irclog2html for #asterisk on 20050611

00:00.14xeet2rocketraid has alot of linux issues, they use the system cpu for some things, kind of like a winmodem did
00:00.16Juxtno, i don't touch anything made by highpoint
00:00.27xeet2good.
00:00.32X-RobI've never had a problem with highpoint stuff
00:00.40xeet2x-rob: give it time
00:00.41Juxti had better luck with software raid under linux
00:00.42X-RobI've got a few sites running their 8 port SATA cards without problems.
00:00.46xeet2its like buying a kia
00:01.11X-RobAdmittedly, I'm just using the card to provide ports, it's not doing any RAID stuff itself.
00:01.17xeet2ah
00:01.25xeet2yeah then it should work without any major issues
00:01.29xeet2just don't turn on raid
00:01.37X-RobI hate doing that - if the card dies you're fucked unless you can find exactly the same card.
00:01.38xeet2or if you do, use software raid =)
00:01.42Juxtactually my card is made by maxtor
00:01.50Juxtbut i don't think the chipset is maxtor
00:02.00Slupdrumkilla mmm ok i'm talking to someone who says that if an external address is recieved by * it just drops it on the floor and to talk to users on an external provider you need to set it up in the dial plan
00:02.01X-Robuse mdadm and it doesn't matter what you plug the drives into.
00:02.09xeet2right
00:02.28Juxti have a box that runs 2 raid1 arrays using mdadm
00:02.34Juxtso far it's been freaking fantastic
00:02.39xeet2slup: sip?  you need to set up localnet, etc...
00:02.44jackxSlup: http://slacker.com/~nugget/asterisk7.php
00:02.46Juxti am seriously considering not buying raid cards anymore
00:02.52Slupdrumkilla but i was saying to that if you did that the callee would be presented with the id of the account you used in the dial plan
00:03.07xeet2they work really well =)
00:03.22Juxtyep, they are also quite pricey
00:03.31xeet2well, yeah, you get what you pay for
00:03.36Juxti concur
00:03.57*** join/#asterisk telephoneman (~chatzilla@67-41-211-223.brbn.qwest.net)
00:04.15xeet2combining them with western digital raptor drives, its quite speedy
00:04.33Juxtso i have this problem where my box has to transcode some calls and in peak hours it gets pegged out
00:04.50xeet2juxt: between what codecs?
00:04.59Juxtulaw and ilbc
00:05.10Juxtit's a call center
00:05.14Juxtso lots of calls
00:05.28jackx...
00:05.40jackxwhy you need conversion for?
00:05.54Juxti use ilbc for trunking
00:05.55jackxi assume all the agents in the call center are calling to "normal" extension
00:06.20Juxtwell all internal calls are ulaw
00:06.24jackxyou connected it to an other pbx?
00:06.34Juxtyes
00:06.40xeet2juxt: how many concurrent calls are we talking about?
00:06.47jackxtrought voip?
00:06.48Juxt150-200
00:06.59jackxbig center :)
00:07.00xeet2and how much bandwidth is available?
00:07.05jackxwe have only 60 seats
00:07.11Juxta T1 and 2 DSL
00:07.27Juxti put about 110 calls thru a data T1 using ilbc
00:07.39xeet2how are you load balancing between the t1 and the dsl?
00:07.44jackxT1 is like 1.5Mbit up/down fd right?
00:08.39xeet2juxt: and what kind of box is it?
00:09.13Juxtxeet2: i use a linux based load balancer
00:09.23Juxtthe box is a dual P3
00:09.31xeet2juxt: how many boxes?
00:09.38Juxt1
00:09.46jackxrespect
00:09.52*** join/#asterisk afaict (~afaict@201.6.255.86)
00:10.02jackxare all the phones IP phones?
00:10.08xeet2yeah, I think you'd need a bigger box than that to handle 200 calls
00:10.12xeet2200 calls of ilbc
00:10.13afaicthi folks!
00:10.26xeet2I would also be quite concerned about your bandwidth
00:10.37Juxtbandwith is cheap
00:10.40xeet2you need another t1 regardless
00:10.45Juxtyeah
00:10.55Juxtactually we're moving to a new location that has metro-ethernet
00:10.55afaictI would like to know. I have a x.25 line connected in my FXO digium card. Can I call more then once if I have multiple X.25 channels right ?
00:10.56shido.
00:11.03jackxJuxt: what company terminates your calls?
00:11.09xeet2yes you could use just under the maximum but when you start having jitter issues your circuit will be useless
00:11.18shidoits HOT
00:11.23shidoI need a new A/C
00:11.32Juxtjackx: i currently use 4 carriers
00:11.39xeet2shido: in mi?
00:11.48shidoyes in michigan
00:11.50Juxthave least cost dial too
00:11.51Juxt:-)
00:11.57xeet2I thought it was always cold out there =)
00:11.58jackxJuxt: can you name some?
00:12.02shidowe get everything here
00:12.07shidohail in summer
00:12.11shidoand summer in winter
00:12.15shidoand contruction
00:12.19jackxother question: What do you pay for the connection and the termination/
00:12.20xeet2earthquakes?  landslides?
00:12.23shidono
00:12.28xeet2how about tsunamis?
00:12.28JuxtRNK, experient
00:12.29xeet2=P
00:12.30shido1 little tremor every 10 yrs
00:12.31jackxexcl. the callkosts
00:12.43harryvvshido: want some of our cool BC air?
00:12.44harryvv:)
00:12.48shidolandslides? not unless they're man made
00:12.51shidoBC?
00:12.52shidoerr
00:12.55harryvvvancouver
00:13.01shidoI know something else you can send me from BC
00:13.13shidoxpress post
00:13.18harryvvwhats that?
00:13.30afaicthi folks.
00:13.31shidoxmas colors grown natural around there
00:13.47harryvvno idea what that is
00:13.47harryvv:)
00:13.52shidoI bet you dont
00:13.55xeet2hehe
00:14.03Juxtjackx: level3 in 2 months :-)
00:14.04shidonow pass the zigzags
00:14.24afaictif I have a X.25 line with 4 channels connected in a FXO Digium card. Will I able to do 4 calls at the same time with asterisk
00:14.32harryvvJuxt what relation do you have with L3?
00:14.37xeet2juxt: you should look at getting a bigger box or two, more bandwidth, and using g726
00:14.40shidome?
00:14.47MiccOk, I'm running stable and I've reinstalled on linux, ztdummy seems to be fine, but I've still got a bit of a delay in meetme. hmmm I should try the echo test.
00:14.54Juxtharryvv: can't disclose that yet
00:15.09Juxtg726?
00:15.14harryvvare you a customer or reseller of there service?
00:15.25Juxtgoing thru a reseller
00:15.28shidoI can jump rope with level3 connections
00:15.30harryvvi see
00:15.43Juxti can't meet their monthy commit yet
00:15.45Juxtbut soon
00:16.00Juxtwhy would i use g729 over say g729 or ilbc?
00:16.08shidoJuxt, drop me a line at greg@nufone.net
00:16.08xeet2g726, in voice conversations, sounds exactly like ulaw...  and its not bad on the cpu
00:16.31*** join/#asterisk newmedian (~crowlther@Quebec-HSE-ppp230300.qc.sympatico.ca)
00:16.33Juxtoh hey greg :-) i was looking for you at VON
00:16.50Juxttoo bad you weren't there, i was gonna throw a huge chunk of loot your way
00:16.57xeet2lol
00:17.03jackxJuxt: can you mean a company?
00:17.13jackxmean = name
00:17.13afaictI have the IAXy behind IPSEC network. I connected two offices with IPSEC. One of the offices has asterisk and another office has IAXy. It works fine right ?
00:17.25Juxtjackx: name of my company?
00:17.32Juxtor reseller?
00:17.34jackxJuxt: no. name of a reseller
00:17.34xeet2afaict: yes, barring any network issues
00:17.40Juxtno i can't just yet
00:17.43Juxtsorry
00:17.45shidowas... cuz...
00:17.55jackxI'm in holland, no competition here ;)
00:17.57afaictxeet2, thanks.
00:18.06Juxti have an NDA
00:18.23jackxnone disclosure Agreement?
00:18.23afaictand for that question do you have the anwser? if I have a X.25 line with 4 channels connected in a FXO Digium card. Will I able to do 4 calls at the same time with asterisk
00:18.24harryvvjack..you lucky dog
00:18.25harryvv:)
00:18.38jackxi am not into business or something
00:19.01afaictxeet2 do you know something about ? X.25 lines and asterisk
00:19.11xeet2afaict: I didn't know zaptel could do anything x.25
00:19.58afaictxeet2, really ?
00:20.28xeet2really, I didn't know, don't know anything about that
00:20.37xeet2haven't touched x.25 in quite a while =)
00:21.13Juxtso what i was thinking is to put another box and a load balancing proxy in front of the 2
00:21.22Juxtthis way i could scale my rig
00:21.30X-RobX25 is 1970's technology. We've moved on since then.
00:21.37X-Robmoved on a -long- way.
00:22.10X-RobAltho, sadly enough, in australia you can dial 01924 and get a 2400bps connection to Austpac, Australias X25 network.
00:22.35X-Rob_still_
00:22.40xeet2haha
00:23.03afaictX-Rob, im from brasil ... and we use x.25 lines when we have multiple dial connections at the same time
00:23.13afaictX-Rob, :-/
00:23.14xeet2juxt: look at bigip load balancers
00:23.29*** join/#asterisk |Vulture| (~V@c-69-180-67-228.hsd1.fl.comcast.net)
00:23.32newmedianAren't a number of EFTPOS style terminals still using x.25?
00:24.05xeet27-11 still has a ton of x.25 stuff
00:24.12X-Robafaict, you poor thing.
00:24.22*** part/#asterisk outtolunc (outtolunc@adsl-69-110-5-162.dsl.pltn13.pacbell.net)
00:24.25X-Robxeet, and that's why it's still around
00:24.33afaictX-Rob, :D
00:24.39Juxtxeet - looking right now
00:24.44Juxtlooks magical :-)
00:24.55xeet2juxt: they are
00:24.58Micccan someone call my asterisk system and test my conference meetme. 1 (888) 206-4443 room #1234
00:25.02xeet2juxt: and white reliable if you can afford them
00:25.10X-Robeven though you can get IPSEC,TCP/IP and PPPoE tunnel on a single chip now, they still use X25.
00:25.15X-Robbecause it's too hard to change.
00:25.21X-Rob(which is why 01924 still works)
00:25.30xeet2x-rob: doex x25 run on that single chip too?
00:25.37X-Robprobably
00:25.38Juxti just don't see the advantage of that over SER, except i can keep IAX
00:25.49X-RobLinux supports X25, so it should.
00:26.15X-RobThe difference is, IPSEC is actually secure, whilst X25 can be intercepted anywhere inside the telco.
00:26.34xeet2juxt: there's alot more you can do with a bigip with sip than what ser can do
00:27.18harryvvX-Rob that VERY important to govermental agencies.
00:27.32Juxtxeet2: that's probably true, alas i just want to accomplish only 1 thing :-)
00:27.40harryvvhere in canada the privacy laws are very strict...almost to strict.
00:27.41shidono
00:27.46shidohello
00:27.51shidoyes hello
00:27.52xeet2hello
00:28.04shidoare you still having echo problems or delay problems with this thing
00:28.10shidochoppy?
00:28.13shidohrmmm
00:28.25shidook let me change some things...
00:29.10harryvvbtw does anyone know of any voip testing software that can simulate say 100,1000 or more calls on my server?
00:29.19shidoshow audio codecs
00:29.33*** join/#asterisk file[desk] (~jcolp@mctn1-4092.nb.aliant.net)
00:29.40shidoI sound bad?
00:29.41shidochoppy?
00:29.49shidolet me try a differnet phone
00:30.43Miccmaybe I have firewall issues.
00:30.47shidohrmm
00:30.53shidonaah i think its me and limewire
00:31.08shidojust cuz you kill the limewire interface doesnt mean java is dead
00:31.16shidolet me kill limewire and call you back
00:31.37*** join/#asterisk Umaro (~umaro@209.140.74.64)
00:31.37MiccIt still seems that there is a bit of a delay. But when I try the echo test it isn't too bad.
00:31.38shidomic chec
00:32.52UmaroHey guys.. I currently have a area code 971 DID from broadvoice, but i'm finally frustrated to the point where i'm looking for someone else to provide it. Does anyone have any suggestions?
00:33.10shidoodd
00:33.15*** join/#asterisk IQ (~IQ@70-59-165-178.omah.qwest.net)
00:33.34IQhi all
00:34.33xeet2umaro: where is 971?
00:34.46afaictX-Rob, im trying to figure out how it will works. I mean guess that I have in one office 5 phones connected in a single IAXy. What IAXy will do if I have 2 phones calling at the same time ?
00:35.09harryvvumaro, first time here?
00:35.17Umaroxeet2: portland, oregon
00:35.20Umaroharryvv: no, why?
00:35.37X-RobI don't know anythign about the IAXy's, cept they have a bad habit of catching on fire.
00:35.46xeet2mmm, wish I could but I can't port oregon =)
00:35.46X-RobYou can't buy 'em in australia, so I've never seen on.
00:35.47harryvvokay there have been complaints about broadvoice discussed here ;) whats the problem waiting to buy one?
00:36.01afaictX-Rob, why u can not but it in australia ?
00:36.11harryvvportland oregon my original home ;)
00:36.13X-Robnot approved.
00:36.21Umaroharryvv: do they include people saying their calls sound like there's a helicopter in the background?
00:36.31afaictX-Rob, hows that ? " they have a bad habit of catching on fire."
00:36.32harryvvUramo, thats jitter
00:36.34afaictX-Rob, they burn ?
00:36.36afaict:D
00:36.46afaictsorry.. it burns ?
00:36.54X-RobWell
00:36.55harryvvThay have a jitter problem
00:37.00X-Robnot actually catching on fire
00:37.09X-Robbut they get extremely hot and I've heard of a couple of the cases melting.
00:37.27afaictX-Rob, nice. :D
00:37.28X-RobThis is only a rumour, by the way. Totally unsubstanciated.
00:37.35afaictok.
00:37.39X-RobIf you want IAX, get some PA1688 handsets.
00:37.44X-Robthey're as cheap as anything.
00:37.46harryvvI have only experainced jitter once on my asterisk install. that was 6 months ago. I dont know enough about it but is it related to bandwith problems?
00:38.31ptiggerdineharryvv, use mrtg to find out
00:38.32*** join/#asterisk SwK (~ken@12-219-156-206.client.mchsi.com)
00:38.35X-Robharry - jitter can be related to _lots_ of things, but yes, lack of bandwidth is one of the things that will cause jitter
00:38.41X-RobGah
00:38.43*** join/#asterisk Strom_C (~strom@66.159.243.60)
00:38.47X-RobI have this dual xeon to build
00:38.57xeet2bandwidth probelms, out of order packets, corrupted packets
00:39.04Umaroharryvv: I don't think it's a bandwidth problem.. T1 in las vegas, using their LAX proxy server
00:39.09UmaroT1 is at like 25% max
00:39.26xeet2umaro: what about the other end?
00:39.49Umaroxeet2: what do you mean?
00:39.57Umaroxeet2: I have the problem with those two end points
00:40.15*** join/#asterisk Eduardo_Henrique (~ehst@201.19.9.10)
00:40.36xeet2this is the problem with voip over the internet
00:40.42xeet2so many variables
00:40.47harryvvX-ROB um running asterisk on my opteron system works fine.
00:40.50Eduardo_HenriqueExist a IAX ActiveX for .NET plataform ?
00:41.12xeet2eduardo: blasphemy
00:41.20Umaromaybe i'll just forward my broadvoice DID
00:41.34xeet2umaro: oh its to broadvoice?
00:42.03xeet2broadvoice has had alot of quality issues lately
00:42.12Eduardo_Henriquexeet2: link ?
00:42.15Eduardo_Henrique=P
00:44.26Miccanyone know what might cause a delay of over 1 second with meetme? I"m using ztdummy
00:44.39Miccand I'm using the stable tree.
00:46.12X-RobHarry, yeah, I know asterisk works extremely well.
00:46.16X-RobI'm tossing up what to do with it.
00:46.34X-RobI've just put Fedora Core 4pre3 on it.
00:47.18*** join/#asterisk drooth (~drooth@ip68-107-113-76.sd.sd.cox.net)
00:47.33droothdoes anyone know how to change the hostname for CentOS
00:47.35X-Robdo I a: Go with CVS HEAD, or b: stick with 1.0.7.. They're not going to be running any TDM hardware in it, so I probably want to go with CVS's ztdummy. Gah. Decisions.
00:47.42afaictX-Rob, what internet rate connection to use with asterisk office-internet-office ?
00:47.58X-Robdrooth - edit the file /etc/sysconfig/network
00:47.59afaictX-Rob, 128k up/downlink link is fine ?
00:48.08xeet2afaict: how many concurrent calls?
00:48.16X-Robxeet2 beat me 8)
00:48.17afaictxeet2, 2
00:48.19harryvvX-ROB thats what I am running is FC3 x86_64 but if you want to run it as a work station I would avoid it. There are still alot of binaries and source code that have alot of missing files that problebly work on a i386 system
00:48.28droothx-rob: you are the man! htanks
00:48.29drooththanks
00:48.41xeet2128k what, dsl?  and is that dsl pppoe?
00:48.52afaictxeet2, yes. dsl and ipsec
00:48.55Juxthas anyone done any benchmarking with the dual-core opterons?
00:49.03X-Robharry, this is a 32bit xeon machine, not the new funky (expensive) 64bit xeons.
00:49.19afaictX-Rob, 64bit xeons... ;~
00:49.23xeet2afaict: you'd be lucky to get 1 ulaw call out of that then.  go with g729, ilbc, gsm, or speex, but yeah you could get 2 calls
00:49.30X-RobI'm gunna build up a nice athlon64 as the gateway box shortly.
00:49.44X-Robafaict, rare as hens teeth, and twice as spensy.
00:49.57afaictX-Rob, :D
00:49.58X-Rob128k shoudl get two ilbc calls.
00:50.18droothx-rob: uname -a still shows the old name, do i need to reset server?
00:50.21Juxteeh 128k should do more than that
00:50.33X-Robdrooth, if you want to change it -now- then type 'hostname new.host.name'
00:50.36Miccis 99.95% accuracy with the zttest good?
00:50.50X-Robmicc, 99.98% is what digium say you should have
00:50.57xeet2juxt: for almost anything but ulaw yeah you can get more than 2 hopefully
00:51.03afaictX-Rob, Juxt thanks.
00:51.13MiccX-Rob, I've got 99.95 average with ztdummy
00:51.24X-Robmicc, are you using linux 2.4 or 2.6?
00:51.31Juxthttp://lists.digium.com/pipermail/asterisk-users/2003-June/014788.html
00:51.44MiccX-Rob: 2.6
00:52.01harryvvX-Rob my opteron works smoothly. for a long time i did not know if the orginal motherboard.cpu ram was bad when it was new..i replaced it with a warentee board and booted right up.
00:52.19*** join/#asterisk xcore (~rmaraschi@200.180.111.2)
00:52.19harryvvmsi neo far 2 had a bad back of boards.
00:52.25harryvvbach
00:52.34*** part/#asterisk Eduardo_Henrique (~ehst@201.19.9.10)
00:53.00file[desk]batch
00:53.12X-Rob<PROTECTED>
00:53.19X-Robmicc ^^ do that
00:53.24X-Robthat'll get the latest CVS of asterisk
00:53.38X-Robyou want the ztdummy from there
00:56.55*** part/#asterisk Juxt (~Juxt@sfl-dsl-64-135-113-4-cust.host.net)
00:58.00MiccX-Rob, is this going to build and install everything?
00:58.08X-Robyeah
00:58.26X-RobThen at line 48
00:58.27Miccit dies on No description for sounds/1-yes-2-no.gsm
00:58.37X-RobWierd
00:58.43X-RobI just did exactly that on my FC4 machine
00:58.52X-Robant it worked flawlessly
00:58.59MiccDo I need to reboot or restart anything before I try it again?
00:59.02X-Robnope
00:59.09MiccI'm using FC3
00:59.18X-Robthis is FC4pre3 I think
00:59.54X-Robanyway. Line 48 of /usr/src/zaptel/ztdummy.c change '#if 0' to '#if 1'
01:01.18xeet2what does that do?
01:01.29X-Robenables the 2.6 RTC support for ztdummy
01:01.39xeet2oh cool
01:01.39X-Robdoesn't use USB, uses POSIX timers, smegloads more accurate
01:01.54xeet2is that documented anywhere?
01:02.00X-RobNot really.
01:02.08X-Robhttp://aussievoip.com.au/wiki-1.10.008-Install-MDK10.1
01:02.15X-RobThat's the mandrake 10.1 docco
01:02.29X-RobI'm going to copy most of that to fedora core, coz it's very similar.
01:07.37MiccMy line ztdummy.c line 48 is #else
01:08.03X-Robaround there
01:08.05X-Robis #if 0
01:09.45Miccit is #ifdef LINUX26
01:10.51*** join/#asterisk telephoneman (~chatzilla@67-41-211-223.brbn.qwest.net)
01:11.11X-RobI don't know what you've downloaded
01:11.48X-Rob. /usr/src/zaptel/ztdummy.c
01:11.55*** part/#asterisk Moc (~Moc@modemcable165.109-70-69.mc.videotron.ca)
01:14.32X-RobHrm.
01:14.41X-RobIt's sucking.
01:15.07X-Rob--- Results after 4 passes ---
01:15.07X-RobBest: 99.975586 -- Worst: 99.938965 -- Average: 99.960327
01:15.09MiccX-Rob, I did that thing you said, so I should have that version.
01:16.36X-Robline 69 has LINUX26
01:17.40X-Robthis is the first time I've tried this
01:17.59Miccyeah I am using hyperthreading too.
01:18.21h3xFC3 dosent work too well with zaptel
01:18.47X-RobThis is strange
01:19.32X-Rob--- Results after 9 passes ---
01:19.32X-RobBest: 99.987793 -- Worst: 99.987793 -- Average: 99.987793
01:19.37X-Robthat's _without_ use rtc
01:19.37Micch3x, does anything work well with zaptel?
01:20.25Strom_Czaptel and debian seem to play nicely for me
01:20.51Hmmhesaysdebian plays nice with lots
01:21.56MiccX-Rob, what is rtc?
01:22.00X-Robreal time clock
01:22.14*** join/#asterisk jpablo (~jpablo@200.94.130.194)
01:22.26X-Rob99.987793 is optimal
01:22.29Miccis rtc supposed to be better?
01:22.36ptiggerdineyes
01:22.36X-Robyeah, rtc is supposed to be perfect
01:23.25MiccI'm getting 99.95 with latest source. what line do I need to change in ztdummy.c?
01:23.58jpablohi, anyone knows how to solve this: I'm not getting ring tone when doing a blind transfer with a spa-2000 in asterisk.
01:24.06MiccAverage: 99.832266
01:24.10jpablothe blind transfer is done using 'flash'.
01:24.28X-RobMicc, ok. You should change the line near the top that says '#if 0' and underneath that says '#define USE_RTC'
01:24.35X-Robchange the '#if 0' to '#if 1'
01:27.14Miccok, I found that.
01:27.27MiccI must not have had the right cvs version before.
01:27.53X-RobOK
01:28.03X-RobI am goign to have to turn off hyperthreading
01:28.08X-Robbecuase with RTC, it's all over the place
01:28.10X-Roband it shouldn't be.
01:28.24X-Rob--- Results after 8 passes ---
01:28.24X-RobBest: 99.963379 -- Worst: 99.938965 -- Average: 99.951172
01:28.30MiccI think I need to build a non smp kernel.
01:28.46MiccBest: 99.963379 -- Worst: 96.923828 -- Average: 99.699910
01:28.52Micclook at that worst. 96
01:29.16*** join/#asterisk kimo_sabe (nick@zappa.azrackspace.net)
01:29.56X-RobI'm turning off HT now
01:30.04X-Robwill try with a non-smp kernel next
01:30.09X-Rob(This is a dual xeon 3.0 machine)
01:30.18*** join/#asterisk Cheng29 (~cheng29@d57-87-253.home.cgocable.net)
01:31.10MiccX-Rob, let me know how that works.
01:31.20MiccI've got a dual xeon 2.4
01:31.21X-Robrebooting... rebooting.
01:31.30X-Robproblem with server boards is that they take _ages_ to boot
01:33.09telephonemananyone interested in a MB that works with spandsp?
01:35.03PatrickDKserver boards don't take ages
01:35.09PatrickDKecc memory takes along time to check
01:35.21X-RobOK. It's _better_ but it's still not right.
01:35.23PatrickDKmy server takes 2.2min to boot, from start to freebsd running
01:35.42PatrickDK4gig ecc memory in it
01:35.45X-RobIt's alternating between 8192 in 8195 and 8187
01:35.48kimo_sabePatrickDK: most do, even without ECC
01:36.05X-RobYou try probing 4 SCSI cards quickly
01:36.07X-Robit doesn't happen
01:36.26PatrickDKya, expectually if you have 2 4port scsi cards :)
01:36.41X-RobHrm.
01:36.45PatrickDKnext upgrade it gets fibre
01:36.45X-RobMicc, I dunno.
01:37.04*** join/#asterisk faqu (pn@OL187-166.fibertel.com.ar)
01:37.09MiccX-Rob, I'm gonna use non smp kernel
01:37.17SwKtry 2 scsi cards, RAID Controller, 2 FC hbAs, and remote management card
01:37.26faqui've a problem i've an ata 186 connected to my switch and i can't set the ip
01:37.39faqui really tried to do it manually and nothing
01:37.42SwKfaqu: hard set the duplex
01:37.44SwKand the speed
01:38.01faqusorry?
01:38.24SwKon the ethernet port on the switch and the ata186 turn off aut speed/duplex negotiation and hard set it
01:38.53faquyep the  lnk is active but the 100 isn't
01:38.56faquhow may fix it?
01:39.07SwKdunno
01:39.28SwKare you sure the ata is still functioning?
01:39.35faquwhy not?
01:39.44faqueverything looks okey with it
01:39.50SwKcant get dhcp or set an ip address manually...
01:40.03SwKcan you get the IVR on the phone attached to it?
01:40.08Slupjackx thanks for that link
01:40.14faquhow can i check it?
01:40.31SwKpre the button on top of it (assuming you are talking abou a cisco ata186)
01:40.39faquyes
01:40.42faquthe function button
01:40.44faqured one
01:40.50SwKyeah
01:40.55faquthen?
01:41.02SwKlisten to the phone
01:41.07faquyep its working ok
01:41.10SwKgot to cisco.com and get the manual for it
01:41.10shido:)
01:41.14SwKtheres a tone of info there
01:41.19*** part/#asterisk faqu (pn@OL187-166.fibertel.com.ar)
01:41.25X-Robah crap.
01:41.27SwKhah
01:41.35X-Rob[root@dhcp5 zaptel]# make
01:41.35X-RobYou do not appear to have the sources for the 2.6.11-1.1286_FC4 kernel installed.
01:41.35X-Robmake: *** [linux26] Error 1
01:41.37X-RobSigh.
01:41.48SwKhah
01:41.53SwKi hate it when that happens
01:42.16SwKand for the record... FC sucks
01:42.26X-Robyeah
01:42.35X-RobI'm coming to that conclusion myself.
01:42.40SwKget ricer linux
01:43.30X-Rob'ricer'?
01:43.42shidoso install the kernel source, X-Rob
01:43.43*** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv)
01:43.57X-RobI dunno what you yanks call a ricer, but there's no way I'd want a customer of mine knowing that I installed 'ricer' linux.
01:44.03X-Robshido, yeah
01:44.04X-Robdoing that
01:44.08YoWoudilolol
01:44.22Sedoroxlol
01:44.27MiccX-Rob, make a symlink linux-2.6 -> /lib/modules/2.6.9-1.667smp/build/
01:45.43SwKX-Rob its gentoo
01:46.21SwKricer linux being a reference to www.funroll-loops.com
01:46.22ptiggerdinei thought ricer was LFS
01:46.37SwKerrr .org
01:47.00Sedoroxlol
01:47.04Sedoroxno.. ricer is Gentoo
01:47.42*** part/#asterisk makhtar (~ageller@mail.bulletinnews.com)
01:47.46X-Robgentoo's a pain in the arse.
01:47.56X-Robcompiling everything everywhere
01:48.10SwKthats only cause you think you have to do that
01:48.25X-Rob*shrug*
01:48.30X-RobI've become lazy in my old age.
01:48.36SwKits like ports on bsd...
01:48.55SwKbuild it if you want (or have a fast machine on slow link) else grab the binary packages
01:49.57ptiggerdinewhat and rpm's better
01:50.30SwKrpms are a pain in the ass
01:50.47SwKatleast emerge will install dependancies for you
01:51.43newland apt or urpmi won't? ;)
01:52.43X-RobWell
01:52.45X-Robfuck this
01:52.57X-RobFedora can blow me.
01:53.00bjohnsonwould adding externip in the general section of sip.conf be responsible for screwing up my lan voip calls?  ie no audio
01:53.47MiccX-Rob, my FC3 came with two kernels, I just switched to the on smp version.
01:53.53X-RobYep
01:53.55X-RobSo did I
01:54.03X-Robbut the kernel source they've supplied is all over the place.
01:54.08shidoits simple... dont use fc for production
01:54.35X-RobMandrake 10.1 I find incredibly good
01:54.47X-Robbut the _name_ 'Mandriva' just sounds .... slimy.
01:55.12X-RobSo. FC is crap. I've got a ubuntu CD here
01:55.14X-Robmaybe I'll throw that on
01:55.21X-RobMicc, you got some time while I do a reinstall?
01:56.59*** join/#asterisk Astermigraine (~psolomon@69-165-217-96.atlsfl.adelphia.net)
01:57.05*** join/#asterisk sudhir492 (~sudhir@pool-71-114-74-131.washdc.dsl-w.verizon.net)
01:57.10sudhir492Hi all
01:57.49*** join/#asterisk mne (~mne@chello080108001212.35.11.tuwien.teleweb.at)
01:58.10sudhir492For most of the callers, we get only the phone number in caller id. Is it possible to get names too ?
01:58.53AstermigraineIf I set my IP phone to use inband DTMF and notify as the type, it causes asterisk to abend. While I can simlpy change it, wouldnt a bug like that possb lead to a possible DOS?
01:59.17MiccX-Rob, I'll be here most of the night.
01:59.28Astermigrainesudhir492 depens what your provider is delivering.
01:59.43mnehi there ! I'm completely new to asterisk. I looked through some available asterisk documentation, but for now it is far to complex for me. Can you suggest anything ? I want to set up a SIP server so that the registered SIP users can talk to each other
01:59.49MiccX-Rob, it looks I'm getting better results with non smp kernel I think, but they are still not the best.
02:00.55sudhir492Astermigraine: does it depend on the telco from which I have PRI, or on the caller's provider?
02:01.38AstermigraineCould be both
02:01.58AstermigraineDepends on what they will pass
02:02.51AstermigraineThe Telco can charge for Caller ID and also passing the ANI's associated account name
02:02.58AstermigraineWith reguard to a SIP client
02:03.01X-Robmne - best thing for you to do is download asterisk@home
02:03.08AstermigraineThats going to be different
02:03.41AstermigraineDepending upon who they are going through, they may block client configured CID
02:03.49mneX-Rob, Astermigraine already suggested it to me, I'm already downloading it ;) thx
02:03.52Astermigraineand pass account based CID
02:05.42sudhir492thx. I will check with my telco on Monday
02:07.12*** join/#asterisk anti (~russ@anti.developer.gentoo)
02:10.28*** join/#asterisk MikeJ_ (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net)
02:11.13*** join/#asterisk MikeJ[Laptop] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net)
02:12.09*** join/#asterisk MikeJ[Jayden] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net)
02:16.40*** join/#asterisk midnight- (midnight@after.12am.net)
02:17.23*** join/#asterisk zeus232 (~zoose@203.49.209.210)
02:17.45X-RobGrr
02:17.52X-RobOK, now no distribution will detect the CDROM
02:17.56X-Rob*sigh*
02:18.32shido?
02:18.42X-RobStupid new server.
02:19.16*** part/#asterisk mne (~mne@chello080108001212.35.11.tuwien.teleweb.at)
02:19.41shidowhat server?
02:20.34X-RobIt's one of the new intel boards with onboard SATA
02:21.09MiccX-Rob, supermicro?
02:21.15X-Robno, intel intel 8)
02:21.32shidono drivers?
02:21.40shidomodprobe is your friend
02:21.44shidoinsmod is too
02:21.49Sedoroxmodprobe > insmod
02:21.56X-RobYeah, it's more a lack of care
02:22.00Sedoroxlol
02:22.08Sedoroxya gotta care man.... ya gotta care...
02:22.18X-RobIf I cared, I'd figure out why it can't detect a stadard smegging IDE DVD burner
02:22.43shidomolex plugged in? :)
02:23.01Sedoroxlol
02:23.10X-RobIt _boots_ off the damn CD
02:23.18shidoerrr
02:23.20X-Robbut the installers are going 'no CD found'
02:23.20shidok
02:23.32Sedoroxlol
02:23.33shidodo you have to remount the new cd?
02:23.34X-RobFedora loaded an 'ata_piix' modeule
02:23.35X-Robmodule
02:23.39X-Robwhich I'm guessing needs to be done
02:23.47X-Robbut, again, lack of care.
02:23.51shidoyou picked fedora cuz?
02:24.01X-Robbecause it was the handiest DVD
02:24.16X-RobI should download SUSE
02:24.30Sedoroxfedora... suck...
02:24.52X-Robhttp://www.intel.com/design/servers/boards/se7520af2/index.htm
02:24.55X-RobThat's the board
02:25.02*** join/#asterisk rcw (~rcw@206-124-146-40.blarg.net)
02:25.04MiccX-Rob, let me know what gets your accuracy up. I'm going to have to install another version of linux too.
02:26.33X-RobI probably should do this sensibly and just upgrade the kernel
02:26.55X-RobI'll download the latest 2.6 and see how I go
02:29.31*** join/#asterisk nemisus (~nemisus@203-217-85-104.dyn.iinet.net.au)
02:29.37*** join/#asterisk wfu (~wfu@203.131.175.66)
02:31.15Umarowhat's the best way to do callerid lookups? I've got an 800 DID, so I don't get callerid name, but i'd really like to.
02:32.08MiccOk, this might be a bit better.
02:32.15MiccShido, you want to give it a try again?
02:32.51Micc(888) 206-4443 room# 1234
02:33.51shidonon smp did the trick
02:33.52*** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com)
02:33.52*** mode/#asterisk [+o anthm] by ChanServ
02:33.52shidook
02:33.57kimo_sabeUmaro: I search our existing LDAP addressbook
02:34.55nemisusis there a tar.gz of 10.0.8rc?
02:34.59Umarokimo_sabe: hmm, smart
02:35.20Umarobrb
02:35.48kimo_sabeUmaro: that way we get meaningful caller id info instead of "US Government" for everybody
02:36.45*** join/#asterisk squirrelv5 (~squirrelv@202.57.81.146)
02:42.15*** join/#asterisk CoolAcid (~jk@216.99.98.39)
02:43.27wfuhi, can i ask anyone to send me samples of sip.conf/extensions.conf/voicemail.conf
02:49.13*** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net)
02:50.03*** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk)
02:50.03*** mode/#asterisk [+o bkw_] by ChanServ
02:51.58zeus232i have fresh fedora1 install and i can't get wcfxs driver to load, although modprobe zaptel works and appropriate output is show in /var/log/message ... error msg is:
02:51.59zeus232/lib/modules/2.4.22-1.2115.nptl/misc/wcfxs.o: init_module: No such device
02:52.00zeus232Hint: insmod errors can be caused by incorrect module parameters, including
02:52.00zeus232invalid IO or IRQ parameters.
02:52.06zeus232anyone got any ideas?
02:52.59SwKwhat fxs card are you using?
02:53.02SwKa freshmaker?
02:53.43bkw_haha
02:53.52bkw_go rm -rf all kernel drivers
02:53.52SwKbkw_
02:53.53bkw_and try again
02:53.58YoWoudiquestion
02:54.06SwKor just rm -rf / and try again
02:54.08zeus232umm it's digium tdm400P
02:54.14*** part/#asterisk wfu (~wfu@203.131.175.66)
02:54.16SwKmodprobe wctdm
02:54.20bkw_NO
02:54.22YoWoudiwhat does the F parameter do swk?
02:54.24bkw_you people aren't listening to me
02:54.30SwKhah
02:54.38SwKignore bkw he doesnt know shit ;)
02:54.38bkw_cd /lib/modules/2.4.22-1.2115.nptl/misc/
02:54.43bkw_rm -rf the modules
02:54.48SwKheh
02:54.49bkw_zaptel was changed for a week
02:54.53bkw_to install in a different location
02:54.55SwKYoWoudi what -F ?
02:55.03SwKyou men th e f in rf -rf?
02:55.03bkw_even if you do modprobe wctdm it might still pickup wcfxs
02:55.04bkw_no go
02:55.10SwKerrr rm -rf
02:55.11h3xgod damn tivo-to-go transfer is slow
02:55.12YoWoudiPERFECT
02:55.13zeus232can't locate wctdm... should it be included in the zaptel
02:55.16YoWoudirm -rf is perfect!
02:55.19zeus232file?
02:55.22YoWoudiwhat does the rf stand for though?
02:55.23bkw_I give up
02:55.27bkw_if you don't know what rm -rf is
02:55.35bkw_YOUR TOTALLY LOST.. you'll need to come to cluecon
02:55.38zeus232r = recursive f = files ?
02:55.39YoWoudii was trying to go do -rm -r withou the F
02:55.42bkw_you can go signup at www.cluecon.com ;)
02:55.43YoWoudioh no wonder
02:55.44bkw_f ix force
02:55.45YoWoudilol
02:55.46SwKYoWoudi try rm -rf / and figure it out
02:55.47*** join/#asterisk Klar (~null@libertarians.STUDENT.CWRU.Edu)
02:55.48bkw_f is force
02:55.54YoWoudiyeah i wasnt sure what the F stood for
02:56.02bkw_-f = FUCKED
02:56.03bkw_haha
02:56.06YoWoudiit was trying to get me to delete one by one if i didnt use F
02:56.06SwKhah
02:56.08YoWoudilol
02:56.10YoWoudithanks
02:56.18YoWoudilol i havent done linux in ages, bear with me
02:56.20SwKrm -rf bkw_
02:57.30MiccI'm cool. I have 3 asterisk servers.
02:58.04YoWoudiare most of you guys recomending to use the stable release over the standard one?
02:58.27YoWoudiof asterisk that is
02:58.44MiccYoWoudi, I like to live on the edge.
02:58.55MiccIt gets me screwed sometimes.
02:58.57YoWoudiis there much difference?
02:59.01MiccBut it makes me cool.
02:59.04YoWoudilol
02:59.15MiccI get to hang with the cool crowd.
02:59.48MikeJ[Laptop]SwK, some things even -f won't work on ;)
03:00.02YoWoudii gotta change this name
03:00.10YoWoudilet me see if some other one is taken
03:00.14SwKunlink works on those tho
03:00.37SwKunlink /dev/zero give about 20% performance boost on solaris
03:02.24*** join/#asterisk pifiu (~myassisbi@208.205.181.170)
03:02.27pifiuok back
03:02.33pifiuyo woudi=pifiu now
03:04.16MiccPRI and single span card is on its way. yah!
03:08.33Micchmmm
03:10.04*** join/#asterisk Cheng29 (~cheng29@d57-87-253.home.cgocable.net)
03:11.27*** join/#asterisk mosty (mostynm@adsl-137-244.swiftdsl.com.au)
03:13.57*** topic/#asterisk by drumkilla -> Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || Cluecon -- PBX Developers Conference in Chicago Aug 3-5
03:16.39MiccI thought cluecon already happened?
03:19.05niZonyeah last year
03:21.12drumkillaum.
03:21.17drumkillayou're making stuff up :p
03:21.39anthmno!
03:22.06drumkillait's coming soon, folks
03:24.39*** join/#asterisk edy2 (anonymous@202.134.128.49)
03:28.43pifiui need to get an ethernet phone to test my implentation of asterisk into our office
03:28.53pifiuany phones you guys recommend in general that is good all around?
03:29.07pifiui would need to buy around 8 or so eventually
03:29.29pifiubut i would start with one to test then buy the rest if i am happy with it
03:29.40*** join/#asterisk dca (~dca@c-67-166-37-218.hsd1.co.comcast.net)
03:29.56mostyi'm playing with a linksys pap2 with a regular telephone attached, it's pretty simple to setup
03:29.58telephonemanwould a Java programmer get anything out of cluecon???
03:30.42telephonemansomeone not developing code for Asterisk?
03:31.59telephonemantest 1 2 3   test 1 2 3
03:32.32Syncrosclues
03:32.34Syncrosi guess
03:33.25X-Robpifiu - The GXP 2000 is pretty shithot at the moment
03:33.30X-Robbest bang-for-buck
03:33.38X-Robhttp://www.aussievoip.com.au/wiki-GXP-2000
03:36.23pifiulet me see rob one sec
03:38.36*** join/#asterisk squirrelv5 (squirrelv5@202.57.81.149)
03:44.39pifiuhey rob question
03:44.47pifiuin the review in the end it says this
03:45.07pifiuThe layout of the phone implies that these can be used for BLF's, but they can't
03:45.10pifiuwhat are BLF's?
03:45.17X-RobBusy Lamp Fields
03:45.23X-Robeg, program them to be othe rpeoples phones
03:45.25pifiuwhich does what?
03:45.29X-Roband they light up when they're on the phone
03:45.38X-Roblike a key system
03:45.46X-Robthe extensions are mapped to a lighted button
03:46.10pifiulol so confused
03:46.17pifiucan you give me an example in use?
03:46.23Slupjust read an artical on asterisk 2  is it true it's out?  i can't seem to find it anywhere
03:46.27pifiugive me a sceneraio
03:46.29X-RobNearly every cheap pbx does it
03:46.36X-Robslup, no, it's an april fools joke
03:46.54X-Robpif - ok. Have you ever seen a phone system that has buttons marked 'line1', 'line2' 'line3' etc?
03:47.30X-Roband the buttons light up when the line is in use?
03:47.36pifiuyeah
03:47.53SlupX-Rob thanks did sound a little to much
03:48.00X-RobThe odds are, those phones would -also- have buttons for extensions
03:48.08X-Robthat light up when that extension is in use, or is ringing
03:48.29X-Rob-that's- a busy lamp
03:48.40X-Robit lights up when the extension/line/whatever is busy
03:49.29edy2i saw ying only
03:50.15bkw_telephoneman, sure
03:50.33bkw_telephoneman, I know of some people that have setup a java fast agi server
03:50.51bkw_we'll be releasing res_js soon too
03:51.27bkw_http://bkw.dyndns.org/test.txt
03:51.28*** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc)
03:53.32*** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net)
03:53.35shidohrmm
03:54.24*** join/#asterisk Kaos76k (~chatzilla@CPE-24-163-140-163.new.res.rr.com)
03:54.39Kaos76kHey I received a Zyxel 2000W V2 today and it works great!
03:54.49bkw_and?
03:55.40Kaos76kThe wiki says they don't work at all.
03:55.49Kaos76kAll kinds of issues.
03:55.55bkw_the wiki says alot of shit that isn't true
03:56.30bkw_those phones are pieces of shit
03:56.35bkw_feel like NOTHING
03:56.42bkw_too light weight
03:56.42Kaos76k??
03:56.55bkw_if you want a real wifi phone get a Hitachi
03:56.56Kaos76ktoo light weight?
03:57.12bkw_the phone doesn't have any meat on its bowns
03:57.15bkw_er bones
03:58.19bkw_the funny part is
03:58.19bkw_CODEC: G711, G.729a(G.729 is recommended for better voice quality)
03:58.27bkw_thats an exact quote from zyxels website
03:58.51*** join/#asterisk cluecon (HeadachesA@wsip-68-99-73-32.tu.ok.cox.net)
03:58.51bkw_thats funny as hell
03:58.56bkw_cluecon, my man
03:58.57bkw_whats up
03:59.01bkw_where at in Oklahoma are you?
03:59.10clueconTulsa
03:59.17drraytulsa from his address
03:59.21drrayoops
03:59.29bkw_drray, you would say the same about me
03:59.35bkw_but i'm not in tulsa
03:59.44Godseyat last I have exim setup w/ sql and cyrus sasl
03:59.48bkw_adsl-69-154-144-52.dsl.tulsok.swbell.net
03:59.51clueconip addresses can be faked...not like i would.
03:59.51pifiuI <3 asterisk, its changed so much
03:59.52pifiui LOVE IT
03:59.54bkw_and i'm 110 miles away
03:59.58Godseyauto-create auto filtering spam into Spam, viruses in Virus :)
04:00.07drraywell, having grown up in austin, I don't trust anything that SWBELL says
04:00.08Godseydspam is slick bkw, have you used it?
04:00.23*** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net)
04:01.30Godseytrust goes as far as someone is willing to pay me ;)
04:03.13*** join/#asterisk waz- (~tjs@208.218.27.24.cfl.res.rr.com)
04:09.28X-RobAnd verily, rob did say, GCC 4.0.0 is an abomination undo Zog, and it should be stricken from the filesystem.
04:09.46antiX-Rob: whys that.
04:09.53X-Robit's giving me the shits, that's why
04:10.07MiccX-Rob, which side of the bed did you wake up on this morning?
04:10.13X-RobThe normal one.
04:10.26*** join/#asterisk Cresl1n (~matt@216.207.245.23)
04:10.33X-Robmy wife gets upset when I clamber over her to get out of bed.
04:10.34Miccdo your machines normally give you this much trouble?
04:10.37X-RobNo
04:10.39X-Robno they don't.
04:10.45X-RobBecause usually I don't use fedora.
04:11.01clueconX-Rob: what do you normally use?
04:11.07X-RobMandrake 10.1
04:11.13X-Robhowever
04:11.23X-RobI'm not impressed with 'Mandriva'
04:11.34Slupanyone got any info on when the 1.0.1.9 firmware for the gxp 2000 is comming?
04:11.39clueconah, therein lies the issue.  Fedora is a beast which, much like Debian, takes some tweaking to make it happy.
04:11.42X-RobSlup, it's on aussievoip.com.au now
04:11.43GodseyI've been using CentOS 4 and Gentoo
04:11.48X-RobI have it in my phone
04:11.51X-Roband it's _nice_
04:12.04Slupcheers X-Rob
04:12.37X-RobIf you have a fast net connection, you can download them from this website — set the 'Firmware Upgrade via HTTP' to point to 68.178.145.188/gxp2000/
04:13.10SlupX-Rob do you have one of these, just they still look a bit toy like in the photos wondering if they are like that in the flesh?
04:13.35X-RobYeah, I've got one. It -is- reasonably plastic compared to a cisco, but, so is everything else
04:13.49X-Robif I was them, I'd be sticking a big chunk of metal in the handset
04:14.00X-Robbecause it just feels too light. You want to have a bit of weight in it
04:14.27*** join/#asterisk faqu (pn@OL187-166.fibertel.com.ar)
04:14.27Slupyea and for the money all you can ask is that it dose what it says without complaints
04:14.58Slupi'm sure i cn find a bit somewhere to stick in it ;)
04:15.07X-RobHehehe
04:15.26X-RobActually, there's no screw on the handset to easily pull it apart
04:15.26Slupthe bugitones wernt even good as paperweights
04:15.42Godseywhy take it apart?
04:15.46Godseygold plate it
04:15.47Sluphmmm
04:15.59Sluplolololol pmsl
04:16.05*** part/#asterisk faqu (pn@OL187-166.fibertel.com.ar)
04:16.15Slupthe gold plating would double the value of the phone
04:16.47Slupcluecon thats about the general concensus
04:17.29Slupi'm trying to find didtant relatives i can palm the few i have on to
04:17.38Slup*distant
04:17.45*** join/#asterisk dynaqueue (~wbevan@adsl-63-205-16-227.dsl.mtry01.pacbell.net)
04:18.24X-Roblooking at the prices, I'm paying AU$120 for 10+ BT101's, or $160 for 10+ GXP2000's
04:18.34X-RobYou can't beat that.
04:18.50X-Robuh. That's $170 for 10+ GXP's even
04:19.03Slupnope and if they work they make an excelent entry level phone
04:19.48X-RobObviously, someone at grandstream heard everyone going 'God, these budgettone's look like $9.95 specials. I'm not putting that in an office!'
04:20.11X-RobSO they thought' Make it black with shiny bits. Give it grey buttons. And lots of lights'
04:20.16X-Rob*tadaah*
04:20.35Godseywith a cp3o ring?
04:21.01X-RobHaven't heard a cp3o ring?
04:21.04Miccdoes babblefish have a r2d2 translator yet?
04:21.22Godseyoh ya
04:21.26Godseyr2d2 not cp3o
04:21.26Slupi wish they had scratched the shiny bits befor they get scratched
04:21.37Godseyor that little buck rogers wiggy thing
04:23.31dynaqueueSo anyone tried sox and the GXP to get their own ring tones on it ?
04:23.46X-Robapparently it works with 1.0.1.9
04:24.15dynaqueueJust got hold of one this week works a treat especially with the beta firmware
04:24.49dynaqueuealthough managed to lock it up I think cos my DHCP server renews IP's every 24hrs so need to look into that
04:26.00GodseyI have my dhcp server hand out 1 hour leases :)
04:26.05X-Robyeah, so do I
04:26.52pifiuwhat does hanging out ips every hour do?
04:26.57dynaqueueweird I went to the GXP this morning and went off hook and that was it, light on but no body at home
04:27.00*** join/#asterisk L|NUX (~linux@202.63.195.147)
04:27.26Godseykeeps a clean lease.db
04:27.29*** join/#asterisk jasonkw (~me@CPE000cf18d47f6-CM00137188afce.cpe.net.cable.rogers.com)
04:27.58pifiui still have no idea what that would do, but if it works ill take it as a tip i guess
04:28.18X-Robpifiu, if you have lots of different machines that aren't always on
04:28.23jasonkwdoes any one here have any idea on how to setup a D-Link DVG-1402S w/ Asterisk?
04:28.26X-Robit doesn't make sence to have a long lease
04:28.44X-RobEither my machines are always on, so they'll never lose it, or they're in here temporarily
04:29.07X-Robooh
04:29.08X-Roboooh
04:29.11X-RobI've just found a new bug
04:29.18X-Robwho here is running 1.0.1.9?
04:29.26X-Rob(dyna?)
04:29.35dynaqueueWell for me the 24hrs works fine for the dhcp renewal besides I have plenty of IP's to doll out
04:29.45dynaqueueme......
04:29.49X-RobYou got two phones there?
04:29.54X-Robtwo of anything
04:29.57*** join/#asterisk DIEMACO (~diemaco@Toronto-HSE-ppp3751032.sympatico.ca)
04:30.21dynaqueueYeah external and the GXP
04:30.29dynaqueuewaiting for more to come :-(
04:30.43X-RobOK, GXP line 1 is 306, and I have a sipura 2000 as 300
04:30.45X-Robline1
04:30.47X-Rob300 [send]
04:30.55X-Robanswer 300
04:31.02X-Robpush 'TRNF'
04:31.13X-RobI get dialtone on the GXP and nothing on the sipura
04:31.22X-Robhrm
04:31.23*** join/#asterisk starsoldier (~joey@laptop.arsenic.org.uk)
04:31.26X-Robdidn't happen that time
04:31.53X-Robbut what happened a couple of times whas that after you pushed 'TRNF' then pushed 'Line2' then pushed 'line2' again to put the sipura on hold, the sipura didn't get MOH.
04:32.00X-Robmaybe I need to wait a bit longer
04:33.04dynaqueueMay be but I'm still not convinced they have the transfer sequence right. On the previous firmware you had the issue of hit transfer and both parties heard the DTMF tones.... So I still think that there is a lot to be done in that area.
04:33.12*** join/#asterisk brc-tux (~brc-tux@212.70.34.110)
04:33.20X-RobBah, it's working now
04:33.21*** part/#asterisk brc-tux (~brc-tux@212.70.34.110)
04:33.26X-Robbut it definately happened twice.
04:33.27dynaqueueI worked for a ACD company, and I'll tell you the hardest thing to get right are transfers and conferences... !
04:33.43X-RobHeh. Their problem was that they weren't expecting people just to push what they watn to do
04:33.52X-Robyou weren't ever meant to just push trnf
04:34.03dynaqueueLOL.....
04:34.08pifiuwhen is a new update for the gxps coming out?
04:34.15X-Robyou were meant to put the call on hold, then use another line, then push trnf and the on hold line
04:34.28dynaqueueCant wait for them to electrify the chair we are using !.....if we push the wrong button !
04:34.28X-Robpifiu, it's on aussievoip now, grandstream shoudl be releasing it on monday our time.
04:34.41pifiuoh wow not bad
04:34.46pifiuso these updates are pretty often?
04:34.59X-Robthese phones are brand new
04:35.06X-Roblike -really- brand new
04:35.15X-Robtheir first run of 5000 phones sold out in about 3 weeks
04:35.17pifiuso that means a lot of updates?
04:35.20dynaqueuePifiu its hands better than what is there at the moment on the firmware that coems with the phones...
04:35.25pifiudamn so where can i get one?
04:35.31X-Robthey've just finished another run of 10,000 phones (apparently) and are shipping new ones now
04:35.44pifiuwhere are the phones from?
04:35.48X-Robchina
04:35.54pifiuand where can i purchase one?
04:36.00X-Robgrandstream dealer?
04:36.01pifiui want to try one out
04:36.06X-Robdepends where you are
04:36.08X-RobI'm in australia
04:36.09pifiumiami, fl
04:36.13pifiuUSA
04:36.16pifiuonline website?
04:36.37X-Robif you can't type 'grandstream distributor' into google, I'm so not going to help you any more.
04:37.29pifiulol wait
04:37.36dynaqueueSo Rob... other than the transfer issue have you seen any other weird behaviour as yet. The speakerphone definitly has improved.
04:37.50X-RobHurm.
04:38.15X-RobI want them to be able to support SIP ANSWER_IMMED, so they can actually be used as an intercom
04:38.18MiccNow all I need to do is build freetds and unixodbc and I'll be all set.
04:38.23*** join/#asterisk faqu (pn@OL187-166.fibertel.com.ar)
04:39.03dynaqueueIsnt that what the auto answer feature is for. Not tried it, but wondered if that would work ?
04:39.30*** join/#asterisk michael1234 (~mick@staff.tsn.cc)
04:39.36X-RobIt does work
04:39.39X-Robbut it's not really what you want
04:39.39michael1234does anyone in here run ser
04:39.56X-RobYou don't want to configure a seperate account on there purely for intercom
04:39.57michael1234or is there a channel for ser
04:40.43X-Robbecause then the user may pick the intercom line up inadvertently
04:40.56MikeJ[Laptop]who uses head?
04:41.05dynaqueueThats where the electrified chair comes in handy !
04:41.15MiccI use current cvs head
04:41.32MikeJ[Laptop]can you test a patch for me?
04:41.51Miccsure.
04:42.04pifiuwow rob
04:42.11dynaqueueRob, well I would like for Grandstream to allow more access to the LCD they have......
04:42.12pifiuthey arent that expensive at all!
04:42.20X-RobMicc - by the way, I haven't had any luck with using RTC
04:42.20pifiui found one for 94 US
04:42.25X-Robyeah, they're cheap as.
04:42.29X-Robwhich is why they're so popular.
04:42.35dynaqueueDamn... Where Pifio.... $114 for my one !
04:42.46MiccX-Rob, yeah its broken, but I got it to be good enough with non smp kernel.
04:42.54pifiuhttp://www.myphonecall.co.uk/voip/iptelephones/grandstream/default.aspx
04:42.57pifiu+shipping though
04:43.01MikeJ[Laptop]http://bugs.digium.com/view.php?id=4511
04:43.07pifiuso it might be more since shipping is from europe
04:43.10*** part/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net)
04:43.17dynaqueueyeah true
04:43.49MikeJ[Laptop]it just give background an option to stop streaming only on certain keys, instead of any dtmf
04:44.31MikeJ[Laptop]I need outside feedback as to if it works or not
04:45.48*** join/#asterisk heath__ (~heath__@12-215-32-56.client.mchsi.com)
04:47.58Sluppifiu is that not GBP?
04:49.14MikeJ[Laptop]if you can comment feedback on the bug that would be wonderful
04:52.14*** join/#asterisk darwinfish (~funxion@69-165-200-17.miamfl.adelphia.net)
04:53.54darwinfishanyone still up
04:54.24Pete_Largoif that is British pounds (GBP) then it's more like $144
04:54.39Pete_Largobut I did find them on ebay for $109 free shipping.
04:55.26cluecondarwinfish: who says we ever go down?
04:56.52darwinfishlol
04:57.04darwinfishjust checking cuz I didnt see any activity
04:57.15darwinfishguess I didnt wait long enuff
04:58.05MiccMikeJ[Laptop], I've never applied a patch before, what do I do with this file? I've got it in my source directory now.
04:58.21rikstapatch -p0 < patch.diff
04:58.25MikeJ[Jayden]thanks
04:58.32Miccactually, thats not true, but its been about 10 years since I've ran patch.
04:58.39rikstamight have to adjust the -p value
04:58.57MikeJ[Jayden]Micc, then compile and install
04:59.16dynaqueueOffTopic < Anyone aware of the largest Asterisk deployment running in an ACD configuration ?
04:59.33MikeJ[Jayden]the patch adds a 4th option to background
05:00.01MikeJ[Jayden]you just set the 4th option to be the list of digits you want to make background stop playing and try that ext
05:00.14MikeJ[Jayden]any other digit, it will just ignore
05:00.24rikstaif he cant apply a patch how is he going to test it :)
05:01.11darwinfishanyone using astcc
05:01.25MikeJ[Jayden]anyone else happy to test :D
05:01.36MiccI'm on the phone, hold on.
05:01.38rikstai use stable, dont like to walk on the wild side
05:01.44Juggiemikej, doesnt playback already do that
05:01.57Juggiei know in agi stream file will break on what you specify
05:02.08MikeJ[Jayden]allow you to break when hit a key?
05:02.19MikeJ[Jayden]background does
05:02.24MikeJ[Jayden]playback does not
05:02.42Juggiewell if background does, what is the extra patch for
05:03.00MikeJ[Jayden]to specify which digits to break for, istead of any digit
05:03.58MikeJ[Jayden]it's for a project a friend is working on, but I have no box to test on at the moment
05:04.07Juggieoh
05:04.10Juggieagi already does that
05:04.19Juggieweird that you cant do it with background
05:04.45MikeJ[Jayden]in agi you do start and stop stream manually don't you?
05:04.57MikeJ[Jayden]I am not an agi user
05:05.01Juggieno
05:05.06Juggieits blocking
05:05.10Juggieit will return @ the end of file
05:05.18Juggieor when one the dtmf's in your list is pressed
05:05.46MikeJ[Jayden]hmmm
05:06.09MikeJ[Jayden]well... I need to go to bed.. if anyone tests that can you post in the bug please  :)
05:06.20Miccok. I'm off the phone.
05:06.53Miccok it built.
05:09.22michael1234what versions of ser do you guys run
05:10.08pifiuok i think enough for tonight
05:10.16pifiumy brain is getting overloadead with info
05:10.20pifiuMUST SLOW DOWN lol
05:11.04mostyi'm following this walkthrough http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html and i'm trying to listen to my voicemail. i get a menu system that states when my messages were, but i can't figure out how to actually hear the messages
05:14.43mostyahh, the messages were just zero sized files. fixed now
05:16.47*** part/#asterisk Cresl1n (~matt@216.207.245.23)
05:17.42darwinfishanyone using astcc?
05:23.19MikeJ[Jayden]Micc, now background needs to be tried out both to make sure that it works like it used to with the 4th param ommited, and that when you specify the 4th param it works as it should...
05:23.29MikeJ[Jayden]ok.. I'm really going to sleep now
05:26.25Micctalk to me tomorrow mikej
05:27.39MiccI don't think I'll get to it tonight. I built and am using it now but I won't test it until later
05:31.00X-RobAh
05:31.01X-Robsmeg
05:31.09*** join/#asterisk outtolunc (~me@adsl-69-110-5-162.dsl.pltn13.pacbell.net)
05:31.14X-RobMicc, with the RTC patches, it looks like they -were- right
05:31.34X-Robhttp://bugs.digium.com/view.php?id=4301
05:33.04outtoluncupdating
05:33.06outtoluncoops
05:33.37*** part/#asterisk heath__ (~heath__@12-215-32-56.client.mchsi.com)
05:34.08MiccX-Rob, do I need to apply these patches or are they already in CVS-HEAD?
05:34.26Nuggetmeetme and ztdummy need to die.
05:34.26*** join/#asterisk Astermigraine (~psolomon@69-165-217-96.atlsfl.adelphia.net)
05:35.19X-Robthey're already there
05:35.22X-Robthat's what you enabled
05:35.27X-Robyou can go back to hyperthreading and SMP
05:35.29X-RobI think
05:35.32X-Rob*checking now*
05:35.44X-Robhyperthreading and SMP breaks ztdummy badly -without- RTC
05:35.51MiccNo, I tried that and it didn't work.
05:36.01X-Robit should
05:36.12X-Robyou won't get 100&, nor will you get 99.98
05:36.18MiccI might need to apply that patch still, there was more to it than just the #if
05:36.21X-Robbut you'll get it bouncing between more and less than per seconds
05:36.37X-RobMicc, that patch is allready in ztdummy
05:36.46X-Robthat ifdef just comments it out
05:36.59MiccOk, well then, it doesn't help with smp kernel.
05:37.04rikstadoes anyone in here use ADM ?
05:37.15X-RobRebooting now and verifying this fact.
05:37.26X-RobI've been researching this in my long quiet period here
05:37.35MiccX-Rob, I've got pretty good response time now with my meetme I think.
05:38.03MiccX-Rob, I'll give you my 800 number if you want to see how mine sounds.
05:38.09X-RobI'm in australia 8)
05:38.14Miccdoh
05:38.17X-RobAnd FWD won't let me dial 1-800 numbers at the moment
05:38.20X-Rob(for some reason)
05:38.34Micccan you dial regular US numbers?
05:38.36X-RobActually, I should be able to
05:38.42MiccI've got about 10 DIDs
05:38.50X-RobGive me an 800 number and I'll see what I can do
05:39.02Micc(888) 206-4443
05:39.57Nuggetif you enable enum lookups there's some host out there that will grab any US toll free number you dial and let you route via iax.
05:39.57X-Robwhichroom no?
05:39.59*** join/#asterisk jerlique (~jerlique@lnk254.adl0.adsl.esc.net.au)
05:40.02Micc1234
05:40.14NuggetI dunno who they are, but they want you to route toll free calls through them
05:40.40Nuggetenum lookups are good karma anyway.
05:41.03Nuggethttp://www.e164.org/config.php
05:41.08X-Robheh
05:42.33Nuggetif you had enum lookups you could call me, and about 320,000 other phone numbers, for free.
05:42.53X-RobZapata Telephony Interface Registered on major 196
05:42.53X-Robztdummy: using RTC
05:42.53X-RobRegistered tone zone 0 (United States / North America)
05:49.19pifiuquick question
05:49.22pifiuin the sip.conf
05:49.36pifiuthe callerid=
05:49.42pifiuwhen you write it for example like this
05:49.54pifiucallerid=pifiu <?> what is ?
05:49.56Slupone quick Q befor i go to bed ....
05:50.19MiccI reboot that machine.
05:50.33Slupanyone here got good ethernet and phone cabeling knowledge?
05:51.04X-Robwhat do you want to know?
05:51.58Slupon a ethernet cable only half the pairs are used and convienently the middle two pins on the blue pair are free which is what a rj11 phone lead would connect too ....
05:52.50X-RobThat's correct, unless you're using gigabit
05:52.56X-Robgigabit uses all 4 pairs.
05:53.12Slupbut the local earth 'rinning' cable would normaly be on orange white which is used
05:53.13Slupyes
05:53.38X-RobThat last line made no sence to me whatsoever.
05:53.40Slupnot got that luxaury yet :(
05:54.00Slupmayby you don't have it in Oz
05:54.22X-RobConvetnion has it that blue/white is a pstn line
05:54.33X-Robor red/wite is voice on a MDF
05:55.08Slupon the UK system you have a master socket with the ringing cap and resister so it is not needed in every phone or secondary socket
05:55.23Slupstandard pstn lines that is
05:55.28X-RobOPoh yeah, this was mentioned on asterisk-users today I think
05:55.35X-RobNever had that in oz
05:55.39X-Robwe have two wire phones
05:55.43X-Robno ground thing.
05:56.11Slupwhich goes out on orange white which is one of the ground wires on the ethernet side of things
05:57.00Slupyou can have that in the uk too but you need to use phones that have the ringing cap and resister in the phone or it won't ring
05:58.29X-Robdon't use colours, use pin numbers 8) -- RJ45/100BaseT uses 1,2,3 and 6.
05:58.54Slupso i'm just wondering if you had ringing on the orange and white wire shaired with the orange and white of the network am i going to start shafting things
05:59.16Slupmmm just sent my other pc whith all that kind of stuff on into hybernate
05:59.18X-RobI'm guessing that sending a 48v signal into a network card is going to be bad.
05:59.26X-Robsmoke and sparks bad.
05:59.28Slupi thought the coulours were an int standard
05:59.42Sluplol yea thats that i was thinking
05:59.45Sluphmmm
05:59.57X-RobMicc, how'd your reboot go?
06:00.28Miccmachine freaked out and didn't boot.
06:00.36MiccHad to call up and have someone at the datacenter reboot it.
06:00.41X-Robick
06:00.42Miccmy boss just happened to be there still.
06:00.51X-RobThat could be good or bad
06:01.21Miccok, zttest looks ok.
06:01.42Sluplooks like i'd have to either use a master mod tap with the ringing cap in it or move the ringing wire to the brown pair and make moded rj11 cables
06:01.44jerliquedoes anyone have a url to a flow diagram of how SIP and RTP sessions work with SER and say an * media proxy??
06:02.04Micchow do I see how many cpus my machine thinks it has?
06:02.46X-Robcat /proc/cpuinfo
06:02.51SlupX-Rob do you know of any site where i can find info on the various rj11 standards i've had a google but didn't find much
06:03.28Slupwell not amoungst the multitude of component catalogues which didn't have much info
06:04.32Slupor anyone else whos still alive
06:04.50Pete_Largopifiu the extension number: callerid=pifiu <1002> for example
06:06.01X-Robhttp://www.faqs.org/faqs/LANs/cabling-faq/
06:06.31*** join/#asterisk fugitivo (~ajf@201.255.104.46)
06:06.33fugitivohello
06:06.56Sluprobs the man !!!
06:07.46Sluprob i hope while your not on here your goign out and earning good money and not just wastiing your talents on us
06:08.00Pete_Largoputting ring voltage on a UTP cable that has data running on it is probably not a good idea re: the continuity of the data....  probably be less of a headache to run a second cable.
06:08.14MiccX-Rob, could you call in again?
06:08.17fugitivowhat's the best codec for a adsl 256kbit/s conection?
06:08.41Pete_Largo~codec
06:08.42jbotsomebody said codecs was http://snipurl.com/wiki_codecs
06:09.03fugitivothanks
06:09.21SlupPete_Largo yup thats what i was thinking but you never know so i thought i'd stick the question in
06:10.11fugitivois iLBC a good option?
06:10.12Pete_LargoI've never done that before Slup, that's my story and I'm stickin' to it :)
06:13.28MiccThanks for the help, X-Rob.
06:13.46X-Robno probs
06:13.55X-RobI'd like to find out where all that latency is coming from tho
06:13.57X-Robcoz that was crud
06:14.15X-Robwell
06:14.28X-Robactually it would be voip -> us -> pstn -> voip -> you
06:14.38X-Robat least 4 conversions
06:15.18MiccI'll test it some more later.
06:15.28X-Robah
06:15.37X-Robfwd is now letting me dial 1800's
06:16.26*** join/#asterisk drooth (~drooth@ip68-107-113-76.sd.sd.cox.net)
06:16.31droothat long last i have QoS!!!!!!!!
06:16.38droothI am using latest PFsense
06:17.08fugitivoanyone is using qos with linux?
06:17.31droothi am maxing out at 3 calls however, 4th call and it's choppy
06:18.11*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
06:19.44*** join/#asterisk loud (~roots@cypher.punk.net)
06:19.48Pete_LargoHey, I just had a brainstorm... IP Phone with an analog port on the side... so you can plug in your regular house phones like an ata...
06:42.50*** join/#asterisk drooth (~drooth@ip68-107-113-76.sd.sd.cox.net)
06:42.57droothhow do you transfer a call in *
06:44.04X-RobUm
06:44.10X-RobDEpends
06:44.19X-Roblook at /etc/asterisk/features.conf
06:44.26X-Robthat's where you configure the hotkey
06:44.35X-Roband you need to have the dial string with 'Tt' on it
06:44.46X-Robusually, you use your phone to do transfers.
06:45.09droothyeah i am using a spa-841
06:45.16droothdisplay is kinda crappy
06:47.16droothX-Rob: I don't have any hotkeys there
06:47.25droothX-Rob: do you have an example you could paste
06:47.38droothX-Rob: like hitting # twice or..?
06:48.32droothto put the caller on hold and then allow to transfer
06:48.51X-Rob[featuremap]
06:48.51X-Rob;blindxfer => #1                ; Blind transfer
06:48.51X-Rob;disconnect => *0               ; Disconnect
06:48.51X-Rob;automon => *1                  ; One Touch Record
06:48.52X-Rob;atxfer => *2                   ; Attended transfer
06:49.01X-RobThat's possibly only in CVS HEAD
06:49.04droothwow
06:49.13droothcool
06:49.41droothCVS HEAD?
06:50.16droothlet me paste that in and restart
06:50.34X-Robyou'll need to uncomment them
06:50.37droothright
06:50.38X-Rob; == comment
06:51.00X-Robheh
06:51.04X-Robnew to IRC are you? 8)
06:51.10droothno
06:51.12droothtired
06:51.14X-Robhehehe
06:51.16droothbeen on for 8 yrs
06:51.20droothapprox
06:51.34droothso those are universal hotkeys i can use when people call
06:51.46droothbut can only be used if logged into queue
06:51.55droothand i have answered a call
06:52.38X-Robhttp://www.voip-info.org/wiki-Asterisk+cmd+dial
06:52.41X-Robread that too
06:53.04drooththx
07:00.23*** join/#asterisk cmk (~cmk_@p54A3F060.dip.t-dialin.net)
07:01.00droothread it...
07:01.07droothinteresting
07:02.16*** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode)
07:02.29droothAttempts to establish a new outgoing connection on a channel, and then link it to the existing input channel.  not quite sure what that means actually
07:02.34droothconference?
07:03.14X-Robwhere's that?
07:03.19X-Robconferencing is done with apps
07:03.22X-Robapp_meetme and such
07:03.28droothi have conferencing
07:03.34drooththat is from the link ou gave me
07:03.58X-RobOh
07:04.10X-RobThat's a complex way of saying 'It dials this number'
07:04.21X-Robbut
07:04.22X-Robuh
07:04.27X-Roblemme rephrase that
07:04.27droothso why did you give me that link?
07:04.39X-RobBEcause on the Dial command, you need to have 't' and/or 'T' options
07:04.45X-Robbefore you can transfer a call
07:05.54droothi hit # and i was able to transfer! wow
07:06.06droothguess I have it already
07:06.09droothusing a modified config
07:07.14X-RobYou're using Asterisk@Home I guess?
07:07.50X-Rob(It helps to say that at the start 8)
07:08.59droothyeah
07:13.55*** join/#asterisk Cheng29 (~cheng29@d57-87-253.home.cgocable.net)
07:15.38jerliquewhy is something like rtp proxy used for helping traverse nat'ed clients? My understanding is that SER can re-write the packets to show the correct ip addresses....
07:16.09droothseems that if I hit # I can do a blind transfer. dope
07:16.30*** join/#asterisk RestLessGemini (~umairbari@202.142.189.86)
07:18.42X-Robyou'll need CVS-HEAD to do attended transfers
07:19.28*** join/#asterisk cmk (~cmk_@p54A3F060.dip.t-dialin.net)
07:21.33droothok
07:22.46xeet2attended transfers aren't in stable?
07:24.52X-Robafaik not
07:24.55X-RobNot that I've tried it
07:38.28*** join/#asterisk MikeJ_ (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net)
07:38.29droothwhere are the voice messages located for each extension?
07:39.26mostydrooth, they should be in /var/spool/ somewhere
07:39.31Hmmhesaysjoan jett is scarey now
07:40.05droothumm /var/spool/asterisk/vm
07:42.57*** join/#asterisk loud (~roots@cypher.punk.net)
08:04.54*** join/#asterisk djin (~djin@gridfox.xs4all.nl)
08:08.30droothanyone know how i can change the codec per call?
08:08.42droothlike the first 2 calls get highest codec 80k and then all calls after get lower quality
08:10.31droothhow do you record a greeting by calling into * via a phone
08:12.52jerlique80k?? what codec is that?
08:15.09*** join/#asterisk dasuberdavid (~david@pcp01540029pcs.huntsv01.al.comcast.net)
08:15.11droothhighest quality
08:15.21droothincluding overhead
08:15.29drooth64k + IP overhead
08:23.02droothhow do you dial a SIP URL from firefly?
08:30.47*** join/#asterisk Godsey (lanny@204.17.223.9)
08:46.57X-Robdrooth, I don't think you can.
08:47.18droothhmm i just installed new version
08:47.28droothit looks like they took out features to dial your own IAX network??
08:57.59droothhttp://www.virbiage.com/firefly/download/firefly-thirdparty.exe
09:07.36*** join/#asterisk zoa (zoa@82.103.76.147)
09:50.07*** join/#asterisk MikeJ[Laptop] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net)
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10:49.06*** join/#asterisk ruski (ruski@adsl-160-122.swiftdsl.com.au)
10:50.16ruskiI've setup asterisk to work with my SIP provider, they dont know about it, when i make an incoming call from the outside world, it bounces through my provider then into my asterisk then onto the voip phone, when im calling i cant hear any ringing on the handset, it's just silent .. any ideas?
10:51.07Mavvieyou should tell them about it.
10:51.25*** join/#asterisk twisty (twisty@ndn-165-143-17.telkomadsl.co.za)
10:51.30twistyhi all.
10:51.34Mavvieare you registering?
10:52.06ruskiMavvie: they'd most probably not be very happy, yeh i get registerd, i can even make calls to the voip phone
10:52.14ruskiit's just the calling party doesnt hear any ringing
10:52.14twistythis will probably sound stupid but how do you tell what version of asterisk you are running other than the show version
10:52.28ruskijust silence until the reciever answers the call
10:52.59Mavvieit's the local phone who makes the ringing sound until it's picked up.
10:53.40Mavvieuntil the call is answered that is
10:54.42twistyok. is there anything that one should specificaly do when updateing. other than backup the config files.
10:55.04cochir-option to the Dial command or Ringing command in the extension on lower prio? ;)
10:55.13*** join/#asterisk outofjungle (~outofjung@61.17.134.230)
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10:59.28twistyare there many people following this chat ?
10:59.47Mavvieno
11:00.26twistyMavvie : am about to update asterisk installation. is there anyhting i should watch out for or do specificaly to ensure that it comes up again without problem
11:00.50Mavviejust make sure you have the old binaries still, just in case.
11:00.59twistyhave done that thanks/
11:12.50*** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net)
11:13.12twistyMavvie : this is not an urgent one but i set the loadzone and defaultzone to za. for south africa but asterisk still does not pickup on the engaged tone and runs through all options before hanging up the line. do you have an idea why this is happening or is there something else to change the zone ?
11:16.49*** join/#asterisk nazgool (~nazgool@p62.246.134.116.tisdip.tiscali.de)
11:16.50nazgoolhi
11:19.22nazgooli just tried to make a modules.conf with autoload=no. after solving my module dependency issues so far, i got asterisk starting up without errors, but:
11:20.15nazgoolwhen i say "extensions reload" i get the error "No such command extensions'"
11:20.56nazgoolwhat module must i load to have the extensions command?
11:22.10*** join/#asterisk nicox (~nicox@h082218027030.host.wavenet.at)
11:22.33nazgool(the modules loaded so far are: res_features.so; app_parkandannounce.so; res_musiconhold.so; chan_sip.so; format_mp3.so; format_wav.so; codec_ulaw.so; chan_capi.so)
11:22.58nazgoolany really crucial module i would be missing to run asterisk with sip + capi?
11:23.38nazgool(well, obviously there's at least one missing: the one that would give me the 'extensions' command)
11:23.42nazgool?
11:24.51*** join/#asterisk Romik_ (~romik@1.fix.netvision.net.il)
11:27.44*** join/#asterisk rvhi (~rv@66.175.65.89)
11:33.59ruskiis there a way in asterisk to work out what codecs a peer // another sip server has available ?
11:36.24Romik_ruski: ty hot' ponyal chto sam to skazal?
11:40.53nicoxis there any specialist with extension.conf?
11:42.33Romik_nicox: there a lot of guru's of this file...;)
11:43.11djinnicox, define specialist.
11:44.10Romik_djin: specialist is 1 steps before guru the step name is expert ;)
11:46.03djinso nicox, phat's the wroblem?
11:49.45*** join/#asterisk stkn_ (nobody@stkn.developer.gentoo)
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11:51.52*** join/#asterisk Yellow_Fuzzy (~yellow@c211-31-41-107.wavrl1.nsw.optusnet.com.au)
11:52.03Yellow_Fuzzyhi
11:53.05Yellow_Fuzzyanyone here got or used a Draytek 2900V
11:54.08djinI use a 2600VG, not sure if that covers your question.
11:55.01newlActually, the specialist would probably be a grunt, then the SME (subject matter expert), then the GURU. :D
11:55.20Yellow_Fuzzybasically i bought a 2900V today and connected it to my optus cable modem and have tested and discovered that my internet speed is about 200KB/s slower when through the 2900V then when i plug directly into a pc
11:55.57newlBlame Optus.  Everyone else does. haha
11:56.46djinYellow_Fuzzy, I don't seem to have that problem. Download is running at maximum.
11:56.55Yellow_Fuzzyi know its not the 2900V that is slowing it down because as i said it works 200KB/s faster running the same speed test
11:57.04Yellow_Fuzzyand ive updated to the latest firmware
11:57.55Yellow_Fuzzyand ive tried turning off QoS & all the firewall features
11:58.14*** join/#asterisk nemisus (~nemisus@203-206-228-13.dyn.iinet.net.au)
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11:58.54*** join/#asterisk RestLessGemini (~umairbari@202.142.189.86)
11:59.25Yellow_Fuzzyany ideas of what i could try to get it to work properly
12:00.36djinNo, I'm sorry + I think this is not the best place to find answers.
12:01.30Yellow_Fuzzyyeh i know its probably not the best place, but ive posted on a few forums & am waiting for answers from them
12:02.39djinnicox, did you try setting the DigitTimeout?
12:02.40djinhttp://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20DigitTimeout
12:03.33djinand ResponseTimeout
12:03.33djinhttp://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+ResponseTimeout
12:05.48nicoxhow do i set this for a PRI?
12:06.33djinFor a PRI? You can set this in the extensions.conf
12:07.03*** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk)
12:07.03*** mode/#asterisk [+o bkw_] by ChanServ
12:09.13MikeJ[Laptop]morning bkw_
12:11.06nicox(msg djin i read this "If not specified in the dialplan, the default is 5 seconds. " on the website but i have no 5 seconds to dial the correct number. so i think this is no solution!?
12:11.12*** join/#asterisk gres (~serg@81.222.48.242)
12:19.19*** join/#asterisk dudes (~dudes@12-215-34-84.client.mchsi.com)
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12:35.47*** join/#asterisk asterbeast (~acidburnn@cust.7.204.adsl.cistron.nl)
12:35.50asterbeastppl...
12:36.16asterbeasti have some probs trying to call out to the 612 number for time
12:36.32asterbeastDial prefix is 393...
12:36.43asterbeastcan someone help?
12:39.48*** join/#asterisk Cybertank (~cybertank@CPE000dbd0f269c-CM00111ae6ff9c.cpe.net.cable.rogers.com)
12:42.23asterbeasti have some probs trying to call out to the 612 number for time.. my prefix is 393 but get -- IAX2/fwd/3 is busy
12:42.36asterbeastcan someone point me to the right direction?
12:45.43djinwhat is your exten line?
12:47.02asterbeastin extensions_additional.conf?
12:47.31asterbeast[outrt-002-test393]
12:47.31asterbeastinclude => outrt-002-test393-custom
12:47.31asterbeastexten => _393.,1,Macro(dialout-trunk,2,${EXTEN:3})
12:47.31asterbeastexten => _393.,2,Macro(outisbusy)       ; No available circuits
12:47.53asterbeastis this my prob?? "No available circuits"
12:48.58asterbeastno.. i dont think so..
12:49.28*** join/#asterisk destro_dm4 (~rdsaffell@dsl092-161-155.wdc2.dsl.speakeasy.net)
12:49.39*** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net)
12:50.11destro_dm4anyone know how to automatically add a prefix (9) to numbers in the call list on Polycom 500 phones?
12:56.50*** join/#asterisk wooden (zaiwot@p54AA03DD.dip0.t-ipconnect.de)
12:56.55woodenhi
12:57.05woodenive a problem with meetme, conferences
12:57.15woodenchan_zap.c:916 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory
12:58.08JerJeryou don't have zaptel timing
12:58.33woodeni use the debian 3.1 asterisk
12:58.44JerJerthat was your first mistake
12:59.13woodenoh there are zaptel package in debian
12:59.35JerJeruse the source - that's why it was written
13:00.18*** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com)
13:00.25RoyK~seen inspired
13:00.26jbotinspired is currently on #asterisk
13:01.38woodeni think the support for chan_zap isnt build in debian?
13:01.42woodenu
13:02.14asterbeasti have some probs trying to call out to the 612 number for time.. my prefix is 393 but get -- IAX2/fwd/3 is busy
13:03.14JerJerthen the number is busy
13:03.29JerJerwooden:  use the source
13:04.30asterbeastok.. JerJer... but this number is the standard for time isnt it?
13:07.11asterbeasteverythig i call (when using the prefix 393 ) ends up with: "Executing Playback("SIP/200-e685", "allison7/all-circuits-busy-now") in new stack"
13:08.22Ariel_asterbeast, are you using amp for your setup?
13:08.37asterbeastand the console yes...
13:09.32Ariel_did you setup fwd as sip or iax2?
13:10.05Romik_any one uses bluetooth channel in asterisk?
13:11.19*** join/#asterisk WeezeyD (Weezey@206.210.111.117)
13:12.11*** join/#asterisk cluecon (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net)
13:13.36WeezeyDcluecon: you have a PBX Developers Conference in Chicago from Aug 3rd to 5th.
13:13.40*** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it)
13:14.16woodeni need for zaptel kernel drivers?
13:14.20CybertankAnyone else install asterisk and digium card in Fedora Core??
13:15.09*** join/#asterisk Laibsch (~Laibsch@G1dfb.g.pppool.de)
13:17.00asterbeastAriel_ : iax2
13:17.47CybertankWhat distro is most recommended for asterisk and digium?
13:19.21asterbeastCENTOS??? from the  asterisk@home project
13:22.12clueconCybertank: I use FC3 and it works.
13:23.46dwmw2_goneRomik_: I do
13:24.11*** join/#asterisk nain (nain@61.5.143.19)
13:24.20nainHi Every body
13:24.32WeezeyDHi Doctor Nain!
13:24.35Ariel_asterbeast, did you follow the hand book on setting up fwd on with Asterisk@home?
13:24.55nainWeezeyD: I am Not Doctor :)
13:25.10WeezeyDBut you play one on TV?
13:25.15Ariel_Cybertank, most will work. There are post on the wiki on how to setup your choice.
13:25.27WeezeyDhow do I grep through a directory full of gzipped files?
13:25.30Ariel_Cybertank, which Distro do you like for your other use?
13:26.08nainWeeseyD: That may be some one else
13:26.29Yellow_FuzzyWeezey: as in you want to search through the contents of thoes gzipped files?
13:26.36nainWell Any one Can help me ?
13:26.59nainI want to recieve IP call in SIP using asterisk ?????
13:27.19Ariel_nain, that is what asterisk does
13:27.38newlwees: something akin to: for i in `ls .`; do gzip -l -c $i|grep foo; done
13:27.41file^^^ that is what chan_sip does
13:28.35nainAriel_: yes but my problem is that carrier is sending in H323 and i have SIP account So i want to directly calls to SIP not in H323??
13:29.04nainI don't want to use chan_h323 or chan_oh323.so
13:29.25Ariel_nain, then get a different carrier
13:29.37fileuh well, there's another channel driver in asterisk-addons too... but if they only do H323, nothing you can do about it
13:30.45nainActually problem is that i was using h323 too but H323 driver has lot of problem to me like no ringtone or dead air, Protocol Error, So i decided to recieve call only in SIP as my provider is USING SIP
13:31.21fileyou're contradicting yourself
13:31.36newlindeed.
13:31.57nainfor the above said reason i don't want to more use H323 . I HAVE DISABLE H323 DRIVER and only want call in SIP
13:32.18*** join/#asterisk Rez (lorez@lorez.staff.freenode)
13:32.22fileyeah... so only use SIP
13:32.45nainbut customer is sending call in h323 how to recieve in SIP ?
13:32.57fileyou have to tell them to do it...
13:33.05filethat's something between you and your provider
13:33.06filenot asterisk
13:33.09nainThey are not ready to do it
13:33.18newlor you and the customer. :)
13:33.32filewell then what are you asking?
13:33.40file"I want to do SIP... but they're using H323, and I don't want to"
13:33.41fileboo who
13:33.45nainActually in Customer and me there is VOIPESCROW which is doing this conversion but calls are failed
13:34.39newlSell the customer a SIP enabled device, set them up with a provider, collect the commission, problem solved.
13:35.22nainBut Whole Sale Carrier Only like to Use H323 not SIP
13:35.48RoyKGuten Tag
13:35.58filenain: and what do you want us/asterisk to do?
13:37.00nainfile: Recieve call from particular IP in SIP and Dial to my Provider
13:37.31filethat's what asterisk does... go Google, tons of examples
13:37.35fileliterally TONS
13:38.22nainbut that examples are with chan_oh323 or chan_h323. I don't want to involve h323
13:38.25RoyK~docs
13:38.28jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
13:39.06Ariel_nain, your going to have to setup h323 of some type on your system.  if not then your not going to get the calls. Again if you don't want h323 get a different carrier there are many that do sip or even iax2
13:39.31RoyKnain: no, just configure a sip account in sip.conf with context=asdf and in extensions.conf add [asdf] with exten => dial(SIP/${EXTEN}@provider)
13:39.36RoyKnain: rtfm
13:39.52fileAriel_: thx
13:39.59filetoday is not turning out to be a good day
13:40.08fileSwiss Chalet got rid of my chicken strips
13:40.09fileSOOOOOOO
13:40.11fileI'm cheating
13:40.20fileand getting two kids chicken strip dinners :P
13:40.22RoyKnain: and if your provider only runs h.323, there's no fucking way you can avoid using it yourself
13:40.56nainRoyK: i have a context like [voip-escrow] host=carrier-ip ...... in context=sip-call in sip.conf and have extenions in extension.conf which is dialing to my provider
13:41.21RoyKnain: have you read the asterisk docs yet?
13:41.50nainRoyK: yes ! But some one tell me that it is still possible without using h323 .
13:42.02newlheh
13:42.04RoyKwhat isn't possible without h323??
13:42.17nainIP to SIP
13:42.26RoyKSIP to SIP?
13:42.46nainofcourse SIP to SIP is no problem but problem is IP to SIP without involvind h323
13:42.51RoyKiirc IP is a layer three protocol and SIP a layer 7 protocol, so IP to SIP is quite hard....
13:43.15newl'e's lost the plot gov!
13:43.38fileRoyK: you're mean
13:43.44RoyKSIP over NetBEUI :)
13:43.53RoyKfile: i'm well aware of that
13:44.03Ariel_nain, h323 is like spanish and sip is like english the don't talk to each other unless there is a translator(asterisk) but for the translator to do this it needs to speak both spanish and english
13:44.04nainSOme one tell me that in sip account set host=IP not dynamic and qualify=yes, insecure=very will work but it is not working
13:44.32filefood ordered, yay
13:44.36Ariel_nain no no no sip is different the h323
13:45.01filenah, german!
13:45.08newljavole!
13:45.11nainI KNOW SIP IS DIFFERENT THEN OH323 BUT FEW PEOPLES ARE DOING AND THEY SAID IT IS TRICK. JUST GO AND FIND
13:45.33Ariel_nain, there pulling your legs
13:45.39nainStrange
13:45.45newlboth of them. :)
13:46.17WeezeyDdamn, why does mgcp do *67 by default, but SIP doesn't.  That's irritating
13:46.28nainThey are very expert and Professional. What do u say to VOIPESCROW TECH. An online billing solution ?
13:46.46fileWeezeyD: SIP doesn't do it, the phone does it - on MGCP, the server (asterisk) does it
13:46.55filenain: SIP
13:46.56filegah
13:46.58nainI m using Asterisk as Server
13:47.19filenain: SIP isn't H323, let's just get that down pat... if you are using H323 in some form, you need an H323 channel driver (chan_h323, chan_oh323, chan_ooh323)
13:47.32newlor a gatekeeper
13:47.45Ariel_nain, if they send you signal in h323 the only way asterisk is going to know it is with the h323 drivers one of 3 out there. You have to have a translator.
13:47.49fileor something else that'll do H323 and SIP on it's own and do the conversion of signalling for you
13:47.56WeezeyDfile: the SIP phone does it 'eh?  Hmm...
13:48.11fileWeezeyD: if it doesn't, you write dialplan logic to
13:48.24fileSIP phones do most everything
13:48.41WeezeyDfile: I think it does, I just have sip.conf setting it, so that would be over-writing it.
13:48.42Ariel_Weezey, you can setup rules to do that.  Also like some sip device sipura it's in there configuration for many *67 and other codes built-in
13:49.14nainok you are right
13:50.28nainbut they said that by setting host=IP    SIP.conf is still able to recieve and dial the extensions. I have succeeded lit bit but calls are failing. The person who guide me is not available due to Weekend and he is using the Trick. So i can't wait for him for 2 days Therefor i m here to ask u for help ????
13:50.50fileif you want SIP to work that's easy, like I said there's TONS of examples out there
13:50.51RoyK~lart nain
13:50.55nainI DON'T KNOW WHAT LOGIC HE IS USING, HOWEVER HE IS DOING
13:50.58newlheh I've got dialplan logic for all that fun stuff.  Didn't take but a couple of days from start to finish.  It'd be nicer to have built in though and controlled through features.conf. :)
13:51.19fileSIP and H323 are two completely different configured things
13:51.34fileso, go find examples and don't give me "oh I have... there's none"
13:51.36filebecause I know there's TONS
13:51.37nainHmmmmmmmmmmmmm
13:51.46fileso go use your brain, common sense, and read
13:51.49RoyKSIP is a nice protocol designed with headers like HTTP. H.323 is more like ISDN over IP
13:51.51nainTHANKS
13:53.29nainOK ONE MORE THING DID GnuGk can convert h323 calls to SIP ?
13:54.21RoyKnain: no, nothing can convert between SIP and H.323
13:54.51nainSO I MUST USE CHAN_OH323.SO DRVIER FOR DOING THIS
13:55.01RoyKnain: but asterisk and possibly yate can perhaps bridge between them
13:55.10RoyKnain: don't use caps you moron
13:55.41nainOK
13:55.44RoyKnain: you will have to use some channel driver for h.323 to use h.323, yes
13:57.03nainOK I have also used asterisk_oh323.0.7.1 and asterisk_oh323.0.7.2pre1 driver but there are some problems with using these drivers for examples, noringtone to carrier, dead air, or protocol errors etc.
13:57.40nainAs these are the latest Channels Drivers for oh323
13:58.29RoyKi beleive chan_h323 with asterisk cvs head is the best you get
13:58.31clueconnain, welcome to the world of h323
13:58.31RoyKby now
13:58.35RoyKalthough i'm not sure
13:58.57newlSo isolate the cause of the bugs, fix, submit patches, enjoy the open source wave. :)
13:58.59Ariel_nain, h323 does work. But there not easy to setup. That is the main reason poeple are going away from h323.
13:58.59nainI have used latest CVS head version of 3 days before of asterisk
13:59.04RoyKnain: but, there is something called aefirion that is an asterisk fork and that has an OPAL-based h323 driver
13:59.09RoyKthat should perhaps work better
13:59.10RoyKor
13:59.12RoyKYate
13:59.16RoyKyate.null.ro
13:59.25RoyKthat uses h323 natively
13:59.34RoyKwhich again may work better than asterisk on h323
13:59.39RoyKbut then
13:59.53nainhmmmmmmmmm but is Yate do translation from h323 to SIP?
14:00.01RoyKno, nain
14:00.14RoyKbut as i said, it may gateway between them
14:00.16nainSo the problem is same still
14:00.22RoyKlisten to what i say
14:00.24RoyKidiot
14:00.37nainok
14:00.51RoyKyou don't  translate protocols
14:00.56RoyKyou gateway between them
14:02.23nainthat' it
14:02.53nainWell so Yate will perform well as a Gateway between h323 and SIP
14:03.01RoyKyes
14:03.15RoyKjust as asterisk or aefirion may do with chan_h323, chan_oh323
14:04.11naini have seen lot of support for asterisk . What about Yate ?
14:04.13RoyKor aefirion's woomera/opal support
14:04.21RoyKnain: ask on #yate
14:04.52RoyKl-fy may be bitchy, but she knows a lot if you treat her well
14:05.17nainWhere is she ?
14:05.45RoyKusually hanging out on #yate
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14:06.10nainThanks
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14:15.42WeezeyDRoyK; I can't get h323 to run with CVS HEAD and I can't get oh323 0.7.1 to compile.  :(
14:16.18WeezeyDI wouldn't care, but I can hook up to my buddy's phone system with oh323.
14:16.25WeezeyDor h323
14:17.05RoyKdon't ask my
14:17.08RoyKi only use iax2 and sip
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14:23.10nainHI Every BOdy
14:23.15WeezeyDyou again.
14:23.37nainuffffffffff you are also here
14:24.01nainActually i didn't find someone in Yate channel so i thought to search out more on other channels
14:26.39WeezeyDand died.
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14:28.27woodenive installed ent extracted the zaptel-source and edited the Makefile how to install this no correct?
14:28.37woodenups wrong channel
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15:00.32`SauronAre there any good CDR post-processing scripts/things?
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15:03.40ersilahey all. can anyone reference me to somewhere in the dial plan where i can add a 911 feature? i'm using sip.
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15:07.46jackxDoes someone know what a standard gsm-gateway (1 sim) may cost?
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15:10.39eKo1ersila: eh, exten => 911,dial...
15:11.11eKo1jackx: you mean for cellular?
15:11.33jackxyes
15:11.43WeezeyDis it possible to dial a pause to a PSTN somehow?
15:11.50Godseyexten => 911,1,Background(not-so-pro-gun-control-now-are-you)
15:12.08tzangerGodsey: hahaha
15:12.15tzangerthat's not nice
15:13.10GodseyI'm a firm believer that people kill people, not guns
15:13.17Godseyand god helps those who help themselves ;)
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15:14.16GodseyWeezeyD: it is, I'm trying to recall how
15:15.33WeezeyDGodsey: Within the phone number, you may use the special modifier w to indicate a half-second pause.
15:15.36eKo1define 'dial a pause'
15:15.41WeezeyDGodsey: sound right?
15:15.52natoxi've a problem when i work with my offline ip my cisco ata 186 is responding but when i try to assign one of my 3 ips isn't working how may i fix that?
15:16.03Godseythat I don't know
15:16.11GodseyeKo1: pause is standard feature
15:16.24GodseyWeezeyD: I did it a hackish way then
15:16.30eKo1really? I've never heard of it.
15:16.39Godseyyour phone has pause
15:16.51WeezeyDGodsey: really? what did you do?
15:17.17eKo1It does? All I see is redial and recall buttons.
15:17.17GodseyI used an agi
15:17.32pigpencould someone post what that cheap fxo pci card is that asterisk is cool with (wildcard or something?)
15:17.34Godseyand had it just continue the dial plan
15:17.52eKo1pigpen: x101p
15:18.05pigpenthanks...I have a bad memory.
15:18.40Godseyoh i see :)
15:18.53Godseymine was a bit different
15:18.59GodseyI did it w/ a connection macro
15:19.13Godseyyou put pause=yes in zapata.conf?
15:19.16eKo1does 'pause' mean 'hold'?
15:19.32Godseyno, delay
15:20.04eKo1aren't delays just added with exten => ...,wait(...)
15:20.13WeezeyDGodsey: no I didn't put pause=yes, but I can.
15:20.30WeezeyDeko1: not if you're dialling
15:21.18eKo1Why would I want a delay when I'm dialing?
15:21.27pigpeneKo1, is the Astralis X101P clone ok?
15:21.32WeezeyDbecause you need time to scratch your nuts before the call connects
15:21.49eKo1pigpen: no clue, i use the wildcards from digium.
15:21.55pigpenk
15:22.04Godseyexten => 911,1,Dial(Zap/911)
15:22.10Godseybut I'm not positive that's correct
15:22.14GodseyI don't use zap devices
15:22.14tzangerno that is not correct
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15:22.21eKo1Weezey: really? i do after the call connects.
15:22.24tzangerZap/[channel_number]/911
15:22.37tzangerand maybe a soft hangup before to make sure the line's free
15:22.51jackx[channel_number'
15:22.57jackxcan't that be like a group of lines?
15:23.00tzangeryes
15:23.02tzangerif you have a group defined
15:23.03jackxso it figures out wich is free by itself?
15:23.09tzangerbut still, you want to make sure *A* line is free
15:23.12tzangeror better
15:23.13jackxtrue
15:23.22tzangerdial, and then if all are busy, soft hangup, wait a second and goto(1) again
15:23.24Godseyi only said that because someone took my joke seriously about gun control
15:23.41tzangeryou can make it more complex not to cut off an emergency call but IMO that's getting a little overboard
15:23.44tzangerbut not impossible at all
15:23.52tzangerGodsey: who?
15:24.16Godseyanon unless they want to say :)
15:24.31tzangerha
15:24.38WeezeyDhow can I set EXTEN ?
15:24.39tzangeralison should do that phrase
15:24.51WeezeyDI do Setvar(EXTEN= but it seems to ignore me.
15:24.55tzangerWeezeyD: Goto(context,exten,1)
15:25.22WeezeyDcool
15:25.23WeezeyDthanks
15:25.57Godseywell
15:26.01Godseythere is an easy way
15:26.07Godseypick channel 1
15:26.15Godseyand kill any call on it :)
15:26.17Godseythen dial
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15:26.35Godseyor drop all zap channels and dial 911 out the group
15:26.48tzangerno need to drop all
15:26.52Godseyunless they're talking to 911 ;P
15:26.59ersilatzanger: me :)
15:27.01Godseydepends on how fast people are w/ redial right? :)
15:27.10tzanger"a 911 call is already in progress, please look for the panicked person and inform them of the new emergency."
15:28.04Godseythere could be more than 1 person having a problem
15:28.17tzanger"please look for the panicked parties" then :-)
15:28.33jackxlol
15:28.49Godseywhat is to happen to nufone and others now that fcc mandates they provide e911 service?
15:29.12tzangerI don't know about hte others but I'm fairly confident jerjer will have zero issue with that
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15:30.53Godseytzanger: 0 issue how?
15:31.00Godseyyou mean easily provide e911?
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15:31.24tzangerGodsey: he terminates directly to TDM himself and it's not little PRIs here and there...  so I don't think doing e911 is going to be a major issu efor him
15:32.05GodseyI'm not so sure
15:32.20Godseydont you have to connect to every ilec in every serviced area?
15:32.29Godseyand we have ss7 connection also
15:32.39Godseymainly for dial IN :)
15:32.44tzangerGodsey: I don't think so, you just need to route to the correct PSAP
15:33.16tzangerI don't think you need to be connected to every PSAP's tandem, just be able to get a call to them (even over standard links)
15:33.36tzangerI'm speaking right at the very fringe edge of my knowlege of "real" telco though so I may stop making sense at any moment
15:33.45Godseyhumm
15:33.57GodseyI thought you had to have direct connection w/ each 911 center
15:35.04tzangercould be but at least for what I do I just call the emergency line of the PSAP (the standard # that 911 routes to)
15:35.11tzangerbut I'm very much small fry :-)
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15:37.07nainHi every body
15:37.40Zeeekeeeek un nain !
15:38.17nainyes me nain
15:38.28nainwell i have little problem
15:38.56Zeeekplease state your case
15:39.09Zeeekan agent will be onlie to help soon
15:39.27nainIs it possible to recieve IP call in SIP without asterisk_oh323 channel driver
15:39.28nain?
15:40.06nainOnly in sip not h323 involvement
15:40.15ZeeekIf you use SIP you probbly don't need h323
15:40.37naingood but what if carrier is sending in h323 and my provider is SIP?
15:41.01Zeeekthen you are forced to use 323
15:41.14nainso i want that i recieve call from carrier in h323 and asterisk dial an sip account to my sip provider
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15:42.58nainAny solution Plz
15:43.26Zeeekmaybe someone can help, I know nothing about h323
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15:43.34nainthanks
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15:48.10Sparkieanyone ever setup a TDM04B before?
15:48.35Sparkienifty.
15:49.01SarahEmmSparkie: what's the issue?
15:49.21Sparkiewell.
15:49.26SparkieI got the card installed, got the module built
15:49.34Sparkienow I'm at setting up zapata.conf
15:49.46SarahEmmokay..
15:49.53Sparkieand I'm not overly certain what to put in there.
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15:50.17Sparkiei'm still rereading the manual ;)
15:50.26ZeeekSparkie there's a LOT of the wiki (if it's working)
15:50.27Sparkieyea
15:50.27SparkieThat's where I'm at right now
15:50.28Zeeeksample configs for zapata stuff
15:50.40SarahEmmSparkie: ahh. 04b is 4fxs no fxo, or the other way?
15:50.40Zeeekzaptel has about 4 lines
15:50.47Sparkiebut the only examples it lists (that i see) are for T-1/E-1 cards
15:51.10SparkieSarahEmm: the other way.
15:51.10Sparkiemy zaptel.conf is just 3 lines
15:51.17Zeeekwell that sounds great
15:51.23cluecon~seen bkw_
15:51.24jbotbkw_ is currently on #asterisk (3h 44m 21s)
15:51.24Sparkiefxsks=1-4
15:51.24Sparkiethen my loadzone and default zone
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15:51.32SarahEmmsparkie: okay, 4fxo no fxs...
15:51.38SarahEmmSparkie: okie, and ztcfg doesn't like it?
15:51.56nainHI
15:52.38ZeeekSparkie there is something for you here: http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html
15:52.38nainis there any h323 expert ?
15:52.38Zeeeka few pages and you'll be there
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15:54.03Sparkie-ride the way
15:54.03Sparkie-wave
15:54.03Sparkie-lol
15:54.16nainany h323 expert ?
15:54.53SarahEmmSparkie: what's ztcfg doing when you try using that config file?
15:55.46Sparkie-Channel 01: FXS Kewlstart (Default) (Slaves: 01)
15:55.46Sparkie-gives me 4 channels
15:55.46ZeeekSparkie here's another one that will help (save 'em for later) http://automated.it/guidetoasterisk.htm
15:55.46Sparkie-4 channels configured.
15:55.53SarahEmmsparkie: okay, what does zttool show after you do that?
15:55.57SarahEmmshould show OK on the spans
15:56.15Sparkie-whoa
15:56.15Sparkie-nice
15:56.26SarahEmmmew?
15:56.35Sparkie-No alarms, Internally clocked, then it gives me 1234
15:56.42Sparkie-and a series of transmit / receive lines
15:56.43SarahEmmok, no alarms is good
15:56.46SarahEmmso what's the problem? :)
15:56.49RoyKsplittetisplat
15:57.00Sparkie-I dunno
15:57.13SarahEmmSparkie-: okay..... it looks like it's configured right to me
15:57.19SarahEmmSparkie-: what problem are you having?
15:57.20Sparkie-well
15:57.29Sparkie-I'm not having a 'problem' really I know it's just something I haven't done
15:57.35Sparkie-to add the 4 lines into the trunk
15:57.39Sparkie-so that I can make outgoing calls
15:57.51SarahEmmahh
15:57.53SarahEmmokay :)
15:58.13SarahEmmhave you edited the zap cfg file in asterisk itself?
15:58.14SarahEmmthat'd be the next step
15:59.02Sparkie-zapata.conf right?
15:59.20SarahEmmyeppers
15:59.25ruskiwhat "give aways" should i be looking for im only hearing 1 way audio on a sip call ?
15:59.40Zeeekruski NAT on either side ?
15:59.56Zeeekruski and what phones?
16:00.20Sparkie-Ok, so if I have the 4 fxo cards on there, it's gonna start at g1 or g0 ?
16:00.38Zeeekg1
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16:01.17Sparkie-wonder if that's been my problem all this time.
16:01.34ruskiZeeek: no nat
16:02.02ruskiZeeek: sjphone and a mobile phone, im using a sip provider and have setup asterisk to recieve incoming calls from them and route them to my sjphone
16:02.36bkw_yo yo yo
16:03.53clueconmorning brian.
16:04.17ariel_hello everyone
16:04.41ariel_I guess some of us got disconnected by freenode...
16:04.42SarahEmmhihi
16:05.05nainHI every body
16:05.27Sparkie-show channels at the asterisk cli should only show channels that are currently in use right?
16:05.44*** join/#asterisk berniv6 (~berni@svr01.mucip.net)
16:05.48nainThere is a tricky question regarding SIP
16:06.00bkw_SIP tricky? NEVER!!
16:06.04SarahEmmlol
16:06.06nainyes
16:06.24bkw_guess you didn't catch the sarcasim
16:06.36nainWhat if SIP client sent only his IP not username neither secret or no registration how to treat it ?
16:06.57nainwhere this will gone in sip.conf ?
16:07.08nainbkw_: is it not tricky?
16:07.19bkw_no
16:07.26bkw_it hits the context in [general]
16:07.38bkw_all unauthenticed sip calls hit that context
16:08.07nainbut my calls are not attaching to general context
16:08.12nainthey are failing
16:08.21bkw_they are if they don't auth
16:08.35bkw_you can also match him by ip
16:08.39bkw_[bob]
16:08.42nainbut i don't need auth i just want call successful
16:08.42bkw_type=user
16:08.45bkw_host=hisip
16:08.54bkw_context=bob
16:08.58naini have done host=IP and it's context but :(
16:09.00bkw_do what I said
16:09.09bkw_you can't have a secret or username
16:09.18bkw_switch it to type=peer
16:09.44naintype=peer ??? i have set it to type=friend bcz i also want to dial it to sip provider
16:09.57bkw_again
16:10.00bkw_LISTEN TO ME
16:10.07nainOK
16:10.22bkw_its evil
16:10.23bkw_but I understand it
16:10.31bkw_if you look at the code you'll understand it too
16:10.38nainok i m listening
16:10.46bkw_you can't have ANY auth data on the entry you want it to match
16:10.53bkw_otherwise it will fail
16:11.08bkw_so you will have to have a user entry with NO username or secret
16:11.10bkw_just for inbound
16:11.12bkw_or
16:11.17bkw_make them match the general context
16:11.20bkw_which is what it will do
16:11.44bkw_unless you have a user and/or peer without auth data
16:11.46Kernel_corehi bkw_
16:11.47bkw_and host=ip
16:11.59bkw_yo Kernel_core
16:12.34bkw_go look at find_user and find_peer in chan_sip.c
16:12.40bkw_then look how they are used
16:12.49bkw_that will spell it all out clear as day if you can read and understand the code
16:13.27nainOK
16:13.30bkw_if I were on the box.. watching calls come in and looking at a sip debug
16:13.33bkw_I could fix it in two seconds
16:13.46nainsure i can give u access if u can
16:14.08bkw_but you'll never learn if I do it for you
16:14.12bkw_give it a few more tries
16:14.20bkw_if you don't get it then i'll look at it
16:14.40Kernel_corebkw_: is CVS-Nv1-0-7-06/11/05-17:43:44 the lastest version of Asterisk ?!
16:14.51bkw_LIAR
16:14.52bkw_no
16:15.02bkw_becuase that no longer had the version string in it
16:15.05naini will do it ok tell me one more thing that the context name will be ignore if host=ip define ?
16:15.24bkw_you need need context= also
16:15.41nainOK
16:15.41bkw_Asterisk CVS-HEAD built by root@shinzon on a i686 running Linux on 2005-06-11 15:58:51
16:15.47bkw_thats what the version string looks like now as of late
16:15.55bkw_you also have a new cli command 'show version files'
16:16.09bkw_that will show you the revision of all files
16:17.20Kernel_coreI use cvsup what should I put  instead of " *default release=cvs tag=." to get the latestet version of Asterisk ?! ( CVS )
16:17.38bkw_your running 1.0.7
16:17.45bkw_that is by no means the latest CVS-HEAD
16:17.50bkw_stable is a pile of shit
16:17.54bkw_you call from stable to head
16:17.57bkw_it will sound like total ass
16:18.19nainbkw_: see it is right ?
16:18.23nain[general]
16:18.24naincontext=global-calls
16:18.24nainport=5060
16:18.24nainbindaddr=0.0.0.0
16:18.24nain[no-auth]
16:18.24nainhost=207.113.96.114
16:18.26naincontext=global-calls
16:18.28naintype=peer
16:18.30nainnat=yes
16:18.32naincanreinvite=no
16:18.34naininsecure=very
16:18.36nainqualify=yes
16:18.38naindisallow=all
16:18.40nainallow=ulaw
16:18.42nainallow=g729
16:18.43bkw_turn auqlify off
16:18.44jontowdude.. pastebin.ca
16:18.46SarahEmmgah
16:18.46jontow:)
16:18.55bkw_and don't paste in here again or i'll slap you
16:18.57filebkw_: I want my Powerbook :(
16:18.57bkw_:P
16:19.00nainOK
16:19.00bkw_and I mean that with love
16:19.12bkw_file what is this you msged me about
16:19.13nainevery thing is OK instead qualify=yes ?
16:19.13*** join/#asterisk ptblank (~MURDER1@68-65-93-235.lmdaca.adelphia.net)
16:19.15bkw_me no understand
16:19.25filebkw_: and I never messaged you anything
16:19.32bkw_yes you did
16:19.34bkw_on my phone
16:19.34fileatleast not that I can remember
16:19.39fileoh that was AGES ago
16:19.43bkw_haha
16:19.45bkw_I got it today
16:19.45filewhen DNS went kaboom
16:19.48nainOK
16:19.57bkw_funny
16:21.18bkw_ouch
16:21.35pigpenok..I just ordered some of the x100p's for cheap fxo's...can one of these ports be an fxs ?  if not, any cheap fxs's out there?
16:21.47bkw_pigpen, no and no
16:21.50bkw_NEXT!!!
16:22.23*** join/#asterisk pjz (~pj@place.org) [NETSPLIT VICTIM]
16:22.25pigpenOk..so just get an IAXy or someting...
16:22.40clueconbkw_: you do plan to run around cluecon screaming NEXT!!! every 5 minutes right?
16:22.44Godseyif you're doing that, why not get SPA-3000?
16:22.46Godsey:P
16:22.58bkw_cluecon, na
16:23.12pigpenI have a 3000...it is kinda clunky since I have an asterisk box at location.
16:23.13bkw_i'm not a bitch in real life.. I just play one on IRC
16:23.15GodseyAND THEN?
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16:25.48clueconthe least you could do is run around screaming next between speakers.
16:26.00clueconit would liven it up a bit.
16:26.52pigpenanyone know how the IAXy would do with fax?
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16:29.40wooden<PROTECTED>
16:29.46woodenknows anyone about this problem?
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16:32.54eKo1Did you change from stable to head or vice versa?
16:35.42WeezeyDhow can I block my number in xlite?
16:40.18bkw_HAHAH I pissed off the neighbors
16:40.22bkw_they are mad as hell
16:40.40bkw_http://homepage.mac.com/brian.west/PhotoAlbum13.html  <-- this is them... I called da cops
16:41.49*** join/#asterisk roamer323 (~sing@toronto-HSE-ppp4075317.sympatico.ca)
16:42.21outtolunc.. /.
16:42.41outtoluncoops
16:42.45outtolunccute
16:42.58outtoluncdid you also take the wheels and tires first <G>
16:44.13bkw_haha
16:44.23bkw_they did that
16:44.28bkw_get
16:44.29bkw_toe
16:44.32WeezeyDthey did that to who?
16:44.39bkw_thats their own car
16:44.41bkw_they did that
16:44.49bkw_guess they needed to sell the wheels to make some money
16:44.50WeezeyDand they're trying to claim?
16:44.52bkw_I wish they would just move
16:44.59bkw_no they are just white trash
16:45.02WeezeyDheh
16:45.11bkw_I wish they would LEAVE
16:45.15MikeJ_so they sold used tires for cash?
16:45.17bkw_they are loud
16:45.20WeezeyDthey needed beer money
16:45.23bkw_no clue
16:45.27MikeJ_you think at that point they would see the car
16:45.28bkw_but the care is not in the road today
16:45.29MikeJ_boom
16:45.31WeezeyDmy new neighbours better not be loud.
16:45.43filebkw_ is uber-sexy in real life
16:45.47bkw_they are the only trouble people in the neighbor hood
16:46.07WeezeyDfile: and his middle name is Kelvin
16:46.19filehaha
16:46.21bkw_well they drive property value down
16:47.07MikeJ_awww
16:47.12MikeJ_it's a slow dance
16:47.19MikeJ_isn't that sooo cute
16:47.38MikeJ_:p
16:48.10MikeJ_is it hot as hell down there too?
16:48.11file:P
16:48.38MikeJ_uhhhh
16:48.45bkw_hahaha
16:48.54bkw_i've been thinkin bout you....
16:49.02filethinking about me eh?
16:49.05MikeJ_it's my high school reunion this weekend
16:49.07MikeJ_hehe
16:49.16bkw_i'll never get to go to one of those
16:49.16MikeJ_scary shit
16:49.21bkw_since I didn't actually graduate
16:49.24MikeJ_I am.. it's kinda funny
16:49.27MikeJ_soo
16:49.30MikeJ_who cares
16:49.38fileMikeJ_: you're old
16:49.43bkw_I don't care to see the people that made my teen years a living hell
16:49.45fileI care about my Powerbook!
16:49.56MikeJ_I barely did.. I was getting an a in one of the required classes, and I never did this one paper..
16:50.02MikeJ_ethics i think it was
16:50.08bkw_I quit and took my GED
16:50.29bkw_MikeJ_, neither did alot of folks it would seem
16:50.35MikeJ_and my prof said a week before graduation that he would give me an incomplete if I didn;t write it and I would not graduate
16:50.59MikeJ_my girl finished at an alternative school
16:51.26bkw_sad part is now I have to get a security camera for the house
16:51.34MikeJ_the people attarcted to this kind of project are probably all non traditional learners
16:51.40bkw_yep
16:51.48MikeJ_bkw_, that's not sad.. that's an excuse to buy toys
16:51.56Sparkie-Hrm. I keep getting a 503 error from my softphone, what have I done wrong now? :)
16:52.04bkw_MikeJ_, hahah
16:52.12MikeJ_996
16:52.14bkw_well I fear those pricks will fuck with our shit
16:52.25MikeJ_you know what you need
16:54.20drumkillaMikeJ_: you've really got to stop using outlook, haha
16:54.25drumkillayou're killing my threading :p
16:57.37MikeJ_sorry
16:57.48drumkillait's all good, just pickin' on you
16:57.49MikeJ_tis my work e-mail...
16:57.50opus__yo
16:58.14MikeJ_and the house is running a test exchange server for the office too...
16:58.18MikeJ_so I have what I have
16:58.24ariel_Sparkie-, 503 means service is not available.
16:58.38drumkillaevolution can talk to exchange ;(
16:58.40drumkillaer, ;)
16:58.57Sparkie-yea I fixed that
16:59.03MikeJ_can't do the integration with sharepoint
16:59.13Sparkie-it's not letting me pickup the outside line, just keeps telling me all circuits are busy.
16:59.36ariel_Sparkie-, are you using amp?
16:59.45MikeJ_Sparkie-, are all the circuits busy :D
16:59.49drumkillaMikeJ_: ah.  well... I don't really know what I'm talking about.  When I worked at an office that used exchange, i just used outlook.  :)
16:59.50Sparkie-ariel_: yes.
17:00.01MikeJ_:p
17:00.12MikeJ_wassup... anything happy going on today
17:00.14Sparkie-They shouldn't be busy. Granted, I only have 1 line plugged into the fxo, but no one is in the office so the phone shouldn't be on
17:00.22ariel_Sparkie-, make sure your trunks are registered.
17:00.42ariel_Sparkie-, do you have your dialing rules set correctly?
17:01.10Sparkie-I have 1 trunk setup, ZAP/g1 with 4 maximum channels, dial rules are 1NXXNXXXXXX, NXXNXXXXXX, and NXXXXXX
17:01.13MikeJ_ariel_, if he has no trunks set up, then all 0 of his circuits would be busy ;)
17:01.18Sparkie-outbound dial prefix is 8
17:01.19Sparkie-err 9
17:01.27Sparkie-and the zap identifier is g1
17:01.34ariel_also there is an area here just for amp #amportal
17:01.47ariel_Sparkie-, you have 4 fxo's?
17:02.12ariel_The default for amp is Zap/g0
17:02.17Sparkie-yes
17:02.19Sparkie-I have 4 fxo's
17:02.25Sparkie-yea ... I deleted g0
17:02.25Sparkie-lol
17:02.32Sparkie-woops
17:04.40*** join/#asterisk trasherr (trasher@dsl-084-058-015-087.arcor-ip.net)
17:04.44trasherrhi
17:04.48ariel_Oh boy it's mango season down here again. Hummm good
17:05.45*** join/#asterisk bofh42 (~bofh42@p548231DA.dip0.t-ipconnect.de)
17:06.27nainbkw_: are you there ?
17:07.03trasherri am using asterisk with sip + capi, forwarding incoming isdn calls via sip to my workstation, and outgoing sip calls through isdn... but: the called-person can't see my ISDN number. how to i configure that?
17:08.14*** join/#asterisk ptblank (~MURDER1@68-65-93-235.lmdaca.adelphia.net)
17:08.40ariel_trasherr, are you setting up your callerID when your dialing out?
17:09.50trasherrif you mean the "callerid=..." in sip.conf -> yes i do
17:10.35Sparkie-from the asterisk cli, should show channels show me something to let me know the channels are configured properly?
17:11.18ariel_Sparkie-, show channels will only show you channels in use.
17:11.48Kernel_coreNo description for sounds/digits/minus.gsm
17:11.48Kernel_coremake: *** [datafiles] Error 1
17:11.57Kernel_coreduring compile I got this ! what is wrong ?!
17:11.57Sparkie-hrm.
17:12.02*** part/#asterisk SarahEmm (~sarahemm_@Toronto-HSE-ppp3681993.sympatico.ca)
17:12.05Sparkie-welp. i'm lost lol
17:13.25ariel_Sparkie-, ztcfg -vvvv will show you what channels are configured. from the linux prompt
17:13.38Sparkie-yea, it shows me the 4 channels.
17:15.33ariel_trasherr, you can set your caller ID in your dial command look up SetCallerID on the wiki for your outbound dialing rules
17:15.43trasherrariel_: oks, thank you
17:16.33ariel_Sparkie-, ok how did you configure them in amp which file did you put the settings in?
17:17.06Sparkie-the trunks?
17:17.13*** join/#asterisk Hmmhesays (~Neg@24-117-130-121.cpe.cableone.net)
17:18.02ariel_Sparkie-, well in which files yes are you using amp 1.10.08?
17:19.12Sparkie-I'm using the AMP that came with asterisk at home
17:20.01ariel_Sparkie-, I see. ok did you do genzaptelconf -s -d at the linux command prompt to configure your zap devices?
17:20.22Sparkie-no
17:20.27Sparkie-genzaptelconf errors out
17:20.30Sparkie-I had to manually set it up
17:21.01Sparkie-the zaptel modules that came with asterisk at home caused problems under linux 2.6, so I manually built the modules and installed them
17:21.53ariel_asterisk at home is preset for 3.4 CentOS 2.6 you need to follow the udev setup. did you?
17:22.17ariel_which is linux 2.4 version
17:22.32Sparkie-yes I did.
17:22.41Sparkie-the stuff in /dev/zap is setup properly.
17:22.54ariel_can you post your zapata.conf on pastebin.ca for us to see?
17:23.02Sparkie-yea ....
17:23.03Hmmhesaysis *@home stable enough for a small production setup?
17:23.04Sparkie-uhm
17:23.06Sparkie-there's nothing there
17:23.14ariel_Hmmhesays, yes
17:23.26Sparkie-That's probably part of my problem.
17:23.32Hmmhesaysfor someone who knows what they are doing
17:23.41ariel_Hmmhesays, yes
17:23.43Sparkie-oh woops i was in zapata-auto
17:24.33ariel_Hmmhesays, they now also include a great CRM called SugarCRM.
17:24.41Sparkie-http://pastebin.ca/13969
17:24.59Hmmhesaysoh really?, i've not watched the dev very closely
17:25.38ariel_Sparkie-, ok what does the additional and the auto have?
17:25.42Sparkie-nothing
17:25.46ariel_your default group is 0
17:26.02ariel_ahh then it's not going to work then.
17:26.05Sparkie-yea. I reset everything back to g0
17:26.59ariel_you need something to say channel=1-4 if you want to use them
17:27.25Sparkie-ok
17:28.30Kernel_coremaye so stupid question , but how do I get the yesterday asterisk Snapshot ?!
17:31.40MikeJ_Kernel_core, head or stable
17:32.05*** join/#asterisk Total-Net (~pbudden@user-1121i7q.dsl.mindspring.com)
17:32.12MikeJ_cvs co -D ....
17:32.41MikeJ_followed by the date of vode you want
17:32.43MikeJ_code
17:33.03MikeJ_if you want a specific date of stable you would add -r v1-0
17:33.45*** join/#asterisk iq (~iq@65-103-165-110.omah.qwest.net)
17:33.55iqhi all
17:34.38MikeJ_high
17:34.53*** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com)
17:35.13wooden[app_capiCD.so]Jun 11 18:29:32 WARNING[10529]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/app_capiCD.so: undefined symbol: capidebug
17:35.14*** join/#asterisk Sparkie- (sparkie@24-50-118-84.lkwnny.adelphia.net)
17:35.27woodenknows anyone whats wrong?
17:36.04Hmmhesayssay hi
17:38.10Kernel_coreMikeJ_: HEAD , not stable
17:40.25*** join/#asterisk brookshire (~matt@207.111.174.1)
17:41.29drumkillai think there are nightly snapshots on the ftp site, as well ...
17:43.34Kernel_corewhere do I get this nightly snapshot ? ftp.asterisk.org ?! or digium !?
17:44.27filea Matt has been detected
17:46.32Hmmhesaysa Matt?
17:47.05filehe is a Matt
17:47.15Hmmhesaysheh, me too
17:48.44*** join/#asterisk dynaqueue (~wbevan@adsl-63-205-16-227.dsl.mtry01.pacbell.net)
17:49.13dynaqueueAnyone else running the latest firmware on a grandstream xps 2000 ?
17:51.48trasherrariel_: i tried it with setting the callerid before dialing the number.. but it doesn't work :| google can't help me
17:54.14*** join/#asterisk faqu (pn@32-76-235-201.fibertel.com.ar)
17:59.16Hmmhesaysyeah ariel_ sugarcrm looks pretty sweet
18:00.04Hmmhesaysactually I was looking for something very similar
18:03.56*** join/#asterisk cluecon (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net)
18:04.00clueconis iaxtel down?
18:05.37*** join/#asterisk ptblank (~MURDER1@68-65-93-235.lmdaca.adelphia.net)
18:08.00Hmmhesaysi do not know
18:08.06brookshirehi!
18:08.46Hmmhesayshello person of the same name
18:08.53brookshireare you a matt?
18:09.04clueconSomebody got an iaxtel number i can call to test?
18:09.11brookshirecluecon: iaxtel is always down
18:09.12woodenwhere to get a good example for capi.conf and extensions.conf for dialin and dialout
18:09.16*** join/#asterisk meppl (mephisto@p54AABBEC.dip.t-dialin.net)
18:09.22woodenisdn
18:09.22brookshirerussell has been working on it, trying to make it stable
18:09.32brookshire:(
18:10.17*** join/#asterisk Matt-E- (~Matt-E-@66-224-125-137.atgi.net)
18:11.39*** join/#asterisk ManxPower (~eric@n34.infopact.nl)
18:13.52*** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net)
18:16.09*** part/#asterisk Matt-E- (~Matt-E-@66-224-125-137.atgi.net)
18:17.06shidoboink
18:17.16Hmmhesayshey jerk
18:17.23Hmmhesayswhats up ;)
18:19.02shidohey num nuts
18:19.04shido:)
18:19.10Hmmhesayshaha
18:19.20Hmmhesaysyeah I gotta quite putting the icy hot on them
18:19.26shidoLOL
18:19.36*** join/#asterisk Xumpi (~SysOp@a81-84-69-130.netcabo.pt)
18:19.57Hmmhesaysyou know anyone using sugarcrm?
18:20.06shidowhats that?
18:20.07HmmhesaysI was tinkering around with the demo today
18:20.11Xumpihi
18:20.23Xumpiany kiax-alike for freeBSD ?
18:20.30Hmmhesayshttp://demo.sugarondemand.com/sugarcrm_os/index.php?action=Login&module=Users
18:20.32Xumpiany kiax-alike-program for freeBSD ?
18:21.09HmmhesaysfreeBSD is definately not my strong suite
18:21.37XumpiOpenBSD also's available here
18:23.00shidowow
18:23.09*** join/#asterisk ManxPower (~eric@n34.infopact.nl)
18:23.31Hmmhesaysmy employers crm software blows
18:23.38shidothis is cool
18:23.51Hmmhesaysyeah ariel pointed it out to me earlier
18:24.17Hmmhesaysit's integrated into the latest release of asterisk@home too
18:25.19HmmhesaysI find that a little bit odd but, whatever
18:25.26Xumpianyone's using voipbuster.com ?
18:28.50*** join/#asterisk cinix (~The@24-52-166-190.lndnnh.adelphia.net)
18:29.16trasherrXumpi: hmm sounds intresting
18:29.27cinixWhat's a name of an IAX2 linux client?
18:30.00Xumpigot it
18:30.01Xumpihttp://david.dw-perspective.org.uk/VoipBuster-Linux-MacOS-Solaris-FreeBSD.html
18:30.02Xumpibrb
18:30.11Xumpisee you guys l8er's
18:30.45*** join/#asterisk WilliamK (~wkeller@c-24-0-130-177.hsd1.tx.comcast.net)
18:31.35*** join/#asterisk kajtzu (~kajtzu@shell1.fi.basen.net)
18:33.27*** join/#asterisk RoyK (~roy@host-n39-135.homerun.telia.com)
18:34.33*** join/#asterisk wacky_ (~wacky@modemcable011.4-37-24.mc.videotron.ca)
18:34.39wacky_hey boys and girls :)
18:35.31wacky_I'm having a hard time configuring Asterisk with my SIP provider. When the provider sends in a call (from the PSTN network), asterisk reply with a: 407 Proxy Authentication Required, thus dumping the incoming call
18:35.42wacky_I'd like to have a little insight about how I could work around this
18:35.49*** join/#asterisk waz (~tjs@208.218.27.24.cfl.res.rr.com)
18:36.03Romik_hmmmmmmmmmm  load average: 179.91, 163.94, 136.86
18:36.16Hmmhesayswacky_
18:36.17cinixout of...
18:36.19cinixhehe wow
18:36.21HmmhesaysI could tell you the answer
18:36.28wazyikes
18:36.31XumpiDoS prolly
18:36.43Kernel_corewacky_: I have the same problem with my SIP phone to asterisk
18:36.43Xumpisome kind of fork(); bomb prolly
18:36.55wazor hitting something in inetd
18:37.06filewho wants a lesson in SIP authentication?
18:37.14fileif so, everyone else shutup and tell me when I should start speaking
18:37.15Romik_CPU Usage: u12974.4 s297.61 cu.04 cs.02 - 62.9% CPU load
18:37.16Romik_2.58 requests/sec - 37.8 kB/second - 14.7 kB/request
18:37.16Romik_135 requests currently being processed, 90 idle servers
18:37.23wacky_I read it may be caused by the 'secret' field being present, or 'auth' or 'md5secret'
18:37.37Hmmhesaysinsecure=very
18:37.44fileHmmhesays: quiet - I wanna explain why/etc
18:37.46shidodoes your provider tell you to use a secure password?
18:37.47wazanyone have asterisk working with AT&T CallVantage?
18:37.52fileoh nevermind
18:38.15Hmmhesayssorry file
18:38.18wacky_file - go ahead, come on :))
18:38.47fileIf you're using a user entry, then asterisk will challenge for credentials if it finds that the incoming INVITE is going to match against a user... that's standard SIP
18:39.02filenormally the other side will respond with the same invite, except a Proxy-Authorization header is there
18:39.07file(this is the case with phones)
18:39.30filenow, depending on how you set it up - your asterisk will do the same thing with a provider, and not all (okay... very few) providers support doing this
18:39.38fileinstead you have to match based on the IP address of the INVITE
18:39.46filethis is done using a peer entry with host, context, and insecure set
18:39.53fileinsecure=very liek Hmmhesays said
18:40.10filethis will cause asterisk to not doing any kind of authentication but simply trust the INVITE based on the source IP address
18:40.40filemmmkthx
18:40.45Hmmhesaysgood job file
18:40.57citatsughh stupid gcc3isms
18:40.57fileSIP happens.
18:41.02Hmmhesays:D
18:41.34*** join/#asterisk Pazzo (~Pazzo@host130-250.pool8172.interbusiness.it)
18:41.36HmmhesaysI figured an explanation would have fallen on deaf ears... I was mistaken
18:42.04filegiving an answer is one thing, but explaining the answer and why is another :) I'd rather you folks learn the why, then just 'here is how to do it'
18:42.33HmmhesaysI forget people actually listen in here most of the time though
18:42.46fileit's rare
18:42.56Hmmhesaysas opposed to paid support customers
18:43.01wacky_hey good, thanks file..
18:44.13Kernel_corefile: what should I do to authenticate My Eyebeam SIP Client to get authenticated with asterisk ?!
18:44.16Pete_Largothanks file
18:44.25fileKernel_core: set it up right? because you obviously didn't
18:44.52fileas I've got Eyebeam working fine and dandy
18:44.53Kernel_corefile: with this config , X-lite can connect , so it seems that my conf is OKey
18:45.13*** join/#asterisk cougar101 (Molad@adsl-69-224-94-166.dsl.irvnca.pacbell.net)
18:45.48wacky_file - only now.. my def contains type=peer, insecure=very, a host= field and a context= field
18:45.53wacky_and I still get a 407..
18:46.00wacky_now maybe I could check the IP adresses ?
18:46.20filewacky_: do a sip reload, and yes - check the IP address
18:46.26Kernel_corefile: what should I specify to get Eyebeam work ?
18:46.31Kernel_corefile: what should I specify to make Eyebeam work ?
18:46.36fileKernel_core: there's nothing special in asterisk you have to specify
18:46.37Hmmhesaysif you are using a host name to register, make sure it matches the incoming IP
18:46.45Kernel_corefile: so ?
18:46.47wacky_I reloaded, and restarted a few times, waiting for the registration to expire.
18:46.53wacky_I'll check IP addresses
18:47.04fileKernel_core: you probably have something configured wrong
18:48.39Kernel_corefile: http://pastebin.com/299002 here is my conf file
18:48.50Kernel_corefor X-Lite
18:49.00fileokay some of this is just unneeded
18:49.00Kernel_coreso what could be misconfigured ?
18:49.07Kernel_coreexample ?
18:49.12wacky_file - which field of the request must match the host= def ?
18:49.38filewacky_: the origin IP address
18:49.45fileKernel_core: http://pastebin.com/299004 use that
18:49.57wacky_oh, okay
18:51.12wacky_file - I'll explain something, see my provider requires the use of an outbound proxy.. so someone found a trick to bypass that.. by placing a def. in the /etc/hosts file and fake the whole thing
18:51.17Hmmhesaysit's odd he's get a proxy authentication required if the origin ip didn't match is registration ip
18:51.26Hmmhesaysbecause the call would just go into the default context
18:51.35wacky_Has support been added in latest releases ?
18:51.40*** join/#asterisk ska (~ska@68-234-182-91.snbrca.adelphia.net)
18:51.58*** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net)
18:52.04fileHmmhesays: almost makes me think a user entry is mucking with it
18:52.12filewacky_: no
18:52.14shidogrrr
18:52.38Hmmhesaysindeed
18:52.42skais it possible with some services to have more incoming calls that you have t1 channels for? or do you need a real voip solution to do that?
18:52.48Hmmhesayspastebin your sip.conf
18:53.06Hmmhesaysdown to where you define your proxy
18:53.08Kernel_corefile: got the same result !
18:54.15dsvanlundI am unable to call dundi_lookup from my own application module. It compiles without any problem. However when I load my module I get "undefined symbol: dundi_lookup". What could I be doing wrong? dundi.h is included, and pbx_dundi.so loaded (if that matters).
18:54.20fileKernel_core: and your eyebeam is setup right?
18:54.37Kernel_coreDisplay name : blank
18:54.41filedsvanlund: you can't, dundi is an isolated module
18:54.42Kernel_coreusername :54321
18:54.48Kernel_corepass: 4511793
18:55.01Kernel_coreand ip address of domain
18:55.05fileska: uhhhhh no
18:55.06Kernel_coreis it enough?
18:55.21fileKernel_core: fill out the rest, realm and sip proxy with your address for asterisk
18:55.34Kernel_coredoiman proxy: register with domain and use as outbound proxy
18:55.53Kernel_coreI don't see any realm
18:55.56trasherri am using asterisk with sip + capi, forwarding incoming isdn calls via sip to my workstation, and outgoing sip calls through isdn... but: the called-person can't see my ISDN number. how to i configure that? i tried it with SetCallerID before dialing, but it does not work..
18:56.02Kernel_coreI am useing the latest version !
18:56.17fileKernel_core: you're going to make me set it up aren't you?
18:56.21fileand it's going to work
18:56.41Kernel_coreif kindly you come :)
18:57.08*** join/#asterisk PakiPenguin (~pingu@202.147.163.11)
18:57.24fileworked fine
18:57.26file:P
18:57.46fileDisplay name, User name, Password, Authorization user name, and Domain
18:57.52filethat's what I filled out
18:58.01fileand checked "Register with domain" and "Use as Outbound Proxy"
18:58.17Kernel_coreI get this message in my debug : Ignoring too old SIP packet packet 1 (expecting >= 2)
18:58.17GodseyI wonder how many drug kingpins use asterisk w/ ssh tunnels for toll bypass and DEA subversion U.S.<->Columbia
18:58.25Kernel_corewhat's wrong ?!
18:58.26Godseythat is a $$ business I bet
18:58.47dsvanlundfile: OK... I believed I could use dundi_lookup if I included dundi.h, where it is declared.
18:58.52filedsvanlund: no.
18:59.10infiasteriskDOTbz: *THE* Choice of Money Launderers Everywhere.
18:59.14infier
18:59.17fileKernel_core: make sure what I told you to have set (see above) is set, close Eyebeam, and then open it
18:59.23infistupid nick completion
18:59.35*** join/#asterisk cluecon (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net)
18:59.44*** part/#asterisk cluecon (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net)
18:59.51*** join/#asterisk cluecon (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net)
19:00.05Kernel_corefile: got the same result
19:00.19Kernel_coreuseing the lateset version of Eyebeam !
19:00.21Godseynarcotel :)
19:00.26filewell I just set it up, and it works fine
19:00.52Kernel_corefile: I have near 700ms latency
19:00.59Kernel_coredoes it affect to my SIP ?
19:01.02*** join/#asterisk L|NUX (~linux@202.5.145.58)
19:01.03Kernel_coreI get this message in my debug : Ignoring too old SIP packet packet 1 (expecting >= 2)
19:01.22Kernel_coremaybe some SIP packets reject
19:01.33Godsey700?
19:01.39*** join/#asterisk at561 (~oiasdf@68.71.213-254.atlsfl.adelphia.net)
19:01.42Godseydirectway? :)
19:01.49Kernel_core:
19:01.50L|NUXevening
19:01.51Kernel_core:|
19:02.34at561is there a way to get two asterisk servers registered as peers to a third, with one operating as primary and the other as backup?
19:02.39clueconafternoon.
19:02.41GodseyKernel_core: I'm not positive but i think asterisk defaults to 300ms qualify=
19:03.03Kernel_coreGodsey: how do I change this value ?!
19:03.15Kernel_coreand where do I ?
19:03.19Godseywait for someone smart to confirm it
19:03.21Godsey:)
19:03.26wacky_does asterisk consider the /etc/hosts file while doing name resolutions ?
19:03.29GodseyI'm not so good w/ ASterisk yet
19:03.29PakiPenguinanyone here using areski calling card system??
19:03.30Kernel_coreOKey
19:03.35clueconat561: so you want server a to use b and c as gateways with failover to c if b is down?
19:03.41at561pakipenguin, what's that
19:03.43Godseywacky_: you may need to disable srv lookup
19:03.44Kernel_corefile: what do you think ?
19:03.47at561yes cluecon
19:04.00fileKernel_core: I think you're a potato
19:04.05at561access to a is minimal, too
19:04.09PakiPenguinat561, its a calling card( prepaid calling card system ) written in php and pgsql
19:04.22at561in other words changing its config would require a huge favor from the admin
19:04.42clueconsome dialplan logic might accomplish this.  i think if b is down when you attempt the dial it will return with a congested status which you could trap and then pass the call over to c.
19:04.56Kernel_corefile :and you act as salt :)
19:05.24wacky_file - is /etc/hosts considered ?
19:05.28clueconat561: that could complicate things.
19:05.30at561yeah. i'm thinking of making c try to dial b if it receives any of its calls
19:05.43at561and have them both register as the same peer
19:06.53fileyes.
19:06.58Hmmhesaysthe areski cdr thing is pretty good, I can't imagine he would have done a bad job on the cc agi's
19:08.20wacky_ok thanks guys
19:11.20*** join/#asterisk bkw__ (~brian@adsl-69-154-144-52.dsl.tulsok.swbell.net)
19:13.18*** join/#asterisk fugitivo (~ajf@168.226.246.239)
19:13.28fugitivohiu
19:16.16Hmmhesaysyo
19:17.03clueconeh
19:17.21Hmmhesaysheh
19:17.43*** join/#asterisk ptblank (~MURDER1@68-65-93-235.lmdaca.adelphia.net)
19:17.47Hmmhesaysso joan jett was pretty scary last night
19:18.53Godseynearly everything is :)
19:19.03citatsWilliamK: should be possible to do with extensions.conf processing and astdb
19:20.31Hmmhesaysi second citats
19:20.33citatshmmm, although unless theres an app to authenticate off voicemail passwords and return then that could be a sticky part
19:20.51Hmmhesaysa simple agi could auth voicmail passwords
19:20.56Hmmhesays*voicemail even
19:21.13Hmmhesaysthere's about a million ways you could do it
19:21.26citatsyeah, but thats not the elegant solution i was thinking of
19:22.13Hmmhesayshttp://www.voip-info.org/tiki-index.php?page=Asterisk+authenticate+using+voicemail+passwords
19:22.27citatssomething already?  cool
19:22.44Hmmhesaysi dunno if it's "elegant" enough for you
19:23.15Hmmhesaysi joke
19:23.24*** join/#asterisk zoa (zoa@82.103.76.147)
19:24.58ChujiAnyone fool with sphinx at all?
19:25.08ChujiDo I have train the damn thing before I can use it?
19:25.24citatsdefinitely not the elegant solution i would use, but it uses my perl module so good enough for me :)
19:25.47HmmhesaysI only go for elegant if I have time to do so
19:25.54hardwireI love you all
19:25.55hardwirereally
19:25.58Hmmhesaysif it's stable and it works...
19:25.58hardwireyou are all so very cool
19:25.59hardwireand
19:26.05hardwireI would be heartbroken if all of you left me
19:26.14Hmmhesaysbeen drinkin' hardwire?
19:26.15hardwirejust thinking about it gives me insecurity issues.
19:26.23hardwireHmmhesays: no.. too much coffee.
19:26.27Hmmhesayshaha
19:26.32hardwiremy poor brain
19:26.42hardwirehah
19:26.43Hmmhesaysi got my drink on last night... i'm moving pretty slow today
19:26.47Hmmhesaysribfest was last night
19:26.48hardwireugh
19:26.54fugitivoi don't like the quality of my gsm files, is any trick to have more quality on the files?
19:26.55hardwireI just got done being on a boat for 5 days
19:27.01hardwireand we headed to the first bar in sight
19:27.10hardwiredrank around 5 pitchers of amber inbetween 3 of us
19:27.20Hmmhesaysfugitivo: use different format?
19:27.20fugitivowith sox -r 8000, quality is really poor
19:27.24hardwireI fell over when we got back to the hotel and was drunk for the next day.. poor!
19:27.36hardwireI had a meeting with the us coast guard that day too.. ugh
19:27.38Hmmhesayshaha nice
19:27.39fugitivoHmmhesays: the default asterisk gsm files have good quality
19:27.56hardwireI got 11 last night.. thats twice as much as usual
19:27.58Hmmhesayscheck the wiki and the mailing list, i'm sure there is info there
19:28.00hardwireI am all happy today
19:28.14hardwireso
19:28.14clueconfugitivo: quality will depend heavily on the source file.
19:28.18hardwirevoice over starband
19:28.37hardwirehttp://bogomip.com/~hardwire/Starband_Install.jpg
19:28.39hardwireworks great
19:28.47clueconhow about voice over microwave?  think it'll work?
19:29.25fugitivocluecon: the source file is a studio recorded audio, great quality
19:29.57hardwireI'm the big one in that picture.
19:30.02Hmmhesaysso use the same software to turn it convert to gsm
19:30.04clueconfugitivo: what is the original format and rate?
19:30.05hardwireeverybody else is so very tiny
19:30.27Hmmhesaysawesome
19:30.55Hmmhesayshttp://www.grumpy-monkey.com/users/3853/huh.jpg <-- Me
19:31.05hardwirejesus christ
19:31.06fugitivocluecon: wav, 44100hz, 16bit
19:31.10hardwireit would be like cuddling a porcupine
19:31.24hardwirecluecon: hah
19:31.24Hmmhesayscluecon where in ok are you from?
19:31.38hardwirecluecon: I would say move somewhere other than anchorage
19:31.39clueconfugitivo: what command are you using to do the conversion?
19:31.40hardwirelike homer
19:31.52hardwireotherwise the misquitos will carry off your pets
19:31.57Hmmhesayscuddling porcupine?
19:32.00fugitivocluecon: i tried "sox file.wav -r 8000 -c 1 file.gsm"
19:32.02hardwireI just had to spray my entire lawn to get rid of all them blood suckers
19:32.38Chujicluecon : you fool with sphinx at all/
19:32.40Chuji?
19:32.42hardwireHmmhesays: whats funny is.. you fit the demographic I would concider saying things like "Monkey Poo"
19:32.48Xumpioh man
19:33.02Hmmhesayshardwire wtf are you talking about? lol
19:33.09Xumpi"You can also catch some of the developers online at irc.freenode.net, #iaxclient, or #asterisk" @ http://iaxclient.sourceforge.net/
19:33.10hardwirehttp://www.grumpy-monkey.com/index.php
19:33.16Xumpiwhere can i get the source code ?
19:33.26clueconHmmhesays: Tulsa, thunderstorms this morning (lightning struck the backyard) and now it's about 85 out and 60% humidity makes it feal like closer to a hundred.
19:33.28Hmmhesaysoh lol
19:33.30hardwireheh
19:33.48Hmmhesaysi once knew a girl from tiawah?
19:33.52Hmmhesaysi think that's spelled right
19:34.00hardwireatleast you have good taste in guitars.
19:34.27Hmmhesaysshe tore my heart out and kicked it across the floor
19:34.29clueconChuji: no, haven't made it to sphinx yet but it is on my list.
19:34.31*** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net)
19:34.56Hmmhesayshardwire: it's a epiphone, couldn't afford a gibson/gibson when I graduated highschool
19:35.08hardwireheh!
19:35.19hardwireI had a peavey nitro so don't be ashamed
19:35.25Hmmhesaysbut gibson is epiphones parent company
19:35.33Hmmhesaysso it plays very well
19:35.42clueconfugitivo: let me see what it is I use, that doesn't look right.
19:35.43hardwireepiphones aren't terrible at all
19:35.47hardwirejust upgrade the strings and be happy.
19:35.56hardwireits all in the strings
19:36.00Hmmhesaysyeah, I use erney ball on those 10's
19:36.01hardwireafaic
19:36.15hardwireI can't play a amn thing now
19:36.18hardwiredamn
19:36.27hardwireI never took lessons
19:36.31hardwirejust played and cried a lot
19:36.34hardwireit was that sad
19:36.45Hmmhesaysme either, i just played a lot.... that acoustic has seen many a drunken fire pits
19:36.46hardwirebut my friends played very well.
19:37.28Hmmhesaysgood or bad doesn't really matter, if you can play a tune that people can sing along to... everyone will think you're awesome
19:37.35hardwirehaha
19:37.36hardwireyeh
19:37.54Hmmhesayswell i've been sitting here for 2 hours... on my day off
19:38.01Hmmhesayshaha time to jet
19:38.06hardwireGOQ
19:38.08hardwireerr
19:38.09hardwirego!
19:38.19Hmmhesaysoh and just for you hardwire ..........
19:38.22hardwireanyhoot.. I have questions about speex for anybody who has mastered it :)
19:38.24Hmmhesays"MONKEY POO"
19:38.29hardwireHmmhesays: YAAAY
19:38.37Hmmhesaysok, later
19:39.35hardwirethe speex implementation in codec_speex is all vbr right?
19:40.40clueconfugitivo: i use sox filename.wav -r 8000 -c 1 filename.gsm resample -ql
19:41.21*** join/#asterisk djin (~djin@gridfox.xs4all.nl)
19:46.09woodenhow to define a default phone in the extensions.conf for all incoming calls?
19:46.33hardwireI have a house to clean
19:48.03clueconwooden: create a s extension in the default context that your calls come into.
19:48.44woodenplease give me example for it
19:51.03woodencluecon: can u give me a exten for a standard phone?
19:51.19clueconwooden: have you setup an s extension before?
19:51.24woodenyes
19:51.31woodenand it works
19:51.36woodenbut dont know it
19:51.46woodenfor standard phone
19:52.10woodenexten => *,1,Dial(SIP/snom190,,r)
19:52.12wooden?
19:52.32*** join/#asterisk mafia (pure@mother.phuck.in)
19:52.35mafiahi
19:52.38clueconwooden: yes.  put that in the incoming context for your calls.
19:52.52woodenok
19:52.58mafiaim new to asterisk, tryin to get it working
19:53.14mafialol
19:53.16*** join/#asterisk drumkilla_ (~russell@user-24-214-77-225.knology.net)
19:53.16*** mode/#asterisk [+o drumkilla_] by ChanServ
19:53.18mafiawebpage
19:53.24mafia:P
19:53.39mafiauhm
19:53.45mafiacan someone help me get it werkin?
19:54.09mafialol
19:54.11cluecondepends on what the problem is mafia.  step 1 is always read the docs.
19:54.19mafiaright
19:54.22trasherrcluecon: you're talking about an "incoming context"... how do i define something like that? i mean.. e.g. if i had a "context=capi" in capi.conf, isn't it the context for incoming AND outgoing calls?
19:54.30mafiawell i changed the one line
19:54.38mafiaexten line
19:54.46mafiato what nufone told me to
19:54.59*** join/#asterisk drumkilla (~russell@user-24-214-77-225.knology.net)
19:54.59*** mode/#asterisk [+o drumkilla] by ChanServ
19:55.02mafiabut Unable to connect to remote asterisk
19:55.05mafia;/
19:55.14cluecontrasherr: by incoming context, i mean the following
19:55.20*** join/#asterisk drumkilla (~russell@user-24-214-77-225.knology.net)
19:55.20*** mode/#asterisk [+o drumkilla] by ChanServ
19:56.38clueconon my zap lines: all incoming calls get dropped into an 'incoming' context because I've set the context=incoming line.  i'm not familiar with capi, but with zap, outbound calls are made using dial commands and the outbound context doesn't matter (at least not in my case).
19:57.05zoacontexts are not used for outgoing calls
19:57.13zoayou have two call legs all the time
19:57.22zoaso for every call you have an incoming context anyway
19:57.24clueconmafia: pastebin your cli output and your extensions.conf with sensitive info removed and i'll take a look.
19:57.30zoasip to zap -> sip context is used
19:57.40zoazap to iax2 -> zap context is used
19:57.46*** join/#asterisk drumkilla (~russell@user-24-214-77-225.knology.net)
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19:57.54clueconyeah, what zoa said...as long as he is right.
20:00.45mafiahm
20:02.18trasherrzoa: ok, i see.. thanks
20:02.53zoathanks for the question, i just added it to my astricon tutorial :)
20:03.19clueconhey zoa: think i could get a copy of that tutorial for my reference purposes?
20:03.55*** join/#asterisk drumkilla (~russell@user-24-214-77-225.knology.net)
20:03.55*** mode/#asterisk [+o drumkilla] by ChanServ
20:04.07zoaof course
20:04.14zoait will be online after the speech
20:04.59clueconthanks zoa.  just point me in the right direction when it's up.
20:05.19clueconif i thought i could get back in the country and had the money, i would be at astricon.
20:05.43zoait will be at asteriskguru.com
20:07.18*** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net)
20:07.23*** join/#asterisk mtgh (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net)
20:08.58clueconbookmarked. thank you.
20:12.51trasherrcould anybody say me, why i can't login to sipgate with asterisk? Jun 11 22:12:14 NOTICE[3580]:    -- Registration for '4213871@sipgate.de' timed out, trying again
20:14.20*** join/#asterisk syle (~blah@wnpgmb06dc1-43-135.dynamic.mts.net)
20:14.58*** join/#asterisk ptblank (~MURDER1@68-65-93-235.lmdaca.adelphia.net)
20:16.16*** join/#asterisk bjohnson (~bjohnson@66.11.165.163)
20:16.41shidofinally
20:16.43shidosome rain
20:17.29PakiPenguinsend us some too shido!
20:18.03clueconwe got slammed yesterday and early this morning.  Not cool when your wakeup call is a lightning bolt in the backyard.
20:18.12shidohehehe
20:18.18shidowe desperately needed some here
20:18.21shidoit was HOT!
20:18.23mtghshido: are you in MI
20:18.26shidoand I have no working A/C
20:18.30shidoyes
20:18.36clueconwe needed it too. but it didn't help the heat.
20:18.36mtghshido: So am I, where are you?
20:18.44PakiPenguinits not hotter then where i am
20:18.46shidoDetroit
20:19.14PakiPenguinThe current temperature in Multan, Pakistan is 91°F (11:00 PM PKT on June 11, 2005). Conditions: Blowing Sand. Humidity: 28%. Dew Point: 62°F. Pressure: 29.55 in 1001 hPa. Visibility: 0.0 miles 0.5 kilometers.
20:19.46citatsshido: you living here now?
20:19.58mtghConditions: Blowing Sand .... nice
20:20.12shido"here" ?
20:20.16PakiPenguinyeah , we had a furious sand storm earlier
20:20.18citatsmichigan
20:20.21shidododo
20:20.28shidoI've always been a michigander
20:20.42citatsthought you were from canada
20:20.45shidono
20:20.48clueconPaki: we are close, would be pretty equal if we hadn't started the day with rain.
20:20.48shidoI visited Canada
20:20.55shidoand got stuck
20:20.58shidodue to sept 11
20:21.35shidowhen we have voip issues
20:21.42shidoI head over to our secret location
20:21.56shidoand reboot a few things or get the crash cart and fix0r them
20:22.21citatsshido: i don't have any gear at that place anymore, so i wont ever see ya there
20:22.33shidowhere are you citats ?
20:22.43citatsi live in howell for now
20:23.12shidowhy'd u move your gear?
20:23.46citatsshido: i was laid off by the company that had gear there, and i moved my personal cabinet to a diff location
20:24.22shidocitats, you know you coulda just asked us
20:24.47shidoour cabinets are relatively empty - due to jerjer's Linux Kung Fu
20:25.08shidoif we filled them we could take over the world finally - but he keeps tellin us "in due time"
20:25.23citatsshido: i was paying for the one there.  i moved my own stuff to another location where i get the cabinet comp'd for helping out every once in a while
20:26.02shidowell we have a ton of bandwidth at each and every one of our locations
20:26.29PakiPenguinanyone care to look at this http://pastebin.ca/13983 <-- i am using areskicc , giving me channel status : 6 , from the agi and then exits
20:26.49shidotalk to areski, PakiPenguin he's very helpful when u drop him some cash
20:27.01PakiPenguin:) alright
20:28.19shidotell him shido6@gmail.com sent you
20:28.50shidoHostans friend
20:29.25shidoengrish not so good but understand him you will, yes
20:29.50shidohell, my engrish isnt so good either
20:30.01asterbeastwhat port do i need to open in my firewall to let an X-lite softphone connect with my PBX?
20:30.11zoalots of them :)
20:30.55X-RobI'd actually say 'most'
20:30.57X-Rob8)
20:30.58hardwireanybody found a good voice processing card for non circut interfacing?
20:31.11asterbeastwhy?
20:31.23shidohardwire, softwire
20:31.25asterbeast5060 udp is for sip ok...
20:31.29shidono need for hardwiring
20:31.42shidohardwiring will cost you $$$$$$
20:31.45hardwireshido: its what I doooo :)
20:31.50shidohehehe
20:31.56zoahttp://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html
20:31.59zoaread this tutorial
20:32.03shidohow many simultaneous calls, hardwire ?
20:32.08zoait might explain something
20:32.11zoasuch as, use iax2
20:32.13zoa:)
20:32.23hardwireshido: no idea.. I just need to work on low power voip systems
20:32.43hardwireso I need to use 266mhz hardware for the most part
20:32.52shidono problem
20:32.53shidohow much ram
20:32.55shido?
20:33.12hardwire128 :)
20:33.16hardwireactually.. 64
20:33.18hardwirehah
20:33.19shidofor 60 calls?
20:33.25X-Robhardwire, look at the VIA EPIA boards, they use bugger all power, and you can get up to 1ghz
20:33.29shidowhat else do u need it to do?
20:33.33hardwireX-Rob: I already use them
20:33.42shidojust calls
20:33.43shidono ivrs
20:33.46shidono musiconhold
20:33.50hardwirewell
20:33.50shidono db?
20:34.01hardwireI need voice prompts
20:34.01hardwireand moh
20:34.05hardwireand usb for timing for meetme
20:34.15shidohrmm
20:34.19shidour gonna have to cut the kernel
20:34.28hardwirewith scissors?
20:34.32shidoand build your own distro
20:34.35hardwireI already have
20:34.40shidono not with scissors
20:34.42shidowith "rm"
20:34.43shido:)
20:34.47hardwireits called bulletdebian :)
20:34.49shidoand vi
20:34.54shidobulletdebian
20:35.01hardwireI need to make a homepage for it
20:35.08X-Robhardwire - USe 2.6 and use RTC.  No need for USB
20:35.15hardwireX-Rob: ztrtc?
20:35.15shidotell me more about bulletdebian
20:35.19shidoI'm running ubuntu here
20:35.26hardwireshido: what I did was pretty simple
20:35.43hardwirecustom kernel based around debians build system for kernels.. targetted for the geode processor
20:35.53hardwireor any target processor.. even Via C3
20:36.02X-Robhardware - cvs ztdummy.c has support for using 2.6's RTC, rather than USB
20:36.07X-Robis much MUCH more accurate
20:36.08hardwireI have a build system for the distro that has a target drive..
20:36.14hardwireit makes a boot drive
20:36.15X-RobReal Time Clock
20:36.23hardwirea squashfs for the root that is partition 2
20:36.29hardwirepartition 3 is the backup storage
20:36.33hardwirepartition 3 is the unionfs mount
20:36.38*** join/#asterisk Sparkie- (sparkie@24-50-118-84.lkwnny.adelphia.net)
20:36.39hardwire4 I mean
20:36.48hardwireevery boot it populates partition 4 with partition 3
20:36.51hardwireand mounts that on top of /
20:37.05hardwirea script rsyncs p4 to p3 as perminant storage
20:37.11hardwirehence.. bullet"proof" debian
20:37.15asterbeastzoa: I want to connect to my pbx from the internet...
20:37.18Sparkie-How do you make asterisk answer the phone faster ... currently it's waiting 2 rings to direct my call to the digital receptionist. Is there a way to speed that up to 1 ring?
20:37.26hardwireyou can disconnect its power at any time and no problems occur
20:37.44X-RobSparkie - turn off caller ID detection
20:37.49shidoSparkie-, dialplan logic
20:38.18X-RobIT's waiting for caller ID to be sent before it answers. it waits ~10 secs before it gives up and answers
20:38.38Sparkie-k
20:39.47hardwireugh
20:39.51hardwirenever drink blue sky energy drink
20:39.57trasherrdoes anybody know a sip program (windows), where i can answer a call with a hotkey?
20:39.57hardwireit tastes like pepto
20:40.22shidoheh
20:40.31X-RobAustralian Mountain Dew tastes like Diesel.
20:40.35X-RobIt's _horrible)_
20:44.12bkw__muhahha
20:44.17bkw__my switch kit ROCKS
20:44.51mafiaswitch kit ?
20:45.12zoai just calculated that an asterisk server could cost you 1.7 million euro per 24 hours if hacked
20:45.22zoaif its has 2x4 e1 cards
20:45.29*** join/#asterisk zotz (~zotz@208.196.247.140)
20:45.40kajtzuhow do you come up with that number?
20:46.13zoacalls to inmarsat are 5 euro per minute
20:46.28*** join/#asterisk cluecon (~asterisk-@adsl-70-244-228-14.dsl.tulsok.swbell.net)
20:47.11kajtzuright. you're still gonna have a problem calling someone over inmarsat with 240 channels 24x7
20:47.26*** join/#asterisk outtolunc (outtolunc@adsl-69-110-5-162.dsl.pltn13.pacbell.net)
20:48.00zoayeah true
20:48.01zoa:)
20:48.02zoabut still
20:49.02zoasome belgian toll numbers would be half that price
20:49.05zoaand thats possible:)
20:50.18kajtzuhmm last time I used inmarsat was in a comms room on a cruise liner ;)
20:52.22zoai never used it
20:52.28zoamust be extremely expensive to call with it
20:53.37kajtzumortals use mtn or one of the other satellite bandwidth providers
20:53.56kajtzuinmarsat tends to be company business or emergency use
20:54.04kajtzu(mortals = paying customers)
21:03.15*** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com)
21:10.48bkw__OMG BECKY realtime OWNZ
21:11.53*** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net)
21:11.56zoabrian
21:11.57zoamr mr mr
21:12.59*** join/#asterisk ptblank (~MURDER1@68-65-93-235.lmdaca.adelphia.net)
21:24.12*** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk)
21:28.16*** join/#asterisk loick (~loick@APuteaux-151-1-28-174.w82-124.abo.wanadoo.fr)
21:31.44WilliamKanyone know howto get the Verticle Service Activation codes to work with *?
21:32.01X-RobWhich one would you like to work?
21:32.49WilliamKhow about all of them? =)
21:32.55WilliamK*67, 82, etc..
21:32.55WilliamK=)
21:33.02X-Rob'all' of them are usually implemented by your handset
21:33.15X-Robany specific one I might be able to hint at using with a dialplan
21:33.31WilliamKSipura has it in there, however it gives a busy signal right after and sends nothing to *
21:33.40shidohehe
21:33.44shidodialplan logic WilliamK
21:34.07WilliamKgotta create a dialplan per service that I want to enable then? =)
21:34.09X-Robeg, call waiting is a function of your phone, and of asterisk
21:34.15shidoerrr
21:34.24shidojust dialplan entries
21:34.24X-Robasterisk can either not send a call and redirect it to voicemail
21:34.31X-Robor it can send a call and get a busy back from your phone
21:34.37WilliamKX-Rob, I'm speaking about outbound services, not the inbound ones =)
21:34.39shidonot an entire plan for each service unless you are creating your services that way - to enable and disable services, etc
21:34.41X-Robor it can send a call and your phone can generate call waiting tones
21:35.06WilliamKI want the user to be able to like hit *67 and make it work
21:35.08WilliamK=)
21:35.11X-RobWhat's *67 meant to do?
21:35.23WilliamKdisable callerID per call outbound
21:35.35WilliamK*82 does it perm.
21:35.41X-RobThat's a phone and asterisk function again
21:35.59X-Roblook at AMP's dialer agi
21:36.08X-Robit uses astdb to figure out wether to send caller ID or not
21:36.16WilliamKah
21:36.24X-Robbut most phones can handle not sending caller id
21:36.44WilliamKthe sipura should be able to, not sure why it's giving back a busy signal
21:37.04X-Robprobably coz it's not enabled
21:37.08X-RobRTFM?
21:37.24X-Robmy 2000 does it
21:37.27X-RobI think
21:37.29X-Rob(checks)
21:37.41opus__hi x-rob
21:37.41WilliamKI am using a 2000 and it is enabled
21:37.51blitzragezoa: have you seen the hotel that Astricon Europe is at?
21:37.55X-RobWell, it gave me a dial tone, but it still sent caller ID
21:38.22X-RobI dialed *67, got a dialtone, dialed the GXP2000, and I got caller ID
21:38.38X-RobDon't know why.
21:39.25blitzragezoa: http://www.hotelauditorium.com/
21:41.25zoayeah i know it
21:41.28zoawill be cool
21:41.37zoai hope im not speaking in that very big room :)
21:42.01*** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-75-202.cust.bezeqint.net)
21:43.55WilliamKokie, I factory defaulted my sipura and now it dials without the busy but gives the callerID
21:45.08skaSorry, I missed the respones to my question about incoming calls.. Is it possible to recieive more incoming calles than you have t1 channels for?
21:46.36zoablitzrage: see private, fast i need to go to bed
21:46.50clueconska: the number of calls will always be limited by the number of available channels.
21:46.55Blissexska: depends what you mean by «receive»
21:47.34skaBlissex: i mean incoming calls to a set of finite numbers.
21:48.10Blissexska: thats not that useful
21:48.39Blissexska: for example, if you get calls routed to you over IAX2, of course many many more.
21:49.22zoaall extra calls will give you busy
21:49.23clueconBlissex: he is asking bout T1.
21:49.27skaBlissex: so if we use IAX we are only limited by available bandwidth?
21:49.45zoabandwidth and cpu
21:50.03Blissexska: if you mean routed to you over POTS or ISDN lines, the routing telephone company cannot route more than N, and will return some ''system busy'' to the caller.
21:50.10skazoa: good point. Never thought about cpu limits..
21:50.26*** join/#asterisk apolinux (~carlos@ppp-105-36.telesat.com.co)
21:50.35clueconska: T1 is limited to either 23 or 24 channels.  Practically everything else is strictly limited by the power of the machine and available bandwidth.
21:50.40skaAre there any decent estimates for cpu/call numbers?
21:50.40Blissexska: CPU limits matter in particular if you use some compressed scheme... But not just because of that.
21:50.47apolinuxehlo pe0ple
21:51.08Blissexapolinux: you speaking SMTP or just a typo? :-)
21:51.52apolinuxjejejeje yes..
21:53.28skaI suppose there are special cards with compression and codecs that can speed things along.?
21:53.50apolinuxsomeone tried to configure a voip like  sua <-> asterisk <-> gnugk <-> gw <-> pstn , to call from sip phone to pstn?
21:54.32Blissexska: I think Digium themselves sell something like that.
21:54.45Blissexska: but it can be also ''just buy a bigger PC''.
21:54.53zoaska, no such cards exist for asterisk (yet)
21:55.02Blissexska: also it is not that difficult to use multiple PCs.
21:55.06apolinuxi'm talking alone.. <:-|
21:55.20zoaapolinux: i did before
21:55.29zoabut there is no good h323 implementation for asterisk
21:55.33Blissexapolinux: be patient...
21:55.34zoai could not get anything stable
21:55.37zoaand i gave up
21:55.41zoaafter months of trying
21:55.56Blissexapolinux: just dtop H323 it is the spawn of the devil and part of the plan for apocalypse...
21:56.08apolinuxi make it works..
21:56.11apolinuxbut
21:56.12skablissex: like in high availability setup with multiple * systems?
21:56.40apolinuxonly the voice goes one way..from sip to pstn
21:56.48blitzragezoa: replied to PM
21:56.55Blissexska: «high availability» is a bit of an ambitious phrase.
21:57.03apolinuxno..i mean from pstn to sip
21:57.28apolinuxbut doesn't works from the sip to pstn..
21:57.31skaska: perhaps simple load balancing?
21:57.33Blissexska: but for example if you setup a nice shared user database, and route calls between the various PCs, its not that hard.
21:57.50apolinuxzoa: which software do you use it?
21:58.28zoai tried everything available
21:58.30zoachan_h323
21:58.33zoachan_oh323
21:58.37zoaand chan_ooh323
21:58.38apolinuxjejejejej
21:58.41apolinuxhm..
21:59.02apolinuxwhich problems do you had?
22:00.21apolinux:-O ...
22:01.06X-RobGive up on h323
22:01.18X-Robas was recently pointed out on asterisk-users, it causes cancer.
22:03.52bkw__h323 works
22:03.53bkw__chan_woomera works
22:04.07bkw__the one in asterisk-addons works
22:04.09bkw__kinda
22:05.03zoabkw, i had bad luck with it a while ago
22:05.06zoawith the ooh323
22:05.20zoabut maybe it was the carriers fault
22:05.22bkw__it works and makes a call
22:05.28bkw__I called from it to woomera
22:05.29bkw__worked
22:05.35bkw__woomera to it worked
22:06.02apolinuxzoa..
22:06.08tzangerbkw__: did we ever figure out a definitive answer about what entailed "linking" to Asterisk?
22:06.17apolinuxwhat is the ooh323? where i can get it?
22:06.23bkw__no but i'm 100% sure agi and talking to the manager does NOT count
22:06.43tzangerfiles don't explode
22:06.51tzangerbkw__: yeah that's the common sense answer :-)
22:06.52bkw__depends on the file system
22:07.14bkw__file my dear
22:07.20bkw__you wanna test my new stuff I put up today?
22:07.26zoawhat what what
22:07.29zoagimme the goodies
22:07.32fileohhhhh not that
22:07.39bkw__its a service goodie for asterlink customers
22:07.50bkw__hehe
22:08.07zoaand what is it ?
22:08.07filebtw, did you try put international into the standard routing stuff, or do I need to add a service flag to the account?
22:08.16files/try//
22:08.25bkw__file, the way it is now you don't have to bother with rating calls or billing records
22:08.28zoaim going to be dnow
22:08.31bkw__thats still done on the asterisk side
22:08.36fileyeah i know that
22:08.40fileI just meant for my regular calling :P
22:08.47bkw__ah call 996
22:09.48*** join/#asterisk yaaar (~todd@12-216-227-245.client.mchsi.com)
22:09.49yaaarword
22:10.09*** join/#asterisk ptblank (~MURDER1@68-65-93-235.lmdaca.adelphia.net)
22:10.15*** join/#asterisk diablopico (~russ@ip68-8-227-7.sd.sd.cox.net)
22:10.41diablopicoi need help
22:13.12diablopicois anyone able to help with a h323-gateway config with 2 digium/zaptel e1 cards /connecting with CAS/r2
22:15.03Hmmhesaysr2 yuck
22:23.54SwKshit shit shit
22:24.18SwKsomething busted my box to hell in a head check out from about 20 minutes ago
22:24.52SwKiax no worky work... dont even get debug output hah
22:24.57*** join/#asterisk fiber0pti (~john@209-234-169-232.gen.twtelecom.net)
22:25.25fiber0ptiWhat kind of tools does Digum offer with their "Asterisk Business Edition?"
22:27.26drumkillanothing additional
22:27.37SwKyou just get to pay for it
22:27.50drumkillait's a commercially licensed and supported version of Asterisk
22:27.53SwK(well I think they throw in some support to)
22:28.42SwKdrumkilla what all support do they provide under that lic?
22:29.03bkw__drumkilla, we need a RAND function
22:29.08fileomg, HI
22:29.14bkw__I need a random number between 1 and 7
22:29.20bkw__and no real good way to do it in the dialplan
22:29.23drumkillabkw__: I think there is an app_rand
22:29.31bkw__that will not give me a result
22:29.35*** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net)
22:29.35bkw__it will only branch
22:29.55drumkillawrite a rand dialplan function
22:29.58tzangerI love getting "deleted your email because it contained sensitive words" emails
22:30.08bkw__tzanger, who sent that?
22:30.14tzangerwhat's the matter, can't take me saying "fuck" or "bitch"
22:30.18tzangerdunno
22:30.22tzangersomoene subscribed to -users
22:30.34tzangermy ex's old job had the same kind of fucked up filter
22:30.53HmmhesaysI got no filter at work
22:30.55tzangerI can certainly understand it for OUTGOING mail, but to block customer's email without notifying the person it was intended for is just as bad if not worse
22:31.21WilliamKanyone know about the outdated modules warning messages? I keep seeing them show up at random when I recompile
22:31.26asterbeastdoes anyone have experience with handling firewall ports trough asterisk?
22:31.27Hmmhesaysmy email says I can add 4 1nch3s to my p3n0r daily
22:31.43SwKbkw_ you know what I dont even see it sending a register packet in tcpdump
22:32.08SwKHmmhesays d00d sure fire way to get in to the guiness book of records!
22:32.39asterbeasti would like to open or close ports in my firewall with a script. when i call asterisk with my mobile.
22:33.09SwK"Hmmhesays: man with the worlds largest p3n0r... with a t00l that measures 56 inches!"
22:33.23bkw__WHACK
22:33.24SwKasterbeast: so write an AGI for that
22:33.26bkw__is iax broken?
22:33.28HmmhesaysSwK LOL
22:33.30Syncrosmoose penis !
22:33.37Hmmhesayswhere!
22:33.49fileHmmhesays: that's sexy
22:33.50SwKbkw__ I dunno
22:34.31SwKbut i can get calls out now just by adding in app_math which i commeneted out cause it would build... it has some f'd up error w/ gcc 2.95 in  if(!foo){
22:34.44SwKgcc repoprts it as a syntx error heh
22:35.04asterbeastSwk doenst anybody already did something like that?
22:35.12tzangerasterbeast hahaha love the nick
22:35.18*** join/#asterisk ToyMan (~stuq@user-12lcqur.cable.mindspring.com)
22:35.19SwKasterbeast: dunno
22:35.27Hmmhesaysstrange conversations in here sometimes
22:35.38tzangerHmmhesays: you don't know the half of it :-)
22:35.57SwKhah
22:35.58shido:)
22:36.07tzangerpower flicker?  wtf
22:36.09Hmmhesaysi can only imagine
22:36.15SwKbkw__: tcpdumping IAX port dont show shit
22:36.20tzangertoo bad I'm not gonna be here for this storm
22:36.37SwKbkw__ till I dial out hah
22:36.43asterbeasttzanger thanks!@
22:36.48HmmhesaysIt just rained hardcore here, I"m going to take pictures of the river
22:36.58tzangerwere's here?
22:37.04Hmmhesaysfargo/moorhead
22:37.05SwKits been raining here for like 3 days
22:37.14Hmmhesaysthe river is the red river of the north
22:37.27SwKmy yard looks like ass cause the yardboy wont come and cut it if it looks likes its gonna rain today
22:37.34SwK(and its looked like that for about 2 weeks now)
22:37.43tzangertell him to get his ass over and mow it BEFORE it fucking rains
22:37.46Hmmhesaysmost of the streets here are flooded
22:38.48SwKi'm just gonna go buy the wife a lawnmower and say "get out there and cut the grass bIatch
22:39.02tzangerSwK: heh
22:39.04tzangerI was out mowing today
22:39.07tzangerabout 1/2 done
22:39.44SwKmy yard wouldnt be that bad... cept for the damned johnson grass all over the place... that shit  grows like foot tall in a week
22:39.48*** join/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net)
22:39.49xbmodderI hope you mean mowing the lawn
22:40.03tzangersounds like my johnson, now I know how it got its name :-)
22:40.05xbmodderlol
22:40.16tzangerit's up before you realize it
22:40.31SwKtzanger you on that same p3n0r plan as Hmmhesays?
22:40.36tzangerhahaha
22:40.39xbmodderbash.org?
22:40.45tzangeryeah mine's on consignment though
22:40.52tzangerI go by and visit it every day
22:40.58SwKhah
22:41.15tzangerer
22:41.17tzangernot consignment
22:41.19tzangerlayaway
22:41.26tzangerdammit I hate hwen I fuck up a good joke
22:41.49SwKtzanger: does that mean "it's in tzanger's wifes purse until she lets you visit it"?
22:42.10tzangerno, I'm wifeless.  Had one, she's WAS on consignment but someone fially picked her up
22:42.17SwKhah
22:42.18tzangeronly she got 50%
22:42.19Hmmhesayssomeone want to sponsor me for cluecon? i just found out work can't pay it
22:42.34tzangerHmmhesays:  get in line, bitch.  I'm first in line for sponsorship :-)
22:43.10Hmmhesaysthey said "we have enough to pay transportation, from old airline miles"
22:43.37Hmmhesayssad thing is, it's only like 600 miles away for me
22:44.04tzangerdamn
22:44.13tzangerwill they spring for half?
22:44.21Hmmhesaysi'm going to see
22:44.26tzangeranyway gotta go
22:44.27tzangerlater all
22:44.30*** join/#asterisk cjk (~cjk@80.92.75.120)
22:44.31Hmmhesaysyeah same here
22:44.34shidohrmm
22:44.39shidoHmmhesays, where are you?
22:44.42Hmmhesaysfargo
22:44.51shidonow for the cheap seats
22:44.53shidowhere is fargo?
22:45.02Hmmhesaysright next to moorhead mN
22:45.06Hmmhesaysfargo ND
22:45.26cjkhi, lets say i need some small custom asterisk things to be changed in the source code? who could i contract? is there a chance that they will be put into the asterisk tree? anyone interested, mostly basic things...
22:45.29Hmmhesaysi'd drive there no problem
22:45.41Hmmhesayswe just restructered though and money is really really tight
22:46.48HmmhesaysI gotta run, BBIA
22:50.31*** join/#asterisk outtolunc (~me@adsl-69-110-5-162.dsl.pltn13.pacbell.net)
22:58.58harryvvreally tight mmm
23:00.23*** join/#asterisk cmk (~cmk_@p54A3FB87.dip.t-dialin.net)
23:01.57*** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv)
23:02.02diablopicois anyone able to help with a h323-gateway config with 2 digium/zaptel e1 cards /connecting with CAS/r2
23:03.28diablopiconobody ?
23:05.13shidoyeah
23:05.17shidowhats up diablopico ?
23:05.28shidofrustrated?
23:05.31*** join/#asterisk jerlique (~jerlique@lnk254.adl0.adsl.esc.net.au)
23:10.12diablopicowhat happened shido
23:12.58harryvvanyone here know where voipsupply is located at?
23:16.03X-Rob716 area code
23:16.06X-Robwherever that is
23:17.47X-RobBuffalo NY
23:18.10harryvvmm thats really to far from the pacific nw.
23:18.26harryvvso voipsupply and atacomm are the two major phone resellers.
23:18.28X-Robspecifically, 454 Sonwil Drive, Buffalo NY
23:18.33X-Robgotta love google.
23:21.10hardwireyou ever pick up the phone
23:21.17hardwireand somebody is just heavy breathing at you
23:21.28hardwireI swear to god it wasn't me
23:23.33hardwireit was him.
23:25.36X-Robwho, him?
23:26.18hardwire:)
23:26.28hardwirethe commercials in anchorage are the worst
23:26.37hardwirethe airtime must be incredibly cheap
23:26.38X-RobAhh. TV
23:27.23X-Robhttp://www.nodedb.com/australia/qld/gladstone/?
23:27.27SedoroxWhat is this... T.V. You speak of ?
23:27.27Sedorox:p
23:27.33X-RobZoom out from there and you'll see where I live
23:27.55X-RobTV ads here are so cheap that there are at least 3 companies that have ads made in the middle 70's still on TV
23:27.58X-Robcurrently.
23:28.17hardwirehaha
23:28.28hardwirewe have a guy here that has a place called the matress ranch
23:28.30hardwirehe sells beds
23:28.37X-RobWe changed our phone numbers from 049-72-xxxx to 07-4972-xxxx about 7 years ago
23:28.39hardwireand he has some of the worst tv ads ever made
23:28.51X-Robtwo of those ads haven't even bothered updating the phone number on their ads
23:28.56hardwirewow.. you are way the hell up there
23:29.11X-RobI'm 42kms south of the tropics
23:29.23hardwireI should come visit :)
23:29.31X-Robfeel free!
23:29.33hardwirewe will go 4ms north
23:29.34hardwireerr
23:29.36X-RobWe have a 6br house 8)
23:29.39hardwire42kms
23:29.43hardwireholy crap
23:29.59X-RobUh
23:29.59hardwiredid it grow from the ground up?
23:30.02X-Robit's not a mansion
23:30.05X-Robit's a highset place
23:30.07X-Robbuilt in underneath
23:30.08X-Rob8)
23:30.33X-Robwhere are you?
23:30.39hardwirenowhere near you
23:30.44hardwireI am in Alaska
23:30.48X-RobAhha.
23:30.55X-RobCold. Mooses.
23:31.00hardwirethe mooses haunt me
23:31.07hardwirewe are trying to escape this weekend actually
23:31.11X-Robdamn mooses.
23:31.12hardwireand have no idea where to go
23:31.38X-Robit's a 14hour flight from LAX to sydney
23:31.47X-Robthen 3 hours flight north
23:32.02hardwirethere goes the weekend?
23:32.04hardwireerr
23:32.05hardwire!
23:32.08*** join/#asterisk Cybertoy (user@ool-4357da9f.dyn.optonline.net)
23:34.19X-Robheh
23:34.28CybertoyI'm implementing the telemarketer torture... anyone happen to have a nice women's voice prompts for the example in voip-info.org ready?
23:34.42Cybertoyotherwise I'll use my own unsexy voice... :)
23:35.14X-RobYou guys have -no- idea how good it is to not live in the US, really.
23:35.26hardwireoh don't start
23:35.27X-RobI can't remember the last time someone tried to call me and sell me shit
23:35.44Cybertoyactually I do have an idea... I just moved to the states from Switzerland.
23:35.48hardwireit helps living in alaska
23:35.54hardwireit costs 5 times as much to call us
23:35.56hardwireso nobody does
23:35.58Cybertoyand once in a while some monkey would call me there to sell me a newspaper...
23:35.59X-RobCybertoy - is it as bad as you expected?
23:36.20*** join/#asterisk |Vulture| (~V@c-69-180-67-228.hsd1.fl.comcast.net)
23:36.29Cybertoythe telemarketers here are worse... but my own phone number is not spoiled yet... but the one in the corporate appartment I'm in now is.
23:36.49Cybertoyactually ... the other day someone called on my swiss voip number and wanted to sell a newspaper... I was talking to him for about 15 minutes...
23:36.52hardwiresell sell sell guys
23:36.53*** join/#asterisk file[desk] (~jcolp@mctnnbsah25-142166147147.nb.aliant.net)
23:36.56Cybertoyand then wanted to give him my address...
23:36.57hardwiredon't you know the US rule of thumb?
23:37.04Cybertoyhe didn't like my us address...
23:37.16X-Robheh
23:37.17Cybertoyand was pretty annoyed that I wasted 15 minutes of his time.
23:37.17|Vulture|Cybertoy: who do you have a swiss DID from?
23:37.23Cybertoywww.sipcall.ch
23:37.34hardwireCybertoy: a friend of mine forwarded to his sat phone from his home phone
23:37.38hardwireand he was in Iran
23:37.41hardwireand he did the same thing
23:37.48hardwirethe newspaper salesman was pretty torqued
23:37.49Cybertoymy old phone company was nice enough to make an announcement on my previous phone number to call the new swiss did...
23:37.52|Vulture|wow... can't read that
23:38.07Cybertoyvulture, it's german...
23:38.14hardwireyerman
23:38.20Cybertoyvulture, but try www.netphone.ch ... there's a bunch of links there to others.
23:38.50Cybertoysorry... that should be www.nettel.ch
23:38.57hardwireCybertoy: no LNP?
23:38.57|Vulture|Cybertoy: okay cause I know one of my clients would love a swiss DID
23:39.25Cybertoyvulture, nettel lists a bunch of them... maybe one of them is in english ...
23:39.41hardwireengrish
23:39.53shidou don like my engrish, boi?
23:40.00Cybertoyvulture, if you want though I can talk you through setting up sipcall.ch ...
23:40.11hardwireugh
23:40.13hardwireI need to lik
23:40.15hardwirefall over
23:40.17hardwireand sleep
23:40.21|Vulture|How do those voip providers send 911 different data? I thought that was via the hardware location
23:40.47Cybertoynow why would a swiss did provider worry about 911 ??
23:41.08|Vulture|no I got a link from there
23:41.15|Vulture|and it was talking about 911 calls
23:41.16*** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com)
23:41.23|Vulture|nikotel
23:41.58Cybertoydunno them...
23:42.07Cybertoyuse sipcall .. they'll give you a swiss did for free
23:42.24|Vulture|I use VPC and Asterlink
23:42.33|Vulture|but neither offer DIDs in remote areas
23:44.07Cybertoywhat's vpc?
23:44.10RoyK~seen inspired
23:44.11jbotinspired <mikael@213.197.167.61> was last seen on IRC in channel #asterisk, 8d 10h 18m 50s ago, saying: 'if it's made at the wood alcohol factory, you'll go blind ;)'.
23:44.15|Vulture|voicepulse connect
23:44.18Cybertoyah
23:45.09Cybertoywell ... the offer stands... I can guide you through sipcall if you want. the setup is free. no need for credit-card .. nothing ...
23:45.48|Vulture|nice, thanx Ill see if he wants it right now he can just dial but I am sure he will want to have a DID
23:46.22Cybertoythat's what you get
23:47.52Cybertoywww.voipbuster.com is nice to make free phone calls in usa and switzerland ...
23:48.07Cybertoyvery good quality.
23:53.57bkw__any asterlink customers here?
23:54.03SwKno
23:54.05SwKasterlink sucks
23:54.12SwKhah
23:54.17nemisuslol
23:54.18bkw__thats bullshit.. NEXT!!!!
23:54.19SwKgod damned register issues
23:54.30SwK</asshole>
23:54.37SwKreally asterlink is a good provider
23:54.37bkw__freebsd ain't my problem :P
23:54.41bkw__haha
23:54.45bkw__but we gotta fix it
23:54.45SwKthey are just bsd clueless
23:54.55bkw__no we are not.. asterisk is
23:55.02bkw__me luvs BSD
23:55.02SwKheh
23:55.04bkw__:P
23:55.05*** join/#asterisk calvinhp (~calvinhp@cpe-65-29-88-222.indy.res.rr.com)
23:55.25CybertoyI thought there's a port for openbsd for asterisk
23:55.34Cybertoybut probably an older version
23:55.35bkw__it compiles on Mac OS X
23:55.37SwKbkw__ I wont tell anyone about your chuck the daemon butt plug
23:55.39*** join/#asterisk R3DB0x (nobody@66.142.28.36)
23:55.54SwKCybertoy: sometimes it will compile
23:55.58hardwireheh
23:56.00file[desk]bkw__: I can't call toll-frees d00d
23:56.09SwKCybertoy just depends on the when you check out head
23:56.22bkw__file[desk], i'll fix that in a few dear
23:56.24bkw__:P
23:56.29SwKCybertoy try it and lemme know id it doesnt we'll see what we can do to patch it up
23:56.30hardwirethats sweet
23:56.52CybertoyI'm not gonna... I have a nice working installation on redhat linux.
23:56.53|Vulture|bkw_: yea
23:56.57file[desk]bkw__: my service doesn't work, *WHINE* :P
23:56.59|Vulture|bkw_: I am one
23:58.42*** join/#asterisk dasuberdavid (~chibski@user-24-214-126-76.knology.net)
23:58.56harryvvWe just had a major down poor and asked if my dog wanted to go out side he was exceited. he then ran anywhere he could to get away from the huge downpoor :)
23:59.06bewesthas anyone got their sipura 3000 to get the pstn line to call * properly?
23:59.18bewestthe only way I could manage it was to mangle the caller id stuff in sipura
23:59.19|Vulture|harryvv: where?
23:59.41dasuberdaviddoes anyone know what context a _9NXXXXXXX extensions should be in
23:59.42dasuberdavid?
23:59.55|Vulture|the outbound one...
23:59.58RoyK~lart himself

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