irclog2html for #asterisk on 20050729

00:00.26laserfoxcan anyone help? my asterisk box is behind a router in my office and i am trying to register xlite from here...
00:00.39sylecvs stable is good if you never want to program with it
00:01.01hardwirelaserfox: forward in the right ports
00:01.03DarthCluecvs stable is good if you like being stuck in the past.  just wait till 1.2 comes out then use stable
00:01.41Qwellholy shit man...UPS sent me a fruit basket
00:01.58hardwirewell.. they do that
00:02.11laserfoxi have set up some port forwarding on my router at the office..  port 5060 and 10000-20000 goes to the * box
00:02.14Qwellhardwire: as part of an apology
00:02.17hardwireah
00:02.21hardwireI figured as a shipment :)
00:02.36laserfoxand here i am behind my debian router
00:03.30laserfoxis it possible?
00:03.43hardwirehave you ever used asterisk?
00:03.54mishehuugh.  it'd be really nice if I could get my bluetooth headset to pair up with my hci
00:04.03hardwiremishehu: its not that hard
00:04.32laserfoxevery day for the past three weeks, i can register xlites in the office no probs
00:04.38*** join/#asterisk stormfr (~StorM@alf94-3-82-66-251-138.fbx.proxad.net)
00:04.39laserfoxgot capi fritz working
00:05.41sylequick Poll: how many people run asterisk in a screen session?
00:05.53anthmme!
00:07.31laserfoxwhat do u reckon hardwire? can u help :)
00:07.57mishehuhardwire: I keep getting an error of Error: Failed to connect to SDP server: Function not implemented
00:08.47opus_if I rebuild chan_sip.so, will 'reload chan_sip.so' load the new binary?
00:08.51mishehuand Can't connect RFCOMM channel: Resource temporarily unavailable
00:09.45sylei can see opus crashing his systems alot
00:10.12opus_System uptime: 6 days, 23 hours, 38 minutes, 6 seconds
00:10.17opus_thats my development box
00:10.28*** join/#asterisk MikeJ[Laptop] (~ircatjerr@adsl-69-209-191-45.dsl.sfldmi.ameritech.net)
00:10.31*** join/#asterisk meppl (~mephisto@87.193.3.117)
00:10.34*** part/#asterisk MasterYoda (~mnicholso@MasterYoda.sustaining.supporter.pdpc)
00:10.57opus_well i guess its my preproduction box now, i don't want to fuck up my uptime! :)
00:11.21laserfoxi read on a forum that it is not possible.. :(
00:11.25sylei guess that depends opus on whether you made a few simple changes or downloaded a whole new cvs
00:11.49syleyou wouldn;t run old kernel modules with a new kernel would you
00:11.51*** join/#asterisk znoG (~gs@200.115.218.81)
00:11.53hardwiremishehu: privmsg me
00:12.02hardwirelaserfox: what did you do wrong?
00:12.26opus_i just made a one line change to try and fix the polycom dtmfmode problem
00:12.29laserfoxif i knew that, i would put it right :P
00:12.35znoGhi guys, question.. i just plugged in my shiny new PCI wireless card, and it seems to be conflicting with my FXO card. When I try and dial out through it, I get this massibly loud and annoying sound making it impossible to hear the other person. Any ideas?
00:12.42syledon;t see a problem then
00:12.52opus_i don't think it actually loaded it hmmm
00:12.59hardwirelaserfox: what changed?
00:13.04hardwireI thought you said xlites were working
00:13.05opus_perhaps i should try deleteing the file and reloading it and checking for an error message to verify this
00:13.09hardwireall inside the firewall.. right?
00:13.14sylei would restart asterisk completely anyways when inserting a new module
00:13.15*** join/#asterisk Cybertoy (~Cybertoy@dsl254-123-241.nyc1.dsl.speakeasy.net)
00:13.16hardwirebut not from the outside?
00:13.22laserfoxcant register my sip from here across the net, the * box is behind a router.. i have no problem when i am on the same lan as it
00:13.30hardwireok
00:13.32laserfoxi am just trying this for the first time tonight
00:13.37hardwireso on the debian firewall you need to do the following
00:13.48hardwiredo you have a static IP ?
00:14.00laserfoxnot static ip here, office one is
00:14.14opus_nope, it still reloads even when i rm -rf the module
00:14.17hardwirelaserfox: ok
00:14.19hardwireso at the office
00:14.26hardwirethats where the firewall is?
00:14.29hardwirethe debian one?
00:14.51hardwireok.. forward in the IAX2 port from your firewalls public IP address.. to the asterisk box inside the office..
00:14.54laserfoxno office router is a conexant or something like that
00:14.59hardwireok
00:15.02laserfoxhere my gateway is my debian box
00:15.07hardwireyou still need to do port forwarding at the office
00:15.25hardwireso.. fw at office forwards port:iax to asteriskbox port:iax
00:15.30laserfoxon debian i port forward the 5060 and the 10000-20000 to my pc here?
00:15.36hardwirethen in iax.conf (dunno if this is needed) you set your external IP
00:15.47laserfoxport forwarding all done on the conexant ok then?
00:15.48hardwireerr
00:15.49hardwireshit
00:15.51hardwirexlite == sip
00:15.54hardwireyeh
00:15.59laserfoxiax?
00:16.02hardwirequit
00:16.04hardwirerewind
00:16.09hardwirewhere is * box?
00:16.13laserfoxlol
00:16.17opus_shit, my fix sort of worked.
00:16.24opus_Is there a way to make dtmf longer?
00:16.27laserfox* box in office behind conexant with static ip
00:16.34hardwireopus_: hold down the button longer :)
00:16.40laserfoxlol
00:16.46hardwirelaserfox: on conexant.. forward 5060 and all RTP ports to asterisk box
00:16.48opus_Make dtmf tones longer in length, but with rfc2883
00:16.54hardwireyou can even limit the range on the rtp in rtp.conf on the * box
00:17.04laserfoxrtp ports being 10000-20000 ?
00:17.11hardwireby default
00:17.16laserfoxcool, thats done
00:17.17hardwireall UDP
00:17.18hardwireok
00:17.20laserfoxyup
00:17.49hardwireso in sip.conf set externip
00:17.58hardwireexternip = conexants_extern_ip
00:18.04laserfoxoooh!
00:18.14hardwireand set the localnet to the local networks network and netmask
00:18.18hardwireand set nat=yes
00:18.42hardwirealso turn off canreinvite for that xlite
00:19.22opus_hmmm.. why is  send_dtmf in rtp.c?
00:19.49*** join/#asterisk snewpy (~markl@203-166-227-227.dyn.iinet.net.au)
00:20.01znoGfound my problem
00:20.10znoGthe USB port and my FXO card are sharing an IRQ
00:21.37znoGnow i gotta work out how to put the FXO on its own IRQ
00:21.51syleeasy
00:22.04sylepull out all your other pci cards
00:22.33znoGjust re-arrange them i guess, eh?
00:22.37znoGi have no free PCI slots
00:22.45xhelioxyou can manually assign an irq to the slot in your bios
00:22.50sylewhy you need more than a video card?
00:23.08znoGi have an FXO card, a wireless card, a PCI video card and a NIC
00:24.11syleits a pain in the ass, i did it once, and personally i;d rather just upgrade the motherboard
00:24.28syleyou;ll just end up having zap problems anyways if you don;t
00:24.28znoGwhat is?
00:24.35laserfoxcant connect :(
00:24.39znoGunfortunately I need all the cards in the system
00:24.50laserfoxdo i have to port forward on my debian box then also?
00:24.51*** join/#asterisk MustDie (~Alik@205.247.13.73)
00:24.54syleupgrade your motherboard
00:25.02QwellznoG: tried switching PCI ports?
00:25.22znoGQwell: am about to try assigning it a fixed IRQ first
00:25.30znoGthen yes, i'll try the swap method
00:25.49sylestart with the wireless card
00:25.54sylei can;t see how you need that
00:26.03znoGits a gateway box acting as an access point
00:26.12znoGi rather keep all gateway functions on the one machine in the house
00:26.37syleman i don;t trust wireless in my house to do anythign more than google something on a laptop on the couch hehe
00:26.51*** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv)
00:27.53sylehow is your accesspoint?
00:27.57syleany packet loss?
00:29.01laserfoxweird.. it all looks right
00:29.03znoGsyle: to be honest, the wireless card has been in the machine for the last 15 minutes, i just bought it. I'll let you know :)
00:29.14znoGbrb gonna do some PCI slot swapping
00:30.10xhelioxexten => s,5(restart),BackGround(blah) ; --- what is the "restart" used for? I can't seem to find a reference to it anywhere
00:37.44rikstaanyone on linux with a bluetooth adapter wanna try our bluetooth presence detection call redirect app?
00:37.59Nuggetno, I'm already running mine.  :)
00:38.07rikstakool :)
00:38.24rikstais it a script...or?
00:38.34Nuggetan agi in perl, nothing fancy.
00:38.39rikstaaha i see
00:39.32Nuggethttp://slacker.com/~nugget/asterisk6.php
00:40.00rikstaslow :) but gettin there
00:40.35rikstathe one we did is x-platform...eventually
00:41.19Nuggetbluetooth in linux is pretty shitty.  I hated working with it to get the presence detection going.
00:41.42rikstaim using a proper api
00:42.14NuggetI didn't really think that the concept warranted that much effort.
00:42.29NuggetI mean, it's a one line shell command
00:43.03rikstanot really, but it's just a plugin of a handy app srt and I are making, with lots of other stuff, plus it's x-platform whereas the agi isn't and has to run on the same box
00:43.15Nuggetbluetooth isn't cross platform, though.
00:43.21rikstaanyway..i'm not knocking it or anything
00:43.32*** join/#asterisk oOlli (www-data@baltz-online.de)
00:43.45rikstathe API is, you provide a different jar for each platform to interface it
00:43.51Nuggetoh, java.  :)
00:44.11rikstayeah, but it's SWT so it's nice
00:44.21rikstahow else you gonna go x-platform ;)
00:44.55Nuggetwell, with asterisk it's safe to assume "unix" so cross platform has a lot of options as long as "platform" equates to "asterisk deployments"
00:45.16rikstayeah for the backend maybe
00:45.22rikstanot the client end
00:45.27*** join/#asterisk wwalker (~wwalker@wwalker.sustaining.supporter.pdpc)
00:45.48Nuggetif your concept of crossplatform is "we play both kinds of music: country and western!" where it's either windows or linux, well, java's a good bet.  :)
00:46.07rikstahhehe, well i'm not getting into a debate about it :)
00:46.10rikstawas just making convo
00:46.23laserfoxu there hardwire?
00:47.31rikstanite
00:47.46CoriantumAnyone know how to add comments in AEL?
00:48.03NuggetWhat is AEL?
00:48.27Nuggetah, I see.  :)
00:48.38Coriantumit's a extension parser
00:48.41Coriantuman
00:50.24*** join/#asterisk dasenjo (~dasenjo@63.245.87.180)
00:50.53*** join/#asterisk TheEmperor (~TheEmpero@210.19.250.122)
00:51.12wwalkerI'm very new to asterisk administration.  Been doing mostly development (mostly AGI/perl).  Everything works great, except everytime I play a sound file, there's a say 60% chance that it will begin with "warbly" audio, then smooth out.  The rest of the time it plays fine.   I'm running HEAD from a few days back.  IAX2, ulaw, I tried jitterbuffer=yes and got no audio at all.  Any pointer to what to read up on?
00:52.04Nuggetif you find out, let me know.  I've been using asterisk for about a year now and that's been my experience too.  Although not quite that bad.
00:52.33CoriantumAre you running X?
00:52.52wwalkerNo X,
00:52.57anthmmaybe it's the cpu spike of your AGI process launching
00:53.22CoriantumWould a wait help in that case?
00:53.29wwalkerAGI is already running for a second or two by then...
00:53.50anthmis it agi's fault, if you do the same thing in the dialplan is it ok?
00:53.59Darwin35anthm need input
00:54.12wwalkerdual proc (Intel(R) Xeon(TM) CPU 3.00GHz) HT
00:54.24Nuggetin my experience it takes a call a second or so to figure out what the hell is going on.
00:54.28Darwin35daw and I are having issues with set($DB()
00:54.34NuggetI just cheat by answering and then sticking a wait(1) on all calls
00:54.41Darwin35its not puting things into the db
00:54.51Nuggettrying to start playback immediately is very problematic
00:54.57Darwin35but dbput is
00:55.01Darwin35on head
00:55.12Darwin35but it gives the depriciated warning
00:55.26anthmyou dont say set($db
00:55.29Juggieanyone ever have a problem with callerid not working on outgoing long distance calls?
00:55.37Juggieseems to work local, but ld seems to be empty.
00:55.42Juggiebut i can receive LD callerid, ok.
00:55.44anthmcheck the syntax of your set
00:55.46Juggieso i'm not sure whats up
00:55.53NuggetJuggie: over what type of channel?
00:55.56Darwin35Set(${DB(family/key)}=${foo})
00:55.58Juggiepri
00:56.03wwalkeranthm Nope,, sounds terrible from the dialplan too.
00:56.04Juggieit works local
00:56.09Nuggetweird
00:56.19Juggiemaybe its the format?
00:56.25Nuggetdunno, sorry.
00:56.30anthmok, so you ruled out agi, 1 step closer
00:56.37Juggiei have like "Donny <613-562-6242>"
00:56.41anthmwhat channel type is it
00:56.42oOllihi folks. i have 2x ISDN BRI here. what interface cards do i need to use tese lines with astrisk? NT-mode? active or passive cards? we are going to use PTP.
00:56.43Juggiedo i need to put a 1?
00:56.51NuggetDitch the dashes, I'd suggest.
00:56.57Juggiek
00:57.32Nuggetand if that fails, experiment with setcidnum() directly to completely bypass any potential parsing issues in asterisk.  (or the HEAD equivalent which is changed)
00:57.39Darwin35exten => *57,1,Set(${DB(SCA/${CALLERIDNUM}=YES) this is not puting anything in the db to make it active
00:57.45Nuggetset(callerid(num)=aaabbbcccc) I think
00:58.07Darwin35http://lists.digium.com/pipermail/asterisk-cvs/2005-May/006297.html
00:58.12Darwin35this is what I am fallowing
00:58.17Nuggetbut generally callerid number should just be digits
00:59.27anthmyou dont use the $ in the set
01:00.03Darwin35then the readme is wrong
01:00.09oOllimay i use 2x passive ISDN BRI cards for PTP?
01:00.24anthmSet(DB(SCA/${CALLERIDNUM})=YES)
01:00.33anthmwhen you set func you do not use $
01:00.35Darwin35ok
01:00.42anthmSet(FUNC(val)=val)
01:01.11anthm$ forces an eval
01:01.34Darwin35${ is used to check the status
01:01.56anthm${ tells the parser to evaluate
01:01.57Darwin35then the readme needs to be fixed
01:02.02Darwin35ok
01:02.07anthmso does $[
01:02.20anthmyou cannot set and eval in the same step
01:02.45anthmtell the readme they are on crack, I made the function engine so I can assure you ;)
01:05.59oOllino one willing to help me=
01:06.00oOlli?
01:06.11Ariel_Darwin35, did you get your phone to take calls? the IAX one?
01:06.12Darwin35ok thnks anthm
01:06.20Darwin35yes it rings
01:06.38Darwin35the iax protocall needs work
01:06.56*** join/#asterisk sigterm (sigterm@devious.info)
01:06.56Ariel_oOlli, we migth now know about ISDN BRI cards.
01:07.02Darwin35if I am on the phone it goes to unavaible vm and not busy vm
01:07.07Ariel_Darwin35, on the phone or asterisk side
01:07.14Darwin35both I think
01:07.49Darwin35ok anthm its all working now
01:07.59*** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net)
01:08.01oOlliAriel_, do I really nead active cards?
01:08.34Ariel_oOlli, I don't know here we don't use ISDN BRI cards.
01:09.24*** join/#asterisk mog_home (~mogorman@user-24-236-84-48.knology.net)
01:09.35*** part/#asterisk dasenjo (~dasenjo@63.245.87.180)
01:09.45Darwin35nope
01:09.48oOlliAriel_, what do you use instead?
01:09.54Ariel_oOlli, most of the people that BRI cards are sleeping at this time.
01:09.59Darwin35it did not put a DND in the database
01:10.06Ariel_oOlli, I use analog and pri cards
01:10.10Darwin35database show shows nothing for DND
01:10.26oOlliAriel_, you are right. i should go sleeping, too :)
01:10.32NewSolelol
01:10.53NewSolewhat are we talking about
01:11.05Ariel_oOlli, well I wish I could help you on that part.
01:11.15Ariel_NewSole, oOlli needs info on BRI cards
01:13.43Darwin35fixed
01:15.33NewSoledid the doctor go snip... snip...
01:19.19*** join/#asterisk jr352k (~jraborg@pcp03618028pcs.univde01.de.comcast.net)
01:24.06jontowso.. HEAD/netbsd works great so far.. made a successfuly call back to the office through the PRI and back to my landline here :)
01:24.15jontowtime to relax a bit
01:24.26fugitivowhy people write scripts in php?
01:25.04fugitivoperl exists for that reason
01:25.08fugitivophp is for the web
01:25.44ptiggerdinejontow, which PRI cards are you using?
01:25.44NuggetI have no idea and I share your puzzlement.
01:26.07Nuggetphp was not designed to be a general-purpose scripting language and it isn't very well-suited to that task
01:26.41fugitivoNugget: god, i wanted to try that wakeup call agi, and it's php! i'm not going to install php in my asterisk box
01:26.50Nuggeteww
01:27.06fugitivonow i'll have to write one in perl
01:27.08NewSolewhy write a script to hack what you want when you can just write a C code moduel to do exactly what you want
01:27.48fugitivoNewSole: yeah yeah, but no time to write C
01:28.14NewSoleyes but its the better way.... uses less cpu
01:28.21fugitivoi know
01:28.28Nugget"better" is such a subjective term/
01:28.34Nuggetdepends on what you need to accomplish
01:29.00fugitivoC is "better" for performance
01:29.14Nuggetsure, but my asterisk box sits at 0.00% CPU all day long.
01:29.22Nuggetand performance isn't always the bottleneck.
01:29.33NewSoleya but how many calls at once do you get
01:29.37fugitivoPERL is "better" for "i have no time"
01:29.43Nuggetnot enough for me to fuss over cpu overhead.
01:30.15fugitivoNugget: i know, and a wake up script will not run more than once a while, so performance is not a problem
01:30.17NewSolealready running at 70%
01:31.07fugitivoNewSole: wow
01:31.56NewSoleya over 300 clients and over 80 peers doing at least 20 calls each at once
01:32.14fugitivoNewSole: what system?
01:32.17NewSolealready ordered 3 more servers
01:35.18NewSoledual xeon server that does ontario canada.... our BC and West cost server is only 7%
01:36.49fugitivowhy xeon and not opteron?
01:37.26NewSolecause I dont like bursting CPU's
01:38.05fugitivowhy bursting?
01:38.08fugitivoopteron works well
01:38.33fugitivoi didn't try it with asterisk, that's why i'm asking
01:38.50fugitivoi used to use it as a server for thin clients
01:39.03NewSoleAMD gives the most active process the most cpu power
01:39.38*** join/#asterisk znoG (~gs@200.115.218.81)
01:39.53NewSoleand thats great if it was just routing calls and it was a single thread
01:40.00znoGafter a bit of card swapping, got my cards up each on their own IRQ :)
01:40.17*** join/#asterisk santiago (~santiago@63.245.86.141)
01:40.24fugitivoNewSole: so you think it's not a good idea opteron for asterisk?
01:40.31NewSolebut drop in multi calls and transcoding codecs...
01:41.11NewSoleeven hyper thead from intel is bad idea
01:41.31Nuggetwouldn't that be more an issue determined by the os scheduler, not the cpu?
01:41.38NewSolecauses more leaks then a kid in a soda shop
01:46.19Nuggetand aren't xeons "bursting cpus"?  I mean, they're based on an architecture named "NetBurst", right?
01:46.40NewSoleonly hyper threding...
01:46.47NewSolethats why i turn it off
01:46.58NuggetI have to turn it off for my DB2 boxes.
01:47.08Nuggetbut what's that got to do with opterons?
01:48.18fmanright, so anyone want to help me compile zaptel
01:48.27fmanshould be a stupid problem I've got in front of me
01:49.26*** join/#asterisk stkn_ (nobody@stkn-active-pdpc.developer.gentoo)
01:49.43Kattymew
01:49.48fugitivofman: emerge zaptel
01:49.51fmanhmm
01:49.56fmanthis is debian
01:50.00fmanit is bitching about You do not appear to have the kernel sources for your current kernel installed.
01:50.12fmanbut I have a static link in /usr/src
01:50.17fmanpoint towards linux
01:50.22*** join/#asterisk DonX (don@tool.sparkhosting.net)
01:50.26*** join/#asterisk Beirdo (~gjhurlbu@beirdo.user) [NETSPLIT VICTIM]
01:50.32Nuggethi hi
01:50.43Kattyhow be?
01:51.04Nuggetfman: a static link to what, though?  does the kernel source you have really match your running kernel?
01:51.27fmanwell I'm running 2.6.11-powerpc
01:51.29*** join/#asterisk dudes (~dudes@12-215-34-84.client.mchsi.com)
01:51.31NuggetCan't complain, Katty.  Been a lazy week and I'm all done with work.
01:51.34fmanthe kernel source is 2-6-11
01:51.42fmanso, in my mind that is the same
01:51.43Katty:>
01:52.12KattyDarthClue: what time are you guys going to be here monday?
01:53.29KattyNewSole: you're running windows? :/
01:54.13NewSoleno... I have this person who keeps calling our main number as a calling card
01:54.29Kattyk
01:54.39NewSoleand wont speak english... just keep hanging up un us
01:55.22NewSoleso I have it black listed and everytime they call now it gets forwarded to 1800-SPANKME
01:55.47Katty:<
01:55.56Katty(:>)
01:56.05opus_Jul 28 18:55:37 NOTICE[14715]: chan_sip.c:3326 process_sdp: No compatible codecs!
01:56.07Nugget]:8) Moo
01:56.08ltersKatty, can I ask a question?
01:56.14NuggetYou just did.
01:56.20Kattydidn't leave me much choice there sunshine.
01:56.21*** join/#asterisk kingtux (~susekid@pool-141-157-9-215.balt.east.verizon.net)
01:56.26ltersanother one :)
01:56.28opus_anyone use asterisk config odbc and have 'No compatible codecs' problem?
01:56.35NewSolejust did
01:56.45opus_I have disallow/allow in that order in both ast_config and in my users table
01:56.55opus_disallow=all;
01:57.04opus_allow=ulaw;alaw;gsm right?
01:57.10kingtuxAnyone know much on a calling card setup
01:57.17ltersI am using 1.0.9 and badly need the HEAD with better queue/transfer
01:57.36Kattyi see
01:57.39lterstheres awful many cvs updates.
01:57.52lterswhat date could I grab that would be decently stable.
01:58.14Kattyuh.
01:58.18Kattythe one that works? (=
01:58.27*** join/#asterisk _m_ (~m@fbta199.fbta.uni-karlsruhe.de)
01:58.45*** join/#asterisk KryoStoffer (~kri@helium.kri.dk) [NETSPLIT VICTIM]
01:59.05ltersI saw JerJer said he was on 7-11
01:59.16kingtuxanyone??
01:59.53*** join/#asterisk Evanrude (~david@wsip-68-15-251-34.dl.dl.cox.net)
02:01.11*** join/#asterisk RandomAndy (~randomand@adsl-63-207-13-143.dsl.snfc21.pacbell.net)
02:01.28*** join/#asterisk stkn_ (nobody@stkn-active-pdpc.developer.gentoo)
02:02.12NewSolekingtux... what are you looking for
02:04.06kingtuxCalling Card system with asteisk
02:04.08kingtuxasterisk
02:04.16kingtuxsome info
02:04.19hardwireastcc
02:04.24hardwirevoip-info.org
02:04.26hardwiregoogle
02:04.31hardwirecalling people
02:04.35hardwiredial 911 and ask
02:07.04Nuggetheh
02:08.13Darwin35has dbdel been changed
02:08.51Darwin35exten => *88,n,DBdel(DB(DND/${CALLERIDNUM}) is not working
02:09.01Darwin35its not removing the line from the db
02:13.25Darwin35fixed
02:15.07dudesin a "permit=" if you put 192.168.1.255/255.255.255.0 would that be correct?  Or would it be permit=192.168.1.0/24
02:17.24*** join/#asterisk blaed (~davidm@220-253-41-248.VIC.netspace.net.au)
02:21.14loud192.168.1.255/255.255.255.0 is wrong for a whole /24, should be 192.168.1.0/255.255.255.0 if you use that syntax.
02:23.34Darwin35yes its all working again
02:23.40Darwin35my full dial plan
02:23.57Darwin35now to  finish the new macro  for stdexten
02:25.43opus_if I have a register in sip.conf, somebody calls, it goes to the 's' in the general context of sip.conf
02:26.26*** join/#asterisk dysan (~ack@202.37.224.27)
02:26.57xhelioxexten => s,5(restart),BackGround(blah) ; --- what is the "restart" used for? I can't seem to find a reference to it anywhere
02:29.29dysanmy voicemail messages arnt getting emailed, sendmail is working fine, and they are getting saved in /var/spool/asterisk/voicemail/default/501/INBOX
02:30.25*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
02:31.46*** part/#asterisk moy (~kvirc@201.135.113.46)
02:34.59TheEmperordysan: i seem to have the same problem
02:35.15Qwellxheliox: There is likely a reference to it on the wiki
02:40.17xhelioxQwell: Yes, but unfortunately the term background shows up nothing of relevance, that I could find. :)
02:40.44QwellI don't think restart and background are related at all
02:42.21*** join/#asterisk Hogie (daniel@alpha.dfwservers.net)
02:43.56DarthCluexheliox: where did restart come from?
02:44.24QwellDarthClue: I've seen it used once or twice before.  I think amp generates it for some things
02:45.25xhelioxDarthClue: I'm starting to wonder that myself, maybe the Wiki. Which isn't a very reliable source always :-/
02:45.46DarthCluexheliox: the wiki is far more reliable than amp.
02:45.57loudi love doing that
02:46.11loudlove removing every single amp out there.
02:46.12xhelioxWell, I was just curious, I don't need it or anything, I just never saw it before.
02:46.35DarthClueloud: Congratulations, you have been chosen as todays lucky winner.  Give me root access so that I can eliminate your worries.
02:47.03sylehow can you get a list of what is on the cvs server
02:47.14sylethere like a cvs list command
02:47.33Qwellsyle: ls?
02:47.55syle?
02:48.06Qwellcvs help
02:48.14syleand
02:49.59*** join/#asterisk kabewm (~kabewm@24.180.28.208)
02:50.52Qwelllook at what there is..
02:51.22sylei wouldn;t of asked if i hadn;t looked already
02:51.25syledo you know how to check?
02:53.42dysanwhen i look at /var/log/mail.info i see Jul 28 22:49:20 asterisk sendmail[2789]: j6T2nKbI002789: to=... <..@..>, ctladdr=asterisk (103/103)...(j6T2nKE5002791 Message accepted for delivery)
02:53.49dysanbut mail not sent
02:53.51dysanany ideas?
02:54.20syleif your on adsl or cable modem i doubt your sending anything hehe
02:56.48bkw_never fear bkw is here
02:56.57Qwellcrap
02:57.01Qwellparties over guys
02:57.09Qwellparty's?  whatever
02:57.18bkw_haha
02:58.04fmanright, got it all working
03:01.20Hogiehow's it hanging bkw?
03:01.51bkw_he is about to kick the shit out of cogent
03:07.52hermiebut, but... Cogent has 30,000 miles of fiber! You can't hate somebody who owns 30,000 miles of glass and teflon!
03:08.16QwellI'd be more impressed if it was 30,000 square miles
03:08.38konfuzedhey there, I think I figured out a big part of the sip registration and auth problem I am still having. I'm sure there is a config error on the provider side and it shows when I run sip debug and then call into the number
03:08.44DarthCluehermie: you obviously don't know bkw_
03:09.28loudwe all know cogent tho.
03:09.35hermieQwell: I won't argue... that would be more impressive...
03:09.40konfuzedcan I post the debug results somewhere and hopefully some one can confirm what this debug report says
03:09.55Qwellkonfuzed: pastebin.ca
03:10.01konfuzedthanks
03:10.25*** join/#asterisk vuvie (~vuvie@bb219-74-46-99.singnet.com.sg)
03:12.14*** join/#asterisk DougRoyer (doug@70-67.69-92-cpe.cableone.net)
03:12.14DarthCluehermie: they have effectively shut down our services.  it ain't gonna be pretty when bkw gets done with them.
03:13.25QwellI'd like to hear that call.  He should pull in 996
03:13.37DarthClueQwell: 996 is down.
03:13.45QwellDarthClue: right...
03:13.52Qwellwell then
03:13.59Qwellparty at Qwell place
03:14.09QwellQwells...today isn't my day
03:14.22Qwellbkw_: recording it?
03:14.30bkw_please hold while I put crisco on this fucking boot so I can kick this guys tonsils
03:14.46DarthClueQwell: i recommend hiding, and you might want to put this raincoat so you don't get intestinal leakage on you.
03:14.49hermiebkw_: do you have an AS?
03:15.06QwellDarthClue: I'm used to shit kicking.  See my UPS rant from Monday.
03:15.22QwellDarthClue: UPS btw, sent me a fruit basket today
03:15.26hermieaka ASN
03:15.40DarthClueQwell: k, just want to be sure that everyone is properly prepared.
03:16.18QwellI also told them I wanted to be reimbursed if I go over my minutes on my cellphone...they said they would reimburse it whether or not I went over.  I'm waiting on a check right now
03:16.20konfuzedok so I added a bit of info for context at http://pastebin.ca/18752
03:16.50Darwin35does the fucking boot come with a vibrator attachment
03:17.04fileooh Darth is interested
03:17.21DarthClueDarwin35: no, and we chose not to use crisco.
03:17.24Qwellkonfuzed: Do you have two IPs on your box or something?
03:17.40konfuzedQwell   I new it could be done
03:17.49konfuzedthe gift basket thing that is
03:18.07konfuzedwell actually to some extent that is correct.
03:18.15konfuzedthere has just been a fresh reinstall
03:18.34DarthCluefile: Darth is bored.  And besides, I have a financial interest in this coming back up.
03:19.07konfuzedtwo ethernet adapters on the box and an openbsd firewall with both those ips pointed at one each of those ether net cards
03:19.17bkw_anthm and I are both on th ephone with cogent now
03:19.19konfuzeda hard binat though
03:19.23bkw_this is gonna be fun
03:19.29file[laptop]oh oh oh
03:19.43DarthCluetoo bad the conference isn't working, this would be a really funny call to be on.
03:19.45Qwellkonfuzed: So, its behind NAT?
03:19.48konfuzedthe ips on the box are actually 192.168.1.136 and 1.137
03:20.08konfuzedim not sure that open bsd's interpretation is the same as yours
03:20.35Qwellso, it is behind a NAT...
03:20.46Qwells/a //
03:20.47konfuzedi did not config the firewall and can't read pf.conf
03:21.01Qwellkonfuzed: 192.168 == NAT
03:21.11konfuzedsufficient for me
03:21.56konfuzedso bviously nat=yes or no
03:22.12konfuzedI take it in that I have it on no you would say set it to yes
03:23.11QwellDoes the fbsd box have multiple public IPs?
03:23.22konfuzedyes
03:23.33konfuzedi've just made the nat=yes change and
03:24.13konfuzedthe sip debug still shows    (no NAT)
03:24.16konfuzed??
03:24.24konfuzedperhaps that just doesnlt matter
03:24.29konfuzedmore importantly
03:25.00konfuzedthe toaddress changed an internal ip 192.168.3.43
03:25.19*** join/#asterisk tonyzhang (~tengulre@61.185.238.166)
03:25.27Qwellif you're forwarding the ports, you don't really need nat=yes, afaik
03:26.24konfuzedwell that time all I changed was nat=no to nat=yes    and then the sip debug   reports   to : 192.168.3.43 instead of 64.119.127.99
03:26.37konfuzedneither of which should be involved
03:27.55konfuzedshite the register line is effectively commented out both times asterisk was reloaded
03:28.12Qwellkonfuzed: try setting externip
03:28.16konfuzedso I only got those damn results in  the first place because the register line was commented out
03:28.22konfuzedhm ok
03:28.24Qwelland likely localnet
03:28.52konfuzedno the same as host=sip.providers.ip   bu my ip obviously
03:28.55konfuzedlocalnet?
03:29.25Nuggetnat is such a pain in the ass.
03:29.29Qwell192.168.1.0/24, probably
03:29.34QwellNugget: indeed
03:29.37konfuzedone of my big conundrums here is that I believe the register=   is required and my provider suggested that he doesn't think its needed
03:30.00Qwellkonfuzed: Do you get incoming calls from the provider?
03:30.06QwellIf so, you definitely need to register
03:30.20konfuzeddo i or don't bother with the register?     I thought so
03:30.38konfuzedthe provider is running asteisk and provides me DID
03:30.58konfuzedhe says that he configured the DID to call my ip 64.119.114.148
03:31.01Nuggetif you don't register, the provider will not know where to send your calls.
03:31.20*** part/#asterisk Cybertoy (~Cybertoy@dsl254-123-241.nyc1.dsl.speakeasy.net)
03:31.20Qwellunless you have to call and update your IP all the time...
03:31.28Qwell(which is stupid)
03:31.28konfuzedsee that's what started me thinking the provider has a config problem
03:31.41QwellWhat provider?
03:31.52konfuzedlocal toronto company
03:32.10konfuzedthey don't want me to tell anyone they offer VoIP and DID yet.
03:32.25Qwellwonder why
03:32.31konfuzedI suppose
03:33.12konfuzedI think they are quite big on SIP and are mostly using asterisk to map DIDs and internal SIP phones and likely for voicemail
03:33.15Nuggetsounds like they DON'T offer it yet.  :)
03:33.28konfuzedI know they like cisco gear but they are also very big opensource operators
03:34.07konfuzedthey consider it live testing stage
03:34.27Qwellhope you aren't paying anything for it
03:35.12konfuzedon the other hand I know its the Hybrid Gateway nature of Asterisks and the firewall benefits of IAX that make asterisk the killer app that it is
03:35.23konfuzedwell I haven't sent in any money yet
03:35.56konfuzedbut perhaps they have not configured it wrong and some screwed up thing is on my end ??????
03:36.09konfuzedI'm trying to validate either scenario :)
03:38.40konfuzedok so if I reactivate the register line then registration fails
03:38.43konfuzedpisses me off
03:39.47konfuzedperhaps the provider has configured my account as though it is going to an external SIP phone.
03:39.54konfuzeddoes this make any sense?
03:39.55*** part/#asterisk vuvie (~vuvie@bb219-74-46-99.singnet.com.sg)
03:40.30Sedoroxwouldn't matter.. asterisk acts the same as a sip phone...
03:40.37konfuzedif it is the case can I set my asterisk to behave as a sip phone
03:40.41konfuzedwell there ya go
03:40.42Sedoroxjust a hell of a lot more features :p
03:40.55Qwelland SIP phones need to register
03:41.06konfuzedso would  a cisco sip phone need a register= function?
03:41.16konfuzedthat would be yes then
03:41.45konfuzedis that also true just on a LAN
03:41.52dudesyes
03:41.53konfuzedwith out a firewall in the middle
03:42.03konfuzedok
03:42.10*** join/#asterisk kks (~kks@202.73.8.130)
03:42.21konfuzedso I better start with woring out this register error
03:42.32*** join/#asterisk PakiPenguin (~uppal@202.147.163.81)
03:42.45dudesI take it you have a sip trunk /w a DID?
03:43.12ManxPowerOnly terrorists use the "r" option of Dial
03:43.37QwellManxPower: I use r
03:43.45`SauronTERRORIST!
03:43.46*** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net)
03:44.09bkw_YA WE SAVED THE DAY
03:44.53DarthClueso how many poor souls at cogent are gonna go home crying about the guy who tortured them today?
03:45.02Juggiecogent sucks
03:45.38fearnordonut hate on cogent
03:45.40fearnorcogent pwns.
03:46.09dudespastries are good
03:46.14fearnorits the large number of idiots who run companies that buy from cogent that suck.
03:46.19*** join/#asterisk dijungal (~ovr@206.113.106.27)
03:46.25dijungalhi guys
03:46.50dijungali have setup asterisk, but the user i setup cannot log in
03:47.00dijungalfrom the softphone
03:47.05konfuzedI already had externip=64.119.114.148
03:47.06*** join/#asterisk vuvie (~vuvie@bb219-74-46-99.singnet.com.sg)
03:47.13*** part/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com)
03:47.16mishehubah.
03:47.36DarthClueevening mishehu.  how are you tonight?
03:48.01dijungaldoes that have anything to do with the error i am getting saying "Unknown type 'test' for 'test' in sip.conf" ..?
03:48.17*** join/#asterisk Mavvie (edwin@252-131-222-203.rev.techex.net.au)
03:48.18dijungalwhere do i setup the types by the way.. ?
03:48.27dudesin the sip context type=
03:48.55Mavviedon't you love half broken implementations?
03:49.13konfuzeddudes you refer to the sip trunk with DID
03:49.42dudesyes ... is it a sip trunk /w a DID
03:50.42dijungalis there a web interface for asterisk.?
03:50.52konfuzedI've only messed with iax.conf and when that wouldn't register we cahged it to a sip setup and now I've only changed the sip.conf , isn't that the only file i need (once asterisk is running in general of course) to add this sip service account
03:51.04Delta34amp
03:51.33konfuzeddo I have to set up a trunk config on my box then ?
03:51.53*** join/#asterisk Rez (lorez@lorez.staff.freenode)
03:51.54dudeskonfuzed - do you have a username/password / and is it a trunk /w DID
03:52.50konfuzedive got user/pass/and.ip   no one said anything to me about a trunk setup until now
03:53.04konfuzedgenerally trunk for out going right?
03:53.19konfuzedI haven't even started to look a placing an out call on this yet
03:53.20dudeskonfuzed - can you drop more than one call through it?
03:53.45konfuzedI can't even register
03:54.18konfuzedAt least the server admin will answer my emails and likely make any config change I ask for
03:54.28dijungalneither can i.. cannot get my sip phone to register
03:54.45dudesin sip.conf register=username:password@ip doesn't work
03:55.02konfuzedbut I don't want to complain about his config unless I can prove i'm not screwed up
03:55.07konfuzedno
03:55.12konfuzedit doesn;t
03:55.20konfuzedI cry every time well not exactly
03:55.41dijungallol..."cry
03:55.44dudeskonfuzed - what does it tell you?
03:56.03*** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com)
03:56.03*** mode/#asterisk [+o anthm] by ChanServ
03:56.50*** join/#asterisk shmooz (~shmooz@H142.C72.B0.tor.eicat.ca)
03:57.58konfuzed*CLI> Jul 29 00:13:11 WARNING[26235]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 11bba984324d2d1765425e5b06b63fef@192.168.1.136 for seqno 102 (Non-critical Request)
03:57.58konfuzedJul 29 00:13:25 NOTICE[26235]: chan_sip.c:4052 sip_reg_timeout:    -- Registration for 'profx@sip.eicat.ca' timed out, trying again
03:57.58konfuzedJul 29 00:13:25 WARNING[26235]: chan_sip.c:6851 handle_response: Forbidden - wrong password on authentication for REGISTER for 'profx' to 'sip.eicat.ca'
03:58.17QwellSo you have the wrong password
03:58.19*** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net)
03:58.25konfuzedits not getting there
03:58.36konfuzedit has been verified twice and I even had him change it
03:59.00dudesmaybe he didn't asterisk -rx "sip reload"
03:59.04konfuzedto somethig totally different that he complained about the lack of security of 8 numbers
03:59.48file[laptop]what is it with me and hacking cellular provider services
03:59.52Qwellpsh, I store money in banks, and they only know me by a 4 digit number
03:59.55file[laptop]so far I've gotten free data on GPRS
03:59.59file[laptop]and now I've gotten free data on CDMA
04:00.03Qwellfile[laptop]: get me data on Sprint
04:00.12Qwellsave me $60/month...
04:00.23file[laptop]I gotta see if it billed me...
04:01.47Qwellhmm
04:01.47file[laptop]I cheated really
04:01.50file[laptop]and the techs were lazy
04:04.13file[laptop]I'm using the access used on regular cellphones for their microbrowser
04:04.13file[laptop]normally it goes through their proxy to be billed, and is only used on regular cellphones
04:04.13konfuzedso http://pastebin.ca/18753  this shows the sip debug which now saya to and from sip.provider.com
04:04.13file[laptop]but I programmed my PDA to use that same access for regular stuff
04:04.13file[laptop]and the techs didn't block regular ports
04:04.13konfuzedthat would seem like the wrong loop to me
04:04.13dijungalhello guys... does asterisk have a graphical or web configuration interface..?
04:04.33dudesAMP!
04:04.42dijungal?
04:04.57DarthClueamp blows.  just modify the config by hand.  it ain't that hard.
04:05.02fearnoramp is homogay
04:05.08konfuzedAsterisk Management Panel -
04:05.09dudesAmp is a web config thingy
04:05.18dijungalok
04:05.25dijungali amd configuring the thingy by hand..
04:05.30dijungalbut i was just wondering..
04:05.41konfuzedvoip-info.org indicates all kinds of compatible interfaces
04:05.51fearnorthey all suck ass.
04:05.56*** join/#asterisk nwhit (~chatzilla@traffic.whittrio.com)
04:06.11dijungalcause right now. i can't register the softphone on the my asterisk server from my softphone
04:06.22shmoozheheh
04:06.33konfuzedha
04:07.02konfuzedI so want to say go look at this one http://XenVox.com   but its just not ready yet
04:07.23dijungalok
04:07.32nwhithey... i am having trouble setting up my connect with a sip provider.... the call goes out, the remote phone rings, but it looks like there is no rtp stream setup.  I have had the same problem setting up two asterisk boxes to talk to each other with sip
04:08.36*** join/#asterisk colinm_ (~colol@VDSL-130-13-9-155.PHNX.QWEST.NET)
04:08.54konfuzedok so when I call my DID number from the regular bell phone, my providers box trys to negotiate with my box and then registration fails and i'm dropped back into a voicemail box on the providers system
04:09.18Qwellkonfuzed: You need to allow a codec that they are willing to use
04:09.41konfuzedI'l lstart with allow=all would that be a good start
04:09.46Qwellsure
04:09.57konfuzedI have gsm ulaw alaw  I better double check
04:10.28konfuzedI prefer to just make work unrestricted and then tighten it down
04:10.28Qwellif it picks something you don't like, try to allow=all, disallow=thecrappycodec
04:10.31Qwelltry again, until you find out whats valid. :p
04:10.42konfuzedyeah
04:10.59*** join/#asterisk SwK (udlijm@12-219-156-206.client.mchsi.com)
04:11.13konfuzedok so with the register I'm sure the password is right but perhaps he messed up on the userid
04:11.50konfuzedI mean he only gave me one user and the register has a userid and authid
04:12.43konfuzedif my user is supposed to be bob and the voicemail extension is 1006      is the 1006 part of authid or user id in any way?
04:15.03*** join/#asterisk riksta (~rick@84.93.243.170.broadband.plus.dyn.plus.net)
04:16.29loudriksta, hello - any news about the 7960 manager now that most people have 7.0.5 firmware ?
04:20.16*** join/#asterisk Dibbler (~Dibbler@snaddy.plus.com)
04:23.30*** join/#asterisk Rez (lorez@lorez.staff.freenode)
04:26.20konfuzedare these two ips and two domains what would typically be seen in a working config?
04:26.20konfuzedit seems to me that the from and to address should be different.
04:26.20konfuzedwhat does it mean?
04:26.21konfuzedSIP/2.0 403 Forbidden
04:26.23konfuzedVia: SIP/2.0/UDP 64.119.114.148:5060;branch=z9hG4bK3b0649a2
04:26.25konfuzedFrom: <sip:profx@sip.eicat.ca>;tag=as01eda3bd
04:26.26jarrodany better presence than fop?
04:26.27konfuzedTo: <sip:profx@sip.eicat.ca>;tag=as3283b34f
04:26.29konfuzedCall-ID: 0e43c1030c85839c16bd4b885114b5fe@192.168.1.136
04:26.40konfuzedthis darn editor
04:32.08nwhitI am having alot of trouble with this scenario  Sip UA -> Asterisk -> Sip Provider -> PSTN -- the Sip Provider makes the call, but no rtp stream appears to get setup
04:32.13nwhitany suggestions
04:32.29*** join/#asterisk cslug (~shane@202.55.153.73)
04:38.02*** join/#asterisk kabewm (~kabewm@24.180.28.208)
04:38.48*** join/#asterisk human39 (~smk@smk-dsl.fyi.net)
04:38.51loudwhats your firewall model nwhit
04:39.03nwhitno firewall actually
04:39.38bkw_lalalal
04:39.40nwhiti have had this problem with sip ua -> asterisk (sip) -> asterisk (sip) -> pstn
04:39.47human39hello all - would anybody have any clues on why my voicemail system is not picking up what I type in.  always getting a "No username but # key pressed."
04:40.02nwhitusing iax between asterisk boxes works fine
04:40.12nwhitso i think this might be a similar problem
04:40.28fearnorwhat up bkw
04:40.29nwhitthe sip ua is behind a firewall, though
04:40.47nwhitbut sip ua -> asterisk -> pstn works fine from the same device
04:46.28konfuzedif someone could please clarify for me, I thikn I just read that for incoming sip calls the only line in sip.conf that is needed is the register line.  where as the [myprovider]  context is to enable outbound calls only .   is that correct or did i misinterpret what I read??
04:48.02*** join/#asterisk tengulre (~tengulre@61.185.238.166)
04:48.04cslugwhats the best way to trunk 30 or so channels between 2 or more asterisk boxes?
04:48.30Mavvietwo PRIs :-)
04:48.35Mavvietwo cables.
04:48.45Mavvieoh screw it, the joke isn't even funny.
04:48.52*** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com)
04:48.55cslugok thanks :-S
04:49.05ManxPowerSometimes I hate Polycoms
04:49.08*** join/#asterisk riksta (~rick@84.93.243.170.broadband.plus.dyn.plus.net)
04:49.16ManxPowerThere doesn't seem to be a way to have a global contacts directory.
04:50.02brendaHow's the speaker phone quality?
04:50.16ManxPowerFor Polycoms?  Incredibly good.
04:50.26ManxPowerMost things about Polycoms are great.
04:50.27brendavs. 7960s
04:50.36*** join/#asterisk remmo (~rem@smack.isp.net.au)
04:50.36konfuzedsilence can be interpretted in so many ways I just get more konfuzed
04:50.39ManxPowerbrenda: They are similar from what I've been told.
04:50.49brendareally?
04:50.56brendawhich polycoms?
04:51.07ManxPowerbrenda: rumor has it that Cisco licensed Polycom's speakerphone tech
04:51.27brendaeven the cheap polycoms sound good?
04:51.27ManxPowerbrenda: Soundpoint IP (listen only speakerphone), IP 500 and IP 600
04:51.39brendalisten only?
04:51.46Qwellno mic
04:51.52brendaodd
04:51.59*** part/#asterisk human39 (~smk@smk-dsl.fyi.net)
04:52.03Qwellused in large conference rooms
04:52.06ManxPowerNo, they want you to buy the 500 8-)
04:52.59brendahmm
04:53.04brendaso which cheap one has a mic?
04:53.13ManxPowerPolycom has no cheap phones.
04:53.15konfuzedQwell perhaps you could unkonfuze me about the register line and context section in sip.conf
04:53.26ManxPowerThe least expeensive Polycom with a microphone is the Soundpoint IP 500
04:53.38Qwellkonfuzed: all of my concentration is being taken right now...sorry
04:53.47brenda$200 is cheaper than a 7960
04:53.57konfuzedits one thing to figure it out its quite another to get someone else to agree
04:53.59ManxPowerThe 500 has three "lines"
04:53.59konfuzed;^)
04:54.11konfuzedQwell I know how it is
04:54.22ManxPowerkonfuzed: the ONLY thing a register => does is inform the remote server what your IP address is.  It does nothing else.
04:54.34ManxPowerand is only needed if the remote side does not know the IP address of your server.
04:54.44konfuzedI can't seem to move forward on installing other things cause I can't think of anything else but this broken phone setup
04:55.32konfuzedManxPower: surely it provides the authentication requirement?
04:55.46ManxPowerkonfuzed: How so?
04:55.47Qwellno
04:56.05QwellThats what your user/peer are for
04:56.08ManxPowerkonfuzed: you need a userid and password in the register so the remote servers knows what userid/password to associate with the IP address you announce.
04:56.25ManxPowerbut it does NOTHING else.
04:56.34konfuzedso if the provider has set asterisk to have the DID forward the call to my ip address then I dont need a register line (which sounds like what the provider was suggesting)
04:56.45nwhiti really could use some help with a problem i am having setting up a sip termination provider
04:56.49ManxPowerkonfuzed: correct, but that would be VERY unusual.
04:57.07fearnorsomewhat unusual, not very unusual. we do that for our wholesale clients.
04:57.26fearnorthere's no freaking point if a client has 100+ dids and a colo'd box to bother our proxy with 100 registers
04:57.31ManxPowerfearnor: Yeah, but I've never heard of that for retail clients
04:57.36konfuzedthis admin believes the register line is not needed or at least he said he "believes it is not needed"
04:57.38fearnorindeed.
04:58.07nwhitI am having alot of trouble with this scenario  Sip UA -> Asterisk -> Sip Provider -> PSTN -- the Sip Provider makes the call, but no rtp stream appears to get setup
04:58.18fearnordo you have a firewall
04:58.21fearnor[yes]
04:58.24nwhitno
04:58.30konfuzedwell he could very well have been setting up a very flexible account for me
04:58.31fearnorput your UA outside of firewall
04:58.36fearnorktnxbye
04:58.56fearnorput *everything* outside firewall. make sure it all works. then deal with nat issues.
04:59.03nwhitthis works fine though  sip ua -> asterisk -> pstn
04:59.04ManxPowerkonfuzed: not using register REDUCES flexibility since you can't change IP addresses on the fly.
04:59.18konfuzedin my case I would be fine with that
04:59.19fearnornwhit: you have firewalls
04:59.36nwhityes... sorry there  is a firewall between ua and asterisk
04:59.42fearnorwell then
04:59.44fearnordisable fireall
04:59.49fearnorktnxbye :)
05:00.02fearnorand read the wiki about nat/firewalls/reinvite etc
05:00.05konfuzedits coming into a static ip and pf.conf on openbsd hard maps it to my lan server ip
05:00.15nwhiti have
05:00.27opus_bugz  bugz
05:00.56nwhitwhy would sip ua -> asterisk -> pstn work but not sip ua -> asterisk -> sip term provider -> pstn no
05:01.08fearnorREINVITE
05:01.11opus_nwhit - you got canrevinte wrong
05:01.22fearnoryour asstricks is telling UA to deal directly with sip provider
05:01.35nwhiti have it set canreinvite=no on both ua and the user for the sip provider
05:01.36fearnorand UA can't get there from here [or other way around] cause of firewall
05:01.48*** part/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com)
05:01.53fearnorthen you screwed something up.
05:02.17opus_whats sip term provider
05:02.22nwhitlevel3
05:02.24opus_can you get asterisk -> sip term provider
05:02.31opus_level3 does your voip?
05:02.38nwhitwill be
05:02.40opus_thast your problem:)
05:02.46fearnorreseller of a customer of an aggregator of l3, i bet ;P
05:02.47nwhitwe do alot of minutes
05:02.57fearnoryou do lots of minutes but cant figure out reinvite? :)
05:02.57opus_how much
05:02.58konfuzedok then so perhaps I should just try out this scenario with the register line at all and see what changes
05:03.02nwhitfearnor,  no level3 itself
05:03.18konfuzedok then so perhaps I should just try out this scenario WITHOUT the register line at all and see what changes
05:03.42nwhitfearnor,  must be
05:03.43opus_you can always dial(SIP/user:password@provider.com/18005551212)
05:03.48dijungali added a user to the end of the sip.conf sample file:
05:03.49dijungal[test1]
05:03.49dijungaltype=friend
05:03.49dijungalusername=test1
05:03.49dijungalsecret=password
05:03.49dijungalhost=dynamic
05:03.51dijungalcontext=tester
05:03.56fearnordij: pastebin.com plz
05:04.00nwhitfearnor,  but i still could use help
05:04.08dijungalbut when i try to register it saying no registraion for user test1
05:04.11fearnornwhit: paste your sip.conf on pastebin.com
05:05.08konfuzedCLI> Destroying call '6827b82f58345553146e38cb16561d44@192.168.1.136'
05:05.12dijungal?
05:05.18konfuzedso without the register line I now get the above message
05:05.29konfuzedwhy is it trying to initiate a call????????
05:05.42opus_no idea.
05:05.43konfuzedits not setup to autocall out anywhere?
05:05.50opus_i see thousands of those fly by per second
05:06.02konfuzedhhmmm
05:06.03opus_the GOVERNMENT
05:06.18fearnorhaxors
05:06.19opus_e.t. phone home
05:06.33fearnorhaxors callin inmarsatte at 10$/minute
05:07.07konfuzedwell that was interesting
05:07.22opus_its osama
05:07.42*** join/#asterisk srt (~nobody@gw0-cgn.reucon.net)
05:07.52konfuzedI call my did and now I get more useful info;
05:07.56konfuzedFrom: "4164613806" <sip:4164613806@66.96.30.25>;tag=as0cc655ad
05:07.56konfuzedTo: <sip:64.119.127.99>;tag=as754d55d1
05:08.10nwhitfearnor,  just pasted as nwhit
05:08.29konfuzedthe to: address I would expect to be 64.119.114.148
05:08.54opus_sheit
05:09.00opus_if you figure that one out
05:09.07konfuzedbut the freaky thing is this line
05:09.10opus_i think they're just rouge udp packets..
05:09.11konfuzedContact: <sip:64.119.127.99@64.119.114.148>
05:09.30fearnorit looks alright.
05:09.35konfuzedthose are both public ips on my system
05:09.56fearnornwhit: i have a feeling something doesn't match something else and calls aren't associated with right peers
05:09.58nwhitfearnor,  yes... but it doesn't work
05:10.23nwhitfearnor,  how could i diagnose?
05:10.32fearnoroutput from console
05:10.34opus_man sometimes i think mysql is more buggier then asterisk..
05:10.41fearnoropus: not possible
05:10.46nwhitfearnor,  i have looked at the sip call setup, but i don't see anything weird
05:10.47fearnorasterisk is buggier than *
05:10.53konfuzedwould that be bcause somehow when my box talks to providers box that my box reports 64.119.127.99   or that some config on the providers box is telling it to call out to 64.119.127.99 ?????
05:11.07fearnorkonf: latter probably
05:11.11nwhitfearnor,  i see the rtp setup between the ua and asterisk... but absolutely no rtp from level3
05:11.19fearnorsip debug
05:11.24fearnorsee if there are reinvites involved
05:11.33nwhitnone
05:11.38fearnorpaste sip debug
05:12.51nwhitjust did
05:12.56konfuzedIf I may reuest a corroboration that Contact: <sip:64.119.127.99@64.119.114.148>  indicates a provider side config issue
05:13.13fearnorthats from you to (3)
05:13.17fearnorwhat happens then
05:13.23nwhitit restransmits several times, but i never see anything come back from level3
05:13.34nwhitnever see anything come it
05:13.35nwhitin
05:13.38nwhitits weird
05:13.47*** part/#asterisk Mavvie (edwin@252-131-222-203.rev.techex.net.au)
05:13.53*** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com)
05:13.55fearnordoes it work to (3) directly from different ua?
05:14.06fearnori mean, maybe (3) just doesn't love you, period.
05:14.39nwhitthat won't work because it is by ip, no username/password
05:14.53fearnorand you sure that the IP in question is permitted?
05:15.07nwhitfearnor,  i am trying to verify that with them
05:15.15fearnorwell
05:15.19nwhitfearnor,  but the call is made ... it rings on the other end
05:15.25fearnorstop aksing questions here then ;)
05:15.28nwhitno rtp
05:15.33fearnorum
05:15.36fearnorif invite isn't acked
05:15.40fearnorhrm
05:15.47fearnori think you have firewall that filters l3's response
05:16.00nwhiti like to rule out everything before i look like an idiot before (3)
05:16.07nwhitno firewall
05:16.12nwhitin front of asterisk
05:16.20fearnorshrug
05:16.21nwhitfor now at least
05:16.22fearnortime to sleep
05:16.34nwhityeah... hopefully it is on their end
05:16.42nwhitand its simple
05:16.57nwhiturg
05:18.00nwhitgnight
05:19.04dijungalok me now..
05:19.56dijungali gots this uers i have setup in the sip.conf but when i try to login with the softphone asterisks says "Noe registration for peer 'test1'"
05:20.01dijungalwassup wid dat man.. :s
05:20.09fearnorok
05:20.14fearnori bet you misspelled something somewhere
05:20.21fearnorktnxbye
05:20.28dijungal<PROTECTED>
05:20.29dijungal[test1]
05:20.29dijungaltype=friend
05:20.29dijungalusername=test1
05:20.29dijungalsecret=password
05:20.29dijungalhost=dynamic
05:20.31dijungalcontext=default
05:20.33dijungalnat=yes
05:20.34opus_ahh
05:20.35dijungal<PROTECTED>
05:20.37dijungal[test2]
05:20.39dijungaltype=friend
05:20.40ManxPowerDON'T FLOOD THE CHANNEL!
05:20.41dijungalusername=test2
05:20.43dijungalsecret=password
05:20.43ManxPower!pastebin
05:20.45dijungalhost=dynamic
05:20.47dijungalcontext=default
05:20.48ManxPower~pastebin
05:20.48jboti guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
05:20.50opus_op me
05:21.06dijungalwhat is pastbin..?
05:21.11dijungal!pastbin
05:21.17dijungal~pastebin
05:21.17jbotwell, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
05:21.21drumkilla/mode +o opus_
05:21.28drumkillahm ... won't go through for some reason.
05:21.31opus_.mdeop
05:21.34drumkilla:-p
05:21.38Qwelldrumkilla: You need to add the channel
05:21.38blitzragedrumkilla: yer still up?
05:21.41Qwell:p
05:22.02drumkillaQwell: joke, silly head!
05:22.19Qwellduh Lo
05:22.22Qwell:p even
05:22.22blitzragedrumkilla: thought you said you were going to op me yesterday? :)
05:22.32drumkillawaiiit.
05:23.03*** mode/#asterisk [-o blitzrage] by drumkilla
05:23.21blitzragelol
05:23.34opus_so you guys all heading to chicago
05:23.41opus_i can't go, to much work
05:23.51drumkillaI will not be in attendance
05:24.00dijungalhttp://pastebin.ca/18755
05:24.13opus_but you got the most cvs points
05:24.20DarthCluedue to an overabundance of geek power, chicago is now closed.
05:24.31konfuzedhhhmmmmmm I think I'm going to try setting the firewall to map those two public ips to the oposite ethernet cards
05:24.31dijungalany ideas guys...?
05:24.50konfuzedand see if that clears up the problem with the wrong ip
05:24.55konfuzedgoing out
05:26.50dijungalJul 28 22:35:17 NOTICE[4774]: chan_iax2.c:3910 register_verify: No registration for peer 'test1'
05:27.04dijungalbut it's a sip registration
05:28.20opus_asterisk realtime extensions is somewhat brokenn..
05:29.17dijungalgeez.. no once can help.. :S
05:29.22dijungalno one
05:29.24opus_fluffy little clouds
05:29.25brendaael is brokened too :(
05:29.40opus_i can't get realtime extensions to find 's' or 't'
05:30.19ManxPowerdijungal: try during the day USA time
05:30.23opus_Jul 28 22:19:23 NOTICE[20291]: pbx.c:1698 pbx_extension_helper: No such label 'from-pstn-rt' in extension 't' in context 'from-pstn'
05:30.26opus_mofo
05:30.43dijungalhmm..
05:30.50*** part/#asterisk dijungal (~ovr@206.113.106.27)
05:30.58brendaseptember 1 is a lofty goal
05:31.22*** join/#asterisk lehel (~lehel@82.79.20.17)
05:31.30ManxPowerbrenda: Well, I don't recall him mentioning Sept 1 of what year.
05:31.31lehelhello
05:31.41brendaManxPower: lol
05:32.04brendaI think some people are afraid to report bugs
05:32.12opus_anyone ever implement app_dbodbc?
05:32.32ManxPowerbrenda: Oh I am afraid to report bugs.
05:32.37opus_what do i do with func.rtdb.c
05:32.46brendaManxPower: someone bite your head off too?
05:33.23brendaI just hate when I report a bug and nobody believes me
05:33.45ManxPowerbrenda: mostly I report bugs on 1.0.x, but then everyone tells me to "try it in CVS-HEAD".
05:34.06ManxPowerI'm not reporting bugs for CVS-HEAD, I'm reporting them for 1.0.x 8-)
05:34.06opus_Segmentation fault (core dumped)
05:34.17brendaeven after 1.2?
05:34.37ManxPowerWhen 1.2 is feature frozen I'll try it out on my personal Asterisk box.
05:35.05DarthClueManxPower: yeah, that does suck.  especially since 1.0.x is supposed to be feature frozen and in bug fix mode.
05:35.05blitzrageI've been using HEAD exclusively at home :)
05:35.28ManxPowerDarthClue: That's WHY I use 1.0.x
05:35.41brendaall our new projects are HEAD
05:35.45ManxPowerWhen 1.2 gets feature frozen and bug fixes only, then I'll try that.
05:35.56opus_1.2 will be a mess
05:36.01ManxPowerThere are many features I want that's in 1.2
05:36.11opus_somebody will need to rm -rf voip-info.org, cuz they'll be way to confused
05:36.14ManxPoweropus_: I agree, but I still have to try it when the time is right.
05:36.20DarthClueManxPower: i don't like using stable, but after 1.2, HEAD won't be stable enough to use.
05:36.30opus_1.2.1 haha
05:36.36Corydon76-homeI dunno why people are afraid to report bugs... It's not like I bite their entire heads off...
05:36.45blitzrageopus_: thats why we wrote the book based on 1.2 instead of stable
05:36.56opus_good
05:37.02brendaCorydon76-home: how much of their head to you leave?
05:37.16DarthCluemy 3 year old likes to bite the heads of off crickets.  any crickets in here?
05:37.18Corydon76-homebrenda: enough to grow back
05:37.27drumkillaI give no mercy!!
05:37.39Corydon76-homebrenda: but slowly
05:37.40opus_cvs update book
05:37.46brendabrenda: that's nasty
05:37.55Corydon76-homedrumkilla: no quarter given?
05:37.59ManxPowerI was VERY disapointed with how one of my bug reports turned out recently.
05:38.14DarthCluebrenda: yes it is.  but i can't convince her of that.
05:38.18opus_Manx which one?
05:38.22brendaCorydon76-home: talking to myself about the image of you biting 'heads'
05:38.29opus_I'm totally fucked on this Polycom dtmf one, its a real show stopper
05:38.30*** join/#asterisk outtolunc (~me@adsl-69-110-15-184.dsl.pltn13.pacbell.net)
05:38.48Corydon76-homebrenda: I don't bite that head...
05:38.48opus_the bug id is basically rotting away
05:38.56blitzrageopus_: that's kind of how the asterisk docs project works - I check in sections of the book via CVS (we use DocBook)
05:38.59fearnorwell, like linux, it'll probably be stable at 1.2.13 :)
05:39.00ManxPowerI spent a day gathering additional information for the bug report, being told that "that's just the way Asterisk works", only to discover that not one of the 3 or 4 people commenting on the bugs realized that the problem was a lack of /etc/asterisk/indications.conf
05:39.19brendaCorydon76-home: me neither
05:39.20opus_blitzrage - sounds like how they wrote bsd unix
05:39.24Corydon76-homeouttolunc: chomp, chomp
05:39.30fearnorberzerkeley
05:39.41ManxPowerRather embarassing for me, but really disapointing nobody realized that was the problem.
05:39.54fearnorat least bsd unix was graduate students, not highschool kids like astrerisk
05:39.56fearnor:)
05:40.26brendaouttolunc: someone ate your h?
05:40.27opus_yeah.. well its pretty bad now days anyways. like i found two exploits in centos the other day
05:40.28Corydon76-homeUh, what's wrong with high school kids contributing?
05:40.35lehelppl what do you think .. how it is possible, that i can call from a softphone, but not with the zap channels plugged phones?
05:40.42outtoluncmany^2 years ago
05:40.47Corydon76-homeI know several who do fine jobs
05:41.03fearnori know far more who are, er, well, just highschool kids. :P
05:41.11*** join/#asterisk Math` (~math@modemcable222.240-37-24.mc.videotron.ca)
05:41.34lehelcould be some context problem ?
05:41.35opus_yo Math`
05:41.49opus_lehel - 'ztcfg' show anything?
05:41.57lehelyess
05:41.57Corydon76-homefearnor: if you can do better, why aren't you?
05:42.13fearnorima too busy making blingbling
05:42.14fearnor:)
05:42.47opus_dumbdumbmode=onn
05:42.49fearnori've done my asterisk contributions
05:43.00fearnorfirst on the list in CREDITS ;)
05:43.43lehelouttolunc: and how do i know wich context i have to define for the zap channels?
05:44.00Corydon76-homefearnor: and what have you done recently?  ;-)
05:44.33lehelnow it's: from-internal
05:44.34brendaCorydon76-home has saved our hides a few times
05:44.39outtoluncwell it would whichever you chose after reading the handout
05:44.45*** join/#asterisk scratchrf (~scratchrf@63-226-200-214.tukw.qwest.net)
05:44.56Corydon76-homebrenda: heh
05:45.10lehel"handout" outtolunc? means..?
05:45.27blitzragebed time, night!
05:45.28outtoluncthe 'draft'
05:45.29lehelfrom-internal.. why it isn't good?
05:45.36outtoluncis it still around?
05:45.40Corydon76-homeI particularly like how somebody wanted to disable the bang command because it was 'dangerous'
05:45.48blitzrageCorydon76-home: :D
05:46.08drumkillaoh well, i'm sure it's like a 3 line patch :)
05:46.11Corydon76-homeYet you can still do CLI> add extension 9999,1,System,"rm -rf /" into internal
05:46.18brendaWhy don't they comment it out if they don't want it?
05:46.20*** join/#asterisk |nix (~inix@218.208.24.248)
05:46.37drumkillabrenda: what if we said that about everything ?  :-p
05:46.54brendadrumkilla: true true
05:47.02ManxPowersome should write a "safe" CLI using just the manager interface.
05:47.12Corydon76-homedrumkilla: there are far worse things than the band command.  See above.
05:47.20Corydon76-homes/band/bang/
05:47.26brendadrumkilla: we could dump half the apps too
05:47.30outtoluncer handbook-draft
05:47.43scratchrfis there an iax softphone that runs in a web browser?
05:48.05Corydon76-homeManxPower: or asterisk -M  M for morons...
05:48.21drumkillahow about a web browser inside of a softphone ?
05:48.29outtolunchttp://www.digium.com/handbook-draft.pdf
05:48.34ManxPowerI was thinking more along the lines of Asterisk CLI Terminal.
05:48.37scratchrfyeah, that too :D
05:48.41drumkillathe new gnophone has that :)
05:48.44lehelthanks outtolunc, got it
05:48.52outtoluncreading it helps
05:48.56Corydon76-homedrumkilla: there's a new gnophone?
05:49.12drumkillaCorydon76-home: well, goatmilk had been working on it and his stuff is in digium cvs
05:49.20outtoluncthe gist is, you can assign 'any' context you want to whatever
05:49.28drumkillaCorydon76-home: to use iaxclient and gtk2
05:49.40Corydon76-homedrumkilla: Ah, interesting...
05:51.40brendathat self destruct extension is good for clients that stop paying
05:51.57lehelouttolunc: is there a possibility to check remotely (or by sofphone) if the context works for the zap channel?
05:52.14Corydon76-homebrenda: actually, that's not a bad idea...
05:52.33brendaJust afraid I might accidently test it
05:52.46Corydon76-homebrenda: so add an Authenticate step to it
05:53.02brendanice!
05:53.27outtoluncmaybe i did too many shots this evening, but i remember 'lehel' as being a nick that has been here before and as such you probably realise to 'test' a context requires 'having it to begin with and an exten to use it'
05:53.54outtoluncso if this is 'attempt to mess with otl night' not gonna happen
05:56.50*** join/#asterisk wasim_ (~wasim@wasim.active.supporter.pdpc)
05:57.32*** join/#asterisk Ayano (~erik_leee@adsl-70-245-190-90.dsl.spfdmo.swbell.net)
05:57.47*** part/#asterisk tuxinator_linux (~tuxinator@ip68-109-146-168.ph.ph.cox.net)
05:58.01Ayanoany asterlink guys on here?
05:58.37opus_nope. we're all microsoft live server guys
05:58.40opus_:)
05:58.56brendaand gals!
06:00.07opus_dude they totally changed the API, i got to rewrite my vb shit
06:00.10outtoluncer s/o/i
06:01.06brendaI'm just happy it runs on my xbox
06:02.53opus_i think eventually asterisk needs to rewritten
06:03.09opus_whats this 'modem' module that it just has to have?
06:03.57fearnorif something needs to be rewritten its the media handling stuff
06:04.12opus_audio/video?
06:04.31fearnoryes
06:04.45fearnorwould love dialplan to be able to deal easily with media
06:04.50lehelouttolunc: maybe just my english is not that expressive ;p
06:04.52brendafearnor: when are you going to start?
06:04.58fearnorlike pipes on unix.
06:06.49outtolunclehel, in zapata.conf you setup the channel and set a context
06:07.04outtoluncso
06:07.23outtoluncif you lift the handset at anytime that 'context' is triggered
06:07.29outtoluncso s,1
06:07.52outtoluncplayback(screaming-monkeys) or whatever
06:08.30outtoluncand if you read the handbook-draft as i suggested it will help
06:09.51*** join/#asterisk pauldy (~pauldy@c-67-166-175-153.hsd1.tx.comcast.net)
06:10.44konfuzedfearnor: do you provide DID services through your own company or one that you work for?
06:12.23*** join/#asterisk PyroSteve (~steve@wsip-70-183-114-254.no.no.cox.net)
06:12.29PyroStevehello
06:12.43PyroStevecan I get a raise of hands of people who dont like using AAH
06:12.51PyroSteveand most importantly .. WHY ?
06:13.09PyroStevebasically Im looking for opioions
06:13.13PyroStevespelling ?
06:13.52*** join/#asterisk darkskiez (~mhb@host-84-9-85-42.bulldogdsl.com)
06:14.17*** join/#asterisk argos73 (~mike@65-85-207-101.client.dsl.net)
06:14.25darkskiezhow does digiums new firmware squeeze more lines into a T1/E1?
06:14.26PyroStevecome on .. this should strike up a good some good conversation !?!
06:14.42Qwelldarkskiez: not more lines.  less load
06:14.50konfuzedfearnor: Are you in Canada
06:14.59konfuzedsorry
06:15.04*** join/#asterisk oej (~oej@apollo.webway.se)
06:15.07konfuzedcurious
06:15.11darkskiezthey say they can get 150 channels to the PSTN on a 4port card!
06:15.22twisteddarkskiez, no they don't
06:15.39twisteddarkskiez, they say that asterisk can TRANSCODE 150 channels of g729 with the new firmware
06:15.47outtoluncif so, never used it, seems like it's fine 'for HOME users'
06:15.50darkskiezFor example, a dual-processor, 3-GHz 800FSB Intel XEON server with 1MB L2 cache, and a Digium 4-port T1/E1 card, can now convert 120 SIP channels with G.729 compression to the PSTN without Digium's echo cancellation module and 150 channels with G.729 compression with the Digium echo cancellation module.
06:16.50twistedoh
06:16.51twistedhmm
06:17.04PyroStevethats referring to the load a server of those specs can handle ?!?!?
06:17.27QwellPyroSteve: transcoding channels
06:17.57PyroSteveright ..i think that what Im trying to say
06:18.36PyroStevethe !?!?!? means Im pretty sure my statement is correct, but could be wrong .. and please correct im if wrong
06:18.59darkskiezi know what they meant, but i think their press release is wayyy too vague
06:19.10PyroSteveahh ok
06:19.24lehelthe codec problem could be a possibility.. k outtolunc
06:19.42PyroStevehas anybody here tried to use AMP and finally said "no way man" and threw it in the garbage
06:20.37lehelyou did PyroSteve?
06:21.12PyroStevelehel: almost
06:21.25*** join/#asterisk Stephnie (dfsdf@203.215.180.254)
06:21.45PyroStevelehel:  you handle is spelled the same way both ways
06:21.51outtolunclehel, my suggestion to you is to reduce the complexity.. meaning just use 2 boxes, try to get those 2 to talk.
06:22.03Stephniehi all
06:22.15PyroSteveHi Honey
06:22.52PyroSteveStephnie: Was your daddy a thief ?
06:22.54StephnieHoney!!!!! :D
06:23.10StephnieYes!!!! how do you know that?
06:23.12Stephnie:p
06:23.58argos73i hate this...  getting a new voip phone sys at work - the geek in me says "go asterisk", but the responsible business side of me is saying "cisco call manager"...  grrrr..
06:23.59PyroSteveBecause I want to know who stole the stars out the sky and put 'em in your eyes
06:24.14argos73(they worry about what happens when I'm hit by a truck)
06:24.21Stephnieaaaawwwwww!!!
06:24.31*** join/#asterisk jiro5281 (~anton281@203.177.242.192)
06:24.44darkskiezargos73: "Job Security"
06:24.49argos73hehe  :)
06:25.38Stephnieok now .. coming out of my eyes....and I need to know...
06:25.41PyroSteveStephnie: How can we help ?
06:25.48argos73our billing vendor is VERY pro-microsloth, and they keep telling my bosses that my way is "bad"
06:26.40PyroSteveargos73: your way ? as in using * ?
06:26.47Stephnie"5551212@myasteriskip"   .....I need to save 5551212 in a variable and then use this variable again in another context.
06:27.13tengulreHi,all! which channel is discussion Microsoft Visual Studio ?
06:27.29PyroStevetengulre: not this one !
06:27.38argos73PyroSteve, UNIX-based solutions in general...  especially any software that I wrote myself...
06:27.46PyroSteveStephnie: look at SetVar
06:27.59PyroSteveor better yet ... just Set()
06:28.03argos73tengulre, #hourlyblowups
06:28.13PyroStevehehe
06:28.23twistedtengulre, #1,000
06:28.50StephniePyroSteve: alright...I check wiki
06:29.20PyroSteveargos73: well give them an agrument ... show them the end product... and the cost ... then have them give you a reason why its not the right solution
06:29.27konfuzedhey any canadian DID providers with a good reputation????????
06:29.54Stephnietengulre: what kind of information you need to know about MS Visual Studio ?
06:30.00PyroSteveStephnie: you basically you set to define a variable and then refer the variable as ${VARIABLE_NAME}
06:30.10PyroStevethrough out your dialplan
06:30.29PyroStevethere are global vars, channel vars, and one or two other types of vars
06:31.05StephniePyroSteve: ok...should I check wiki for an example? or you gonna help me in that too ;)
06:31.07argos73PyroSteve, that's usually what I do...  they accept that reasoning for a while, then come back later and ask "why aren't we running windows?".. again.  and again.  and again..
06:31.31PyroSteveStephnie: Im be glad to you help you !
06:31.36opus_Yoyo
06:31.49PyroSteveargos73: yeah, i hear you
06:32.07StephniePyroSteve:  so sweet of you ...kekeke :-)
06:32.08blitzrageI hate not being able to sleep
06:32.30opus_shit. mysql doesn't support jack shit
06:32.44opus_i need to create a view.
06:32.52argos73PyroSteve, wrote a voice mail system a few years ago for our Merlin Legend system - works fine, but people are always blaming "i got cut off when listening to my messages from a cell phone" on the fact that I wrote it...  go figure.  :)
06:33.17argos73as if I can control where they're driving....
06:33.41opus_i had a client onetime ask me to change the weather
06:33.45opus_jk
06:33.57opus_Whats up with the 'comedian mail' stuff
06:34.07opus_did you write that shit?
06:36.02konfuzedhey is there perhaps some kind of test account server whereby I can test if I can make this setup work with some other sip-server as test to indicate a service provider problem at the server I should be conecting too.
06:37.25opus_testvoip.com or something, hmm
06:37.32opus_testmyvoip?
06:37.39konfuzednot FWD  but something more like how a pay servce provider would have thissetup
06:37.42konfuzedoh wow
06:37.50konfuzedI'll have to find that one
06:40.55opus_nice... apparently i'm getting 60mb/sec to my hd
06:41.06opus_Timing buffered disk reads:  174 MB in  3.01 seconds =  57.74 MB/sec
06:41.12opus_pretty good for ibm 206
06:41.43limbiquehi
06:42.59*** join/#asterisk Kernel_core (Raph@217.218.94.169)
06:43.18konfuzedwell interesting sites   testvoip.com and testyourvoip.com  but neither offer an account I can connect to to send or recieve test calls
06:44.37Kernel_corehi all , is there any Cisco IOS relase that support iax ?!
06:44.57PyroStevewhats wrong with FWD ?
06:45.38PyroSteveopus_: are you looking for a PSTN test gateway ?
06:46.09opus_pyrosteve no konfuzed is.
06:46.24PyroSteveoh sorry ..
06:46.46opus_i run a provider, thank you:)
06:47.16PyroStevehehe ... which company ?
06:49.10*** join/#asterisk stkn_ (nobody@stkn-active-pdpc.developer.gentoo)
06:49.44opus_amd is dog slooowww
06:50.08Kernel_coreopus_: which Model of AMD?
06:51.05outtoluncwhy even ask <G>
06:51.06postelKernel_core: Cisco IOS that "supports" iax? All cisco routers can route iax traffic
06:51.42Kernel_corepostel: I know Cisco IOS support MGCP , SIP , H323 , but does it support IAX Protocol ?
06:51.54opus_Kernel - i've only used amd64 3200 and this is a sempron 3000 socket a
06:51.54NixKernel_core: no
06:51.55postelKernel_core: nope
06:51.56Kernel_coreI mean for telephony , not routing !
06:52.00postelnope
06:52.04NixKernel_core: use sip
06:52.11outtoluncKernel_core: just open that port.. instant support
06:52.39konfuzedopus_:  PyroSteve:   actually I just want test if the only thing wrong with my setup is the did provider, one good way to do so is to try it with another provider who really knows asterisk properly
06:53.10lehelotl it's not a codec problem.. when he wants to dial, phone hangs up .. context:(
06:53.20konfuzedActually in the end I want to setup with multiple DID providers for a certain kind of fialover redundency
06:53.46opus_thats cool
06:54.39outtolunclehel: are you talking about the issue you were having before?  if so that was just 'one' possibility out of many for why 'an exten 'just hangs up''
06:54.59Kernel_coreNix: I know , the only/best way is SIP to connect asterisk to cisco
06:56.21outtolunci never knew one would 'connect sip to a cisco device' unless that device was 'a gateway'
06:56.26outtolunchello
06:56.29konfuzedopus_: can you provide a Canadian DID Service on IAX or SIP ???
06:56.48opus_ummm, I don't know.
06:57.08opus_Canada never even bothered to check that
06:57.32*** join/#asterisk grimse (~grimse@p5481BFA8.dip.t-dialin.net)
07:02.02*** join/#asterisk Zeeek (~icechat5@Zeeek.active.supporter.pdpc)
07:06.16konfuzedCanada is so UnderRated ;^)
07:06.56*** join/#asterisk Uther_P (~uther_p@66.180.120.82)
07:07.10*** part/#asterisk Uther_P (~uther_p@66.180.120.82)
07:08.50opus_dude i'm leaving usa for next month to go to canada
07:08.57opus_recommend any places to go camping?
07:09.18argos73downtown toronto?
07:09.18argos73:)
07:09.35opus_west side..
07:09.49outtoluncgoogle 'bigfoot sightings' <G>
07:10.03opus_i'm convinced there's going to be some uber terrorist attack cuz bush is on vacation again, time to get outside the blast zone
07:10.34opus_my friends dad debunked bigfoot
07:11.12outtoluncwas there a furry costume in the corner of the garage growing up?
07:11.25opus_haha
07:11.39argos73there's a furry something in my refrigerator growing up...
07:11.51argos73think it used to be a tomato.
07:11.54outtolunclet it grow
07:12.24opus_its frig resistant now
07:12.29outtoluncit's only the ones that 'stink' you should toss out <G>
07:12.33*** join/#asterisk Gunnar (~gunnar@bkkb-gw.bitcon.no)
07:14.20argos73can buy some TV time and sell it as the newest "male enhancement elixir"
07:14.46argos73i'm sure somebody would bite...
07:16.02argos73any comments re: the new Digium T1 cards w/hardware echo-cancellation?
07:17.08*** part/#asterisk lehel (~lehel@82.79.20.17)
07:17.28*** join/#asterisk lehel (~lehel@82.79.20.17)
07:18.21opus_don't know, never touched analog shit
07:18.27*** join/#asterisk monkmole (~monkmole@208.187.28.250)
07:19.22monkmoleI'm sad to say that nothing short of a kernel panic has me here today looking for help :(
07:19.29hardwirehttp://www.motorazr.com/home/p2_articleid/22
07:19.31hardwirejust nuts
07:20.27opus_blah, that phone will be like $800
07:20.36opus_not somthing i'd want to lose
07:20.36*** join/#asterisk franekstein (~frankenst@S01060008c7c072ee.ed.shawcable.net)
07:20.55opus_i'd rather use microsoft phone for voip
07:22.53monkmolemodprobe wcte11xp works OK on centos 4.1 and rmmod wcte11xp gives a kernel panic everytime.
07:23.36shidoouch
07:23.42shidothat must really hurt...
07:23.48monkmoleThats on Asterislk and Zaptel etc... 1.0.9
07:23.50shidocvs had? monkmole ?
07:23.52shido+e
07:25.30monkmoleshido: Guess I don't know what you are asking with the "cvs had?"...
07:26.21drrayhead
07:26.35drray(i'd wager)
07:26.35monkmoleahhh
07:26.59monkmolegetting late here and the grey matter is slowing.
07:28.37outtolunctry, grey matter marinated in whiskey
07:28.43shidocvs head
07:28.48shidoare you runnign cvs head, monkmole
07:28.48shido?
07:29.20outtoluncfrom a week ago
07:30.04monkmoleshido: Well I don't know what head is at but I'm running the latest tar (1.0.9)
07:30.20drraymonk did you check it out from cvs
07:30.21drray?
07:30.42outtoluncnotes to drray, you can 'checkout' either
07:30.49opus_uh oh
07:30.51opus_<PROTECTED>
07:31.01drraysure, but you can't check out stable, without knowing about it
07:31.03monkmoleno wget from "http://www.asterisk.org/html/downloads/asterisk-1.0.9.tar.gz"
07:31.30opus_Although officials said the shuttle Atlantis could be launched as part of a rescue operation and was being held "on readiness", they stressed they were "nowhere near doing that".
07:31.34monkmoleI hate cvs
07:31.51lehelheh
07:32.11drrayinstalling asterisk via cvs works well for me
07:32.19outtoluncdo you want to know what i hate?
07:32.25drraypeople who hate?
07:32.35outtolunchehe close <G>
07:32.40monkmoleThe last thing I wan't to do is go OT when trying to get help with a kerel panic that needs to be fixed by morning but......
07:32.57monkmolewhen is Asterisk going to get with the program and offer svn?
07:32.58*** join/#asterisk IronHelix (~irc@ool-45785cfe.dyn.optonline.net)
07:33.11drraysvn being subversion?
07:33.15outtoluncbut yes those that say 'something MUST be fixed by morning' <G>
07:33.28konfuzedIronHelix: How serious is chocolate?
07:33.32shidotar?
07:33.43shidomonkmole, try using s/w out of cvs
07:33.57drraytry STABLE fork as well
07:34.53outtoluncmy first question would be did you post a 'bugs' entry for this 'issue'
07:35.10monkmolesorry I'm so slow shido, but what is s/w?
07:36.11monkmoleouttolunc: I will when I know what the issue *is*
07:36.27drrayhas this ever worked right?
07:36.39monkmolewell - once
07:36.50outtolunccan you at least give us the 'fainted clue' as to what you did before it PUKED
07:36.58outtoluncer faintest
07:37.18shidosoftware
07:37.20shido= s/w
07:37.21monkmoleone test call and then I was going to make sure it was up in the event of a reboot - after rebooting - nada.
07:37.58drrayyou did modprobe zaptel, your hardware, then ztcfg -vv?
07:38.03monkmolewell I did the stuff that you need to do for systems that use udev
07:38.05outtoluncwell it sounds to me like you aren't even sure if the proper drivers/modules are in place
07:38.39monkmoleouttolunc: sure I'm sure
07:38.51drrayI did a yum update one time that broke asterisk and I had to make clean to fix it, I'm still pleased that I figured that out in 10 minutes
07:39.00opus_fuck why can't asterisk send DTMF tones right
07:39.21opus_inband doesn't work, nor does rfc2883 or whichever.
07:39.37outtoluncif you are so sure, why not tell us what hardware you have and what is listed for lspci
07:39.40outtoluncand such
07:40.39drraywhen I was first starting with asterisk i used the livecd to verify that it was working before I started blaming asterisk for my woes
07:40.43monkmolemodprobe zaptel; modprobe wcte11xp; modprobe wcfxs
07:41.07outtoluncyes that 'looks' nice
07:41.13outtoluncand what was the result
07:41.43monkmolewell one TE110P and one TDM04B
07:41.48outtolunc(i'm still wondering why you had to do that manually) but what the hey
07:42.41outtoluncand actually suppling the lspci ???
07:42.43shidoopus - it sends it just fine, whats wrong , tell me more about your setip
07:42.46shidosetup
07:43.04monkmole`lsmod | grep wc` gives "wcfxs 32160 0   wcte11xp 27808 0   zaptel 205956  2 wcfxs,wcte11xp"
07:43.04outtolunchell even a dmesg subset
07:43.07outtoluncsomething
07:43.28monkmoleI can only type to fast.... ;)
07:43.45outtoluncand the lspci
07:43.52monkmole00:10.0 Network controller: Unknown device e159:0001
07:43.52monkmole00:11.0 Communication controller: Unknown device e159:0001
07:44.03outtolunchmmm
07:44.30opus_shido - http://bugs.digium.com/view.php?id=3847 I think thats my problem
07:44.45drraydid you do ztcfg -vv after you loaded your modules? before starting asterisk?
07:44.51outtoluncthe fact that both your cards are sharing the same irq i a prob
07:44.59monkmolethe card where known when I was running FC2 but CentOS 4.1 doesn't see to "know" what they are...
07:45.09opus_shido - polycom, asterisk cvs head, SER with hardware gateway. Dtmf tones are only a few milliseconds no matter what i do. Not enough to send a full dtmf tone.
07:45.19opus_for example, 1800callatt won't accept any dtmf
07:45.32*** join/#asterisk Kraven (kraven@inetandmore.com)
07:45.38monkmole<PROTECTED>
07:45.38monkmole<PROTECTED>
07:46.11monkmolesorry - sleepy
07:46.24monkmole<PROTECTED>
07:46.53opus_arge
07:46.57monkmoleso they don't share with each other but wcte11xp shares with via
07:47.08outtoluncfirst i would disable all the bs like lpt/serial stuff in the mb bios
07:47.20drrayUSB
07:47.20outtoluncopen up some space
07:47.21monkmoleyeah - I'll do that now
07:47.22shidoopus what codec are you using?
07:47.28opus_ulaw
07:47.29outtoluncthe rerun it
07:47.55outtolunchaving your t1 card share an irq is not a good thing
07:48.17outtoluncthis is 'common' stuff
07:49.03opus_http://pastebin.ca/18757 <-- my sip show channel channelid
07:50.58shidohrmmmmmmmm
07:51.14jalsothi
07:51.37opus_here is another guy talking about it
07:51.39opus_http://lists.digium.com/pipermail/asterisk-users/2005-March/096925.html
07:52.00opus_if you hit the dial pad REALLY fast it works
07:52.19outtoluncdialpad on what client
07:52.19opus_or if you put your cellphone up to the mouth peice and send it truely inband it also works
07:52.27opus_on the polycom ip500 phone.
07:53.05outtoluncnever used one of those
07:53.43outtoluncbut on others, especially softphones, there is a huge difference in the 'timeframe' for dtmf
07:54.07outtoluncso much so, that 'transfering' fails
07:54.23lehelouttolunc: it's working!;)
07:54.41jalsotdoes anybody know what causes 'Bad file descriptor' message when using FastAGI? Jul 29 09:38:55 WARNING[896] res_agi.c: Connect to 'agi://127.0.0.1/callhangup' failed: Bad file descriptor
07:54.43outtoluncmeaning, if you wanted for 'transfering' to say, too late
07:54.45lehelwe are the champs:-P
07:55.09outtolunclehel, congrats
07:55.37outtolunci'm a dweeb
07:56.39lehelnow can we make an outside call with capi? ;P
07:58.05lehel:))
07:58.18outtolunche's happy
07:58.35outtoluncso what about opus
07:59.44opus_i'm hard coding it from 800ms in rtp.c to 1800 ms
07:59.48opus_grandma style
08:00.21opus_spaghetti and meatball code
08:00.29outtoluncwalker on quicksand?
08:00.49outtoluncjust wondering why the 1000ms delay
08:00.51opus_i can't believe nobody else has this problem...
08:01.05opus_there has to be other people using polycom phones with asterisk/ser
08:01.17outtolunci'm sure there are
08:01.27outtoluncit's probably something else
08:01.41outtoluncis this a 'remote' client
08:01.48lehelouttolunc: i think it is not a big deal (dial out)
08:02.07opus_its up to asterisk to tell SER when to start the dtmf tone and when to stop it,...
08:02.18outtolunclehel, yes, 'dialing out' is not a big deal (for most)
08:02.36lehelcapi isdn
08:03.37outtolunclehel, i've heard alot of people scream about capi/isdn, but usually it's a config issue
08:03.55drrayor a lack of clue
08:03.57outtoluncone i'm proud to say i'm GLAD i do not have to deal with
08:03.57lehelcorrect!
08:04.45outtoluncas for opus, where is the ser box in relation to the asterisk box?
08:06.01outtoluncthat far eh.. <G>
08:06.05*** join/#asterisk gres (~serg@81.222.48.242)
08:06.29opus_at my buddies house
08:06.36lehelouttolunc: what are you dealing with? besides '#asterisk'
08:06.38outtoluncpingwise
08:07.25outtoluncnot much, just the same old thing i've been dealing with company wise for what the last 25+ years
08:08.13opus_fuck
08:08.16opus_still doesn't work
08:08.21monkmoleIRQ solved but kernel still panics
08:08.21outtolunchaha
08:08.46outtoluncmonk, are you running 'stock' code for whatever you are running?
08:08.59monkmoleyeah all stock
08:09.02monkmoleyum update
08:09.17outtoluncthen what were the last few lines for the prompt
08:09.20monkmoleit's only 2:00 am here :)
08:09.35outtolunc1:07am here whats your point
08:10.05drrayit's 1:02 here
08:10.10outtoluncyum update of what?
08:10.11monkmoleI'm being upbeat - I have a full 5.5 hours before the office start filling up :)
08:10.38monkmoleI'm just saying the the system is stock and updated
08:10.52outtoluncof what means, what dist you running?
08:11.03drraycentos, why did you ditch fedora?
08:11.09monkmoleCentOS 4.1
08:11.13outtoluncoh joy
08:11.34monkmole6 years of updates vs. 6 months.
08:11.47drrayworking asterisk server vs not working
08:11.48outtoluncdoes centos have the ability to regess kernel in grub like fedora?
08:11.50monkmoleI run FC at home
08:11.50outtoluncif so
08:11.59outtoluncdo it 2 kernels back
08:12.01opus_RHEL rocks
08:12.31outtoluncthe rebuild your zaptel, libpri
08:12.31monkmoleyeah it is RHEL4 without the RedHat logos
08:12.35outtoluncer then
08:12.54outtoluncread ^^^^^^^^^^^^^^^
08:13.23monkmoleI'll give er a swing
08:13.44outtoluncif it 'once worked' that is your best bet
08:14.01outtoluncseriously
08:14.15outtolunci've had similar issues with fc3
08:14.43drrayasterisk live cd is his best bet
08:14.47outtolunchaha
08:15.17outtoluncok lets all just abandon all dev work and just to live cd's
08:15.18monkmolethat runs CentOS
08:15.27opus_how can I turn this debug line on :   if (rtp_debug_test_addr(&rtp->them))
08:15.28lehelppl if i define a global var, why after some "reloads" disappear?.. i have ideas too, but.. ?
08:15.29opus_?
08:16.12outtoluncopus, if anything 'triggers' debug its in the Makefile
08:16.18outtolunclook there
08:17.03outtoluncif all else fails "grep rtp_debug *" in various dirs
08:17.32opus_oh, there is a 'rtp debug' cli command
08:17.58opus_Jul 29 01:16:56 DEBUG[27139]: rtp.c:237 process_rfc2833: - RTP 2833 Event: 00000004 (len = 4)
08:19.09opus_duration &= 0xFFFF;
08:19.19opus_hmmmmmm
08:19.22monkmoleanyone ever know vga=x to cause probs?
08:19.48monkmolekernel param that is... Digium guy told me that once.
08:19.48drrayouttolunc - the livecd would tell him that his hardware works and is compat with his mb
08:19.54outtoluncyou are using 2833 (inband) for this?
08:20.34monkmoleRunning it on FC for the last 6 months has told me that :)
08:20.38outtoluncdrray, alot of 'live cd's are for home hardware, i'm pretty sure he had a t1 1 port card also
08:20.45*** join/#asterisk Assid (~assid@203.115.64.60)
08:20.47Assidhi
08:20.52monkmolesup
08:20.56Assidi have a SPA-841
08:21.03Assidits a 2 extension phone
08:21.04drraythe asterislive cd worked with my t1 card/zhone channel bank
08:21.22Assidbut.. when i call on one extension.. and i get another incoming call.. it says the extension is busy
08:21.26drrayback before I knew what I was doing enough to get it working
08:21.29Assidi want to use the same username/info for both
08:22.52outtoluncso ddray, is it your goal to tell new comers with t1 gear to use things like A@H or learn the docs?
08:23.13*** join/#asterisk darkskiez (~darkskiez@194.247.78.146)
08:23.13outtoluncer drray
08:23.43drrayno, my goal would be to help people use simple tools to be able to determine what their problem is.  once I got the hardware working with the live cd, i worked on getting it working with fedora
08:23.47lehel<PROTECTED>
08:24.51outtoluncso, instead of asking questions about what thier configs look like that you just automatically suggest ... A@H or others
08:25.26outtolunci'm just trying to understand how this truely helps the user
08:25.49opus_yes very good point
08:26.10drraywell, I was not suggesting asterisk at home, I was suggesting soemthing that could easily help him determine where the fault was
08:26.23monkmoleI'm still here
08:26.24opus_most of it is triviall
08:26.36opus_if they just had error messages like 'Your nat is fucked'
08:26.50opus_if they just had error messages like 'Your nat is fucked, try nat=yes or natautodetect=yes'
08:26.58outtoluncwell sadly, i can tell you right now, 99.99% of the time it is the user/config
08:27.11opus_you shouldn't have a layer 4 setting in a layer 7 config file
08:27.19*** join/#asterisk Akelavlk (~jansun@82.119.239.141)
08:27.29drrayI agree, but using the live cd would help prove that to people who come in here and say that it must be a problem with asterisk
08:27.35outtoluncso, sending them off to a gui might get it working, but does that solve the issue?
08:27.47AkelavlkHello all
08:27.55drrayok, I'm not really interested in arguing this, I never suggested they go use a gui
08:27.58AkelavlkI have a problem with  modprobe wcfxs
08:28.25opus_ok weird this is so messed up
08:28.26AkelavlkI got error message "/lib/modules/2.4.22-1.2115.nptl/misc/wcfxs.o: init_module: No such device"
08:28.28outtoluncA@H is what i'd call a gui, what do you call it?
08:28.35monkmoleAkelavlk: welcome and joing the crowd :)
08:28.36opus_if i can hit the dtmf pad within 800ms, it works
08:28.40drrayI never suggested Asterisk at home
08:29.01outtolunc(it's actually a shitload of 'side' apps that 'work' with asterisk'
08:29.03outtolunc)
08:29.13AkelavlkHas anyone idea how to solve that problem?
08:29.46monkmoleAkelavlk: What distro?
08:29.48lehelAkelavlk: you need the wcfxs module?.. ain't you have the wctdm?
08:29.52outtoluncthen what was it you were suggesting?
08:30.10AkelavlkI don't have wctdm module.. I have asterisk 1.0.7 version..
08:30.23outtolunc'a live cd'
08:30.26outtolunchmm
08:30.27AkelavlkI think wctdm module was in older versions..
08:30.46opus_shit i hope 1-800-callatt doesn't get pissed at me
08:30.47*** join/#asterisk psycodad (~obiwan@2001:4060:4419:b1:0:0:0:2)
08:31.48Akelavlkmonkmole: What version of Zaptel are you using?
08:32.21opus_i think i fixed the dtmf issue
08:32.22monkmole1.0.9
08:32.27opus_its not fixed, but works a little better..
08:33.39AkelavlkI read documentation there is few words about that modul.
08:33.52lehelwctdm it is definitly not old
08:34.45AkelavlkI have zaptel 1.0.7 version and there is not wctdm module..
08:35.28lehelupgrade! update..
08:35.39*** join/#asterisk zeedo (~zeedo@www.reboot-robot.net)
08:36.46AkelavlkThis is from DIGIUM web site. "Do a modprobe wctdm, (if you have latest cvs otherwise its modprobe wcfxs) which will load the driver for the TDM Card"
08:37.01lehelsorry.. my mistake
08:37.05Akelavlklehel what version do you have?
08:39.34Assidhow do i make a call roll onyto the next extension?
08:39.43Assidi have a SPA-841
08:39.57outtolunclehelDoc/ML/Bugs/Reader <G>
08:39.58Assidbut.... the call only comes on 1 extension.. new calls get busy..
08:40.10lehelDocReader:D
08:41.15outtoluncassid, if one gets thru, the rest get busy (congestion) means you only have 1 path
08:41.25outtoluncone device
08:42.16Assidbut the same phone can have 1 in 1 out.. or 2 out simultanously
08:42.35outtoluncand they would only get congestion if you put that in the next inc of the dialplan
08:42.38Assidits a 2 extension phone..
08:42.44outtoluncnot always
08:43.04Akelavlklehel: Are you using DEV kit from DUGIUM?
08:43.27outtoluncfor a 2 exten phone to actually have the ability to have 2 'rings' incoming calls, it needs both to be setup
08:43.50outtoluncif only 1, well you already know what happens
08:43.54Delvari need a h323 to SIP converter, i rmemebr some asterisk alike software that did it but i cant rememebr waht its called... any ideas?
08:44.03Assidmaybe coz i am using both as same user?
08:44.28outtoluncsame user, 1 context, 1 whatever
08:44.44Assidi have both the same..
08:44.45opus_hmmm, ok next problem
08:44.46outtolunccreate userl1, userl2
08:44.53*** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de)
08:45.06outtoluncand have the phone reg both
08:45.17Assidwhy cant i have both as the same user?
08:45.33opus_each person has a direct line.  when dialing out the caller id is set to that line. however, when internal office calls are made the caller id is of the external line and not the internal extension, is there a workaround for this?
08:45.44outtoluncyou could if, you didn't use the same 'incoming' to the same device
08:45.56monkmoleKinda ugly paste on its way...
08:45.58monkmolekernel panic - not syncing: /usr/src/zaptel-1.0.9.1/wcte11xp.c:333
08:45.58monkmolespin_lock(/usr/src/zaptel-1.0.9.1/wcte11xp.c:dcde0004) already locked by
08:45.58monkmole/usr/src/zaptel-1.0.9.1/wcte11xp.c/887
08:46.08Assidso do i need to make more extensions?
08:46.13Assidor can i atleast keep the extensions same?
08:46.24outtoluncmeaning, IF asterisk thinks that 'device' singular is busy, why would it continue
08:46.54outtolunceach 'line' on the phone needs to be separate
08:47.57Assidso effectively i need 3 extensions per phone?
08:48.02outtoluncno
08:48.05Assid1 to call the phone.. 2 for routing?
08:48.27outtoluncif you have a 2 line phone, effectively it needs 2 'entities'
08:48.48outtolunccall waiting, hold, etc are separate functions
08:48.57Assidi see..
08:49.04Assidwhy cant it just determine based on the port?
08:49.17outtoluncdepends on phone
08:49.24Assidlike user1 on the same ip.. but 2 different entries based on the port
08:49.39outtoluncearlier models required you to 'setup' each line
08:49.51outtolunchense the separation
08:50.17Assidso if there was another unit.. maybe a good cisco phone.. i wouldnt have to do this?
08:50.18outtoluncthink of it this way
08:50.36outtolunccan you plug in a 2 line pstn phone to 1 line and have both lines work?
08:50.41outtoluncno
08:50.55outtoluncnot separately
08:51.25outtoluncso, if you want each to 'act' separately, you must treat them as such
08:51.36outtolunclogical?
08:51.50Assidhrmm
08:51.52Assidi guess
08:51.54Assidlemme try this
08:52.00drrayyou want extensions.conf to handle it, not the phone
08:52.02Assidand make the dial plan with &
08:52.06outtolunctry whatever you like <G>
08:52.24outtolunci was just giving advice
08:52.29Assidyeah..
08:52.38Assidthanks
08:52.50Assidso.. i treat each line seperately
08:52.59Assidand make my dialplan so it dials botht he extensions
08:53.04drrayon my cisco phone 1-5 are extensions 100-105, when they dial 100 it rolls over via extensions.conf
08:53.07Assiderr.. as 1 extension..
08:53.34drray100-105 are the sip.conf names
08:53.47Assidbut i m guessing this doesnt support it?
08:53.52Assidthe 841?
08:54.01outtoluncthen in your dialplan you 'could' (other ways also) just dial(tech/dev1&tech/dev2)
08:54.10outtoluncand both would ring
08:54.21Assidright..
08:54.32Assidthats what i was trying
08:55.08*** join/#asterisk nextime (~nextime@213-140-6-96.fastres.net)
08:55.39outtoluncif you dialed 1 phone line on a 2 line phone (that had 2)  both would not ring, just the one that was
08:56.03outtoluncwhat makes you think this is 'that' different?
08:56.28Assidright.. what i did .. was made 2 users in sip.conf.. and made it dial(tech/dev1&tech/dev2)
08:56.41outtoluncthat is how you do it
08:57.04outtoluncor queue those tech/devs
08:58.05outtoluncand the problem is?
08:58.38Assidnot unable to make 2nd call
08:58.40Assidi think
08:58.40Assid1 sec
08:59.04outtoluncfor a 'phone' to have 2 'appearances' it must register (both)
08:59.22outtoluncif both are not, well you get the idea
08:59.46Assidyeah.. sip show users shows them
08:59.50outtoluncall asterisk can do is 'attempt' to try that tech/dev
09:00.00Assidsip show peers shows them as well
09:00.13outtoluncare they the same user?
09:00.20Assiduser1 user2
09:00.43outtoluncif they truely are, then i don't see an issue
09:00.50Assid1 on 5060  2nd on 5061
09:00.59outtoluncumm
09:01.00Assidboth have OK status
09:01.13outtoluncthey should both be on the same port <G>
09:01.25outtoluncyou got a firewall anywhere <G>
09:01.44Assidbut they registered..
09:01.45Assidso...
09:01.49Assidwhy shouldnt it work?
09:01.58outtoluncthats 1 way traffic, but ok
09:03.04outtoluncmeaning the phone sent a packet, but is the asterisk box able to send it back 'originating'
09:03.27outtoluncpart of the issue with nat
09:03.36outtoluncanyways
09:03.50outtolunci wish you the best, i'm headed for bed
09:04.26Assidhrmm
09:04.29Assiddidnt work :(
09:05.49Assidumm.. how do you dial from console ?
09:09.50*** join/#asterisk wasim_ (~wasim@wasim.active.supporter.pdpc)
09:18.08*** join/#asterisk fenlander (~neils@82.152.81.57)
09:21.25*** join/#asterisk riemensc (~riemensc@83-169-155-92-dynip.superkabel.de)
09:22.27lehelDocReaderpeople could you have a look waht is the problem with this call?: http://pastebin.ca/18761
09:27.37*** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl)
09:27.39puzzledmorning
09:28.42ltersGood Morning
09:29.10lterspuzzled, is something up with the mailling list?
09:29.40lterslast emails are from yesterday mid to late afternoon
09:31.14puzzledlters: dunno but I have suspiciously little email from the list. maybe there is an issue but I'm sure it will be fixed when the guys in the US wake up
09:32.24ltersfancy
09:32.43*** join/#asterisk kaldemar (~kalde@vipunen.hut.fi)
09:32.44ltersu running head?
09:34.52lterswishing I could update my main * to HEAD and don't know what date to shoot for.
09:35.54puzzledlters: I mainly run STABEL but will update to HEAD today. I just get whatever is most recent
09:35.59puzzledSTABLE even
09:38.05ltersI see. how many users tho?
09:38.22lters2 or 3, or 50 or 60 :)
09:38.50puzzledjust me, myself and I :)
09:39.07Assidhow many users would a AMD Athlon64 3400 support?
09:39.13Assidavergae?
09:39.30Assidhrmm
09:39.34lters60
09:39.52ltersor 600
09:39.53Assidi realyl wanna add database support to asterisk.. so i can log my calls correctly  and stuff
09:40.03Assid60/600??
09:40.08*** join/#asterisk grimse (~grimse@p5481BFA8.dip.t-dialin.net)
09:40.08Assidsimultanous calls
09:40.11ltersedit cdr_pgsql.conf
09:40.24Assidgotta install pgsql first.. heheh
09:40.26puzzledAssid: depends if you do codec conversion. with straight alaw or ulaw quite a decent number. I guess 50 or so. test it with sipsak
09:40.28ltersno
09:40.40Assidsipsak ?
09:40.50puzzledgoogle is your friend
09:41.08*** join/#asterisk Stephnie (dfsdf@203.215.180.254)
09:41.11*** join/#asterisk kuonSama (~kuon@alragore.goyman.com)
09:41.15kuonSamahello all
09:41.15Stephniehi
09:41.28puzzledhello
09:41.51StephnieI have RH9 ... VNC server is installed...
09:41.59kuonSamaI did setup a lot of asterisk box, but I did never do the interconnexion with PBX (I always used cisco router as SIP->PSTN)
09:42.03Stephniehow to run it and connect to it?? any help?
09:42.11kuonSamanow, I have to add a voip part to an alcatel 4200E
09:42.23kuonSamaI know everything for the asterisk part
09:42.36puzzledStephnie: join #fedora. this channel is about asterisk
09:42.38kuonSamabut no idea how to interconnect the alcatel PBX with asterisk computer
09:42.56puzzledkuonSama: did you check the mailing list archives and voip-info.org?
09:43.04kuonSamawith google yes
09:43.38Assiddo i have to reinstall/compile asterisk after i install pgsql?
09:43.47Stephnieok
09:44.04puzzledkuonSama: what will you need, an T1/E1/PRI or just a few ISDN channels?
09:44.13puzzledAssid: to do what?
09:44.42ltersAssid, well, you need to only install the pg headers, than doing make in asterisk will compile just the cdr_pgsql.so
09:44.44kuonSamapuzzled: the alcatel is connected with few isdn chan
09:45.18ltersAssid, unless of course u want to also store the cdr's local (not really good)
09:45.36Assidnah.. just want it in the database
09:46.10puzzledkuonSama: get a card from junghanns.net and install bristuff so you can make your * box talk to the alcatel pbx with isdn channels
09:47.16kuonSamago thanks
09:47.38puzzledkuonSama: you can test it cheaply if you can find an HFC-S based isdncard with one BRI
09:51.18kuonSamak thanks
09:51.20StephnieStarting VNC server:                                       [  OK  ]
09:52.03Stephnienow should I type IP of machine in Internet explorer to connect to VNC server?
09:52.27Stephnieoops
09:52.29puzzledStephnie: wrong channel. ask in #vnc, #fedora or something like that
09:52.30Stephniewrong channel :D
09:52.38puzzledhehe
09:52.38Stephnie:D
09:52.41Stephniehehehe...
09:52.51lehelpeople can you have a look pls.. what is going wrong with my call: http://pastebin.ca/18762
09:55.30lterslehel, your channel is not accepting calls
09:55.38*** join/#asterisk oej (~oej@bruno.upsys.se)
09:55.56puzzledmorning oej
09:56.06ltershi oej
09:56.08oejmorning
09:56.15kuonSamahow much is the isdn quab bri card?
09:56.23puzzleddunno
09:56.33kuonSamacan't find any price
09:56.42puzzledemail them and ask for a quote
09:56.45ltersoej, I like your 1.2 presentation. it is good
09:56.52oejIters: Thanks!
09:56.56puzzledlehel: the phone you are trying to call seems to be busy
09:57.10kuonSamak
09:57.43ltersoej, max retries on a sip phone... why does that happen
09:58.11oejIters: Depends on when... When asterisk reboots, that happens due to NAT...
09:58.30oejIn some cases, like REFER, the phone simply doesn't understand and answer properly
09:58.55ltershmm, that makes sense. using 1.0.9
09:59.01oej...and with Grandstreams, it's the phone :-)
09:59.17ltersthis is 7.3/cisco/sip
09:59.41oejIters: Ok, then you have to tell me what type of message that fails
10:00.25ltersoej, I don't know how to have a *controled* debug, because it only happens when there are lots of other calls.
10:00.40oejsip debug peer <name>
10:00.44lehellters: my channels isn't accepting calls? could be codec problem?..
10:01.30lterslehel, not likely. looks deeper than that.
10:01.41lehelpuzzled: unfortunately they are'nt busy ;\
10:01.59ltersoej, but can I get that to a debug file ?
10:02.32lterslehel, set verbose 10
10:02.44oejIters: asterisk -rvvvdddd | tee /tmp/debug
10:03.17ltersoej, with -r works ?
10:03.22oejverbose 4 and debug 4 are the highest levels
10:03.39oejIters: either -c or -r depending whether asterisk runs in the background or not
10:04.01ltersok, I will do this.
10:04.21ltersoej, do you know of any probs with HEAD at the moment?
10:05.34ltersor could I update to the latest from 1.0.9 ?
10:10.50Assidokay finished installing postgres
10:10.56Assidnow..
10:11.03Assiddo i recompile asterisk?
10:13.21*** join/#asterisk modulus_ (~modulus@rm-f.net)
10:13.25ltersmake && make install
10:13.31modulus_damn lotsa more ppl in here
10:13.56modulus_where'd all these people come from?
10:13.58modulus_jeezes
10:14.36AkelavlkSlovakia
10:15.09puzzledsomeone call me?
10:15.30AkelavlkSure, I have problem with load module..
10:16.10AkelavlkWhen I run this modprobe wcfxo I get error.
10:16.38konfuzeduhm I just called my DID number from a landline and got this message in the CLI;
10:16.38konfuzedCLI>
10:16.38konfuzedJul 29 06:25:25 NOTICE[31330]: chan_sip.c:7295 handle_request: Failed to authenticate user "$callerstelnumber" <sip:$callerstelnumber@66.96.30.25>;tag=as5ef4700f
10:16.38konfuzedCLI>
10:16.39konfuzedrepalce the $callerstelnumber listed with the actual 10digit phone number of the caller.
10:16.41konfuzedmore to the point why is it using the callers phone number as a userid@mysip.provider ????
10:16.49konfuzeddamn editor
10:17.22konfuzedkvirc has this built in editor that keeps opening
10:18.45*** join/#asterisk pigpen (~mark@fw.seamans.cc)
10:20.12modulus_Akelavlk, what does ztcfg -vvv show?
10:20.37InfraRed|Birthdyspam
10:24.21modulus_# ztcfg -vvv
10:24.21modulus_Zaptel Configuration
10:24.21modulus_======================
10:24.21modulus_SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)
10:24.24modulus_*grin*
10:24.30modulus_Channel map:
10:24.30modulus_Channel 01: Individual Clear channel (Default) (Slaves: 01)
10:24.30modulus_Channel 02: Individual Clear channel (Default) (Slaves: 02)
10:24.32modulus_...
10:24.36modulus_Channel 23: Individual Clear channel (Default) (Slaves: 23)
10:24.36modulus_Channel 24: D-channel (Default) (Slaves: 24)
10:24.36modulus_24 channels configured.
10:25.47modulus_does that look ok for my t1 pri?
10:25.51modulus_anyone?
10:26.02puzzledlooks ok to me
10:26.22modulus_ast-lax2*CLI> zap show status
10:26.22modulus_Description                              Alarms     IRQ        bpviol     CRC4
10:26.22modulus_Digium Wildcard TE110P T1/E1 Card 0      OK                  0          0          0
10:27.08modulus_stupid zaptel rc script installer loads wcusb
10:27.20modulus_# lsmod
10:27.20modulus_Module                  Size  Used by    Not tainted
10:27.20modulus_soundcore               6916   0  (autoclean)
10:27.20modulus_3c59x                  31184   1
10:27.20modulus_wcusb                  20128   0  (unused)
10:27.21modulus_wcte11xp               22464  24
10:27.23modulus_zaptel                183552  52  [wcusb wcte11xp]
10:27.27puzzledI always load modules from /etc/rc.d/rc.local
10:27.31modulus_should i bother taking out the wcusb?
10:27.47puzzledi always do
10:27.53puzzledonly load what I need
10:27.57modulus_yeah don't need to use up extra mem
10:28.16modulus_# grep wcusb zaptel
10:28.16modulus_<PROTECTED>
10:29.18modulus_eureka: /etc/sysconfig/zaptel
10:29.54modulus_rebooting....
10:33.11*** join/#asterisk truescot (~truescot2@213.201.171.186)
10:33.28modulus_# lsmod
10:33.28modulus_Module                  Size  Used by    Not tainted
10:33.28modulus_soundcore               6916   0  (autoclean)
10:33.28modulus_3c59x                  31184   1
10:33.28modulus_wcte11xp               22464  24
10:33.29modulus_zaptel                183552  52  [wcte11xp]
10:33.31modulus_w00t
10:33.51truescothello people, does any one know of a good fax client for windows that will connect to asterisk??
10:35.03puzzledthat does not make sense to me. what are you trying to do
10:35.58truescoti want to send a fax from windows through asterisk if its me u mean
10:37.04tuxinator_linuxMNight guys
10:37.21puzzledtrig_hm: ah right. sorry I have no idea
10:37.39tuxinator_linuxMappropriately named puzzled
10:37.49puzzledhehe
10:40.15*** join/#asterisk Poincare (~jefffnode@dD5779BD2.access.telenet.be)
10:41.05truescotahh well just have to keep lookin
10:42.31lehelPeople Asterisk is AMAZING!!.. you can do wwwowww things.. huhh
10:43.01modulus_no it isn't
10:43.03modulus_asterisk sucks
10:43.04modulus_voip sucks
10:43.36modulus_i just use it b/c it's cheap
10:43.52modulus_and phone calls aren't important to me
10:43.53truescotcheap in the extreme
10:44.19truescotas in we just got rid of a pbx that cost a fin fortune and did hardly any of the stuff asterisk will do
10:45.14truescoti really do like it and have found no drawbacks moving to it
10:45.25lehel"b/c" < ??
10:45.26truescotso far its been running 32 days without problem
10:45.38truescotb/c = because
10:45.53modulus_"lehel" < ??
10:45.58lehel:P
10:50.56*** join/#asterisk jaike (~a@203.177.242.192)
10:51.59Assidumm.. where do i find the table structure for pgsql cdr table?
10:52.16puzzledAssid: voip-info.org or maybe the docs included in asterisk src
10:52.18jaikecan anyone suggest any good ip phones? within the $150 - $200 pricerange
10:52.28puzzledPolycom IP300
10:52.36jaikewe were using sipura 841 but theyre not very good
10:52.56*** part/#asterisk vuvie (~vuvie@bb219-74-46-99.singnet.com.sg)
10:53.08*** join/#asterisk riemensc (~riemensc@83-169-155-92-dynip.superkabel.de)
10:54.06jaikehow bout the ciscos?
10:56.22InfraRed|Birthdycisco are expenisve
10:56.32InfraRed|Birthdyyou have to pay for
10:56.35InfraRed|Birthdyphone
10:56.37InfraRed|Birthdysip license
10:56.40InfraRed|Birthdypower cube
10:57.13*** join/#asterisk jiro5281 (~anton281@203.177.242.192)
10:57.20InfraRed|Birthdysaying that
10:57.36nDuffbut that said, the Snom 360 is a much better phone -- only thing is that it's more like $240 at the low end of the price range
10:58.02*** join/#asterisk paskifr (~paskifr@evr91-1-82-230-215-102.fbx.proxad.net)
10:58.24nDuffjaiger, be sure to avoid the Grandstream -- not just the Budgetones, but the GXP-2000 as well; they're all crap.
10:58.26nDufferm
10:58.29nDuffjaike, be sure to avoid the Grandstream -- not just the Budgetones, but the GXP-2000 as well; they're all crap.
10:59.10nDuffs/history/implementation/
10:59.48jaikeok..thanks guys
11:00.20jaikeanyone experience any problems with the ip300?
11:00.29truescoti again i have got to say, i have 12 gxp-2000 phones and with the latest firmware i have had no problems with them at all
11:01.48truescotgranted the old firmware made them practically unusable tho
11:10.07*** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca)
11:11.12*** join/#asterisk zotz (~zotz@24.231.36.100)
11:14.15Assidif i want to set the callerid as the name <number> how do i do that?
11:19.39*** join/#asterisk Newbie___ (me@211.24.146.12)
11:22.52*** join/#asterisk blazint (~blazin@cm225.epsilon203.maxonline.com.sg)
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11:27.27*** part/#asterisk lehel (~lehel@82.79.20.17)
11:27.58Assidhow do i access the variable being used in src column for cdr's?
11:28.30*** join/#asterisk jaike (~a@203.177.242.192)
11:28.53jaikehi again...got this from voipsupply.com
11:28.55jaikePolycom IP500 Phone
11:28.55jaikePlease Note: Polycom phones are not supported under Asterisk Open Source PBX. Polycom certified platform partners include Path Navigator, Broadsoft, Interactive Intelligence, Sphere, Sylantro, Vertical Networks, VocalData, Alcatel and 3COM. For more information on Polycom supported IP Communications platforms Click Here
11:29.14jaikepolycom wont work with asterisk?
11:29.38truescotno, they will work they jsut wont give u support for them
11:29.48truescotat leas i know som of them do
11:31.49Newbie___hi, can anyone tell me where to look for a ANI callback feature for asterisk ?
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11:34.55*** part/#asterisk truescot (~truescot2@213.201.171.186)
11:37.22drrayI don't think cisco phones are supported under Asterisk either
11:37.35drray"supported"
11:41.50*** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de)
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12:19.35TrailBlazerManhello all
12:21.40TrailBlazerManwho has experience to compile iaxclient library to WIN32 ?
12:23.10TrailBlazerMani have some problem to do this with Cygwin+MinGW
12:23.45*** join/#asterisk JohnJacob (~JohnJacob@pcp0011543387pcs.mainf01.in.comcast.net)
12:25.54*** join/#asterisk sgnome` (~sgnome@ip68-226-115-127.ph.ph.cox.net)
12:26.14xhelioxIs the Iaxy not password secured? Couldn't anyone provision it? The docs don't seem to ask for a password.
12:26.36sgnome`what sym links do i need to make on a system with a 2.6 kernel to get a TDM400P to work?
12:27.14*** join/#asterisk sgagers (vrega@151.97.29.77)
12:27.32sgagershi all
12:27.37sgagersi need help
12:28.08sgagersi'm not able to checkout from cvs head of asterisk 1.2
12:28.22sgagersit response that asterisk directory does not exists
12:28.45sgagersi send command:
12:29.18sgagerscvs checkout zaptel libpri asterisk
12:31.27*** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net)
12:31.54sgagerssomeone can help me?
12:32.09jaikeanyone using polycom ip301s?
12:32.47DarthCluejaike: why?
12:33.17jaikejust making sure its working with asterisk..u never know
12:33.50TrailBlazerManwhen i try to do 'make shared' i got errors like
12:34.12*** join/#asterisk xylome (~asterisk@hg-msq-hol.levigo.de)
12:34.41DarthCluejaike: it should work fine.  i don't actually have an ip301 but i know people who do and i have an ip501 that works fine as well.
12:34.55*** join/#asterisk astoria (~haydenth@66.235.201.217)
12:34.55TrailBlazerManaudio_file.c:99: error: conflicting types for 'iax_set_files'
12:35.09xylomehi, is the digium mailing list server down?
12:35.12astoriaOMG! Pizza Party at CLUECON! I'm so excited! :)
12:36.39jaikethanks darth
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12:55.31Broomhelo
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13:00.18*** part/#asterisk Akelavlk (~jansun@82.119.239.141)
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13:03.46*** join/#asterisk Feral_Kid (~kcallis@red-corp-200.56.96.178.telnor.net)
13:05.34magically_fooAny CISCO 7960 owners on right now?
13:05.49mutwe've got a few why
13:05.51crash3muhhh, yeah
13:05.54zoai have some
13:05.59zoayou need firmware right ? :p
13:06.04mutheh
13:06.06Hogieheh
13:06.06magically_fooOk, here is my problem...
13:07.37muthmm
13:07.50mutmy mail queue musta been bouncing messages like mad last night
13:08.08muthad ~300 in thre and now theres 40
13:08.10magically_fooI can get into the setting, and my phone was set to a static IP which I can access... If I fire up knoppix (or something like that), set the IP address to fall in my old IP range on the phone, and plug in a x-over, I should be able to tftp the firmware, correct?
13:08.27magically_fooI can't get into my settings I meant to say...
13:08.43*** join/#asterisk zeedo (~zeedo@www.reboot-robot.net)
13:08.43*** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net)
13:08.51*** join/#asterisk smurfix (~smurf@smurfix.developer.debian)
13:09.43crash3mmagically_foo: if your IP is set to that of the tftp server sure
13:10.07crash3myou should also be able to set your admin password through the config file...while your messing with tftp and all
13:16.40*** join/#asterisk oej (~oej@bruno.upsys.se)
13:17.07zoahey ho olle
13:18.03blitzrageho hey olle
13:18.16*** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net)
13:19.40*** join/#asterisk Ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
13:19.48zoahey ho blitz
13:20.02Ariel_morning all
13:20.17mountieXML is like violence. If it doesn't solve the problem, use more.
13:20.22blitzragezoa: zup!
13:20.31Ariel_quick question is the user list broken? I have not gotten any email since yesterday afternoon
13:21.39riemensci´ve got a problem i call via sip a person and i can hear the person the person cannot hear me
13:22.26DarthClueAriel_: a couple of people have stated that they (as in the people, not the lists) seem to be having problems as well.
13:22.32Ariel_riemensc, check your firewall is allowing rtp sound stream outbound.
13:23.03Ariel_DarthClue, do you get the emails if you do then it might be a spam rule on my side
13:23.15DarthClueAriel_: i don't subscribe to users.
13:24.07DarthCluei have a -dev from shortly before midnight.  but that's not unusual.
13:24.16Ariel_DarthClue, do you get the -dev or cvs?
13:24.29Ariel_sorry -dev i got one a 1 am
13:25.03DarthClueAriel_: title of the message on -dev?
13:25.11Ariel_it's just strange coming in to work and not having over 100 emails to skim through from the asterisk lists.
13:25.16*** join/#asterisk JohnJacob (~JohnJacob@pcp0011543387pcs.mainf01.in.comcast.net)
13:25.29Ariel_DarthClue, pizza
13:25.41DarthClueAriel_: IP_ID in RTP UDP packets was the last one i got on -dev.
13:26.05elzhovHello. I read in extensions.conf sample line with comments:;
13:26.05elzhov; We also create an example user, 1234, who is on the console and has
13:26.05elzhov; voicemail, etc.
13:26.05elzhov;
13:26.05elzhovexten => 1234,1,Playback(transfer,skip); "Please hold while..."
13:26.06elzhovWhat "user is on the console" means? Thanks :)
13:26.11Ariel_I got that one before brian's pizza message
13:26.31DarthCluewell, i got pizza before that one so i think we are even in that respect.
13:26.40mutanyone know whats with dice.com's date stamps for their job postings?
13:26.49mutthey're almost always the current date
13:26.53mutno matter when ya visit ti
13:27.01Assidanyone know any good providers.. incoming free.. where i can port a NY DID
13:27.03Ariel_DarthClue, strange that I may get a reply to an email I send before I see the message in the list.
13:27.47DarthClueAriel_: yeah, i've seen that as well.  it's a by-product of the list software i'm sure.
13:29.05riemenscmy firewall is check and in dmz running my server with no port filtering
13:29.06Ariel_Assid, incoming free well the only ones I know that is fair is voicepulse's connect.  Broadvoice has it's lite but there so-so for service.
13:29.26DarthClueAriel_: according to the list-archives, a -users message should have gone out as late as 7:47 CDT today.
13:29.28riemensci call the person and hear on my phone a echo from my voice
13:29.41riemensci hear the person they talk on other telephone
13:29.47Ariel_riemensc, if you only get one way audio it's your rules or some type of sip setting not set correctly.
13:30.13*** join/#asterisk zurab (~zurab@ws2.sitel.com.ua)
13:30.25Ariel_riemensc, do you have the external IP address set in your sip.conf?
13:30.43riemenscyes via dyndns.org
13:31.04DarthClueAriel_: and there are at least 5 messages on -dev that i haven't gotten yet.
13:31.37Ariel_externip=kasipbx.homedns.org
13:31.37Ariel_localnet=192.168.XXX.0/255.255.255.0
13:31.37Ariel_<PROTECTED>
13:31.54zurabHello, Everybody!!!
13:32.05AssidAriel_: need consecutive/simultanous calls
13:32.07crash3mcan anyone tell me what has changed in the uniden firmware since 4.56?
13:32.38Ariel_Assid, you said inbound free
13:33.00riemenscmy local ip is 192.168.199.100
13:33.33Ariel_your local needs to be 192.168.199.0  this allow everyone one in your local to get the sound correctly.
13:33.46riemensclocalnet = 192.168.199.0/255.255.255.0
13:34.03Ariel_riemensc, yes that allows all your local systems to get sound.
13:34.50lathos42Ariel_: I just saw your message about the -users list..  I havent received any messages from it since yesterday afternoon as well
13:34.50riemenschow can allow this?
13:34.57bkw_Ariel_, its externhost now isn't it
13:35.10*** join/#asterisk Feral_Kid (~kcallis@red-corp-200.56.96.178.telnor.net)
13:35.18lters_lathos42, something is definitely wrong with the asterisk-users mailling list
13:35.19Ariel_bkw_, not yet as far as I see. But I am on stable mostly.
13:35.31riemenscexternip = frconsulting.dyndns.org
13:35.57lters_bkw_, what is up with the mailling list?
13:36.20Ariel_bkw_, I just changed my head test box and it's taken the extenip= statement.
13:36.33zurabI'm trying to do this : http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20mysql%20peers on a FreeBSD (5.4-RELEASE) (from the ports)but the only thing i'm getting is : Jul 29 15:34:47 WARNING[39287]: /usr/local/lib/asterisk/modules/chan_sip.so: Undefined symbol "mysql_real_escape_string" Jul 29 15:34:47 WARNING[39287]: Loading module chan_sip.so failed! Maybe someone can help ?
13:36.35riemensci use a netgear fvs318 router
13:36.56riemensc192.168.199.100 is in DMZ
13:37.01*** join/#asterisk oOlli (www-data@baltz-online.de)
13:37.07riemensci use debian 2.6 kernel
13:37.14riemenscas standard installation
13:37.29Ariel_riemensc, ok but still you need to have your localnet as 192.168.199.0/255.255.255.0
13:37.37oOllihi folks. do I need an active ISDN card in order to use P2P ISDN (in germany: Anlagenanschluss) ??
13:39.01riemensci use localnet = 192.168.199.0/255.255.255.0
13:39.59DarthCluezurab: do you have mysql and the client libraries installed?  if so, what version?
13:40.46riemenscgo not
13:41.20zurabyes, I have mysql-server-4.1.13
13:42.03*** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca)
13:43.18*** join/#asterisk RedHatUser (~acabi@baydeinter-27-83.baydenet.com.br)
13:44.47otmaroOlli: no, that's pure software
13:45.06DarthCluezurab: what about the client libraries?
13:45.19DarthCluezurab: you may need the dev libraries as well.
13:45.24oOlliotmar: that means?
13:45.44zurabmysql-client-4.1.13, everything from fresh ports collection
13:45.53newlPTP and PTMP is normally configured in the node data.
13:46.43*** join/#asterisk nighty- (~nighty@fr-reims-gw.origami-systems.com)
13:46.54nighty-Hi
13:47.52*** join/#asterisk tengulre (~tengulre@219.144.170.96)
13:48.04nighty-I have a problem with a PAP2-NA not being able to talk between the 2 PAP2 line (no reinvite) using Asterisk
13:48.16nighty-this must be something very easy to do
13:48.23nighty-but I fail to find how
13:48.53nighty-PAP2 NA is receiving and calling just fine thru asterisk to any other numbers
13:48.58nighty-but the local extensions
13:49.10nighty-can any help me on this ,
13:49.11nighty-?
13:49.14oOlliotmar: am I able to use 2 passive Fritz! cards than for 2x ISDN BRI (p2p) on asterisk??
13:49.59*** part/#asterisk RedHatUser (~acabi@baydeinter-27-83.baydenet.com.br)
13:50.16Ariel_nighty-, are they in the same context the extensions?
13:50.27nighty-Ariel_: they are both in the same context
13:50.35nighty-Ariel_: that is why I don't understand
13:50.54Ariel_nighty-, what error do you get on the CLI when you dial the other extension?
13:51.28puzzledoOlli: supposedly it is possible. there is a doc floating around explaining how to do it. search google
13:51.43jake1932antone know where i can ge the SIP firware for the 7940?
13:51.52puzzledcisco.com
13:52.15jake1932tnx
13:52.35nighty-Ariel_: I get (in SIP debug) :   == No one is available to answer at this time
13:52.54*** join/#asterisk RoyK (~roy@b-argyle-adsl.demon.co.uk)
13:53.12newlWhat?  No bag pipes?
13:53.14*** join/#asterisk ctooley (~ctooley@rrcs-24-227-212-181.sw.biz.rr.com)
13:53.27Ariel_nighty-, what is your dial string to the devices?
13:54.01nighty-Dial(SIP/ata12,70,r)
13:54.12Ariel_something like exten => 100,1,Dial(Sip/100) is a simple setup
13:54.18nighty-yes
13:54.58nighty-Ariel_: to be exact it is
13:54.59nighty-exten => 6201,1,Dial(SIP/ata11,70,r)
13:54.59nighty-exten => 6202,1,Dial(SIP/ata12,70,r)
13:55.10*** join/#asterisk mhnoyes (~mhnoyes@user-38lc13t.dialup.mindspring.com)
13:55.25nighty-ata11 and ata12 and 2 ports of the same ATA (PAP2-NA)
13:55.28ManxPowerIs now a good time to say 'only terrorists use the "r" option to Dial'?
13:55.40nighty-ManxPower: sorry :)
13:55.44Nuggetheh
13:55.50nighty-ManxPower: I can remove this
13:55.52nighty-:)
13:55.56ManxPowerNugget: that's my new .sig file.
13:55.56Ariel_ManxPower, there you go again.... wow.
13:56.15nighty-ManxPower: can you explain why it is bad ?
13:57.10ManxPowernighty-: "r" hides the REAL sound you should be hearing and forces a ringing sound, even if you should be hearing something like "the cell phone you dialed is not reachable"
13:57.32nighty-ManxPower: oh good to know, thank you I'll remove this
13:57.44Assidshido: you up?
13:57.51Assidi need intl dialling on my account man
13:57.51ManxPowernighty-: %80 of the time that won't be an issue.  The other %20 of the time you'll spend hours trying to figure out why you are hearing a ringing sound instead of what you should hear.
13:58.06ManxPowerThere are a SMALL number of situations where you want to override with "r"
13:58.21oOllipuzzled: found! http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO
13:58.35nighty-ManxPower: well to be honest when I did not have this "r" option on , my PAP2 did return a very strange sound
13:58.42oOllipuzzled: but THAT seems not to be stable in a production server :-)
13:58.47nighty-ManxPower: only when calling between the PAP2 ports
13:59.26ManxPowernighty-: then the real problem is why you are hearing the wrong tone.
13:59.53nighty-ManxPower: well I don't have this problem when calling my DIDs from outside
14:00.02zurabDarthClue: can you tell me a liitle bit more about dev libraries?
14:00.02ManxPowerAriel_: I guess I should change it to "only gay men use the 'r' option to Dial"
14:00.07nighty-ManxPower: they stil end up being routed to the same ATA
14:00.21Ariel_rofl
14:00.22nighty-ManxPower: I am clueless I must say
14:00.32ManxPowernighty-: time will change that
14:00.53ManxPowernighty-: the problem is that SO many examples were posted by newbies and those examples have 'r'
14:00.54nighty-ManxPower: I don't get this strange sound when routing call from outside to this same PAP2 unit
14:00.58Ariel_nighty-, what codec are you using?
14:01.12nighty-Ariel_: g729
14:01.21*** join/#asterisk coppice (~chatzilla@4.199.17.210.dyn.pacific.net.hk)
14:01.22nighty-Ariel_: but I don't have yet licences
14:01.25Ariel_nighty-, ok the pap2-na can only use one
14:01.30nighty-Ariel_: for asterisk
14:01.33Ariel_so change the codec to ulaw
14:01.40Ariel_nighty-, no the device
14:01.44nighty-Ariel_: oh shit
14:01.50nighty-Ariel_: they are crap
14:02.16nighty-Ariel_: you can't receive 2 voice calls on these with g729 ?
14:02.39Ariel_nighty-, correctly they only support one of there ports as g729
14:02.48nighty-Ariel_: BAD
14:02.54riemenscis bindaddr = 0.0.0.0
14:02.55nighty-Ariel_: not handy at all
14:02.56riemenscbindaddr = 0.0.0.0
14:02.57Ariel_same with the sipura 2000.
14:02.58ManxPowernighty-: That's a pretty common limitation
14:03.02riemenscis bindaddr = 0.0.0.0
14:03.11riemenscis bindaddr = 0.0.0.0 correct?
14:03.26nighty-ManxPower: well it is one that I did not know about (until now)
14:03.27jake1932ok - so this is going to be additional for the 7940 SIP firmware?
14:03.28Ariel_riemensc, yes if it's means listen to all nic
14:03.34ManxPowerriemensc: just don't use bindaddr at all. It will default to something correct
14:03.40*** join/#asterisk adjacent (~scott@office.bftwave.com)
14:04.04nighty-ManxPower: may I say "it sux" ? (stress relief)
14:04.32ManxPowernighty-: The Cisco ATAs also have that limitation, I believe that the Cisco phones have that issue as well.
14:04.38ManxPowerThe Polycoms may also have that issue.
14:04.56riemenscwhat is the correct configuration for bindadd ?
14:05.18ManxPowerriemensc: That would depend on what you want to do.  Most of the time just leave it commented out.
14:05.19Ariel_riemensc, it's used more when you have more then one network card in the system.
14:05.50puzzledanyone have a clue what's causing tons of clicks in MoH and not during normal calls as those are crystal clear
14:05.50riemensci´ve got 1 network card can i disable this option?
14:06.01Ariel_nighty-, most devices don't have the horse power for more then one g729 setup.
14:06.18ManxPowerriemensc: LISTEN.  just comment it out.
14:06.39riemenscokay
14:07.08jake1932anyone in here a cisco approved reseller?
14:07.16nighty-Ariel_: better use G711u (ulaw) or G711a (alaw) ?
14:07.24ManxPowerjake1932: What is your specific question?
14:07.30jake19327940 firmware
14:07.34Ariel_nighty-, where are you? US/Canada or EU?
14:07.35ManxPowernighty-: what country are you in?
14:07.46ManxPowerjake1932: What is your question about the 7940 firmware?
14:07.50nighty-Ariel_: Europe (France, the land of frogs :))
14:07.52jake1932I need it
14:07.54Ariel_alaw
14:08.10ManxPowerjake1932: It's about $100 or so.
14:08.14riemensci´ve comment out and have the same problem
14:08.25jake1932ah - can't you cut me a break?
14:08.26nighty-Ariel_: that is also true for FAX ?
14:08.26ManxPowerriemensc: then bindaddr was NOT your problem.
14:08.39riemenscwhat do you think is my problem
14:08.47Ariel_riemensc, firewall
14:08.53ManxPowerjake1932: Do you also want me to burn you a copy of MS Office and give you my CD key too?
14:09.02jake1932right
14:09.21puzzledManxPower: any idea what could cause tremendous clicks/screeches in MoH and not during normal calls. this is stable rfom today
14:09.22jake1932what I'm saying is - aren't there some high volume reseller's that can get a better deal?
14:09.27Ariel_jake1932, lisc is around 100 dollars and smartnet to get it is about 9 dollars.
14:09.32riemenscmy firewall is not active i use fvs318 netgear and forward all from extern to 192.168.199.100 this is the server ip
14:09.39ManxPowerjake1932: One of the significant reasons we don't use Cisco phones is because the firmware costs extra
14:09.55jake1932hmm - that's bs
14:10.01jake1932i'm boycotting them
14:10.01ManxPowerriemensc: So Your Astrisk is behind NAT?
14:10.05riemensces
14:10.06riemenscyes
14:10.13Ariel_jake1932, that is why I use polycoms mostly.
14:10.19NuggetIt can be a real challenge to buy the contract for just a single cisco phone.  I spent over a month bickering back and forth before cdw finally accepted defeat and sold me one
14:10.23ManxPowerriemensc: and are you also using externip and localnet options?
14:10.29riemenscyes
14:10.35nighty-Ariel_: whats the differnce between alaw and ulaw ?
14:10.39jake1932does SCCP work well with *?
14:10.41riemenscexternip via dyndns.org
14:10.45Nuggetjake1932: no
14:10.53ManxPowernighty-: ulaw is what north america uses, alaw is what the rest of the world uses.
14:10.54jake1932:(
14:11.09ManxPowerriemensc: I don't believe externip supports a hostname.
14:11.22jake1932$300 for a phone and it better make me breakfast
14:11.22Ariel_jake1932, yes but with lots of limitations
14:11.26Hmmhesaysomg someone in nigeria wants to send me 170 million dollars
14:11.45pauldyor at least come with its own pr0n feed
14:11.49ManxPowerHmmhesays: Be a nice guy and give him your Boss's banking info.
14:11.50Ariel_Hmmhesays, I can send you the one from the Philipines to go along wit that one.
14:11.55jake1932hehe
14:12.05HmmhesaysI'm on the internet... wheeeeeeeee!
14:12.27riemenscshould I rather register 0.0.0.0?
14:12.28ManxPowerCorrection: I don't believe externip supports a hostname in 1.0.x, but should in CVS-HEAD
14:12.35NuggetIf I do buy more ciscos, there's zero chance I'll be shopping for price.  All I will look for is a vendor who can sell me the whole kit all at once.
14:12.48Ariel_ManxPower, it does on mine
14:12.51nighty-Ariel_: you mean , I have to configure both ports with 'alaw' I can't have 1 port with G729 and 1 port with 'alaw' ?
14:12.57ManxPowerAriel_: you are using CVS-HEAD?
14:13.00Ariel_is use externip=kasipbx.homedns.org
14:13.03Ariel_stable
14:13.15ManxPowernighty-: Yuo can use different codecs for different ports
14:13.18Ariel_I also set it up in my hosts file
14:13.20riemenscmy ip configuration for dydns.org is correct
14:13.31Hmmhesaysheh debian packages 1.0.7 as stable
14:13.39riemenscwhat have you setup in hosts ?
14:13.54ManxPowerAriel_: *nod*, and if your external ip address changes, how long does it take for Asterisk to see the change?
14:14.05puzzledHmmhesays: amazing that they are already at 1.0.7 :)
14:14.14riemenscevery 24h change my ip
14:14.20Ariel_ManxPower, I really don't know have not seen it as a problem. (
14:14.26Hmmhesaysmust be a pretty good one
14:14.28Hmmhesays<chuckle>
14:14.35nighty-ManxPower: well does not seem to make it work any better
14:14.35ManxPowerIn my experience is that if there is ever any transient DNS issue, Asterisk will never retry the lookup
14:15.01riemensci starting the server every 12h again via cronjob
14:15.03ManxPoweras far as I'm concerned that means "it doesn't work"
14:15.16riemenscwhat have you write in /etc/hosts
14:15.24*** part/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com)
14:15.35Ariel_I have my local address there.
14:15.45riemenscNAME.dyndns.org localhost is this correct
14:16.03Ariel_192.168.xxx.xxx    kasipbx.homedns.org
14:16.03riemensci´ve got no my local adress in hosts
14:16.33xhelioxManxPower: Yesterday you said the outer pins on the TDM400P weren't used for anything, right? I failed to ask, are they connected to anything on the card? Reason I ask is that if I use a 2 pair wire, the card doesn't work right, but when I use a 1 pair, it's fine. Is there anything I have to do to disable those pins? (I'm 1000 miles away from the site, so it's not feasiable to change the cables right now)
14:16.35riemensci change and testing again
14:16.39So3krishello i'm bizzy with musiconhold and i recive noise thure the phone. mus i have alsa installed or is mpg123 enugh ?
14:17.21Ariel_So3kris, no
14:17.43Ariel_mpg123 should work. But in head there is a new way to do moh
14:17.44pauldySo3kris, maybe you need the zaptel dummy driver loaded
14:17.58Ariel_moh does not need ztdummy
14:18.06xhelioxOh, he left. :)
14:18.09*** join/#asterisk yaaar (~chatzilla@lifebook.tranquility.net)
14:18.12yaaarword
14:18.17DarthCluesentence.
14:18.20pauldyreally one of the wiki pages says you need it for timming
14:18.30yaaarDarthClue: that's not a sentence!
14:18.35DarthClueyes it is!
14:18.35Ariel_pauldy, you need it for meetme and iax2 trunking
14:18.57So3krisi have no meetme and iax2
14:19.09*** join/#asterisk wunderkin (kev@24.137.147.163)
14:19.25Darwin35you must not have ztdummy or rtc
14:19.31Darwin35setup
14:19.31So3krisi have looked in al the wiki pages but have not seen a otherway than i have
14:19.56Darwin35you have to uncomment in the make file to make ztdummy
14:19.56*** join/#asterisk hullam (~hullam@61.68.149.118)
14:19.59pauldyhmm I must have read it wrong but mine is working great
14:20.06*** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com)
14:20.22Darwin35you have to have timing for iax and meetme
14:20.45Ariel_Darwin35, there talking about mpg123 and moh not meetme or iax2 trunking
14:20.54hullamwhere can i buy digium cards from?
14:21.00Nuggetdigium.com
14:21.29Darwin35look on the digium site
14:21.35Darwin35it tells you
14:21.42hullamany place cheaper?
14:21.51Ariel_also you can get them from voipsupply, atacomm and other distributors
14:21.56hullamlooks very costly
14:22.03Darwin35what card
14:22.21Darwin35no telecom hardware is cheap
14:22.35hullami need one with 4 fxo on single pci
14:22.41Darwin35and you get support with the purchase
14:22.48nighty-Ariel_: now it ring the other extension , but I get a strange noise
14:22.55nighty-Ariel_: no voice goes thru
14:23.10Ariel_nighty-, put both on alaw and test it out.
14:23.24hullami am from australia, those cards work with australian telco?
14:23.37Ariel_hullam, the 4 fxo board from digium is the least cost way to get 4 fxo's.
14:24.20*** join/#asterisk echo465 (Edward@4.17.192.145)
14:24.21hullamany other card with 4 fxo that works with asterisk?
14:24.26Ariel_unless you get 4 sipura 3000's and use the 4 fxs as extensions. Then the 400 dollars is a better deal
14:24.34Hmmhesayssangoma
14:24.44hullamany other vendor with price around 100 bucks?
14:24.53Ariel_Hmmhesays, he is looking at less money
14:25.17Hmmhesaysnod, I just answered the question of "<hullam> any other card with 4 fxo that works with asterisk?" didn't see the money part
14:25.25riemenscthe same problem again
14:25.36Hmmhesaysthe 4 spa-3000's isn't a bad deal
14:25.37Ariel_Hmmhesays, yes I figured that
14:25.47Nuggetthe 4 spa-3000's is probably a lot less grief too.
14:25.52Hmmhesaysprobably uses a little more power
14:26.04Nuggetmaybe you can buy one of ManxPower's old tdm400ps for cheap.  :)
14:26.12riemensci cannot hear the other person
14:26.24riemenscthe other person hear me
14:26.29Ariel_I wounder if we can take the 4 put them into a box and use one powersupply and well never mind....
14:26.47HmmhesaysAriel_ probably not with the stock psu
14:27.41Ariel_riemensc, I still think you have an issue with the netgrear I had a customer yesterday with weeks of problem with it and B/V until he upgraded his firmware it's working now.
14:28.02HmmhesaysI agree with Ariel_ netgears can be real trouble
14:28.53hullam<PROTECTED>
14:29.21hullam200 bucks each
14:29.34Ariel_hullam, but you get one fxo and one fxs.  There around 99 dollars here
14:29.34Hmmhesaysyeah but they have 1fxs and 1 fxo port each
14:30.21hullamseems like i am better off buying 4 pci card with 1 fxo each from ebay
14:30.26hullam9.99 each pci
14:30.49nighty-Ariel_: both to alaw , they work
14:30.50crash3ma lot cheaper than a 4 * fxo card
14:30.53riemensci´ve change the rtp.conf and i hear my not and the other here me not
14:31.04nighty-Ariel_: but when calling from outside , now I get the strange sound again
14:31.28nighty-Ariel_: I think codec negociation problem maybe ?
14:31.33Ariel_nighty-, that is the transcoding
14:31.45nighty-Ariel_: what is wrong ?
14:31.49greg_workhullam: then you probably need 4 IRQ's to run them without echo/delay problems
14:32.01greg_workhullam: a tdm400p only requries 1 irq
14:32.09Ariel_hullam, yes but you will not get 4 of them working correctly in a system.
14:32.17nighty-Ariel_: in this case transcoding can not work because it needs real codecs ?
14:32.24nighty-Ariel_: on the asterisk side ?
14:32.25hullamshoot, didnt know that
14:32.41hullamhow many pci max can be used in one box?
14:32.43Ariel_nighty-, did you install the lisc for g729 on the asterisk box?
14:32.51Hmmhesaysas many as you have irq's for?
14:32.53nighty-Ariel_: no
14:32.55riemenschow can i find the actual version number from my netgear?
14:33.03nighty-Ariel_: not that I know
14:33.09Ariel_nighty-, that is your problem then.
14:33.14nighty-Ariel_: I don't know how to install lisc :)
14:33.16Hmmhesaysriemensc: you have to smash it apart to get the serial number
14:33.21nighty-Ariel_: I will look in the docs
14:33.31hullami have 6 pci slot with 2.6 kernel ... how many irq do i have? is that set by bios?
14:33.37*** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com)
14:33.37*** mode/#asterisk [+o anthm] by ChanServ
14:34.07nighty-Ariel_: what you call lisc is the g729 codecs licences ?
14:34.45Ariel_hullam, I have never gotten more then 2 x100p to work correctly in any system.
14:34.54*** join/#asterisk HaHaOok (~norman@CPE-203-45-194-185.qld.bigpond.net.au)
14:34.57Ariel_nighty-, digium 10 dollars per lisc.
14:35.13Ariel_nighty-, yes
14:35.44hullamgnugk and asterisk - can they both run in one box?
14:36.07hullamasterik has any gk functionality in it?
14:36.13Nuggetzaptel is flaky even in the best of circumstances.  There's no amount of cost savings that would make a four way clone x100p make sense.
14:36.33riemensci use at the moment V1.3 Jan. 29 2003 can i upgrade to 3.0
14:36.35Nuggetif you can't afford to buy the right hardware asterisk just isn't for you.
14:36.41riemenscnetgear fvs318
14:37.06nighty-Ariel_: ok :)
14:37.22*** join/#asterisk doughecka_ (~Miranda@doughecka.user)
14:37.43riemenscwhat do you think is better Firmware Version 2.4 or Firmware Beta Version 21 RC2?
14:37.45nighty-Ariel_: how many licences you need depend on how many simulatenous calls your are handling ?
14:37.48Darwin35by the year 2025 all homes will have to have thier own pbx
14:37.56BeirdoNugget: that's a specious statement
14:38.26Hmmhesaysby the year 2025, we will all be wearing star trek style communicators
14:38.29hullamanyone here from australia?
14:38.30echo465Darwin: by the year 2025, most people will still have appliances flashing 12:00
14:38.33Darwin35if you to cheap to buy the tight hardware then asterisk is not for you
14:38.41NuggetI disagree.  THe guy's going to go out and blow $40 on a bunch of clone x100p cards which he'll never be able to get working.
14:38.42Hmmhesaysand thinking of the good ol' days where the kids weren't so lazy
14:38.55NuggetHe'd be better of just not trying at all.
14:38.55*** join/#asterisk shmaltz (~chatzilla@69.28.255.210)
14:39.13BeirdoNugget: my zaptel crappy cards work 90%+ of the time
14:39.18Nuggetmaking asterisk work well with pstn is expensive.
14:39.20newlhullam: I'm IN Australia but not FROM, does that count? :)
14:39.36Beirdoand it will get him to a point where he can justify buying the expensive stuff
14:39.38hullamhell ya ... what card do u use newl?
14:39.47newlcard?
14:39.51hullamanyone using voicetronix?
14:40.02Nugget90%?  good god your expectations are low.
14:40.26Nuggetand I'll bet you don't have four of them in one machine.
14:40.39Broomanyone has a working config/example on how to create conference rooms? im getting an invalid conference room error even though they are at meetme.conf
14:40.40Broom??
14:41.33*** join/#asterisk santiago (~santiago@63.245.86.141)
14:41.44*** join/#asterisk PatrickDK (patrickdk@dyn-19-218.myactv.net)
14:41.51BeirdoI have two of them in there
14:46.55Darwin35its simple
14:47.03Darwin35its in the wikik
14:47.08echo465Question:  I have a T100p, connected to a Verizon PRI.  It worked.  Then we lost power.  Now, outbound calls are returning 'all circuits are busy', and inbound calls are just dropping.  The console is showing 4x '== Primary D-Channel on span 1 up', then 1 '== Primary D-Channel on span 1 down'.
14:47.20Darwin35learn to read the wiki pages at www.voip-info.com
14:47.25echo465any suggestions on troubleshooting this problem?
14:47.53shmaltzecho465, call Verizon
14:48.09*** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com)
14:48.23nighty-Ariel_: just one more question :)
14:48.26shmaltzBroom, you have a timer source?
14:48.45Ariel_nighty-, ask away
14:48.52nighty-Ariel_: the lisc .so goes where ? (I guess some library path /usr/local/lib ???)
14:49.16ManxPowernighty-: you mean for G729?
14:49.18nighty-Ariel_: I ordered the codecs but they say it takes 24 hours to get
14:49.19nighty-:(
14:49.39nighty-ManxPower: yes I downloaded the 5.2.1 FreeBSD and the Register program
14:49.50ManxPowernighty-: then you need to follow the directions
14:49.55BroomBeirdo: you answered to me?
14:50.00ManxPowereither the README or the instructions that come with the license key
14:50.07Beirdowho me?  no
14:50.17nighty-ManxPower: uhmm so I have to wait 24Hours then :)
14:50.19Broomoh ok, thought so
14:50.34nighty-ManxPower: I took 5.2.1 FreeBSD even though I am using 5.4 and 5.3 :)
14:50.37shmaltzthis guy is over I think:
14:50.39shmaltzhttp://www.thesun.co.uk/article/0,,2-2005340731,00.html
14:50.58ManxPowernighty-: If you think 24 hours is a long time, just wait until you order your first E-1
14:51.13*** join/#asterisk santiago (~santiago@63.245.86.141)
14:51.46Beirdoheheh, yeah
14:51.46RoyKilt~sex
14:51.47jbotupdatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep, or super extractor, http://sf.net/projects/sex/
14:52.48*** join/#asterisk mithro (~tim@87.76.42.120)
14:53.01nighty-ManxPower: E1 card ?
14:53.06nighty-ManxPower: or E1 line ?
14:53.32nighty-ManxPower: I think you are talking about Digium so must be the card
14:53.39shmaltzinteresting, just opens up new posiblites with asterisk:
14:53.41shmaltzhttp://news.yahoo.com/s/cmp/20050729/tc_cmp/166401966/nc:1817;_ylt=Aszk.chdFR2ew5FJzOSqoDKor7oF;_ylu=X3oDMTBiMW04NW9mBHNlYwMlJVRPUCUl
14:53.59nighty-ManxPower: I have not ordered anything from Digium yet....except the codecs
14:54.23nighty-ManxPower: I am just starting to use Asterisk as you certainly noticed :)
14:54.57*** join/#asterisk _T3_ (~rposada@35.229.uio.satnet.net)
14:55.34nighty-ManxPower: You are in Europe ?
14:55.40ManxPowernighty-: No, but I wish I was.
14:55.51nighty-ManxPower: why ?
14:56.00ManxPowerso a /whois ManxPower
14:56.13*** join/#asterisk Cresl1n (~Cresl1n@207.111.174.1)
14:56.15ManxPowernighty-: because the USA is becoming a theocracy
14:56.24zoacreslin!!!!!
14:56.41Cresl1nzoa!!!!!
14:56.45nighty-ManxPower: lol
14:58.37Cresl1nManxPower!!!!!
15:00.04Darwin35ok gays to the right straights to the left Dragqueens centerstage with the women
15:00.44blitzrageDarwin35: you keep getting wierder and wierder
15:00.54*** join/#asterisk doughecka_ (~Miranda@doughecka.user)
15:01.04_T3_and the people that needs some help where stay?
15:01.05blitzrageanyone going to TAUG tonight?
15:01.06Darwin35?
15:01.12*** part/#asterisk Darwin35 (~darwin@ip70-186-117-198.ma.dl.cox.net)
15:01.21blitzragetzanger: I'm looking in your direction :)
15:01.22*** join/#asterisk Darwin35 (~darwin@ip70-186-117-198.ma.dl.cox.net)
15:01.34Darwin35grr
15:02.09*** join/#asterisk sparky0001 (~mark@mark.keele.netcentral.co.uk)
15:02.16Beirdoblitzrage: Toronto?
15:02.29blitzrageBeirdo: yes sir
15:02.32Beirdohmm, I don't think I can make it yet again.  blargh!
15:02.38blitzrageBeirdo: North York to be specific
15:02.42tzangereh?
15:02.45blitzrageBeirdo: doh!
15:02.46tzangerno I have my kids tonight
15:02.50Beirdowhere, when?  I don't remember if I got the email
15:02.51tzangerfriday nights don't work well for me
15:02.51blitzragetzanger: bring'em along! :)
15:02.54tzangerhahaha yeah
15:03.05Beirdowonder if John Sellens will be there tonight
15:03.06blitzragetzanger: yah, we gotta think of a different night to do the meetings
15:03.06Beirdoheh
15:03.21tzangerthursday nights were good but fridays can get good
15:03.21nighty-blitzrage: and place ? :)
15:03.25blitzragetzanger: what nights work good for you? (and anyone else for that matter)
15:03.26tzangerI'm just busy with other stuff too
15:03.29nighty-blitzrage: how about Paris France ? :)
15:03.37sparky0001Does anyone know why when using a Digium E1 card in the UK the caller ID by default drops the leading 0 of the telephone number as this is really annoying me now and I cannot seem to correct it?
15:03.47blitzragenighty-: ummm... TAUG - Toronto Asterisk Users Group .... so I think we'll host it in Toronto :)
15:03.49tzangersparky0001: add a 'w' before the #
15:03.51Beirdoblitzrage: well all nights pretty much equally are suckily booked :)
15:03.57blitzrageBeirdo: lol
15:04.11nighty-blitzrage: bummer :)
15:04.17blitzragesparky0001: sounds like you've got a ${EXTEN:1}
15:04.27_T3_hi every body
15:04.34nighty-Any webpage for this ?
15:04.36puzzledphew, figured out that MMX support in zaptel was causing the clicks in MoH
15:04.37tzangersparky0001: oh wait
15:04.38tzangercaller ID
15:04.39tzangersorry
15:04.40nighty-blitzrage: URL ?
15:04.46blitzragenighty-: www.taug.ca
15:04.50blitzrage~taug
15:04.50jboti heard taug is The Toronto Asterisk Users group. The website can be found at http://www.taug.ca
15:05.06filemy name is Willy Wonka!
15:05.08mutlook at those ugly busses
15:05.20ManxPowerpuzzled: enableing MMX in zaptel will also cause random crashes
15:05.33_T3_i need help with SendText cmd i need how to send the message to the caller not to callee....
15:06.01Kattymew
15:06.01puzzledManxPower: I asked the other day and people seemed to use it do I thought I'd go with it. clearly that was not a good idea
15:06.03sparky0001I have not got the ${EXTEN:1} in my scripts the dubug on the PRI card seems to just show the caller ID without the zero ?
15:06.16Beirdoblitzrage: I'll see if I can free myself for that
15:06.45_T3_i read the code but i dont understand how to identify the callee channel
15:07.03Beirdo5095 Yonge...  hmmm
15:07.04ManxPowersparky0001: Um, leading "0" and "00" is NOT part of callerid
15:09.11Beirdoso that's the North York town-centre stop, right, blitzrage?
15:09.52_T3_sorry
15:10.29sparky0001ManxPower: I guessed it may not be it's just annoying on the phones as they do no show the full number and so you cannot hit redial when this happens any ideas?
15:10.49ManxPowersparky0001: prepend the leading 0 or 00 in your dialplan to set the callerid
15:12.40*** join/#asterisk |nix (~inix@cm11.gamma116.maxonline.com.sg)
15:12.46sparky0001ManxPower: I have got it to do this per extension is there a quick way to add the pre-pend as a global ?
15:12.50doughecka_ManxPower: howdy
15:13.09doughecka_ManxPower: zttest is showing 99.963379... thats not good correct?
15:13.48ManxPowersparky0001: I wrote an AGI script to do that.
15:13.55blitzrageBeirdo: yep, Toby's is right across the road from that subway stop
15:14.02Beirdoperfect
15:14.10BeirdoI'll try to make it
15:14.11blitzrageBeirdo: yah, super easy to get to
15:14.17blitzrageBeirdo: sweet!
15:14.24BeirdoI live in harbourfront, work downtown
15:14.32blitzrageI'm in Oakville :)
15:14.39blitzragebut I just take my bicycle on the trains
15:14.42*** join/#asterisk wrmem (~monnin@wireless-test-227.slip.uiuc.edu)
15:15.17Beirdonice
15:15.28Beirdoso yeah, easy for me to get to
15:15.47puzzleddoughecka_: don't think that is too good. is that with or without that RTC patch thingy enabled?
15:15.52Beirdoand I might be able to get there, have to talk with the woman who manages to use all my time :)
15:15.55Beirdohehe
15:15.56_T3_hey some attention here please
15:16.00blitzrageBeirdo: damn women :)
15:16.00Beirdonot that I'm complaining
15:16.42_T3_cmd sendtext help?
15:17.13ManxPower_T3_: Since you have not told us what the problem is....
15:17.30blitzrageBeirdo: just bring her along, we're all fun people! :)
15:17.32puzzledManxPower: do you enable MMX support (-K6OPT) in asterisk or leave it disabled?
15:17.37Beirdohah, good luck
15:17.42Beirdoshe's not in this country
15:17.56blitzrageBeirdo: lol
15:18.01blitzrageBeirdo: then what's your excuse! :D
15:18.24ManxPowerpuzzled: I leave zconfig.h:/* #define CONFIG_ZAPTEL_MMX */ commented out
15:18.24blitzragebrb, gotta go rotate the laundry
15:18.26Beirdowell BECAUSE it's at a distance...  you gotta spend all the time you can together, etc
15:18.29Beirdohehe
15:18.34*** join/#asterisk doughecka_ (~Miranda@doughecka.user)
15:19.00puzzledManxPower: yup have done that and it is a great improvement but what about the -K6OPT in asterisk/Makefile that enables MMX stuff. do you enable that?
15:19.29ManxPowerpuzzled: Um, I never touch the makefile except for one option to make SIPura's using G726 work with Asterisk
15:19.41puzzledManxPower: ok, thanks
15:20.05doughecka_yay
15:20.10doughecka_--- Results after 206 passes ---
15:20.10doughecka_Best: 99.975586 -- Worst: 99.951172 -- Average: 99.963355
15:20.17doughecka_ManxPower: thats not good eh?
15:20.25*** part/#asterisk wrmem (~monnin@wireless-test-227.slip.uiuc.edu)
15:20.39ManxPowerdoughecka_: not really.
15:20.43doughecka_meh
15:20.53ManxPowerdoughecka_: try running zttest while doing a "find / -print"
15:21.30_T3_ManxPower: i have a pa1688 phone that handle sip messages i change the software to show the message in the lcd screen
15:21.36doughecka_it jumped to 100%, then its back to 99.975
15:21.54*** join/#asterisk Saaib (~nabudocon@200.76.231.14)
15:22.05doughecka_I assume that could introduce echo
15:22.11_T3_but know i want asterisk to send sip messages to phone the problem is that the cmd SendText only send messages to the callee
15:22.15doughecka_now its at a steady 99.9755
15:22.23puzzleddoughecka_: exact same value for me
15:22.32doughecka_--- Results after 79 passes ---
15:22.32doughecka_Best: 100.000000 -- Worst: 99.975586 -- Average: 99.976204
15:22.41doughecka_puzzled: do you have issues?
15:23.01*** join/#asterisk wrmem (~monnin@wireless-test-227.slip.uiuc.edu)
15:23.07doughecka_call comes in, and you pick it up and its dial tone
15:23.15_T3_anybody knows how to change the code is the first time i read asterisk code i dont know how to change the info to send the message to the caller
15:23.15puzzleddoughecka_: without MMX none but it is only a very lightly loaded box
15:23.25doughecka_so is mine
15:23.30doughecka_thats without any calls going through
15:23.37*** join/#asterisk ilTizio (~pelo@adsl203-149-051.mclink.it)
15:23.59puzzleddoughecka_: it's an old PII-400 with an Eicon Diva Server card and * from today's stable cvs
15:24.25doughecka_meh, mines a fast pentium 4 with a T1 card and an adtran
15:24.32doughecka_oooh
15:24.33puzzledhehe
15:24.36doughecka_its 100%
15:25.02puzzleddoughecka_: did you turn off usb and ide if you use scsi?
15:25.12puzzleddoughecka_: and parallel and serial ports
15:25.27mutanyone in here use Imail?
15:25.42PatrickDKI wonder if you can tern off the console, vga, keyvboard
15:25.56_T3_ManxPower
15:26.14doughecka_yep
15:26.18doughecka_turned everything off
15:26.21puzzledPatrickDK: remove the vga card and specify console=ttyS0 but then you would use a serial port with a null modem cable
15:26.22ManxPower_T3_: I still don't understand wha you are trying to do.
15:26.27doughecka_huh, its now at 99.98
15:26.43_T3_which part?
15:27.05ManxPower"send message to caller"
15:27.39_T3_ok read this: you can use the sendtext command like this:
15:27.51ManxPower_T3_: The device has to support sendtext
15:27.58*** join/#asterisk Veto (mdkuser@cpe-66-69-38-192.satx.res.rr.com)
15:27.59ManxPowerI don't know of any device that supports sendtext
15:28.13_T3_no no
15:28.22ManxPower_T3_: What are you trying to DO?  Inform the caller of something?
15:28.35_T3_i use a sip phone that support sip messages and info method
15:28.38ManxPower_T3_: change the callerid?
15:28.45ManxPower_T3_: Asterisk does not support that.
15:29.07_T3_asterisk sends sip messages
15:29.18_T3_using the sendtext command
15:29.28twisted[asteria]HACK TEH PLANET!
15:29.33ManxPowerI've managed to send an "IM" to a Polycom phone using sipsak
15:29.33blitzragetwisted[asteria]: lol
15:29.42twisted[asteria]lol - sup blitz
15:29.46ManxPower_T3_: Try it and watch your calls get disconnected.
15:30.10ManxPower_T3_: So, what happens when you try to use sendtext?
15:30.17_T3_but asterisk send the messages in one way --> to callee
15:30.21blitzragetwisted[asteria]: not too much, doing some laundry and working on Asterisk. Trying to get sheat done before the TAUG meeting tonight, them I'm going camping for the weekend (I might even dabble in some intoxication)
15:30.29_T3_i want to send the message to the caller
15:30.40twisted[asteria]blitzrage, camping?  In canada?  What about the polar bears?
15:30.48blitzragetwisted[asteria]: I've got a shotgun
15:30.51twisted[asteria]ah
15:30.55blitzragetwisted[asteria]: makeup shotgun
15:31.00ManxPower_T3_: What message do you want to send to the caller?
15:31.08blitzrage"I think you have it set to whore"
15:31.18_T3_rate info
15:31.29twisted[asteria]blitzrage, lol
15:31.29_T3_or time left info
15:31.34twisted[asteria]file!
15:31.37filetwisted!
15:31.39_T3_like calling card
15:31.49ManxPower_T3_: I cannot help you.
15:33.37_T3_why?
15:34.11twisted[asteria]because you've been bludgeowned
15:34.42*** part/#asterisk santiago (~santiago@63.245.86.141)
15:35.28*** part/#asterisk sparky0001 (~mark@mark.keele.netcentral.co.uk)
15:35.44*** join/#asterisk focks (~cbruender@nsc66.147.95-93.newsouth.net)
15:36.16fockswhat do you need to do to enable changes made to zapata.conf? reload/restart?
15:36.44twisted[asteria]focks, what version of *, and what sort of changes?
15:36.58focks1.0.7 adjusting echo settings
15:37.01twisted[asteria]ooooh
15:37.03twisted[asteria]yeah, restart
15:37.32ManxPoweror unload chan_zap.so and load chan_zap.so
15:37.40*** join/#asterisk Godsey (~lanny@Godsey.sustaining.supporter.pdpc)
15:37.55fockstwisted[asteria], i'm having pretty nasty echo . using a PRI with SIP and Polycom soundpoint 501s
15:38.20twisted[asteria]focks, good luck with that
15:38.31fockslots of fun
15:38.42*** join/#asterisk pbxbart (user@p54B01C7C.dip0.t-ipconnect.de)
15:38.51*** join/#asterisk florz (nobody@2001:1a50:503c:0:0:0:0:1)
15:39.21So3krishello iÅm new with asterisk but can;t you transfer a call. not directly ? extern call-->respionist-->message-toe-sells-from-receptionist->extern-->sells(man/woman)
15:39.49florzis there any free softphone, sip or iax, that is at least technically good quality? The user interface doesn't matter?
15:40.02florzs/\?$/!/
15:40.07So3kriswww.sjlabs.com
15:40.23So3krisor unix ?
15:40.28So3kriskphone i use
15:40.35*** join/#asterisk mrme (asdf@209.50.203.45)
15:40.57mrmecan someone recommend a sip provider
15:41.12*** join/#asterisk JohnJacob (~JohnJacob@pcp0011543387pcs.mainf01.in.comcast.net)
15:41.21florzSo3kris: Yep, linux. And nope, kphone sucks.
15:41.46So3kriswhy ?
15:42.21mrmethinking about broadvoice since i read alot of comments, but is it the best choice at the moment?
15:42.36*** join/#asterisk harryvv (~none@S010600a0c93f6f7e.vs.shawcable.net)
15:42.42DarthCluemrme: what do you want to do with said sip provider?
15:43.04So3krislinphone ? mythphone ?
15:43.05florzSo3kris: I don't know the most recent version, so maybe I'm wrong, just had a look at the changelog. Most of all, it doesn't have any sensible jitter buffer.
15:43.18*** join/#asterisk doughecka_ (~Miranda@doughecka.user)
15:43.31florzSo3kris: I just tried linphone, latency is far too high.
15:43.34*** join/#asterisk mithro (~tim@87.76.42.120)
15:43.36doughecka_grr
15:43.41doughecka_ManxPower
15:44.02florzSo3kris: around 650 ms echo RTT over the LAN.
15:44.13So3krisflorz: how can you see that
15:44.26florzSo3kris: Hear, not see =:-)
15:45.11So3krisi have a home network and it soundśs not to bad. i have ordered a cisco
15:45.24mrmeanyone using the broadvoice service
15:45.48ManxPowermrme: search the mailingl ist for info about broadvoice
15:46.13|nixhey guys, does anyone knows if asterisk 1.2 intends to have simple?
15:46.41|nixfor instant msg and stuff?
15:46.45harryvvsimple what
15:46.54doughecka_ManxPower: ever see a dialtone trigger a channel bank to act as if its recieving a call?
15:46.56|nixmsg, presence for example...
15:46.58doughecka_its like its too sensitive
15:47.02doughecka_plus I still have echo
15:47.06ManxPowerdoughecka_: no
15:47.24doughecka_any ideas?
15:47.38doughecka_it was working fine yesterday
15:47.44doughecka_of course I take it onsite and it starts acting up
15:47.45florzSo3kris: Yes, it doesn't sound bad if the "line" is good, like a LAN. However, add 100-200 ms latency over the internet and it becomes a problem. Add a few tens of ms of jitter and it starts crackling, too.
15:47.51ManxPowerdoughecka_: You should never ever get echo on calls Analog->channelbank->asterisk->T-1toTelco
15:48.18doughecka_this is telco -> channel bank -> asterisk -> voip
15:48.24doughecka_no analog side
15:48.32doughecka_channel bank's FXO lines
15:48.34So3krisflorz: but how can you see if you have latency ?
15:48.58florzSo3kris: Just call an asterisk echo extension and make some noises? :-)
15:49.05ManxPowerdoughecka_: echocancel=yes echotraining=900 and lower the gains from the channel bank to the telco
15:49.06mrmeManxPower what mailinglist
15:49.13ManxPower~mailinglist
15:49.13jbotsomebody said mailinglist was Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
15:49.15doughecka_gaims are 0
15:49.21ManxPowerdoughecka_: then lower them
15:49.38ManxPowerwe need txgain=-6.0 on our PRI to eliminate echo
15:49.42doughecka_oh?
15:50.05florzSo3kris: Basically, if you can measure the latency that way, it's too much ;-)
15:50.55So3krisi wil look at it.
15:51.21So3krisiḿ bizzy with thransfer of a call
15:51.33harryvvmanx, doesnt that reduce the volume of the call?
15:51.52doughecka_hmm
15:51.58doughecka_it seems to improve it a little
15:52.02doughecka_still have echo
15:52.12doughecka_before it was perfect, then I hear a click and then eacho starts
15:52.15ManxPowerkeep lowering it
15:52.28ManxPoweryou hear a clock?  maybe you have a frame slip issue
15:52.34ManxPower..er... click
15:52.44harryvvdoughecka_ thats interesting because I think one time I recall hearing a click then the echo apeared once.
15:53.08doughecka_my zttest jumps all over the place
15:53.12ManxPowerharryvv: Correct.
15:53.20ManxPowerdoughecka_: that could be a problem too.
15:53.23harryvvSo what would cause that
15:53.29doughecka_keep lowering txgain?
15:53.34ManxPowerharryvv: a frame slip
15:53.36ManxPowerdoughecka_: yup
15:54.00*** join/#asterisk collink (~collin@207.111.174.1)
15:54.03harryvvseems a frame slip is the effect..what would cause it.
15:54.17doughecka_--- Results after 1078 passes ---
15:54.17doughecka_Best: 100.000000 -- Worst: 99.975586 -- Average: 99.990318
15:54.19Corydon-wI've also had success removing echo by leaving txgain alone and lowering rxgain
15:55.03*** join/#asterisk jfonsecausa (~jfonseca@c-66-176-57-28.hsd1.fl.comcast.net)
15:55.46ManxPowerharryvv: missed IRQ or a wrong timeing source for the line
15:56.16doughecka_crap
15:56.19*** join/#asterisk cresl1n (~Cresl1n@207.111.174.1)
15:56.20doughecka_this is crazy
15:56.25doughecka_I keep getting fake calls
15:56.28So3krisare there differnd ways to transfer a call ?
15:56.36mutmaybe someone is just screwing with your head
15:56.48ManxPowerSo3kris: yes
15:56.50doughecka_noo
15:56.51doughecka_:P
15:56.59doughecka_ManxPower: span=1,1,0,esf,b8zs
15:57.01doughecka_that better?
15:57.20ManxPowerdoughecka_: I don't know.  Is the channel bank providing timeing or expecting asterisk to?
15:57.25doughecka_expecting
15:57.28ManxPowerusually a channel bank expect to get it's timing from asterisk.
15:57.38doughecka_its expecting twins actually
15:57.39doughecka_:
15:57.40doughecka_:P
15:57.45ManxPowerso the span=1,1 is telling asterisk to get the timing from the channel bank
15:57.58doughecka_meh
15:59.22So3krisManxPower: know you some names off it or have you a link ?
15:59.53ManxPowerdoughecka_: try  span=1,0,0,esf,b8zs
16:00.09ManxPowerSo3kris: SIP Transfers and DTMF transfers
16:00.20doughecka_ok
16:00.39doughecka_now calls dont even go through
16:00.41ManxPowerunless you mean supervised/consultative transfer .vs. blind transfer
16:00.59So3kristhanx
16:01.30*** join/#asterisk BuckRogers (~steve@ool-44c29ac5.dyn.optonline.net)
16:01.33ManxPowerdoughecka_: t-1s need a min or so to "calm down" when they are reset
16:01.42BuckRogershello all
16:04.24*** join/#asterisk gaffneyc (~gaffney@70.88.90.25)
16:05.40*** join/#asterisk doughecka_ (~Miranda@doughecka.user)
16:05.57astoriadoughecka_: wait for the b channels to come up.
16:06.11*** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com)
16:06.12ManxPowerastoria: he's not using PRI
16:06.16doughecka_meh, still not workin :)
16:06.19astoriaManxPower: oh.
16:06.27astoriadoughecka_: never mind.
16:06.40ManxPowerdoughecka_: what does zttool report?
16:06.43astoriaDid you guys know today is systems admin appreciation day!?
16:06.49doughecka_99.975586
16:06.53doughecka_er
16:07.01fearnorlast tnt
16:07.02fearnorerr
16:07.02doughecka_it says OK
16:07.08doughecka_but...
16:07.26ManxPowerdoughecka_: don't worry about ttest right now
16:07.28doughecka_calling via a group doesnt work, picking up the line and manually dialing works
16:07.32ManxPowerzttest, even
16:08.26blitzrageastoria: I did not! Have a link to a website or something?
16:08.41doughecka_no alarms
16:08.43doughecka_internally clocked
16:09.05ManxPowerdoughecka_: the clocking info in zttool is a know bug
16:09.10doughecka_oh, ok
16:09.32Hmmhesaysanyone ever try using local channels with asteriskcc?
16:09.35Hmmhesayser.. astcc
16:09.46doughecka_<PROTECTED>
16:09.56bkw_is this a PRI?
16:10.11astoriabkw_: no, it's not
16:10.25ManxPowerdoughecka_: I don't know for sure if you can have a group=0
16:10.29astoriablitzrage: http://www.sysadminday.com/
16:10.30zoazttest is broken i think
16:10.33bkw_doughecka_, is it a PRI?
16:10.36gaffneycI'm dealing with Cisco phones but the version of cfgfmt will only work with glibc 2.1 or 2.0. Is there any place other than Cisco that I could get an updatE?
16:10.41bkw_if its  PRI don't do a w
16:10.48zoadont do what ?
16:10.50ManxPowerbkw_: doughecka_ is not running a PRI
16:10.52bkw_ah I see not PRI
16:10.58bkw_I seen astoria ask about d channel
16:10.59bkw_sorry
16:11.03zoabrian, dont do what ?
16:11.11bkw_zoa you punk
16:11.13zoaim writing a tutorial for e1/t1 cards
16:11.16zoai need all info i can get
16:11.16astoriabkw_: sorry, i didn't pay much attention and am cuasing all sorts of problems!
16:11.17zoa:)
16:11.35blitzrageastoria: thx!
16:12.06`SauronHAPPY SYS ADMIN DAY
16:12.18*** part/#asterisk collink (~collin@207.111.174.1)
16:13.46*** join/#asterisk Brumle (~brumle@brumle.com) [NETSPLIT VICTIM]
16:13.46*** join/#asterisk KryoStoffer (~kri@helium.kri.dk) [NETSPLIT VICTIM]
16:13.53*** join/#asterisk jeffgus (~jeffgus@216.86.199.4)
16:14.25*** join/#asterisk doughecka_ (~Miranda@doughecka.user)
16:14.30*** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com)
16:14.43doughecka_blah
16:14.45doughecka_ok
16:14.47doughecka_what did I miss?
16:14.49doughecka_:0
16:14.51doughecka_unstable internet
16:14.55doughecka_would dsl effect the adtran?
16:15.03ManxPower`Sauron: Does that mean I'll get a break from mobs of users waving torches and screaming "Kill the geek!"?
16:15.36ManxPowerdoughecka_: do you have a filter on the DSL line before it connects to the Adtran?
16:15.42doughecka_not sure
16:15.45doughecka_probebly not
16:15.48doughecka_let me see if I can find one
16:15.50ManxPowerdoughecka_: you need to
16:15.57doughecka_would that be causing some issues?
16:16.18doughecka_brb, let me find one
16:16.22doughecka_cause the dsl keeps dropping as well
16:16.25ManxPowerdoughecka_: is the line connected to the phantom ringing port?
16:16.34doughecka_possibly, but I need to make sure
16:16.39ManxPowerdoughecka_: no filter would make the DSL drop
16:16.59*** join/#asterisk Corydon-w (red@vcchgate.vcch01.springfield.tn.us.vcch.net)
16:17.06doughecka_wopuld the adtran causer the dsl to drop though
16:17.27Beirdoone would think not, but who knows
16:17.29ManxPowerdoughecka_: ANYTHING plugged into the line without a filter (except the DSL modem) can make the DSL service drop.
16:17.42doughecka_k
16:17.46doughecka_let me go find one
16:18.01BeirdoManxPower: not in the testing I did when designing DSL gear.  usually it just degrades a touch if anything
16:18.07jake1932has anyone found a reliable IAX hardphone?
16:18.16Beirdothe analog phones can go ape-shit though from the higher frequency stuff
16:18.20ManxPowerBeirdo: I said "can". 8-)
16:18.24Beirdoahh :)
16:18.25Beirdoheh
16:18.26Beirdoyeah
16:18.30*** join/#asterisk Brumle (~brumle@brumle.com) [NETSPLIT VICTIM]
16:18.30*** join/#asterisk KryoStoffer (~kri@helium.kri.dk) [NETSPLIT VICTIM]
16:18.38Beirdofilters are a good plan
16:18.41ManxPowerBeirdo: If I don't have a filter on my TiVo, for example, I'll get slow speeds and random drops
16:18.47Beirdonice.
16:18.50ManxPoweron the DSL
16:19.06Beirdothey gotta try real hard to bugger DSL
16:19.11*** join/#asterisk nettie (~nettie@213-140-17-96.fastres.net)
16:19.23Beirdoit's surprisingly resiliant
16:19.39ManxPowerBeirdo: I try not to tempt fate
16:19.41Beirdobetcha they are feeding high frequency crap out the line
16:19.48Beirdoyeah, better safe than sorry :)
16:19.59BeirdoI have one phone that REALLY doesn't like DSL filterless
16:20.19Beirdoit will ring sometimes...  with a real weird warble
16:20.26Beirdowhen nobody's calling
16:20.47Beirdoso I learned early to filter it anyways
16:21.02BeirdoHmm, I should go for lunch before I forget
16:21.16nettieHi guys, how's going? I'm trying to configure my analog to voip adapter (Cisco ATA186) with my current voip provider. We're using SIP. The problem is that I can't complete the registration process. The service works flawless using X-lite. Anyone have experience on the matter and a couple of minutes to hook me up please?
16:21.30Nuggetdealing with the phone company is like dealing with the government
16:23.11doughecka_ManxPower: dsl line is not one that causes the random calls through the adtran
16:24.14doughecka_they have a dsl filter on the fax machine
16:24.17doughecka_its the fax line
16:24.37doughecka_I can leave it unplugged as they have a fax machien and I am not using the fax features on the phone system
16:26.11harryvvnetti, give the symptoms
16:26.35nettieharryvv well it basically doesnt registers
16:26.43harryvvto the asterisk box?
16:26.51nettieI cant hear any ringtone
16:26.57harryvvdialtone
16:26.59*** join/#asterisk Brumle (~brumle@brumle.com) [NETSPLIT VICTIM]
16:26.59*** join/#asterisk KryoStoffer (~kri@helium.kri.dk) [NETSPLIT VICTIM]
16:27.03nettieyeah, the voip service provider is using asteris
16:27.06nettieasterisk
16:27.06harryvvnettie, whats the ip of the ata
16:27.19nettieit's in a private netowrk behind nat
16:27.20harryvvohh thats good who it that?
16:27.27*** join/#asterisk gniretar (~mark@198.173.197.15)
16:27.28mrmeis there a way to setup a skype trunk?
16:27.30harryvvand xlite did work behind nat?
16:27.37nettiesure, perfectly
16:27.40harryvvk
16:27.50harryvvthat would eliminated the fw as suspect
16:27.55gniretarhey, i'm having a wierd problem with the Asterisk CDR
16:28.00harryvvbrb
16:28.11nDuffmrme, no -- there's a bounty for adding support, but given that skype's protocol is proprietary, it's not an easy thing to do.
16:28.22gniretarcalls from sip phones are showing up with the correct extension in 'src' but zap pohnes dont
16:28.23nettiexlite registered perfectly, I wasnt able to receive calls but I opened a rang of udp ports in my fw and worked prefctly
16:28.34gniretaranyone had this problem before?
16:28.37nettieI think it's a nat traversal problem
16:28.43nettiewhich for some reason xlite doesnt hav
16:28.44nettiee
16:29.09nettieI also have a Planet VIP-150 POE SIP phone and it doesnt work
16:29.54harryvvagain, xlite eliminated the fw as suspect if you could make two way communications but out of curiosity, what fw is it and what ports opened up?
16:30.06*** join/#asterisk Qwell (~north@24-50-66-194.vnnyca.adelphia.net)
16:30.08nettiecisco IOS
16:30.19nettieopened udp from 1024 to 65535
16:30.30nettieof course I permitted only the sip server eheh
16:30.36ManxPowernetnameus: your router is a Cisco?
16:30.48nettieManxPower yu
16:30.50nettieyup
16:30.57gniretaranyone know anything about CDR?\
16:31.02harryvvhis ata is cisco
16:31.10nettieyeah the ata too
16:31.17harryvvohh, you have both
16:31.17ManxPowerno ip nat service sip udp port 5060
16:31.23ManxPoweryou want that on the cisco router
16:31.26harryvvso the fw is a cisco router?
16:31.28nettieManxPower I tried that
16:31.37nettiewith the sip phone .. didnt work
16:31.39ManxPowernettie: Is Asterisk behind NAT?
16:31.45nettieManxPower nope
16:31.51nettieManxPower but lemme retry
16:31.57nettiemaybe the ata likes that ;)
16:31.58nettiehangon
16:31.59harryvvnettie, I thought you said it was behind nat and that it worked
16:32.04ManxPowernettie: you are screwing everythnig up and making it more complicated.
16:32.06harryvvon xlite at least
16:32.19harryvvbrb
16:32.20nettieharryvv: the CLIENT is behind NAT
16:32.36nettiethe ASTERISK server is on the NET with a public ip
16:32.50ManxPowerIn the SIP ata do NOT enable any NAT settings.  In the Cisco do NOT portforward anything and set "no ip nat service sip udp port 5060" in asterisk set nat=yes in the sip.conf section for the phone.
16:33.16nettiewith CLIENT I mean: xlite, ata186, poe sip phone
16:33.17ManxPoweroh, and put qualify=yes in the sip.conf section for that device as well.
16:33.35gniretarnoone has any experience with Asterisk CDR?
16:33.44nettieManxPower: I'm pretty sure it's like that but I have no control of my ISP asterisk nat configuration.
16:33.48ManxPowerwhen the device registers with asterisk "sip show peers" should show the public ip address for the NATted device.
16:33.57harryvvmanx, he is the client and the isp uses asterisk. he is a remote user
16:34.12ManxPowerharryvv: then why isn't he asking his ITSP these things?
16:34.32harryvv:)
16:34.44ManxPowerIt's pretty usless to ask us.
16:35.35ManxPowerHell, maybe the ITSP is using SER or Cisco and that would invalidate anything we could tell nettie
16:35.47nettieguys, I was just asking if anyone had experience on the matter.
16:36.05ManxPowernettie: lots of experience, but I know how the asterisk part needs to be setup.
16:37.36ManxPowerBTW, setting  "no ip nat service sip udp port 5060" will make the Cisco NAT work like all the dumb home NAT routers
16:37.48nettieManxPower I definitely get this, the point is that I was looking for some feedback regarding possible NAT traversal device configuration.. considering the current setup works with a softphone but has problem with 2 hardware device which should definitely support NAT traversal
16:38.05ManxPowernettie: Well, I did give you that info
16:38.17nettieManxPower this is the actual configuration.
16:38.17odie_floconbuy a WRT54G and install asterisk on it.
16:38.25*** part/#asterisk gniretar (~mark@198.173.197.15)
16:38.26ManxPowerBut really, your ITSP is the place to ask
16:38.54nettieodie_flocon I already have one.. running openwrt ;)
16:39.25*** join/#asterisk JohnJacob (~JohnJacob@pcp0011543387pcs.mainf01.in.comcast.net)
16:39.26nettieManxPower you're definitely right..  thanx for your time.
16:39.47nettiehey harryvv thanx to you too ;)
16:41.57Hmmhesaysthe sounds in astcc are great, lol they don't match
16:46.52harryvvnettie, who is your tisp?
16:47.09nettiewww.messagenet.it
16:47.40*** join/#asterisk wrmem (~monnin@monnin-win.cso.uiuc.edu)
16:48.25harryvvcute female on there page. Italy interesting.
16:49.01nettiehehe
16:49.09harryvvfree fax?
16:49.19nettieuhmm
16:49.24nettieyeah could be
16:49.27nettiefree incoming
16:49.34nettiemax 10 pe rmonth I think
16:49.38nettienot 100% sure
16:50.02mrmei know its probably not too hard to find, but does anyone have broadvoice setup files or docs
16:51.10Nuggethttp://justfuckinggoogleit.com/?q=broadvoice+asterisk+configuration   :)
16:53.11mrme:P
16:55.48blitzragehrmmm, how much would you guys charge for the configuration of a fairly basic Asterisk system (simple IVR, MoH, Voicemail, DUNDi between two servers, provisioning of 20 devices). I'm thinking like 8-16 hours... all in.
16:56.03*** join/#asterisk jsaunders (jsaunders@S01060060971c5817.vs.shawcable.net)
16:56.20ManxPowerblitzrage: you sure are optimistic
16:56.23*** join/#asterisk essobi (kstone@75.137.26.216.host.teledvance.com)
16:56.25harryvvyour trying to estimate the cost to the customer?
16:56.33ManxPowerblitzrage: I'd charge about $2k
16:56.52blitzrageManxPower: this isn't for a business... its for a guy to setup a couple of boxes for his family...
16:57.07blitzrageManxPower: going to send ATAs or phones to people overseas
16:57.16ManxPowerblitzrage: Then I would charge about $5k
16:57.20blitzrageManxPower: LOL
16:57.35ManxPowerblitzrage: No joke.  Home users are horrible.
16:57.35*** join/#asterisk CoaxD (coax@shell1.cornernet.com)
16:57.40blitzrageManxPower: I hear that!
16:57.59coppiceif he wants you to put his family in boxes, i'd expect you to charge more :-)
16:58.03harryvvwe have large east indian famillies here. Unfortunaly thay rip everyone off just to drive there vipers ;)
16:58.40Hmmhesaysheh
16:58.41harryvv6-8 illegal bedroom houses, 20k in all and the city does not give a squat.
16:58.58Nuggetwhat is an "illegal bedroom house"?
16:59.01Hmmhesaysharryvv: where are you?
16:59.23essobiI got two sip peers.. both I'm setting G729 then G711 for the codec weights, then I do the same in my sip.conf peers.. but when I try to make a call.. It's negotiating 711 on one and g729 on the other and dropping the call cause I don't have any licenses..
16:59.33Qwellharryvv: illegal bedrooms?
16:59.33essobiAnyone have an idea why it's negotiating like that?
16:59.38*** join/#asterisk wizhippo (~wizhippo@Quebec-HSE-ppp237166.qc.sympatico.ca)
16:59.45harryvvsay its frame was designed for 6-8 or 5-7 and then its zones single residential 3-4. once the inspector leaves thay wall in the porch.
17:00.00ManxPoweressobi: Because Asterisk will always prefer G711 over any other codec.
17:00.09essobiManxPower ?
17:00.10Hmmhesayswe have a large influx of kurds here, who some to have a lot of money
17:00.18ManxPoweressobi: Because Asterisk will always prefer G711 over any other codec.
17:00.31essobiManxPower Why, when it's specified on both bridges as the codec preference order?
17:00.39harryvvthen the husband/wife kids, grandperents and a aunt or uncle may live there. I heard of one case 10 people living in one of these mega homes.
17:00.43essobiIt does negotiate it on one leg properly.. but not the other.
17:00.43ManxPoweressobi: No idea.
17:00.50*** join/#asterisk drooth (~drooth@ip68-111-235-172.sd.sd.cox.net)
17:00.53Hmmhesaysset the codec per phone
17:00.57ManxPoweressobi: one leg prolly allows ulaw and one leg doesn't.
17:01.02`Sauronessobi: You could just remove 711 (or 729)
17:01.12`Sauronand then it'll force the other :p
17:01.13essobiBoth allow 729 and 711.. and I need both to work.
17:01.14BuckRogershey im looking for sip termination middle volume about 100,000 mins a month any one in the business here
17:01.20ManxPoweror buy some damn g729 licenses and save yourself hundreds of dollars
17:01.34essobiI don't want to transcode on this machine.
17:01.45ManxPoweressobi: then don't use G729
17:02.03essobiwhy won't it just pass through then
17:02.05essobi:P
17:02.15ManxPoweressobi: CVS-HEAD's codec selection is supposed to be better then 1.0.x, but I still don't trust it to do the right thing.
17:02.21lters_ManxPower, is something up with the mailing lists?
17:02.32ManxPoweressobi: if you want passthru then don't allow ulaw.
17:02.43ManxPowerlters: Why would I know?
17:03.08MikeJ[Laptop]cuz!
17:03.10lters_just wondered because I am not getting msgs since yesterday
17:03.22MikeJ[Laptop]lters, I have.
17:03.59lters_hmm, I can see on the archive that there were a few, but am currious why I dont get them.
17:04.19BuckRogershey any word on why simpletelcom went off the air?
17:04.44blitzrageharryvv: yep, I lived in one of those - 6 people in one house, each paying $400 + utils
17:04.46file[laptop]SIP codec negotiation and asterisk is fun
17:05.04lters_can u use the setvar to get the codec set?
17:05.40blitzrageharryvv: actually, at one point it was 8 people - fire marshall found out and deemed the house unsafe
17:05.49harryvvohh
17:05.58harryvvdid it have foam for the ceiling?
17:06.00ManxPowerthe SIP_CODEC only works for OUTGOING calls from Asterisk
17:06.12ManxPowerby the time an incoming call hits the dialplan the codec has already been selected.
17:06.12blitzrageharryvv: had a hanging ceiling
17:06.24harryvvfalse ceiling
17:06.25harryvv;)
17:06.26lters_I see, thanks..
17:06.36blitzrageharryvv: aye
17:06.49blitzrageharryvv: did all the wiring himself, I think that was the problem
17:06.56lters_ManxPower, so it will not change codecs on the fly for a passing thru call
17:07.12harryvvblitzrage idiot!
17:07.16harryvvhe was a idiot
17:07.23blitzrageharryvv: yeppers
17:07.45blitzrageharryvv: not sure what happened to that house, I luckily found someone to sublet for 4 months so I got out of there
17:07.50BuckRogershey any word on why simpletelcom went off the air?
17:08.01harryvvI saw this wonderfull view home burned to the ground only because the onwer wanted to save a few bucks on the hotub electrical install and did it hom self :)
17:08.02blitzrageBuckRogers: didn't know the first time you asked, still don't know
17:08.25blitzrageharryvv: yah, people are dumb sometimes - and insurance probably didn't cover anything right?
17:08.33BuckRogersthanks for your response thier are others in the room
17:08.41harryvvwell, if I heard about it then its obvios :)
17:08.55blitzrageBuckRogers: well, you just asked like 3 mins ago
17:09.12BuckRogersdo you every take your eyes off of the screen
17:09.33blitzrageBuckRogers: nope, never, I'm here 24x7
17:09.45BuckRogersright on
17:10.15BuckRogersto my other question then is any one in the voip termination business here
17:10.24*** join/#asterisk Tili (~Tili@202-133-65-37-dialup.sat.net.pk)
17:11.21*** join/#asterisk wunderkin (kev@24.137.147.163)
17:11.26KattyBuckRogers: :>
17:11.41BuckRogerskatty
17:11.53BuckRogers;)
17:11.57*** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com)
17:11.58Kattyi liked buck rogers
17:12.00Kattyit was fun
17:12.05BuckRogersthe show
17:12.10BuckRogersim much better
17:12.19BuckRogersany how are you in the termination business
17:12.43Kattyi see
17:12.49twisted[asteria]Katty!
17:12.56KattyBuckRogers: way to be arrogant
17:13.01Kattytwisted!
17:13.04Kattytwisted josh (=
17:13.08twisted[asteria]haha
17:13.22harryvvwith the truck strike here, walmart is suing them :)
17:13.29harryvv1,000 truckers on strike :)
17:13.36BuckRogersi guess the truth may sound arrogant at times but tip toeing around it gets nothging done
17:13.37Kattyneat
17:13.38twisted[asteria]woo, two hugs
17:13.56KattyBuckRogers: you can be arrogant in front of mirror, kthx (=
17:14.16harryvvas far as im concerned, wall mart can go bankrupt ;l)
17:14.20BuckRogersyea or on the street or typing on them computer
17:14.27BuckRogerswhats your point
17:14.30Corydon-wHmmm
17:14.31Kattyharryvv: well that would sorta be bad too
17:14.35Hmmhesaysheh
17:14.37Kattyharryvv: think of all the jobs lost..
17:14.48Kattyharryvv: and all the people that would lose their health insurance
17:14.53harryvvkatty, mmm thay take jobs away from the mfg bussiness.
17:15.15Kattywalmart going bankrupt would be bad for our economy
17:15.21harryvvhahah yea right
17:15.29Kattyit would be
17:15.32ManxPowerKatty: it would be bad for MY economy
17:15.36droothwhat's up with nufone?  I have asked them to change my failto on my DID and it's just not happening, and they ignore emails/chats.  Geez....
17:15.45BuckRogersit would be devastating they are the largest employeer many towns are completely depent on them
17:15.46gordonjcpyou shouldn't have to have health insurance
17:16.02gordonjcpthe USA is the only 1st-world economy that doesn't have socialised healthcare
17:16.03wizhippocanada rules!!
17:16.05Kattygordonjcp: i think you're right. i wish everyone had free medical
17:16.12gordonjcpyou really need to get that fixed...
17:16.12BuckRogersyeah with good reason
17:16.26wizhippofor health insurance that is
17:16.37Kattygordonjcp: though all the money tied up in healthcare does tend to lend towards development of new and better things
17:16.48Hmmhesayslike TANG
17:16.51gordonjcpKatty: not really, it all goes to line shareholder's pockets
17:16.52harryvvgord, the health care system in this socialised goverment is not all what its cracked up to be...long waiting times even for critical care. Some have died waiting to long.
17:16.54Kattygordonjcp: there's profit to be had in new drugs :/
17:17.04BuckRogersyes harryvv
17:17.11BuckRogersit really is not all its cracked up to be
17:17.16gordonjcpharryvv: and in private-only healthcare, people have died because they couldn't afford treatment
17:17.17KattyHmmhesays: slurper
17:17.21*** join/#asterisk wunderkin (kev@12-215-218-160.client.mchsi.com)
17:17.24BuckRogersyou look at real stats we do have the best heathcare here
17:17.27BuckRogersin the usa
17:17.38Kattynow i want a slurpy
17:17.46BuckRogersnow when is the last time anyone was refused treatment here
17:17.47Kattygoshdangitanyhow Hmmhesays
17:17.52gordonjcpBuckRogers: but you pay disproportionaly more for it
17:17.54Kattyand dagnabbit too!
17:18.04Hmmhesayswhat now?
17:18.18BuckRogersshore you do but i havent been sick for three years
17:18.23harryvvgord, actually there are countries americans can go to get top care health care and its cheap! some where in south america saw it on tv. Go on vacation, have surgery that is top quality and cheap.
17:18.27BuckRogersi have a savings account incase i do
17:19.21BuckRogerswhy sould i have to pay into a tax pool for everyone elses health needs
17:19.40BuckRogersnot like the illegal imigrants aren't already draining that tax pool
17:19.51nighty-social security is bankrupt almost everwhere in europe
17:19.55nighty-social security is bankrupt almost everywhere in europe
17:19.56astoriatouche BuckRogers.. but this is probably not the place..
17:20.10BuckRogersyeah they have 12% unemployment
17:20.25nighty-it means mo' taxes for the tax payers too
17:20.30BuckRogerssocialism does not work its proven in the economic stats time and time agian
17:20.43nighty-BuckRogers: yep
17:20.55nighty-BuckRogers: but we are stuck with it
17:21.08nighty-BuckRogers: in France (thanks to those shithead at gov)
17:21.12gordonjcpharryvv: I mean, I can choose to get stuff done on the NHS or privately
17:21.25gordonjcpI use a private dentist but NHS for everything else
17:21.33BuckRogersand here in america we have our equilent problem with our tort system of law
17:21.42BuckRogerspeople sue everyone over every thing
17:21.48ManxPowerBuckRogers: Europeans that I've bet are SHOCKED by the high rate of crime, violence, poverty, and illiteracy in the USA
17:21.56nighty-BuckRogers: it's begining to happen in Europe too
17:22.01BuckRogershigh crime really
17:22.15harryvvbuck, its not going to be such a problem with class actions anymore. the limit is now 250k jury award max.
17:22.49harryvvyes, math scores are better over seas.
17:22.55nighty-BuckRogers: Europeans are shocked by everything they don't know , it's a common illness around here
17:22.58coppiceManxPower: yet any little terrorist incident or disease issue and the americans run away. weird.
17:23.00BuckRogerswell our education rating has been steadly increasing world wide, our nations crime level is at its lowest in over 20 years
17:23.11BuckRogershow could you say we havent improved dramitcally
17:23.25harryvvBuck, what country are you talking about?
17:23.46BuckRogersdude stop reading the ny times and get some real stats
17:24.00BuckRogersgo to the census.gov and check your history
17:24.10ManxPowerBuckRogers: Real stats: New Orleans has had over 300 murders since Jan 1 of this year.
17:24.13bkw_coppice, I don't ;)
17:24.17gordonjcpyour crime rates are still incredibly high though
17:24.21nighty-coppice: would'nt you run away ?
17:24.23bkw_I think its all bullshit out the media scares the shit out of people
17:24.31bkw_SARS.. BRING IT ON!!!
17:24.36BuckRogershaha
17:24.37bkw_TERRORISTS... BRING IT ON!!!
17:24.37Hmmhesayslol
17:24.40harryvvI have a Math Proffesor friend of mine who works at UW and he is worried at the constant decline of the level of first year students bring to his classroom.
17:24.43BuckRogersno no no
17:24.49*** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk)
17:24.50Hmmhesaysbad B.O. BRING IT ON
17:24.54ManxPowerOnly terrorists use the 'r' option to Dial
17:24.58nighty-:)
17:25.01coppicenighty-: why? the worse that.eu has ever been for terrorism is far safer than the us
17:25.06bkw_Hmmhesays, no I have a can that can take care of BAD B.O.
17:25.11blitzrageharryvv: same thing at Sheridan in the Telecom Program
17:25.13BuckRogersyeah well traditional college levels are down but tradeschools is up
17:25.19nighty-coppice: depends on where you are when it happens hey :)
17:25.20gordonjcpcoppice: you sure about that?
17:25.23blitzrageharryvv: apparently its really really bad for the next 2 years of students
17:25.33coppicedamn sure
17:25.35BuckRogerspeople are tired of having to take classes that are meaningless in the real world
17:25.37nighty-coppice: of course it will never happen to you :)
17:25.42harryvvI have a been trained in anti terrorist activities in the 80s. Its nothing new to me.
17:25.43nighty-coppice: it never does :)
17:25.48bkw_muslim in oklahoma.. you would notice it!
17:25.48Hmmhesaysbkw_: what a novel idea
17:26.03BuckRogersi was a international buinsess eco student, i had to take thearter appreciation
17:26.15bkw_BuckRogers, what fucking good is that?
17:26.17BuckRogersis bs and the market will dictate it
17:26.17nighty-bkw_: I bet you would not
17:26.31BuckRogersno good at all
17:26.31bkw_nighty-, i'm in southeastern oklahoma
17:26.32bkw_I would
17:26.38BuckRogersi learned shakesphere
17:26.52bkw_school in general is bullshit except for dentists and medical peeps
17:26.52nighty-bkw_: how do you know one is a muslim ?
17:26.53ManxPowerHell, the "homeless" guy that asked me for spare change in Antwerp spoke 2 languages.
17:27.04Nuggeteasy!  all brown people are muslim, right?  :)
17:27.05harryvvbkw, flat there ;) i was stationed in witchita falls tx
17:27.05nighty-bkw_: it is not written on his face
17:27.14bkw_nighty-, I woudln't know because I never leave the house
17:27.16ManxPowerMaybe more, he switched to English when he realized I didn't speak Dutch.
17:27.16Hmmhesaysoklahoma: I dislike that state
17:27.26BuckRogersfocused courses of study are the wave of the future not this moneymakeing porkbarrel traditional college system that exist today
17:27.38bkw_nighty-, but really you can tell if you pay any sort of attention
17:27.46harryvvBuck, yea true.
17:27.52bkw_BuckRogers, ya really
17:27.53astoriaBuckRogers: not as long as the ones from traditional colleges are still the ones doing the hiring :)
17:27.56nighty-bkw_: I don't think so
17:28.03harryvvchina is a serios threat..eating at the US economy.
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17:28.11BuckRogersits called startup companies they are kicking ass
17:28.15bkw_nighty-, you're saying you can't tell someones decent by how they look?
17:28.21BuckRogerssmallcap stocks
17:28.26nighty-bkw_: nope
17:28.26coppiceManxPower: I think in .nl it is an absolute requirement to repeat yourself in english when you get a blank expression in reponse to dutch :-)
17:28.35bkw_nighty-, have you ever been to oklahoma?
17:28.37ManxPowercoppice: 8-)
17:28.38gordonjcpcoppice: are you seriously saying that the EU is generally safer for terrorism than the US?
17:28.38nighty-bkw_: yep
17:28.45BuckRogersyeah right
17:28.45bkw_south eastern oklahoma?
17:28.51BuckRogersthe eu is falling apart
17:28.54nighty-bkw_: yep
17:28.56bkw_where?
17:28.57BuckRogers2000page constiution
17:28.59bkw_what town?
17:29.06harryvvTerrorism on a whole, is over blown. Lots of other ways to die then Terrorism.
17:29.07BuckRogerswhen will everyone every adgree on that
17:29.16bkw_harryvv, yes
17:29.23ManxPowerThe EU countries have stopped killing each other and started working togather.  They will become VERY powerful if they can keep doing that.
17:29.46harryvvTerrorism is like the lottery, just hope though you loose every time.
17:29.50HmmhesaysI onced dated a girl from oklahoma
17:29.54bkw_harryvv, and if we keep loosing rights like we are the terrorists will win
17:30.10bkw_brad pitt is from oklahoma
17:30.12nighty-bkw_: pittsburg ?
17:30.13BuckRogersthe eu counties need to look at why thier populations are indeclien
17:30.28bkw_nighty-, OMG you sure as hell would notice someone out of place in that tiny place
17:30.30BuckRogersits a historic sign of a failing society
17:30.41nighty-bkw_: out of place yes
17:30.49ManxPowerBuckRogers: as opposed to the USA where people on welfare have 10 kids?
17:30.58nighty-bkw_: but I would not know if the guy is a muslim or not
17:31.00ManxPowerYeah, we really do beat them in that respect.
17:31.02BuckRogershaha i hear man
17:31.02coppicewhy? educated people want few children. same happens everywhere
17:31.04nighty-bkw_: just by looking at him
17:31.08bkw_nighty-, thats true
17:31.14BuckRogersnot few childern but no children
17:31.29BuckRogers10 kids on welfare is a very large overstatement
17:31.33bkw_nighty-, I'm in McAlester
17:31.36BuckRogersmaybe in the 80's
17:31.41nighty-bkw_: have not been there
17:31.52nighty-bkw_: I have been to south carolina too
17:31.57nighty-bkw_: pretty poor
17:32.05nighty-bkw_: and pretty religious
17:32.09harryvvbkw, isnt there a DOD contracted bomb making facility there?
17:32.18ManxPowerBuckRogers: When was the last time you spent any time in Europe?
17:32.19BuckRogerswant to talk about well fair, the england bombing suspects have been on thier wellfare system for years
17:32.22coppicethe germans will be more or less extinct in a couple of generations. the only way to stop that would appear to be to close all the schools and plunge them into destitution :-)
17:32.53harryvvdestruction
17:33.40BuckRogersthe eu needs to get fired up, bring back a sense of nationalism, and start showing thier highly compeditive teeth agian
17:33.40nighty-BuckRogers: nah
17:33.41ManxPowerBuckRogers: You are a complete lunatic.
17:33.47harryvvheheh
17:33.48nighty-BuckRogers: that's dangerous
17:33.51ManxPowerNationalism is Europe's PROBLEM.
17:33.59ManxPowerNot their solution.
17:34.01nighty-BuckRogers: we've been there once
17:34.02coppicenationalism in .eu usually means everyone attacks the french
17:34.03BuckRogerslets face it people no matter how PC everyone tries to be
17:34.05nighty-BuckRogers: remember
17:34.26BuckRogerswe are in a compeditive world
17:34.32ManxPowerBuckRogers: When was the last time you spent any time in Europe?
17:34.33Ariel_There is no perfect place. Every country has there own problems. Until we get ride of the UN and start talking correctly and evenly to all it's still going to be a mess.
17:34.41BuckRogershell yeah ariel
17:34.44BuckRogersscrew the un
17:34.48harryvvI dont know about you all, but as a 6 year old asked mom why this item was made in japen..some kind of vase. I said "Isnt
17:34.57harryvvAmerica good enough to make it?"
17:35.17harryvvThose were the evil nixen years
17:35.22harryvvnixin
17:35.23Ariel_harryvv, it's cheaper there due to they pay there people shit to make it.
17:35.24NuggetWhen I'm elected king I will abolish religion and nationalism.  Nothing but trouble.
17:35.26BuckRogersun officals need to start showing their books (finacial spending)
17:35.28nighty-harryvv: never asked these kind of questions to my mum
17:35.33harryvvAriel_ I know
17:35.39BuckRogersno need to abolish religion
17:35.51BuckRogersjust the extremist
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17:36.00Nuggetit's all extreme.
17:36.08coppiceAriel_: they kept saying that about japan, even as its labour rates went well above the eu and us
17:36.12ManxPowerBuckRogers: You're right about THAT, at least.  Religion, when properly used, is one of the best ways of controlling the populace
17:36.24BuckRogersmost of europes and american laws are based off of biblical ideologies most of whitch are wholsome values
17:36.38BuckRogersreligion is a guidline to life and the treatement of others
17:36.45harryvvIm not picking on japan but the fact that alot of our goods are now made over seas..thats a massive tax base loss to the united states.
17:36.50nighty-BuckRogers: isn't that true of all occidental countries ?
17:36.52ManxPowerBuckRogers: only for people that can't think for themselves
17:37.01jake1932is anyone reliably using VOIP termination for SOHO as the primary line?
17:37.09nighty-BuckRogers: now if you are talking about state and religion separation , that's another matter
17:37.10BuckRogerswhy would you read phylosfy then
17:37.35harryvvIm awake on 3 hours of sleap. time to head off to the pool.
17:37.37Ariel_as you see there are allot of the goods that were made in Japan now made else where. Like cars look at the Honda they actually make more in the states and export some back.
17:37.52Nuggetyour claim that western law is based on religious principles is simply not true.
17:38.06harryvvAriel_ I dont have much of a problem like that.
17:38.13jake1932or is everyone using either analog or PRI?
17:38.16harryvvPolycom are made where?
17:38.27harryvvThailand
17:38.46Ariel_religious does play a part of everything. Even if we don't want it to. It's just part of the over all mix.
17:39.06coppicea lot of stuff is being made in thailand now, including lots of hondas
17:39.07harryvvFluke electronics meters are made where? Everett washington :) yea pricy but good quality.
17:39.07*** join/#asterisk cire` (~e@adsl-215-65-134.mia.bellsouth.net)
17:40.00jake1932just want to see if it makes sense to colocate a server next to a VOIP provider
17:40.01nighty-harryvv: you should not complain , US have all electronic makers
17:40.05coppiceits a long time since I saw a fluke instrument made in the US
17:40.09nighty-harryvv: France as almost ZERO :)
17:40.10harryvvI worked at fluke, amazing to see how there meters are made. The automated test lead machine cost a million dollars just to make miles of test lead for the meters.
17:40.30nighty-harryvv: we make trains (TGV) we sell to noone
17:40.31harryvvcoppice, long time since when?
17:40.39nighty-harryvv: planes (almost the same)
17:40.55nighty-harryvv: and Weapons that don't work
17:40.57coppicewell, probably since I started living in asia :-\
17:41.24coppicenighty-: framce exports a lot of trains
17:41.30harryvvHeard the TGV is a good train. Thay use turbines?
17:41.33nighty-harryvv: and we have an national telco that tries to screw everyone (FT)
17:41.53nighty-harryvv: they maybe good trains, but did US buy any from us ?
17:42.04nighty-harryvv: or any other countries ?
17:42.09harryvvAmtrak has the Talgo
17:42.14coppicemost of asias metro systems have french trains
17:42.29nighty-coppice: metro :)
17:42.32nighty-coppice: yeah
17:42.34Hmmhesaysinteresting, mediatrix has a new wifi fxs box coming out
17:42.40nighty-coppice: cool at least we sold these :)
17:42.50*** join/#asterisk wizhippo (~wizhippo@Quebec-HSE-ppp237166.qc.sympatico.ca)
17:42.53Nuggetto make a truly glib argument... pick your preferred ten of the eleven commandments and you'll find that only two of them are illegal.  If that's truly the cornerstone of us and european law, we sure have done a terrible job of it.
17:43.06ManxPowerHmmhesays: I wonder if they will be as hard to configure and lack the support of their existing products?
17:43.18coppicenighty-: you have something against metro systems?
17:43.19nighty-coppice: France problem is the ENA ( National School of Administration :))
17:43.25HmmhesaysManxPower: no, web based, and you need to know who to talk to at mediatrix
17:43.34Hmmhesaystac blows
17:43.45nighty-coppice: it is SHITHEAD factory, unfortunately , they all go to gov after this :(
17:43.55Hmmhesaysif you want answers from them you have to go to the engineers
17:43.58harryvv2102 - Residential VoIP Access Device has T.38 support.
17:44.13HmmhesaysI actually like the 2102 it is a nice solid device
17:44.23coppicenighty-: well the trains seem pretty good
17:44.28Hmmhesayssnmp config aside
17:44.32*** part/#asterisk netnameus (~netnameus@pcp05000344pcs.shrpsr01.tn.comcast.net)
17:44.50HmmhesaysI auto provison them anyway so it doesn't matter
17:44.54ManxPowerThe Thalys I was on was not all that impressive, but it was FAST for the non-urban parts of the trip.
17:44.56Hmmhesays*provision
17:45.02nighty-harryvv: can you explain , what is T.38 (I know this is for FAX) but does it go over g711 ?
17:45.12nighty-harryvv: or it has nothing to do with it ?
17:45.20Hmmhesaysgoogle nighty-
17:45.31Hmmhesaysyou will find more information about t.38 than you ever wanted to know
17:45.39nighty-Hmmhesays: if I wanted to go look in google I would have done so
17:45.45coppicenighty-: T.38 is how you avoid G.711
17:45.47harryvvnighty dont know its been discussed in here.
17:45.58Hmmhesayshe does now
17:46.04nighty-coppice: oh ok
17:46.09*** part/#asterisk cire` (~e@adsl-215-65-134.mia.bellsouth.net)
17:46.13Hmmhesays~t38
17:46.13jbott38 is, like, see http://www.brooktrout.com/whitepapers/pdf/fax_over_ip.pdf for a decent overview of how it all works
17:47.02coppicethat's pretty salesy, but it give yout the idea
17:47.03nighty-coppice: but in asterisk T.38 can be handled ?
17:47.05file[laptop]tries to make faxing work half decently
17:47.11nighty-coppice: I mean easily
17:47.16coppicenighty-: not yet
17:47.28*** join/#asterisk eric- (~e@adsl-215-65-134.mia.bellsouth.net)
17:47.29ManxPowernighty-: Asterisk does not support T.38
17:47.31nighty-coppice: so what do people do ?
17:47.42ManxPowernighty-: they do not use VoIP for Fax.
17:47.50Hmmhesaysyou can't use t.38 on a reinvite?
17:48.00file[laptop]dejavu
17:48.01ManxPowernighty-: or they use ulaw/alaw and pray it works most of the time.
17:48.05nighty-ManxPower: ok , g711 seemed like a decent idea
17:48.07Hmmhesays<chuckle>
17:48.14nighty-ManxPower: I guess it is not that reliable then
17:48.21ManxPowernighty-: not when you have latency and jitter like you have on VoIP.
17:48.24`SauronManx: organized religion is for people who can't/won't think. Faith is a different story.
17:48.29`SauronShrug, the end. :)
17:49.07coppicenightly-: try http://www.soft-switch.org/foip.html for some notes on why G.711 fails
17:49.08nighty-ManxPower: could T.38 be handled by Hylafax (I think I read about this somewhere) ?
17:49.27ManxPower`Sauron: I say that anyone that believes in God does so because they don't want to think.
17:49.36ManxPowernighty-: that would be outside of Asterisk
17:49.47coppicethey is a t.38 to hylafax interface which is a part of open h.323
17:50.04harryvvdoes ser use t.38?
17:50.19*** join/#asterisk mutilator (~animenodv@65.111.201.79)
17:50.31coppiceanyone know what that announcement about t.38 on the mailing list is really about?
17:51.47Hmmhesaysyeah its an advertisement
17:51.55Hmmhesaysas far as I can tell
17:52.06coppiceobviously, but what is really advertising?
17:52.13jake1932anyone using Nufone's services for business?
17:52.40Hmmhesaysnow that is a good question coppice
17:53.06jake1932or any other VOIP provider for business?
17:53.07ManxPowerjake1932: I would never, ever trust the internet to carry business calls.
17:53.25jake1932ManxPower: even if you were 1-2 hops away?
17:53.40ManxPowerjake1932: even then
17:53.53tzangerjake1932: I am
17:53.56jake1932ManxPower:so either PRI or analog?
17:54.03tzangerManxPower: why not?
17:54.05ManxPowerjake1932: PRI
17:54.21ManxPowertzanger: because I don't trust the internet.
17:54.28jake1932ManxPower: but what about eh small businesses?
17:54.46ManxPowerjake1932: what we plan to do is ise VoIP for overflow when our PRIs are full.
17:55.14jake1932tzanger: any problems at all with call quality?
17:55.16greg_workthat's how i use voip (for business)
17:55.28tzangerManxPower: from a security standpoint it's really not that much different.  PRI ain't that difficult to tap
17:55.35tzangerjake1932: not with nufone
17:55.49greg_workthough, we just have 3 POTS.. but if all 3 are busy, it calls out on Voip, and calls get forwarded to a voip DID
17:55.50ManxPowertzanger: I'm more concerened about reliablity
17:55.53nighty-how many simulataneous g729 channel can a P4 3.2Ghz handle (roughly) ?
17:56.02nighty-single proc
17:56.11jake1932tzanger: how close are you to nufone (inrouter  hops)?
17:56.12DarthCluejake1932: asterlink employees use asterlink voip for business.
17:56.38nighty-g729 is CPU hungry it seems
17:56.47tzangerManxPower: ok you have a good point there but again that kind of thing can be worked around...  every call in and out o fthis office has gone over a 1-hop SDSL loop (IAX2) since June 2004
17:56.52tzangerfaxes too
17:57.01greg_workjake1932: i had problems with quality with livevoip
17:57.01*** join/#asterisk MattH (MattH@63.174.244.175)
17:57.09greg_workbut then they went out of business...
17:57.17jake1932much more of a problem then :)
17:57.21tzangerjake1932:
17:57.27ManxPowertzanger: It's not really going over "the internet" if it's only 1 hop, now is it?
17:57.31MattHHi... can anyone point me in the direction for this answer?   I'm trying to figure out how to set the receive jitter buffer on asterisk.. where do I tell it how much to buffer from what it is receiving from a user?
17:57.38tzangerjake1932: 11 hops
17:57.47tzangerManxPower: well we do termination to nufone and asterlink too
17:57.50mutilatorwell
17:57.58ManxPowerMattH: what protocol?  what version of asterisk?
17:58.01jake1932tzanger: 11 hops and no call quality issues?
17:58.05tzangerManxPower: but yes every call does go over an ethernet network :)
17:58.06mutilatorwhat if he had an ethernet cord laying on top of his dsl modem
17:58.16mutilatorthen it'de be going "over" the internet
17:58.22tzangerjake1932: nope nothing I've really noticed or heard complaint about
17:58.32ManxPowertzanger: I send calls over IP all the time, just internal LAN and WAN that we control
17:58.38Nuggetif you're going to play linguistic semantics, you can start by not calling a cable a "cord"
17:58.39tzangerManxPower: gotcha
17:58.54jake1932tzanger: did Nufone let you port over a toll free number?
17:58.55MattHManxPower: SIP 1.0.7
17:59.10ManxPowerMattH: 1.0.x does not have a jitter buffer for SIP.
17:59.20MattHahh
17:59.22tzangerjake1932: I don't do inbound wit them, only termination
17:59.42MattHManxPower: so it doesn't buffer at all eh?   what version DOES have jitter buffer?
17:59.56ManxPowerMattH: the developement version of Asterisk -- CVS-HEAD
18:00.09jake1932tzanger: ok - one more q - what is your ping time from * to nufone?
18:00.29nighty-anyone is using CIRPAK ?
18:00.38ManxPowerI don't use NuFone much anymore, but I still use it for a few things, personal stuff
18:00.49tzangerjake1932: between 50-75ms
18:00.58MattHI'm just trying to figure out an issue and thought that might be it... outgoing (to the sip client) sounds fine.. and inbound seems to breakup.. yet ping times are rock solid between 40 and 50ms
18:00.59*** join/#asterisk Bile_One (~bile_one@pcp03281999pcs.gillst01.ar.comcast.net)
18:00.59ManxPowerI've micrated away from them because of their lack of DIDs
18:01.14ManxPowerMattH: that's a TEN ms jitter.
18:01.32MattHVoIP should be able to handle that
18:01.35jake1932ManxPower: what would you recommend for small businesses that want to use *?
18:01.50MattHbut of course.. there is no jitter buffer
18:02.00*** join/#asterisk Brumle (~brumle@brumle.com) [NETSPLIT VICTIM]
18:02.00*** join/#asterisk KryoStoffer (~kri@helium.kri.dk) [NETSPLIT VICTIM]
18:02.00hardwireMattH: I have a 500ms jitter
18:02.07jake1932ManxPower: a PRI is overkill for many small businesses
18:02.08MattHhardwire: and asterisk works ok for you?
18:02.09ManxPowerjake1932: A Channel bank w/FXO and FXS ports, a Digium T-1 card, and Polycom IP 500s
18:02.11Bile_OneAnyone know what the contract agreement cost is for Cisco, to be able to download the firmware etc...
18:02.17Bile_Onefor a 7960.
18:02.22jake1932ManxPower: analog
18:02.24hardwireMattH: for the most part
18:02.25ManxPowerjake1932: I don't trust the TDM400P anymore.
18:02.30QwellBile_One: about $100?
18:02.30hardwireMattH: whats your issue?
18:02.35ManxPowerjake1932: Channel bank w/fxo will convert analog to T-1
18:02.45tzangerManxPower: have you tried recently?  They've fixed several driver issues including the "stops answering after 25days" one
18:02.47jake1932any particular channel bank?
18:02.48Bile_OneQwell, is that for the year, ot life time?
18:02.49*** join/#asterisk PakiPenguin (~uppal@202.147.163.177)
18:02.53PakiPenguinhello everyone
18:02.54Qwellgot me
18:03.05MattHhardwire: inbound audio (to the sip phone) sounds crystal clear... outbound audio (to the asterisk server and ultimately to the PSTN) gets choppy sometimes...
18:03.07tzangerjake1932: I do NOT recommend Access Bank 1 or 2 for FXO.  FXS is fine though.  ABI/2 do not support CPD
18:03.08ManxPowertzanger: All the fixes seem to be for FXO
18:03.28nighty-ManxPower: BRI cards are no good ? :)
18:03.44ManxPowerI recommend Adtran Total Access.  Under $400 from ebay for FXS ports, slightly more for FXOandFXS
18:03.45jake1932nighty-: i'll answer that for you in a few weeks
18:03.54nighty-jake1932: :)
18:03.57ManxPowernighty-: if you are not in the USA or Canada you could use BRI
18:04.02tzangerManxPower: hmm
18:04.11Qwell~cpd
18:04.12tzangerManxPower: I just don't have issue with FXS
18:04.18nighty-ManxPower: oh I see ISDN is not popular in US ?
18:04.18jake1932nighty-: as for the AVM Fritz it gets a big thumbs down for USA use
18:04.35nighty-jake1932: well it is german
18:04.40ManxPowertzanger: on 2 out of the 6 system we have FXS ports on we don't have problems.
18:04.46nighty-jake1932: so mostly for euroisdn
18:04.47tzangerahh
18:04.52jake1932nighty-: exactly
18:04.54nighty-jake1932: dss1 type
18:05.02ManxPowerthe others have to be rebooted every month or two
18:05.02*** join/#asterisk doughecka_ (~Miranda@doughecka.user)
18:05.03tzangerManxPower: anything in common with the 4 that don't work that isn't in common wit hthe 2 that do?
18:05.04nighty-jake1932: or vn4
18:05.08jake1932nighty-: hope i have better luck with the eicon diva server
18:05.14ManxPowertzanger: nope.
18:05.17nighty-jake1932: this is french :)
18:05.27doughecka_or rather, analog lines
18:05.29nighty-jake1932: at least was :)
18:05.40rikstahas anyone got a very basic AGI script that i can use as an example of handling DTMF
18:05.45rikstapls
18:06.33doughecka_ManxPower: I am having wierd issues, one line wont hang up, I pick it up and get a fast busy. another line I get lots of flutter and stuff, but the line itself sounds fine... but I get the noise on any channel I put it on...
18:07.02ManxPowerdoughecka_: Adtran is usually good.
18:07.05jake1932riksta:  WAIT FOR DIGIT 5000
18:07.10ManxPowerbut you did get like the cheapest model of adtran
18:07.18doughecka_ManxPower: true, but still...
18:07.20doughecka_adtran is adtran
18:07.30jake1932riksta: that's very basic
18:07.31doughecka_what good is a product to a vendor if its crap
18:07.31rikstajake1932: can you capture say "1234" not just cut off at 1?
18:07.39doughecka_ManxPower: it WAS working this morning
18:07.39ManxPowerdoughecka_: did you buy it new or used?
18:07.43doughecka_brand new
18:07.59doughecka_brand bkw_ spanking new
18:07.59ManxPowerdoughecka_: then it prolly doesn't have any weird non-default settings.
18:08.07doughecka_I also reset to factory defaults
18:08.30ManxPowerdoughecka_: Let me give you some patches we use for Adtrans
18:08.35doughecka_ok
18:08.41jake1932riksta: exec READ
18:08.44doughecka_what do they fix?
18:08.57rikstajake1932: what im after is an example config etc that i can look at
18:09.00PakiPenguinweird issues doughecka_
18:09.01PakiPenguin:p
18:09.02PakiPenguinhehe
18:09.03rikstanot just a few commands thrown about
18:09.05jake1932riksta: don't know the exact syntax - but that should get you on the right track
18:09.06doughecka_I hope so :)
18:09.10doughecka_cause this is wierd
18:09.22Bile_OneAnyone know where I can get the firmware updates for a 7960?
18:09.29doughecka_and I keep getting the fake calls in
18:09.30PakiPenguincisco!
18:09.43PakiPenguindoughecka_, sounds like fun!
18:09.45PakiPenguin:p
18:09.48Bile_OneNo S,,, PakiPenguin.
18:09.51doughecka_not really
18:09.54ManxPowerdoughecka_: 3 patches http://pastebin.ca/18792
18:10.02PakiPenguinBile_One, yup :p
18:10.03*** part/#asterisk brookshire (~matt@207.111.174.1)
18:10.05ManxPowerdoughecka_: what signaling did you say you use?
18:10.07jake1932riksta: check this page out: http://64.233.161.104/search?q=cache:KtoqwaLxy6oJ:www.voip-info.org/tiki-index.php%3Fpage%3DAsterisk%2BAGI%2Bphp+agi+%22exec+READ%22+asterisk&hl=en
18:10.16[TK]D-FenderQuestion that is strangely Asterisk related : Anybody know around how much a solution from Sylantro or Broadsoft goes for these days?  My head office is looking at Avaya, and I want them on commodity SIP gear as we are going *.
18:10.17rikstatnx
18:11.04[TK]D-FenderAs Broadsoft & Sylantro are very strong SIP-wise I could integrate them with * fairly easily I expect and maybe even create an * application server for them.
18:12.25*** join/#asterisk sedwards50 (~chatzilla@adsl-67-125-150-70.dsl.irvnca.pacbell.net)
18:12.54ilTiziohas anyone got problem with * in debian sid?
18:13.09hardwireI use 1.0.9.dfsg
18:13.14doughecka_ManxPower: where do I split the patches?
18:13.24doughecka_where does patch 2 end?
18:13.37sedwards50I need help configuring a Adtran 750 talking to a t100p -- the fxo lines won't hang up!
18:13.55fockswith the directory, is there a way to have it match spelling with 3 letters of the last name as it does, but then play recorded names
18:13.56ManxPowerdoughecka_: easiest thing to do is just change the 3 or so lines manually
18:14.19ilTiziohardwire: any trouble with voip service provider who keep disconnectiong and get you connect again only stopping and starting * again after some minute?
18:14.33hardwirethats not a deb issue.. flr sure
18:15.04ManxPowerdoughecka_: patches start where filename is.
18:15.08ManxPower*** zaptel/zaptel.h.orig
18:15.16MattHquestion on asterisk... how easy is it to upgrade from 1.0.7 to CVS-HEAD to be able to get sip jitter buffer on server side?
18:15.24MattHie.. will things break or should it be a seamless upgrade?
18:15.25doughecka_ah
18:15.44NuggetMostly painless, Matt.  You'll get warnings until you change some things in your dialplan, but it's not bad.
18:15.49hardwireanybody else here depressed for no good reason
18:15.49hardwire?
18:15.50ManxPowerMath`: README.upgrade or UPGRADE.txt or something like that is included in the CVS-HEAD source
18:15.58NuggetJust be cautious, HEAD isn't always stable.
18:16.04ManxPowerhardwire: I
18:16.12ManxPowerhardwire: I'm depressed but I have reasons 8-)
18:16.16hardwireI think I do to
18:16.19hardwiremainly my job
18:16.27*** join/#asterisk brookshire (~matt@207.111.174.1)
18:16.33hardwireand a lot of it is me always being an angry little fuck
18:16.49hardwireand for a guy that pets kitties as much as I do.. I shouldn't be depressed
18:16.49MattHManxPower: is it possible to download the CVS-HEAD in a tarball rather then getting it via CVS?
18:17.05NuggetIf you are intimidated by using CVS, you probably shouldn't run HEAD.
18:17.11*** join/#asterisk Brumle (~brumle@brumle.com) [NETSPLIT VICTIM]
18:17.11*** join/#asterisk KryoStoffer (~kri@helium.kri.dk) [NETSPLIT VICTIM]
18:17.17drraycvs is easy
18:17.28MattHNot intimidated... I've used it... just prefer tar balls :)
18:17.38drraythe digium instructions are 100% simple
18:17.38ManxPowerMattH: no.  CVS-HEAD is not a release.
18:17.57MattHManxPower: right... realize that.. just didn't know if they made nightly tarballs of it or anything
18:17.59hardwirehoping to run into the teliax guru today
18:18.00hardwirehe is in town
18:18.02ManxPowerit's the developement version of Asterisk, meaning it may or may not work, depending on when you download it
18:18.03hardwirehe needs beers
18:18.32Nuggetand "HEAD" is a moving target, changing many times an hour on busy days.
18:18.50Bile_OneI need whiskey, and Rum, and Port wine, and beers.
18:18.56blitzrageNugget: no shit - try documenting that!
18:19.00coppicesounds like production apple ][s :-)
18:19.03drrayand beer for my horses
18:19.44hardwireyou have beer horses?
18:20.27Bile_Onebearded horses?
18:20.37hardwireyay for digitally imported - Trance
18:20.43hardwireits going to make me happier really damn soon
18:20.43*** join/#asterisk pa (~Paolo@pa.user)
18:21.07blitzrageexport CVSROOT=:pserver:anoncvs:anoncvs@cvs.digium.com:/usr/cvsroot ; cvs login ; cvs co asterisk zaptel libpri asterisk-sounds
18:21.34blitzrageheck, its easier than tarballs!
18:21.38*** join/#asterisk pbxbart (user@p54B01C7C.dip0.t-ipconnect.de)
18:22.06lathos42blitzrage: Not as easy as installing on Gentoo..  "emerge asterisk" :)
18:22.17hardwirehttp://www.logicsupply.com/
18:22.20hardwireanybody here use these guys
18:22.20DarthClueCVS-HEAD, the easy way : http://www.darthclue.org/categories/3-Chalkboard-Examples
18:22.29lathos42blitzrage: Granted, you can't get CVS-HEAD with Gentoo, but... :)
18:22.33hardwireI am thinking of not ordering via mini-itx.com any more
18:22.39blitzragelathos42: exactly!
18:22.46jake1932just had a thought - if i have a provider that does both origination and termination and I use SIP, could I essentially use asterisk as a call director - but keep it out of the voice loop?
18:23.23blitzragejake1932: you mean your phone behind Asterisk and that attached to your VSP? Of course
18:23.38blitzrage1.0.7? 1.0.9 is out :)
18:24.01jake1932blitzrage: having all the phones on the TDM
18:24.42blitzragejake1932: huh?
18:24.51jake1932blitzrage: assuming asterisk would only be doing call control - but the call would stay inside the VOIP provider
18:25.11*** join/#asterisk fugitivo (~ajf@201.255.100.39)
18:25.12Ariel_jake1932, how are you going to do that TDM needs asterisk to be able to connect to a sip account. Unless you use canreinvite=yes sip accounts that use the same codec as your provider can do that.
18:25.13fugitivohello
18:25.20blitzragejake1932: I think I'm confused as to what you're doing
18:25.33blitzrageor trying to do...
18:25.41blitzrageif its TDM, then Asterisk has to handle the calls...
18:26.08jake1932blitzrage: hadle the call - but from what i understand SIP will allow 2 endpoints to connect directly
18:26.29jake1932if the origination and termination is a VOIP provider - shouldn't it work?
18:26.34blitzragejake1932: it can... depending on configuration
18:26.54jake1932ok - canreinvite seems to be the question I guess
18:26.54blitzragejake1932: signalling would still be going through Asterisk - media _could_ go between end points directly
18:27.00Ariel_jake1932, see my message yes it works but you have to use same codec and canreinvite=yes in the sip.conf
18:27.01jake1932blitzrage: exactly
18:27.08blitzragejake1932: canreinvite=yes
18:27.34jake1932blitzrage: then this might work :)
18:27.38blitzrageand not use 'r' or 't' or 'T' (and probably 'w' and 'W') in the Dial() app
18:27.47jake1932thanks blitzrage and Ariel
18:28.13blitzragew00t, laundry done
18:28.13ManxPoweronly terrorists use the 'r' option to Dial
18:28.19blitzrageManxPower: LOL
18:28.24tzangerManxPower: unfortunately I have to use it
18:28.26fockswith the directory, is there a way to have it match spelling with 3 letters of the last name as it does, but then play the voice recordings of the corresponding names rather than spell them out?
18:28.34blitzrageManxPower: sometimes you need it though
18:28.35tzangersomething changed in HEAD and I get no ringback in some circumstances
18:28.47blitzragetzanger: really?!  me too! And Jim too (even with 'r')
18:28.50hardwirehmm
18:28.53tzangerevn with 'r' ??
18:28.56ManxPowerfocks: If the user "record name" in their voicemail then it will play that
18:28.59blitzragetzanger: even with 'r' on both sides
18:29.03Ariel_focks, it plays the recording if you recorded it.
18:29.04tzangernow that's odd
18:29.05hardwireso I am going to have a few extensions that have to be routed via an iax2 server
18:29.09focksManxPower, by default?
18:29.13hardwireback to the main pbx
18:29.15hardwirewhere the voicemail is
18:29.17ManxPowertzanger: Make sure you have an indications.conf
18:29.20hardwirehow would mwi workk in that situation?
18:29.21ManxPowerfocks: yes.
18:29.24focksthanks
18:29.26*** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com)
18:29.28tzangerManxPower: oh I do
18:29.39blitzragetzanger: yeppers. Gotta find out why... wonder if its because theres no 182 Ringing being sent or something
18:29.40hardwireI don't think it would
18:30.10blitzrageManxPower: but that wouldn't matter when its all SIP would it?
18:30.14doughecka_crap
18:30.22*** join/#asterisk Brumle (~brumle@brumle.com) [NETSPLIT VICTIM]
18:30.22*** join/#asterisk KryoStoffer (~kri@helium.kri.dk) [NETSPLIT VICTIM]
18:30.33blitzrageSIP <--> Asterisk <-- IAX --> Asterisk <--> SIP
18:30.44blitzrage^^^ no ringing (even with 'r')
18:30.47ManxPowerblitzrage: indications.conf comes into play if you need to hear ringing AFTER the line is answered.
18:31.05ManxPower"r" almost NEVER solves a "no ringback" problem.
18:31.06blitzrageManxPower: even in a pure VoIP scenario as above?
18:31.12ManxPowerblitzrage: Yes.
18:31.19blitzrageManxPower: hrmmm... will need to check that then
18:31.27ManxPowerblitzrage: I could not get ringback with a simple SIP->Asterisk->Dial
18:31.30doughecka_ManxPower: now I have my old clicks and pops again...
18:31.34QwellManxPower: IAX2/guest@24.50.66.194/s  I can't do that with IAX unless I use 'r'
18:31.47ManxPowerdoughecka_: did you change the 1 to a 0 on the span
18:31.52doughecka_yea
18:31.57doughecka_I am doing a make clean
18:31.59ManxPowerQwell: It works for me just fine
18:32.10doughecka_ManxPower: of course echo is still there :)
18:32.11QwellManxPower: call it, you'll see what I mean
18:32.33ManxPowerQwell: I would have to see exten => s,1,whatever to be sure.
18:32.50QwellManxPower: I do an explicit playtons
18:32.52Qwellplaytones
18:33.35ManxPowerQwell: try a Wait(5) and see if you get ringing
18:33.46doughecka_hmm
18:33.47blitzrageManxPower: hrmmm.. might be right :)
18:33.49blitzrageManxPower: Jul 29 14:31:32 WARNING[30107]: channel.c:1617 ast_indicate: Unable to handle indication 3
18:33.50Qwellits not ringing.  I do a custom playtones
18:33.52doughecka_thats a little better
18:34.00ManxPowerblitzrage: Yup!  you don't have indications.conf
18:34.17blitzrageManxPower: I do now - I copied it over and did a restart now (before I placed that call)
18:34.59ManxPowerQwell: It's kind of pointless to have me test something if you don't have it configured the way it should work.
18:35.03blitzrageManxPower: what is the '3' referring to?
18:35.13ManxPowerblitzrage: 3 is ringback
18:35.18blitzrageManxPower: ahhhh, I see
18:35.25QwellManxPower: wasn't to have you "test" it.  Just to listen to the reason I use 'r'
18:35.26blitzrageManxPower: wonder why it can't handle it...
18:35.40QwellIf I wanted standard ringing, then no, you're right, I wouldn't need it
18:35.42ManxPowerit doesn't know how to indicated it because there was no indications.conf and the call was answered, so asterisk needs to send the indication inband and it uses indications for that 8-)
18:36.03*** join/#asterisk doughecka (~Miranda@doughecka.user)
18:36.07dougheckameh
18:36.08ManxPowerQwell: Um, "r" will give you standard ringing.
18:36.21QwellManxPower: unless you do an explicit Playtones()
18:36.35*** join/#asterisk zoo (nobody@ip-132-16.travedsl.de)
18:36.39ManxPowerQwell: then you should not need 'r' if you are using playtones
18:36.41ManxPowercab is here!
18:36.52Qwelldunno, doesn't work without it
18:39.10dougheckaDANG IT
18:39.32nighty-ok now I have my licence
18:39.58nighty-so it seems to work but I get a strange noise still
18:40.10nighty-when talking between my 2 extensions
18:40.14Hmmhesayswhy do people schedule things for 3pm on friday
18:40.17Hmmhesaysduring the summer
18:40.21dougheckait just refuses to pass calls now
18:41.02DarthClueHmmhesays: because it works...for them.
18:41.24HmmhesaysI believe there is a more sinister motive there
18:41.58DarthClueHmmhesays: nope.  you see, they had a 9am tee time and by 3pm they will be ready to get to work.
18:42.53Hmmhesayswell I want to play some frisbee golf at 3 dang it
18:43.28*** join/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.hsd1.fl.comcast.net)
18:43.51*** join/#asterisk Jearil (colin@67.151.160.162)
18:43.54obsidian-studiosgreetings doing some research for * pres for the local LUG, I know god help them ;)
18:44.16obsidian-studiosis there a reason to use or not to use IAX over SIP for softphones?
18:44.45pawhat is Ringing()? if i use it should i hear the "free line" sound? i tried it followed by a wait() but i cant hear that sound..
18:45.08nighty-anyone would know why I get this helicopter noise in the background (even though I do have g729 licences) ?
18:45.32nighty-there is a g279 to g711 transcoding in this case
18:45.40*** join/#asterisk citats (james@miles.gnuinter.net)
18:46.41*** join/#asterisk oej (~oej@apollo.webway.se)
18:46.44*** join/#asterisk trimi` (Pharrel@62.162.243.27)
18:47.29coppicemaybe you are using bell 212? :-)
18:47.30DarthClueobsidian-studios: IAX is great if everyone implementation of * is the exact same version.  Otherwise, it will cause problems.  SIP is great unless you have to deal with a firewall, in which case, you have to do some additional config depending on what the firewall can do.
18:47.47nighty-coppice: me ?
18:47.53nighty-coppice: bell 212 ?
18:47.59obsidian-studiosDarthClue:  with regard to actual phones are there IAX phones?
18:48.12nighty-coppice: oh bell helicopter :)
18:48.17trimi`<obsidian-studios> yes
18:48.17obsidian-studiosDarthClue:  I assume IAX was best for use between * boxes
18:48.25coppicebell 212 was a modem standard
18:48.26coppicebell 212 was a popular model of helicopter -)
18:48.29DarthClueobsidian-studios: i think there may be a IAX hardphone but not one in wide use.  SIP phones are the best right now.
18:48.31Delta34so i been trying to get my * to * talking together using IAX, but no luck so far, I read the wiki pages
18:48.43Delta34can someone help me with my IAX issues?
18:48.51obsidian-studiosDarthClue: I have heard that SCCP/Skinny can be better than SIP just not as easy to setup etc/
18:48.58nighty-coppice: you have an idea ?
18:48.59dougheckagrr
18:49.00*** join/#asterisk mkrufky (~mk@68.160.103.77)
18:49.01Hmmhesaysstateful routers, and short sip registration expires have made sip much more manageable in a nat situation
18:49.25*** join/#asterisk rayvd (rayvd@arthur.bludgeon.org)
18:49.39obsidian-studiosis IAX a standard?
18:49.43rayvdIs there a good way to check within my dial plan whether or not a user has a voicemail box?
18:49.51DarthClueobsidian-studios: could be.  right now, i've had the best luck with polycom ip phones using sip.  stay away from grandstream at all costs.
18:49.52rayvdI want to send to voicemail if so, otherwise do Congestion and hang up...
18:50.07Delta34if i do a iax2 show peers
18:50.13DarthClueanthm: that's just being mean to the toilet paper.
18:50.20HmmhesaysI dunno if you want to be using toilet paper with a bunch of ink on it
18:50.23Delta34i see the other * box
18:50.24obsidian-studiosDarthClue: I tend to use Cisco stuff of eBay, if I have to buy new was considering Sipura's stuff, I think Cisco bought them though so ?
18:50.32*** join/#asterisk pbxbart (user@p54B01C7C.dip0.t-ipconnect.de)
18:50.42coppiceanthm: toilet paper is too good for it
18:50.56*** join/#asterisk Twister (Twister@24-179-88-187.dhcp.chtn.wv.charter.com)
18:51.04DarthClueobsidian-studios: if you buy new, i would recommend polycom.  i haven't used a sipura yet, but thus far i am completely impressed by the polycoms.
18:51.11harryvvcisco bought out sipura.
18:51.24obsidian-studiosDarthClue: cool thanks I will check them out
18:51.32*** join/#asterisk capouch (501@12.176.248.4)
18:51.36DarthClueobsidian-studios: btw, you can get polycom ip501s for 172ish
18:51.44harryvvdarth, i have sipira...good units. you can even make it dial 9 within the dial plan.
18:51.45obsidian-studiosis IAX just * or is it a standard used in say CCM?
18:51.52Delta34can anybody help me?
18:51.53obsidian-studiosno experience with CCM FYI
18:51.58Cresl1nwhat's CCM?
18:51.59coppicepolycom doesn't echo cancel their handsets, which is pretty crappy
18:52.06DarthClue~IAX
18:52.06jbotrumour has it, iax is port 5036 for the original (deprecated) IAX protocol. Port 4569 is for the the current IAX2 protocol. IAX is pronounced "Eeks". stands for  Inter-Asterisk Exchange
18:52.06obsidian-studiosCisco Call Manager
18:52.22obsidian-studiosok so IAX is pure *
18:52.22fileis crap
18:52.25obsidian-studioscool
18:52.32Cresl1nno IAX in CCM :-)
18:52.41fileif there was IAX in CCM, I would be afraid
18:52.55obsidian-studios:) just wanted to make sure IAX was * based
18:53.08*** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com)
18:53.12obsidian-studioswill it be used in the future for phones, or is it's purpose mainly between * boxes?
18:53.18harryvvI LOVE the sound quality on a ip500 is so clear.
18:53.44fileit's already used with a chinese VoIP chipset
18:53.53nighty-coppice: no more choppy noise , did not do anything , strange
18:53.53fileand softphones
18:53.54nighty-:)
18:54.29obsidian-studiosso in the future there may be more IAX phones, but they are destined to only work with * unless others systems pick up the protocol
18:54.47Cresl1nAsterisk is a could system to work with :-)
18:54.53DarthClueobsidian-studios: yes
18:55.22*** join/#asterisk scud (~scud@12-214-190-139.client.mchsi.com)
18:55.29obsidian-studioscool, so not always practical for those migrating from other VOIP platforms or for a wide variety of equipment to choose from
18:56.05scudi've guessed you guys have seen the newsforge article about asterisk@home?
18:56.54obsidian-studiosare there benefits to SCCP/Skinny over SIP?
18:57.14coppiceyes, if you are cisco :-)
18:57.32blitzragecan you not filter on a range of ports with tcpdump?
18:57.38blitzrageor am I just not seeing it in the manpages?
18:57.49Delta34I am getting this when I try to dial my IAX2 peer?
18:57.50Delta34<PROTECTED>
18:57.50Delta34<PROTECTED>
18:57.50Delta34<PROTECTED>
18:58.28obsidian-studioswell like some Cisco phones can have SIP firmware or SCCP/Skinny. In the past I worked with 7910's and thought it was a bad idea because they did not use SIP. I was told by a chan_sccp developer from the one on sourceforge, that SCCP/Skinny is actually a bit better than sip, just not easy to install
18:59.26dougheckahmm
18:59.29dougheckathis isnt right
19:00.00dougheckamy adtran wont accept calls strait from asterisk, but if I tell asterisk just to pickup a particular channel and _I_ dial the digits, it works
19:00.05dougheckarecieved calls have no audio
19:00.14*** join/#asterisk JohnJacob (~JohnJacob@pcp0011543387pcs.mainf01.in.comcast.net)
19:01.23nighty-coppice: arrghh no the voicemail does not work strange noice when calling from a G729 to the VoicemailMain
19:01.34_DAWHey everyone.. I need a little help here with variable math using the Set command.
19:01.47dougheckaManxPower: hi
19:01.57_DAWI just want to increment the count variable by one each time.  I am using exten => 1,1,Set(COUNT=${COUNT}+1)
19:02.19_DAWand its obviously wrong.  my var just comes back as exten+1 (no math performed)
19:02.34nighty-something is wrong in transcoding land
19:02.37nighty-:)
19:04.28obsidian-studiosthanks to all for the info, I will back later to pick your brains more. Just do not want to mislead the local LUG, at least not to much :)
19:04.57blitzrageok, off to get ready and head out to TAUG
19:04.59blitzrage~taug
19:04.59jbothmm... taug is The Toronto Asterisk Users group. The website can be found at http://www.taug.ca
19:05.10blitzrage7:30 @ Toby's!
19:05.31harryvvhow about the vancouver asterisk group
19:05.45harryvv~vaug
19:06.03harryvvguess we dont have on here
19:06.05harryvvone here
19:06.34shidoif I leave now I'll just make it
19:06.52harryvvto where
19:07.12blitzrageshido: leave now!
19:07.27shidowhats up doughecka
19:07.28shido?
19:07.58dougheckawell, when I dial out with a group (g1/18005558355) it just sits there...
19:08.09dougheckaif I dial(zap/20)
19:08.15obsidian-studiosanyone have the link a large publicly available dial plan. I seem to remember on with spocket or spracket or something along those lines
19:08.19dougheckaI get dial tone, and can dial...
19:08.30dougheckaif I recieve a call, it acts like its working, but no audio
19:08.34obsidian-studioslost bookmark, and can't remember name :)
19:09.18obsidian-studiosah this one http://sprackett.com/asterisk/
19:10.28dougheckathey aint talking like they should
19:10.53*** part/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.hsd1.fl.comcast.net)
19:11.17Delta34shouldnt it be (zap/g1/18005558355)
19:11.23hardwireugh
19:11.25hardwiremy boss won't budge
19:11.27hardwirehe wants to use 3 digit extensions
19:11.28dougheckawell, thats what it is
19:11.31hardwireis there a clue x 4 that I can hit him with?
19:11.33hardwire"they won't remember 4 digit extensions.. 3 digit is easier!!
19:11.35hardwireheh
19:11.45DarthCluehardwire: how many locations?
19:11.50hardwire12
19:11.55shidoso hy not stick with 3 digis, hardwire ?
19:11.57shido+w
19:12.10hardwireshido: because we have too many locations.. and too many departments
19:12.11dougheckaanyone know adtran?
19:12.16hardwireso far I have 100 through 600 filled up
19:12.19DarthCluehardwire: 12 extensions or 12 different physical locations?
19:12.19hardwireand I am not done yet
19:12.20shidomore than 10?
19:12.40shido100 1 location , 200 another 300 another
19:12.42hardwireshido: we are going to be installing over 50 phones for 20 different departments
19:12.47hardwirefor the most part
19:12.50shidoI see
19:12.51hardwireit may not make sense..
19:12.55shidoyeah
19:12.58hardwirethe ratio of phones to departments is silly
19:13.02hardwirebut we have too many departments
19:13.02shidoyeah
19:13.19hardwirelike we have one guy.. in a room somewhere.. and he is a department
19:13.25Delta34can someone help with my IAX2 setup test?
19:13.27shidowell tell the boss that people can remember 4 digits - stop underestimating your employees
19:13.31mutilatoryour company
19:13.33shidoand if they dont know the extension
19:13.39hardwireand we need to have a general extensino for that department (hi howdy welcome too..) then the operator extensino and then the person extensinos +_ room for expansion
19:13.39shidothen make a directory
19:13.42shidoweb accessible
19:13.45shidoso they cna look ppl up
19:14.04hardwirethe problem is.. we have like.. a lot of room for expansion.. all of these departments could go up to 10-20 people per in a matter of a year
19:14.07*** join/#asterisk dudes (~dudes@12-215-34-84.client.mchsi.com)
19:14.11DarthClue4 digits are going to be a minimum, the boss will just have to deal with it.
19:14.24Qwellhardwire: make up some BS about local prefixes
19:14.27hardwireDarthClue: for instance.. we have a corp headquarters in anchorage and on an island
19:14.35hardwireand in each location we have say 5 departments
19:14.36solarno biggie it looks like they have a bugzilla type of thing we can use
19:14.43hardwireso using 11xx for anchorage
19:14.51hardwireand 12xx for the island makes sense
19:14.52dudesAnyone know how to limit the number of invites on a SIP trunk /w asterisk?
19:14.57hardwireor.. even better
19:15.01hardwire1xxx for anchorage
19:15.04hardwireand 2xxx for st paul
19:15.14hardwirethen 11xx for finance
19:15.20hardwire12xx for my department
19:15.21hardwire....
19:15.35Qwellhardwire: if there aren't 1000 people in each dept, that might be a bit...wide
19:15.37hardwireand then I could do things like set aside extension blocks for things like conference rooms and misc phones
19:15.43hardwireQwell: yeh.
19:15.50hardwirelemme pastebin my preliminary dialplan
19:15.53hardwireits really immature
19:15.55hardwirelike myself
19:16.01hardwirebut maybe you guys can advise
19:16.08hardwiredoughecka: /me intercepts and ebays
19:16.15dougheckayou wouldnt
19:16.16*** join/#asterisk tim27 (~tim27@97-70.dr.cgocable.ca)
19:16.23hardwireI almost think that should be an ebay commercial
19:16.25dougheckadarn thing wont talk anymore
19:16.30hardwiresomebody gets pissed.. throws office equipment out a window
19:16.38QwellI'd personally do (where y is static and x is variable)  y1xx for a location, y11x for a dept, y12x for another dept, y2xx for another location, etc
19:16.39hardwireand somebody runs by .. catches in mid air.. then freezeframe.. ebay
19:16.43Qwellbut, don't mind me. :p
19:17.02KattyDarthClue: make me an iax2 j2me app, plskthxbi
19:17.02hardwireQwell: thats sorta what I want
19:17.04Qwellthat limits you to 100 users in 10 depts in 10 locations though
19:17.18fockshow would I go about setting up a manual night mode? these people insist on setting night mode manually
19:17.26tim27any know a good VOIP provider with CAN/US DID
19:17.33Qwellhardwire: screw it.  randomize :p
19:17.38hardwirehah
19:17.46hardwireQwell: thats almost worth it
19:17.53dougheckaget my adtran talking to my T1 card
19:17.54Qwellhardwire: its what we do at my work, heh
19:18.22Nugget#asterisk advice:  <ManxPower> Move to Europe   <Nugget> Don't use MySQL   <bkw_> Avoid Bank of America   <JuNkY> File it as a bug in Mantis    <jbot> Read the damn docs, already"
19:18.37Qwellhardwire: I have 6257, the person sitting next to me has 6401, and voicemail is 6262
19:18.38hardwireok people
19:18.40hardwirehttp://pastebin.ca/18796
19:18.48hardwiream I just insane
19:18.55QwellNugget: Complain whenever possible.
19:18.59nighty-can someone tell me why phone to phone sip call work , but voicemail seems to have a transcoding problems ?
19:19.00hardwireI am trying to like..; set aside blocks of 5-10 in a group of 100 for certain things
19:19.01DarthCluehardwire: yes you are insane, but what's the problem?
19:19.10capouchDoes anyone know if the mailing lists are down right now?
19:19.11hardwireDarthClue: my damn dialplan initiative
19:19.13nighty-I mean what could be some causes ?
19:19.20hardwiresee how like.. I have the auto answer extensions
19:19.22capouchI haven't gotten any -users or -dev mail since late last night
19:19.25DarthCluecapouch: probably are.
19:19.33tim27any know a good voip provier with 800 number DID ???
19:19.42Qwelltim27: asterlink
19:19.46hardwirefor when the DID is dialed... or somebody from another department needs to transfer to the menu of another one
19:19.48sedwards50I need help configuring a Adtran 750 talking to a t100p -- the fxo lines won't hang up!
19:19.48blitzragecapouch: I've been hearing that from a few people now... so probably are
19:19.49filemy latest post came from ManxPower, entitled "Re: [Asterisk-Users] delay on pri dialling when asterisk is splicedbetween E1-Pri and legacy pbx"
19:19.50DarthCluetim27: yes asterlink
19:19.59DarthCluehardwire: looks reasonable at a first glance.
19:20.02capouchThx
19:20.03blitzragehas anyone contacted Digium yet? :)
19:20.04hardwireand then I have the operator extensions.. that are set up to ring multiple phones per department
19:20.11mutilatorthats a lotta phones
19:20.11hardwireDarthClue: please let me know where I fall short :)
19:20.19hardwireI am keeping my pants on so you can't comment on a few things
19:20.20tim27Qwell Darth they have 800 for canada ???
19:20.27Qwelltim27: You never said Canada
19:20.29capouchblitzrage: Yes, I sent a mail to Rick a while ago
19:20.34DarthCluetim27: you really should have said that before.
19:20.35blitzragecapouch: cool
19:20.39hardwireDarthClue: like line 14.. I have that open for more Tenant space
19:20.45Qwellblitzrage, capouch: They might not ever get it. :p
19:20.47fileif you want low rates for Canada, you have to look far and wide
19:20.49hardwirefofor just the IVR
19:21.04hardwireand yet we only have like.. 10 available extensinos
19:21.05blitzragetelecom is just more expensive in Canada
19:21.07hardwirein seperate small blocks
19:21.14hardwirethats where I want to have 4 digit
19:21.30blitzragebut we have pretty good service as a whole, so its worth it in my opinion
19:21.33hardwiremore space to breathe
19:21.43mutilatordo you think you're going to expand that much though?
19:21.52hardwiremutilator: my area will expand by 10 people
19:21.57hardwireI didn't leave space for that many
19:22.03hardwireand we are all just a bunch of startups for the most part
19:22.04hardwireso yes
19:22.05tim27i sign on link2voip
19:22.10hardwireI have to leave that much room
19:22.10tim27but it seem to suck
19:22.14Qwellmutilator: All that has to happen, is one single dept to expand over a set limit, and it screws the whole structure
19:22.27mutilatoronly screws that department
19:22.30hardwireQwell: so I am not insane to try to leave buffer space?
19:22.34mutilatoryou can move a department to another range later
19:22.36hardwireI like your idea.. completely random :)
19:22.39Qwellhardwire: I'm still for randomizing it
19:22.43Qwellexcept for some preset blocks
19:22.53hardwireQwell: but what about structure.. is it too hard to maintain?
19:22.55Qwelllike, 1100-1199 for conf calls, or whatever
19:23.03hardwirelike.. I want to be able to take a block of 10.. and just move it to another location if I want
19:23.15mutilatorhardwire: i also personally suggest 4 digits
19:23.22Qwellyeah, 4 digits is a given
19:23.26hardwiremutilator: thanks
19:23.29hardwireok
19:23.37hardwirewell I am going to twell my boss that some very smart people told me so
19:23.40hardwireand therefore my word goes
19:23.44mutilatorheh there ya go
19:23.44blitzrageManxPower: haha, res_indications.conf wasn't loaded - that was the ringing problem :)
19:23.47hardwireI love how they trust me to do what they think is best :)
19:23.54Qwellhardwire: except we aren't very bright. :p
19:24.03mutilatorGENIUS is what i prefer
19:24.05mutilator;)
19:24.07Qwell;]
19:24.08hardwireQwell: when you say my name.. your nickname is bright yellow.. that counts
19:24.18Qwellgood point
19:24.26tim27so any know a VOIP provider than can provide me a 800 DID number ... that work from canada ???
19:24.28twisted[asteria]it's  bright green on my screen
19:24.38Qwelltwisted[asteria]: even when I say your name? ;/
19:24.45hardwiretim27: isn't there a list of VoIP providers on voip-info.org ?
19:24.47twisted[asteria]when you say my name it turns bright red
19:24.49mutilatormine ignores the line totally
19:24.50Qwellevil
19:24.51hardwirefor canada.
19:24.53mutilatori don't like people
19:25.00hardwiremutilator: you mutilate them
19:25.05hardwire?
19:25.11mutilatorcuz
19:25.12mutilatori don't
19:25.16mutilator*shrug*
19:25.24hardwirethis sucks
19:25.26tim27hardwire: the list dont tell who is good and not
19:25.27rikstaanyone in the UK that can provide 0906 -> ip?
19:25.31tim27link2voip suck
19:25.32hardwireI was supposed to meet up with the teliax dude and I lost his damn cell phone
19:25.52hardwiretim27: ah.. I always figured you just learn as you go :)
19:25.56mutilatorfrickin hell, i still have an hour and a half to work
19:25.59hardwirewasting money is a valuable way to learn
19:26.05mutilatorand no paycheck in hand yet, so i HAVE to stick around
19:26.14hardwiremutilator: heh
19:26.20hardwiredirect deposit is a lovely thing
19:26.31hardwireout of sight.. out of mind
19:26.31mutilatori have to learn how to do ACH first
19:26.36mutilatorthen i can set that up here
19:26.43hardwireACH?
19:26.47tim27i already spent 50 $ with link2voip and all i got is crappy voice
19:26.52hardwiremutilator: wtf is this MIMO crap?
19:27.11Qwelltim27: I can give you crappy voice for $25
19:27.14Qwelldeal?
19:27.17blitzragetim27: what you looking for? Per minute or unlimited?
19:27.29hardwiretim27: I can too.. but for free.. just call and I will chew you out
19:27.30Qwellblitzrage: for tollfree DID?
19:27.36hardwireits good therapy
19:27.48mutilatorhardwire?
19:27.50Qwelldo unlimited tollfree plans exist?
19:27.53blitzrageQwell: oh, tollfree?  Sorry, wasn't really paying attention I guess :)
19:27.59blitzrageQwell: not as far as I know
19:28.00hardwiremutilator: MIMO wireless equipment
19:28.02tim27Qwell: good deal
19:28.11tim27blitzrage: per minute
19:28.22tim27or maybe unlimited
19:28.35Qwelltim27: You probably won't find an unlim tollfree provider
19:28.35tim27tool free will be per minute
19:28.40tim27i guess
19:29.00tim27voicepulse was fine
19:29.06tim27but they dont have 800 number
19:29.11tim27hmmm
19:29.16dougheckaNOTHING WORKS
19:29.20Qwelldoughecka: send it on over
19:29.35blitzrageyah, I've never seen unlimited toll free
19:29.56hardwiretim27: whats wronte with link2voip?
19:30.04hardwireother than the obvious bad name
19:30.10hardwirewrong
19:30.16mutilatorlooks like just multiple antennas
19:30.22konfuzedoh boy I finally got my "provider" live on irc so that this matter can be resolved
19:30.24mutilatorfor one device
19:30.26mutilatornever used it before
19:30.28konfuzedmy lack of connect that is
19:30.28ManxPowerdoughecka: breathe dude.
19:30.29hardwiremutilator: yeh.. it just seems odd
19:30.35Delta34anyone mind tackling my iax setup? just trying to figure out how to do it? no luck so far =(
19:30.36hardwireand useless
19:30.44mutilatormight be useful
19:30.44hardwireDelta34: pastebin whatcha got
19:30.50mutilatoryou get more area to receive signal
19:30.56hardwireI am not going to work until the number "4" registers completely in my head
19:30.56mutilatorincreasing link quality
19:31.15hardwiremutilator: I use dual antennas on b equipment
19:31.17hardwireit works well
19:31.19mutilatorinstead of just throwing champ amps on anything
19:31.26hardwiremutilator: heh
19:31.26hardwireso
19:31.34hardwireI have these 400mw 802.11a and 802.11b/g cards
19:31.39hardwireand I am istalling them in a week
19:31.41hardwireI can't wait
19:31.46mutilatori wanted to try those
19:31.53hardwire-94db
19:31.57hardwireor some shit like that
19:31.59mutilatorhalf watt on a card is sweet
19:32.02hardwireyes
19:32.04hardwireand they are cool
19:32.10mutilatorbet they get hot tho
19:32.11So3krishardwire: know you the brand and is it linux comp. ?
19:32.12hardwireI am going away from u.fl however
19:32.18hardwireSo3kris: eh?
19:32.24hardwireoh
19:32.29So3krisof you wifi card
19:32.46hardwireyes
19:32.47hardwireits atheros
19:32.48mutilatoryea
19:32.50mutilatorthey're atheros
19:32.52So3krisok then
19:32.58mutilatorshould work in linux..
19:33.04So3krishardwire: are you form the usa ?
19:33.05hardwirelemme get you a link
19:33.09hardwireplease hold
19:33.40hardwirewisp-router.com
19:33.47hardwirehttp://www.wisp-router.com/product_info.php?products_id=400
19:34.04mutilatorthey sure do suck the juice tho
19:34.08hardwirehttp://www.wisp-router.com/product_info.php?cPath=35_51&products_id=399
19:34.14hardwiremutilator: already using a 233mhz geode
19:34.19hardwirenot like I am having juice issues
19:34.23mutilatorheh
19:34.33hardwireI need to update my embedded debian buildroot
19:34.38hardwireso I can get it all stablized
19:34.43hardwirethen use these cards
19:34.55hardwireI got shipped back one of the embedded routers I made
19:34.59hardwireits an outside one
19:35.00Delta34here's my iax problem, http://pastebin.ca/18804
19:35.03hardwirethe sea -salt welded it shut
19:35.15hardwire-98dbm
19:35.16hardwiredamn
19:35.24So3krishardwire: http://www.netgate.com/index.php?cPath=21
19:35.47hardwireSo3kris: ok
19:35.50hardwirewhats up?
19:35.54hardwireDelta34: whats the problem?
19:36.04mutilatorhow much those cards run?
19:36.12Delta34i cant call the other iax box
19:36.21Delta34i get line congestion
19:36.29So3krishardwire: http://www.demarctech.com/products/reliawave-mnt/cable-assemblies.html
19:36.31hardwiremutilator: it says on the site :)
19:36.40hardwireSo3kris: I have lots of pigtails
19:36.44hardwireso.. what gives?
19:36.46Delta34<PROTECTED>
19:36.46Delta34<PROTECTED>
19:36.46Delta34<PROTECTED>
19:36.54So3krishardwire: ufl connectors
19:37.00hardwireI have lots of them :)
19:37.03So3krisoh
19:37.04hardwireI am moving away from them
19:37.13So3krisi touth you are searching
19:37.20hardwiregoing to use the MMCX connectors
19:37.24hardwireand I can't use those pigtails
19:37.29konfuzedhey this may seem like one of those obvious questions but I just cant rely on my own interpretaion.........
19:37.30hardwireI can only use the hyperlink ones
19:37.44hardwirethey are the only ones that are rubber sealed that I like.. and they extrude a bit further
19:37.53*** join/#asterisk DarthClue (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net)
19:37.57hardwireDelta34: so whats the problem?
19:37.59Delta34so is  this a syntax issue
19:38.07konfuzedmy "provider"   got this message on his CLI when I called in
19:38.10Delta34i cant call the other * box
19:38.14konfuzed[15:36:34] <DaveCanoe> Jul 29 15:31:37 WARNING[642]: chan_sip.c:6861 handle_response: Forbidden - wrong password on authentication for INVITE to '"4164613806" <sip:4164613806@66.96.30.25>;tag=as0f4b9eac'
19:38.14konfuzed[15:36:34] <DaveCanoe>     -- SIP/profx-f8d4 is circuit-busy
19:38.20hardwireDelta34: whats your verbosity set at?
19:38.25Delta343
19:38.29hardwirego to 8
19:38.32hardwiremake the call again
19:38.33hardwirethen psot it
19:38.35hardwirepost it
19:38.39konfuzedobviously the password fails
19:38.48mutilatorhm
19:38.52mutilatormini pci tho
19:38.59hardwiremutilator: whats wrong with mini pci?
19:39.14mutilatorcan't get a laptop with extra mini pci slots
19:39.20Delta34setting to 8 same error msg, not informative at all to me
19:39.24konfuzedbut what else does this message say
19:39.34hardwiremutilator: hah
19:39.39hardwirethey make a 400mw pcmcia card
19:39.52Delta34i dont even see the call coming in on other *
19:39.53mutilatordo they? i saw up to quarter wat..
19:40.13hardwirethese guys don't sell them
19:40.22konfuzedthe number 4164613806   is the number I am calling from not the number I am calling too yet it appears to be using that phone number as the username for authentication
19:40.38blitzragehardwire: verbosity past 4 doesn't do anything
19:40.48konfuzedis my interpretation close to accurate here?
19:40.50hardwireDelta34: maybe it would be informative to me?
19:41.04Corydon-wverbosity of 10 or greater does things for ODBC...
19:41.16hardwireblitzrage: he is at 3
19:41.27hardwirerule of thumb seems to always be 8
19:41.32hardwiredon't ask me why
19:41.41blitzragehardwire: code doesn't seem to have anything past 4 though :)
19:41.46Corydon-wverbosity beyond 4 doesn't change the output beyond what it looks like for 4
19:41.52hardwireblitzrage: do I sound like somebody thats in the code all day long?
19:41.59konfuzedwhy does asterisk appear to be using the incoming phone number as a username for authentication?
19:42.15blitzragehardwire: I'm not in the code either, but I'm just letting you know that it doesn't get affected past 4 FYI
19:42.22Delta34yeh so same error as before
19:42.26blitzragekonfuzed: that's how it works in SIP
19:42.27hardwireblitzrage: from now on that will be stamped in my mind
19:42.35hardwireDelta34: ok.. so you are not even getting anything on the remote iax
19:42.40hardwiremake sure firewall ports are opened
19:42.45hardwirealso.. iax2 debug coudl be usefull
19:42.52rikstaanyone in the UK that can provide 090x -> voip?
19:42.57Delta34nope
19:43.02Delta34no firewall
19:43.08hardwiremutilator: http://www.ubnt.com/src/index.htm
19:43.13hardwireI thought this was a 400mw one
19:43.15hardwirebut no
19:43.18hardwire300mw for 2.4ghz
19:43.23konfuzedso this message then is very misleading beyond a password mismatch
19:43.40hardwirekonfuzed: I guess
19:43.42konfuzeddoes it indicate any other problem beyond password mismatch
19:43.52hardwire:)
19:44.09hardwiremutilator: http://www.ubnt.com/sr5/index.htm
19:44.12hardwireI want this one..
19:44.17hardwirewith the SMA
19:44.20hardwirethat would freaking rule
19:44.21hardwirebut..
19:44.25hardwireit would fry the mainboard I am using
19:44.27mutilatorwonder if terabeam makes bin's for that chipset
19:44.33Delta34so is my syntax wrong?
19:44.37hardwireas the connector would officially sit on the mainboard
19:44.39So3kriscool stuff
19:44.46hardwireSo3kris: so.. whats the deal with your nick?
19:44.52hardwireits freaking me out
19:44.57So3krisik got 1 4521
19:44.58*** join/#asterisk DarthClue (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net)
19:45.19hardwireok
19:45.20konfuzedSo3kris:  are you with Soekris Hardware or you just use their hardware ?
19:45.24hardwireI guess I don't have much room to talk
19:45.25hardwirelike
19:45.28hardwirewith my nick
19:45.33hardwirebut yours is jsut freaking me out still
19:45.33So3krisi use the hardware
19:45.36hardwireah
19:45.38mutilatorheh
19:45.41hardwireI have tons of soekris hardware :)
19:45.44hardwireand netgate
19:46.04hardwireand pcengines
19:46.05hardwiremmm
19:46.07hardwireyummy pcengines
19:46.15So3kriscool i live in holland and there is it not easy to get 1
19:46.17konfuzedsoekris has kewl accessible gear
19:46.24harryvvdo you guys use any fuseable protection between the demarc point and the asterisk box?
19:46.59mutilatorcat5..
19:47.20mutilatorstorm last week left black streaks on a few cables when it fried the pbx
19:47.25harryvvlike a big voltage spike or lightning
19:47.44mutilatorshoulda heard it when it happened
19:47.51mutilatorwas the coolest thing i ever heard
19:47.55harryvvthay are used on each post termination point at the central office
19:48.09harryvvpots not post
19:48.37harryvvI have heard of lightning hitting the ground and moving up the grounding point.
19:48.58mutilatorheh we had a lightinin rod on a tower
19:49.00So3krishardwire: whats the price of a 4521 ?
19:49.02hardwireI had a really cool setup
19:49.05mutilatorturned into a ball of steel
19:49.06hardwireSo3kris: I don't fucking know
19:49.09mutilatorwas pretty sweet
19:49.18mutilatorwas just a glob on a lil stubby pole
19:49.19So3kris:D
19:49.20hardwirewww.soekris.com probably does
19:49.32harryvvmut, well since lightning is as hot as the surface of the sun.
19:49.49lathos42+s
19:49.57So3krisit could be that the are kost less than on soekris.com. in holland the are expencive
19:49.57mutilatoryea, it's cool to watch lightnin blow up everything we put out
19:50.00_DAWdoes anyone here know the correct syntax for variable math in * using the Set command?
19:50.12harryvvvariable math?
19:50.24hardwirehttp://happy9.bogomip.com/albums/work/tdx/dutchharbor/img_4713.jpg/view
19:50.27_DAWyes doing math with variables..
19:50.28hardwiremutilator: thats the boards I use
19:50.35harryvvdoes asterisk have a transfering sound?
19:50.38hardwirereplacing the top and bottom mini-pci with the 400mw 802.11a
19:50.41hardwirefor our backbone on the island
19:50.55hardwirethats my own mish mosh
19:51.04hardwirethat board is set up for getting its RTC from GPS time
19:51.09mutilatorwhere do you get those boxes?
19:51.10hardwirewhich is the white cable
19:51.13hardwirewww.minibox.com
19:51.19_DAWharryvv - it is doucmented in the wiki for setvar, but I am having no luck getting it to work with SET
19:51.23mutilatorum
19:51.25mutilatorya sure?
19:51.26hardwireI get the wrap 1d something w/ 4 holes in the chassis
19:51.38mutilatorthats a like.. adware page
19:51.40hardwireshhh
19:51.45hardwirewhat am I a goddamn arms dealer
19:51.49hardwireyou 3rd worlders keep bugging me
19:51.54mutilatorfoo
19:51.56hardwirehttp://www.mini-box.com/site/index.html
19:51.57hardwirethere
19:52.02harryvvhardwire did you make that wap?
19:52.08hardwirehardwire: I assembled it
19:52.15hardwireand coined the term BulletDebian for my uown purposes
19:52.37hardwirea distro I have set up to use a unionfs and non-volitile storage
19:52.40gtigeneHow does Asterisk support vertical service activation codes like *69, etc. I have Polycom phones.
19:52.51harryvvhardware, what parts?
19:52.54hardwirebut it has the option of making root level filesytstem changes
19:53.32hardwirehttp://happy9.bogomip.com/albums/work/tdx/dutchharbor/photoalbum_photo_view?b_start=1
19:53.36droothwhat is wrong with nufone? I cant get them to do one simple thing like change my fail to.  they just ignore chats and emails
19:53.38hardwiretheres the installed box on top of the sat receiver
19:53.41harryvvhardwire okay yea remember you made that satuplink
19:53.42harryvv:)
19:53.53hardwirehttp://happy9.bogomip.com/albums/work/tdx/dutchharbor/img_5083.jpg/view
19:53.57hardwireand there it is.. on top of our boat
19:54.07*** part/#asterisk _DAW (~bob@68-114-110-210.dhcp.slid.la.charter.com)
19:54.09hardwireyou can't really see it
19:54.14*** join/#asterisk _DAW (~bob@68-114-110-210.dhcp.slid.la.charter.com)
19:54.21hardwiregot o original size.. and you can atleast make out the mast (top left boat)
19:54.45harryvvso where did you get the case,antennas,jacks and board?
19:54.58hardwirewww.mini-itx.com and hyperlinktech
19:55.18hardwirethe CF Came from techdata.com (since they are our supplier for common computer crap)
19:55.30hardwirehttp://happy9.bogomip.com/albums/work/tdx/dutchharbor/photoalbum_photo_view?b_start=0
19:55.31konfuzedok so, in that I am a bit konfused right could someone clarify  which conf file will contain the userid which one has the authid specified and which conf files have their corresponding password (if its all the same file please say so)
19:55.38hardwireall that came from different places
19:56.05hardwirehttp://happy9.bogomip.com/albums/work/tdx/dutchharbor/Photo_042905_002.jpg/view
19:56.07hardwiremuch better view
19:56.08So3krishardwire: know you good hardware whit 3 wifi cards/interfaces and not the 4521
19:56.12hardwiremy damn palm pilot took that picture
19:56.25harryvvhehe
19:56.28konfuzedhardwire:   have you ever boated near   http://iwavecanada.com  locations ?????
19:56.29harryvvthats pretty good
19:56.30hardwireSo3kris: 3 .. no
19:56.35hardwirekonfuzed: no
19:56.36*** join/#asterisk Grog (~danievh@ndn-165-131-91.telkomadsl.co.za)
19:56.42hardwirejust in dutch harbor and the bering sea
19:57.11hardwireSo3kris: I would think what you would want would be 3 pci boards.
19:57.18gtigeneHow does Asterisk support vertical service activation codes like *69, etc. I have Polycom phones.
19:57.26hardwireso I would grab a small system w/ one pci slot
19:57.34hardwireand install the 4-mini-pci to pci adapter
19:57.35konfuzedbetter yet what conf files do I ask the provider to show me so that I can prove what how his side is configured
19:57.40hardwirethen use up 3 of the mini-pci slots
19:57.45gtigeneI mean, does Asterisk support *69 etc with digital phones?
19:57.46hardwirehave fun with those irq's kids
19:57.55mutilatoryea
19:58.00hardwiremutilator: how?
19:58.08hardwirecause I wanted to do that the other day
19:58.11GrogGood evening... Anybody here close to South Africa?
19:58.17hardwiresince the channels are dynamic.. I wouldn't know how that would work
19:58.37Hmmhesayson a universal scale i'm close to south africa
19:58.39hardwireanybody in seattle/
19:58.40hardwire?
19:58.46hardwireI need a case of mac-n-jacks
19:59.03Hmmhesaysbut then again, on that scale we all are close
19:59.08GrogOK...I need help quick...hoping for somebody close
19:59.16Hmmhesayshelp with what?
20:00.19GrogWe were willing to fly somebody in, but since you ask... Getting details.
20:00.34hardwireI want their damn beer damnit
20:00.35hardwireon keg
20:00.36hardwiretonight
20:00.39Hmmhesayskeg yuck
20:00.40konfuzedbetter yet what conf files do I ask the provider to show me so that I can prove how the providers side is configured
20:00.54harryvvgtigene I just checked on my ip500 and yes, it does support last number called.
20:01.01blitzragespeaking of beer... good idea
20:01.12konfuzedof course just the files that pertain to the DID they setup for me and of course only the parts that refer to my account
20:01.15gtigeneharryvv: thanks
20:01.16blitzrageworking at home r0x0rz
20:01.19HmmhesaysGrog: what problem are you having?
20:01.49hardwiredamn
20:01.56hardwirethey only distribute to washington oregon and idaho
20:01.57harryvvstill trying to find a voip or telephone domain and 99% of everything i type in its taken. ;)
20:01.59hardwirebest beer ever
20:02.05hardwireand I can't have it short of a plane ticket
20:02.11HmmhesaysI prefer budweiser for middle class beer
20:02.18hardwirehttp://www.beerpal.com/Mac-and-Jacks-African-Amber-Ale-Beer/5208/
20:02.25*** join/#asterisk TrevorSHarrison (~trevorsha@24.49.36.218)
20:02.46Hmmhesaysnewcastle is good
20:02.52hardwirewhat the fuck are you smoking?
20:03.21hardwireI'm a beer snob I think
20:03.29hardwireI spend a good time sniffing the beer before I drink it
20:03.29Hmmhesaysi'm poor
20:03.33hardwireis that just overkill?
20:03.37twisted[asteria]i'm a beer snob
20:03.39hardwireHmmhesays: well I only have like one beer a week
20:03.42hardwire"_
20:03.43hardwireerr
20:03.43hardwire:)
20:03.44harryvvmabey you listned to your dog :)
20:03.48gtigeneharryvv: Do you think the *69 support is in the phone setup or in Asterisk? My phone doesn't do it.
20:03.50hardwiremy dog speaks in tounges
20:03.57harryvvasterisk setup
20:03.57hardwirelike chewbacca
20:04.02harryvvheheh
20:04.19hardwiretwisted[asteria]: any largely distributed beers you prefer
20:04.21hardwire?
20:04.22blitzragegtigene: some phones have some of those codes built in, but more than likely you're going to have to patch on those features in dialplan logic and make it work that way
20:04.34twisted[asteria]hardwire, guiness
20:04.37hardwireheh
20:04.43harryvvwe have a sheltie and its so smart..some times he is funny.not as fast when giving commands unless there is a reward :) trying to make it limp but thats a hard one.
20:04.44blitzragetwisted[asteria]: never had a Guiness
20:04.56hardwirein Fort Collins, CO we had a large amoutn of excellent beers
20:04.56harryvvmy dog counts
20:05.00harryvvcounts fingers
20:05.01harryvv:)
20:05.08gtigeneblitzrage: Not good new but thanks for telling me.
20:05.10hardwireI am a heffy fan
20:05.12hardwirefor the most part
20:05.18pbxbarthi, ist there any one willing to answer me a few basic c/asterisk questions?
20:05.33gtigenepbxbart: basic questions yes
20:05.35konfuzedwell fine my time is up for now perhaps later i can get simple clarifications
20:05.38gtigenebasic
20:05.55blitzrage~docs
20:05.55jbotsomebody said docs was Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
20:06.01hardwirehttp://www.beerpal.com/New-Belgium-Sunshine-Wheat-Beer/7499/
20:06.03hardwireone of my favorites
20:06.42konfuzedhows this for basic, a plain list of the conf files my provider had to touch to Add my DID service and have it call out to my asterisk ip
20:06.48hardwiremy new typical beer.. http://www.beerpal.com/Alaskan-Amber-Beer/221/
20:06.51hardwireits a damn good beer
20:07.23konfuzedDeep Thick Guiness thats real beer
20:07.31hardwireits good
20:07.32hardwireits easy
20:07.34hardwireits in a can
20:07.36gtigeneharryvv: So for *69 I would like, save the caller id in ast db and retrieve it and use it when the user punches *69?
20:07.37Hmmhesaysyou chew guiness
20:07.41Hmmhesaysits like paste
20:07.47hardwireheh
20:07.55hardwireHmmhesays: maybe next time you have one
20:08.01hardwireyou will go get some woodchuck granny smith apple cider
20:08.05konfuzedyes it is high end food
20:08.06Hmmhesayslol
20:08.09hardwirepour in half a pint into a glass
20:08.24hardwirethen distribute over the top of a spoon onto the top of the cider.. the guinness
20:08.24blitzragegtigene: yep, then you could use saynumber() to say the number for you
20:08.35hardwireand enjoy atleast 5 of those by the end of the night
20:08.40hardwireits a wonderfull thing
20:08.41Hmmhesaysmy drink of choice is shakers on the rocks
20:08.42pbxbartmy questions are very basic c stuff
20:08.49konfuzedgee I should try that
20:08.51hardwire#asterisk-beer
20:08.52hardwireheh
20:08.55gtigeneblitzrage: thank you
20:09.02hardwirekonfuzed: its amazingly good
20:09.09konfuzedgot organic apple cider vinegar in the fridge
20:09.10hardwireits like a black and tan.. but I hate those
20:09.12pbxbartI'm just started hacking chan_sip to locate some problem
20:09.16konfuzedwait a minute
20:09.17hardwirekonfuzed: that would be bad.
20:09.22hardwirebad bad konfuzed
20:09.32blitzragegtigene: going to be a bit tricky if you have multiple extensions. You'll have to save the last number for each extension so that when that extension dials *69,it gets the last number that called them, and not someone else
20:09.34konfuzeddid you say just apple cider juice or cider vinegar?
20:09.37hardwireuse either the woodchuck or hornspy draft
20:09.42hardwirehornsby
20:09.43Grogok...here we go... I got a aah1.03 system with 3 fxo ports connected to a samsung pbx. I figured out most of the stuff in the config files as I normally work with different distros of linux....However, we installed the system, transfers, etc all working fine...but.. if I dial an outside line, connecting me to the main pbx with the pstn lines, I can dial any extension without problems...the trick is, to access an outside from the samsung pbx
20:09.53hardwirekonfuzed: I mean hard apple cider
20:10.02hardwireboy you are confused.
20:10.17konfuzedas in fermented but not to the point of vinegar
20:10.17gtigeneblitzrage: I am extension 223, my boss is extension 218, etc. Is that what you mean by multiple extensions?
20:10.24blitzragegtigene: yes
20:10.28konfuzedeasily confused
20:10.43hardwireok
20:10.43konfuzedI can't even trust my own interpretation of addition
20:10.44Hmmhesaysyou got analog station lines from the samsung connected to fxo ports on the aah box?
20:10.44blitzragegtigene: so save the last number going to each extension
20:10.45hardwireshfs is a neat module
20:10.48hardwireI like it a lot now
20:11.04harryvvhardwire have you tested your setup for range?
20:11.10blitzragegtigene: like, /lastnumber/223/${CALLERID}
20:11.18blitzrage/lastnumber/21/${CALLERID}
20:11.25blitzrage/lastnumber/218/${CALLERID}*
20:11.27konfuzedanyway have to run before the bank wont give me beer money for the weekend
20:11.44blitzragegtigene: not literal, but that gives you an idea of the DB structure
20:11.44hardwirehardwire: 8 miles
20:11.49Hmmhesays8 mile
20:11.53Hmmhesaysfrom the trailer?
20:11.54KattyHmmhesays: poke.
20:11.54hardwireerr
20:11.57hardwire8 miles :)
20:12.02HmmhesaysHey Katty
20:12.09mishehubah.
20:12.09KattyHmmhesays: mine phone has been kattified
20:12.17KattyHmmhesays: www.copi-rite.com/phone (=
20:12.17Hmmhesayshow so?
20:12.34HmmhesaysI am not authorized to view that page
20:12.38harryvvis the wifi near the ground?
20:12.42gtigeneblitzrage: I think I see. Does the asterisk you put in the third example signify anything? :)
20:12.57KattyHmmhesays: k, moment
20:12.58harryvvAre there any wimax phones out yet?
20:13.00blitzragegtigene: no, just to signify that I fixed a typo in the previous line
20:13.15blitzrageharryvv: none that are worth a damn
20:13.28harryvvblitzrage just poor engineering?
20:13.47*** join/#asterisk moy (~kvirc@201.135.113.46)
20:14.01gtigeneblitzrage: Have you seen extensions.ael in cvs head samples?
20:14.02blitzrageharryvv: until I find one with decent battery life and a build in web browser, they are useless. Best bet is to just get a nice PDA that will support a softclient and get a BT headset to work with it somehow
20:14.05KattyHmmhesays: try now
20:14.15pbxbarthow can i print out a complete structure for debuging?
20:14.20blitzragegtigene: yep, I use it at home - got rid of extensions.conf entirely :)
20:14.23harryvvalso, is wimax regulated? that would be nice to keep the frequencies from interfearing with other wimax trancivers.
20:14.27hardwireI so need to go to lunch
20:14.37hardwiremy g/f is probably going to kill me today if I am late
20:14.43gtigeneblitzrage, so its stable, etc?
20:14.44blitzrageharryvv: oh sorry, wimax, I thought you meant just wifi - don't think there's any wimax phones out yet
20:15.09blitzragegtigene: well, its a work in progress, but a bunch of things work with it - whatever else doesn't work you can just use in regular extensions.conf logic and merge the two together
20:15.16harryvvblitzrage from what I have read wimax would fill in the void that wifi supplies..more power at the transcivers.
20:15.17Hmmhesayswhich picture?
20:15.55blitzrageharryvv: yep, longer range for sure
20:16.19gtigeneblitzrage: sounds a bit trick... can you embed extensions.conf-style statements in extensions.ael?
20:16.23KattyHmmhesays: they're on rotate, every 5 minutes
20:16.54hardwireok
20:16.55hardwireit cliked
20:16.56Hmmhesaysi got a directory listing
20:16.58hardwire4 digit extensions
20:16.59harryvvI can see selling a wifi to wimax transverter to boats and ships at sea and extend the range much further out. Then sell the wifi phones to the boat owners.
20:17.02hardwireI can now do
20:17.04hardwirestuff
20:17.10hardwireand not fear the reaper
20:17.12blitzragegtigene: not really. You'd just use both extensions.conf and extensions.ael at the same time - Asterisk automatically merges the files together if both modules are loaded at the same time. Works since AEL is just parsed and turned into extensiosn.conf language anyways
20:17.14hardwirehey..
20:17.18hardwireI can also have it quick dial
20:17.21gtigeneblitzrage: a bit "tricky" I meant
20:17.26hardwireso they can still use their damn 3 digit extensions for local
20:17.30blitzragegtigene: nope, not tricky...
20:17.32hardwireits going to psych them out
20:17.32KattyHmmhesays: all the stuff in there is on my phone, except sweet home alabama
20:17.45twisted[asteria]did someone say alabama? :P
20:17.48harryvvkatty, like josh lucas?
20:17.53Hmmhesaysahh ok
20:17.54blitzragetwisted[asteria]: nope
20:17.55Kattytwisted[asteria]: shh (=
20:18.10twisted[asteria]Katty, *mew*
20:18.13hardwireIU feel like giving somebody a nice big jet li punch right in the neck today
20:18.15hardwirewho's with me
20:18.23gtigeneblitzrage: Is there any documentation on extensions.ael? I know parts of it are pretty obvious.
20:18.52essobiWhat part of the sip session describes the media type of the RTP for the completed call?
20:18.56twisted[asteria]yea blitzrage, where's the documentation on AEL?
20:18.57blitzragegtigene: just the README.ael doc file - should be able to figure most of it out from there. I've not had time to write a document on AEL yet.
20:19.08blitzragetwisted[asteria]: how about a bit fuck j00? :)
20:19.17Hmmhesayshaha
20:19.21gtigeneblitzrage: Are you one of the developers of it?
20:19.21twisted[asteria]blitzrage, use shorthand
20:19.26Kattyit's absolutely insane that these mobile companies want you to way for midi files and gifs
20:19.29blitzragetwisted[asteria]: wish i could!
20:19.33Kattyand wavs too.
20:19.36Kattyit's all MADNESS
20:19.41blitzragegtigene: nope, I just write documentation :)
20:19.42twisted[asteria]Katty, pay? HAH
20:19.47twisted[asteria]Katty, that's why you get the data cable
20:19.48twisted[asteria];)
20:19.48Kattypffft, pay
20:19.54Kattymmm, usb cable
20:19.56blitzragegtigene: http://www.asteriskdocs.org and http://www.oreilly.com/catalog/asterisk
20:20.07Kattyi need a pretty asterisk logo
20:20.10gtigeneblitzrage: Well, thanks for all the input..
20:20.21blitzragegtigene: np
20:20.24twisted[asteria]Katty, webjal hates my phone for some reason
20:20.38Kattytwisted[asteria]: did you install the patch?
20:20.44twisted[asteria]Katty, yeah
20:20.45Kattyhrmm
20:20.52Kattydid you try pk2man?
20:20.54twisted[asteria]but it like, freaks out half way through most of the ringtone uploads
20:20.56twisted[asteria]pk2man?  nope
20:20.59Kattyoh
20:21.03Kattymake sure they're mid0
20:21.03Kattyand not mid1
20:21.10HmmhesaysI got freaking charlie browns mom on the phone here "wah wah wah wah wah"
20:21.12*** join/#asterisk doughecka_ (~Miranda@doughecka.user)
20:21.36doughecka_ManxPower: ping
20:21.49ManxPowerdoughecka: what happened?
20:21.52twisted[asteria]Katty, i'm using wav and mp3
20:21.55doughecka_dsl dropped
20:21.56twisted[asteria]my phone supports upload of both
20:22.32Kattyanyone have a big asterisk logo up their sleeve?
20:22.36Kattyhopefully on black
20:23.04twisted[asteria]Katty, i have the bmp that i use for the cisco
20:23.13Kattypost url
20:23.24Kattyalso, bmp :/
20:23.34twisted[asteria]pbx.indigent-networks.com/asterisk-tux.bmp
20:23.53Kattyoh, that's tiny
20:24.00twisted[asteria]it's big on the cisco ;)
20:24.37blitzragetwisted[asteria]: I used to have a cool picture of boob's for my Cisco :)
20:24.45twisted[asteria]blitzrage, uhm.
20:24.49twisted[asteria]Katty, http://www.comesfa.org/images/asterisk-1927_640x480.gif
20:24.52mutilatorfscking accounting
20:25.03mutilator"i want the PO system to do this and that and do it this way not that way"
20:25.30znoGthey're called requirements
20:25.41mutilatoryea, they should have laid them out when i made the whole thing
20:25.47blitzrageok, I'm out! Peas
20:25.49znoGwell, yes. true
20:25.49Kattytwisted[asteria]: yeah, i just grabbed that one
20:25.53mutilatorinstead they say "make a po and inventory system for ys"
20:25.54twisted[asteria]Katty, ;)
20:25.55Kattytwisted[asteria]: yay for google :>
20:25.56mutilatorus
20:26.01twisted[asteria]yup
20:26.07mutilatori say, what do you want it to do, they say make it work
20:26.07znoGmutilator: so then you stop and say, where is the full list of requirements?
20:26.14mutilatori did
20:26.21HmmhesaysI should not have to explain to someone when they need to use a crossover cable v. a straight thru
20:26.32znoGmutilator: or, as a software engineer, you go and find out for yourself through whatever system you use to research requirements.
20:26.41twisted[asteria]Hmmhesays, relate it to a garden hose
20:26.47*** join/#asterisk file[laptop] (~file[lapt@mctn1-6719.nb.aliant.net)
20:26.51mutilatorznoG: you kidding
20:26.53znoGmutilator: and you are satisfied with an answer like that?
20:26.56mutilatorto them thats wasting precious time
20:26.56Hmmhesaysor point them to webopedia
20:27.04twisted[asteria]Hmmhesays, if you want to push water from one spigot to another, you need a "crossover"
20:27.10znoGto me it seems like what they've told you is wasting precious time
20:27.14twisted[asteria]Hmmhesays, if you want to take water out directly to the nozzle, it's a straight through
20:27.18znoGbut hey, i presume they're paying you for it, so what are you complaining about
20:27.20mutilatoryea, but i'm not the one that cause it
20:27.32mutilatorcause i make shit
20:27.34Hmmhesaysyeah that'd work about as well as trying to swim with 100lbs of brick on your back
20:27.51znoGoh well, it takes you more time to fix it, more $$, sounds good to me. They deserve to pay more
20:28.07Hmmhesaysand, the statement was... I shouldn't have to explain, not I don't know how to explain
20:28.19mutilatorwas the same thing with my time clock
20:28.25mutilatori finally told them no
20:28.29mutilatori'm not changing it anymore
20:28.40mutilatorand they just shutup about it, it was great
20:28.57*** join/#asterisk xlyz (~xl@213-140-17-96.fastres.net)
20:29.07xlyzhi
20:30.06xlyzany1 knows if it is possible to send sms with asterisk through italian telecom isdn ?
20:30.55mutilatorhalf hour
20:31.25harryvvI dont think my wife would like this asterisk sound. http://www.loligo.com/asterisk/sounds/you-sound-cute.gsm
20:31.45mutilatorsexah
20:32.02Delta34finally got my * to * iax2 working, woot woot =)
20:32.25harryvvnow try sip
20:32.41Delta34got that one working too
20:32.55mutilatornow try h323
20:32.59mutilatoro_O
20:33.04echo465now put it behind a nat
20:33.05sedwards50What kind of cable do I need to connect a Adtran 750 to a t100p?
20:33.14mutilatorand then throw it in a blender with some vodka
20:33.46ManxPowera T-1 crossover cable
20:34.08gtigeneblitzRage:  Does all the stuff in README.ael work?
20:34.10mutilatoryou probably don't use radius at all do ya hardwire?
20:34.25mutilatorin your hyper boards
20:34.43gordonjcpmmmm radius
20:34.56hardwiremutilator: I would love to have it all set up the right way
20:35.01hardwireI can't even grok radius
20:35.04hardwireif I couild
20:35.08hardwireI would have paying subscribers
20:35.21Delta34if u reconfigure iax.conf does it require a restart of service for the changes to take affect or reload
20:35.32hardwiremutilator: brb.. lunch
20:36.06*** join/#asterisk astpbx (~astpbx@d235-143-242.home1.cgocable.net)
20:36.23astpbxshido: you tehre?
20:36.28*** join/#asterisk JohnJacob (~JohnJacob@pcp0011543387pcs.mainf01.in.comcast.net)
20:37.25*** join/#asterisk mithro (~tim@87.76.42.120)
20:37.54gtigeneAnybody else use extensions.ael?
20:39.08DarthClueDelta34: iax2 reload
20:40.26konfuzedjust to be sure if the extensions.conf file   has a space character before   exten=>   would this be considered a typo or cause a problem?
20:41.06Qwellkonfuzed: no
20:41.12konfuzedthanks
20:41.13Qwelloh, before
20:41.14Qwellmaybe
20:41.17konfuzedhmmmmmm
20:41.20*** part/#asterisk xlyz (~xl@213-140-17-96.fastres.net)
20:41.27Qwellthought you meant exten => vs exten=>
20:41.28konfuzedno im back to Konfuzed ;^)
20:41.42mutilator
20:41.59mutilatorpoo
20:42.08astpbxshido: hello?
20:42.18KattyHmmhesays: mew?
20:42.26mutilatormer
20:42.26Hmmhesayshey
20:42.41*** join/#asterisk ropeguru_work (~ropeguru@141.152.37.26)
20:42.43Katty:>
20:43.13Hmmhesaysthe roosters are playing tonight, i'm looking forward to it
20:43.25Hmmhesayseven if my date does ditch me
20:43.35ropeguru_workOk, this may have been asked already, but is there a probblem with the list server for ASterisk??
20:43.37konfuzedwell removing said spaces has not had an impact on my problem
20:43.48mutilator.. that just means you're free for other chics
20:44.03mutilatoror guys.. i dunno what ya prefer
20:44.03Hmmhesaysi'm not on the prowl though
20:44.06gtigene"chics"?
20:44.20ManxPowergtigene: young rooters
20:44.23twisted[asteria]haha
20:44.24ManxPowerroosters
20:44.25mutilatoryea... chics dig roosters
20:44.30Hmmhesaysslang for young females
20:45.00mutilatorslang for available femals
20:45.03gtigenenot very respectful
20:45.06DarthClueropeguru_work: yes there is.
20:45.11ropeguru_workThanks...
20:46.08ropeguru_workdarthclue: was hoping they had not banned me. :-)  Have you also been getting a lot of double posts in your email from the list?? I got one for the pizza party again today and it seemed like it came from an IP address in Italy..
20:46.10mutilator15 minutes
20:46.24Hmmhesaysanyhoo, if she ditches me I'll enjoy the band anyway, cause live music just rocks
20:46.26Hmmhesaysand i'm a dancing fool
20:46.40astpbxanyone know where i can get a canadian did using paypal?
20:46.45mutilators/dancing//
20:47.03Hmmhesaysheh
20:47.11Hmmhesaysi suppose that is the opinion of some
20:47.28tim27any here know Link2VOIP
20:47.31mutilatorgotta love insults in code
20:47.59Hmmhesaysyes, that are both whitty and funny
20:48.06Hmmhesays*they
20:48.27mutilatorwell
20:48.27konfuzedastpbx: hey there
20:48.32mutilatorhm
20:48.36konfuzedI can facilitate that for you
20:48.38astpbxkonfuzed: hey
20:48.42mutilatoranyone ever heard of a company called authentium?
20:48.48astpbxkonfuzed: what you got to offer?
20:49.09konfuzedastpbx: in that I am here in toronto and can provide you a DID through my noc partner
20:49.29astpbxkonfuzed: query me please?
20:50.34astpbxim trying to talk to shido about nufone account I just created, but he isnt answering atm
20:51.31*** part/#asterisk ropeguru_work (~ropeguru@141.152.37.26)
20:51.36astpbxbrb
20:52.46ManxPower~docs
20:52.46jbotmethinks docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
20:52.48ManxPower~mailinglist
20:52.48jbotrumour has it, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
20:52.50ManxPower~fxofxs
20:52.50jbotit has been said that fxofxs is An FXO port expects to receive dialtone and receive ring voltage.  An FXS port expects to provide dialtone and provide ring voltage.
20:53.43mishehu~theanswer
20:55.02astpbxback
20:55.32*** join/#asterisk ignite (ben@65.94.119.7)
20:55.33mutilatorhttp://forums.speedguide.net/image.php?s=ef4cee9d4bfde9adccc978513af4b8e3&u=27930&dateline=1118880846
20:55.36mutilatormakes my eyes hurt
20:57.36DarthCluemutilator: is she coming to cluecon?
20:57.43mutilatori hope not
20:57.44ManxPowerThe single MIS person that's not in vacation opened a trouble ticket complaining that it was too quiet.
20:57.46mutilatori'de run away
20:57.58ManxPowerSo I added a note to the ticket that I would reboot everything to liven things up for her.
20:58.03DarthCluebut we could use her for security.
20:58.07*** part/#asterisk zoo (nobody@ip-132-16.travedsl.de)
21:01.12mutilatorwell
21:01.15mutilatorit's quittin time
21:01.18mutilatoradios muchachos
21:01.26mutilatortil monday!
21:02.38harryvvwe are on a three day holliday weekend
21:03.01mutilatorcelebrating what
21:03.04mutilatorthe first of the month?
21:03.05*** join/#asterisk jeremywhiting (~jeremy@71-37-74-109.slkc.qwest.net)
21:03.12jeremywhitinghi all
21:03.57jeremywhitinganyone know why asterisk would leave a zap channel as OffHook after a call on the channel ends?
21:04.12jeremywhitingdoes that mean asterisk doesn't hang up?
21:04.24Qwelljeremywhiting: fxo?  x100p perhaps?
21:04.43lathos42If I have a call coming into a specific DID, and that call is received by a receptionist, is there a way to treat that call differently between being transferred and having a call from the receptionist?   (ie, we want to block any telemarketers coming in on a certain number, but still want the receptionist to be able to call us)
21:04.54jeremywhitingtdm actually
21:04.59jeremywhitingwith 1.0.9
21:05.46jeremywhitingI usually end up doing a /etc/init.d/zaptel restart to fix it and then asterisk sees the line as OnHook again
21:06.08Qwelljeremywhiting: after every call?
21:06.08KattyHmmhesays: omgdate?
21:06.15KattyHmmhesays: omgwtfbbqdate?
21:06.32jeremywhitingno, just when the boss complains that the lines are all busy
21:07.32Ariel_jeremywhiting, are you using kwel start or loop start?
21:07.42jeremywhitingkewl
21:07.44Ariel_hello Katty hope your doing fine today?
21:08.42jeremywhitingany ideas?
21:08.50Ariel_jeremywhiting, how long does it take before this happens days weeks hours?
21:10.10jeremywhitingor how can I debug this problem, as I'm pretty sure it's my config somewhere
21:10.17jeremywhitingI'm checking
21:13.47*** join/#asterisk oOlli (www-data@baltz-online.de)
21:14.31jeremywhitingif zap show channel 1 says Actual Hookstate: OffHook with my tdm card, does that mean the line actually off the hook?
21:14.53jeremywhitingsomeone said the tdm cards don't have line detection
21:15.04*** join/#asterisk b0ef (~b0ef@062016141085.customer.alfanett.no)
21:15.17oOllihi folks. just a simple question. how do i switch a call from SIP-phone-1 to SIP-phone-2?
21:15.21KattyAriel_: sick :<
21:16.06oOllinot permanently. by click
21:16.06jeremywhitingsorry if my questions are dumb I'm sort of new at this
21:16.06DarthClueKatty: you know that you have to get well by Monday.
21:16.06b0efanyone tried latest cvs and having problems with iaxcomm?
21:16.06Ariel_oOlli, transfer it to the other phone
21:16.09KattyDarthClue: no, i'm going to make you SICK
21:16.18KattyDarthClue: no one has informed me of vegan details yet
21:16.23KattyDarthClue: am i going to starve at cluecon?
21:16.33KattyDarthClue: or just remain sick on junk food?
21:18.59Ariel_jeremywhiting, there have been some problems with some of the boards that need to be reset every week.
21:18.59KattyDarthClue: you don't have to answer. i knwo that's Not Your Department
21:18.59*** join/#asterisk Meaty (~cp_simbul@office.abi.ca)
21:18.59DarthClueKatty: junk food sounds good.  of course, we can just take you shopping.
21:18.59oOlliAriel_ is X-Lite able to do that for me?
21:18.59jeremywhitingoh yeah
21:18.59*** part/#asterisk capouch (501@12.176.248.4)
21:18.59KattyDarthClue: there's only so much french fries, potato chips, and soda you can handle before a vegan requires protein source :<
21:18.59Ariel_oOlli, oh only if you use the # key hack.
21:18.59DarthClueKatty: how about chocolate?
21:18.59KattyDarthClue: most chocolate bars aren't vegan. they've got milk in them
21:18.59oOlliAriel_: and then the extensions.conf will do the rest?
21:19.00KattyDarthClue: byebye
21:19.00Ariel_oOlli, well yes if you set them up that way.
21:19.49oOlliwhat line do I need in my extensions to switch calls?
21:21.42jeremywhitingis there a way to see if a zap channel is in use from the asterisk cli?
21:22.09jeremywhitingI'm sorta doing everything remotely, which isn't helping much
21:23.37astoriajeremywhiting: zap show channel 1
21:23.46astoriajeremywhiting: but use whatever channel you want
21:24.10Ariel_jeremywhiting, show channels
21:24.27Ariel_jeremywhiting, you should be able to do it all remotely.
21:24.30jeremywhitingrigh, so if it says Hookstate: OffHook, then asterisk is holding the line open?
21:24.54Ariel_then do soft hangup zap/?-?
21:25.25Kattyevolution won't run :<
21:26.30jeremywhitingZap/1-1 is not a known channel
21:27.00jeremywhitingfor all variations of {Zz}ap/
21:27.29bkw_if you wanna hang up everything do a stop now
21:27.32bkw_that usually does it
21:27.33bkw_:P
21:27.37bkw_wait it always does it
21:28.01gtigeneDo all the dialplan features in README.ael actually work?
21:28.04*** part/#asterisk pbxbart (user@p54B01C7C.dip0.t-ipconnect.de)
21:28.10jeremywhitingbut something is wron gif it always says OffHook right?
21:28.24Qwellbkw_: /sbin/shutdown -hn now works too
21:28.29jeremywhitingafter any call before I reset it
21:28.33jeremywhitingyeah
21:28.44oOlliAriel_: what line do I need in my extensions to switch calls?
21:30.05jeremywhitingwhen the call ends cli says Hungup 'Zap/1-1'
21:30.13gtigeneblitzrage: are you there?
21:30.25jeremywhitingbut then zap show channel 1 still says OffHook
21:30.49*** join/#asterisk Ahewes (~rsb@209.234.96.194)
21:31.59*** join/#asterisk valence (~valence@Quebec-HSE-ppp230300.qc.sympatico.ca)
21:32.46ManxPowergtigene: If they don't, post a bug report.  They are supposed to work, but there was a bug fixed in CVS a few days ago relating to AEL
21:33.25bkw_AEL is such a hack.. the end result is still a dialplan just like extensions.conf is
21:33.39Darwin35AEL is a bug
21:33.47Ariel_bkw_, great someone agrees with me.
21:34.02Ariel_I don't like AEL..(But I am along in that).
21:34.11bkw_along?
21:34.36Ariel_when I say something about it people say I don't know what I am talking about here.
21:34.46bkw_no you're perfectly clueful
21:35.18bkw_Asterisk - The only Telephony Lego kit with some blocks super glued together!
21:35.46Ariel_super glue I was thinking more like elmers glue
21:36.59*** join/#asterisk santiago (~santiago@63.245.87.180)
21:37.42Darwin35more like rubber cement
21:38.15oOllidoes anyone have a working solution fo email-->fax for me? instead of ast_fax?
21:38.20Darwin35and BKW is the one block that you just dont know where to put it to finish the project
21:38.33bkw_oOlli, its easy to do
21:38.38QwellDarwin35: its the queerly shaped block :D
21:39.02_DAWbkw_ - could you shed some light on how I would go about setting a channel variable from one set in the * db?
21:39.18*** join/#asterisk modem-WWE (~captinmom@70-96-104-242.br1.rmn.wv.frontiernet.net)
21:39.31oOllibkw_: ast_fax is easy? or YOUR idea? let me hear! in time i am looking for a simple solution for both sides FAX <--> EMAIL with chan_capi
21:39.55ManxPoweroOlli: There is no simple solution for email->fax
21:40.01*** join/#asterisk calavera (~none@200.115.174.198)
21:40.08calaveraHi everybody
21:40.16_DAWI am doing this exten => *75,2,Set(COUNT=(DB(${CALLERIDNUM/MAINCOUNT:0:1})
21:40.18bkw_ManxPower, yes their is
21:40.43oOllibkw_: lets hear your idea!
21:40.44xhelioxDoes anyone know where I could get the PSAP # for Seminole County, FL?
21:41.00bkw_first find an easy way to get anything to tiff
21:41.04calaveraI have a problem with asterisk... if Im receiving a call in a zap channel and one of my internal extension is trying to call to the PSTN .. the extension receive the incoming call... this is wrong... how can I fix this?
21:41.06bkw_the rest is just simple
21:41.15*** join/#asterisk Goshen (~Goshen@67-40-107-29.slkc.qwest.net)
21:41.19jeremywhitingManxPower, and oOlli: Hylafax and a fax/modem work great for me
21:41.31oOllibkw_: with ast_fax ?
21:41.35bkw_no
21:41.39Qwellcalavera: You mean, when a call is coming in, and a user answers, you don't want it to answer?
21:41.49calaverano qwell
21:41.49bkw_you can link the hylafax bits into the rx/txfax stuffs
21:41.53bkw_just write a sendfax wrapper
21:42.02ManxPowerjeremywhiting: how do you handle attachment conversion?
21:42.28ManxPowerSIMPLE faxing of tiff files from e-mail is EASY.
21:42.30jeremywhitingcan,t remember I set it up a while ago
21:42.35calaveraif im making a call to the pstn lets say 233xxxxxx and a call in entering from 245xxxxxx my extension receive the call without ringing .. it just appear and say hello
21:42.54Qwellcalavera: with the phone offhook before the call comes in?
21:42.58Qwellerm, onhook
21:43.01Qwellno, offhook
21:43.09calaverathey meet in the zap channelo
21:43.11jeremywhitingactually mine just uses sendfax from within php code on the same server
21:43.24calaveraQwell, what do you suggest
21:43.41Qwellcalavera: Is the receiver in the persons hand, or on the phone when the call comes in?
21:44.20calaverahmm
21:44.27Qwelland are you sure it wasn't just a fluke that they picked it up JUST before it rang?
21:44.31Qwellits not that uncommon...
21:45.06calaveraQwell, let me explain it again....if the pbx is receiving acall on zap channel 1 and Im making a call to other number through teh same zap channel... I receive the incoming call.
21:45.10Qwellwhen I'm talking to my wife on our cellphones, if my signal fades and I call back immediately, her phone thinks the call never dropped
21:45.16calaverawhich is not the same im dialing
21:45.16Qwelland it just resumes the call
21:45.28Qwellyeah, its called a fluke
21:45.36*** join/#asterisk jhava (~icechat5@200.58.26.21)
21:45.40Qwellhappens all the time
21:45.56calaveraQwell,, are u talking to me?
21:46.02Qwell"hmm, no dialtone." "hey bro!" "wtf...dude, I was just about to call you..."
21:46.15QwellI've had that exact conversation countless times
21:46.28jhavash
21:46.41jhavaSorry, hello all....
21:46.43Qwellcalavera: unless you aren't explaining it right...
21:46.50calaverayes... but im not calling the guy Im calling other person
21:46.55Qwellcalavera: it doesn't matter
21:47.05Qwellif a call is coming in, and you pickup the phone...its gonna answer it
21:47.26calaverabut the incoming call is not supposed to ring in my extension
21:47.30jhavaAnybody knows how to stop a SIP call that is on the asterisk, but both endpoints have already hanged up ?
21:47.32calaverastill I get it
21:47.38Qwellcalavera: yeah, see, thats something you should probably mention
21:47.42_DAW_bkw - I am trying to set a channel variable from a entry stored in the * DB.  Is that possible?  I am having no luck at all.
21:48.00calaveraso is it a bug or something?
21:48.14Qwellcalavera: doubt it.  I'd have to see the relevant parts of your dialplan though
21:48.18oOlliManxPower: installed ast_fax. by ast_fax.call says: Channel: CAPI/$[PHONE]. my asterisk-log says: "Unable to request channel CAPI/2849784". help :-)
21:48.29dantcalavera, bob calls you, at the same time, you dial fred, you call goes through, line is picked up, you speak to bob, not fred as bob was already there when you picked ip
21:48.34moyhi :) some time ago this site ( http://kem.p.lodz.pl/~peter/qnet/ )  has some patches for quality of service,  for iptables, iproute2 and linux, but the site does not work anymore, any one know where can i get the patches, the one im looking is called iproute2-2.6.9-041019-2.6.9-qnet1.bz2
21:48.38ManxPoweroOlli: I can't help you with that.
21:48.46Qwelldant: more like
21:48.49calaveraqwell, exactly
21:49.02Qwelldant: bob calls you, at the same time, you dial fred, bob hears your dtmf, and calls you names
21:49.24*** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv)
21:49.26calaveradant and qwell that is what Im talking about
21:49.26Qwell(because Bob is mean...damn Bob.  You were probably calling Fred to complain about Bob anyways.)
21:49.28dantQwell, well yeah, but you'd not hear that as * would mask it nicely :)
21:49.49calaveradant, qwell,,, is there any fix?
21:49.59Qwellcalavera: I'm still not seeing a problem
21:50.00dantcalavera, you ever picked up a normal phone to call someone and found someone already on the line?
21:50.31QwellThings that are supposed to happen - volume 3, chapter 4, page 16, paragraph 2: "When answering a line...the line gets answered."
21:50.32calaverayes
21:50.49calaveraha!
21:50.57jeremywhitinglol you guys crack me up
21:51.21Sedoroxisn't that subsection 8?
21:51.22Sedorox:p
21:51.24calaveraI know what Im going to do
21:51.38calaveraIm going to use a line as main outgoing line
21:51.41dantcalavera, get the telco to block incoming calls, then you won't have any problems
21:51.42QwellSedorox: umm...you must have volume 2
21:51.45calaveraand another as main incoming line
21:51.47Sedoroxooo ok
21:52.10Qwelloh, I see the problem now!
21:52.13QwellYou only have ONE fxo?
21:52.27calaverano we have 3 fx0
21:52.29calaverafxo
21:52.36calaverapots line
21:52.52dantanalogue is a cludge
21:53.04calaveraline 1 for main line, 2 for fax only, and 3 for pstn IVR
21:53.15calaveraim goint to switch line 1 and 3
21:53.28Darwin35BKW why have you not responded to DAW question
21:53.34calaveraso the slot 1 which is the first option when dialing out is the less used
21:53.47calaveramakes sence?
21:53.59Sedoroxif it does to you... sure :p
21:54.02*** join/#asterisk Obad (~eltarzi-a@82.194.47.115)
21:54.10Qwellheh, silly people
21:54.27calaveraany other workarougn
21:54.35calaveraworkaround
21:54.35Qwellcalavera: accept that it might happen once in a while
21:54.45*** part/#asterisk moy (~kvirc@201.135.113.46)
21:54.46calaverahmmm
21:54.53calaverayes it happen very often
21:54.59calaveratoday like 3 times
21:55.02calaverato me only
21:55.04calaverawe are 6
21:55.19calaveraand we are offering the pbx to many companies here in panama
21:55.24Qwellits slightly funky, but its supposed to happen
21:55.28Sedoroxbah.. get a fraction T1... get 10 lines... only have them assign a DID to 5... and then you can use the rest/not used ones for outgoing
21:55.44*** join/#asterisk pbd (~pbdavidso@12.144.118.37)
21:55.48Qwellso wait...
21:55.56QwellYou have ONE line dedicated to an IVR?
21:55.57calaveramake sence sedorox..but it is not the case for everyone
21:56.04pbdOk, is it me, or is digium's listserv messed up today?
21:56.04calaverayes
21:56.04Sedoroxwell yea...
21:56.05Qwellthat means quite a few people will get busy signals trying to call in...
21:56.20Qwellpbd: tis
21:56.45calaverayeahh the 3th line is avaiable
21:56.51calavera90% of the ime
21:56.59Qwell90% isn't very high...
21:57.01twisted[asteria]the 3th/
21:57.14Qwelltwisted[asteria]: I would have said it, but I'm being nice today
21:57.34enderwhen calling international from the US, is it allways 12 digits after the 011 ?
21:57.40twisted[asteria]11.
21:57.40Sedoroxwha? there's nothing wrong with being 3th Ti :p
21:57.48twisted[asteria]011+1+NPA+NXX+XXXX
21:57.52Sedorox(if you have a lisp that is...)
21:57.53calaverathx for all your help folks
21:58.03calaveranow I understand it may happen
21:58.11*** join/#asterisk drooth (~drooth@ip68-111-235-172.sd.sd.cox.net)
21:58.26twisted[asteria]oh wait
21:58.26Qwellcalavera: Sedorox was right.  You should get a fractional T1 or something...especially if you have 6 people there
21:58.30calaveraQwell waht is wrong on having a dedicated line for ivr?
21:58.31twisted[asteria]when calling FROM the US
21:58.35Qwellsurely, more then one person will want to use the phone at once
21:58.41twisted[asteria]when calling FROM the us, it can be any number of digits after the country code
21:58.47endertwisted[asteria]: hrm, odd.  I was given 2 numbers where are 011 XXX XXX XXX XXX and 011 XX XX XXXX XXXX
21:58.51Qwellfrom 1 to 500
21:58.52enderboth being 12 digits.
21:58.56SedoroxQwell: well he did agree.. but he said with other people... so I guess he's just testing different situations.... *looks at calavera*
21:58.58sedwards50I need help configuring a Adtran 750 talking to a t100p -- the fxo lines won't hang up!
21:59.00enderneither starting w/ 011 1
21:59.04Qwellender: different countries have different lengths
21:59.16calaveraexactly sedorox
21:59.19Qwellsome has as few as 5-6, right?
21:59.21*** join/#asterisk mover (~dlu@gw-dus-net.dus.de.ncore.net)
21:59.32moverhi
21:59.33twisted[asteria]ender, read after that line
22:00.03twisted[asteria]ender, i thought you said TO the us, if FROM the us it can be any length after the 011, depends on the country code and the number of digits dialed within the country
22:00.19enderoh I see.
22:00.19Sedoroxcalavera: I don't see much of a point of that kinda setup.. I mean if you need a lotta lines... and you don't wanna spend for a T1.... then get a FXS channel bank and get a T100P and get it where it'll use the non-used lines for outgoing....
22:00.30enderhrm.
22:00.48Sedoroxbut yea.. with 3 lines... and 6 people... and fax and ivr... yea.. your gonna get overlap
22:00.51endertwisted[asteria]: how best to handle allowing for international calling within a dial plan?  Do I have to list every single digit length possible?
22:01.05twisted[asteria]ender, no, i use _011X.,1,blah()
22:01.07Sedorox_011.
22:01.11Sedorox:p
22:01.15ManxPowerender: exten => _9011XXXXXX.
22:01.17ManxPowernotice the .
22:01.20enderright.
22:01.27ender. matches until the end no?
22:01.28moveri have a strange behavier with snom 190 phones on * head and MWI. Any one are able to help me out? I havre read tons of doc and the source but no idea :-(
22:01.35Sedorox<PROTECTED>
22:01.36Sedorox:p
22:01.41Sedoroxits the complete opp
22:01.47twisted[asteria]the '.' waits for the end of dialing
22:02.01Sedoroxor the digit timeout?
22:02.20twisted[asteria]Sedorox, yeah, but the digit timeout tells it that it's done dialing ;)
22:02.27Sedoroxwell yea :p
22:02.30Qwellforced end of dialing :p
22:02.31*** part/#asterisk mkrufky (~mk@68.160.103.77)
22:02.34Sedoroxheheh
22:02.41Qwelllike a bartender cutting you off
22:02.47Sedoroxlol
22:02.50Sedoroxyour flagged! :p
22:03.00twisted[asteria]Qwell, not as abrupt as beating the user with a pipe after they've dialed so many numbers, but yea
22:03.07SedoroxLOL
22:03.10Qwelltwisted[asteria]: depends on the user
22:03.17twisted[asteria]lol
22:03.55enderthanks.
22:04.09file[laptop]29-Jul-2005 5:30 AM adjustment non financial transaction -$0.25
22:04.15file[laptop]non financial transaction?!?
22:04.22QwellThey gave you money?
22:04.28file[laptop]no, they took money
22:04.36enderfor a non financial?
22:04.40file[laptop]apparently
22:04.45file[laptop]I wasn't even awake at 5:30
22:04.46enderis there a 'fee' on all non-financial transactions?
22:04.50*** join/#asterisk xkev (~kevin@orbit.xmission.com)
22:04.52file[laptop]I've never seen this before
22:05.18file[laptop]well I'm not concerned
22:05.23file[laptop]I'm getting data for free :P
22:05.29Sedoroxlol
22:05.30Sedoroxy?
22:05.38file[laptop]I found a flaw in their system
22:05.38endermmm, wish I was.
22:05.41Sedoroxlol
22:05.46*** join/#asterisk JoiIto (~jito@JoiIto.silver.supporter.pdpc)
22:05.47Sedoroxnextel by any chance? :p
22:05.55file[laptop]Telus Mobility
22:06.06Sedoroxah.. but not Telus Mike huh? :p
22:06.13SedoroxI like the treo....
22:06.15xkevI'm using * as a media gateway, when I Dial() my registrar proxy, I want to catch the reply code (e.g. 404).  Is this possible (asterisk internals say 'circuit-busy', but ${DIALSTATUS} = CONGESTION is too vague)
22:06.18file[laptop]my WAP account has full data access without going through the WAP proxy
22:06.19file[laptop]:)
22:06.23Sedoroxbut I got nextel.. so HOPEFULLY when they merge with sprint
22:06.24enderpbd: I had a 650 at my previous job, where they paid the bill.  I can't really afford to buy one for myself.
22:06.29Sedoroxinteresting
22:06.48xkev..like I want to then hang up the pri with cause 1 (not in service) on 404 or somethign
22:07.01enderpbd: come on... you'd turn down a free Treo?
22:07.20Sedoroxyea.. most companies let you keep the phone :p
22:07.54*** join/#asterisk Twister (Twister@24-179-88-187.dhcp.chtn.wv.charter.com)
22:08.05pbdI'd turn down a free treo, yes.  It would hurt, though.
22:08.11*** join/#asterisk Obad (~eltarzi-a@82.194.47.115)
22:08.19fileI bought my Audiovox
22:08.20fileon eBay.
22:08.25fileand then got the unlock code from Telus
22:08.27oOlliany ideas why txfax does not accept this channel?? --> http://pastebin.com/325001
22:08.33SedoroxI don't like the audiovox's
22:08.43Sedorox<---- moto fan
22:08.57fileI like mine... it's big, but it can do a lot of stuff
22:09.04filehttp://www.telusmobility.com/nb/pcs/handset_audiovox_ppc6600.shtml
22:09.04Sedoroxhehe
22:09.09SedoroxI got a i860 at the moment
22:09.25*** part/#asterisk wrmem (~monnin@monnin-win.cso.uiuc.edu)
22:09.32Sedoroxew.. windows mobile..
22:09.33Sedorox:p
22:09.43fileI like it
22:09.57Sedoroxthe new i930 thats suppose to be coming out for nextel is windows mobile
22:10.03ObadCould someone point me to where I can get some help with Asterisk@home please ?
22:10.07*** join/#asterisk JoiIto (~jito@JoiIto.silver.supporter.pdpc)
22:10.31oOlliObad: what do you want to know?
22:10.32Twisterim trying to update to cvshead, im being told i need a new libpri, ive gotten the cvshead version and make && make install then i went back to the asterisk src dir, did make clean then make and it still errors out on compiling chan_zap.c telling me i need a new libpri..
22:10.45xkevfile, I hear the speaker on those is poor
22:11.05fileI have a bluetooth headset
22:11.14filethe speakerphone though, yeah, it's weak
22:11.28enderfile: does your audiovox do ssh?
22:11.31bsdfreakheh
22:11.36Obadhmm.. everything I guess, but I'm having trouble setting a DID (from iConnectHere) to be received by A@H. I also had some trouble setting up a Polycom phone (but got over most of that).
22:11.38fileender: it can sure
22:11.39*** join/#asterisk SarahEmm (~sarahemm_@2.35.220-216.q9.net)
22:11.43enderI'm trying to decide on a phone we can use as an IT pager so we can get 'pages' and ssh into systems to fix them.
22:11.46filemy provider gives me a public IP too ;)
22:11.47SarahEmmhihi
22:11.54Twisterkitrich!
22:11.54filethe keypad takes some getting used to
22:11.58SarahEmmtwistyone!
22:11.59enderfile: linky to audiovox phone?
22:12.00SarahEmm*snugglez*
22:12.02enderfile: oh.....
22:12.10fileender: http://www.telusmobility.com/nb/pcs/handset_audiovox_ppc6600.shtml
22:12.14Twistermy precious
22:13.07enderooooh it's a windows phone.
22:13.21Sedoroxnokia's suck even worse
22:13.46SarahEmmTwister: how's you?
22:13.52Twisterim hot
22:13.54Sedoroxbbl
22:13.57TwisterSedorox
22:14.00Twistergrr
22:14.09Twister*hole
22:14.12Twisterlol
22:14.25Twistermake a comment like that and run like a lil wimp :P
22:14.56Twisterive had better signal from my nokia than any phone ive ever had
22:15.39endersorry, I can't usea  windows phone.
22:15.57enderGuess my best bet is still the Treo.
22:17.09ObadoOlli: Is there a list discussing Asterisk@home ? I feel like I'm intruding (just not used to this IRC business)
22:18.07*** join/#asterisk cfrank (~cfrank@bi01p1.co.us.ibm.com)
22:21.05*** join/#asterisk scud (~scud@12-214-190-139.client.mchsi.com)
22:21.50hardwireblah
22:21.53hardwireyear of the kick my ass
22:26.19*** part/#asterisk Obad (~eltarzi-a@82.194.47.115)
22:26.57*** join/#asterisk wunderkin (kev@12-215-218-160.client.mchsi.com)
22:27.56hardwireperson asks.. do you want a sales position with our company.. cause we have one of those
22:27.57hardwireheh
22:28.09*** join/#asterisk Obad (~eltarzi-a@82.194.47.115)
22:28.10*** join/#asterisk Netgeeks (~Chris@68-185-24-2.static.mdfd.or.charter.com)
22:28.11hardwireconcidering the first few minutes of the converastion I couldn't even tell who was calling me
22:28.20hardwireI am gonna say no.. due to poor people skills
22:28.46harryvvhe is confused
22:28.57hardwireheh
22:28.58hardwireman cups
22:29.23harryvvhardwire, that sat system, how many calls do you think it can handle at one time?
22:29.30hardwire2
22:29.34hardwirestarband is very limited
22:29.39hardwireif it were another service provider
22:29.41hardwiremuch much more
22:29.41harryvvfricken mmm
22:29.45hardwireI would try to get 256/256
22:29.49hardwirefricken mmm?
22:29.52hardwiregot you some chicken?
22:29.53harryvvyea
22:30.02harryvvthat sucks
22:30.09harryvvand how much for the bandwith?
22:30.36Darwin35who did it
22:30.53*** part/#asterisk Obad (~eltarzi-a@82.194.47.115)
22:31.16hardwirehardwire: $750 a year
22:31.25hardwirefor the 256/256 it would be like.. $1500/mo
22:31.34hardwireso I might as well install more starband modems :)
22:31.44hardwirecause it hass less reoccuring costs :)
22:31.46hardwiredramatically less
22:31.52hardwirefor $1500 I could buy a new modem and service
22:31.58hardwirefor another $1000 a new dish and amp
22:31.58hardwireso
22:31.59hardwireheh
22:32.20hardwirewe talked with starband engineering.. best they could do was move us over to a newer cluster
22:32.22hardwirewith less people on them
22:32.29hardwirethey need a VIP cluster..
22:32.31harryvvi see
22:32.40Beirdowow
22:32.44hardwireharryvv: whywhatsup?
22:32.48harryvvnot much
22:32.51Beirdowho took the pic on the taug page, I wonder?
22:32.53harryvvcuros on cost
22:32.56harryvvcurios
22:32.58Beirdothe night cityscape...
22:33.03hardwiretaug?
22:33.06*** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk)
22:33.10Beirdowww.taug.ca
22:33.15Twisterim trying to update to cvshead, im being told i need a new libpri, ive gotten the cvshead version and make && make install then i went back to the asterisk src dir, did make clean then make and it still errors out on compiling chan_zap.c telling me i need a new libpri..
22:33.19BeirdoToronto Asterisk Users Group
22:33.28hardwirelike I would just know that :)
22:33.38Beirdoheh
22:33.43Beirdowell, SOME people here do
22:33.47hardwireyeh
22:33.48hardwirewell
22:33.53hardwiredon't make me feel so unspecial
22:33.55hardwireI need to feel special
22:33.57Beirdohehe
22:34.08Beirdooh, I'm sure you're VERY special
22:34.10Beirdowe all are
22:34.44Beirdowell, if I wanna go to that social, I should likely head for the subway soon
22:36.58Beirdoanyone else here going?
22:37.06rikstahey Beirdo
22:37.07BeirdoSarahEmm for instance?
22:37.20Beirdoheya, riksta, how's it goin?
22:37.26rikstaBeirdo: good thanks!!
22:37.35rikstaand you?
22:37.36Beirdocool
22:37.40Beirdonot bad at all
22:37.46rikstasweet :)
22:37.49Beirdobeen using asterisk altogether too much
22:37.53rikstaam just writing some AGI
22:37.55Beirdoaveraging 4h/day
22:37.57Beirdohehe
22:37.59rikstajeez
22:38.03Beirdojust for me
22:38.42Beirdoand since she's out with friends, I'm thinking of going to the asterisk user's group social tonight
22:38.46Beirdohehe
22:38.57rikstahow many ppl are at ure asterisk group
22:39.08rikstai wouldn't have thought it was popular enuff to have a group, thats cool
22:39.20Beirdodunno, have never managed to make it
22:39.37harryvvwe dont have one in vancouver
22:39.47*** join/#asterisk znoG (~gs@200.115.218.81)
22:40.24rikstacan someone tell me, as i've never had to think about doing anything like this before, if a telecoms company has a proper switch, what equipment would i need to hook directly into it
22:40.29rikstajust some kinda E1 card?
22:40.36DarthClue~cluecon
22:40.37jbotwell, cluecon is http://www.cluecon.com - The Open Source Telephony Expo and Developers Conference featuring presentations by Mark Spencer, Greg Boehnlein, Ken Rice, Brian West, Vikrant Mathur, Craig Southeren, David Sugar, Bob Andreasen, Joshua Colp, Brian Fertig, Peter Nixon, Marc Olivier Chouinard, and Anthony Minessale II.
22:40.50Twisterriksta: what kinda connection are you wanting to it?
22:41.00Twistera t1/isdn or pots?
22:41.35rikstaTwister: it''s gonna be in the same place as the switch, so i thought i could do something else than that?
22:41.40rikstalike some direct cable
22:41.49rikstai have absolutely no idea about this kinda hardware
22:41.52Twisterumm...
22:42.26konfuzedcould someone confirm for me that if both the providers sip.conf and my sip.conf have set type=peer then this completely bypasses authentication attempts ??
22:42.28*** join/#asterisk Netgeeks_ (~Chris@68-185-24-2.static.mdfd.or.charter.com)
22:42.33Twisteras the telco's switch??
22:42.33rikstai don't even know exactly what im talking about
22:42.38rikstaTwister: yeah
22:43.02*** join/#asterisk dalabera (~Dalabera@pmr.pmrtechnologies.com)
22:43.04DarthClueriksta: that depends on what the telco has for you to use.  it is either gonna be an e1/t1 card or maybe a direct connection via an ethernet side if they equipment capable of doing the translation.
22:43.07rikstamy friend has a normal telco, i want to get asterisk in there so i can run these AGIs
22:43.19rikstaDarthClue: e1, i thought as much
22:43.27rikstaso i just buy sum kinda pci card right?
22:43.41DarthClueriksta: yup, either sangoma or digium.
22:43.49mover<PROTECTED>
22:43.49mover<PROTECTED>
22:43.49mover<PROTECTED>
22:43.57rikstaDarthClue: thanks dude
22:44.00Twisterhttp://www.digium.com/index.php?menu=product_detail&category=hardware&product=TE210P
22:44.07Twisterriksta: theres a link
22:44.25rikstawhy thanks :)
22:44.31konfuzedmover: just gave me a possible clue
22:44.36DarthClueriksta: http://sangoma.com/products/p_voice-data.htm
22:44.37rikstaapprox how much
22:44.53rikstai'd prob buy from digium to support them
22:45.14BeirdoI'd probably go with what is available at the right price and locally
22:45.21Beirdowhatever that turned out to be
22:45.34DarthCluekonfuzed: that depends on entirely on how the calls are connected.
22:45.43Netgeeks_both digium and sangoma offer great support, so no worries there
22:45.55DarthCluei would probably buy whichever one answered the phone first.
22:46.08bsdfreakhi
22:46.34konfuzedDarthClue: regarding the authentication right?
22:46.40konfuzedlet me put it another way
22:46.46DarthCluekonfuzed: yes
22:47.29konfuzedTo bypass authentication all together what else has to be done besides both asterisk boxes having the sip.conf context set to type=peer?
22:48.10DarthCluekonfuzed: how do you dial the other box?
22:48.20DarthCluekonfuzed: and how does the other box dial you?
22:49.32konfuzedwell a DID is setup on the providers box. then it forwards a sip call to my box via ip
22:50.11konfuzedso I dial regular bell phone number -> providers setup -> sip.conf to send it to me via ip
22:51.16konfuzedis that a complete connect picture or is something missing?
22:52.57DarthCluekonfuzed: you would have to configure it and then test it to verify the config.  there are some issues that have to be addressed or it won't work and i don't recall exactly what they are at the moment.
22:52.59*** join/#asterisk jsaunders (jsaunders@70.70.75.206)
22:53.44konfuzedI would like to say that the term "provider" is more generic like the source of the account and asterisk operator as opposed to big company with piles of staff
22:54.08konfuzedintereseting thank yo ufor that info
22:54.32konfuzedin other words its doable but the fine details must be just right
22:55.28DarthCluekonfuzed: yep.
23:02.59hardwireok
23:03.00hardwire4
23:03.01hardwiredigits
23:03.03hardwireits just going to happen
23:03.04konfuzedthat's it(the possible clue I had from mover)  - if   auth=md5      does that require you to use   md5secret:   or do away with needing to use md5secret:   and there by set all secret:   entries to use md5 by default???
23:03.07hardwirefirst digit I think will be company
23:03.20hardwireI hate that.. but it will have to do
23:03.25hardwirethe second will be location/department
23:03.46hardwireit seems odd
23:03.50hardwire11xx being anchorage
23:03.53hardwire12xx being another town
23:04.06hardwireor 11xx-14xx being anchorage w/ departments
23:04.21hardwireand 15xx-17xx being another town w/ departments
23:04.28hardwirethen 18xx-19xx being tenants
23:05.06hardwirethe DID's will just have to be assigned as I put people in the dialplan I guess
23:05.12hardwirejesus my dialplan is going to be large
23:05.34hardwireI hate this.. this isn't a problem I want.
23:07.58Netgeeks_what exactly are you trying to do hardwire?  a multi-site business pbx?
23:08.04hardwireyes
23:09.25Netgeeks_you thought about using dundi to make things easier?
23:09.26file[laptop]konfuzed: auth=md5 just tells Asterisk to use MD5 authentication with the other side
23:10.14konfuzedso when does md5secret come into use?
23:10.27hardwireNetgeeks_: no
23:10.30hardwire:)
23:10.42hardwiredoes 1.0.9 support it?
23:10.50Netgeeks_you could use dundi internally to make your routing of internal extensions much easier
23:10.55hardwireand that still doesn't help me just plain needing to get a dialplan working
23:11.10hardwireNetgeeks_: that seems fine.. I was just going to take a block and route it
23:11.17hardwiredundi is a more ad-hoc way of doing that I take it?
23:11.22*** join/#asterisk emp (~emp@70.57.239.37)
23:11.24hardwiresharing dialplan contexts?
23:11.45hardwireas well as establishing routes?
23:12.02hardwireI just need to get a rule of thumb dialplan schematic layed out before hand
23:12.14Netgeeks_yes, you would use it to advertise routes amongst the servers
23:12.19hardwirethis would be easier if everythign were in base8 :)
23:12.35konfuzedDarthClue: perhaps you can look at my conf file paste at http://pastebin.ca/18817
23:12.57konfuzedI have listed some of the "providers" configs aswell
23:14.01*** join/#asterisk mrtwister (~mrtwister@bc9a1976e1c792b8.session.tor)
23:14.08*** join/#asterisk colinm_ (~colol@VDSL-130-13-9-155.PHNX.QWEST.NET)
23:14.43moveris there a reason why callergroup and pickupgroup # only from 0 to 63?
23:15.30moveri need thousands of caller and pickup groups. is this possible?
23:19.07konfuzedok so I actually had this working before LiveVoIP went bankrupt. I had one server at the time but it has been reinstalled fully fresh installed from scratch and now having copied the extensions.con and sip.conf and iax.conf, modules.conf and queus.conf and voicemail.conf
23:20.13konfuzedit seems to run fine but the added config for new provider still won't connect.
23:20.13harryvvohh livevoip went bankrupt interesting.
23:20.24SarahEmmmover: why thousands?
23:20.30konfuzedat least thats what there website says
23:20.49konfuzedI wonder if it s really creative restructuring to a new name
23:21.54*** join/#asterisk Inv_arp (junya@adsl-11-73-192.mia.bellsouth.net)
23:22.01konfuzedOnce I have the incoming call working I can then look at making these other 2 asterisk servers act as peers to the first one
23:22.31konfuzedthen we can leisurely work out the complexities with out pressure
23:23.00Inv_arphmm... need an incoming DID provider besides BV (ulaw only)... who else is ok these days?
23:23.31konfuzedthen bouncing our brains around making that setup work should eliminate most of my config konfuzions
23:24.21konfuzedits like forced learning kurve
23:24.37konfuzedapparently somebody thinks that is why I was born
23:26.32*** join/#asterisk mago1-cn (~maxglucks@200.109.166.83)
23:27.10mago1-cnHi! Can anyone tell me please how to verify how many licenses I have installed in Asterisk?
23:27.58harryvvmago, you a green bannana?
23:28.18konfuzedwhat kind of licenses do you have?
23:28.22PatrickDKI do, I do
23:28.26mago1-cnDeginitely!
23:29.02mago1-cnDefinitely sorry...
23:29.47konfuzedmago1-cn: possibly the more you know the more sorry you will be.
23:29.53konfuzedUntil you have mastered it all
23:30.06konfuzedlike puzled has ;^)
23:31.44konfuzedperhaps  someone might be able to look at thesemy conf file paste at http://pastebin.ca/18817
23:31.49konfuzedooops
23:31.52mago1-cnSo, anyone will please? Is just that I'm trying to record a call coming from the Internet in g729, an then I get a message saying no g729 channel are available, but I have 2 licenses an I only see one channel in g729 working under the sip show channels command...
23:31.52konfuzedwasnt done
23:32.23konfuzedwhat kind of licenses do you have?
23:32.38konfuzedfrom where?
23:32.39mago1-cnthe $10 g729 channel
23:32.43mago1-cndigium directly
23:33.14mago1-cnthe call was done, but only one party heard
23:33.42konfuzedperhaps  someone might be able to look at these conf files pasted at http://pastebin.ca/18817  there is info for both the providers config and my config
23:34.16Qwellkonfuzed: You really should have masked your password
23:34.28konfuzedshite I thought I got all those  with   ********
23:34.30konfuzedoh well
23:34.36Qwellyou missed one or two
23:34.36*** join/#asterisk santiago (~santiago@63.245.87.180)
23:34.38konfuzedI'll have it changed anyway
23:34.41*** join/#asterisk meshuga (meshuga@neldor.com)
23:34.47Qwellbut, whatever
23:35.12Qwelland also pastebin some CLI output from when the problem occurs
23:35.18konfuzedQwell: I didn't mena to p[ut it there really but I was curious if maybe someone else might try to connect
23:35.26hardwireheh
23:35.30konfuzedoh I suppose the host=  part may prove to be a problem for that
23:35.36hardwirethis stream I am listening to is making my speakers do very wrong things
23:35.40hardwirewhich is efecting my CRT's
23:35.48Qwellhardwire: wtf?  link?
23:36.02hardwireits just an mp3 stream
23:36.10hardwirehttp://www.somafm.com/groovesalad.pls
23:36.27hardwireweird looooow digerydoo sound
23:36.33konfuzedhmmmmm  emf leakage from wiring or speakers beside the monitor?
23:36.35hardwirecow in heat sound
23:36.51*** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net)
23:36.56hardwirekonfuzed: no the speakers are right next to the emmiter for the tube
23:37.08hardwireat the base of the CRT area
23:37.08konfuzedquick seperate that equipement before it blows ;^)
23:37.12hardwirenaw
23:37.19hardwireits just making all vibrant colors shake a lot
23:37.23hardwirebut its over now
23:37.29hardwireI turned down the speakers and all is good
23:37.31Ariel_hello everyone
23:37.34hardwirehi Ariel_
23:37.38Qwellis the part still going that was doing it?
23:37.42Qwellbecause I'm not getting it, heh
23:37.44konfuzedoh gotta love that EMF
23:37.55hardwireQwell: see.. maybe it has to do with my speakers being where they are
23:38.00hardwireand my monitors being shitty
23:38.02Qwellmine are in the same place
23:38.07Qwelland my monitor is shitty too :p
23:38.17hardwirewell sorry I couldn't help make your life funner today :)
23:38.20SarahEmmbut i'm at work
23:38.41konfuzedQwell: did that cofig paste indicate anything besides my password?
23:38.48Qwellkonfuzed: username
23:39.05konfuzedthat should be profx
23:39.17QwellWhat does the CLI say?
23:39.42Qwelland what is the actual problem?
23:39.44mago1-cnkonfuzed, would you help please, i don't think i'll receive support from digium until monday...
23:40.21konfuzedright after I run asterisk I get CLI> Jul 29 19:55:35 WARNING[3528]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 6be3946d5cc35d492462b3003e9dda52@192.168.1.136 for seqno 102 (
23:40.26hardwiresomebody in seattle needs to send me a keg
23:40.31hardwireso I can be happy
23:40.47Qwellkonfuzed: You have the type=friend in your sip.conf?
23:40.58konfuzedthen when I turn on sip debug and call in ..............................
23:41.26konfuzedthe [profx]  copntext is identical to what the "provider" config has
23:41.44Qwellwell, your host= line is wrong.  its pointing to yourself
23:41.47Qwelltry host=66.96.30.25
23:41.49konfuzedhe has indicated that this is usually all that it takes
23:42.54konfuzedi really have to kick myself on that one.   I'd swear I had that
23:43.06konfuzedi'll change it before pasting the  sip debug
23:43.46Qwelland type=friend is horrible for testing
23:44.55*** join/#asterisk optimator (~opti@130.130.217.216.transedge.com)
23:45.13*** join/#asterisk cgcorea (~cgcorea@63.245.14.194)
23:45.43optimatoranyone have good suggestion for tracing down a PSTN line crossover on a 66 block?  no visible signs, very frustrating...
23:46.12harryvvhand held phone tester
23:46.13harryvv:)
23:46.20SarahEmmoptimator: tone it
23:46.23Ariel_optimator, it's called a butt set
23:46.26harryvvinjext a tone into it
23:46.31harryvvinject
23:46.31SarahEmmoptimator: or use a butt set
23:46.41optimatorbutt set? ok... ill look that up
23:46.50Ariel_when you have a xo the sound travels between them as well.
23:47.12Ariel_xo = Crossover
23:47.25optimatorhehe yeah i noticed ;)
23:47.50optimatorhmm..  gah..  these old blocks are hell
23:47.59SarahEmmheh
23:48.09*** part/#asterisk nettie (~nettie@213-140-17-96.fastres.net)
23:48.11harryvvAriel_ I was watching two contracted out techs on a mini phone valt trying to figure out what goes to where. Telus being the main telephone company is on strike ;)
23:48.53*** join/#asterisk iswm (iswm@iswm.user)
23:49.01SarahEmmheh
23:49.03konfuzedso http://pastebin.ca/18818  has the sip debug
23:49.04SarahEmmyes, that they are.
23:49.05SarahEmmugh.
23:49.08Ariel_harryvv, and did the find what they were looking for? or was it a case of the who's on first
23:49.23harryvvthay looked alot bussier then telus
23:49.24harryvv:)
23:49.56harryvvWe have two big strikes here. 1,000 independent truckers are on strike and both main shipping ports are closed down.
23:50.11harryvvWalmart is in the process of suing them...BoooHoo
23:50.38harryvvits costing vancouver 30 million dollars a day in losses.
23:50.40Netgeeks_optimator:  this would work for what you need... http://www.twacomm.com/Catalog/Model_26000-900.htm?sid=1CPKV7M5EGV38GQHMVB6FXN213J330H2
23:50.41*** join/#asterisk iq (~iq@204-26-74-86.omah.qwest.net)
23:50.42Ariel_well I (I know some don't agree) don't like unions.
23:50.44iqhi
23:51.09harryvvthe truckers were brining hom 50 dollars a day after fuel cost and running there trucks.
23:51.30harryvvmostly the fuel cost and the fact those that pay them would not pay them anymore
23:52.02harryvvso imagine saying to your familly, I am now being payed 50 dollars a day in wages.
23:52.13Ariel_harryvv, well that is where you need to re-do your contract ahead of time for problems like fuel charges. That is why you see a sur-charge on our ups bill.
23:52.35harryvvopec is costing everyone
23:52.39harryvvits OPEC
23:52.55Ariel_harryvv, no it's not only opec but the world in general
23:53.00Inv_arpAriel_: sup man... still using VP
23:53.01harryvvalberta oil is ties to opec cost
23:53.15*** join/#asterisk meppl (~mephisto@87.193.3.25)
23:53.18harryvvand thats rediculios. Canada does not import oil.
23:53.26Ariel_Inv_arp, yes and due to allot of my customers have bv I am also using them now. argh I hate bv
23:53.51Inv_arpAriel_: BV still only have ulaw?
23:53.52Ariel_harryvv, there all tied. But it's called politic's.
23:53.57harryvvyea
23:53.58Ariel_Inv_arp, yes
23:54.08harryvvif i had a diesel truck..i would go biodiesel
23:54.33Ariel_besides the us also has enough oil in the ground but due to the enviormentalist it's going to stay there.
23:54.35Inv_arpbah
23:55.24PatrickDKbiodiesel isn't all that great
23:55.53harryvvactually, i would burn pre filtered and dryed waste oil
23:55.54*** join/#asterisk galel (~galel@63.245.8.164)
23:55.59galelhello
23:56.20galelis there anyone from digium?
23:56.27Ariel_well I hope that the cost really gets high enough so we can back to making newer better engines like the one that runs on water.
23:56.44Ariel_galel, if you have a question ask away.
23:56.56Ariel_I am not from digium.
23:57.12galeli got problems with iaxy
23:57.31Darwin35burn it
23:57.35galelok
23:57.45galelgood answer
23:58.30galeli provisioning the iaxy with 2 servers one inside my lan and other from internet
23:59.09galeland in the lan the iaxy connects wright but in the internet don't
23:59.56galelit's a special configuration on my asterisk?

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