEBrkrRoyK: ew
19:36.02IOscannerI am not sure if it is the card I am using
19:36.05P4C0iCEBrkr, please tell me that you don't look me as a lammer and somehow understand my frustration!! I really read the manual but when I saw more that 1 command that do the same, and more than 1 definitions of that command I panic...
19:36.25iCEBrkrP4C0: well 4 people have handed you the answer.. does it work?
19:36.26IOscannerI have an openvox TDM 4 port card and this is the first time I have used openvox
19:36.39IOscannerI have always used Digium cards before
19:36.41iCEBrkrIOscanner: That'll learn ya :)
19:36.43P4C0iCEBrkr, no
19:36.51iCEBrkrOMFG
19:37.01IOscannernot sure if it is a card probelm or config issue
19:37.21iCEBrkrP4C0: http://pastebin.ca/35085
19:37.33iCEBrkrP4C0: now if that doesn't work for ya, you got other issues.
19:37.38IOscannerAnyone else use an openvox 4 port card?
19:38.07P4C0iCEBrkr, what's the s for!???
19:38.25iCEBrkrIf you read the Wiki for VoiceMailMain, you'd know
19:38.34iCEBrkrP4C0: http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands
19:38.38iCEBrkrGo there and figure it out
19:39.03badboyzIf the mailbox is preceded by 's' then the password check will be skipped
19:39.04P4C0iCEBrkr, that wasn't in the example where I get it...
19:39.16iCEBrkrP4C0: SO what?
19:39.29iCEBrkrGo read the VoiceMailMain() app description
19:39.43iCEBrkrhere's the link
19:39.44iCEBrkrhttp://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMailMain
19:39.47iCEBrkrsince you can't figure it out yourself.
19:40.03*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
19:40.12*** join/#asterisk dudes (n=dudes@12-215-32-62.client.mchsi.com)
19:40.18iCEBrkrYou people expect every example to be cut-n-pasted in your config and just work.
19:40.31iCEBrkrYa never go read up on the apps and function calls when shit doesn't work.
19:40.43badboyz'you people' i hate when those generalizations occur =/
19:40.49P4C0I want the password check... besides the s our lines are equal...
19:40.54iCEBrkrbadboyz: INCLUDING YOU! MISTER!
19:40.56iCEBrkr:D
19:40.57badboyzdamnit!
19:40.59iCEBrkrhahah
19:41.20P4C0:p
19:41.24iCEBrkrP4C0: Ok, then you don't have something set correctly in sip.conf
19:41.36*** join/#asterisk Entegrity (n=Entegrit@c-65-96-116-121.hsd1.ma.comcast.net)
19:41.40P4C0iCEBrkr, do I need to put the externsion number in sip.conf??
19:41.51IOscanneranyone know of anything else I can check that might cause zap lines to hang?
19:41.53iCEBrkrP4C0: Personally, I set the Mailbox= in my sip.conf
19:42.05P4C0iCEBrkr, I do as well
19:42.18iCEBrkrP4C0: and I don't use gay names in my [sections] I use the extension.
19:42.25iCEBrkrbadboyz: Got anymore?
19:42.34badboyzgave you my last one :(
19:42.37iCEBrkr:(
19:42.50iCEBrkrbadboyz: I figure I can go into a coma for the next 3hrs.
19:42.54Kattymy brain has insaned.
19:42.59P4C0iCEBrkr, humm maybe the gay names... but all the manuals are with gay names!!! i haven't see any one with extensions!!!
19:43.20*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
19:43.28Kattysomeone in here is from denmark.
19:43.42iCEBrkrP4C0: Well, personally it's easier to maintain.  Besides, what if that 'extension' has a new person there?  You gotta go rename the entry in sip.conf??
19:43.55iCEBrkrKatty: you're quick!
19:44.05twisted[asteria]Katty!!
19:44.08Kattysomeone i talk to a lot.
19:44.09iCEBrkrbadboyz: It was only 3 !'s.
19:44.15Kattytwisted[asteria]: hihi
19:44.16KattyiCEBrkr: hi.
19:44.19iCEBrkrKatty: Yea, you have a lot of netsex?
19:44.24Katty...
19:44.28KattyAriel_: are you from denmark?
19:44.30P4C0iCEBrkr, ok, well, I'll put the extension... but I will like to see the name of ther person that calls insted of the externsion number....
19:44.35KattyiCEBrkr: i'm trying to figure out who this mew years card goes to.
19:44.42Ariel_morning Katty
19:44.45KattyiCEBrkr: the name is drawing a blank in my head.
19:44.54KattyAriel_: hewwo (=
19:45.02iCEBrkrP4C0: Trust me.  Use extensions in your sip.conf [sections]
19:45.05KattyAriel_: was it you?
19:45.30Ariel_Katty, no I am American about 10th generation from Spain back in 1520....
19:45.38*** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw)
19:45.39*** join/#asterisk davidw_ (n=davidw@81-174-34-171.f5.ngi.it)
19:45.40iCEBrkrP4C0: If you want CallerID to show all the right info on the called extension, set 'callerid=' in sip.conf
19:45.58*** part/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted)
19:46.01KattyAriel_: kk
19:46.07iCEBrkrKatty: Apparently, you don't talk to them all that often or you wouldn't be struggling for their name. :)
19:46.09Gh0stythx twisted[asteria] and Druken
19:46.09P4C0iCEBrkr, in sip.conf? the function? ok I will... thanks
19:46.15KattyiCEBrkr: indeed.
19:46.22Ariel_Katty, but if people here still think I am Cuban.... but that is ok
19:46.26Gh0styfound the solution
19:46.35iCEBrkrP4C0: eh like [2101]. and use the callerid= option
19:47.26iCEBrkrP4C0: http://pastebin.ca/35086
19:47.28*** join/#asterisk implicit (n=implicit@200.12.227.205)
19:47.29*** join/#asterisk bkw___ (n=brian@ppp-69-155-251-101.dsl.tulsok.swbell.net)
19:47.30[TK]D-Fendercallerid="name" <number>
19:47.33P4C0iCEBrkr, thanks again
19:47.48[TK]D-Fenderthats what I thought the problem was... garbage in
19:48.00Katty[TK]D-Fender: maybe it was you.
19:48.04Katty[TK]D-Fender: but i don't think so.
19:48.09Katty[TK]D-Fender: in fact, i know it's not you. nevermind.
19:48.33iCEBrkrA friend in need is a friend indeed, a friend with weed is better!
19:49.05iCEBrkr[TK]D-Fender: The kid ain't yours.. Congrats
19:49.23SwK[Work]anyone around ATL with Comcast Business Cablemodem services?
19:49.28malverian[work]If a call is set up with Cfwd to another phone, the call is transfered to Local/{other_phone}
19:49.32Kattyi found his website!
19:49.38SwK[Work]katty :P
19:49.43iCEBrkrKatty: Stalker
19:49.44malverian[work]Is there some way for me to see what the original destination number was?
19:50.27KattySwK[Work]: he's on a different server ;)
19:50.34malverian[work]Or maybe a way to set a different context for 302 response calls.
19:50.37[TK]D-FenderiCEBrkr, kATTY .... :/
19:50.38Kattyhttp://www.slashnet.org/users/M0ffe/
19:50.41iCEBrkrNow where was I?  Oh yes.. PHP.. interface... yeah
19:50.42Kattyhardly stalking ;)
19:50.46P4C0iCEBrkr, now my clients can't connect... username/auth name mismatch...
19:51.04iCEBrkrP4C0: well, if you changed it in sip.conf, you gotta change it in the phone too
19:51.26P4C0the user name?? I only changed the [section] no the secret or username
19:51.57iCEBrkrP4C0: huh.. [section] is the username
19:52.10P4C0iCEBrkr, so the unsermae field dosen't mean anything?
19:52.25iCEBrkrP4C0: Did you see a username= in my sip.conf I pastebin?
19:52.45*** part/#asterisk Gh0sty (i=ghosty@kiekeboe.x-plose.be)
19:52.50P4C0iCEBrkr, :( there's no aliases here!????
19:52.59*** join/#asterisk riddlebox (n=blah@24-171-40-167.dhcp.stls.mo.charter.com)
19:53.11*** part/#asterisk alephcom (n=alephcom@openbsd.hagenhomes.net)
19:53.20riddleboxStrom_C:you around?
19:54.01iCEBrkrP4C0: aliases?
19:54.20Kattyexcellent.
19:54.22[TK]D-Fender?
19:54.27Kattyall the cards are going in the mail this evening!
19:54.35Kattyif you're on my list, you know it.
19:55.01iCEBrkrKatty: Ya know Christmas was 3 days ago, right?
19:55.36KattyiCEBrkr: you're clearly not on my list.
19:55.41KattyiCEBrkr: else you'd know what htis is all about.
19:55.41iCEBrkrPerfect.
19:55.48KattyiCEBrkr: kthxbi.
19:56.09iCEBrkrOMGZ! OH NOEZ! I was kthxbi'd
19:56.23implicitkaty
19:56.27iCEBrkrWhat shall I do now?
19:56.34KattyiCEBrkr: run away! </quote>
19:56.37implicitiCEBrkr, change your name?
19:56.39implicitperhaps?
19:57.09iCEBrkrShit, I can't use my sad face when I have a I DON'T GIVE A SHIT face on.
19:57.18iCEBrkr: |
19:57.26Kattythere you go.
19:57.28Kattygood enough.
19:57.49[TK]D-Fender./join #angst
19:57.59iCEBrkr[TK]D-Fender: yea, I'm PMSing I think.
19:58.25P4C0iCEBrkr, thanks dude, it's working now :)
19:58.25iCEBrkrP4C0: Of course it is :)
19:58.38iCEBrkrP4C0: now do you see how everything is lining up?
19:59.42P4C0iCEBrkr, yes, but I still have [mysipprovider-out] instead of the phone number that my sip provider gave me
19:59.55iCEBrkrThats fine.
20:00.02*** join/#asterisk razu_ (n=razu@ip220.cab17.mus.starman.ee)
20:00.05iCEBrkrP4C0: You're still unable to make outbound calls?
20:00.27iCEBrkrMmmmmmmmmmm Coookie!!!! 8()
20:01.00Kattymidol's for the weak.
20:01.14[TK]D-FenderNo, its not almond... its actually cyanide...
20:01.21[TK]D-Fender:D
20:01.34Katty;>>
20:01.59*** join/#asterisk Gimpy (n=d_akosh@h24-207-33-168.dlt.dccnet.com)
20:02.17*** join/#asterisk backblue (n=moo@87-196-41-95.net.novis.pt)
20:06.21*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
20:06.31shmaltzgmail is down
20:06.49backbluehi, with asterisk 1.2.1 should i have possibility to register the same user 2 times, and when someone try's to contact the user, both clients rings?
20:06.58shmaltzgmail is back up :)
20:07.09shmaltzbackblue, nope
20:07.11P4C0iCEBrkr, yes I can now :) thanks
20:07.16shmaltzjust the last one to register
20:07.24iCEBrkrbackblue: no
20:07.51iCEBrkrshmaltz: I think he wants both phones to ring
20:08.06iCEBrkrP4C0: oh coo
20:08.06backbluebut it will be implemented someday?
20:08.11shmaltziCEBrkr, thats what I'm assuming
20:08.19filebackblue: not as you think
20:08.20shmaltzbackblue, it's already implemented
20:08.21iCEBrkrbackblue: You can make both extensions ring in your dialplan
20:08.27shmaltzusing dial &
20:08.37Vijayanyone knowing the configuration of sip.conf and extension.conf for audiocode mp108 fxs
20:08.51iCEBrkrbackblue: Dial(SIP/1000&SIP/1001)
20:09.14shmaltziCEBrkr, you jewish?
20:09.18iCEBrkrbackblue: It's documented in the wiki
20:09.19fileoej is working on something that's cool... that *should* make it possible...
20:09.26iCEBrkrshmaltz: No, but my wallet wishes I was.
20:09.31filein the way you think.
20:09.32shmaltzlol
20:09.45P4C0Is ther a way to configure the mail options?? like the from and host to pass to sendmail?
20:09.58shmaltzP4C0, it's in voicemail.conf
20:10.02iCEBrkrP4C0: voicemail.conf
20:10.03shmaltzserveremailaddress
20:10.07shmaltzjinks
20:10.15iCEBrkrshmaltz: you owe me a beer.
20:10.17P4C0ok
20:10.25backblueiCEBrkr: ok, tks
20:11.39iCEBrkrw00t beer
20:11.47fileI have work to do and an email to complex :P
20:11.49fileno time to sleep
20:11.58shmaltz~sleep
20:12.01jboti heard sleep is overrated, and a poor substitute for caffeine.
20:12.01shmaltzuse caffeine
20:12.47iCEBrkrrequire_many("clsCaffeine.php");
20:12.48Twistercaffine > sleep =)
20:12.48Kattyfile: i'll sleep for you.
20:13.09Kattyfile: and you can handle my job, and yours too...all at the same time.
20:13.23iCEBrkrfile: That sounds like a shitty deal.
20:13.57iCEBrkrfile: Now it really sucks cuz you're gonna have drool on your shoulder.
20:14.24KattyiCEBrkr: i don't drool on file.
20:14.35linlinwhat does it mean if i pick up an extention and its likea delayed dialtone
20:14.40iCEBrkrWhat's that?
20:14.43linlinkinda like the voicemail indicator, but slower
20:14.44iCEBrkrerrrr. ummmmm.
20:14.46iCEBrkrnevermind.
20:15.04KattyiCEBrkr: you've clearly insaned.
20:15.12KattyiCEBrkr: please reboot yourself.
20:15.13iCEBrkrKatty: Correction.. Unsane
20:15.33iCEBrkrNot quite sane but not quite insane.  Unsane.
20:15.34*** join/#asterisk brookshire (n=nubb@gateway.digium.com)
20:15.36*** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr)
20:15.45*** join/#asterisk malcolmd (n=malcolmd@gateway.digium.com)
20:16.03Kattybrookshire: :>
20:16.05*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
20:16.07*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
20:16.39[TK]D-FenderERROR IN REALITY.SYS.  PRESS ANY KEY TO REBOOT UNIVERSE.
20:18.01[TK]D-FenderYou were supposed to press the "any" key!!!!
20:18.23Kattythere there, mister fender.
20:18.31Kattylet's not abuse trout.
20:18.37filetrying to access the work NFS on a home server = hahahahaha
20:18.48iCEBrkrfile: when you explode do you show the Mac Unhappy face?
20:18.49[TK]D-FenderNo File Security :D
20:18.56fileiCEBrkr: yes.
20:18.59iCEBrkrha
20:19.10*** join/#asterisk riddlebox (n=blah@24-171-40-167.dhcp.stls.mo.charter.com)
20:19.31iCEBrkr305 people in here.. 10 talk.
20:19.49riddleboxI like to watch you guys talk so I can learn
20:19.54filew... t... h...
20:20.02fileI talk too much
20:20.18riddleboxI dont want to say anything stupid
20:20.26[TK]D-Fenderriddlebox : just remember to do the exact opposite as we say or you'll insane!
20:20.32riddleboxlol
20:21.00fileo... m... g...
20:21.09[TK]D-Fender....yes my child? :)
20:21.10filewhoever wrote this SQL query needs to be shot
20:21.14fileno wonder it takes so long
20:21.20RoyKoh my gods....
20:21.39Kattyoh my apollo!
20:22.12RoyK8( ( Loke )
20:22.18riddleboxdoes anyone have charter cable internet?
20:22.27*** join/#asterisk areski (n=areski@223.Red-83-55-102.dynamicIP.rima-tde.net)
20:22.42Kattynext people will be saying douglas adams.
20:22.55RoyKKatty: that a known god?
20:23.01KattyRoyK: something like that.
20:23.03shmaltzriddlebox, yes:
20:23.05shmaltz===riddlebox <n=blah@24-171-40-167.dhcp.stls.mo.charter.com> riddlebox
20:23.06shmaltz===riddlebox: member of #Mandriva and #asterisk
20:23.08shmaltz===riddlebox: attached to irc.freenode.net http://freenode.net/
20:23.10shmaltz===riddlebox is identified to services
20:23.10utherdon't be silly.. Douglas Adams isn't a god... :P
20:23.11shmaltz---End of WHOIS information for riddlebox
20:23.20Kattyuther: pfft.
20:23.24iCEBrkrshmaltz: you ok?
20:23.40shmaltziCEBrkr, I'm not sure, why you asking?
20:23.42KattyRoyK: i learned that boys suck.
20:23.50utherthe babel fish would clearly proove that he is not a god
20:23.53iCEBrkrKatty: Girls have cooties
20:24.01BeirdoKatty: some boys suck.
20:24.06KattyBeirdo: yes, some are nice.
20:24.12riddleboxI was just wondering, in their contract it says you cannot have any server running, does a voip device like asterisk, or vonage or broadvoice count?
20:24.15RoyKKatty: that wiew of life won't help too much....
20:24.23iCEBrkrChicks dig jerks and nice guys finish last.. Amen.
20:24.24KattyRoyK: it's helped lots so far!
20:24.34utherriddlebox: that depends on how you use it... and whether they catch you
20:24.38shmaltzriddlebox, they define it whichever way they want
20:24.39BeirdoKatty: let me rephrase...  MOST men are dicks
20:24.44Beirdohehe
20:24.47*** join/#asterisk Vijay (i=Vijay@203.122.28.109)
20:24.54fileKatty: am I nice?
20:25.07*** join/#asterisk Vijay (i=Vijay@203.122.28.109)
20:25.15Beirdothankfully, some of us are odd, and try to be nice :)
20:25.16iCEBrkrRoyK: Way to go
20:25.24utherI'm nice, and sophisticated
20:25.26RoyKwtf? trying to rsync backup this powerbook, but...  7073700 files...
20:25.33RoyKit's not that large a drive.....
20:25.34riddleboxshmaltz:so I should try not to draw to much attention to myself huh, and not connect to my asterisk server from work or anything
20:26.06utherriddlebox:  I don't see how they would catch you anyway
20:26.22BeirdoKatty: how's your holidays been? :)
20:26.35riddleboxthats what I was wondering, unless I have huge amounts of traffic coming and going right
20:26.42shmaltzriddlebox, no, use it as you please, they will let you know if you are running anything, general it means anything for public use, so your personal server is not considered as such, kaza is considered a server
20:26.55utherriddlebox: if you're really worried, you could always use ssh to tunnel the voip traffic :P
20:27.09iCEBrkrMySpace is horrible slow at the moment.. It's only 3:30
20:27.10riddleboxuther: that would be a server technically
20:27.26iCEBrkrTypically only gets this way when it's 6pm on the west coast
20:27.29utherremotely accessing you system doesn't mean you're serving anything
20:27.34riddleboxiCEBrkr: I have been trying to update our stamps.com stuff all day, I think they have huge problems
20:27.53iCEBrkrriddlebox: SOmeone musta deleted the internet off their desktop
20:27.53Beirdostamps.com?
20:28.00iCEBrkrIt r broken
20:28.14Beirdoanother stamp collector type around?
20:28.24utherheh
20:28.27Kattyfile: you're dreamy.
20:28.41KattyBeirdo: blah, meh, an ehhn.
20:28.44iCEBrkrKatty: You're twisted.
20:28.46riddleboxBeirdo: it is a site for the USPS to purchase stamps and print them out
20:28.50trixterI need to work on my etel presentation ...
20:28.55KattyiCEBrkr: i'm so twisted i'm not even straight anymore.
20:29.01BeirdoKatty: well that's too bad.  sorry to hear it.
20:29.07trixterI almost dont want to go anymore becuase I dont have enough time to do everything I need to :/
20:29.10iCEBrkrKatty: Yea, yeah, that's it.
20:29.11KattyBeirdo: s'ok (= thanks for asking.
20:29.13Beirdoholidays should be relaxing along with the blah
20:29.26Beirdoriddlebox: ahhh, damn. :(
20:30.27BeirdoKatty: any time. :)  mine were a bit blah too.  stupid caffeine-withdrawl headaches
20:30.47Beirdoonce I figured out the problem, I stopped em right quick though
20:30.56KattyBeirdo: yeah i had that too ;)
20:31.06iCEBrkrNap time
20:31.11KattyiCEBrkr: nini
20:31.22Beirdotwo stupid caffeine pills, turn off the lights, almost sleep for 30min..  headache gone
20:31.37Beirdoshoulda just made me coffee, but forgot all about it
20:31.53utherholidays, meh!   I don't need an excuse to blow money
20:32.06Beirdobought those pills in that big power outage in the north-east...  they come in handy when you can't make coffee :)
20:32.38Beirdocaffeine withdrawl sucks.  I need to kick it some day, but coffee tastes so good
20:32.53utheryou could always switch to crack
20:33.09Beirdonah
20:33.15BeirdoI think caffeine's bad enough
20:34.06uthermaybe crack would be better for you
20:34.14Beirdodoubt it
20:34.21BeirdoI don't wanna be a crack whore
20:34.34BeirdoI think my fiancee would agree :)
20:34.37utherwhy would you?  could you not afford it?
20:34.44fileBeirdo: VoIP whore!
20:34.53Beirdonow, a VoIP whore..  yeah
20:35.01utherwow
20:35.02utherhow....
20:35.05utherbranded
20:35.13Beirdousing VoIP has saved me soooo much money :)
20:35.27utheryou should get 'file' tatooed on your ass
20:35.28*** part/#asterisk bkw_ (n=bkw_@ppp-69-155-251-101.dsl.tulsok.swbell.net)
20:35.30slappingthave you guys watched the asterisk vidcast on http://revision3.com/systm/ ?
20:35.43fileuther: ...or not
20:35.44utherhaha, and only we would know what it was truly refering to
20:36.08utherfile: aye, you don't wanna be embeded on Beirdo's ass
20:36.11Beirdooooook...   I think uther needs to put down the bong
20:36.14*** join/#asterisk kenrstone (n=krstone@ool-4573f3dc.dyn.optonline.net)
20:36.19utherits not a bong!
20:36.28utherits a vaporizer!
20:36.29uther:D
20:36.54seele_I need help here please... got an FXO with a very high response time, i mean a PSTN call from outside is taking too long
20:37.19seele_too many rings without the phone actually ringing
20:37.30utheryou know you want it
20:38.03Beirdoheh
20:38.09BeirdoI only do caffeine
20:38.17Beirdoand occasional alcohol
20:38.23utherhey, I graduated DARE.... (Drugs Are Really Excelent)
20:38.27fileuther: it's not safe to IRC from work
20:38.27utherhaha
20:38.31riddleboxcoffee soda, whatever gets the job done
20:38.36Beirdoand I used to do occasional cigars
20:38.58Beirdobut my fiancee told me she wouldn't kiss me if I smoked em.   Bye bye cigars.
20:38.59utherseele_: turn off caller id
20:39.20uther.. hrmm... I'd have stuck with the cigars
20:39.28Beirdono way
20:39.34*** part/#asterisk kenrstone (n=krstone@ool-4573f3dc.dyn.optonline.net)
20:40.22utherno need for gratuitous kissing unless its goin' somewhere... and I can brush my teeth before I put those moves on
20:40.29Beirdobah
20:40.32Beirdobah and bah
20:40.42Beirdooh, and humbug
20:40.43Beirdo:)
20:40.48uthernot that any non-smoking woman has ever complained
20:41.08Beirdotrust me, cigars can't just be brushed away
20:41.14Beirdothe taste is with ya for days
20:41.29Beirdoand it's in your hair even after several showers, etc.
20:41.39BeirdoI can live without em easily.
20:41.42utherI smoke a pack-a-day of camels, and a pipe
20:41.47*** join/#asterisk riddlebox (n=blah@24-171-40-167.dhcp.stls.mo.charter.com)
20:41.54riddleboxman I hate it when that happens
20:42.06utherdid they catch on to your server?
20:42.08uther:D
20:42.35riddleboxno I hit the touchpad which was on the disconnect from server button and it cut me off
20:42.58Beirdohehe
20:43.09utherriddlebox: might adjust your pad's sensitivity
20:43.12Beirdotouchpads get disabled on my machines
20:43.19BeirdoI'll use the clit-mouse, thanks
20:43.30*** join/#asterisk squinky86 (n=ASGjon@64.89.118.139)
20:43.34utherI always prefered the little joystick in the middle of the keyboard
20:43.44Beirdoyup
20:43.47Kattythinkpads++
20:43.53Beirdothinkpads are great
20:43.54Beirdo;)
20:44.02Beirdouse one at work, got two of my own at home
20:44.33Beirdogeek toys++
20:44.43utherwhy
20:44.55Beirdowhy what?
20:45.01utherwhy would you have 3
20:45.07Beirdoone belongs to work
20:45.12Beirdotwo are mine
20:45.13Beirdo;)
20:45.20Beirdoone runs Linux, the other Windows
20:45.23utherbut uhh... I think you just might be missing the point
20:45.26utherof a laptop
20:45.33Beirdonope
20:45.36*** join/#asterisk alexhopper (n=a27386@CPE000103d29ae2-CM001225dfdfe0.cpe.net.cable.rogers.com)
20:45.51Beirdowhen I no longer work here, I'll still have laptops
20:46.20Beirdoand the point is to lie on the couch using wireless
20:46.26Beirdoand eventually, in the back yard :)
20:46.32shmaltzinteresting:
20:46.33shmaltzhttp://www.breitbart.com/news/2005/12/28/051228195053.3uxtz5v6.html
20:46.34utherI have a laptop... I never use it... unfortunatly I;ve been totally burned out on computers... if I'm not working, I just don't even like fucking with them
20:46.46SkramX:)
20:46.47BeirdoI play games, chat, etc
20:47.03shmaltzimagine a DOS on a phone system network that runs Asterisk
20:47.16riddleboxuther:I am trying to learn asterisk,c++ at the same time
20:47.54utherasterisk isn't too hard
20:48.00utherof course, that depends on what you're doing
20:48.04riddleboxnahh not bad at all
20:48.31utherc++ and asterisk put in the same context though is a bit baffeling...
20:48.52seele_How do i turn off the caller ID??
20:48.52riddleboxwhy?
20:48.54nswintbenjk sent me to http://www.automated.it for some Home Automation AGI's.. anyone know where I can find any examples.  I saw John Todd on Systm with Kevin Rose and he mentioned an example where this guy has his his X10 system controlled by Asterisk and gives his kids math tests at their extensions
20:49.14utherwith c++, you gotta learn a language, with asterisk you just gotta hammer through the whacky configuration
20:49.22*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
20:50.27riddleboxI got the Oreilly book for christmas, and I am reading it, its not to bad at all
20:50.44backbluedoes anyone know what do i have to put in package.use (in gentoo) to emerge asterisk-1.2.1?
20:50.57utherseele_: zapata.conf   usecallerid=no
20:51.48nswintriddlebox Mark Spencer put the pdf online ya know
20:52.06utherwow, I think something lost in translation
20:52.11seele_thanx
20:52.11riddleboxits ok, I like print
20:52.33_Sam--how would you dial different extensions based on which ZAP interface a call comes in on...example, i have a client with a tdm2400 w/8 fxo....how do i make calls that come to the first 4 fxo dial a certain extension
20:52.47*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
20:53.29uther_Sam--: put them in a different context
20:54.50_Sam--i see thanks
20:54.54utherno problemo
20:55.48_Sam--different contexts for the channels in both zapata.conf and extensions.conf right?
20:56.26utherthe context in zapata define what context calls from that channel are dumped into
20:56.42utherthe context you specify in zapata must exist
20:56.42_Sam--yep...i see it...thanks again, just wanted to be sure.
20:57.09*** join/#asterisk dasuberdavid (n=david@gateway.digium.com)
20:57.16lesouvageWhy is passing of parameter ${inputnummer} failing in this line <<<<<    exten => s,1,System(/var/spool/asterisk/script/callbestandscript ${INPUTNUMMER})  >>> The script itself is working fine
20:59.06utherlesouvage: you sure that variable has anything in it?
20:59.28lesouvageuther: yes it even shows on the cli
20:59.43nswintSomething interesting to think of... wouldn't Asterisk be a much cheaper tool for all those podcasters using ISDN lines and uber expensive radio mixer boards
20:59.43utherlesouvage: how do you know its not passing it then?
21:00.34*** join/#asterisk P4C0 (n=paco@200.124.22.34)
21:00.44utherlesouvage:  try something like  System(echo ${INPUTNUMMER} > /tmp/somefile)
21:01.09utherlesouvage: and see if it echos its value in that file
21:02.21P4C0hello again :)
21:02.22nswintthey could mute their sip iax or softphones when not talking and they can record callers and  have asterisk to process everything.. they could just edit the individual files afterwards
21:02.36nswintI need to try that
21:02.41nswintmake a podcast using asterisk
21:02.50utherwouldn't the quality suck?
21:03.08nswintpodcasts are usually at 64 Kbps
21:03.11P4C0I'm having a little problem... I'm using asterisk to registry to my sip server, it goes fine, but after a while I get handle_response_register: Forbidden - wrong password on authentication for REGISTER for 'mynombre' to 'myserverip'
21:03.45P4C0and then sip show registry says Request Sent
21:04.15utherP4C0: aye, its retrying...  if I had to guess, I'd say wrong password, heh
21:05.19P4C0uther, yes I know, but the first tiem goes ok
21:05.39utherdoes the cli ever say "Registered"?
21:06.14P4C0uther, I mean, I launch the * it registers with my sip provider I can make call from the outside into * and from * to outside using my sip provider, the sip show registry sais registered, and after about 30 minuts I get what I told...
21:06.14badboyzlesouvage: check the permissions to that script
21:06.48utherP4C0: hrm... who's the provider?
21:07.42P4C0uther, local one, the strange is that if I put my jsphone directly it works fine... really fine
21:07.55lesouvagebadboyz: I check it now
21:08.32lesouvageuther: the number was written to the file
21:08.59P4C0uther, is like it send the registry with the correct values, and after that when it re-registry it dosen't sent the password, or miss the ip address or I don't know it's strange
21:09.35utherlesouvage: well, that removes asterisk from the troubleshooting process... its either permissions like badboyz said, or the script is written incorrectly
21:09.40nswintstupid question... with a TDM400p with FXO can you dial back out on those channels
21:10.23lesouvagebadboyz: this are the permission (output ls -aRl) -rwx------   1 asterisk asterisk  422 2005-12-28 21:52 callbestandscript. It looks OK.
21:10.27badboyznswint: yes
21:10.38nswintbadboyz: kewl
21:10.52badboyzFXO you can recieve & place calls, FXS is for terminating to a device only
21:11.16badboyzlesouvage: is the script not running whatsoever?
21:11.20nswintyeah I thought so but I thought that the TDM400P might have been different
21:11.28utherlesouvage: might try the insecure option
21:11.32badboyzlesouvage: is it a .sh script?
21:11.40P4C0and my sip provider is using asterisk as well, so I'm not sure what can be the problem
21:11.47badboyznswint: same here
21:11.58badboyznswint: i got the same card rather
21:12.57nswintI'm so freaking excited about asterisk... that and mythtv are the best things that have come to opensource in the past few years
21:13.08lesouvagebadboyz: I put it on pastebin.ca
21:13.09badboyzyea, they both are hot stuff :)
21:13.14badboyzlesouvage: link me
21:13.21lunknswint: screen should be mentioned too, but it's not that new
21:13.23utherP4C0: do you have the pw entered in the register line only? or also for the sip entry for that server?
21:13.47nswintlunk: that's true.. forgot how often I use that too :-)
21:13.48P4C0uther, in bouth
21:14.21iCEBrkrlesouvage: su to asterisk and then test that script..
21:14.48iCEBrkrlesouvage: if that works, then try again but with /usr/sh -c and your script name
21:15.13badboyzi know in the past ive have to drop the .sh extension from the script to make it work using System()
21:15.32iCEBrkrShouldn't have to.
21:15.37badboyzi realize that
21:15.54badboyzbut regardless it did the trick
21:16.23utherP4C0: you might try forcing a shorter expirey time
21:17.00[av]baniwhat? you can certainly receive and place calls on FXS... my phone certainly rings on the FXS port, and I can certainly dial out with it :)
21:17.13[av]banibetter to say, FXO is switch-facing and FXS is device-facing
21:17.47uthereasiest to remember,  fxO devices are Objects, and fxS is the Service
21:17.57lesouvagebadboyz: the script is on http://pastebin.ca/35090. I would really appriciate it if you take a look.
21:18.23badboyztaking a peek
21:18.31[av]banior, fxo is office, fxs is station
21:18.42P4C0uther, how, I have sip debug but I can't see any package to my provider... strange
21:19.31utherP4C0: in your sip.conf:   defaultexpirey=120   for example
21:19.38*** join/#asterisk J4k3 (i=j4k3@dhcp-12-197-128-55.intrastar.net)
21:19.54iCEBrkrlesouvage: You could optimize your script by using /bin/sh instead of /bin/bash
21:19.55uther[av]bani: I like mine better :D
21:21.20P4C0uther, when I got registered the refresh changes to 105...
21:21.23badboyzlesouvage: check your PM
21:21.41utherP4C0: yea.. its counting down to its reregistration
21:21.51P4C0uther, right now I'm registered (I reload the sip)
21:21.56*** join/#asterisk dja (n=alden@zaphod.math.ohio-state.edu)
21:22.01P4C0uther, that's in minutes?
21:22.04utherP4C0:  no
21:22.18utherP4C0: what was the registration time before you forced it?
21:22.22P4C0uther, is that is'n in minutes it's not counting down...
21:22.35utherP4C0: its seconds
21:22.40P4C0uther, the default.. I did have that line in my sip.conf
21:22.52utherit was already in there, set at 120?
21:23.04P4C0uther, no, it wasn't
21:23.19lesouvagebadboyz: did you saw my reaction. I have some strange message about blocked messages
21:23.28P4C0but when sip show registry was registering the refresh was in 120, not it's on 105
21:23.48badboyzlesouvage: no -- make sure that you are currently registered w/ nickserv to reply to messages
21:24.00badboyz./msg nickserv IDENTIFY <password>
21:24.14*** join/#asterisk Meaty (n=cp_simbu@office.abi.ca)
21:24.32utherP4C0:  maybe it took 15 seconds to register and list
21:24.45utherP4C0: it wasn't set before now?
21:24.49lesouvagebadboyz: it's done, I'm recognized
21:24.55*** join/#asterisk DrScriptt (n=gtaylor@208.47.119.117)
21:24.58badboyzlesouvage: kk resend your reply to me
21:25.01P4C0uther, no, should I set it to 105?
21:25.27utherP4C0: no, leave it... after 2 minutes it should reregister
21:25.51utherP4C0: do you have verbose turned on in the cli?
21:26.04DrScripttHi all.  I have an odd issue with my VoIP provider (TelIAX) that I would like to get some peoples oppinions on.
21:26.11P4C0my command live have a lot lot lot of vvv :p
21:26.32DrScripttI have an Asterisk v1.0.7 system that is using SIP to register with my VoIP provider via the G.729 coded.
21:26.52P4C0uther, yes
21:26.55badboyzYES YES AND DR!?!
21:26.57DrScripttToday my provider experienced a failure such that G.729 would report as a possible codec yet fail.
21:27.01P4C0uther, should I enable the sip debug?
21:27.01badboyzOH NOES
21:27.18badboyzwho is your provider?
21:27.21DrScripttIs it possible for SIP in general to attempt to use one codec and partialy suceed and then fail to another coded.
21:27.24DrScripttTelIAX
21:27.32badboyz$T@>$KJGadsf. GROAN
21:27.41DrScriptt?
21:27.44grandyIf I'm recording a call and the h extension is called, can I assume that the recording file is completely flushed to disk?  Or is there a way to make sure it is?
21:28.30badboyzgrandy: flushed to disk?
21:29.07grandybadboyz: that the recording isn't buffered somewhere...
21:29.19uthergrandy: umm, I think thats an effect of your filesystem, not asterisk
21:29.31DrScripttBadBoyz:  Was that groan towards TelIAX or something else?
21:29.37uthergrandy:  ext2?
21:29.47badboyzDrScriptt: yea it was toward teliax, have heard some not so pleasant stories about them
21:30.06uthergrandy: if the recordings are *that* precious, I suppose you could do a  System(sync)
21:30.19grandyuther: well, i mean will the file be 100% written and usable as a file when the h extension is called...as opposed to being in some intermediate state (such as in an asterisk buffer)...
21:30.42grandyuther: not to preserve them to disk, just to know that it's OK to copy  the file elsewhere on the filesystem, etc.
21:30.50badboyzgrandy: from my experience when the h is set, the file is written to the disk
21:30.51utherI don't think asterisk buffers recordings...
21:31.01iCEBrkrgrandy: when recording is done, it's done.. no if's about it
21:31.12grandyiCEBrkr: excellent... thanks uther and badboyz
21:31.26badboyzthank the iceman too
21:31.27DrScripttIs it possible for SIP or IAX to attempt to negotiate a 2nd codec to try to use if for some reason the first agreed upon coded fails?
21:31.30iCEBrkrgrandy: I know this cuz I run my .wav files through sox
21:31.33grandythanks iCEBrkr
21:31.35badboyzlol
21:31.41badboyziCEBrkr: mp3 compression FTW
21:32.05iCEBrkrbadboyz: well, you have to merge the two streams and of course MP3 + downsample them.
21:32.11badboyzyou muxing them or converting em?
21:32.13badboyzwell yea
21:32.20grandyiCEBrkr: ok then i have another question for you...  when i do gsm to mp3 compression i find that if i "upsample" to recording to 16Khz it sounds way better even though i'm not adding any information to the file
21:32.34iCEBrkrThe only down side is that the channel is 'busy' or still 'offhook' while the merging and converting is going on
21:32.42grandy.
21:32.43grandyoops
21:32.45utherDrScriptt: well... the codec is agreed upon when the call is established
21:32.49grandymy term is acting up
21:32.55grandytest
21:32.57badboyzhi
21:33.03iCEBrkrSo if you have a LONG conversation, the line could be busy for abit
21:33.24badboyzeh, i dont see how "upsampling" something improves its clarity
21:33.42badboyzbut im not very up on my resampling either
21:33.51badboyzid assume, like the saying goes, cant polish a turd :)
21:33.56utheriCEBrkr: does it still hold the channel even after you issue a hangup?
21:34.08*** join/#asterisk grandy (n=mmmurf@pcp05305753pcs.wanarb01.mi.comcast.net)
21:34.18iCEBrkruther: You can't hang up until your done converting.
21:34.48grandyoops.. i'm back... iCEBrkr, i was just wondering if you have any ideas on why my upsampling makes the recordings sound so much better, and if there's a way to do gsm --> mp3 conversion and preserve as much quality as possible...
21:34.57iCEBrkrgrandy: you can't upsample.
21:34.59utheriCEBrkr: but if you recorded, then hungup then called a script to convert... would it still hold the channel until its terminated from the dialplan?
21:35.07*** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt)
21:35.34badboyziCEBrkr: your using the m flag w/ Monitor, right?
21:35.37uthergrandy: the only reason you would double your sampling rate is if you were converting it from digital, to analog, to digital more than once
21:35.38iCEBrkruther: the phones are hung up, the script fires to merge + convert, then the line is hungup.
21:35.41*** join/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net)
21:35.50iCEBrkruther: If you hangup() before then, you lose all your channel data/variables.
21:36.01utheriCEBrkr: good point
21:36.10grandyuther: it's weird, though, that sox file.gsm -r 16000 file.mp3 resample results in a much better quality mp3 than sox file.gsm file.mp3
21:36.17iCEBrkrbadboyz: I'm using my own routines Record() and such.   Monitor is new
21:36.30*** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk)
21:36.45badboyziCEBrkr: gotcha, Monitor has a nice m flag that calls an external script after the recording finishes
21:36.49uthergrandy: better quality when played out of what?... and through what?
21:37.18grandyuther: through headphones ... just using mpg123
21:37.21seele_My asterisk was behind an analog PBX, now that i put it first it doesnt recieve calls...
21:37.23seele_help
21:37.30seele_outside calls get normal tone
21:37.31uthergrandy: not through asterisk at all?
21:37.37iCEBrkrbadboyz: Yeah, but you most likely still lose your channel data
21:37.49badboyziCEBrkr: define channel data?
21:37.55iCEBrkrbadboyz: ${VARS}
21:37.56grandyuther: not playing the recordings back through asterisk, just through mpg player
21:38.00iCEBrkrCALLERID(name)
21:38.02iCEBrkretc
21:38.02badboyziCEBrkr: nope, it passes that into the file
21:38.09seele_My asterisk was behind an analog PBX, now that i put it first it doesnt recieve calls.. help
21:38.10badboyzrather into whatever script you call
21:38.16iCEBrkrseele_: Switch it back.
21:38.24seele_what do u mean?
21:38.33iCEBrkrPut your PBX in front of Asterisk.
21:38.36iCEBrkrProblem solved.
21:38.41uthergrandy: thats odd... could it be a self-fufilling prophecy?  I don't think sox by default would do a FFT
21:38.43seele_No io can't i replaced the PBX for asterisk for good
21:38.49seele_it was an old Alcatel
21:38.58iCEBrkrseele_: Well then, maybe you should have tested it before going live??
21:39.24seele_I had the asterisk behind just for making tests
21:39.24utherheh
21:39.24_Sam--[av]bani you around?
21:39.24seele_Yes and it was fine
21:39.24utherseele_: I did the same converstion from a panasonic dbs (of course, mine worked :P )
21:39.27grandyuther: not sure... i was just experimenting with various sox options and i was unhappy with the quality of sox file.gsm file.mp3
21:39.32utherseele_: whats your setup?
21:39.44seele_Which one (of all)
21:39.53seele_I got 3 FXO's trunks
21:39.59iCEBrkrbadboyz: I'll have to tinker with Monitor() if it frees up the channel and preserves all channel data, that'll work swell.
21:40.01utherseele_: "calls aren't comming in" doesn't help much
21:40.15*** join/#asterisk zishanov (n=mail@d57-249-149.home.cgocable.net)
21:40.25seele_uther, i know but im kind of a newbie
21:40.41utherseele_: have you configured your channels in zapata.conf ?
21:40.47iCEBrkrSo many people get themselves in trouble doing shit like that.
21:40.58seele_uther, i haven't i do it through aah
21:41.01utherseele_: does the context you placed the channels in actually exist and have an 's' context?
21:41.30utherseele_: I don't know what aah is
21:41.38seele_Asterisk At home
21:41.45seele_AMP
21:42.02utheryou should take the config tools out of the equation
21:42.03badboyziCEBrkr: its good stuff, i recommend taking a shot at it when you have time
21:42.05fugitivocrAMP
21:42.14seele_KO
21:42.16seele_ok
21:42.30seele_but i am totally new to this
21:42.34DrScripttDifferent question:  How many people are running their VoIP setups on a Layer 2 (or 3) managed network with their voice traffic on a different VLAN than their data traffic?
21:42.47utherseele_:  www.voip-info.org is your friend
21:45.05DrScriptt*nod*  VoIP-Info.Org has a LOT of GERATE information.  It is very easy to find what you need on their site too.
21:45.13_Sam--how much config on external FXO gateways are required?   basically you tell it to register to asterisk, and you tell asterisk to DIAL(SIP/externalgw/${EXTEN}) ?
21:45.33utherfxo gateway?
21:45.41_Sam--ive never used any external devices...and i just am not clear how you would dial out from asterisk to an FXO gateway
21:46.01utheryou mean a sip to fxo device?
21:46.06hypa7iaDrScriptt: when i was working for an ILEC, we did nothing but separate-vlan installs
21:46.10iCEBrkr_Sam--: Try Dial(ZAP/g1...)
21:46.23DrScripttI have not used an FXO gateway per say but I have used a Cisco ICS 7750 MRP that used a VWIC to interface with a T1 and it's dial statements were not difficult, I'd be happy to get you an example if you need / want.
21:46.23_Sam--how is it ZAP if its in an external FXO gateway?
21:46.37iCEBrkrWTF is an external FXO gateway
21:46.42uther_Sam--: you keep saying FXO gateway
21:47.01hypa7iaDrScriptt: it's in the best practices for cisco and i think nortel
21:47.01uther_Sam--: try  SIP to ATA converter, or ata device... or freakin sip to fxo maybe
21:47.02fugitivoa gateway with FXO ports?
21:47.09_Sam--finding you the URLs
21:47.09fugitivoFXO to SIP?
21:47.18_Sam--gateway with FXO ports
21:47.29iCEBrkrThat still doesn't tell us anything
21:47.38Mothermultiple greetings
21:47.42DrScripttiCEBrkr, are you trying to find one or asking what they are?
21:47.53badboyzi sure hope he isnt refering to a TDM as a gateway :)
21:47.53P4C0what's the best way to send / receive fax? for rxfax I need to rebuild asterisk right!?
21:47.59Mothercan anyone recommend a good quality voice for festival? from the file size I guess HTS are best?
21:48.00fugitivowhat's difficult with a FXO gateway?
21:48.08badboyzP4C0: dont do faxes w/ asterisk
21:48.08iCEBrkrDrScriptt: huh?  Sam over there is spewing nonsense, I'm trying to figure out what he means
21:48.12_Sam--something like this:   http://www.voipsupply.com/product_info.php?products_id=1230
21:48.25_Sam--that is called a quad external fxo gateway...same as i been calling it
21:48.26P4C0badboyz, why? how can I do fax then??
21:48.31fugitivoP4C0: spandsp
21:48.32iCEBrkrP4C0: Dude, you couldn't even get voicemail callerid stuff working, stick with basics before you get faxing
21:48.34badboyzbuy a fax machine
21:48.40_Sam--like if you plugged 4 fxo lines into that thing...how do you make asterisk call out it?
21:48.48Motherfax? asterisk? buy a fax machine :)
21:48.53uther_Sam--: whats the other end hooked up to?
21:48.54P4C0fugitivo, spandsp = rxfax txfax right?
21:49.00fugitivo_Sam--: register asterisk to sip accounts like a sip service
21:49.07lesouvageMother: Cepstral isn't festival but with an impressive quality and not expensive (unfortunately not available in Dutch)
21:49.13P4C0iCEBrkr, fax is basic :)
21:49.21*** part/#asterisk mkrufky (n=mk@68.160.103.77)
21:49.22badboyzseriously
21:49.22iCEBrkrP4C0: LOLOLOLOL
21:49.23_Sam--uther:  the other end is on the same LAN as the asterisk box...
21:49.23Motherlesouvage: thanks for that tip :)
21:49.25badboyzdont fax w/ asterisk
21:49.25fugitivo_Sam--: plug the lines to the FXO ports, setup the SIP accounts, and use that accounts with asterisk
21:49.25iCEBrkrP4C0: No it's not.
21:49.31fugitivobadboyz: why not?
21:49.40badboyzUDP
21:49.42badboyzis not your friend
21:49.43P4C0iCEBrkr, I have found better docs for spandsp than voicemail...
21:49.45iCEBrkr_Sam--: You'd register your asterisk box with that device.......
21:49.47fugitivobadboyz: ??
21:49.52lesouvageMother: if you wait a moment I will paste a interesting link
21:49.53iCEBrkrP4C0: Well, good luck man..
21:49.55_Sam--i see...so asterisk registers TO that device
21:49.55utherP4C0: fax over ip is a bitch... you're converting digital, to analog, to digital, to analog
21:50.12_Sam--not vice versa?
21:50.12fugitivoP4C0: you need fax over ip or just fax with asterisk?
21:50.16P4C0iCEBrkr, thanks!
21:50.16_Sam--just reigstering to, say, broadvoice or teliax
21:50.19iCEBrkr_Sam--: yea, and you'd still dial(sip/)
21:50.24badboyzwhen you fax over voip, you are begging for muggled documents
21:50.30badboyzyes i said muggled
21:50.32P4C0fugitivo, i just need to receive fax with asterisk... I think that not even send it
21:50.35fugitivohe didn't say he wants to fax over iP
21:50.40_Sam--thank you all
21:50.41iCEBrkrbadboyz: Who says anything about doing faxing over IP?
21:50.45uther_Sam--:  dial(sip/extension@fxogateway)  heh
21:50.48badboyziCEBrkr: chances are he wants to ;)
21:50.49fugitivoP4C0: using pstn or voip?
21:51.00P4C0fugitivo, voip, I dont have pstn
21:51.03_Sam--how would you deliver calls FROM that device to *?
21:51.04fugitivobadboyz: no, chances are 50 and 50
21:51.06badboyzbooyah!
21:51.08badboyztold you !!
21:51.12badboyzhaha
21:51.12_Sam--sorry to sound dumb i just never have seen it
21:51.17iCEBrkrbadboyz: Regardless, he had issues figuring out how to get CALLERID() working. How's he gonna get faxing working even over PSTN??
21:51.20_Sam--that thing has a dialplan or something?
21:51.24DrScripttI have an install that is using a Cisco ICS 7750 that receives the PSTN lines via a Voice T1 and that has an FXS prot that the fax is connected to and it is functioning just fine, but that is not VoIP.
21:51.28fugitivoP4C0: then forget it, get an analog line for faxing
21:51.35Motherlesouvage: thanks!
21:51.40iCEBrkrP4C0: Do you even understand faxing and codec compression?
21:51.48lesouvageMother: check   http://www.voip-info.org/tiki-index.php?page=Asterisk+text2cepstral+www+demo
21:51.51badboyziCEBrkr: i knew he was shooting for faxing over voip though based on all the other idiotic questions he asked ;)
21:51.55iCEBrkrP4C0: I wish you even MORE luck faxing voip. :)
21:52.01iCEBrkrbadboyz: lol
21:52.17P4C0:'(
21:52.27fugitivobadboyz: lol
21:52.38badboyznow i dont discourage using asterisk for faxing, if you have a fxo setup in place
21:52.46P4C0iCEBrkr, I know that I can't use gsm  but I'll check with my provider
21:52.57DrScripttThe only real faxing over IP solution that I've even read about (no experience with it) is T.38 (someone also said T.37 (different)) gateways that convert the analog fax to a data docuemnt and transmit the document and then convert it back to a fax.
21:53.01Motherlesouvage: thanks a bunch for that, very helpful
21:53.23Uther_Pthere is a real fax over ip solution.. its been around forever and it works great!
21:53.25iCEBrkrDrScriptt: That'd be the way to go.
21:53.26Uther_Pits called EMAIL!
21:53.30badboyzlol
21:53.31fugitivot38 is the solution, cisco has it working, but it's experimental under asterisk
21:53.32badboyztrue true..
21:53.45_Sam--i use hylafax on my * box but i have an external modem with fxo
21:53.49_Sam--works great as well
21:53.58iCEBrkr_Sam--: Yea, that'll work fine
21:54.01badboyzyea hyla is nice for that
21:54.05Uther_Psomeone needs a beating for faxes still being a nessesity
21:54.12DrScripttLOL
21:55.06*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
21:55.08N9URKYou can say that again Uther
21:55.25Uther_PI've got a server in the closet with a fax modem, running freebsd and efax.. it takes the freakin fax, converts it to a pdf file, makes a pretty little tiff thumbnail and emails it to a mailing group... simplest freakin solution... and think of all the trees I've saved!
21:55.26N9URKFaxes = Pony Express
21:56.15Uther_PI also have setup where I can send to an email address, with a phone # as the subject, and it freakin faxes it out
21:56.33Uther_Phard copies?  plleehhh!
21:56.58*** join/#asterisk kio (n=kio@ool-4577adba.dyn.optonline.net)
21:57.05_Sam--thats fine and dandy...but what happens if oyu need to sign something and then fax it?
21:57.07badboyzUther_P: well freaking freak freaker!
21:57.11_Sam--and im not talking digital signatures
21:57.24badboyz_Sam--: hit print?
21:57.29_Sam--not with Uther's setup
21:57.37N9URKYou scan your signature and place it in the file and send it back
21:57.39_Sam--i guess you still hit print
21:57.40Uther_P_Sam--: *shrug* .. scan in a signature and impose it on the images :D
21:57.41_Sam--then you can it
21:57.44_Sam--er then you scan it
21:58.18_Sam--all in all, i like faxing!  i like hyla...i like faxing from the desktops...why all the negativity!
21:58.21Uther_P_Sam--: nooo too much trouble.. just scan a copy of your signature, and do a transparent paste on the received fax
21:58.33Uther_Pcause faxing is antiquated
21:58.47DrScripttJust because something may be antiquated does not mean that it is not viable.
21:59.06DrScripttFaxing has become the least common denominator that everyone expects to work.
21:59.12iCEBrkrUther_P: I need a solution like that.. Where I can email a PDF to an address and it'll fax it out
21:59.16Uther_Panalog is not ment for text!
21:59.20lunkDrScriptt: i bought my house totally on an abacus
21:59.27_Sam--iCEBrkr:  its not that using procmail /sendfax
21:59.34Uther_PiCEBrkr: efax
21:59.37DrScripttYou can count on their being a fax in a 3rd (or 4th) world country that you can count on being able to send a fax somewhere in it.
22:00.24N9URKI guess I could still send messages by a horse ridding courier.
22:00.31Uther_PiCEBrkr:  mine is a combination of efax (for receiving the fax), imagemagik for converting, ghostscript for making a pdf, and sendmail for emailing
22:00.41iCEBrkrYup yup
22:00.46iCEBrkrI just haven't had time to tinker with it
22:00.55N9URKWhy not use efax out?
22:00.55Uther_PiCEBrkr: I wrote fancy perl scripts to handle all that crap for me
22:00.56iCEBrkrPlus, I don't have an analog line anymore
22:00.59N9URKis that too colsty?
22:01.15Uther_PN9URK: efax out as opposed to what?
22:01.37N9URKyou have a maching tied to a POTS line now?
22:02.00Uther_Pdude... the business needs a fax line... we just don't need a fax *machine*
22:02.07leenuxg33kbadboyz: yay.. I have outgoing phone again.  telasip is working great
22:02.08badboyziCEBrkr: whats the best way to recompile a .c file that ive made changes to?
22:02.14badboyzleenuxg33k: greats!! :))
22:02.18badboyzleenuxg33k: was pretty simple eh?
22:02.22Uther_Pplus... the line doubles as a dial-up way into the network if the fibers go down
22:02.23leenuxg33kbadboyz: very!
22:02.27badboyzright on man
22:02.31Motherlesouvage: I think Cepstral killed that one...prolly were getting tons of requests :)
22:02.35*** join/#asterisk sterne (n=art@246-84.customer.cloud9.net)
22:02.36leenuxg33kbadboyz: I can't believe galaxyvoice is so screwed up
22:02.40N9URKgood reason to keep it then
22:02.49badboyzleenuxg33k: yet another voip provider to add to the hate list :(
22:03.01leenuxg33kunfortunately
22:03.38*** join/#asterisk slayer192 (n=chrisc@66.138.39.225)
22:04.38Uther_Pthe only reason I consiter using asterisk for faxes is because I've got a 50 DID pool... if I ever find the time to jack with it and make it work, then I won't need the fax mailing list... I can use the did's and give everyone their own fax #
22:05.21*** join/#asterisk brockj49464 (n=brockj49@63.87.56.159)
22:06.13Uther_Pheh, someone should make an image-analysing version of spam assasin for faxes
22:06.36eKo1fax spamming? never heard of it
22:06.37N9URKwhat about sending them to the Mechanical Turk at Amazon?
22:06.40Uther_PI'd make it my mission to bombard the calling numbers with every piece of spam I got
22:06.48[TK]D-FenderUther_P : pipe it throough an OCR, then through SpamAssassin <-
22:06.59Uther_PeKo1: it exists dude... this office gets craploads of it
22:07.09_Sam--us too
22:07.22Uther_Pthats why I made the fax gateway in the first place.. cause it was always running out of paper printing out these dumbass advertisments
22:07.22eKo1hehe, good thing i live in a third world country and nobody cares enough to spam people
22:07.28_Sam--"Disney Vacations for $49"
22:07.43Uther_PeKo1: at least noone there is spamming people *there* :P
22:08.00badboyzif i modify a single .c file, do i have to completely recompile * ?
22:08.10eKo1spamming here would cost too much money
22:08.25eKo1and the returns would me negative at best
22:08.36eKo1badboyz: well duh
22:08.45eKo1s/me/be
22:09.13Uther_PI got the funniest freakin fax spam once.... they were trying to sell us a list of fax numbers for spamming!  they gave one phone number to fax to if we were intersted, and another number to fax to if we wanted to be REMOVED from that list they were selling!  but get this.. the number to fax to in order to get removed from the list was a 900 number! haha
22:09.14badboyzhmm thought i read where you can recompile certain parts
22:09.47eKo1well, unless it is a module
22:09.48Uther_Pit was a fuckin piramid scheme where they won both ways!
22:09.52eKo1which is dynamically loaded
22:10.56*** join/#asterisk mtnbkr (n=mtnbkr@c-67-165-9-234.hsd1.ct.comcast.net)
22:11.22Uther_Poh here comes another freakin fax.... some company selling helical gears
22:11.32Uther_P*sigh*  i'm too disturbed now
22:11.36Uther_PI'm going home
22:11.47Uther_Pl8rzzzz humans
22:11.58badboyzholy overload man
22:11.59badboyzlol
22:12.24Uther_Pi'm reminded of a song....
22:12.42eKo1hehe
22:14.53PoWeRKiLLI can't get DIALSTATUS variable anymore one * 1.2.1 and it was working on 1.2.0 any idea ?
22:16.21PoWeRKiLLon a php agi script i get error Invalid or unknown command
22:16.51_Sam--im not sure if you said that earlier or someone else did...but i saw someone complaining about the DIALSTATUS in 1.2.1
22:16.53*** join/#asterisk Rav1974 (n=r@static-70-19-119-112.ny325.east.verizon.net)
22:17.14Rav1974hi guys, is there a default remote answer code?
22:17.29Rav1974AKA remote call pickup
22:17.34*** join/#asterisk mog_work (n=mogorman@gateway.digium.com)
22:18.16Rav1974ok found it finally its #8
22:23.02fileI don't make mistakes; I make unintentional improvisations.
22:23.38file"It's not cheating, it's being highly intelligent"
22:27.40*** join/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it)
22:28.09markithi :) I'm updating the italian voices translation for asterisk 1.2, I need some clarification about some new voice message
22:28.28markitvm-saveoper.gsm "press 1 to accept this recording, or continue to hold"
22:28.35markitwhat is the meaning?
22:28.51markitif I press 1 I accept the recording, but if I continue I...
22:29.17*** join/#asterisk bkw__ (n=brian@ppp-69-155-251-101.dsl.tulsok.swbell.net)
22:29.45eKo1no, if you don't press 1 you'll stay on hold
22:30.02markiteKo1: ah, ok, thanks
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22:43.35Asterisk_downundhi
22:44.28Asterisk_downundis anyone there :)
22:53.27RoyKshould've been in bed
22:53.49Asterisk_downundhah
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23:01.34iaxcallhi
23:02.03iaxcallI am new to this chat room
23:02.28iaxcalln e one wanna fill me in on the rules ?
23:03.08iaxcallhello!!!
23:03.10TheCopsbe intelligent
23:03.24iaxcallare u talking to me ?
23:03.57anthmrule 1: be arroagant at tout whatever you think is the best way(tm) for things that don't matter at all
23:04.13iaxcallhehe
23:04.19anthmrule 2: be a dick to every newbie
23:04.24tzangeranthm: shhhhhhhhhhhh stop giving away my business secrets
23:04.26iaxcallthank you
23:04.43iaxcallgood to see some sence of humor
23:05.12iaxcallany pros interested in helping me with some wiered issue
23:06.24iaxcallI have an issue with asterisk connecting to an E1 line and receiving the ${RDNIS} properly
23:07.02iaxcallis this too complicated for u all
23:07.21*** join/#asterisk outofjungle (n=outofjun@61.247.254.133)
23:08.16iaxcallwell then
23:08.29iaxcalllet me ask the question differently
23:08.35TheCopsiaxcall, RDNIS, what's the problem ?
23:08.58iaxcallMy carrier does not seem to be sending RDNIS
23:09.00iaxcallinstead
23:09.15iaxcallthey send the Redirecting number at the end of the facility
23:09.35TheCopsok
23:09.51iaxcall< Facility (len=40, codeset=0) [ 0x91, 0xa1, 0x23, 0x02, 0x01, 0xa2, 0x02, 0x01, 0x0f, '0', 0x1b, 0x02, 0x01, 0x01, 0x0a, 0x01, 0x01, 0xa1, 0x13, 0xa0, 0x11, 0xa1, 0x0f, 0x0a, 0x01, 0x02, 0x12, 0x0a, '0394434568' ]
23:10.09iaxcallthis is the debug that shows the number at the end of the facility
23:10.15iaxcallthe question is
23:10.23iaxcallhow do I get that to the CDR for billing
23:10.24iaxcall?
23:10.29tzangeriaxcall: write some code
23:10.36iaxcallhehe
23:10.46tzangerseriously
23:10.54iaxcallhas anyone haad this prob before ?
23:10.55tzangerlibpri will see that, obviously, so you need to get it into asterisk
23:11.07iaxcallyeah libpri does
23:11.10tzangerI haven't heard of it, thankfully Bell Canada sends RDNIS logically
23:11.38iaxcallI tried to get write some code
23:11.53iaxcallunfortunately it crashes at the end of the call
23:11.57TheCopstzanger, and they sell PRI very expensive :P
23:12.13iaxcallwell beleive it or not my prov i MCI
23:12.20TheCopsha!
23:12.27iaxcalland I pay a lot of money for it
23:12.36iaxcallover $500
23:12.38TheCopsin USA ?
23:12.39iaxcalla month
23:12.45iaxcallno Australia
23:12.49TheCopsok
23:13.04iaxcallseriously
23:13.22iaxcallanyone seen this before, or heard of a patch ?
23:14.34tzangerTheCops: well I'm in rate group four anyway
23:14.34iaxcallI have managed to write a pacth for libpri that will pass the RDNIS as the caller ID Name variable to asterisk
23:14.42tzangerthat's the official "bend me over and don't use the lube" rate group
23:14.58tzangerhowever no other carrier will even return my calls
23:15.00TheCopsiaxcall, I can't help you, but why are you using RDNIS ?
23:15.19TheCopstzanger, how much you pay by month ?
23:15.37iaxcallwell, we are testing asterisk as a Long distance Pre-Select voice switch
23:15.58tzangerTheCops: trying to recall the exact number but I think close to 800/mo IIRC
23:16.06TheCopsouch
23:16.16tzangerthere are *no* breaks...  business likes are like $55/line and this is that + Dchan + loop charge
23:16.29iaxcallthe problem we have is when a person diverts their home phone to a long distance number, we need the redirecting number to know who to bill
23:16.30tzangerI only have 15bchan too
23:16.46TheCopsI'll probably pay 300$/month via Videotron telecom
23:16.51tzangeryeah
23:16.54TheCops+ 2$ each DID
23:16.54tzangerI'd love to pay that
23:17.01tzangeroh yeah 30 dids included
23:17.54iaxcallanyone got that ?
23:18.43iaxcallhere is the deal
23:19.17iaxcallI am willing at this point to pay a reasonable price for a patch to fix this issue for me
23:19.51TheCopsahhh the money
23:19.52TheCops;)
23:20.12iaxcallwell this is what makes this planet revolve unfortunately
23:20.22TheCopshehe
23:20.28TheCopsI know i have a business
23:20.34De_Moniaxcall yeah i see people crying all the way to the bank
23:20.39De_Monevery day
23:20.44iaxcallhehe
23:20.46TheCopslol
23:20.52iaxcallthis is waay we try
23:21.00iaxcallwhy :(
23:21.43iaxcallany C gurus that are willing to lift their hand ?
23:22.07De_Monthere has to be a better way to post this bounty
23:22.18De_Monmaybe the voip wiki or asterisk mailing list
23:22.36iaxcallthere is nearly 300 people here
23:22.36De_Monbut i'm sure you want a quicker response...
23:22.50De_Monyeah, and about 15 of them are active
23:22.52iaxcallthat's it :)
23:23.00TheCopsDe_Mon lol
23:23.25De_Moniaxcall a reply within the next 3 hours would be impressive!
23:23.53iaxcallI know where to point somebody to make it quick for them to write the patch
23:24.10iaxcallbut I have hit a block cause my C is a bit old
23:24.12iaxcall<PROTECTED>
23:25.12iaxcallAs I mentionned before, I have written a patch that will pass the RDNIS through the CALLERID NAME variable to asterisk
23:25.26TheCopsiaxcall, anyone who know C can do a patch ?
23:25.47iaxcallbut the problem is that it only works when I have "pri intense debug" on
23:26.38iaxcallcause I used the Facility dump function to save the last variable in the facility in a global variable which then is passed to the callerIDNAME if set
23:27.13iaxcallvery bad, should not use global variables
23:28.34seele_Which is the password for the webmail in aah??
23:28.37seele_Which is the password for the webmail in aah??
23:28.53seele_the default, obviously
23:29.57mrdigitalpassword
23:30.11mrdigitalAll passwords in AAH = password
23:30.48seele_but the login
23:30.49seele_!!
23:31.38seele_webmail login please?
23:34.51*** join/#asterisk Jzalae (n=sk@dsl-66-63-110-48.gwi.net)
23:35.44*** join/#asterisk _cleric_ (n=dacleric@p5482A46A.dip0.t-ipconnect.de)
23:36.16*** join/#asterisk sansna (n=cas@213-63-26-86.static.jdsl.net.artelecom.pt)
23:36.18distortionanyone use g729 w/voicemail?
23:36.41sansanhi all
23:37.17sansanhow can i start troubleshooting a kernel panic when loading the zaptel module?
23:37.25sansanusing centos 4.2 zaptel 1.2.1
23:37.36sansantried svn 1.2 and tarball 1.2.1
23:37.37distortionmy voicemail has issues when trying to write w/g729 on 1.2, here is the output if anyone has a min: http://pastebin.ca/35106 specifically lines 252-255.
23:39.38sansancan it be a problem that i have a zaph and a tdmp400p card
23:39.42sansanon the same system?
23:39.50tzangerwhat's a zaph card?
23:39.59sansancologne chip  i mean
23:40.27tzangerare they sharing interrupts?
23:40.30sansanzaphfc
23:40.32sansanno
23:42.58sansanthe panic > http://pastebin.ca/35109
23:43.38tzangeryou're getting a panic??
23:43.45sansanit just panics when i do modprobe zaptel
23:43.48sansanyeah
23:44.05tzangerwell
23:44.08tzangeryou're using 2.6.9 to start
23:44.12tzangerthat was a nasty kernel in general
23:44.14sansanyes
23:44.20sansancentos 4.2 here
23:44.23tzangerbut take that oops and run it through ksymoops
23:44.39*** join/#asterisk jcwunder (n=chris@a192.lrz.vpn.lrz-muenchen.de)
23:44.49tzangerso it's a nonstandard kenrel and very likely nonstandard zaptel drivers too
23:45.00tzangersorry man but I think you're the poster boy for "on your own"
23:45.12iaxcallanyone interested in writing some C code patch for libpri ?
23:45.24tzangerto do
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23:45.56iaxcallextract some variables and pass them to asterisk run_env
23:46.21tzangerwrite up a bounty and email it to me:akohlsmith@mixdown.ca I'll poke at it and see if it's something I'm interested in
23:47.25iaxcallI can even point you towards exactly what you want and all you need to get if done, been researching it for a week
23:47.45iaxcallbut my C is a bit rough though
23:47.54iaxcallpointers hater :)
23:48.07tzanger:-)
23:49.45sansanhummm, no ksymoops for this kernel
23:50.03sansanthought tht zaptel would run ok, on this popular distro
23:50.52SkramXctooley around
23:51.18*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
23:51.26sansanBTW, will the tdm400p wotk for a fax machine?
23:51.44sansani have a multifunction HP 6210 printer
23:51.58sansanwotk > work
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23:52.37sansanor can i just connect the fax on that printer to the tdm400p FXS module?
23:53.18tzangerit should work but honestly you won't know until you try
23:53.47malverian[work]Dec 28 18:31:30 WARNING[15386]: chan_sip.c:1314 __sip_autodestruct: Autodestruct on call '\uffff\uffffSIPCALLID' with owner in place
23:53.57malverian[work]Segfault :-P
23:54.24malverian[work]I love when asterisk crashes .. woo
23:55.21tzanger:-)
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