irclog2html for #asterisk on 20051013

00:00.15Vlat-if they're now wanting to do - we use STUN
00:00.28epabloOK
00:01.14epabloHas anyone used EAGI with perl.  I need to access that FD 3 to send audio, but I don't know how
00:01.47Vlat-clyrrad: 're you here ?
00:03.53*** join/#asterisk Delta34 (n=delta34o@198.87.24.253)
00:04.09*** join/#asterisk _DAW (n=bob@adsl-150-43-153.msy.bellsouth.net)
00:05.42*** join/#asterisk file[laptop] (n=jcolp@142.166.94.161)
00:05.55*** join/#asterisk tris (i=tristan@camel.ethereal.net)
00:06.23*** join/#asterisk dan__t (n=dant@ip70-176-120-15.ph.ph.cox.net)
00:06.27*** part/#asterisk dan__t (n=dant@ip70-176-120-15.ph.ph.cox.net)
00:06.53*** join/#asterisk dan__t (n=dant@ip70-176-120-15.ph.ph.cox.net)
00:07.01Delta34hi all, i been trying to answer for this but no luck, is their a way to display the called party's name to the calling party, say if i dailed 4000 it will tell me that i dialed the operator
00:07.07dan__tbah#*@!&($#!$#*@
00:07.23Delta34this is using cisco 7960 phones and asterisk
00:07.38Delta34i read its a limitaion of sip
00:08.12Vlat-Delta34: CallerID is evil
00:08.52Vlat-usually it doesn't work...unless you're in US, and have got the perfecit gatway chain
00:09.21Delta34no its for internal users on the asterisk box
00:09.30syle2i bought one of those rhino channel banks, anyone played with them before?
00:09.32*** join/#asterisk cio (n=na@adsl-072-149-159-016.sip.bhm.bellsouth.net)
00:09.37Delta34old analog pbx has this feature
00:09.52leszqbye
00:09.54*** part/#asterisk leszq (n=leszq@82.177.97.254)
00:09.54cioJust joined - What feature Delta34?
00:10.38Delta34so if i dialed ext 4000, on my cisco 7960 phone, it will tell me i dialed the opeartor at ext 4000
00:10.39MikeJ[Laptop]Delta34, yes, you can do that
00:10.47Delta34how?
00:11.01MikeJ[Laptop]wait..on the dialing phone or the called phone?
00:11.20Delta34on the dialing phone, i know callerid shows up on the called phone
00:11.28*** join/#asterisk kingtux (n=kingtux@pool-151-196-44-240.balt.east.verizon.net)
00:11.30MikeJ[Laptop]it works now to show the called phone whatever you want, for example if it is a que call, what que it is coming from
00:11.35kingtuxhell all
00:11.40kingtuxhello
00:11.47*** join/#asterisk supaigtr (n=yurplsl@152.53.17.1)
00:11.58MikeJ[Laptop]on the 7960's I beleive you can send xml stuff back to the screen, I do not know if that is implemented anywhere
00:12.46supaigtrIs there a way to disable a zaptel card and use asterisk with card still intact?  I'm trying to see if my audio problems are card.
00:12.59Delta34dont think u can send xml stuff that way, the xml stuff is for directories or services features
00:13.11kingtuxhaving trouble with ztdummy mod....I'm getting this error when I try to modprobe it. I have no digium cards just VOIP....FATAL: Error inserting ztdummy (/lib/modules/2.6.9-11.EL/misc/ztdummy.ko): Unknown symbol in module, or unknown parameter (see dmesg)
00:13.11kingtuxFATAL: Error running install command for ztdummy
00:18.14supaigtrkingtux: is the module installed in the modules directory for the running kernel?
00:19.03cioOn debian, I should be able to compile cvs with just my kernel-headers, right?  I shouldn't have to use the entire source tree?!
00:20.14*** join/#asterisk coppice (n=chatzill@48.201.17.210.dyn.pacific.net.hk)
00:20.55*** join/#asterisk drumkilla_laptop (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
00:20.55*** mode/#asterisk [+o drumkilla_laptop] by ChanServ
00:21.30marc324how do you force * to use thr db instead of conf files
00:22.43*** join/#asterisk wolfson` (n=hehe@usr-kdh-208-6-58-26.beachlink.com)
00:23.30kingtuxyes
00:23.42kingtuxi see it there
00:23.52kingtuxztdummy.ko
00:24.19*** join/#asterisk Math` (n=math@modemcable231.182-70-69.mc.videotron.ca)
00:25.10supaigtrKernel compiled with symbols?
00:29.09*** join/#asterisk simprix (n=simprix@24-231-248-225.static.aldl.mi.charter.com)
00:29.27simprixDoes anyone have experience with voicepulse connect ?
00:31.29brookshire[home]i do!
00:31.55fileMattttttt
00:33.28*** join/#asterisk Godsey (i=lanny@pdpc/supporter/sustaining/Godsey)
00:33.51cioHow stable is the 1.2beta1?
00:33.59brookshire[home]more stable than 1.0.9
00:34.00brookshire[home]:)
00:34.24brookshire[home]in some situations
00:34.26cioHow about vs 1.0.7?
00:34.41brookshire[home]i think 1.0.7 has a security hole
00:35.16simprixbrookshire[home]: how do you like voicepulse
00:35.26brookshire[home]voicepulse is awesome
00:35.39brookshire[home]i'm setting it up on my new computer as we speak actually
00:35.42filebrookshire[home]: wazzup?
00:35.45simprixcool
00:35.48brookshire[home]but i've had them for 6 months
00:35.49simprixthe service is ok
00:35.50*** join/#asterisk Connor (n=billy@198-144-165-65.knx.tn.nxs.net)
00:36.02brookshire[home]and they have a number in my area
00:36.03brookshire[home]:)
00:36.54kingtuxman
00:36.57*** part/#asterisk kingtux (n=kingtux@pool-151-196-44-240.balt.east.verizon.net)
00:37.49cioAnyone running 1.2beta1 in production?
00:38.40supaigtrcio: I was as of today.
00:39.02InfraRedanything to report
00:39.06brookshire[home]hey file
00:39.06cioMeaning you're still running it or you change from that version to another?
00:39.38supaigtrI'm backing down but I think that'll create more problems.  RIght now I have major problems with IAX2 to IAX2 that have been ongoing.
00:40.13*** part/#asterisk epablo (n=epablo@WLL-24-pppoe197.t-net.net.ve)
00:41.33shido6taco bell
00:41.50supaigtrHow do I stop the real zaptel driver loading and force ztdummy to load instead?
00:42.36simprixbrookshire[home]: do i need a normal voicepulse account also
00:42.37ciommmmmmm... taco bell....
00:42.42*** join/#asterisk kingtux (n=kingtux@pool-151-196-44-240.balt.east.verizon.net)
00:42.56*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfj0e.dialup.mindspring.com)
00:43.05brookshire[home]i don't think so
00:44.49RoyK<PROTECTED>
00:45.46supaigtrcio: Its gotten so complicated I don't know whats broken.
00:45.53NetSkierAnyone here attending Astricon?  I am wondering how good it has been so far.
00:46.03RoyKrunning cvs head in production is for madmen and jerjer
00:46.05RoyKthat is
00:46.14InfraRed|<-------------------->| this good
00:46.18InfraRednot to scale
00:46.19NetSkierlol
00:46.37InfraRed:)
00:46.53RoyKs/madman/other madman/
00:46.57brookshire[home]netskier: #astricon
00:47.08NetSkierI have been wondering if I should drive thru rush hr traffic tomorrow to crash the party.
00:47.12NetSkierbrookshire: thnx
00:49.59*** join/#asterisk brookshire[home] (n=matt@esbrooks3.traveller.com)
00:50.17supaigtrAnyone seen calls to drop to on-hold music then back?
00:51.44*** join/#asterisk Uberbot (n=Uberbot@69.252.219.76)
00:52.38marc324how do you force asterisk to load from db?
00:54.01brookshire[home]as in realtime?
00:54.07marc324yes
00:54.21marc324i have the db setup (hopeso).
00:54.43brookshire[home]http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime
00:54.44marc324but now as a a test, i want to load the extensions from db
00:56.28brookshire[home]basically in extensions.conf you will just do something like
00:56.31brookshire[home]switch => Realtime/mycontext@realtime_ext
00:56.40blitzrageyo!
00:56.57brookshire[home]BLITZ!!!
00:57.04blitzragebrookshire[home]: !!!!!!!!!!!!!!!!!!!!!!
00:57.10blitzragebrookshire[home]: we're missing you at astricon
00:57.21HiltonThow is Astricon?
00:57.33*** join/#asterisk dos000 (n=dos000@i216-58-62-73.cybersurf.com)
00:57.38brookshire[home]http://www.midsouthmarketplace.com/~krice/gallery/view_photo.php?set_albumName=album02&id=IMG_3334
00:57.41HiltonT(if only I could get A@H to work, I could actually play with *!!!
00:57.42brookshire[home]BEST PIC EVER!
00:57.48Vlat-btw, by the astericon
00:58.06HiltonTrofl
00:58.07brookshire[home]HiltonT: just drop A@H and do a hardcore install ;)
00:58.15Vlat-if there's a consolidated solution, like ser+asterisk together
00:58.30brookshire[home]much less headache imho :)
00:58.33HiltonTI would, but I want to play a little first before I get my head around Linux/BSD again
00:58.36Vlat-i would be happy to receive any info about it
00:58.37*** join/#asterisk cio (n=na@adsl-072-149-159-016.sip.bhm.bellsouth.net)
00:58.44Vlat-in other words - let me know :)
00:58.53brookshire[home]hilton: can you install debian?
00:58.53HiltonTbeen quite a while since I had Linux/BSD here to work on
00:58.55cioin extensions.conf, if I have 4 lines in a TDM400P, my TRUNK should say ZAP/g1-g4?
00:59.06HiltonTDebian - shouldn't be an issue
00:59.16dos000HiltonT, last time i tried A@H it went smooth. I just did not like centos thz all
00:59.27HiltonTAsterisk fails to start here
00:59.46brookshire[home]install debian
00:59.54HiltonTthi the 100-clone card gets detected as does the rest of the hardware (P2B-DS, dual Cel-522, 512MB, SCSI HDD)
00:59.55brookshire[home]then follow this: http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+Debian
01:00.00Vlat-merge ser and asterisk together...and you won't need 5300/5350 anymore
01:00.05dos000HiltonT, tell what is the error, someone can maybe help
01:00.10HiltonTbrookshire; thanks, reading
01:00.13cioOr ZAP/g1-4?
01:00.47HiltonTno http loaded, AMP shows "Asterisk" as not started, php script errors when creating an extension - which error do you want me to start with?  :)
01:01.14dos000Vlat-, except for the g729 stuff ... not sure how many simultaneous g729 you can do on a P43G machine
01:01.14brookshire[home]lol
01:01.14HiltonTXorcom Rapid - that worth a look? (since it is Debian anyway)
01:01.18brookshire[home]yeah
01:01.20brookshire[home]i like it
01:01.32Vlat-dos000: g729 isn't needed anymore i guess
01:01.36cioAnyone?
01:01.38Vlat-yes, it's a good codec
01:01.40dos000Vlat-, why ?
01:01.50Vlat-it have...maybe 2-3 years
01:02.07brookshire[home]i agree
01:02.16dos000Vlat-, fill me in .. i just came back few months on the voip arena
01:02.17Vlat-right now ILBC sometimes can suffer a concurency to 729
01:02.19brookshire[home]g729 will be replaced with a codec with higher quality
01:02.20brookshire[home]:)
01:02.28Vlat-for example
01:02.30brookshire[home]like mp3 quality
01:02.31HiltonTI have Xorcom here, may try that first
01:02.35dos000brookshire, which one ?
01:02.38Vlat-we're using 729/711 world-wide
01:02.57brookshire[home]dunno.. quess we'll find out
01:03.02Vlat-and when it's no 729 compatibility - 711 play a best
01:03.02simprixbrookshire[home]: did you have to have a phone number or did they give you oner
01:03.03brookshire[home]maybe skype's
01:03.10Vlat-erm
01:03.17tzafrir_laptopHiltonT, xorcom is good :-)
01:03.17Vlat-skype must die :)
01:03.24brookshire[home]simprix: you do not have to have a phone number with voicepulse connect
01:03.26HiltonTcool, will look at that
01:03.28dos000hell with skype
01:03.29brookshire[home]but you can buy one extra
01:03.29Vlat-they're using hacked ilbc
01:03.39HiltonTSkype, schmype
01:03.59Vlat-we had several customers, switched to skype
01:04.11tzafrir_laptopVlat-, no, they're using another codec that is also called ilbc. Or actually: anohter variant of ilbc. A patented one
01:04.15dos000Vlat-, but seriously how many can you do in asterisk ?
01:04.16Vlat-after some month they're went back
01:04.22NetSkierHiltonT: tzafrir  wrote Xorcom.
01:04.55Vlat-tzafrir: any difference? we can use uncompressed 711 now
01:05.02NetSkierI like xorcom too.
01:05.08Vlat-and it make no difference on current dsl speeds
01:05.24Vlat-9kbit or 64kbit...
01:05.38Vlat-is there any difference if user have 3mbit ?
01:05.45tzafrir_laptopTheir variant uses much less bandwidth. But if you can use g711, don't bother
01:05.53dos000Vlat-, ha ... i have people on dial up calling from remote third world countries over vsat !
01:06.03Vlat-dos000: asterisk is our media backend
01:06.10tzafrir_laptopBut with our current DSL, uplink is still a problem
01:06.15brookshire[home]yeah.. in that case you would need g729
01:06.23dos000Vlat-, any reboot horror stories ?
01:06.30brookshire[home]g729 is about the only codec that can fit on dialup
01:06.31Vlat-dos000: we don't use it for primary, coz it have a ..interesting.. implementation of sip protocol
01:06.42Vlat-as PBX it acts the best
01:06.54tzafrir_laptopNetSkier, I'm trying to add more providers to the add-trunk script. Any ideas?
01:07.14Vlat-dos000: we have ser + asterisk, and these working the best
01:07.21NetSkierNot really; I am an asterisk newbie.
01:07.36Vlat-ser can handle a 10000 connection at dual Xenon
01:07.40Vlat-Xeon
01:07.50dos000Vlat-, but tell me did you experienced unscheduled down time at all due to ser and/or asterisk
01:08.00Vlat-asterisk can do anything with codec conversion
01:08.09Vlat-dos000: no
01:08.29NetSkiertzafrir: I just hit upon xorcom to jumpstart my debian efforts.  Great way to get started quickly.
01:08.38Vlat-dos000: we have backup boxes. but, to be honest, we did not need them in the past 3 years
01:08.57dos000Vlat-, wow ... how do you rate asterisk over sems for vmail ?
01:09.01Vlat-i don't want to tell the asterisk suck (it's great software)
01:09.16Vlat-neither (ser is great) (great software, but the config........)
01:09.18HiltonTthen, I suppose tzafrir is a tad on the biased side, which isn't necessarily bad  :)
01:09.35Vlat-together they're making a real-working thingie
01:09.44*** join/#asterisk kingtux (n=kingtux@pool-151-196-44-240.balt.east.verizon.net)
01:09.59NetSkierHiltonT: Quick, install it quickly, while he is here, so you can bug him for help.  ;)
01:10.00CoaxDOkay, who's gonna /msg me an NT Server 4.0 reg key? :)
01:10.18dos000Vlat-, what do you use for failover ?
01:10.27brookshire[home]coax: astalavista.com ?
01:10.31CoaxDMy license arrives tomorrow. i need to do the install today.  C'MON. hah
01:10.45CoaxDbrookshire: Microsoft keeps most of these search engines pretty damn well tied down in that arena. *g*
01:11.28*** join/#asterisk starman (n=inshift@host06.alica.hyatthsiagx.com)
01:11.36Vlat-dos000: just to make things clear
01:11.42brookshire[home]first off.. why nt?
01:11.45dos000CoaxD, NT is'nt that the os from the time of lion heart ?
01:11.46Vlat-also we're using 3 5350
01:11.49brookshire[home]you might as well use window 95
01:12.15Vlat-and "Asterisk suck/SER rulez or vice-versa"...sorry, but it make no difference
01:12.20brookshire[home]windows N(o)T
01:12.37CoaxDdos000: Pretty much
01:12.40CoaxDThat said, i still need it
01:13.20dos000Vlat-, you mentioned you had a failover solution .. what is it based on ?
01:13.26Vlat-dos000: there're a rack, in about 5 meters from me
01:13.53Vlat-dos000: 2 Intel2.4 ghz servers as ser
01:14.20Vlat-if one doesn't respond - every transaction forwardet to the second by hardware
01:14.26dos000Vlat-, how do you do failover
01:14.49dos000Vlat-, my question is how does the forwarding happen
01:14.52Vlat-dos000: 2 SER machine and self-written scripts
01:15.06Vlat-dos000: it does on t_on_failure[x]
01:15.13kingtuxi'm trying to make asterisk-addons and i'm gettting this error ..
01:15.17kingtuxcdr_addon_mysql.c:24:22: asterisk.h: No such file or directory
01:15.17kingtuxmake: *** [cdr_addon_mysql.o] Error 1
01:15.22Vlat-ser's default feature
01:15.44brookshire[home]that's not an error.. it's a feature!
01:16.00*** join/#asterisk NoRemorse (n=axel@202.161.68.2)
01:16.04Vlat-t_on_failure[1] {
01:16.18NoRemorsehello, how do I disbale CLI presentation to a peer please? is it just SetCallerID() ?
01:16.19kingtuxwell i'm looking to store cdr data...so this will screw me up right
01:16.21Vlat-<PROTECTED>
01:16.28Vlat-<PROTECTED>
01:16.29Vlat-{
01:16.30dos000Vlat-, i dont get it. does your failover run (on hw) below sip or in sip ?
01:16.47brookshire[home]kingtux: looks like it can't find the asterisk.h file
01:16.53Vlat-dos000: all we have is a sip, nothing more or less
01:17.13Vlat-IAX is the thing we don't really like (and we can describe why)
01:17.26kingtuxnow i'm getting this error ...
01:17.35brookshire[home]of course.. i don't have that either
01:17.39kingtuxFATAL: Error inserting ztdummy (/lib/modules/2.6.9-11.EL/misc/ztdummy.ko): Unknown symbol in module, or unknown parameter (see dmesg)
01:17.39kingtuxFATAL: Error running install command for ztdummy
01:18.18brookshire[home]that seems like the kernel module is not matching up with the kernel version
01:18.19Vlat-dos000: asterisk is the best for PBX solutions. Very great
01:18.21brookshire[home]uname -a
01:18.40Vlat-dos000: but I have no idea, why the people using it at ITSP level
01:18.59Vlat-yes, it's more confortable
01:19.09brookshire[home]kingtux: you did compile asterisk first and install it before you tried to compile the addons right?
01:19.29MikeJ[Laptop]brookshire, arn't you supposed to be in CA?
01:19.31Vlat-but less configurable. so IMHO, SER is for signalling, * is for anything other
01:19.39brookshire[home]mikej: who would run digium?
01:19.40brookshire[home]:)
01:19.44*** part/#asterisk NoRemorse (n=axel@202.161.68.2)
01:20.08kingtuxyes compiled it fist
01:20.14brookshire[home]did you install?
01:20.18brookshire[home]make install
01:20.34Vlat-some time ago i was in ser devteam
01:20.49Vlat-and it make me cry, the people using asterisk for the proxy
01:21.07Vlat-becose it's several hundreds time SLOWER
01:21.12kingtuxmake install
01:21.21Vlat-'comz it's for PBX solutions
01:21.41brookshire[home]i'm sure one day it will have a proxy
01:21.47brookshire[home]heh.. just not anytime soon
01:22.14Vlat-brookshire[home]: you're alredy have it
01:22.21HiltonTok - Xorcom installing as I type...
01:22.33brookshire[home]hehe.. not a real one
01:22.35brookshire[home];)
01:22.49Vlat-brookshire: grep the source
01:22.53supaigtrVlat: Any pointers on SER TNT and * vm?
01:23.11Vlat-supaigtr: vm like voicemail ?
01:23.15supaigtrYea.
01:23.17Vlat-if so, forget about SEMS
01:23.24supaigtrSEMS?
01:23.36Vlat-SER Media System
01:23.41brookshire[home]vlat: you know.. as long as i've used asterisk.. i've never had to deal with sip.. so i don't really care, lol
01:23.50Vlat-Asterisk voicemail is JUST GREAT
01:24.07brookshire[home]mainly just iax and tdm :)
01:24.09Vlat-you just need forward() your ser to local asterisk backend
01:24.37supaigtrVlat: I'd like use the best of * meetme, etc, best of maxtnt PSTN - SIP, and SER lot of normal users with SIP to the TNT with some voicemail.
01:24.40dos000Vlat-, how you rate asterisk against sems ?
01:24.48Vlat-brookshire: sorry, i think i just did a mistake, thought about only sip
01:25.26Vlat-dos000: SEMS is 10%, Asterisk is 100%
01:25.41Vlat-all: SER is for routing
01:25.50dos000Vlat-, what is missing .. just on the voicemail side of things
01:25.51Vlat-all: Asterisk is for user's pleasure :)
01:25.55*** join/#asterisk Weezey (n=ohno@CPE001195cf5c03-CM0014e8267934.cpe.net.cable.rogers.com)
01:26.03supaigtr* vm and meetme seems very stable, other things like IAX trunking, SIP support seem a bit flaky at times.
01:26.05NetSkierVlat-: I would like to know why you don't like iax2.
01:26.08Vlat-SER is just a router, nothing more
01:26.28dos000Vlat-, i am talking about sems not ser per say
01:26.30Vlat-NetSkier: stop. we doesn't work with IAX
01:26.42brookshire[home]iax is good for asterisk to asterisk
01:26.49brookshire[home]but.. for ip phones..
01:26.54supaigtrSER = softwitch vendors worst nightmare.  Something that works that doesn't cost 6 digits.
01:26.54brookshire[home]iax is lacking for support
01:26.54Vlat-i had several (negative) thoughths about IAX
01:27.10Weezey(IAX slept with his dog)
01:27.11cioIs there a trick to getting tapi notify turned on with asterisk?  I can place tapi calls, but I can't get it to notify on inbound ...
01:27.19Vlat-for example - rtp/sincro broke cause rtp/sincro broke
01:27.20supaigtriax between boxen doesn't work so hot all the time.
01:27.33Vlat-e.g. 30ns timeout for rtp
01:27.35NetSkierWeezey: Heh.  Well, I sleep with my dog too.
01:28.03dos000supaigtr, but its funny how all things ser related are still discussed behind doors. chec the #ser for example
01:28.08Vlat-dos000: sems (f)worked at me...for 15 minutes
01:28.21HiltonTNetSkier; I think I slept with your dog, too  :)
01:28.22supaigtrdos000: True.
01:28.31Vlat-dos000: then I removed the package and installed back asterisk
01:28.32brookshire[home]wow this channel is getting freaky
01:28.37HiltonTlol
01:28.58Vlat-dos000: SEMS can do nothing right now. To be honest, it's not beta. It's the pre-alpha
01:29.05Vlat-look at the CVS
01:29.09dos000Vlat-, thanks
01:29.31NetSkierHiltonT: Well, my dog is a male, if that helps at all.  ;0
01:29.40NetSkier;)
01:29.48Vlat-so, i'll say once more
01:29.53supaigtrVlat: know of any publicly accessable example of ser, tnt, and asterisk?
01:29.55brookshire[home]that's hot
01:29.56Vlat-SER if for signalling
01:30.04Vlat-Asterisk is for media
01:30.14Vlat-supaigtr: www.onsip.org
01:30.22dos000supaigtr, do you have tnt in your hand ?
01:30.27Vlat-supaigtr: completely working example
01:30.34supaigtrYep.  Bunch of them.
01:30.41brookshire[home]asterisk is also for in-house pbx :)
01:30.42supaigtrLeft over from dial up switching to PON.
01:30.48dos000supaigtr, check voip-info.org as well
01:30.51Vlat-o!!!
01:30.57supaigtrVlat:Close I can get.
01:31.14Vlat-The CISCO Hungary motherfuckers brought a beer for me
01:31.26supaigtrI read the article and have it working with * no problems.  Just turn off auth but SER is where i have problems.
01:31.51Vlat-rather bring me a $12000 for hardware-testing
01:32.11dos000supaigtr, what kind of problems. I am about to buy tnts myself
01:32.50*** join/#asterisk nnnnnn (n=killfill@pc-200-74-17-222.asturias2.pc.metropolis-inter.com)
01:32.51Vlat-supaigtr: let's examine. asterisk cand handle x connections at amount of time as router. SER handles x*500 connections at the same configuration
01:32.53nnnnnnhi..
01:33.02supaigtrPain to get setup. U'll want someone who know TNT.  I'll be glad to help of course.  It took like a week to got thru the millions of options to get it working.
01:33.07Vlat-i'm not asterisk addict, neither se
01:33.08Vlat-r
01:33.30dos000supaigtr, did you had someone in here helping ?
01:33.34supaigtrvlat: I just want to use * for the VM and meetme.  10,000 account in SER / TNT and 1000 or so VM.
01:33.45Vlat-i just own the company in EU. and ser+asterisk is the most cost-effective solution right now
01:33.48supaigtrdos000: Nobody in here except BKW he helped.
01:33.55dos000supaigtr, i would really like to get in touch with you if it no a problem
01:34.05Vlat-'coz Cisco5350 is $20000
01:34.16nnnnnni have a abox directly connected to the internet. if i make x-lit connect to the public IP the phone doesnt recieve any sound, but, when i connect to 127.0.0.1 it works..
01:34.30supaigtrNo problem.  I'd really like some extra help getting everything working with ser.  The TNT is noauth so I've got that part.
01:34.33nnnnnnhow can i see where the problem is?..
01:34.56dos000supaigtr, did you buy it used ?
01:34.57supaigtrdos000 U getting tnt for ser or *?
01:35.08Vlat-the only problem - i have to moderate myself sometimes
01:35.11dos000supaigtr, ser+asterisk for vmail !
01:35.19supaigtrdos000: New.  They were used for dialup and fax applications.
01:35.26supaigtrWe have the same goal.
01:35.28Vlat-it's asterisk channel, and i don't want to make someone cry :)
01:36.06dos000supaigtr, mind to take this in pv ?
01:36.12supaigtrk
01:36.36Vlat-when asterisk will get the real REGISTAR, we'll change to it
01:36.50Vlat-SIP REGISTAR
01:37.02Vlat-currently it's sip1.1 i guess
01:37.08supaigtrI kinda think it should stay seperated.
01:39.31Vlat-Asterisk is the PBX solution, and it do it at the best. I just don't understand the people trying to get it to ITSP level
01:39.52supaigtrRight.
01:40.29supaigtrTheres alot to do on the PBX side before trying to replace ser.  Also I have problem using * as TDM gateway. TNT hardware works much better.
01:41.01Vlat-SER is just a proxy
01:41.09Vlat-Asterisk+TDM <> SER
01:41.12Vlat-works fine
01:41.23Vlat-but rather borrow a C5300
01:41.32supaigtrecho, interrupts are a problem.  TNT = no problem
01:41.37Vlat-5300 is onlu $11000 now
01:42.31supaigtr:)
01:42.32Vlat-so I don't really understand
01:43.08Vlat-asterisk maillist is full of replies like "Our customers aren't sastificated with our service"
01:43.14*** part/#asterisk kingtux (n=kingtux@pool-151-196-44-240.balt.east.verizon.net)
01:43.33Vcoi belive SER is actually a "router" not a proxy
01:43.35Vlat-damned god, and who told you Asterisk is business-grade solution???
01:43.55Vlat-Vco: you're right, i didn't choise a right expression
01:44.15Vlat-use ser as router/proxy. use Asterisk as the backing end
01:44.17MikeJ[Laptop]plenty of business use asterisk
01:44.19Vlat-or maybe SEMS
01:44.30MikeJ[Laptop]just use it right
01:44.33Vcoasterisk suxours as SIP routing compared to SER :(
01:44.37Vlat-use Cisco equipment
01:44.42Vlat-maybe Vocaltek
01:45.08Vlat-but just to make it sure. SW solutions never will reach the HW
01:45.25MikeJ[Laptop]asterisk isn't a proxy, it's not meant to be... if you are doing high volume sip termination, yeah use ser.. ser and asterisk together allow you to do some good stuff
01:45.37Vlat-Btw, aster1.2 SIP is much more than buggy
01:45.44Vcoi mean even simple concepts like multiple SIP registrations etc..
01:45.55Vlat-see it before
01:46.06Vlat-Ast. is the best PBX we had found
01:46.13Vlat-even hardware ones
01:46.25Vcohell ya...
01:46.30Vlat-but like ITSP-grade solution it's nothing
01:47.00Vlat-it's the way I hate SER and I like SER
01:47.02Vlat-it's config
01:47.06Vcoya
01:47.10supaigtrI've had more than one customer revolt on * and buy a lucent or panasonic.
01:47.14Vlat-i can do virtually everything with it
01:47.44brookshire[home]supaigtr: i would love to know the outcomes of that, are they happier with that?
01:47.47*** join/#asterisk chidex (i=richard_@82-45-239-141.cable.ubr01.enfi.blueyonder.co.uk)
01:47.58Vlat-i have the except customer ? ok. i'll write the codes to config and the customer's HW will work
01:48.16chidexI take it that asterisk 1.0.9 is reasonably stable?
01:48.21Vlat-in the case of Asterisk we'll need to submit CVS change and wait for a month
01:48.27Vcoanyone else have problems with Asterisk <-> Asterisk SIP and DTMF?
01:48.35supaigtrbrookshire: Most are happy with the pana.  Not the lucent.  But * has lots of problems in the business world.
01:48.37Vlat-chidex: more than stable
01:48.49Vlat-chidex: using it at primary
01:49.30*** join/#asterisk forrestc{hm} (n=forrestc@206.127.77.82)
01:49.42forrestc{hm}Hello everyone.
01:49.43chidexvlat: didn't get that last bit
01:49.51forrestc{hm}Got a bizzare one tonight
01:50.12Vlat-chidex: pastle it in, please
01:50.12nnnnnnhey Vlat-, when my software sip phone connect to asterisk, but doesnt get any sound from it, i.e. the ansear mashine voice record, what does it mean?..
01:50.31clyrradWhat is the best codec to use for IAX to IAX communication?
01:50.33nnnnnnit can send sounds..
01:50.41Vcois it behind nat?
01:50.45Vlat-nnnnnn: set your phones to use nat+stun. btw, grandstream ?
01:50.46forrestc{hm}nnnnn: probably a firewall issue, assuming you don't have an audio problem
01:50.53forrestc{hm}clyrrad: it depends
01:50.57clyrradon what?
01:50.59Vlat-nat=yes usually helps
01:51.03brookshire[home]clyrrad: inhouse or over the internet?
01:51.07clyrradover internet
01:51.16brookshire[home]gsm or g729
01:51.21forrestc{hm}clyrrad: bandwidth versus quality versus packet loss.
01:51.23nnnnnnhm..  Vlat- you mean on sip.conf?
01:51.37forrestc{hm}clyrrad: I personally use ulaw for everything.
01:51.59forrestc{hm}clyrrad: no transcoding issues, better quality, no license issues, etc.
01:52.04Vlat-nnnnnn: 1) set up ip forwarding on customer routers 2) set up stun server at customer hardware to stunserver.org
01:52.05clyrradI have g711u right now, makes a strange sound until the call is connected for some odd reason
01:52.15Vlat-and don't forget about DNS
01:52.17forrestc{hm}clyrrad: but takes more bandwidth.
01:52.22HiltonTxorcom rapid: what are the default passwords for the extensions, and how are these change?
01:52.39Vlat-if you want just to do it work at everyone
01:52.45brookshire[home]hilton: you mean for voicemail?
01:52.47Vlat-than forget about it
01:52.57HiltonTnope, to attach my SIP hardphone to it
01:53.03Vlat-it's impossible even at B2BUA
01:53.07HiltonTand then later, for voicemail  :)
01:53.28forrestc{hm}vlat: I've had almost zero problems with the sipura stuff with nat=on and no stun.
01:53.37clyrradforrestc{hm}.... Any idea why some IAX to IAX calls pass the Caller ID and other show up as Anonomous?  If I call from the same number to the same source some times the CID will show up othertimes it wont.....
01:53.53Vlat-forrestc{hm}: and the local network ? :))
01:54.03brookshire[home]clyrrad: that is a configuration problem ;)
01:54.13forrestc{hm}vlat: we're an ISP with dozens of different types of customer routers.
01:54.18brookshire[home]you can set the caller id to be whatever usually
01:54.33Vlat-forrestc{hm}: NAT user trying to call another NAT user. 192.168.0.155 trying to call 10.0.0.221
01:54.35forrestc{hm}vlat: Asterisk is on a real IP address which helps
01:54.44forrestc{hm}vlat: oh..  That does suck.
01:54.49brookshire[home]if no caller id specified.. then anonymous
01:54.56clyrradbrookshire[home].... Its kind of hard to spot a configuration issue with this when some times it works and other times it wont under the same testing conditions.
01:55.03clyrradCould it be CVS HEAD?
01:55.04Vlat-forrestc{hm}: we had solved it by the ser way
01:55.23Vlat-forrestc{hm}: asterisk does the rtp proxying, when it needed
01:55.32Vlat-+2 line to config
01:55.44brookshire[home]intresting.. do you control the end that is giving you an anonomous back?
01:55.59clyrradYes I control both
01:56.08supaigtrclyrrad: It does that. I have the same problem.  Sometimes it send CID sometimes not.
01:56.11Vlat-so I have no idea
01:56.15forrestc{hm}vlat: we basically decided just to use asterisk as a B2BUA and our PSTN gateway (actually a farm of asterisk boxes).
01:56.30clyrradsupaigtr.... Are you using CVS HEAD as well?
01:56.35Vlat-forrestc{hm}: to be honest - have no idea
01:56.56brookshire[home]intresting.. i'll look into it
01:56.59Vlat-forrestc{hm}: we're using SER as proxy. It have more than flexible config to do everything
01:57.02brookshire[home]if you can reproduce it
01:57.11brookshire[home]i would post it to the bug tracker
01:57.11Vlat-forrestc{hm}: so we solve our problems with SER
01:57.32Vlat-when we need the MEDIA we forward to ser
01:57.34clyrradbrookshire[home] was that message to me?
01:57.38Vlat-for example for 404
01:57.40brookshire[home]yes sir
01:57.42supaigtrclyrrad: Yep.
01:57.42brookshire[home]:)
01:57.51brookshire[home]vlat: we get it.. ser is great for sip :)
01:58.15clyrradHow to reproduce it is a damn good question LOL.... there is no rhyme or reason to it that I can see, sometimes it works and other times it wont
01:58.25Vlat-[default] exten 404 on asterisk say "User is not available, press 1 to leave the message"
01:58.32clyrradsupaigtr.... have you had any luck finding the EXACT condition that causes it to happen?
01:58.34forrestc{hm}clyrrad: have you looked in the Asterisk console with debugging/verbose turned up
01:58.48supaigtrclyrrad: I have a simlar intermittent problem with muting audio in one direction as well with iax
01:59.08Vlat-press 2 to call it again (+call time to my billing(
01:59.20brookshire[home]brb
01:59.21Vlat-+3 to contact the supprot team
01:59.24HiltonTXorcom seems to have installed fine and be running fine, but NFI how to add/remove extensions, nor how to register my hardphone to ext 400
01:59.25Vlat-(payable)
01:59.32clyrradforrectc{hm}, yes when CID is not sent I do not see anyting different on the CLI and I am running about 10 v's when I load *
01:59.55forrestc{hm}clyrrad: have you done a "set debug 255"
02:00.04clyrradsupaigtr are you using CVS head?
02:00.14Vlat-Asterisk = PBX. The guy wrote SIP module for it it's a fat german guy
02:00.15supaigtrYep.
02:00.25Vlat-(2 month ago it's married man)
02:00.33Vlat-so don't expect a lot
02:00.33clyrradforectc{hm} i just did that now
02:00.35NetSkierHiltonT: try asteriskguru.com for some phone configuration hints.
02:00.41*** part/#asterisk cio (n=na@adsl-072-149-159-016.sip.bhm.bellsouth.net)
02:01.05NetSkierHiltonT: google for asteriskguru if I got the url wrong.
02:01.11HiltonTjust don't know the password set by default in Xormcom Rapid, and apparently little documentation, but reading that site...
02:01.26HiltonT(I know how to config this phione fine, just not Rapid)
02:01.34NetSkierHiltonT: basicly edit /etc/asterisk/extensions.conf
02:01.36forrestc{hm}clyrrad: I'm trying to figure out how to spit a debug out of some sort with the CID Just before the dial.
02:01.45HiltonTk, editing now...
02:01.54supaigtrnoop
02:01.57forrestc{hm}clyrrad: how often does this occur?
02:01.59clyrradforrestc{hm} even with that debug set the same thing is happening, with no errors on the CLI
02:02.05clyrradrandomly
02:02.10clyrradbut all the time
02:02.15clyrradif that makes sense*
02:02.27forrestc{hm}clyrrad: what did you say the calling source was?
02:02.31clyrradWhat I mean is every time I call it will happen, but the odd time the CID will work
02:02.47clyrradits IAX to IAX
02:03.35chidexvlat: so are you using 1.0.9 in a production environment then?
02:03.43supaigtrclyrrad: U had any audio problems?  Not often but it happens.  1 -3 sec audio lost on far end IAX?
02:03.58forrestc{hm}clyrrad: an IAX to IAX trunk, or actually using an IAX ATA or phone.
02:03.59Vlat-chidex: yes
02:04.26clyrradYes i have Audio problems like a JITTER or a lag at the beginning of every call, takes about 3 seconds, then its gone for the remainder of the call, its like it is doing handshaking of some kind
02:04.38chidexvlat: did you apply any custom patches?
02:04.46clyrradI have an * box, and an IAX ATA
02:05.08forrestc{hm}clyrrad: so you basically want the CallerID to work on the IAX ATA.
02:05.10Vlat-chidex: nothing
02:05.17*** join/#asterisk jets (n=jets@dsl001-136-136.lax1.dsl.speakeasy.net)
02:05.22chidexvlat: cool
02:05.36forrestc{hm}clyrrad: is the a call from * to the ATA or from the ATA to the *?
02:05.36clyrradYes
02:05.40chidexvlat: how big is your setup?
02:05.54Vlat-chidex: 1000+ customer
02:05.58clyrradThe call is from my cell phone to the * box which sends to the ATA
02:06.17forrestc{hm}clyrrad: how are you getting the call into the * box?
02:06.18chidexvlat: oh yeah, using ser! :)
02:06.28clyrradfrom my VOIP provider
02:06.53forrestc{hm}clyrrad: so you're taking a did from a voip provider into your * box and passing onto an IAXy or similar?
02:07.01clyrradexactly
02:07.09Vlat-chidex: it make no differnece for me. if ser is working for that setup - it's OK. if * would work - that would be OK
02:07.27forrestc{hm}clyrad: what protocol are you talking to your voip provider?  IAX also?
02:07.32Vlat-there's no time for the holy wars
02:07.33clyrradyes
02:07.46clyrradIAX to the VOIP provider to Me, then from Me IAX to the ATA
02:07.58Vlat-fuck
02:08.14forrestc{hm}clyrad: have you done anything which indicates that you are actually receiving CID On the * box every time?
02:08.20Vlat-Uncensored Terminator II it's absoluterly different movie !!!
02:08.43clyrradsorry didnt follow your last question
02:09.10Vlat-about IAX and ATA
02:09.16NetSkierVlat-: I have not heard of the uncensored version.  Where can I get it?
02:09.27Vlat-if you have 100+ customers you would like to do it
02:09.34Vlat-- don't do it
02:09.42forrestc{hm}clyrad: there are two places this could be broken... either the cid from your provider, or cid from * to the ATA.
02:09.57Vlat-there're several problemms with IAX
02:10.04Vlat-for example E164
02:10.12forrestc{hm}clyrad: what I'm wondering is if you know whether or not you are getting the CID On the * box from the provider each time
02:10.26clyrradhrm.... its a good question
02:11.03forrestc{hm}Hold on..
02:11.11Vlat-NetSkier: i have it here. and i heared neither. but it's here!!! and the TII movie end do not stops with T1 melting
02:11.19HiltonThhmmm, seems no simple (as in straightforward) way to edit the files in Xorcom Rapid (extensions, for example) and no docs, and no idea what the default passwords are, and no way to connect my hardphone to ext 401, for example  :(
02:11.33forrestc{hm}clyrrad: try a iax2 debug on the command line.
02:11.36Vlat-NetSkier: there're a 15 minute trail by Connor's thinks
02:11.58clyrradok done
02:12.02clyrradand got an annon that time
02:12.24forrestc{hm}clyrrad: did you get the anon from the iax box?
02:12.31forrestc{hm}er iax debug?
02:12.49clyrradI called from a phone connected to the * box to my cell phone that time and got Annon on my cell CID
02:12.50NetSkierHiltonT: I booted it into single user to set the root password.  Then booted into multiuser mode.
02:13.11HiltonTI'm ssh'ed into it
02:13.21forrestc{hm}clyrrad: that was backwards from what I heard you say before.
02:13.36NetSkierHiltonT: Then I used the editor nano, a super small emacs, to edit /etc/apt/sources.list to add the regular debian repositoriers.
02:13.37clyrradyes, right now I tried it the other way around
02:13.50forrestc{hm}clyrrad: both ways work differently
02:13.56forrestc{hm}clyrad: different set of issues.
02:14.01clyrradthat what im checking right now :)
02:14.10HiltonTof the 3 distros - Astlinux, A@Home (and its latest Beta) and Xorcom Rapid, at least Rapid actually starts running Asterisk!!!
02:14.22NetSkierThen I became root, and ran apt-get'ed some packages, like emacs, and whatever else I wanted.
02:14.27clyrradseems to work evrey time when i call from cell phone to the IAX box
02:14.32HiltonTI don't want to make it a full Debian system, I want some docs on how to drive the distro!
02:14.36clyrradAsterisk box i mean *
02:14.43NetSkierThen, with my favorite editor, I edited /etc/asterisk/extensions.conf.
02:14.45HiltonTit has vim on it - I'm happy with vim
02:14.49forrestc{hm}clyrrad: you get cid every time from the cell phone.
02:14.54forrestc{hm}clyrrad: right?
02:14.55NetSkiergreat; use vim then.
02:15.04NetSkierbut you need to be able to become root.
02:15.12Vlat-NetSkier: also, the guys are messing tottally (shocker, rude bite for example) with Mrs. Connor at the start. There no such thing in the uncens movie
02:15.14forrestc{hm}clyrrad: doesn't work from the phone to the cell phone.
02:15.19NetSkierto edit those files, IIRC.
02:15.22HiltonTI ssh'ed in as root  :(
02:15.32clyrradYes, every time i call from my cell phoen to the * box CID works, but this is a SIP phone connected directly to the * box, thats the one that always gets the CID
02:15.34NetSkierok
02:15.43clyrradso I think your right, the problem is between * and the IAX box
02:15.50clyrradnot with the VOIP provider
02:15.55Nuggetno, it's his left.
02:16.08forrestc{hm}clyrrad: does the iax box ever work?
02:16.20Vlat-about IAX... it has no native CID
02:16.35Vlat-in the latest revisions it was diverted
02:16.36HiltonTthe extensions.conf in /etc/asterisk doesn't match the extensions shown in its console UI  :(
02:16.46Vlat-but still to buggy imho
02:17.09clyrradThe IAX box works in the sense that i can send and receive calls, but its CID never works when I call from the IAX box to my cell phone
02:17.35forrestc{hm}clyrrad: are you sure you have callerid set right int he iax.conf?
02:17.36*** join/#asterisk wolfson (i=hehe@nc-71-2-31-68.dyn.sprint-hsd.net)
02:17.50clyrradwoops that was a lie, it just worked this time
02:18.05clyrradthat was the first time I got CID off the IAX box
02:18.17forrestc{hm}clyrrad: could you let me know what happens in these cases:
02:18.20clyrradand I have made no changes to the configs
02:18.23forrestc{hm}SIP ATA Call to cell phone
02:18.28forrestc{hm}IAX ATA Call to cell phone
02:18.33forrestc{hm}Cell phone call to SIP ATA
02:18.38forrestc{hm}Cell phoen call to IAX ATA
02:18.57forrestc{hm}SIP ATA might mean SIPphone if it's not an ATA>
02:19.03clyrradOk SIP to Cell I get random if the CID will work or not
02:19.22clyrradSame with IAX ATA to Cell, its random if it will work or not
02:19.31clyrradCell phone to SIP ATA works EVERY time
02:19.32Vlat-XXXX <-> AABBCC
02:19.41clyrradCell phone to IAX works Randomly
02:20.00forrestc{hm}clyrrad: one more:  SIP phone to IAX phone,
02:20.11Vlat-forget about getting the proper CID every time
02:20.12forrestc{hm}on and one more after that:  iax phone to sip phone.
02:20.37*** join/#asterisk NoRemorse (n=axel@202.161.68.2)
02:20.40clyrradNope it sais NO Data when i do that
02:20.48Vlat-CallerID is the unstardatesising piece of shi... the thing sometimes don't work
02:20.51forrestc{hm}vlat: I'm about ready to tell him the same thing... sounds like his provider is screwing with CID.
02:21.02Vlat-for example
02:21.02NoRemorsehello, has anyone got a h323 gateway O can test some calls to please? trying to debug my openh323 setup here
02:21.08Vlat-US has the own CID format
02:21.22NoRemorse*I
02:21.23Vlat-Russia don't evem has CID, but it's working
02:21.44Vlat-fucking Europe has something standartized, but it doesn't works
02:21.48forrestc{hm}clyrrad: what do you have in your sip.conf and iax.conf
02:22.02Vlat-Korea had found the OWN CID
02:22.16clyrradwell in SIP.conf I dont have anyting for the IAX phone or ATA if that is what you mean
02:22.17Vlat-Japan can convert from US CID
02:22.32clyrradand in IAX.conf i have the register that registeres the DID with my provider
02:22.35Vlat-and there're only the primary countries
02:22.46forrestc{hm}clyrrad: no, I'm just wondering how/where you are setting the CID for the ATA's.
02:22.46clyrradVlat... sounds like we have a real mess huh
02:22.56Vlat-it's a REAL mess
02:23.07clyrradOh, the CID is being passed from my VOIP provider
02:23.23Vlat-i have about 10 exceptions for CID-s in my ser.cfg\
02:23.26clyrradso then i do this SetCallerID(<${CALLERIDNUM}>
02:23.28forrestc{hm}vlat: In all reality, the US CID system seems to work pretty well as  long as everything is provisioned for it.
02:23.29clyrradin the dial plan
02:23.45forrestc{hm}clyrrad: have you tried hardcoding $CALLERIDNUM
02:23.47Vlat-forrestc{hm}: US CID works 100% in US
02:23.48*** part/#asterisk NoRemorse (n=axel@202.161.68.2)
02:24.12clyrradforrestcm{hm} yes i get the exact same results some times it works and other times it does not
02:24.18Vlat-i will tell the other thing ;)
02:24.21clyrradI thought i may be a CVS HEAD bug
02:24.28Vlat-There's the Elcatel
02:24.41Vlat-the ATC manufacture operator
02:25.03Vlat-there's the Alcatel too
02:25.08forrestc{hm}clyrrad: I suspect more likely that your provider is not honoring CID from you for some reason.
02:25.12Vlat-and there's the Ericcson
02:25.12forrestc{hm}clyrrad: which provider?
02:25.22clyrradunlimitel
02:25.31Vlat-also there're about 5-th of Chineese ATC providers
02:25.58Vlat-and EVERY OF THEM is incompatible at CID level with each other
02:26.12forrestc{hm}clyrrad: see http://www.voip-info.org/tiki-index.php?page=Unlimitel
02:26.18clyrradVlat-> I see that has been frustrating you for some time huh? :P
02:26.22Vlat-personally me had to check this, so i know what about i speaking
02:26.48Vlat-clyrrad: i'm rather hardware-man than SW-man...so of course it's frustating for me :))
02:27.12clyrradforrestc{hm} LOL
02:27.15clyrradNice link
02:27.25clyrradhere I am pulling my hair out over my configuration
02:27.39clyrradI wonder why that is a problem
02:28.00Vlat-HW-man means 3-floor ATC + Ser :))
02:28.03forrestc{hm}clyrrad: they probably don't use PRI or SS7 or similar for their trunks, which don't support CID.
02:28.25Vlat-atc can not do a trick in this case
02:28.34forrestc{hm}I said that wierd... but you get the idea hopefully.
02:28.52Vlat-by the way
02:29.09forrestc{hm}On the ATA side, I suspect it *might* be an issue where the ATA isn't 100% callerid-friendly or there's a setting to get it to pass CID Correctly, etc.
02:29.16Vlat-CID is a little circuit in the end-customer's ATC
02:29.41Vlat-if it's inside ATC, you can do any thing
02:30.07forrestc{hm}clyrrad: might also be the CID box if it's really an ATA.
02:30.10clyrradThe ATA I am using does not seem to have any such setting its a GNET VP168I not sure if anyone is farmiliar with that one or not
02:30.15Vlat-but it'll replace the append_rpid_hf by the own nubmer
02:30.33Kattymew.
02:30.35forrestc{hm}clyrrad: not here, but like vlad indicates CID sometimes is a mess.
02:30.53clyrradLOL, yup and i think this is a perfect example
02:30.54forrestc{hm}Hmmm.. back to my bizzare SIP problem myself.
02:31.00clyrradthanks for helping me get to the bottom of it
02:31.08clyrradthat link was the final proof that my configs are fine
02:31.38clyrradBut there is a seperate issue with the ATA not sending and receiving caller ID
02:31.42Kattymew?
02:32.07Vlat-btw, that circuit has Atmel ROM :) $xxxx and we can reset it :)))))
02:32.22Kattylet's trying rephrasing.
02:32.24Kattyhi!
02:32.31supaigtrSup.
02:32.32Kattyobviously no one speaks kat around here
02:32.54Vlat-to be serious
02:33.11Vlat-the last-end GW forms the CID
02:33.28Vlat-if it don't want to do - the user receive CID you set up
02:33.54Vlat-if ANY GW at the path trying to change CID - you have nothing to do with it
02:33.57forrestc{hm}Is anyone aware of a SIP call audio distortion issue with HEAD?
02:34.06forrestc{hm}ulaw codec
02:34.44clyrradforrest what kind of distoration is it?  Like a LAG or JITTER?  or just scrambled audio?
02:34.52forrestc{hm}scrambled audio.
02:35.03Vlat-forrestc{hm}: what doest the ulaw have to do with sip ?
02:35.07Vlat--t
02:35.10*** join/#asterisk brookshire[home] (n=matt@esbrooks3.traveller.com)
02:35.15forrestc{hm}calls *all* start out fine but get more and more distorted.
02:35.24*** join/#asterisk Brijn (n=bas@S0106004063c0fa1f.vn.shawcable.net)
02:35.36forrestc{hm}vlat: you can run different codecs on sip
02:35.40BrijnGood evening all
02:35.58Vlat-forrestc{hm}: let's start it again: what equipment have you got. CISCO? Vocaltec? PC with SER ?
02:36.00forrestc{hm}vlat: i.e. ulaw (aka 711u), 729, etc.
02:36.14Vlat-forrestc{hm}: of course
02:36.19forrestc{hm}vlat: I have an asterisk box with a 4 port digium card.
02:36.31forrestc{hm}vlat: I am passing *MODEM* calls through this asterisk box with no problems.
02:36.44*** join/#asterisk crash3m__ (n=crash3m@unaffiliated/crash3m)
02:36.48Vlat-forrestc{hm}: codec is codec. signalling is signalling. billing is billing. etc....
02:36.52forrestc{hm}SIP calls to/fron the asterisk box sound fine, then slowy get worse.
02:37.06Vlat-dos000: #ser is dead
02:37.37forrestc{hm}digium card means digium T1 card aka TE405.
02:37.45dos000Vlat-, long live to <> ?
02:37.49JamesDotComVlat-: so :~
02:37.52Kattywickets.
02:38.02JamesDotComstill need to build the channel up
02:38.08Vlat-dos000: of course i can tell the * is the ....!!!! ..........!!! ..!!!!
02:38.19clyrradforestc{hm} does the distortion happen all the time no matter how many calls the PBX is taking on?
02:38.21Vlat-but it wouldn't be the true
02:38.25forrestc{hm}clyrrad: yep.
02:38.37clyrradso even with 1 call in and out you get this problem?
02:38.40forrestc{hm}and every call starts out fine.
02:38.40dos000Vlat-, long live to openser ?
02:38.43forrestc{hm}clyrrad: yep.
02:38.48Vlat-forrestc{hm}: are you serious????
02:38.52clyrradhow long is it before they go sour?
02:38.53Vlat-the MODEM ????!!!!!
02:38.59forrestc{hm}clyrrad: different.
02:39.04Vlat-dos000: never tried openser
02:39.05forrestc{hm}clyrrad..  usually a few seconds.
02:39.13clyrradthats what I was goning to suggest too the modem or the card may be bad
02:39.34forrestc{hm}clyrrad: *VOICEMAIL* calls to the asterisk box do the same thing.
02:39.36clyrraddo you have another card you can try to rule out defective hardware?
02:39.58clyrradalways from the same phone?  or with any phone?
02:40.01tamp4xwhat u trying to do with ser
02:40.12Vlat-modem with 10+...
02:40.13forrestc{hm}Different ATA's... and were working *fine* with an earlier version of asterisk.
02:40.16Vlat-interesting
02:40.20Kattyomgwtfulolzkthxbi
02:40.22tamp4xi just set up a stateless roundrobin router with it
02:40.29Vlat-but if the proto is IAX
02:40.32forrestc{hm}I'm also passing 56K modem calls at 53K through the T1 ports.
02:40.35Vlat-it would suck well
02:40.53forrestc{hm}Got 2 T1's attached to Telco PRI's and 2 T1's attached to the modem bank... all is well.
02:40.57tamp4xi had one of my programmers write an iax2 router =]
02:41.35clyrradhrm.... that is strange, so different ATX's and different phones, all produce the same audio problem even if the calls are not going accross your digium card?
02:41.49forrestc{hm}Heck I have one caller which has been on a modem call 2:15 between two channels on teh digium card.
02:41.52forrestc{hm}clyrrad: yep.
02:42.04forrestc{hm}clyrrad: and what's weird is it sounds codec related.
02:42.18forrestc{hm}clyrrad.  I.E. I get "phonetic" distortion, not loss.
02:42.20clyrradI would agree
02:42.34clyrraddoes sound like codec to me too
02:42.38*** part/#asterisk n3u7 (n=neutrin0@CPE000d8802a707-CM0011e6c7edb1.cpe.net.cable.rogers.com)
02:42.40clyrradI wonder if you can re-install the codec's
02:42.42forrestc{hm}but I'm running 711u, which isn't really a lossy codec.
02:42.51forrestc{hm}it's either there or it's not.
02:43.09clyrradyou compiled * right?
02:43.13forrestc{hm}yes.
02:43.13clyrradfrom CVS
02:43.19forrestc{hm}yes.
02:43.20clyrradwhen?
02:43.29forrestc{hm}last 48 hours.
02:43.48forrestc{hm}let me get a version.
02:43.49clyrradhave you checked for any updates since then?
02:43.57forrestc{hm}somewhat..
02:44.04forrestc{hm}looked at the CVS Commits.
02:44.11clyrradhow are you checking your cvs version?
02:44.13forrestc{hm}didn't see anything seemingly related.
02:44.54FuriousGeorgeanyone ever seen fax detection not working despite the user setting it up right?  is there some bug i dont know about
02:45.06clyrradFax over VOIP?
02:45.25FuriousGeorgeclyrrad: no
02:45.28FuriousGeorgepots
02:45.40forrestc{hm}clyrad: from the file i touch when I do a cvs
02:45.41clyrradsorry cant help u on that one
02:45.58forrestc{hm}October 10th 21:39
02:46.10forrestc{hm}MDT
02:46.13clyrradMy CVS was from October 1st
02:46.17clyrradtake your CVS from there
02:46.41clyrradbecase we know I have that CVS, and I dont have the problem your having....
02:46.57clyrradso if you date back then and get your files from there and still have the same problem then it must be hardware
02:47.47forrestc{hm}I'll play with that a bit later tonight.
02:48.18clyrradits the only other thing i can think to suggest, becase I am not sure if you can manually re-install the codecs to see if that is the problem
02:48.54forrestc{hm}Before I do that I'm going to see if I can find a tool which will let me grab the RTP packets and play them back "non-streamed" to determine if it's bad data being sent by asterisk or if it's a network issue.
02:49.44forrestc{hm}I'll write a quick chunk of perl to strip the ulaw stream out of a tethereal dump if I can't find a tool to do it automatically.
02:50.46crash3m__I know one exists to rip some type of stream, wanting to say they called it a forensic tool heh
02:50.48clyrraddo you have alot of traffic accross your network?
02:50.58*** join/#asterisk cwetter (n=cwetter@68-114-46-75.dhcp.stbr.ga.charter.com)
02:51.27clyrradif not and this is all on your internal LAN you should have to worry about a network issue
02:51.39clyrradshould'nt*
02:51.55supaigtrforreestc: You can tell if network is bad with ipperf.
02:52.30cwetterwhen I run modprobe wcfxo I get 'ZT_CHANCONFIG failed on channel 4: No such device or address (6)'.  What do I do?
02:53.03forrestc{hm}clyrrad: I really doubt it's a network issue.   If it is it's related to the network card in machine, not the transport to the SIP Devices.
02:53.48forrestc{hm}hey this looks promising...  rtpdump
02:53.57supaigtrforrestc: ipperf can tell you if UDP iax2 traffic can make it from point a to b realiably.
02:54.59cwetterIs s110m an FXO or FXS Card?
02:55.18clyrradforrestc are you using VOIP at all with your Asterisk box?
02:55.38crash3m__wtf else would you be doing besides VoIP with an asterisk box?
02:55.39forrestc{hm}supiagtr: what's really bizzare is that RTCP indicates *no packet loss*  or other issues on the call
02:56.00forrestc{hm}clyrrad: yep.   The call problems are SIP.
02:56.14forrestc{hm}crash3m: We use it as a call router around here T1 to T1.
02:56.40forrestc{hm}cupiagtr: and RTCP usually *KNOWS* what is going on.
02:56.46clyrradBut for me the Jitter or Lag whatever it 'SHOULD' be called, it goes away after 2-5 seconds like handshaking is taking place.... know what i mean?
02:57.21forrestc{hm}clyrrad: probably an issue with the jitterbufer tuning up.  It might also be something to do with echo cancellation.
02:57.49clyrradits one of the 2 havent been able to fix it yet, wanted to see if you were having the same issue or if you solved that part already
02:58.49supaigtrI have IAX2 muting in one direction every so many calls.  I can't find any problems.  Its like a ghost problem.
02:58.54*** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com)
02:59.46clyrradDont have the Muting problem, I use IAX2, both directions sound good, it is just a bit of jumping of the call at the beginning, sounds more like a lag than anyting else
03:00.02supaigtriax2 show netstats?
03:00.28clyrradhrm... never tried this before
03:01.49clyrradChannel                    RTT  Jit  Del  Lost   %  Drop  OOO  Kpkts  Jit  Del  Lost   %  Drop  OOO  Kpkts
03:01.49clyrradIAX2/unlimitel-1            1000   -1    0    -1  -1     0   -1      0    0    0     0   0     0    0      0
03:01.53clyrradhow does that look to you?
03:02.23supaigtrRTT looks bad.
03:02.32clyrradReally?
03:02.35supaigtrOh.  Is jitterbuffer on?
03:02.42supaigtrBoth sides?
03:03.04supaigtrRTT is Round Trip
03:03.19clyrradon the IAX box all I have is jitter size and its set to 0
03:03.19supaigtrI have like 2-6 ms
03:03.43clyrradin IAX.CONF i have forcejitterbuffer=no under [general]
03:03.52Vlat-damned good UK production
03:04.15clyrradsupaigtr... what do you have set for jitter?
03:04.16Vlat-by the way
03:04.26*** join/#asterisk docE (n=docE@dsl001-136-136.lax1.dsl.speakeasy.net)
03:04.26Vlat-if you're using the jitter
03:04.26supaigtrI've been using the new jitterbuffer which fills out the rest of those values.  Seems to work. Ihave since reverted to stable astersik but I think my muting was related to zaptel card.  I unloaded that module and loaded ztdummy and going to test tommorow.
03:04.33Vlat-just add +-10 to it
03:04.49supaigtrI just turned on jitterbuffer=yes.  the new jb is supposed to do things for you.
03:04.55Vlat-it would be more or less real value
03:05.07docEsup sup from Astricon!
03:05.11clyrradsupaigtr in iax.conf under [general]?
03:05.20*** join/#asterisk KaBewM (n=kabewm@66-215-7-106.dhcp.psdn.ca.charter.com)
03:05.26*** join/#asterisk Soulz--- (n=Soulz---@host-137-132-43-194.imcb.nus.edu.sg)
03:05.28docEpictures available @ http://astri2005.netdr.biz  More coming!
03:05.30Soulz---hi all
03:05.32clyrradVlat are you refering to setting Jitter Size =10 in my IAX ATA?
03:05.32*** join/#asterisk xheliox (n=jeff@user-0c6s3h2.cable.mindspring.com)
03:05.36Vlat-docE: what's new at asteric ?
03:05.55Vlat-clyrrad: i meant to set up the adaptive jitter
03:06.11Soulz---i am getting  Set Absolute Timeout to 15 timeouts when ever a incomming call
03:06.14supaigtrclyrrad: there or in each iax section.  I do it in both.
03:06.15docELots of more companies going to be supporting asterisk
03:06.18Soulz---any idea how to fix that please?
03:06.41clyrradsupaigtr... you have jitterbuffer=yes in your general and each context of your iax.conf?
03:06.58Vlat-<PROTECTED>
03:07.00clyrradVlat, i do not have such a setting in my ATA
03:07.20Vlat-e.g. endpoint to ....austria [xxx]xxxxxxxx for example
03:07.28supaigtrYep.
03:07.37clyrradhrm... let me give that a shot
03:07.42Vlat-c.
03:07.46supaigtrVlat-:  I just want to connect two offices.
03:07.58Vlat-clyrrad: ATA like ATA186?
03:08.55Vlat-supaigtr: look, i really have less experinece with IAX. But with SIP it usually take 10-20 minutes
03:09.01Vlat-with H323 - 30 minutes
03:09.19docEI test for dCAP tomorrow!   YAY!
03:09.24Vlat-(10mins to tell the people WHY they're shouldn't use H323)
03:09.28brookshire[home]yay!
03:09.42brookshire[home]everyone uses h323
03:09.44brookshire[home]hehe
03:09.50brookshire[home]well... all the big boys
03:09.54clyrradsupaigtr LOL i think that fixed it :)
03:10.01Vlat-give me the man use h323
03:10.10Vlat-i'll make Uncle Bens from him
03:10.41Vlat-or maybe Uncle Vlad
03:10.58Vlat-3 years ago i had to wrote h323 applications
03:11.17Vlat-openh323 made a more than bad impression to us
03:11.22Vlat-with the fscking bugs
03:12.37brookshire[home]:(
03:13.37brookshire[home]katty: he's getting drunk in cali
03:13.39brookshire[home]:(
03:14.01Kattybrookshire[home]: i think he's actually is disneyland
03:14.06brookshire[home]haha
03:14.08Kattybrookshire[home]: are you at astricon as well?
03:14.08brookshire[home]probably so
03:14.12brookshire[home]no..
03:14.14Kattyk
03:14.15brookshire[home]i had to stay home
03:14.24brookshire[home]are you his nextel bud? ;)
03:14.46Kattyoh, perhaps ;)
03:14.53*** join/#asterisk epablo (n=epablo@WLL-24-pppoe197.t-net.net.ve)
03:14.53Kattyi do beep beep on occasion
03:14.59brookshire[home]haha
03:15.03docEKatty?   Female edition?   nice
03:15.04brookshire[home]always while we are playing pool
03:15.05epabloHi people..
03:15.18fileKatty!
03:15.22Kattybrookshire[home]: i can't help it if i call him while he's playing pool!
03:15.24epabloOn what var is the result/return code of a call get set
03:15.30Kattyfile: yay, file!
03:16.07fileheyyyyyyyy
03:16.09filewazzup?
03:16.31Kattymy muscle content
03:16.42Kattyforearms ripple now :>
03:16.46epabloIf I dial, lets say zap.  On what var do I get my diconnect cause?
03:17.01fileooh
03:17.47docEep its lemme check
03:17.48docEI have it
03:17.52supaigtrclyrrad: What was the fix.
03:18.10docEyou know this is on the WIKI right?
03:19.06epablolet me see if I can find it, sorry for my lazyness.  I think it is late and I'm low on tea
03:19.48docEhttp://www.voip-info.org/wiki/view/Asterisk+variables
03:20.13epabloThanks!
03:20.14docEGet some JOLT, Mt Dew, or something
03:20.48docEI have 4 cans of Mt Dew next to me in the Code Room
03:20.55Kattyi /will/ have these forearms: http://www.propstore.com/images/products/410/tombraid-angjolgundisplay2.jpg
03:21.35docEKat do you  have a picture of you now?
03:21.40epabloThose cans give me big headaches.. I think it is the Caffeine level, or the other stuff they put in them
03:21.47KattydocE: 92 of them, at last count.
03:21.57docEJust caffeen!   they are good!
03:22.06docECan I have the URL?
03:22.09KattydocE: no
03:22.13docE:(
03:22.17docEwhy?
03:22.25docEIm not a bad person..  Just curious..   :D
03:22.26Kattyi don't know you
03:22.37docEok who in here knows me?
03:22.54*** join/#asterisk chrisdavid (n=cjones@69.90.193.151.novuscom.net)
03:22.57wunderkinsure i know you docE
03:22.59docETwisted or Damin could vouch for me if they were in here..  but I think they are down stairs
03:23.02wunderkin/msg docE whats the url?
03:23.20docESorry wouldnt give it out if she gave it to me.
03:23.30docENot in my nature to kill someones trust.
03:24.13epabloKatty:  Blur out the face (TV style) and let us see you. ;)
03:24.48Kattyfile: zomg, lookit! http://perso.wanadoo.fr/psylo-vision/_images/wallpapers/_first/Tomb-Raider-cot%E9.jpg
03:24.52Kattyfile: purrty!
03:25.42KattydocE: just because twisted knows you doesn't make it All Better
03:25.45fileoooooooh
03:26.09docEsigh..
03:26.11docEwomen
03:26.29KattydocE: oh yes, i'm unreasonable now because i don't want to share.
03:26.34KattydocE: shame on me.
03:26.43docEhehe..  no not even like that..
03:26.51Kattyk
03:26.54docEWomen just like to be secretive..   thats all..   :)
03:27.04docEMy wife does the same thing to me and drives me nuts
03:27.05file[laptop]Katty is silly
03:27.19Kattyfile[laptop]: you'll have to braid my hair like miss croft at the next convention!
03:27.40file[laptop]oh no!!!
03:27.49Kattyoh yes!
03:27.49*** part/#asterisk Sedorox (n=Brandon@smartserv/cna/Sedorox)
03:27.51Kattyyes you will
03:28.03Kattyor i will steal your danish and OJ
03:28.35file[laptop]nooooooo :(
03:29.03Kattyhair or death! little red cookbook!
03:29.33file[laptop]redrum
03:29.38L|NUXplease donate www.pakistanhelp.com
03:30.55file[laptop]Katty: how was your day?
03:31.30Kattyfile[laptop]: medium rare.
03:31.40file[laptop]ooh
03:31.43Kattyfile[laptop]: with a hint of windows, actually
03:31.51Kattysilly routing and remote access causing me headaches
03:31.51*** join/#asterisk jdv79 (n=jdv79@u1057064.ul.warwick.net)
03:31.55jdv79Oct 12 23:31:12 WARNING[10400]: chan_zap.c:1573 zt_set_hook: zt hook failed: Device or resource busy
03:31.58jdv79anyone?
03:31.59Kattyi threatened it with a stick
03:32.08Kattyand by stick i mean reconfigured it
03:34.02*** join/#asterisk fiber0pti (n=johndoe@pcp01876618pcs.sandia01.nm.comcast.net)
03:34.23supaigtrIf you had to get something fixed and had to bet on HEAD is broken or the zap card you aren't using is causing audio problems with IAX2 to IAX2 which one would you go with by morning.  Revert to stable, or stay with HEAD but figure out how to disable the zaptel card remotely and use ztdummy.
03:34.46fiber0ptiAnyone have any suggestions for some good and inexpensive voip phones? Somewhere between $75-$150 retail..
03:34.57clyrradCAD or USD?
03:35.17supaigtrclyrrad: What was the fix?
03:35.30clyrradjitterbuffer = yes :) :) :)
03:35.38*** join/#asterisk tim_scott (n=war@d198-166-220-253.abhsia.telus.net)
03:35.41tim_scottHello all.
03:35.50tim_scottIs there anyone here who could spare a moment to answer a question or two?
03:36.30supaigtrclyrrad: I think my audio loss is the fact that the machine still has a zaptel TDM card but it isn't used.
03:37.32supaigtrtim_scott: Ask away.
03:37.32tim_scottAlrighty.
03:37.32*** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net)
03:37.54tim_scottOkay.
03:38.12clyrraddid you disable all intrupts that your not using in your BIOS? I read that can screw things up
03:38.31epabloAnyone know where a file generated call returns it's disconnect cause.  Just checked dialstatus and hangupcause.. with no luck
03:38.46tim_scottI doing a presentation on Asterisk on Friday at ny local college. I was going to set up an Linux machine running Asterisk there, which would communicate over IAX with a machine at work.
03:39.15Soulz---hello all
03:39.18tim_scottThe college has port 4569 blocked, so I set the bindport=80 in iax.conf, on both the machine at work and at college.
03:39.30Soulz---when ever i dial a sip/iax2 extenion, it always goes to voicemail
03:39.37Soulz---though the phones are working
03:39.44supaigtrI did but it still had problems.  I unloaded the module but nobody here seems to know if that will make things work. I know it doesn't show up in interrupts now.  The problem is intermittent and I won't know until tommorow. By then they'll be ready to rip * out of the business.
03:39.47tim_scottthe work machine will not register with the machine at college, but I can still place calls from work -> college.
03:40.04tim_scottBut I can't make calls from college -> work, because "work" is the machine doing the register=>'ing.
03:40.07*** join/#asterisk st3v (n=st3v@netblock-66-245-213-120.dslextreme.com)
03:40.19tim_scottAnyone have any suggestions? Where is the registration traffic going? :/
03:40.31Soulz---can anyone give me to some pointers on where i should look?
03:40.31*** join/#asterisk kb1_kanobe (n=krisbout@h24-207-96-50.cst.dccnet.com)
03:40.42epablotim_scott:  sounds like a FireWall problem
03:40.59clyrradsupaigrt.... LOL yah some people just wont get it
03:41.11kb1_kanobeevening all.
03:41.12tim_scottSo where does the registration traffic go?
03:41.28kb1_kanobeanything scary in cvs-head at the moment?
03:41.32tim_scottThe firewall port cannot be unblocked, it's their security policy to be a massive pain in the ass.
03:41.41Kattyanyone know who's hosting the cluecon gallery?
03:41.49Kattyor, more importantly, where Junky's gallery is?
03:41.58Kattycause i know who's hosting it
03:41.59docEIm hosting Astricon's..  :)
03:42.14Soulz---when ever i dial a sip/iax2 extenion, it always goes to voicemail, anyone?
03:42.21st3vI am setting up an asterisk server, using a Rhino channel bank. Should I get the 24 port POTS FXS bank and replace one of the 4 port FXS modules with a $380 FXO module, or just keep the channel bank all FXS, and get the digium card with 4 FXO ports built in?
03:42.49tim_scottSoulz, can you be more specific?
03:42.52Vcowell
03:42.56Vcodo the math
03:43.14Soulz---timscott: when ever i dial from different ip phones (internally and externally)
03:43.23Soulz---all sip calls/iax seems to go to vm
03:43.27st3vI mean performance wise, they are almost the same price.
03:43.43epablotim_scott: You are right can't be the FW.. both side are sure to have port 80 in/out open
03:44.02Vcowhats a TDM40B go for?
03:44.07epablotim_scott: unless it is set only for specific IP's
03:44.15supaigtrTCP UDP theres a difference  http is TCP
03:44.34st3v<PROTECTED>
03:44.44tim_scott:/
03:44.58st3vI would get the TDM04B
03:45.48docEyes and asterisk only uses UDP ports.  Not tcp
03:45.50tim_scottsupaigtr: I'm not following. Was that comment directed at me? :/
03:46.02Vcomore importantly, will you need the extra space for FXS later...
03:46.13Soulz---externally meaning when external did calls a internal sip extension
03:46.21Soulz---i get to go straight to vm
03:46.24Vcoeither way you're going to need a card if thats the case
03:46.26Soulz---this is quite strange
03:46.33Vcoso then it really wouldn't matter
03:46.34st3vwell we only need 18 extensions now, but maybe more later
03:46.36glm2kst3v: same functionality. are you on a budget?
03:46.58Vcogo channelbank...
03:47.10glm2ki would do the same as well.
03:47.15glm2ksaves a pci slot too
03:47.36Soulz---http://pastebin.ca/25359
03:47.45Vcoand who knows...maybe you'll need more that the 4 ports later...
03:47.52Vcomaking the TDM card a waste
03:47.54supaigtrtim_scott: Yep.  UDP is likely blocked
03:48.02Soulz---AGI Script dialparties.agi completed, returning 0 or Everyone is busy/congested at this time
03:48.08Soulz---dunno where i should look
03:48.12st3vyeah
03:50.22*** join/#asterisk bmg505 (n=leon@rndf-146-13-10.telkomadsl.co.za)
03:50.25tim_scottsupaigter: Possible. I'll find out.
03:50.27tim_scottThanks.
03:50.49epablotim_scott: Colleges normally block UDP.. At least mine did
03:50.49kb1_kanobeis it possible to delete specific voicemail messages from specific boxes through the manager interface?
03:51.11tim_scottSo the registration traffic is going over UDP?
03:51.39Soulz---tim: do u need more info?
03:52.35epablotim_scott: it all goes over UDP
03:54.06tim_scottUDP port what? 4569?
03:56.40epablotim_scott:  Thats the defualt.. but if you can manage the FW guy to open any of the to you.. You can set it up
03:56.47Kattyiax is 4569, me thinks
03:57.09tim_scottHe won't open up any of the ports.. URG. He said it's a security risk... >_<
03:57.15Kattypffft
03:57.19tim_scottExactly.
03:57.25tim_scottSo is there another option? Would it help if I pasted what I see on my screen?
03:57.37tim_scottI'm really screwed on this one, presentation is tommorow and friday. :/
03:57.41tim_scottAnyone have any suggestions?
03:57.57epabloCan you set up a VPN?
03:58.03Kattyyes, VPN will work
03:58.10Kattyand make yourself part of the internal network
03:58.21tim_scottI can't do any port-forwarding stuff. I can only use port 80.
03:58.23*** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au)
03:58.36Kattytim_scott: you don't have to if your firewall does vpn
03:58.41Kattytim_scott: and most do
03:58.55tim_scottThe firewall at college? Hell, I don't know.
03:59.00Kattyi'm sure it does
03:59.05Kattyeven the cheapies have vpn
03:59.12tim_scottI would assume.
03:59.19Kattythe symantec model 100 for 10 users has vpn
03:59.21tim_scottSadly, I don't know how to do that.
03:59.31epabloI don't think he will let you doit.  but you can use openvpn on both machines..
03:59.31Kattynot going to learn any quicker (=
03:59.34tim_scottI don't have enough time to mess with VPN patches.
03:59.40Kattyk
03:59.51tim_scottI have like, 8 hours before I have to start setting up.
03:59.53tim_scott:/
03:59.58Vcooh so you're saying you have a life or something..
04:00.16Vcoahh
04:00.47tim_scottI know I probably asked this already, but could someone explain to me in detail why simply setting bindport=80  in iax.conf on both sides of the firewall will not work?
04:01.01tim_scottI mean, I would assume it would work, but the one end can't register with the other. :/
04:01.12*** part/#asterisk chrisdavid (n=cjones@69.90.193.151.novuscom.net)
04:01.54tim_scottPardon my ignorance, I just don't understand why it isn't working. :S
04:02.06kb1_kanobetim_scott: sorry, just kind of coming in mid-discussion. What are you trying to do?
04:02.14tim_scottHeh >_<
04:02.23supaigtrtim_scott: They allow TCP on port 80 not UDP
04:02.35tim_scottAssuming UDP is allowed.
04:02.52epablotim_scott: Then their is no problem
04:02.54tim_scottkb1_kanobe, I'm setting up an asterisk system at my local college that needs to communicate via IAX with an Asterisk server at work.
04:03.03kb1_kanobeserver to server?
04:03.09tim_scott4569 is blocked inbound and outbound, so I set bindport=80 in my iax.conf file.
04:03.11supaigtrIt would work if udp was allowed.
04:03.12tim_scottYes, server to server.
04:03.23tim_scottWell, UDP is allowed.
04:03.30supaigtrIt should work then.
04:03.34tim_scottBut the one server isn't register=>'ing with the other.
04:03.37tim_scottSee, it's _not_ working.
04:03.39Kattyfile[laptop]: i suddenly got sleepy :<
04:03.53tim_scottWhat additional information do I need to provide to help someone help me ? :S
04:03.56kb1_kanobeI don't use register between my servers.
04:04.05*** join/#asterisk JerJer[mobile] (n=jj@dsl001-136-136.lax1.dsl.speakeasy.net)
04:04.17supaigtrThat has nothing to do with registering.  If both * boxes are doing IAX over 80 and it worked on the orignal port it should work on 80.
04:04.18kb1_kanobeas longs as udp is allowed both ways and there is no nat, it should be simple enougg.
04:04.28tim_scottThere is NAT.
04:04.32Vcoor a proxy server?
04:04.33kb1_kanobes/enougg/enough.
04:04.36supaigtrOn both sides NAT?
04:04.46tim_scottBoth sides.
04:04.57epabloBoth need to reg
04:04.59kb1_kanobeewwww... out of my league. Sorry.
04:05.12tim_scottLike I said, I can Dial() work -> college, but not back.
04:05.13supaigtrIf you can't get a public ip on one side it won't work.  You only need one side to register. Side behind nat.
04:05.44kb1_kanobecan both sides register w/ a third party server elsewhere?
04:05.45tim_scottOh.
04:05.54tim_scottsupaigtr: that sounds like it makes sense.
04:06.06tim_scottSo they can't simply register with each other?
04:06.10docEkb1_kanobe yes
04:06.32docEif bedind nat No. . You would have to setup DMZ or tunnel your port to the private IP
04:06.40kb1_kanobedocE: if he were to use notransfer on the man-in-the-middile, that could work for him...?
04:07.14tim_scottAUGH
04:07.24tim_scottThat's what I thought.
04:07.38tim_scottkanobe: latency would probably be killer if I did that.
04:07.46Vco?
04:07.59tim_scottI tried setting up a proxy at work, but bouncing my calls work -> proxy -> college added too much latency.
04:08.02kb1_kanobeunless you can get access to something in the DMZ at the site, or nearby (network-wize)
04:08.24tim_scottI'll have to find out tommorow. I have a few hours setup time. *shudder*
04:08.30tim_scottLet's hope this isn't a total flop. Thanks guys.
04:08.33kb1_kanobeVPN will be the way to go...
04:08.38tim_scottAt least my suspicians were confirmed.
04:08.46kb1_kanobegood luck :-/
04:08.50tim_scottI'll see if I can plug into a DMZ, that'll probably be easier.
04:08.55kb1_kanobeoptimal.
04:09.08tim_scottGoodnight.
04:09.10*** part/#asterisk tim_scott (n=war@d198-166-220-253.abhsia.telus.net)
04:10.01docEthe man in the middle option would not work for the nat issue
04:10.45kb1_kanobeNo? Damn nat. I assumed he'd be able to register both with a 3rd party and, as long as notransfer was enabled then all rtp traffic would relay through there.
04:15.22fiber0ptiDoes anyone have experience with setting up multiple Sipura SPA-841 phones with asterisk?
04:17.18Kattynewp.
04:20.22docEyes
04:20.23docEwhy?
04:20.46docEthey are like setting up any other ATA or Sip phone.. You assign extensions and they are happy
04:20.59Kattyand they frolic about the network
04:21.00fiber0pticool.. so it was easy?
04:21.08fiber0ptiI might be setting up 100 of them
04:21.12Kattywith their happy rtp packets.
04:21.15fiber0ptiI'm looking at them because of price
04:21.50Kattythe only phones i've really worked with are the polycom500s
04:22.11fiber0ptiKatty; I will be setting up 10 of those soon.. any suggestions?
04:22.16fiber0ptianything that stood out?
04:22.31Kattyfiber0pti: if you're going to setup a .wav file for a ringtone, make sure it's ulaw 8 mono
04:22.44fiber0ptihehe.. k
04:22.58docEif you setup 100 of them might I suggest using a tftp config on the phones
04:23.05docEit will make your life MUCH easier trust me!
04:23.05fiber0ptiI got 16 of them from ebay for $100 a piece
04:23.09Kattytftp--
04:23.11Kattyftp++
04:23.24Kattyunless you have a firewall
04:23.25supaigtrftp or https are only options for the poly
04:23.29fiber0ptidocE: I've used tFTP on a few cisco 7960.. it was great
04:23.31Kattywhich i hope you do, since you have IP phones.
04:23.39supaigtrThe thing that sucks is buddy lists and DHCP.
04:23.41Kattysupaigtr: no, they do tftp as well
04:23.52supaigtrNot with the newest bootrom.
04:23.55Kattysupaigtr: that's what i originally tried. and then decided it was silly.
04:24.04Kattysilly like a...
04:24.06Katty...balloon.
04:24.09fiber0ptiI would actually prefer ftp over tftp just becuase of some issues I've ran into with tftp
04:24.26supaigtr:)  Katty..  How did you handle DSS buttons / console?
04:24.27HiltonThhmmm, can AMP be run with Xorcom Rapid?
04:24.36docEat anyrate fiber good luck with that..  All I can say is take a week and plan out your deployment.  Cause a poorly planned deploymnet will make your life hell
04:24.44*** join/#asterisk brookshire[home] (n=matt@esbrooks3.traveller.com)
04:24.54Kattybrookshire[home]: yay!
04:24.54fiber0ptidoce: thanks.. I will try my best
04:25.46Kattyfiber0pti: and if your polycom 500s require rebooting a lot...let me know.
04:25.53docEbut if you need help by all means do ask questions..  I believe the only dumb question is the one not asked..
04:26.01Kattyfiber0pti: i haven't /quite/ figured out why mine do that. like 5 times during the day.
04:26.05docEI am usually here at docE, Docelm0 or Docelmo
04:26.12Kattyfiber0pti: there's this weird issue of they can hear you, but you can't hear them
04:26.16fiber0ptiweird.. how long do they take to come up after rebooting?
04:26.17Kattyfiber0pti: yet, it's not an rtp problem
04:26.27*** join/#asterisk _Thor (i=Christia@user-vc8fl7l.biz.mindspring.com)
04:26.33Kattyfiber0pti: i'm all confuzzled and want to look at sip debug, but haven't had a chance yet
04:27.03fiber0ptiKAtty: interesting.. does the reboot render them useless at that time? What if someone is on the phone?
04:27.09supaigtrKatty: is that an intermittent problem?
04:27.11docEanywho..  off for niccotine bbiaf..
04:27.13Kattysupaigtr: no
04:27.18supaigtrHmm.
04:27.22Kattyfiber0pti: hmm?
04:27.27Kattyfiber0pti: they reboot and the phone is fine
04:27.33supaigtrThe reboot is usually from timestamp changing on FTP.
04:27.35Kattyfiber0pti: tis very odd.
04:27.44Kattywe're not using ftp
04:27.45fiber0ptiindeed.. I will see if mine do it too
04:27.51Kattywe're MANUALLY rebooting
04:27.59Kattykthx
04:28.05Kattyfiber0pti: excellent
04:28.25Kattyalso! this is a girly rant.
04:28.28Kattyno need for solutions.
04:28.35fiber0ptihehe..
04:28.39enderI use IP301s and IP501s.  I've never had to reboot them aside from config changes.
04:28.49fiber0ptigood to know, as well
04:29.00supaigtrWe built a we UI to mass configure phones here. IT works ok but we have problems with * still.
04:31.20JerJer[mobile]girly rants are fun
04:31.32HiltonTgirly pants are more fun
04:31.40Kattybye.
04:31.41JerJer[mobile]if they are on the floor
04:31.49JerJer[mobile]damn i scared katty off   :(
04:35.00DaminKatty: Was someone talking shit to you?
04:35.36Kattymrow?
04:36.33KattyDamin: have you insaned?
04:36.46Vcoi think JerJer was making some mention of wanting to wear her pants......
04:36.49KattyDamin: why would /anyone/ do something like that?
04:38.42KattyDamin: that's what i thought.
04:39.25Kattysupaigtr: sorry, you missed half the conversation.
04:39.26*** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com)
04:40.07DaminI think I missed half the conversation too..
04:40.14Kattyexactly.
04:40.15DaminKatty: Should I smack him for you?
04:40.32Kattyviolence never solved anything except deadlocks.
04:40.41MikeJ[Laptop]yes.. you should smack him
04:40.45MikeJ[Laptop]:P
04:40.56Kattyright. now i'm quite confused.
04:41.02MikeJ[Laptop]hey Damin... what the hell are you doing online.. go get a drink boi
04:41.06Kattybecause Brain Surgery Sucks. or so i hear.
04:41.52DaminMikeJ[Laptop]: I would.. but I don't know where people are drinking at...
04:41.59MikeJ[Laptop]the bar?
04:42.00DaminMikeJ[Laptop]: And I'm catching up on Email right now..
04:42.00Kattyi think i'll just stop while i'm ahead (=
04:42.12*** join/#asterisk Eight (n=blake@12-227-171-175.client.mchsi.com)
04:42.16DaminKatty: Probably a good idea.. Maybe I should smack you?
04:42.30KattyDamin: not a very wise idea.
04:42.32*** join/#asterisk dudes (n=dudes@12-215-34-84.client.mchsi.com)
04:42.34MikeJ[Laptop]yay cygwin!!!
04:42.42DaminMikeJ[Laptop]: Huh?
04:42.43MikeJ[Laptop]soooo close now
04:42.45*** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim)
04:42.53MikeJ[Laptop]almost have it all compiling
04:42.59docEYAY!
04:43.07MikeJ[Laptop]figured out all the makefile ugliness
04:43.18MikeJ[Laptop]and figured out how to do it not sooo ugly
04:43.27Kattywhere is twisted?
04:43.39JerJer[mobile]last i heard they were going to dinner
04:43.42DamindocE: Who are you?
04:43.50MikeJ[Laptop]and did it without all the junk that the asteriskwin32.com guy had to do to make a loader and do all kninds of weird stuff...
04:44.21MikeJ[Laptop]so I still build .so files on top of dll's so all the module loader stuff does not have to be totally re-written
04:44.38Kattyg'night.
04:44.56MikeJ[Laptop]which is good
04:45.10docEIm sitting across from you dude..
04:45.13MikeJ[Laptop]makes the makefiles a bit fun tho
04:45.20MikeJ[Laptop]not from me
04:45.25*** join/#asterisk kshumard_home (n=ksh@pcp01931374pcs.huntsv01.al.comcast.net)
04:45.29MikeJ[Laptop]cuz I'm in deroit
04:45.43MikeJ[Laptop]:P
04:46.02JerJer[mobile]detoliet
04:46.07MikeJ[Laptop]yes
04:46.11MikeJ[Laptop]is mark there ?
04:46.16JerJer[mobile]somewhere
04:46.27MikeJ[Laptop]tell that boi to get a drink...
04:46.28JerJer[mobile]Alison is too  :)
04:46.32DaminMikeJ[Laptop]: Mark is totally pissed off at you dude.
04:46.40MikeJ[Laptop]at me?
04:46.41MikeJ[Laptop]why?
04:46.43DaminMikeJ[Laptop]: Yeah..
04:46.55DaminMikeJ[Laptop]: I don't know..
04:47.02MikeJ[Laptop]well that's silly
04:47.17DaminMikeJ[Laptop]: What did you do?
04:47.20MikeJ[Laptop]dunno
04:47.32Kattywhy don't you /ask/ like grownups?
04:47.34MikeJ[Laptop]been a stranger sense I got back from boston..
04:47.42MikeJ[Laptop]ummm.. ask what?
04:47.48Kattyif he's upset with you.
04:48.01Kattyor i suppose you could do the girly could shoulder thing for a few years.
04:48.08Kattys/could/cold/
04:48.10MikeJ[Laptop]well.. I would assume if somone is mad at me, that they would say somthing to me
04:48.17HiltonTI like the girly cold shoulder thing  :)
04:48.23Kattyright. i was going to bed.
04:48.33MikeJ[Laptop]so I am guessing that they are just joking w/ me
04:48.36HiltonTnite Katty
04:48.42MikeJ[Laptop]if not.. then that's just silly
04:48.50HiltonTMikeJ[Laptop]; 'zactly
04:49.20*** part/#asterisk Uberbot (n=Uberbot@69.252.219.76)
04:50.17filetomorrow is shopping time... oh joy
04:50.23MikeJ[Laptop]for me?
04:50.27filenope!
04:50.30MikeJ[Laptop]:(
04:50.33MikeJ[Laptop]why not?
04:51.28jetsFILE
04:51.36filewell if you want to pay for it all, sure - then shopping time
04:51.37HiltonTbecause you are a dude, MikeJ[Laptop], and shopping isn't something we enjoy doing
04:51.37fileoh no it's jets
04:51.37jetswhat are you shopping for hMMM?
04:51.44filesneakers.
04:51.49jetsoh whatever have u even talked to me this whole trip HMMM?
04:51.50DaminOK..
04:51.52Damingoing to drink..
04:52.02*** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl)
04:52.03jetswho's drinking and where? :)
04:52.13docEProbably in the bar..
04:52.31docEJet arent you like 18 or something?
04:52.58jetsNope i'm 22 bitches, i was at the bar lastnight with a bunch of geeks
04:53.04jetswhich bar
04:53.24docEthe one down stairs..   :)
04:53.31docEI am in the code room
04:53.36docEI may go drink..  dunno..
04:54.09Vcothats not a br, that the gay crack house
04:54.35docEwho are you?
04:55.05jetsi'm brian... fun hair
04:55.09jetsgonna speak about callmanager
04:55.14jetsbeing replaced by *
04:55.32*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
04:55.41HiltonTCallManager == Hell Expensive!
04:55.47HiltonT(well, it IS Cisco!!!)
04:56.09VcoAnd thats even coming from someone with the name Hilton.........
04:56.10jetscome on baby light my fire!
04:56.14HiltonTlol
04:56.19jetsheh
04:56.23HiltonTI look nothing like Paris (thank fsck)
04:56.24jetsi'm a hilton
04:57.21*** join/#asterisk bweschke (n=bweschke@m810f36d0.tmodns.net)
04:59.01jetstyler hilton was at von for a few hours in march
04:59.15HiltonTwho's that?
04:59.31jetsgoogle
05:01.30HiltonToh, yay - teenybop
05:01.50HiltonTanother Britney Spears...
05:02.54jetsya
05:02.59jetsfscking loaded though
05:03.01jetsand likes asterisk
05:04.03loudjets
05:04.13loudnext call manager will be SIP enabled.
05:04.18loudi saw it today.
05:04.21MikeJ[Laptop]g'night all
05:04.27loudcisco noticed they are losing money.
05:04.31bweschkeon second thought.... now that I'm back to my room I'm pretty damn tired... not gonna come back down to the code zone. good night damin and jeremy.. catch you guys tomorrow
05:05.12jetsloud: "sip enabled" doesn't 4.0 do sip already?
05:06.15jetsanything fun going on at the code zone
05:06.26loud5.0
05:06.35loudQ1 2006
05:07.28jetshrmmm no a deployment of 4.0 at a college were using soft phones and doing some sip calls to asterisk........  maybe not trunking, etc,etc, but endpoints like polycom hardphones, softphones and asterisk
05:07.52loudyou sure they are not using cisco's sip proxy ?
05:08.22jetsyep i'm sure they are using callmanager
05:08.50jetshttp://www.cisco.com/en/US/products/sw/voicesw/ps556/products_qanda_item09186a00801f8e18.shtml
05:19.15clyrradAnyone here had custom messages recorded by Alyson?
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05:23.33kb1_kanobeerrr.... has the +101 jump behaviour been depreciated, or deactived in cvs-head?
05:23.59wasimthat would not be good
05:24.17*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:24.21kb1_kanobeindeed.
05:24.40kb1_kanobeI must be missing something... more digging.
05:26.10kb1_kanobeoh, for ff.... sake. Why oh why is my shiney newly-upgraded asterisk server imploding.
05:29.31*** join/#asterisk svadoe (n=svadoe@216.230.147.187)
05:29.59kb1_kanobeHmmm... weasles appear to be eating my platters.
05:30.51*** join/#asterisk SplasPood (n=sp@brooklyn.paravolve.net)
05:36.43*** join/#asterisk jtodd (n=jtodd@host06.alica.hyatthsiagx.com)
05:42.53*** join/#asterisk niZon (n=ilt@S0106deadbeefbeef.wp.shawcable.net)
05:43.07niZonanyone here ever done any kind of point to point link over dry copper pairs?
05:43.16supaigtrYep. Everyday.
05:43.30tessierWet copper pairs tend to corrode.
05:43.41*** join/#asterisk Qwell (n=chatzill@pool-71-108-241-125.lsanca.dsl-w.verizon.net)
05:43.41niZonsupaigtr what kind of speed can you get?
05:43.41SplasPoodI prefer mine lightly salted
05:44.01supaigtrDepends on eq. Theres stuff out there to do 10 meg +
05:44.02kb1_kanobeSDSL w/old netopia 7100C were quite popular.
05:44.22niZonI'm looking to do it over about 1.5Km or so
05:44.24supaigtrGot boxes of those.  Back to back required firmware.
05:44.30niZonas fast as possible
05:44.39supaigtrniZon: just one connection or multiple?
05:44.45kb1_kanobethe 7100c could do it out of the box if memory serves.
05:44.55niZondepends on what kind of speeds i can get with one connection
05:45.06supaigtrnetopia had a firmware for the "CO" end
05:45.14supaigtr1.5 is going to be SDSL
05:45.45niZondo you have links to any info on required hw?
05:45.49supaigtrYou could get a coppermountain concentrator and bond to 6 meg or so.
05:45.55niZonhm
05:46.04supaigtrYou'd need three pair.
05:46.04kb1_kanobeniZon: http://www.pbs.org/cringely/pulpit/pulpit20010823.html
05:46.45supaigtrnet2net has some small stuff thats current tech.
05:46.53niZonok
05:47.24supaigtrFiber works better.  Microwave ain't so bad either.
05:47.32niZonI'd like to get about 10Mbit
05:47.44niZonfiber is a nono, can't get it to the building
05:47.47kb1_kanobehell, even 802.11 wifi (interference excepted)
05:47.54supaigtrI've got a pair of new waverider eq that does 10 mb.
05:48.04niZonwifi, maybe if we can get roof space
05:48.30supaigtrI'll love to get rid of it and it don't require much.  Ethernet to a little 1ft panel.  self contained.
05:48.38*** part/#asterisk epablo (n=epablo@WLL-24-pppoe197.t-net.net.ve)
05:48.43supaigtrI replaced with a 90meg link.
05:48.47kb1_kanobeniZon: http://www.star-os.com
05:49.01niZonhm
05:49.34supaigtrI think the wr is rated at 72meg but expect 10 -30 in real world.
05:49.50supaigtrThey setup in a day and work well.
05:49.55*** join/#asterisk swm_ (n=admin@digitaldatabits.net)
05:50.28niZonthey look interesting
05:51.17supaigtrNLOS but I recommend little in the way. Couple trees isn't a big deal but it won't work thru mountain of dirt and rock.
05:51.22Vcomotorola canopy wireless
05:51.24Vco900mhz
05:51.28supaigtrSucks!!!!
05:51.45supaigtrI've got 2 of those systems in boxes.
05:51.56Vcofired them up ?
05:52.00niZon900mhz is a tad slow..
05:52.27supaigtrYea for about 6 months.  too much in that spectrum even in rural areas.  Most pager system interfer.
05:52.49supaigtrIts hard to compete with a 120watt pager system in more areas than we have cell service.
05:52.50Vcoi noticed they have new higher speed backhauls now too...
05:53.07supaigtrthe waverider is 5.8
05:53.10supaigtriSM band
05:53.42swm_What happened to S2SM band?
05:53.47Vcois still say this is cool
05:53.48Vcohttp://cgi.ebay.com/ebaymotors/Mobile-Cellular-Cell-Tower-Truck-Antennas-Ham-Radio_W0QQcmdZViewItemQQcategoryZ63739QQitemZ4581983789QQrdZ1
05:53.52supaigtrTHe only microwave backhaul that is realiable for carrier use is the 10+gig licensed bands.
05:54.13Vcojust paint WARDRIVER across the sides and drive around
05:54.56supaigtrNo last mile wireless works good unless its point to point and well designed.
05:55.20supaigtrPlayed with mesh but we named it mush networking :)
05:55.43supaigtrFiber is sooooo much better and more realistic in meeting demands.
05:55.58Vcooooh.. they have 300mb backhauls actually..
05:56.07supaigtrThats not user rates.
05:56.49supaigtrA good 655mb sonet microwave system starts at about 120,000 installed and licensed.
05:56.50swm_Anyone tried infrared transmissions with a high powered foocused infrared beam between two points?
05:57.49Vcoyou mean a laser pointer, morse code chart and a twitchy finger?
05:57.57kb1_kanobenever tried it, but canon used to make an atm155 system using short range lasers.
05:58.41supaigtrThat went belly up. canobeam or something like that. It looked like an atm camera.
05:58.49kb1_kanobeyeah.
05:59.06supaigtrSaw one on ebay for around 8000 a few months ago.
05:59.20kb1_kanobelol - I think I saw same.
05:59.28swm_Well a 130 Watt Laster (not millawatt like those stupid pens) but, it can go clear half way around the world, with repeaters of course. 35.8 miles and you need another reeater or your shooting off earth into the space
06:00.19kb1_kanobebah - piggyback on one of those reflectors they left up on the moon for earth-moon distance measurements.
06:00.22kb1_kanobe;-)
06:00.31supaigtrAnyone have any sonet hardware?  whiterock or dmx?
06:01.04supaigtrIf you're will to break the rules there all sort of things to do.
06:01.05tuppa$INSERT_OBLIGATORY_AUSTIN_POWERS_LASER_JOKE_HERE
06:02.51*** join/#asterisk mthem (i=merlintm@64.235.245.133)
06:02.54swm_Why not install a 250,000,000,000 Watt Laster powered from a 680 array of satellite solar conductors with a huge 65,000 Micro Ferit capacitors in outer space and nuke things around the worls, starting with Geoge Bush
06:03.01*** join/#asterisk srfrog (n=cag@209-250-4-64.convergentaz.net)
06:03.52supaigtrnite ppl
06:03.54Vcobecaue that would make you some sort ofpinko commie terrorist
06:04.18*** part/#asterisk srfrog (n=cag@209-250-4-64.convergentaz.net)
06:04.38*** join/#asterisk gres (n=serg@62.152.85.99)
06:04.41swm_Yeah totall self destructive too... Launches it into deep space and explodes heh... Not recoverable, no proof
06:05.15*** join/#asterisk NoRemorse (n=axel@202.161.68.2)
06:05.47NoRemorsehi, does OH33 use a hardwired CID? (ie the "aliases" in oh323.conf) or can it pass on correct CID for each call?
06:05.53NoRemorse*OH323
06:06.29HiltonTok - is there anywhere I can find a doc that lets me get my head around the extensions.conf file and Asterisk dialplans?  I guess that this is the thing I need to grok before I move forward
06:06.47NoRemorseextensions.conf.sample
06:07.36swm_-BOOM-BOOM-BOOM-BOOM-BOOM Bannnnnnnggggggggggggggggggggggggggggggg
06:07.40HiltonTdoesn't exist in Xorcom Rapid  :(
06:20.12*** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1)
06:20.36*** part/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1)
06:23.08*** join/#asterisk dasuberdavid (n=david@dsl001-136-136.lax1.dsl.speakeasy.net)
06:26.22*** join/#asterisk JerJer[mobile] (n=jj@68.123.154.34)
06:27.14JerJer[mobile]MikeJ[Laptop]:  :(
06:27.45*** join/#asterisk brookshire[home] (n=matt@esbrooks3.traveller.com)
06:28.44FuriousGeorgeanyone ever install a doorphone?
06:29.27*** join/#asterisk bongfrog (n=winston@dsl001-136-136.lax1.dsl.speakeasy.net)
06:29.41brookshire[home]glorified doorbell?
06:29.49brookshire[home]:)
06:31.46HiltonTFuriousGeorge; yeah, we have as part of a Home Security package, but not (yet) a VOIP one
06:36.12*** join/#asterisk [hC] (n=hardcore@dsl001-136-136.lax1.dsl.speakeasy.net)
06:37.00HiltonTbut, if I can get my head around Asterisk, that's likely to change
06:38.06*** join/#asterisk jeffik (n=Administ@CPE0011505c92d3-CM014350000760.cpe.net.cable.rogers.com)
06:38.14FuriousGeorgebrookshire[home]: yeah, but its for a business in the city so you dont wanna just buzz anyone in
06:38.22*** part/#asterisk NoRemorse (n=axel@202.161.68.2)
06:38.27FuriousGeorgeso its only glorified in a very practical way
06:38.39HiltonTwe do this for residential installs
06:38.48HiltonT(and I mean EXPENSIVE res installs)
06:39.08FuriousGeorgeHiltonT: b/c you sell proprietary hogwash :)
06:39.26HiltonTwhere there's a tradesman's entrance to the house which can be remotely opened and locked, as can the internal door from that room to the rest of the house
06:39.32HiltonTFuriousGeorge; yes, that we do!  :)
06:39.36HiltonTand I want to change this
06:39.54FuriousGeorgeHiltonT: you've come to the right place
06:40.03HiltonTthat's why I'm here!
06:40.33HiltonTI just need to grok how to config * to let me have my hardphone register to it, and I'll be underway (Xorcom Rapid)
06:40.53HiltonTand then I can figure out dialplans and such funishness
06:41.01FuriousGeorgedid you check voip-info
06:41.02wasim~docs
06:41.04jbotit has been said that 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
06:41.46HiltonTreading that now (and have been for a while now)
06:41.54HiltonTthanks jbot you rock
06:42.15HiltonThandbook-draft is, well, not really useful
06:42.29HiltonTasteriskdocs, heading over...
06:43.30brookshire[home]voip-info.org
06:43.38FuriousGeorgeHiltonT: how hard can it be? [username]  secret=,dynamic=,allow=,regexten=,callerid=,etc.
06:43.41brookshire[home]:)
06:43.47FuriousGeorgeHiltonT: voip-info.org is where its at
06:44.06HiltonTyeah, but there's sooooo much there to read thru to get to the real stuff  :)
06:44.23brookshire[home]asterisk o'reilly book?
06:44.26brookshire[home]:)
06:44.35FuriousGeorgebrookshire[home]: got it, havent looked at it yet
06:44.37HiltonTvoip-info
06:44.42*** join/#asterisk jets (n=jets@dsl001-136-136.lax1.dsl.speakeasy.net)
06:44.50jetsI MIGHT BE
06:44.51jetsi might not be
06:44.53jetshmmM!
06:44.56FuriousGeorgeHiltonT: just search for Xircom
06:45.15FuriousGeorgeor Xorcom or whatever
06:45.18Daminhttp://www.speakeasy.org/~gomez/owlhouse/images/misc/CamelToads2.jpg
06:45.18HiltonTor maybe even Xorcom  :)
06:46.11HiltonTcamel toads - rofl
06:46.33brookshire[home]Xircom was an old modem, lol
06:46.41HiltonTyup
06:48.19rikstahahaha
06:48.27mutilatoranyone know how to fix win2k only showing one processor when 2 are installed? and using HT so it should actually show 4 processors..
06:49.02brookshire[home]umm... no
06:49.07HiltonTFuriousGeorge; sure, but where does that go...
06:49.33HiltonTmutilator; was W2K installed in SMP mode, or was the 2nd CPU added later?
06:51.34FuriousGeorgeHiltonT: why, in sip.conf of course, sometime after register, if youve got one of those
06:51.45HiltonTsip.conf?
06:51.54HiltonToh, someone earlier said extensions.conf
06:51.57mutilatornot sure, i got it tho
06:52.00HiltonTNFwonder I'm lost!
06:52.09mutilatorhad to change my computer type to ACPI multiprocessor PC
06:52.13mutilatorwas Standard PC
06:52.21FuriousGeorge[general] blah blah blah register=> blah blah [user] blah blah [peer] blah blah [friend] blah blah <---  sip.conf
06:52.29HiltonTstandard PC is single CPU most definitely
06:52.37mutilatoryea
06:52.40HiltonTFuriousGeorge; ta, blah, ta
06:52.48mutilatorfound teh ms docs
06:52.57mutilatortook longer than i expected googling ~3 minutes
06:52.58HiltonTand MS doesn't support the change like that to make it SMP properly
06:53.00mutilatorso i asked here in the mean time
06:53.04brookshire[home]well.. extensions.conf pretty much ties everything together
06:53.05FuriousGeorgeHiltonT: seriously, thats where it goes in the doc
06:53.12brookshire[home]but sip.conf setups your sip channels
06:53.24HiltonTyeah, I gathered, I "blah" a lot, too, that's all  :)
06:53.44FuriousGeorgeyou got your general section with a register=>  and section for friends peers and users
06:54.05FuriousGeorgewhats the problem?  if you can grasp that go fill in the values from the example
06:54.11FuriousGeorgedocs
06:54.17HiltonTthere's no problem
06:54.20HiltonTI said "thanks"
06:54.27FuriousGeorgeoh
06:54.29FuriousGeorgeyour welcom
06:54.32HiltonT:)
06:54.55FuriousGeorgeso you got your hardphone going?
06:55.00HiltonTnot yet
06:55.15HiltonTI only installed * this morning, never played before
06:55.37HiltonT(tho, the hardphone was working with AstraTEL (an Aussie SIP supplier) earlier)
06:56.16FuriousGeorgeHiltonT: you know how to change settings in the phone to point it to ur * server?
06:56.19mutilatorwell shit
06:56.21mutilatorthat wasn't good
06:56.30mutilatorBSOD changing it to acpi multiproc
06:56.32FuriousGeorgemutilator: rm -rf?
06:56.34HiltonTFuriousGeorge; know how to drive the phone fine, just not *
06:56.45HiltonTmutilator; exactly why they don't support it  :)
06:56.55HiltonTFuriousGeorge; fdisk works better
06:57.08mutilatoryeh our other server works fine tho =\
06:57.25HiltonTreally, running Win 2K these days is asking for problems
06:57.34FuriousGeorgeHiltonT: username of phone corresponds to [ ] entry after general in sip.conf
06:57.35HiltonTesp. if it EVER connects to the Internet
06:57.58*** part/#asterisk Qwell (n=chatzill@pool-71-108-241-125.lsanca.dsl-w.verizon.net)
06:58.04FuriousGeorgesecret is obviously password
06:58.49FuriousGeorgedisallow all and allow a codec that will work with the phone
06:59.25FuriousGeorgepick a context for the phone (maybe something like admin_caller, or outgoing or something)
06:59.37FuriousGeorgethat covers the essentials
07:00.00HiltonThhmmm, me needs to do some more reading ... "context"?
07:00.06HiltonTwas fine up until there!
07:00.32HiltonTand I gather [user] would be [4001] or whatever the extension number will be
07:00.36FuriousGeorgedont think to much into it.  an incoming call has a context "its incoming"
07:00.39brookshire[home]http://www.asterisk.org/glossary
07:00.41brookshire[home]:)
07:00.44mutilatorgood thing they added the "last known good config" boot option
07:00.53HiltonTmutilator; definitely  :)
07:00.57FuriousGeorgeif its from your sip provider thats a context
07:01.15dan__thi
07:01.17brookshire[home]context is like a group
07:01.20dan__ti'm back, who missed me.
07:01.32brookshire[home][default]
07:01.35HiltonTnah, want this to be internal only right now, then expand it to go external, then use AstreSIP (for now) to terminate to a POTS number
07:01.36brookshire[home][menu[
07:01.41brookshire[home][menu]
07:01.42brookshire[home]etc
07:02.03FuriousGeorgeHiltonT: oh yeah i forgot, you gotta make your hardphone a friend (type=friend) if u want it to take and make calls
07:02.06FuriousGeorgethats important
07:02.15HiltonTI really need to read more!
07:02.51HiltonTnow I'm totally in the dark!
07:02.56FuriousGeorgeHiltonT: you can get your phone logged in but it wont make or take calls till u hook that up in the dialplan
07:03.29HiltonTI can't make it log in - that's why I need to read up on how * handles extensions
07:03.41HiltonTand half you said made sense!
07:03.43HiltonT:)
07:03.52brookshire[home]well.. first you need to get the channel working
07:03.56HiltonThalf *what* you said made sense  :)
07:04.08FuriousGeorgeits easy, under [hardphone] in sip.conf you say context=hardphone, and under [hardphone] in extensions.conf you do a exten=> _X.,1,dial(sip/provider/${EXTEN})
07:04.09HiltonTfirst I need to understand what I'm doing!
07:04.13brookshire[home]then you can hook it into extensions.conf :)
07:04.33HiltonTsure, for ppl who understand this!
07:04.47FuriousGeorgeHiltonT: forget calling, lets focus on logging your phone in
07:04.56HiltonTsure
07:05.00*** join/#asterisk \PsyKo\ (n=xxxxxxxx@golia.caltanet.it)
07:05.23FuriousGeorgethe first section of sip.conf is general, forget that for now, the second section is for peers,users,and friends (peer+user=friend)
07:05.48HiltonTok, but there's no "second" part in Xorcom default install
07:05.53HiltonT(listening)
07:05.58FuriousGeorgemake a friend entry for your hardphone there.  whats in the [] is its username  what comes after the secret= on a subsequent line, is the pw
07:06.08*** join/#asterisk voipguy (n=voipguy@196.200.26.42)
07:06.10HiltonTusername = extension number?
07:06.30FuriousGeorgeusername in the phone = whats in the [ ] in sip.conf
07:06.31FuriousGeorgeget it
07:06.44Dr_Ray[ray]
07:06.45HiltonTyup
07:06.49Dr_Rayextension=100
07:06.56Dr_Rayuser = ray
07:07.05Dr_Raysecret = ray
07:07.12Dr_Rayerm username
07:07.18FuriousGeorgeyeah username
07:07.52FuriousGeorgeand i believe its regexten=101
07:08.15FuriousGeorgetype=friend , you probably want
07:08.25kb1_kanobeis newjb in stable or only cvs-head?
07:08.39HiltonTok, got that done
07:08.48FuriousGeorgekb1_kanobe: afaik, only 1.0.9 is stable
07:08.54FuriousGeorge1.0.X
07:09.09HiltonTwith regexten
07:09.35FuriousGeorgefire up a console
07:09.39HiltonTdone
07:09.39FuriousGeorgeasterisk -rvvvvvvvv
07:09.50FuriousGeorgeset verbose 50
07:09.58FuriousGeorgesip show peers
07:10.09*** join/#asterisk mbranca (n=matteo@host-210-mi.espia-net.net)
07:10.18HiltonTdone
07:10.37FuriousGeorgeand
07:10.49HiltonTshows 501 - 510, none of which is what I just configured (4001)
07:11.02FuriousGeorgeoh no, u using a@h or something
07:11.11HiltonTXorcom
07:11.29voipguyany asteriskwin32 users here?
07:11.33FuriousGeorgehow come you have extensions 501 -510
07:11.41HiltonTI don't want to get my head around both Linux (been quite some time) and * at the same time
07:11.51HiltonTNFI - I didn't config 'em
07:12.07FuriousGeorgeNFI?
07:12.14HiltonTaahhhhh, I see my phone trying and failing to register (with an incorrect username and such)
07:12.21HiltonTno fscking idea
07:12.40FuriousGeorge~pb
07:12.43jbotfrom memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/
07:12.46Dr_Rayisn't anything above set verbose 10 useless?
07:12.49FuriousGeorgesip.conf
07:13.14brookshire[home]above 4
07:13.16brookshire[home]i think
07:13.18brookshire[home]maybe 3
07:13.53FuriousGeorgei heard it was 11
07:14.06FuriousGeorge~verbose
07:14.08jbotfrom memory, verbose is try running the verbose option on that command and looking at output for likely problems.
07:14.13brookshire[home]69 is the only true level of verbose ;)
07:14.24Dr_Rayall I ever get is 68
07:14.25HiltonTlol - guess what my password is!
07:14.36brookshire[home]secret?
07:14.38FuriousGeorgeid harly call slurping and muffled moaning verbose
07:14.51Dr_Raythe you do me, and I'll owe you one bit
07:14.53FuriousGeorgehardly*
07:15.38brookshire[home]heh..
07:15.40brookshire[home]so yeah
07:15.49brookshire[home]not being picky
07:15.54brookshire[home]but wouldn't it be byte
07:16.02HiltonTok - I cleaned up the comments from sip.conf, and here 'tis...
07:16.05HiltonT[general]
07:16.05HiltonTcontext=disabled-sip-insecure-read-getting-started
07:16.06HiltonTport=5060
07:16.06HiltonTbindaddr=0.0.0.0
07:16.06HiltonTsrvlookup=yes
07:16.06HiltonT#include "sip-reg.d/*.conf"
07:16.07HiltonT#include "sip-phones.d/*.conf"
07:16.10HiltonT[test]
07:16.12HiltonTregexten=4001
07:16.12FuriousGeorgeHiltonT:
07:16.13HiltonTusername=test
07:16.15FuriousGeorge~pb
07:16.17jbotpastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca/
07:16.17HiltonTsecret=69
07:16.17HiltonTtype=friend
07:16.28brookshire[home]<PROTECTED>
07:16.32HiltonTsorry  :)
07:17.04HiltonThttp://pastebin.ca/25372
07:17.15brookshire[home]much better ;)
07:17.26HiltonT:) now I know aboot it, I'll use it  :)
07:18.14brookshire[home]so.. does it register?
07:18.23HiltonTnope
07:18.34HiltonThang on...
07:19.28FuriousGeorgehttp://pastebin.ca/25373 <--  that looks good but ur commenting wrong
07:19.51FuriousGeorgeu sure you got the credentials right on your phone?
07:20.19FuriousGeorgeif you do put a context=something and a correspond [something] in extensions.conf
07:21.00HiltonTyup - user/authuser = "test" and password = "69" and sip proxy and domain = "192.blah"
07:21.13FuriousGeorgeand what does cli say
07:21.19HiltonTOct 13 17:21:04 NOTICE[2084]: chan_sip.c:7761 handle_request: Registration from 'test <sip:test@192.168.69.253>' failed for '192.168.69.27'
07:21.37*** join/#asterisk Pikoro (n=pikoro@db.sunny-net.ne.jp)
07:21.46HiltonTwhat's that aboot extensions.conf?
07:22.06FuriousGeorgeput a context=something and a correspond [something] in extensions.conf
07:22.36Pikorook, finally got my outbound calls working with sound via sip, however, i can't figure out why i can't receive incoming calls via sip.. gotta be routes but i can't find anyting in extensions.conf or sip.conf that might be affecting it
07:22.57FuriousGeorgewhat context is your general section of sip.conf in?
07:23.06Pikorofrom-trunk
07:23.15HiltonTwhereaboots in extensions.conf - me know nosing
07:23.25Pikoroahh, not talking to me :D
07:23.47HiltonTI have no idea what context it is in
07:23.58FuriousGeorgeHiltonT: extension.conf should have only a [general] [global] and [something] in it for now, one on each line
07:24.16HiltonTsome macro-stdexten stuff too
07:24.28HiltonTand macro-stdmeetme
07:24.34FuriousGeorgesave that one as extensions.conf.example
07:24.39FuriousGeorgeand make your own
07:24.51FuriousGeorgewith those three lines
07:24.57HiltonTexactly what I'm doing  :)
07:25.18FuriousGeorgePikoro: do you have a user entry for your sip provider in sip.conf
07:25.32Pikorouser entry?  yes
07:25.47HiltonTok - 3 lines with those headings
07:25.58HiltonTnowt else
07:25.58FuriousGeorgeis that in the incoming from sip context you want it to be in
07:26.11HiltonThuh?
07:26.12FuriousGeorgeHiltonT: reload at cli and see if phone registers
07:26.16HiltonT:q
07:26.19HiltonT:)
07:26.50FuriousGeorgeif it doesnt leave secret= blank and same for phone
07:27.12HiltonTfailed
07:27.41HiltonTclearing secret=
07:28.11HiltonTI still have all those extensions listed
07:28.22FuriousGeorge"reload"
07:28.27HiltonTdid that
07:28.37FuriousGeorgestop asterisk and start it
07:28.52HiltonTon debian, how?
07:29.00FuriousGeorge?
07:29.05HiltonTnever used Debian before
07:29.09FuriousGeorgejust do a stop gracefully at the cli
07:29.14HiltonT(only DeadRat)
07:29.19FuriousGeorgeand then start it with an asterisk -cvvvvvvvvvvvvvvvvvvvv
07:29.25HiltonTk
07:30.01HiltonTNo such command 'stop'
07:30.05FuriousGeorgeyour iax.conf could have peers in it too
07:30.10FuriousGeorgestop gracefully
07:30.26HiltonTsorry
07:30.43HiltonTready
07:30.48Dr_Raystop now
07:30.49FuriousGeorgeready
07:30.54HiltonTall still there  :(
07:31.04FuriousGeorgecheck iax.conf
07:31.11Dr_RayI don't like stop gracefully, it stops taking new calls
07:31.29HiltonTI have no  calls in/out, initial config  :)
07:31.30FuriousGeorgeDr_Ray: he's not quite in production yet :)
07:31.35Dr_Raytrue dat
07:31.36Dr_Ray:)
07:31.39HiltonTFAR, far from it  :)
07:32.40HiltonTok - iax.conf...
07:33.04FuriousGeorgeHiltonT: doesnt your Xorcom distro come with phone example configs?
07:33.20FuriousGeorgefor what your supposed to put in sip.conf
07:33.26HiltonTno example configs
07:33.31HiltonTjust pre-configured files
07:33.38HiltonTwhich are obviously not too great
07:33.55FuriousGeorgepreconfigured sip.conf with example config for your phone?
07:33.57HiltonThence why I'm flailing here
07:34.02HiltonTnope
07:34.08FuriousGeorgewhat phone is it?
07:34.17HiltonTnetcomm v85
07:34.20HiltonT(on eval)
07:34.25HiltonTdon't like it
07:34.50HiltonTbut, I can config it in its web interface without issue - can connect to a number of SIP providers
07:35.26HiltonTcan connect it to my AstraSIP and also (one at a time) my FWD account
07:35.46FuriousGeorgehttp://www.netcomm.com.au/VoIP/#V100
07:35.48FuriousGeorgethat thing?
07:36.14HiltonTnope
07:36.16HiltonThang on
07:36.31HiltonTthe V85 a few line sup
07:37.53FuriousGeorgechange your credentials on phone from test to 4001
07:38.06HiltonTreally basic web config - can't backup/reload data, can't pre-config phonebook nor speed dial entries...
07:38.18HiltonT2 secs...
07:39.36HiltonTstill no go
07:39.54FuriousGeorgehmm
07:40.11*** join/#asterisk Corydon76-home (i=green@pdpc/supporter/sustaining/Corydon76-home)
07:40.13FuriousGeorgeok i got an idea, get a softphone installed on a pc on that network and see if that thing gets logged in
07:40.17FuriousGeorgetry x-lite
07:40.20FuriousGeorgewww.xten.com
07:40.32HiltonTI decided a while back (nice pie for lunch) to not get frustrated at this, just to get it working - MUCH better revenge  :)
07:40.40HiltonTk, running now
07:42.00HiltonTk, failed as "test", trying "4001"
07:42.16HiltonTfailed
07:42.24HiltonTblank secret
07:42.28FuriousGeorgeso the softphone is failing too?
07:42.30HiltonTyup
07:42.46HiltonTOct 13 17:42:31 NOTICE[4607]: chan_sip.c:7761 handle_request: Registration from 'Quark HT <sip:4001@192.168.69.253>' failed for '192.168.69.28'
07:42.49mutilatorman what a pain in the ass
07:42.55HiltonTyup
07:42.59mutilatorwonder if i do a full restore
07:43.00FuriousGeorgethrow in a host=dynamic in there, and make sure you have a context=something that corresponds to extensions.conf
07:43.03mutilatorif it'll detect it
07:43.08mutilatori just updated to the latest bios
07:43.11HiltonTfull restore, or "over the top reinstall"
07:43.28HiltonThost=dynamic in where?  sip.conf?
07:43.29mutilatori don't wanna overtop reinstall our main mail/web server
07:43.37FuriousGeorgeyah sip.conf
07:44.12HiltonTthere's no context=something
07:44.20HiltonTthat under [test] as well?
07:44.34FuriousGeorgeunder test put a context=something
07:44.41FuriousGeorgein extensions.conf [something]
07:44.55HiltonTdone
07:45.54FuriousGeorgereload try to log softphone in
07:45.54HiltonTain the process
07:45.54mutilatorgrrr
07:45.54FuriousGeorgesip reload
07:45.55mutilatorthe promise drivers didn't wor
07:45.55mutilatork
07:45.55mutilatornot detecting the raid now
07:45.55mutilatorone problem after another
07:45.55FuriousGeorgepromises promises
07:45.55HiltonTsame error
07:45.55*** join/#asterisk konfuzed (n=konfuzed@H129.C72.B0.tor.eicat.ca)
07:45.55HiltonTI HATE PROMISE CONTROLLERS
07:50.25*** join/#asterisk ^X-works (n=drttrtr@81-208-62-98.ip.fastwebnet.it)
07:51.53mutilatorheh
07:52.05mutilatori shoulda stayed in bed tonite
07:52.06mutilatorand said forget it
07:52.14mutilatori wanted to be outta here an hour ago
07:53.42*** join/#asterisk [Outcast] (n=bill@222-152-255-158.jetstream.xtra.co.nz)
07:54.17[Outcast]greetings any bug marshalls in the room?
08:03.09xminganyone using chan_missdn with 1.2-beta1?
08:07.04shido6heh, poor mutilator
08:10.19xmingstrings.h seems to be gone in the beta1, while I have it in the CVSHEAD of several months ago
08:12.09xmingoops
08:12.40xmingstrings.h is still in beta1 but not on the latest CVS ?
08:15.58*** join/#asterisk Poincare (n=jefffnod@dD5779806.access.telenet.be)
08:20.07*** join/#asterisk mutilator (n=animenod@65.111.201.79)
08:20.09mutilatorargh!
08:22.35skrustyanyone know why in cvs head i keep getting "not a local SIP domain"
08:22.47skrustywhen a device tries to register
08:32.13*** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62)
08:32.28nfi|ermeshi all
08:33.06uterhow do i disable hisax?
08:34.14iDunnormmod hisax
08:35.38uterwell, i don't want it to be started at all
08:36.51wasimthen don't modprobe it
08:38.09iDunnoif you're using hotplug, add it in to the blacklist
08:40.02*** join/#asterisk darkskiez (n=darkskie@194.247.78.146)
08:40.15uteron one of my debian systems it works without blacklist
08:41.23uterwell, i think i'll try to find the difference between both systems
08:44.31*** join/#asterisk case_ (n=case@pdpc/supporter/student/case-)
08:44.31*** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net)
08:44.38case_hello
08:44.49case_i'm making some tests with Asterisk and i would like to generates some calls without any special devices, any ideas?
08:46.18*** join/#asterisk Koshatul (n=evangeli@ip157.net65.ipnetworks.net.au)
08:47.03iDunnojust set up some sip channels and use softphones?
08:48.02case_do you have a softphone to suggest?
08:49.01iDunnowell, there's xlite for windows, or kphone or gnome-meeting for linux
08:49.15*** join/#asterisk e3g (i=ee@u15157627.onlinehome-server.com)
08:50.45case_iDunno: thanks a lot
08:50.45e3ghow to allow only 1 IP Address to Hit my Asterisk???? IF SIP Registration is not there....??
08:51.32e3gI meant to say...a specific IP Address?
08:52.57nfi|ermes<PROTECTED>
08:53.29nfi|ermessometimes it works, sometimes not
08:53.38nfi|ermesi think some nat prooblem
08:53.43nfi|ermesany suggestion ?
08:53.53e3ghave you tried NAT=yes ?
08:53.57*** join/#asterisk tsetane (n=tse@212.4.33.75)
08:54.10e3gIs your asterisk at public IP Address?
08:54.29case_e3g: can't you do that with your firewall ?
08:54.57nfi|ermesn
08:55.02nfi|ermesit's in my lan
08:55.04e3gcase_:  I havent worked on firewall stuff.....I think there is LOKKIT .... or let me know about one?
08:55.20case_e3g: what is your OS ?
08:55.24e3gRH9
08:55.27nfi|ermesi tried nat=1 or nat=0
08:55.42case_e3g: i'm suggest you to have a look on firestarter
08:55.47e3gnfi|ermes :  try NAT=YES
08:56.04case_it's a gui to controll iptables, the linux kernel embended firewall
08:56.34e3gwhat about if I have minimum Installation?
08:56.47e3gor FREEBSD?
08:57.07*** join/#asterisk JoseHap (n=Jose@202-197.246.81.adsl.skynet.be)
08:57.13JoseHaphi
08:57.30JoseHapdoes anyone know poe cabling scheme of polycom phones
08:57.30case_e3g: you can controll iptables with scripts (not that hard), for netbsd i don't know
08:57.43*** join/#asterisk tsetane (n=tsetane@212.4.33.75)
08:57.45JoseHapis it possible to make it ourselves ?
08:57.47case_s/netbsd/frebsd/
08:58.10HiltonTPOE is pins 7+8 (normally)
08:58.20e3gcase_: alright....thanks
08:58.40nfi|ermese3g, it's incredible . With nat=1 , it works after i deactivate and reactivate my lan adapter in the cleint
08:58.40case_hum... can someone tell me if my gnomemeeting 1.2.1 can do SIP or if i need the cvs version?
08:58.44case_e3g: you're welcome
08:58.54JoseHapbecause the polycom 300 doesn't work with a poe 802.2af switch and std cable
08:59.02JoseHapi read i need an adaptor
08:59.47HiltonT300 is supposed to be af compliant, therefore a standard Cat5 should do it
09:00.22JoseHapon the datasheet is mentionned 802.3af and cisco poe, but optional ;-)
09:00.48HiltonTand you bought the option?
09:00.54JoseHapi test it on a small netgear poe switch which works great with grandstream phones
09:01.06*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
09:01.08JoseHapno, i didn't buy it
09:01.18HiltonTand you wonder why it doesn't work?
09:01.23HiltonT:)
09:01.28JoseHapbut as it's an external adaptor i suppose it only changes pins layout
09:01.35JoseHapbut i do not know how
09:01.48HiltonTt's pi$$poor
09:02.38*** join/#asterisk RoyK (n=roy@80.239.107.70)
09:04.35JoseHapok, so maybe i revert 8 and 7 pins to test
09:04.43HiltonTmaybe
09:04.52HiltonTdo you really like the phone?
09:04.55JoseHapmaybe i kill the phone :D
09:18.13*** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de)
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09:18.24*** mode/#asterisk [+o twisted|astricon] by ChanServ
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09:21.23*** part/#asterisk twisted|astricon (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted)
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09:35.30*** join/#asterisk kippi (n=chrisfro@untrust-gct.equinoxit.net)
09:35.31kippihey
09:35.48kippiI need a ISDN 2 card, what would people recomend?
09:36.50*** join/#asterisk zobia (n=laura_sh@218.6.242.212)
09:37.12zobiahello everyone. do u know how to transfer a zap channel to a meetme conference?
09:39.12wasimzobia: exten => 1,1,Meeme(blah)
09:39.34wasimzobia: where blah exists in meetme.conf and exten 1 is in a context that is accessible to the zap channel
09:39.57wasimzobia: there was an obvious typo in that command above, show application meetme for more details
09:40.39zobiaokay. thanks a ot
09:40.53htimswasim: and how does it work when i want to tansfer someone i've called already to an conference?
09:41.00wasimsure, donate a little something to earthquake relief :)
09:41.24wasimhtims: use manager api to transfer the call, or use # transfer, see features.conf
09:41.57*** join/#asterisk areski (n=areski@polar.es6.egwn.net)
09:42.06wasim40k+ dead, 50k+ injured, 2 million homeless, winter approaching, is not a pretty sight ...
09:42.13*** join/#asterisk fulgas (n=fulgas@213.58.130.46)
09:45.23RoyKshit
09:45.32RoyKwasim: and hospitals are standing?
09:46.13*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
09:46.15puzzledmorning
09:46.49case_errr... i don't understand how to generate a call from a sidphone to asterisk (well, i can call asterisk from the sidphone, but how to configure Asterisk to accept this call?)
09:47.49*** join/#asterisk corne (n=corne@ndn-165-157-254.telkomadsl.co.za)
09:47.52JoseHapdo you mean sipphone ?
09:48.06case_yes i mean it
09:48.25RoyKsidphone, sid, the kid that breaks toys......
09:48.26RoyK:)
09:49.08case_that's the revenge of the sid :)
09:49.21RoyKhttp://www.nextron.no/main.php3?PI=11&PNO=13167
09:49.23RoyKlol
09:49.41JoseHapcase_ i cannot understand the question about your phone
09:50.16JoseHapyou want asterisk to make something with the call, like answering and playing beeps or music ?
09:51.19case_JoseHap: i want asterisk to queue the call
09:52.03case_i'm in a call center scheme here. there are incoming calls, some queues, and operators to answer
09:52.15case_i don't bother about operators right know
09:52.35case_what i want is generating incoming calls with a sipphone and queue the call (forever)
09:55.10*** join/#asterisk johnm (n=johnm@gentoo/developer/johnm)
09:55.16*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
09:56.55zobiahello. any one how to transfer a call with agi?
09:58.15zobiaactually i want to transfer a to a meetme in the agi.
09:58.58cornehi case_ i have just join, if i understand you correctly you want to set up queueing on you asterisk. if that is the case then i can tell you that it is very easy to setup a queue if you are using asterisk@home
10:00.10*** join/#asterisk folsson (n=filip@h147n1fls32o985.telia.com)
10:09.22RoyKeeeeeerrrrrr
10:09.25RoyKstrange
10:09.32RoyKI try to dial into my new snom 320
10:09.36RoyKand it answers 404
10:09.38RoyKnot found
10:09.41RoyKstupid
10:11.20case_corne: thank you very much but JoseHap is helping me in private...
10:23.45*** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net)
10:47.58gordonjcphello
10:48.43gordonjcpis there a good reason why asterisk wants so many rtp ports?
10:50.37malabarwho says asterisk wants many rtp ports?
10:51.07gordonjcpwell, in rtp.conf it's set up to use 10000-20000
10:51.25gordonjcpten thousand possible rtp ports seems a tad excessive
10:51.33RoyKgordonjcp: well
10:51.52RoyKgordonjcp: http://bugs.digium.com/bug_view_page.php?bug_id=3986 is a good reason to allocate a bunch
10:53.09gordonjcpwell, it seems that only a couple of people are having the problem
10:53.54malabaradjust it to your needs, what is it? 2 per SIP-connection? 1 per IAX-connection? Havent't used 1.0.7...
10:54.19RoyKgordonjcp: there aren't that many people that use asterisk in large itsp setups
10:56.31HiltonTRoyK; any reason in particular?
10:56.35*** join/#asterisk contrabanda (n=G@213.131.37.202)
10:56.41contrabandahiii
10:56.49contrabandaplease help me with voicemail
10:57.14*** join/#asterisk Akelavlk (n=jansun@82.119.239.141)
10:57.22RoyKHiltonT: reason for what?
10:57.31AkelavlkHello, has anybody experience with spanDSP?
10:57.41HiltonTnot being used in large ITSP setups
10:57.42contrabandaexten => 111,1, Answer
10:57.43contrabandaexten => 111,2, Dial(SIP/dito,10);
10:57.43contrabandaexten => 111,3, Playback(custom/111)
10:57.43contrabandaexten => 111,4, Voicemail, 111
10:57.43contrabandaexten => 111,5, Hangup
10:57.43contrabandaexten => 111,103,Voicemail,b111
10:57.45contrabandaexten=> 111, 104, Hangup
10:57.48AkelavlkMy question is if spanDSP is ok or not..
10:57.52contrabandahere ia my conf
10:58.09RoyK~pb?
10:58.10jbotit has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/
10:58.21*** join/#asterisk zotz (n=zotz@24.231.36.100)
10:58.42contrabandaok sorry
10:58.57razucontrabanda : do you have an account in voicemail.conf ?
11:00.20contrabandayes
11:00.28contrabandai got such error
11:00.31contrabandaOct 13 17:53:44 WARNING[2934]: app_voicemail.c:2406 leave_voicemail: No entry in voicemail config file for ' 111'
11:00.45razuwell
11:00.58razuthe answer should be infront of u
11:01.09contrabanda111 => 111, Dito Sekhniashvili, naxalovka@mail.ru
11:01.09RoyKHiltonT: stability, perhaps?
11:01.25HiltonTdunno - wondered if there was a reason, not a guess is all
11:01.34HiltonTmaybe no-one's played that much  :)
11:01.45HiltonTas for stability, have you found * to be unstable?
11:01.47AkelavlkWhat about spanDSP?
11:02.08contrabandawhats a problem with my voicemail?
11:02.17HiltonTit doesn't work
11:02.28RoyKHiltonT: yes. lots of things needed fixing. and still there are lots of more
11:02.42HiltonTaha - getting better, though
11:02.52HiltonTstable in a 30-50 user office, I gather
11:03.01RoyKasterisk is good for small systems, but never designed to run anything larger than the small office
11:03.06HiltonTI assume 1.2.0 will rock fairly hard
11:03.20RoyKHiltonT: we're running a few thousand users on a custom patched 1.0.7
11:03.27HiltonTwhat's the definition of "small" tho?
11:03.44RoyKHiltonT: do not assume that. it's not really well tested, and prolly won't be when it comes out as 'stable'
11:03.49*** part/#asterisk Akelavlk (n=jansun@82.119.239.141)
11:03.50HiltonTcool - sounds like a reasonable size!
11:04.00HiltonTwhat was patched, roughly, to handle this?
11:04.22HiltonT(I'm *just* getting into *)
11:04.25razucontrabanda : get rid of the annoyng spaces after commas. and check again
11:10.13contrabandaok
11:10.40HiltonTRoyK; patches?
11:10.48contrabandai got the same error
11:12.36contrabandais there any rule for creating voicemail mailbox?
11:14.54RoyKHiltonT: it's not patched to handle the load. it's patched to stop crashing and to stop blocking certain clients from sending INVITES and to handle mysql integration better and a few other things
11:15.52HiltonTaha - I gather you passed these patches, especially the "stop crashing" ones back to the team for inclusion?
11:16.24HiltonTMySQL integration would be useful, as would Outlook/Exchange integration (tho I gather it does this thru CAPI)
11:18.34HiltonTbecause less crashing is obviously nice  :)
11:18.39HiltonTin any situation
11:19.50RoyKall are open
11:29.14*** join/#asterisk skydiver (i=skydiver@skydiver.no)
11:30.27skydiveranyone here who has the correct setup to get a skinny/SCCP phone working with asterisk? I've got it all set up, but I can only dial out to the PSTN and to other SIP phones, but not to other skinny/SCCP and not from SIP/PSTN and inwards. guess this relates to missing extensions, but I haven't been able to find any good examples on how this is done. :-)
11:33.06*** join/#asterisk aor (n=bob@202-197.246.81.adsl.skynet.be)
11:33.15aorHi everybody ...
11:33.22JoseHaphi aor ;-)
11:34.02aorI'm (desesperately :) trying to make zaptel 1.2 working with a  centos 4.1 with standard kernel 2.6.10
11:34.17aorI follow the README for udev
11:34.35aorthe devices are created, even if I get this message on the modprobe wctdm
11:35.07zobiahello can asterisk tranfer a iax2 phone to a meetroom?
11:35.25mbrancaeh, you need to wait some secs after modprobing before launching ztcfg ....
11:35.30mbrancaudev is not immediate
11:35.31mbranca:)
11:35.50aorwell, it seems that black magic is with me :)
11:36.04aori just rmmod and modprobe again wctdm and it is working :s
11:36.30aormbranca : yeah i even checked that the devices where created, but I got an error with chanzap
11:36.35aorbut now it is working
11:36.44aorthe machine is damn to slow to my fingers ;-)
11:39.34*** join/#asterisk mmmToop (n=chatzill@196.31.11.194)
11:42.11aorok, got the tip : I need to modprobe zaptel first, wait a few seconds, then launch wctdm and everything is ok
11:45.10skydiveranyone with some skinny/SCCP experience here? =)
11:47.25*** join/#asterisk Ldr (n=Lode@194.206.157.226)
11:51.29JoseHapaor is definitely too fast ;-)
11:52.49zobiahello. anyone how to transfer a established call to a meetroom in agi.
11:54.54*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
11:56.27kippihi
11:57.53*** join/#asterisk MuppetMaster (n=MuppetMa@62.37.171.235)
11:58.01MuppetMasterHello
11:58.30MuppetMasterPer this post:  http://forums.digium.com/viewtopic.php?t=1892 - does anyone have, no of where to find an Asterisk server connected to an incoming line in India/Pakistan?
12:00.15kippiif I have ISDN line and I want to put my fax number on it, Do I just get a grandstream box for the fax and just route the calls over asterisk to the fax?
12:03.09skydiveranyone here who has the correct setup to get a skinny/SCCP phone working with asterisk? I've got it all set up, but I can only dial out to the PSTN and to other SIP phones, but not to other skinny/SCCP and not from SIP/PSTN and inwards. guess this relates to missing extensions, but I haven't been able to find any good examples on how this is done. :-)
12:03.39*** part/#asterisk MuppetMaster (n=MuppetMa@62.37.171.235)
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12:18.12kippiI need some kinda reporting software, whats the best on out there?
12:24.51RoyKkiko69: reporting what?
12:24.58RoyKecho is a good reporting command
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12:27.21freaekgood evening all
12:28.44melvenhello has anyone used a digium s100u on a fedora core 3 machine ?
12:30.20freaeksorry melven, not me, I'm using slackware :)
12:38.12*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
12:42.13melven<PROTECTED>
12:42.26melveni'm just trying to get it to work period .
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12:49.05lehelhello
12:49.26tzafrir_laptophi
12:50.48Ariel_Morning all
12:58.34*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
13:02.40uteris there a way to tell asterisk the general language?
13:02.56Kattymew.
13:03.05uternot just for an extension or channel, but for all the things?
13:06.15*** join/#asterisk mistral (i=mistral@jstevenson.plus.com)
13:06.16*** join/#asterisk Corydon76-home (i=two@pdpc/supporter/sustaining/Corydon76-home)
13:06.22Kattymew?
13:06.34mistrali am a little stuck on a problem to configure outbound calls
13:06.35Ahrimanesmuw?
13:06.45tzafrir_laptoputer, you can speak English in any channel
13:06.52mistralhow would i just perform a catch all ? and dial it on a linue ?
13:07.20tzafrir_laptoputer, but you refer to SetLanguage?
13:07.33Ariel_So anyone know if the 1.2 version of asterisk is office out yet?
13:07.39Kattymew :<
13:08.22Katty:>
13:08.31Kattyyay, someone answered me for a change
13:08.34tzafrir_laptopAriel_, if it's not in $TOPIC, it didn't happen
13:08.43tzangerI always say good morning, Katty
13:09.02mistralso i guess nobody can hlep me then
13:09.03Kattytzanger: :>>
13:09.07Ariel_tzafrir_laptop, yes I did not see it on any mail list either, bummer
13:09.17*** join/#asterisk jontow (i=jontow@ws.woflsys.net)
13:09.20tzangermistral: you *really* and I mean *really* need to read the handbook
13:09.21tzangerit's covered there
13:09.23utertzafrir i don't want to use setlanguage in every extension
13:09.27*** join/#asterisk IzNoGooD (n=marc@iznogood.demon.nl)
13:09.31Kattytzanger: i never read the handbook.
13:09.38tzafrir_laptopmistral, your answer is not clear
13:09.42Kattytzanger: more like browsed several pdfs.
13:09.44tzangerKatty: you're not asking questions that are answered clearly in the handbook.  :-)
13:09.49Kattytzanger: and bugged the crap out of Hmmhesays ;>
13:09.51uterso i would like to specify it as a global
13:09.55IzNoGooDIs there any special I need, to get asterisk to play the demo?
13:09.56Ariel_mistral, a catch all is like exten => _.,Dial(Blah) but it's dangerest and I would not use it.
13:10.10IzNoGooDI have a connection through sip and misdn, but both I don't hear any sound
13:10.17tzafrir_laptoputer, you can set it in sip.conf/iax.conf . But I know of no global way in the dialplan
13:10.19IzNoGooDBoth connect to the asterisk demo
13:11.07utertzafrir, i tried it in zapata.conf for incoming calls, but it didn't work :(
13:11.12iDunnohmm.
13:11.17mistralyou see i have read the aterisk hand book i just dont follow how the pattern matching works on the numbers
13:11.29Kattytzanger: does jim van mag(etc) talk in here?
13:11.37Kattymeg(etc) i mean
13:11.50*** join/#asterisk coppice (n=chatzill@190.196.17.210.dyn.pacific.net.hk)
13:11.58iDunnoNow using the zaphfc driver, and a zaphfc card, it just doesn't appear to register at all :(
13:12.36Ariel_mistral, the files in /usr/src/asterisk/configs/extensions.conf.sample has lots of samples for you to see dialing pattern matching.
13:13.07Ariel_Katty, it's an isdn type card used in the EU
13:13.36KattyAriel_: isdn type card?
13:13.44KattyAriel_: i take it it's not for analog lines.
13:13.51tzangerKatty: yes, he's JimVanM when he's here
13:13.52tzangerbut it's not often
13:13.56tzangernicfe
13:14.03Kattytzanger: k...does he show up at conferences?
13:14.08tzangerI just got a request to fax something to 750 different locations
13:14.17Kattytzanger: email it instead!
13:14.21tzangerthis'll be a good test for txfax :-)
13:14.24Kattytzanger: save trees!
13:14.32*** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com)
13:14.36tzangerKatty: a LOT of these places do not have or prefer not to use email
13:14.51tzangerKatty: I'd agree with you but I know what our customers are like :-S
13:14.51Kattyif they have any sense they'll have a kyocera copier which takes the fax and -> to email
13:14.58tzangerconferences?  I have no idea
13:15.03KattyWHAT
13:15.07tzangerhe tends to frequent the TorAstricon meetings
13:15.14Kattywhat sort of backworld place to these people live in?
13:15.20tzangerKatty: if they had any sense they'd prefer email in the first place
13:15.22Kattywe've only got 10 people here and /we/ have email!
13:15.36tzangerKatty: a lot of our customers are literally in the middle of nowhere
13:15.42iCEBrkrDude, my parents use email religiously..
13:15.49Kattyof course we're the ones who setup networks too and stuff, but anyway
13:15.51tzangeriCEBrkr: I'm not saying old people can't use email
13:15.54iCEBrkrThat's a scary concept considering my dad is a motorhead mechanic
13:15.56tzangerI know a lot that do
13:15.58Kattytzanger: i'm in the middle of nowhere :P
13:16.05Kattytzanger: missouri has corn, cows, and fields. that's it :P
13:16.10tzangerbut I'm saying that some of our customers don't have email
13:16.14Kattysad.
13:16.17tzangerKatty: heh, I'm in rural ontario
13:16.17Kattysad sad sad.
13:16.27tzangersome of our customers don't have computers
13:16.35Kattyeek!
13:16.40Kattyi'd die.
13:16.43tzangerwhere's coppice
13:16.48iCEBrkrtzanger: But I know where you're coming from. I worked at a mortgage place and a bunch of the appraisers didn't have/use email.  They expected faxs and Fed-Ex'd documents.
13:16.49Kattynapping.
13:17.09tzangerI wonder if a P4/whatever can send 23 simultaneous faxes
13:17.12IzNoGooDAny ideas why I don't get any sound?
13:17.24Kattygreat
13:17.27Kattyi get to go answer phones.
13:17.34Kattyand take care of /walk ins/
13:17.35iCEBrkrKatty: Sweet
13:17.50Kattyour nextel staff is going to an All Day conference :<<<<
13:18.02tzangerfun
13:18.05*** join/#asterisk IPmonger (n=ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net)
13:18.36Kattyno, /not/ fun
13:18.40Kattypeople are evil
13:18.44KattyEVIL
13:19.11iCEBrkrKatty: Yea, well. Ya know, birds of a feather. :P
13:19.26tzangeriCEBrkr: exactly
13:19.28Kattybig difference between birds and people.
13:19.29iCEBrkr:)
13:19.37Kattybirds flock, and they get along with flocking.
13:19.49Kattythey don't go visit nextel stores and bitch at the staff who work in the IT department
13:20.01iCEBrkrKatty: My phone doesn't work!!! *Whine*
13:20.03Kattybecause they think everyone is a Dedicated Nextel Team Member
13:20.11iCEBrkrKatty: I keep getting dropped calls!! *Whine*
13:20.16*** join/#asterisk funxion (n=nunya@mtnuser.icgws.com)
13:20.25iCEBrkrYou warmed-up yet?
13:20.27KattyiCEBrkr: nono, it's more like this
13:20.42KattyiCEBrkr: zomgidroppedmyphoneinthetoilet MAKE IT WORK NOW OR I"M TALKING TO YOUR MANAGER
13:20.52iCEBrkrlol
13:20.59tzangerKatty: do they hand the phone to you?
13:21.04iCEBrkrOH NOEZ!!!
13:21.09Kattytzanger: actually, no.
13:21.14Kattytzanger: i know nothing about nextel stuff.
13:21.17tzangerTHANKFULLY no I'd think :-)
13:21.21Kattytzanger: except that it's amr and IDEN network.
13:21.34tzangerI have no idea what that means
13:21.36Kattytzanger: and that sms is just an email address
13:21.46Kattytzanger: nextel stuff isn't my department ;)
13:21.59iCEBrkrSpeaking of SMS.
13:22.03Kattyyay, SMS
13:22.12Kattyif you put your number, under gaim, under uhh...aim stuff, me thinks
13:22.15Kattyand talk to it
13:22.20Kattyit'll send a message to a mobile
13:22.22tzangerKatty: can you get me the format of an SMS message that lights up the phone's MWI?  I'm sure it's just a specific header, something like MWI: active,8095124533 or MWI: inactive
13:22.25iCEBrkrI want my SMS to be able to SMS people.  I guess I could just use their providers gateway...
13:22.48Kattytzanger: no clue. all i know is that if i send an email to myphonenumber@messaging.nextel.com it's an SMS
13:23.03Kattytzanger: which is darn handy when a server hiccups (=
13:23.11Kattytzanger: or when i get a voicemail, etc.
13:23.24tzangerKatty: yeah that's pretty standard, I was hoping to light up MWI on the phone (so it shows voicemail waiting, not just an SMS waiting)
13:23.37Kattytzanger: oh, i've been working on that, too.
13:23.53Kattytzanger: i tried an email with an attachment ..with several different encodings too....never worked :<
13:24.04tzangeryeah
13:24.04Kattytzanger: apparently they use a different type of server for that.
13:24.13Kattytzanger: it's not an email server.
13:24.16tzangerthere are SMS gatweays that charge $0.17/SMS that can light/delight MWI they claim
13:24.19tzangerso I know it's possible
13:24.27tzangeryeah they use their SMS gateway
13:25.09iCEBrkrtzanger: There's gotta be a 'free' SMS provider out there. :)
13:25.19tzangernah
13:25.21iCEBrkror one that offers like 100 SMS/mo
13:25.24iCEBrkr:(
13:25.24Kattytzanger: i'll ask.
13:25.32Kattytzanger: but our nextel staff doesn't know
13:25.35tzangerI want to do it myself I don't want to rely on an external gateway
13:25.38Kattytzanger: and this area's staff doesn't know
13:25.44tzangerI'd love to set up an email address on my domain and say to send it there so I could see
13:25.45Kattytzanger: so i'll have to pester corporate level
13:25.49Kattytzanger: and i'm sure they wouldn't tell me
13:25.53tzangerbut I don't think they'll do it
13:26.03iCEBrkrtzanger: Yeah, I'd like to get my own SMS thing set up without relying on the providers gateway
13:26.11Kattynextel people are really dumb.
13:26.11tzangerKatty: yeah :-(  I was bugging telus here in ontario and they just gave up and said it wasn't psosible which is a bullshit answer
13:26.20Kattyand when you get to the ones who know stuff, they're too busy to talk to you
13:26.22tzangereven if I had to call an 800# TAP gateway to do it I could
13:26.35iCEBrkrTAP
13:26.35iCEBrkrlol
13:26.37iCEBrkrOld skewl
13:26.49tzangeriCEBrkr: my other idea was to just get a phone and send the message that way
13:26.50Kattyskwerly
13:27.00*** join/#asterisk mlynch (n=mlynch@email.gcom.com)
13:27.16iCEBrkrI wrote a TAP interface in Turbo Pascal when I was like 15?
13:27.28Kattyi did some HTML when i was 13
13:27.30tzangeriCEBrkr: :-)
13:27.35iCEBrkrSo I could send pages to my PageNet pager :P
13:27.35Kattyi was all bouncy and excited
13:27.36coppiceTAP was a areal PITA
13:27.38Kattyit sucked ;)
13:27.42Kattybut i was happy!
13:27.42tzangersome of the documentation I have says that that's all it is is high speed TAP these days
13:27.54mutilatorgeocities w00t
13:28.01tzangerI had a pagenet pager!
13:28.04Kattypfft, geocities
13:28.08Kattyi never needed geocities.
13:28.14mlynchCan anyone help out a noob?
13:28.15mutilatorheh
13:28.16Kattyi was on irc from like 10
13:28.19tzangercoppice: do you forsee any issues in faxing two pages to 750 faxes with app_txfax?
13:28.28iCEBrkrtzanger: Yeah, but did you WarDial pagers with a targets phone number?
13:28.31Kattythere's always someone on irc that will let you use their server heh
13:28.31mutilatorso what'de ya do host it off your 28k modem
13:28.33iCEBrkr>:)
13:28.38mutilatorah
13:28.44tzangerI have two .tiff files, I need to figure out how to make one 2-page .tiff file yet
13:28.45iCEBrkrmlynch: What's up noob!!
13:28.54Kattymutilator: yay apache on 28k ;>
13:28.57tzangeriCEBrkr: Hahahahaha no no I never did that
13:29.00Kattymutilator: hey! i did that for awhile!
13:29.04iCEBrkrtzanger: Can't ImageMajick do that?
13:29.05*** join/#asterisk Uther_P (n=uther_p@66.180.120.82)
13:29.08mutilatorheh
13:29.09mutilatorme too
13:29.10KattyUther_P: (=
13:29.13coppicetzanger: those who have trouble with spandsp normally have it when sending. I have yet to find out why. many people seem to send in volume OK
13:29.19tzangeriCEBrkr: perhaps I'm not sure :-)  I have to make sure these tiff files are in the right format too
13:29.22mutilatormy dialup stayed connected for days at a time usually
13:29.28Kattymutilator: woah.
13:29.30mutilatormost i ever did was 7 day straight connection
13:29.38Kattymutilator: mine stays connected for a few hours now.
13:29.38mutilatorwas wicked cool
13:29.46tzangercoppice: ok, this will be an interesting test.  I'm curious to see if my CPU can handle 24 9600 baud fax sessions at once
13:29.49Kattymutilator: i bet.
13:29.56mutilatorya most isp limit it now, this was back when i got it through the community college
13:29.57iCEBrkrtzanger: Yeah.  Give IM a shot, there's a ton of command line tools to manipulate different image formats.  I'm learning stuff almost every day
13:29.59tzangeractually I think I only have 15 B channels turned up so that might be the limiting factor
13:30.00mlynchInstalled asterisk with samples on RHFC2 with 2.6.13.3 kernel.  Everything works but I can't seem to get the connection to digium IAX demo server to do anything
13:30.06mutilatorhope on and browse yahoo with lynx and stuff
13:30.10mutilatorhop*
13:30.19Kattymmm, lynx
13:30.20Kattyand pine
13:30.22Kattyremember pine?
13:30.25coppicetzanger: 24 should OK on a modern machine
13:30.25mutilatoryep
13:30.27Kattyit was hottt.
13:30.28mutilatorused pine for out email
13:30.30Uther_Ppine is bad
13:30.31mutilatorour*
13:30.32tzangeryeah this is a P4/something (single processor)
13:30.43KattyUther_P: it was dreamy in the mid 80s
13:30.45Uther_Pbuggy buggy buggy
13:30.52tzangerP4/2.4G
13:30.56mutilatorhaving fun with my .plan file
13:30.57Uther_Phad a few code exploit butz
13:31.01KattyUther_P: whiine whine whine ;)
13:31.04Uther_Per bugz
13:31.05mutilatorput all kinda cool ascii art in it
13:31.07iCEBrkrtzanger: cat /proc/cpuinfo
13:31.12tzangeriCEBrkr: I just did
13:31.14iCEBrkrhehe
13:31.24coppicetzanger by dual 2.4GHz machine handles more than 48 in tests :-)
13:31.37tzanger:-)  okay then
13:31.41Uther_Peek... try mutt... its cooler anyway in my opinion
13:31.42iDunnoKatty: poor child.
13:31.48KattyiDunno: don't poor child me.
13:32.08iDunnoKatty: using pine, though... only unfortunates do that ;)
13:32.09KattyiDunno: if you're going to poor child me, say it over the fact i have to use exchange
13:32.20Uther_Phaha
13:32.21KattyiDunno: with outlook...and windows
13:32.22*** join/#asterisk Assid (n=assid@203.115.64.57)
13:32.27*** part/#asterisk IzNoGooD (n=marc@iznogood.demon.nl)
13:32.29tzangercoppice: it's curious how the 'ht' indicator bug is still there in 2.6.13
13:32.40iDunnoKatty: *shudder* I feel exceptionally sorry for you all of a sudden
13:32.47KattyiDunno: kthx.
13:33.13*** join/#asterisk aminorex (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com)
13:33.22Uther_Pthe samba team emulated the window networking... I wonder why noone has tried to emulate the exchange server
13:33.33Kattywho would want too?
13:33.36Assidi think there is a free exchange
13:33.40Kattyfor the calendar and tasks and stuff?
13:33.45Assidthere is one i think
13:33.50Uther_Pfor efficiency
13:33.50Assidopenexchange i think its called
13:33.55Assidsf/freshmeat for it
13:33.55tzangerI used that
13:33.56tzangerit's crap
13:34.03tzangerwe use exchange4linux now
13:34.08Kattymy next one is going to be Hula Project
13:34.22Kattymister nixon recommended it
13:34.25tzangerSuSE OpenExchange Server blew goats with amazing speed
13:34.39coppicewho? Richard?
13:34.52mutilatortzanger... exchange4linux actaully works for ya?
13:34.54tzangerExchange4Linux rocks -- 100% python, uses Postgres, stores EVERYTHING in postgres, Outlook client is in python too (but closed source)
13:34.54Kattycoppice: peter
13:34.54coppice"There will be no whitewash in the open source"
13:34.56tzangermutilator: sure does
13:35.11mutilatorya gotta pay for the windows driver thing tho right?
13:35.20tzangerI don't care about that.  $50/seat is fine
13:35.23mutilatori think thats why i opted out of it
13:35.37tzangerthe backend is open source
13:35.37tzangerthat was the key
13:35.37*** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au)
13:35.37mutilatorya
13:35.37tzangerI want to migrate off outlook eventually
13:35.37Kattynow then!
13:35.43Kattyso i had that polycom 500 problem
13:35.47mutilatorit's a one time forever updates free fee right?
13:35.54coppicetzanger: do it today
13:35.55nfi|ermesin my default dialplan there is: $[foo${ARG3} = foo]?3:2 . Anyone knows what is this ?
13:36.00Kattyif anyone recalls, we were having the issue of we couldn't hear them, but they could hear us.
13:36.05Kattyand then you manually rebooted the phone
13:36.09Kattyand it was magically all better.
13:36.13Kattyuntil it goofed up again.
13:36.16mutilatorheh well my office wanted to use it for the outlook calendaring feature
13:36.20Kattysoooooo.......i watched the rtp packets
13:36.23mutilatorso i just built a nifty one into the website
13:36.23Assid30G ?
13:36.24tzangercoppice: I hope to do it tonight actually
13:36.26Kattyand it's not a port thing
13:36.33Assidnice.. what the hell you storing in there
13:36.36Kattyany Other suggestions of things to look at?
13:36.46tzangernot keen on tying up all the B channels during working hours
13:36.55Kattyand someone better answer me :P
13:37.06Kattyor i shall start poking.
13:37.06tzangerKatty: exchange4linux
13:37.19Kattytzanger: i don't think we're talking about the same problem.
13:37.19tzangeroutlook doesn't know it's not talking to Exchange, which means that all the good stuff works
13:37.35Kattytzanger: nonono, the polycom problem.
13:37.36tzangerand all your data sits in an open system
13:37.36Kattytzanger: you silly sod :P
13:37.48tzangerthere's something really *cool* about doing arbitrary SQL queries on your contacts, todos and even emails :-)
13:37.51tzangerKatty: oh :-)
13:38.05tzangerit's solved a few problems for us already (being able to access all our outlook data)
13:38.18*** join/#asterisk Caede (n=caede@sentry.zoom.com)
13:38.28tzangerhmm I could poke this all through nufone's fax gateway too
13:38.29Kattytzanger: after roughly 8-10 calls person can no longer hear any incoming audio. manually reboot phone, everything is ok.
13:38.36tzangerKatty: hmm
13:38.37Kattytzanger: explain /why/
13:38.46Kattytzanger: also, please note, i watched rtp packets.
13:38.51tzangeris it allocating RTP ports outside of what Asterisk is saying it can do?
13:38.52Kattytzanger: nothing seemed unusual.
13:38.58Kattytzanger: no, that's what i looked at
13:39.04Kattytzanger: it's not too high.
13:39.09tzangerKatty: have you called polycom support?
13:39.15Kattytzanger: :<
13:39.20Kattytzanger: why would i do a silly thing like that?
13:39.40Kattytzanger: you know they'll give me the run around.
13:39.51tzangerKatty: then you politely tell them you'll send the phone back
13:39.54Kattytzanger: and then it'll come down to a We Don't Know type thing.
13:40.03iDunno*YAY*!
13:40.04Kattytzanger: but it's not one phone. it'll all 10 of our phones.
13:40.11iDunnoI R TEH W1NN0R!
13:40.17Kattytzanger: something tells me they're not /all/ bad
13:40.21tzangerKatty: then you say, "that's terribly saddenning.  I guess I'll send my phones back and go with the Cisco solution.  I thought you'd be able to help."
13:40.23iDunno(or, rather, at bloody last I've got working ISDN)
13:40.34Kattytzanger: ha, that won't work.
13:40.36tzangerKatty: do you have another phone to test?  Have you done a factory reset?
13:40.49Kattytzanger: yes, i've already done a factory reset.
13:41.02AssidKatty: restart network
13:41.10KattyAssid: that's not the issue either.
13:41.26AssidKatty: try pinging the ip from the linujx box
13:41.26KattyAssid: we've had power outages where everything was reset. changed nothing
13:41.33KattyAssid: that's fine too.
13:41.40KattyAssid: these phones recieve calls perfectly fine
13:42.05Assidhrmm.. weird
13:42.09Kattytzanger: i really thought it was the rtp port issue problem thingy
13:42.21Kattytzanger: but it went up to something like 2257 and dropped back down to 2222
13:42.28Kattytzanger: the person never rebooted their phone in between
13:42.42Kattyi /really/ need to take a look at sip debug
13:43.01*** join/#asterisk dalabera (n=Dalabera@pmr.pmrtechnologies.com)
13:44.22sylewhat is happening with intel and asterisk? they are talking about shipping some intel machine with asterisk business edition in nov/05
13:44.35Ahrimaneslink?
13:44.45sylehttp://blog.tmcnet.com/blog/tom-keating/voip/intel-and-asterisk.asp
13:45.03tzangerhmm
13:45.18sylehttp://www.voipplanet.com/solutions/article.php/3551016
13:45.39*** join/#asterisk KeX_WorX (n=chris@83-65-129-46.paris-lodron.xdsl-line.inode.at)
13:45.44KeX_WorXhi
13:46.06KeX_WorXcan some1 tell me how i tell asterisk to log infos bout queues?
13:46.24KeX_WorXthis should be an queue_log file in /etc/log/asterisk or?
13:46.29*** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd)
13:46.44KeX_WorXsaw this on an asterisk srv. now i set up a new one and there isn't such a file : /
13:46.54syleidk what your logger.conf is set to log dude
13:46.58mutilatori ought to get one of them fancy asterisk certifications
13:47.03CaedeOkay... frustrated-at-the-end-of-my-wits question.  I have a zaptel TE110P card connected to a PBX using E&M (wink-start) signalling.  We get constant clicking/popping sounds on any call traversing through the zap interface.  I don't see the clicks/pops when using ztmonitor, and it's only heard by the TDM-end (at the PBX side).  Does this sound like timing or something else?  If I turn on...
13:47.04Caede...extra extra debugging for zaptel, I get oodles of "T1: Lost our place, resyncing" (offset 28, although that's not in the message)
13:47.09mutilatorprobly start being a resume stuffer here soon
13:47.24mutilatorone of the useful ones anyway
13:47.30KeX_WorXsyle, what do i have to write in logger.conf ?
13:47.54KeX_WorXi've these files from both srv's one with quele_log file, one without
13:48.01KeX_WorXthe logger.conf files are exactly the same
13:48.35syleenable your debug log
13:48.36KeX_WorX*grml : )
13:48.39KeX_WorXqueue_log
13:48.39KeX_WorXsry
13:48.41*** join/#asterisk apardo (n=w0w0@1.Red-83-46-192.dynamicIP.rima-tde.net)
13:49.03nfi|ermesin my default dialplan there is: $[foo${ARG3} = foo] . Anyone knows what is this ?
13:49.19*** join/#asterisk Akelavlk (n=jansun@82.119.239.141)
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13:49.31nfi|ermesdoes it check if the ${ARG3} is empty ?
13:49.41AkelavlkHello. Has anybody experiences with HylaFax of spanDSP?
13:50.00Uther_Pnfi|ermes: yes,
13:51.16sylei think arg3 is a call to a macro possibly
13:51.31syleso check where its calling it
13:51.38nfi|ermesthx Uther_P
13:53.02Kattyyay, new server
13:53.13Kattyrev drive, 2 300 gig hard drives with raid 1
13:53.18Corydon76-homenfi|ermes: you could also do $[${LEN(${ARG3})}]
13:53.20Kattypretty little box.
13:53.45Kattytoo bad windows is going on it :<
13:53.47Corydon76-homebut that would be the opposite
13:53.58funxionwhy cant I find wct2xxp anywhere in zaptel
13:53.58Beirdoawww, poor little box.
13:54.02KattyBeirdo: i know :<<
13:54.09Corydon76-homesince an empty ARG3 would have length 0, evaluating to false
13:54.13KeX_WorXsyle, had to uncomment queue_log => in logger.conf
13:54.16Beirdothese things happen, I guess
13:54.19Beirdogood morning, Katty
13:54.39KeX_WorXsyle, but on the srv where the loging works, this line is commented and it logs? how is that? u know?
13:56.25syledid you read the file?
13:56.31sylecause it defaults to yes
13:57.55IzNoGooDWhat is the best channel to get help on using asterisk?
13:58.17jake1932here is the asterisk channel
13:58.52IzNoGooDOk, the question is, I try to use asterisk with misdn, and the connect works
13:59.04IzNoGooDand asterisk starts playing the demo-congrats
13:59.09IzNoGooDbut I don't hear any sound
13:59.31*** join/#asterisk FABRIZIOxxx (n=FABRIZIO@81-208-26-86.ip.fastwebnet.it)
13:59.45*** join/#asterisk mlynch (n=mlynch@email.gcom.com)
14:00.13jake1932IzNoGooD: can you get audio using other devices (SIP/IAX/etc)?
14:00.22IzNoGooDno
14:00.23FABRIZIOxxxhello all .. how can i test the efficiency of the lan .. i'm using tracepath and i'm getting strange results ... is it ok .. ?
14:00.28IzNoGooDsip isn't working either
14:00.36jake1932IzNoGooD: any error messages
14:00.51mlynchCan anyone help a noob get digium iax demo server demo working?
14:01.06IzNoGooDjake1932: none
14:01.12funxiondoes anyone here have a TE210p?
14:01.20jake1932IzNoGooD: have you done "rtp debug" from the CLI?
14:01.30IzNoGooDTrying
14:02.25IzNoGooDjake1932: outputs <Sent RTP packet to 82.161.0.227:49194 (type 0, seq 60201, ts 528960, len 160)> which is my external ip
14:02.50IzNoGooDbut both hosts asterisk and sip-client are on localnet 10.0.0.x
14:02.59jake1932IzNoGooD: should be sending from the asterisk box to your softphone or hardphone ip
14:04.01IzNoGooDshould I set the localip somewhere?
14:04.07IzNoGooDWhere does it get my externel ip from?
14:04.17jake1932IzNoGooD: using NAT?
14:04.26IzNoGooDNot in between
14:04.39IzNoGooDI have a seperate router
14:04.43*** part/#asterisk Akelavlk (n=jansun@82.119.239.141)
14:04.48IzNoGooDwith nat
14:04.52jake1932IzNoGooD: sip or iax?
14:04.57IzNoGooDsip
14:05.02jake1932IzNoGooD: sip is sip.conf - iax is iax.conf
14:05.10IzNoGooDyes, sip.conf
14:09.32KeX_WorXsyle, where is that what u said? that queue_log defaults to yes ? can't find it
14:10.12sylein your logger.conf dude
14:10.20KeX_WorXcan't find it
14:10.26KeX_WorXasterisk 1.0.7
14:10.28syle; This determines whether or not we log queue events to a file (defaults to yes).
14:10.28syle;queue_log = no
14:10.28syle;
14:10.54KeX_WorXnop. that's not in my file
14:11.02syleoww no idea then dude, i use latest cvs myself
14:11.11tzangerok app_txfax returns 0 or -1, how do I see that in the dialplan?
14:11.26tzangern+101 isn't around anymore I don't think and README.variables doesn't show any kind of "application result code"
14:11.52*** join/#asterisk mithro (n=tim@tagung-233-198.tagung.uni-hamburg.de)
14:12.13sylehow is it not around
14:12.18KeX_WorXthis is my logger.conf http://pastebin.com/392243
14:12.34KeX_WorXuse the deb package. perhaps it's in a new version
14:13.18*** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com)
14:13.45syleis 1.0.7 latest stable?
14:14.04KeX_WorXähm, dunno : )
14:14.14file[laptop]no, 1.0.9 is the latest
14:14.15KeX_WorX'latest' stabel on debian
14:14.16KeX_WorXk
14:14.45syleis realtime stuff at least supported?
14:14.52file[laptop]in stable? no
14:15.04IzNoGooDah sip works now
14:15.08KeX_WorXlooked at a logger.conf on a 1.2-beta1 asterisk. ther is the queue_log = no
14:15.13syleie: extconfig.conf
14:15.19Ariel_1.0.9.2 does not support realtime
14:15.28syleiaxusers peers etc
14:15.45file[laptop]stable does not support realtime.
14:15.51sylewell that is gay
14:15.57file[laptop]not really, it's bug fixes stuff
14:15.59Ariel_no it's great that it does not support realtime
14:16.08syledepends on what your doing
14:16.12sylefor me it would suck
14:16.26file[laptop]then don't use stable
14:16.26Ariel_if you need realtime you don't use stable
14:16.26sylei don;t :)
14:17.14*** join/#asterisk gambolputty (n=gambolpu@72.240.241.108)
14:17.22syleumm how can you not see advantage is realtime
14:17.26sylein
14:17.39syleyou new to asterisk?
14:18.02tzafrir_laptopsyle, it has some atvantages, but a whole slew of problems.
14:18.10Ariel_syle, no I have been working with asterisk for over 3 years. I can do allot faster things with normal setup then realtime.
14:18.30sylehow can you automate adding iaxusers etc from webpage without it
14:18.32Ariel_it's not complete and has issues
14:18.43tzafrir_laptopsyle, sure
14:18.43Ariel_manager api
14:19.04sylemanager api? how so?
14:19.06tzafrir_laptopsyle, edit config file and reload
14:19.19tzafrir_laptopAMPortal does that
14:19.40sylei;ve played with manager api in perl, never saw a way to add realtime users
14:19.56*** join/#asterisk Essobi (i=kstone@75.137.26.216.host.teledvance.com)
14:20.10tzafrir_laptopto add real-time users you simply add them to the table, right?
14:20.16syleright
14:20.17Ariel_syle you submit then your perl does a write then a reload
14:20.29sylea reload on production!
14:20.31syleyou nuts
14:20.41Ariel_syle, it works great
14:20.49EssobiAnyone remember the name of that sip program you could use to hand craft packets to check for responses to servers?
14:20.51syleyou must not have many users
14:20.57Ariel_you can reload sip by it's self and also extensions by them self
14:21.03tzafrir_laptopsyle, but then Asterisk has much more work to do at call time, just when it needs to be most responsive
14:21.22sylei;m to bleeding edge as it is it is scary
14:21.24Ariel_prayer time bbl
14:21.33sylecvs head and mysql 5.x for trigger support
14:21.53Essobi;)
14:21.59EssobiThat is bleeeding edge.
14:22.09EssobiBut triggers are sweeet.
14:22.17syleyes i love them!
14:22.25CaedeOkay... guess my first question was too long.  How 'bout this: any reasons NOT to use icc to compile Asterisk?
14:22.29sylegod i don;t ever want to go back to 4.x
14:22.34EssobiWe're about to migrate some stuff to 5, for triggers and real sub-selects, and views
14:22.54tzafrir_laptopsyle, then why not use pgsql and have tried-and-tested triggers?
14:23.01tzafrir_laptopand sub-selects
14:23.36sylebecause that would take more days of learning postgres and i really would rather not commit that, + i hate fact you have to export a whole table just to back it up
14:23.46mutilatoranyone know if it's possible to upgrade a win2k pc from Standard PC mode to ACPI multiproc pc mode or MPS multi proc mode w/o reinstalling win2k over top.. i tried to just update from standard to acpi last night and i had to restore windows cause it bsod..
14:24.04sylesame with innodb, but not my myisam tables
14:24.31tzafrir_laptopsyle, BTW: did you read about oracle and innodb?
14:24.34syle+ realtime support is native in asterisk with mysql , not postgres
14:24.40sylepostgres needs ODBC
14:24.54tzafrir_laptopisn't there native postgres in head?
14:24.56syleis supported natively in asterisk with mysql
14:25.34*** part/#asterisk mlynch (n=mlynch@email.gcom.com)
14:25.37Essobisome poeple are just comfortable with what they know
14:25.42EssobiI know I am. :)
14:25.49sylepretty much, i been using mysql since it first came out
14:26.17Essobimysql does outshine pgsql when it comes to speed in large large large databases too.
14:26.23tzafrir_laptopOracle recently bought the company that develops Innodb: http://www.oracle.com/corporate/press/2005_oct/inno.html
14:26.35EssobiNice.
14:26.40sylereally? i thought mysql shined on smaller tables
14:26.58syleread an article somewhere postgres performs better on more data tables
14:27.07darkskiezand then there was sqlite
14:27.29sylebut bastards were probably doing comparison on myisam instead of innodb so maybe they were getting table locking instead of row locking
14:27.31*** join/#asterisk _T3_ (n=rposada@53.228.uio.satnet.net)
14:27.46tzafrir_laptopactually, if you don't need to access the data from a different server, sqlite would probably mean much less administration headache
14:28.20sylethats scary, oracle will no longer maintain it opensource i bet
14:29.02sylei am very found of triggers
14:29.13sylespent a whole day with them over the weekend
14:29.22tzafrir_laptopsyle, it's GPL. The worst they could do is stop maintaining it. But from the POV of MySQL AB the worst that could happen is they could no longer resell it as part of their non-free product
14:29.30syleabit tricky to learn to do complex shit, but its great when you know it
14:30.26sylealthough i guess monday wasn;t a holiday for you guys
14:30.39sylewas canadian thanksgiving here
14:31.28syleanyways, i;ve never seen how to add a user to a config file over manager API
14:32.38sylebut i disagree with calling sip reloads all the time for realtime stuff, i hope extconfig.conf makes it into stable soon
14:34.47Ariel_syle, only way it makes it is if the 1.2beta1 becomes stable release.
14:37.37tzangerok why the fuck isn't * seeing my outgoing .callfiles
14:38.28tzangerwasn't there some kind of module I had to load ot get that functionality
14:38.39tzanger/var/spool/asterisk/outgoing exists, I drop a file in and it isn't picked up
14:38.48syleyou use mv right?
14:39.34sylewell i;ll make my script that places calls into that directory
14:39.36sylepaste
14:39.49sylecp /etc/asterisk/callfile.txt /etc/asterisk/callfile.txt.old
14:39.49sylechown asterisk.asterisk /etc/asterisk/callfile.txt
14:39.49sylemv /etc/asterisk/callfile.txt /var/spool/asterisk/outgoing/callfile1.txt
14:39.49sylecp /etc/asterisk/callfile.txt.old /etc/asterisk/callfile.txt
14:39.55sylesimple shell script
14:40.08iDunnoanyone using the zaphfc driver?
14:40.13iDunnoand has working callerid?
14:40.25tzangeryeah
14:40.35tzangerbut it's not getting picked up. usually the file is deleted within a few seconds of being put there
14:40.51syletail /var/log/asterisk/debug
14:40.55syleafter you do it
14:41.14tzangerI am tailing it
14:41.15tzangernothing
14:41.40syleCLI show anything?
14:42.17syleidk if you have verbose on or not
14:42.37Dr_Rayis the file a valid call file?
14:42.46tzangerno cli norally would show it being processed
14:42.53tzangerit's like the part of asterisk that schedules this is not on
14:43.05syledid you try restarting asterisk?
14:43.54syleonly reason i;ve ever heard of it doing that is if your timestamp on that callfile is in the future
14:44.16Dr_Raysyle - that's an awesome feature btw
14:44.31tzangersyle: yeah
14:44.37sylewakeup call stuff
14:44.40syleyeah
14:44.41syle:)
14:44.47Dr_Rayyeah, saves a cron job
14:45.00*** join/#asterisk case_ (n=case@pdpc/supporter/student/case-)
14:46.01tzangersyle: hmm
14:46.03tzangerdon't think so
14:46.14tzangerI mean even if the callfile's garbage asterisk usually deletes it
14:47.26tzafrir_laptopthe call file needs to be readable by asterisk. If it's not immedietly deteled , it is probably not readable
14:47.29syleit just sits there?
14:47.49syledid you chown it to the asterisk users before placing it there?
14:47.55tzafrir_laptopchown/chmod it before you move
14:47.59*** join/#asterisk Brijn (n=bas@S0106004063c0fa1f.vn.shawcable.net)
14:48.03nfi|ermeshow shuold i choose wich codec to use ?
14:48.04sylewell he says it does not show up in his debug log at all though
14:48.27syledebug would show you reason it failed usually, permission issue or whatever
14:48.31BrijnQuick q, where do you define on what IP's asterisk will listen for incoming registration requests..
14:48.31*** join/#asterisk darwin35 (n=darwin35@208.139.193.178)
14:48.35tzafrir_laptopnfi|ermes, sip show channels / iax2 show channels
14:48.35tzangerfound it syle
14:48.39tzangerpbx_spool.so was noload'ed
14:48.48*** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net)
14:48.56syleplaying to much with modules.conf?
14:49.01darwin35good morning my fellow asterisk tramps,whores, sluts, and all others
14:49.09sylehehe
14:49.22darwin35have you Prosatuted your asterisk box latly
14:49.36darwin35have you made it work its tits and ass off
14:50.26darwin35has it made you so happy you could just burst with joy
14:50.56darwin35no I am just happy got my first Pay check from new job
14:51.05sylekewl, what are you doing?
14:51.05darwin35its nice to be working again
14:51.08*** join/#asterisk Ocean_big (n=morey@ax113-4-82-227-179-176.fbx.proxad.net)
14:51.12darwin35www.teliax.com
14:51.14Ocean_bigHi!
14:51.19darwin35Support office
14:51.39syleyou do asterisk stuff?
14:52.05syleor you the new phone monkey hehe
14:52.07darwin35everything from support to loading up new servers with asterisk realtime to
14:52.23darwin35testing
14:52.38sylekewl what database you using for realtime?
14:52.38limbiquehow to do pickup with asterisk?
14:52.38*** join/#asterisk royth1 (n=royth1@200.121.129.178)
14:52.44Ocean_bigi need help with asterisk...  i 'm newbee. Someone to help me ?
14:52.46darwin35I am going to be building 4 new boxes today if the boss gets here with them
14:52.58tzangerok that sent WAY too fast to be successful
14:53.01*** join/#asterisk rikstah (n=rick@84.93.87.216.plusnet.ptn-ag2.dyn.plus.net)
14:53.06tzangerpostgres
14:53.08tzangerpostgres
14:53.08darwin35dont ask for help . just state the issue your having and wait for a responce
14:53.11syle:(
14:53.24rikstahhey, i just registered on the voip-user.org wiki and clicked the validation link i was emailed, but i still can't sign in...any ideas pls?
14:53.30tzangerwhat's worng with pg?  It Just Works
14:53.53darwin35how is pg clustering now days
14:53.58sylei;m not saying anything is wrong with it, i;m asking what db darwin uses for his realtime stuff
14:54.22darwin35here we use MYSQL at the min but I want to move to PG
14:54.44darwin35but I have not heard the update on the clustering issues they had 4 months ago
14:55.09Ocean_bigmy problem : asterisk dont answer any incomming call, I can only call (SIP or non-IP phone)
14:55.14sylei heard people switching from postgres to mysql for backup db servers heheh
14:55.31tzafrir_laptopOcean_big, what do you see on the CLI? set verbose 3
14:55.52tzafrir_laptop~pb
14:55.54jbotpastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca/
14:56.19*** join/#asterisk hotgrits (n=hotgrits@192.160.238.156)
14:57.10Ocean_bigtzafrir_laptop, I 'm newbee.. I use Asterisk@home, i don't understand what is the CLI.
14:57.34*** join/#asterisk gdh (i=foobar@bum.net)
14:57.40tzafrir_laptopOcean_big, as root, run: asterisk -r
14:57.46darwin35well I am going to setup a 2 pg boxes today and test clustering and 1 fbsd box
14:57.50Ocean_bigok
14:57.53darwin35and go on from there
14:58.01gdhQuickie - is 1.2.0 final likely to be announced at Astricon. or am I better just installing beta1 on a fresh box? :)
14:58.32case_i still can't call my asterisk server with a sipphone :(
14:58.44*** join/#asterisk salmandr (n=salmandr@mdsnwinas02pool2-a226.mdsnwi.tds.net)
14:59.00Ocean_bigtzafrir_laptop,it's ok.. i can see " Verbosity is at least 3
14:59.50tzangerok zaptel bug
15:00.22tzangerIAX2 call to a * box which dial(Zap/g1) -- if the zaptel channel is busy it sends back BUSY but IAX2 sees hangupcause 21 - rejected... not busy
15:01.03tzafrir_laptopOcean_big, good. Now what happens when you call?
15:01.29*** join/#asterisk bongfrog (n=winston@dsl001-136-136.lax1.dsl.speakeasy.net)
15:01.54tzafrir_laptopgdh, if it were to be announce, it were already be announced, I guess
15:02.11tzangerI have to explicitly say Busy() in the dialplan or I get a rejection not busy even though chan_zap is clearly saying busy
15:02.30case_when i try to call asterisk with a linphone, it says "user cannot be found at given address". i want that anybody can call the asterisk server, what should i do?
15:02.36Ocean_bigtzafrir_laptop, nothing ^^
15:02.38tzafrir_laptopdgh, and if it is going to be released today, you really don't want to install it. Give it a day or two to sort out the most obvious problems
15:02.39gdhtzafrir: I wondered if they'd leave it until the last day, tho :)
15:03.11tzafrir_laptopIf you want to experiment, you can always take current HEAD
15:03.29Ocean_bigtzafrir_laptop, but when i call, all seem to be OK
15:03.49tzafrir_laptopOcean_big, how do you know that the call attempt did get to the Asterisk machine? Is it SIP?
15:04.07gdhtzafrir: Don't suppose you've got any 1.2.x xorcommed debs? :)
15:04.25tzafrir_laptopof the beta. Uploading them now...
15:04.52gdhhaha nice timing :) paste a sources.list line when you're ready :)
15:06.20Ocean_bigtzafrir_laptop, i don't know.. i think the call doesn't attemp asterik machine... but why ? asterisk machine is connect on my phone line. I call it with my mobil phone
15:06.54tzangercoppice: you around?  txfax seems to be dying immediately and even with option caller|debug I'm not getting any output saying why
15:07.16coppicesad, isn't it?
15:07.16Ocean_bigi 'm french, there is something to customize for french phone ? :s
15:07.55tzangercoppice: ha
15:08.09*** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net)
15:08.33*** join/#asterisk _DAW (n=bob@adsl-150-43-153.msy.bellsouth.net)
15:08.52tzangercoppice: I just called my own number with txfax and the * cli says it hung up, but I am still hearing beeping
15:09.24*** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com)
15:10.45johnmHas anyone else had problems with * HEAD, where we redifne the sent CLI. If the CLI we recieve is less than 5, we set it to somethitng else. Basically what happens is: If the original CLI is something (anything) it works. if the original CLI is blank, then we set it.. just like we do any other time.. and it doesn't work. it simply doesn't send a CLI.
15:11.13johnmis anyone else having problems with this?
15:11.56*** join/#asterisk wolfson (n=hehe@usr-kdh-208-6-58-26.beachlink.com)
15:14.23*** join/#asterisk srsergio (n=asterisk@194.red-217-216-155.user.auna.net)
15:14.44*** join/#asterisk [_gordo_] (n=gordo@bl5-187-197.dsl.telepac.pt)
15:15.09[_gordo_]zt_maxpulsetime for Matra Nortel pbx ?
15:15.17*** join/#asterisk viLeR (i=1000@66.128.47.232)
15:15.38*** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)
15:15.46generalhanwhats up everyone ?
15:16.55*** join/#asterisk fabsoft (n=settete@mercedes.cs.unibo.it)
15:17.07fabsoftsalve a tutto
15:17.08tzangerahh it's the local channel that is hanging up
15:17.10fabsofti*
15:17.11tzangeronce the call is bridged
15:17.51tzafrir_laptopOcean_big, so go a bit below to the network level
15:18.01generalhani have a question about the asternic FOP: i want to have 2 different FOPs to run so i copied all the files do a new dir and adjusted where the flash_dir is, but when i run the op_sever.pl from the new directory it still shows the old FOP, even on the new URL ? any ideas ?
15:18.30*** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net)
15:18.38rikstahis anyone using a bluetooth headset and asterisk to talk on SIP(etc) calls ?
15:18.46tzafrir_laptopyou could run a sniffer to check that SIP packets are indeed sent. I like to use tcpdump for that: tcpdump -n 'udp port 5060'
15:18.56syzygyBSDI have used it before, not right at this minute though
15:19.12rikstahsyzygyBSD that to me? if so is it realiable
15:19.32syzygyBSDrikstah: have you used a bluetooth headset with a cellphone?
15:19.42rikstahsyzygyBSD yeah of course
15:19.47Uther_Pwhat is 'syzygy' BSD ?
15:19.47fabsofti live in italy, a have got a telecom isdn TN mode network card, and also a hfc-pci chipset card, what channel i need to use hfc-pci for send/receive call ?
15:20.07tzafrir_laptopOcean_big, you can also run 'sip debug' on asterisk. This dumps every sip packet asterisk recieves, and is very verbose . Don't try to understand them. But you'll easily see when sip packets arrive
15:20.13Ocean_bigtzafrir_laptop, now, i 'm sure that the call attempt the astersik machine, because i had plugged a non-ip phone behind the card.
15:20.21syzygyBSDUther_P: the only word in english that has 3 y's and no other vowels
15:20.35Ocean_bigthe Soft PHONE is OK, i can call with it
15:20.45syzygyBSDrikstah: it is just as reliable as witha cell phone I think
15:20.55*** join/#asterisk klasstek (n=nunyobiz@sta-206-168-218-202.rockynet.com)
15:21.04rikstahsyzygyBSD did you use the app_bluetooth thing?
15:21.09tzafrir_laptopI don't understand exactly what you sa. What is working and what isn't?
15:21.23Ocean_bigthe network is very basic only one PC and the asterisk machine
15:21.27*** join/#asterisk snitt (i=snitt@snitt.info)
15:21.34Ocean_big(and a routeur)
15:21.38Uther_PsyzygyBSD: ahh... a solar or lunar eclipse, for examples, are syzygy
15:21.41Uther_P's
15:21.43syzygyBSDrikstah: no, I just had it setup like a bluetooth headset in windows using IBM's drivers
15:21.53syzygyBSDUther_P: also any 3 objects in alignment
15:21.56rikstahahh sorry i forgot to mention i'm on LINUX :P
15:22.11syzygyBSDrikstah: ya, I haven't done it on linux
15:22.44Ocean_bigtzafrir_laptop, outgoing call are OK, and Incoming call are not OK :(..
15:22.50*** join/#asterisk fugitivo (n=ajf@201.255.102.19)
15:22.51fabsofti live in italy, a have got a telecom isdn TN mode network card, and also a hfc-pci chipset card, what channel i need to use hfc-pci for send/receive call from pstn network ?
15:22.55Uther_Pheh... like the old saying  about needing to "get your ducks in a syzygy"
15:22.56fugitivohello
15:23.14Uther_P...or a game of 3 card monty :P
15:23.30Uther_Pinteresting word... it is forever burned into my vocabulary
15:23.31tzafrir_laptopOcean_big, outgoing calls and incoming calls to where? to that certain IP phone?
15:23.56darwin35back in while calls are taking off
15:24.08[_gordo_]how do i dial a flash hook from my softphone
15:24.09darwin35billibg/support/wheres the boss
15:24.16[_gordo_]i can't make it work
15:24.32*** part/#asterisk darwin35 (n=darwin35@208.139.193.178)
15:26.39*** join/#asterisk MuppetMaster (n=MuppetMa@62.37.171.235)
15:26.41MuppetMasterHello
15:27.16MuppetMasterAnyone know how to invoke the 'On-Demand' recording that may be configured for an extension within AMP?   Per this screen on the demo:  http://demo.coalescentsystems.ca/config.php?display=3
15:27.26*** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca)
15:27.43IzNoGooDUnhandled Message: prim 281 len -6 from addr 51400201, dinfo 0 on port: 1
15:27.50IzNoGooDThat's what I get on an misdn channel
15:27.59IzNoGooDsound isn't working
15:28.09IzNoGooDany idea's?
15:28.23Ocean_bigtzafrir_laptop, outgoing and incoming from non-ip phone
15:28.39tzafrir_laptopAnybody for some asterisk 1.2 (beta1) debs?
15:28.50MuppetMastertzafrir_laptop:  ?
15:28.53Ocean_bigIp phone are OK
15:29.04tzafrir_laptopdeb     http://rapid.dotsrc.org/ experimental/
15:29.33*** join/#asterisk ret28 (i=rt@82-71-120-246.dsl.in-addr.zen.co.uk)
15:29.36leheli'll try it tzafrir_laptop
15:29.44IzNoGooDme 2
15:31.11tzafrir_laptopI'll commit quite similar changes to the pkg-voip svn soon
15:31.27tzafrir_laptopI just hope everybody's happy with my version
15:32.00Ocean_bigtzafrir_laptop, the CLI doesn't show anything when i call the asterisk machine with a non-ip Phone
15:32.00ret28Does Asterisk's implementation check against From: forgery ?
15:32.54Uther_Phow would a forged sip packet do any good?  other than maybe annoying the crap out of someone
15:32.59gdhtzafrir: cheers :)
15:33.42ret28Uther_P: Pretending to be from a SIP domain that it had no right to be ... or it's possible I'm misunderstanding SIP addressing
15:34.01ret28(that is, calling somebody pretending to be somebody else)
15:34.26Uther_Pbut doesn't it use the uri to know how to send reponses back to the originator?
15:34.44ret28I thought it just sent them back along the same connection
15:34.50syleanyone using SER?
15:34.52mutilator[11:33:59] <Pyro> Dark-Fx: walked up to the sdc..it went like this "yeah my arm might be broken", "we can fit you in after the weekend"
15:34.52mutilator[11:34:03] <Pyro> so i drove to hancock
15:34.55_DAWHello
15:34.57ret28(especiall true for a TCP SIP connection)
15:35.12Uther_Pbut asterisk doesn't use tcp sip
15:35.20ret28Mmm, but it'll probably do the same for UDP
15:35.24_DAWHas anyone here used the TOUCH_MONITOR variable with one touch record?
15:35.31ret28That is, sending it back to the originating IP address
15:35.36Uther_PI would think that the protocols would be the same regardless of its transport proto
15:35.59ret28I suppose I ought to test this internally (say by forging a voip provider), and seeing what it tries to send to the outside world
15:36.16tzangercoppice: listening to txfax and the fax machine (xerox) talk
15:36.49Uther_Pret28: if my memory serves me,  the uri is used to responses... if you hit a proxy first, it encapsulates your sip message in its own,  then reinvites (if applicable) later to take itself out of the middle
15:36.53*** part/#asterisk MuppetMaster (n=MuppetMa@62.37.171.235)
15:36.54tzangercoppice: I can hear them try to negotiate, the fax warble is replaced quickly by normal modem sounds (noisy hiss) but then it stops and goes back to warble a few more times, hissing again, then just dropping out
15:38.01Uther_Pret28: I don't think sip cares what the udp packet header says the originating ip address was
15:38.44Uther_Pret28: but... I imagine someone could do something similar by faking themselves as a proxy
15:38.46*** join/#asterisk heka (n=heka@82.114.68.123)
15:39.08ret28Damn, this client really doesn't like me trying to make it spoof :>
15:39.10Uther_Pand encapsulating its own message
15:39.34Uther_Pret28: so formulate your own messages
15:39.51*** join/#asterisk danzig (n=chatzill@130.226.169.177)
15:39.59ret28At the moment, I'm thinking that'll be more effort than beating this client about, but I may turn out to be wrong :)
15:40.28ret28I was under the impression that proxying was done by keeping the SIP signalling going via the proxy (as it'll be low bandwidth), but redirecting the RTP
15:40.41ret28I can't think of a particularly good reason to want to offload the SIP signalling anyway
15:41.29Uther_Pret28: the media transport is established by means communicated inside the sip messages
15:41.48Uther_Pthe sip can work just fine without the media working at all
15:41.49InfraRedhttp://docs.info.apple.com/article.html?artnum=86816
15:42.10Uther_Phaha
15:42.12Uther_Pwho bothered
15:42.42*** join/#asterisk TK9 (n=Administ@p54B28C56.dip0.t-ipconnect.de)
15:42.56ret28"Walk to your destination by putting one foot in front of the other. /Do not walk into walls with your iMacG5/."
15:43.43Uther_Prefrain from jumping, hoping or skipping while carrying your g5 as you are likely to trip on your own stupidity
15:43.51InfraRedit took 2 months of fine tuning the document too
15:44.02ret28Is it bad politics to say something offensive about Mac users here? :)
15:44.03Uther_Pdidn't know quite how to word it?
15:44.21Uther_Pits only safe politics to degrade microsloth
15:44.43ret28Well, many of my best friends are Mac users ...
15:44.50*** join/#asterisk [hC] (n=hardcore@dsl001-136-136.lax1.dsl.speakeasy.net)
15:46.16*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
15:46.38danzigHello * Gurus :-) Question: I have a lot of Grandstream GXP2000 hardware phones belonging to students in a dorm, each student has a SIP account (with Musimi.dk, think Vonage or something) which is a SIP peer, when the phone tries to dial out Asterisk does a Dial SIP/the-persons-trunk . Asterisk says bridging call natively, it works fine, but I give very bad error messages. If the trunk says...
15:46.40danzig...408, the phone inside gets told 503. If teh trunk says 503, the phone inside gets 503. If the trunk does not exist, the phoen inside gets told 503. Is there a good/easy way of letting more informative error messages through?
15:46.40*** join/#asterisk marc324 (n=marc3234@206-248-159-56.dsl.teksavvy.com)
15:47.50Uther_Pthe trunk says 408 to your asterisk box, and it reports 503 to the phone?
15:47.53*** join/#asterisk n0where (n=kc@dsl001-136-136.lax1.dsl.speakeasy.net)
15:48.34ret28Hrrmm, this isn't good
15:50.00ret28I've successfully spoofed a From, and asterisk doesn't even try to lookup or contact the real server for that domain
15:50.05danzigYep, I think so. I am doing
15:50.06ret28(as far as tcpdump can tell)
15:50.07danzigexten => _XXXXX.,5,Dial(SIP/${account}/${EXTEN},120)
15:50.31ret28If the SIP module gave me another variables, I could execute a System call to a check in the dialplan
15:50.34ret28But nooo ...
15:50.47*** join/#asterisk wolfson` (n=hehe@usr-kdh-208-6-58-26.beachlink.com)
15:51.11ret28*more variables
15:52.22*** part/#asterisk darkskiez (n=darkskie@194.247.78.146)
15:52.43Uther_Pdanzig: do the grandstreams even have a different action to take when receiving a 408?  I wouldn't think so... the trunk says to forward somewhere else to asterisk, asterisk has nowhere else to forward, so it gives a resource unavailable to the phone... seems like what its doing is reasonable to me
15:52.44ret28Is this a Major Security Hole ?
15:52.56af_mhh, there is a webphone for asterisk, with sources available?
15:53.12Uther_Paf_:  several
15:53.21Uther_Pyou mean a soft phone?
15:53.36Uther_Pret28:  did the sip responses come back to you?
15:53.41af_no a stuff Icould phone without install nothing
15:53.47af_not a softphone
15:53.47Uther_Phah, what?
15:54.02af_I push a button on a web page, that's all
15:54.13ret28Uther_P: Yeah, it all validated as a call, finished dialling, went to my voicemail
15:54.31Uther_Phrm... I've never seen one... but I bet you could write a front end for a softphone
15:54.52Uther_Pret28:  what did the caller id say for who the caller was from?
15:55.18Uther_Pret28:  if you wanna see the sip messages as they come in, then turn sip debug messages on for your clients ip
15:55.33Uther_Pthat'll tell you if your "spoof" even worked
15:55.49tzafrir_laptopret28, how have you authenticated?
15:56.10danzigUther: yes, it is reasonable. The GXP displays the error number on the display. The problem is my users get confused - they do not understand the diffenrence between congestion as in you have no trunk and congestion as in party B is on the phone. Is there anything I can do, easier than a big block of 'jump here on chan-unavailable', 'jump there on something-else'  etc.?
15:56.39Uther_Pdanzig: yea... you can have your dialplan report congestion if the dial fails
15:56.44ret28My dialplan uses SetCallerID(${CALLERID}@${SIPFROMDOMAIN}) , which gives me a callerid of "totallybogususer1234@voiptalk.org"
15:57.03ret28callerchan=SIP/voiptalk.org-b5a00470 (from the voicemail txt file)
15:57.08ret28So it just believed what my client fed it
15:57.29ret28tzafrir_laptop: It hasn't, I'm supposed to be simulating a totally foreign call by a 3rd party trying to talk to my SIP address
15:57.47ret28(I've not got a client connected and registed with my SIP at the moment, so it's dropping to voicemail)
15:57.52tzafrir_laptopret28, AFAIK caller-ID is something that can be spoofed. You should only trust it if you have a good reason
15:58.19ret28tzafrir_laptop: But the actual SIP From address is being spoofed successfully as well
15:58.30ret28(note that I set the callerid for incoming SIP based on the From address)
15:58.35Uther_Pyea, heh... caller id is as irrelivant as the From field in an email... you can make it look like it's comming from whoever you want
15:58.41tzafrir_laptopret28, you authenticate the sip user, right?
15:58.58ret28tzafrir_laptop: Erm, no. That's the point, it's for people who aren't local users, total 3rd parties trying to call me.
15:59.02tzafrir_laptopret28, give a caller-id based on that sip user
15:59.15danzigUther: But I always get congestion/503, no matter what goes wrong... So I do not really need more reporting of congestion/503. What I really need is congestion only if the called party is engaged, and other errors (maybe recorded messages) on other errors. Anyone already done this?
15:59.36tzafrir_laptopret28, well, you shouldn't trust their caller-IDs. If you do: don't complain
15:59.37ret28And did the SIP designers really make the mistake that's long been realised with SMTP? ;)
16:00.02Uther_Pdanzig:  called party is engaged?  you mean on the phone?   if you mean the called party is using the phone, that messages is 486, not 503
16:00.04ret28tzafrir_laptop: How do I actually know that somebody calling from @voiptalk.org is a VoIPTalk user then?
16:00.07ret28(for example)
16:00.09tzafrir_laptopret28, there is no simple way to verify that address on-line
16:00.12ret28This isn't a closed system
16:00.21Uther_Pret28: its not a mistake, per se... its pointless
16:00.24ret28It's "I have a SIP address, call me from whoever your provider is"
16:00.37tzafrir_laptopret28, otherwise I can easily concive a DOS attack on voiptalk.org .
16:00.51ret28tzafrir_laptop: With callbacks?
16:00.53tzafrir_laptopNot to mention a dictionary attack to discover valid names
16:00.56ret28*verification callbacks
16:00.56Uther_Pret28: what if you are bouncing of a proxy, and a firewall?  then you couldn't get back to the originator to verify it
16:01.01terrapenI prefer an OS/2 attack.
16:01.04Uther_Pyou would jsut have to go back the way you came
16:01.24Uther_Pheh, the old smurf method
16:01.34ret28OK, what about by DNS?
16:01.46tzafrir_laptopa DOS attack leaves you with the ability to run DOS alone. That is why nobody would do a OS/2 attack
16:01.52ret28Do a lookup for the A, and both SRVs, compare against From host
16:01.57tzafrir_laptopret28, spofable just the same
16:02.03ret28I'm not suggesting verifying the originating user, just the domain
16:02.18Uther_Pret28: that call could be bouncing off many different places... proxies, firewalls?  your lookup wouldn't work the way you think
16:02.29*** join/#asterisk IronHelix (n=irc@ool-45785cfe.dyn.optonline.net)
16:02.37Uther_Pyour proxy could be (and often is) a different domain the the originator
16:02.39danzigUther: Hmmm.... Am getting 503 on _everything_ Must go check log files... Maybe my provider is causing the problem? I thought it was my fault, with the bridging, that everything ended up as a 503 inside...
16:03.04ret28So, back to the original question ... is there any way to have trustworthy sources in SIP?
16:03.05*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfjs5.dialup.mindspring.com)
16:03.34ret28Uther_P: I was under the impression that users typically proxied through their VoIP provider
16:04.25*** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
16:04.35Uther_Pdanzig:  if your truck reported 408 (timeout).. then your server would try other providers if it had any more... when it exhausts them, then it reports to the caller that 503, the service is unavailable... the service the user is trying to call, which would include any and all paths to that service the sip server has
16:04.44*** join/#asterisk terracecomm (n=terracec@164.216-123-230-0.interbaun.com)
16:05.21Uther_Pthe phone would only get a 408 if it was going to the trunk directly... as the trunk would expect that it may not be the only method out
16:05.58Uther_Pdanzig:  some people have external proxies that bounce their calls through firewalls
16:06.09Uther_Perr,,  that was directed to ret28
16:06.25terracecommdid anyone find the problem with building 1-2-0-beta1 on centos 4
16:06.35ret28Howcome their voip provider isn't doing that?
16:06.37terracecommseem to be missing a dependency
16:07.00redder86coppice: hi
16:07.06ret28(where 'doing that' == 'providing a proxy' btw)
16:07.44Uther_Pbecause sometimes both sides are behind nats and/or firewalls... and need someone outside to bounce off of first, to establish the natd entry
16:07.47danzigUther: Thanks, now I am beginning to understand. I apologize for my stupidity. There is always only one path out - what do I write in extensions.conf to get Asterisk to give the internal phone the error the trunk reported, _not_ go on to the 'next stage' - which is nothing/timeout/congestion/hangup in most cases?
16:08.33Uther_Pdanzig: not sure...hrm.   I don't know that you can through the dialplan... probably can with an agi though
16:08.48Uther_Pwhy would the end users need to know *why
16:08.52Uther_P* they can't get out
16:08.56ret28Uther_P: Confused ... is the implication that one's VoIP provider wouldn't have public IPs?
16:09.14Uther_Pisn't getting the beeping telling them they can't get out enough?  why would they need to know why?
16:09.24Uther_Pret28:  welcome to MY world, ahah
16:09.31ret28I think my 'solution' would be to just add 'Suspicious' to the callerid where lookups don't match :)
16:09.40*** join/#asterisk vp7 (n=vp7@193.27.41.43)
16:09.41ret28Uther_P: Running your own SIP domain/server behind NAT? :)
16:09.47Uther_Pret28: my VoIP provider is also my T1 provider, and they used RFC1918 addresses for their SIP and media servers
16:09.54Uther_Pit was a major pain in my ass
16:10.14ret28Don't their SRV records point to the same place as they NAT out of then? :/
16:10.21Uther_Pret28: yea, its behind my firewall... being translated to a public ip address from there
16:10.43Uther_Pret28: they DON'T nat out... heh... the packets come to ME from internal addresses
16:10.46mmlj4suggestions on where to buy digium hardware?
16:10.49vp7Hello! Could anyone tell me, if it's possible to run asterisk with H.323 support on FreeBSD? I'd like to run Asterisk as office PBX and our provider can give us only H.323
16:10.56Uther_Pthey can do that because they are also the T1 provider
16:11.08mmlj4vp7: yes, of course
16:11.24ret28Uther_P: :/
16:11.34ret28My sympathies
16:11.58ret28It's still utterly mindboggling that the SIP drafters didn't think "Hmm, e-mail From: forgery ... hey, we ought not to fall into that!"
16:12.06Uther_Pret28: so it's technically a local network to them... but it was a bitch, because I had to allow the rfc1918 addresses inbound from the external interfaces... then translate them... THEN make sure that my firewall didn't try to translate or drop the packets on the way back out
16:12.57danzigUther: A valid point... Main reason is that we have people getting frustrated because they cannot tell the difference between the 3 most common cases:
16:12.58danzig- the remote party is actually engaged
16:12.59Uther_Pfinally I just setup the cisco router to act as a nat FROM the T1 to the internal network as the public ip address of mydefault gateway
16:13.00danzig- the person has not even set a trunk up for themselves and has absoloutly no chance of ever getting out (I could handle this with a jump on chan-unavailable)
16:13.01danzig- our service provider is foobarred
16:13.03danzigTherefore they keep on trying, thingking that the remote party is home, talking on the phone, so now is a good time to call.
16:13.40Uther_Pdanzig: yea, for an admin's purpose, I can understand... but unless your users are admins... it does'nt really make any difference to them whether they know... since they can't do anything about it anyway
16:13.56ret28I think I'll just have to live with a small number of users coming up with "(SUSS)" in the callerid on my phone :D
16:14.15Uther_Pdanzig:  naa... if the other party was buzy the phone would get back a 486 message, not a 503 message
16:14.46Uther_Pdanzig:  503 == service unavailable.... 486 == Busy Here
16:14.51*** join/#asterisk JerJer[mobile] (n=jj@68.123.154.34)
16:14.53ret28I've seen people suggesting an SPF equivalent, which makes more sense than for e-mail (as there's no forwarding issues, because SIP forwarding is done via HTTP style redirects)
16:15.37*** join/#asterisk konfuzed (n=konfuzed@H129.C72.B0.tor.eicat.ca)
16:17.29*** join/#asterisk Pr0ph37 (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net)
16:17.40*** part/#asterisk Pr0ph37 (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net)
16:17.43Uther_Pret28: I think as far as reinvites from one nat to another... in the sip messages, the clients should specify to each other their external ip's (which they do), and have both send packets out at the same time.... currently one of them tries to start the media pathway to the other which is listening.... but if both of them were to just send a few packets out before the pathway is established, then both sides would have already poked a hole throught their
16:18.19ret28Oh, STUN
16:18.37Uther_Pthat way the initial packets don't get dropped at the nat with no entry to forward them
16:19.01*** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
16:19.31danzigUther: Thanks. Maybe solution is that last step in dialplan should not be congestion, but a recorded mesasge. The important thing is that they can easily (by the sound) tell the difference between busy and service unavailable...
16:19.31ret28I'm not so concerned about the actual media IP origination, just the SIP messages being authentic (which can go through a provider proxy without being too heavyweight)
16:19.39lvphello
16:20.20Uther_Pdanzig:  but their phone's display should say "486" on it if the call is busy.... besides... isn't it a faster busy tone for a 503 than for a 486?
16:20.28danzigOr maybe I should fiddle with indications.conf?
16:20.37lvpany idea if any hard SIP phone has support for using LDAP queries to implement a central phone book ?
16:21.01lvp(or any other way which is scalable and has search features)
16:21.14danzigHmmm... am having trouble testing, Do not have any relaiably busy numbers to call :-( May have to make one...
16:21.45Uther_Plvp: you could probably create an agi on asterisk and map a prefix of an extension to do the lookup
16:21.57danzigHmmm... too close for normal users, I fear. ;-)
16:22.08Uther_Pdanzig: call the phone number that you are dialing out of, heh
16:22.57lvpU: hmm..
16:23.15danzigNo, there are 4 lines available. Maybe I have set it up silly, but one can perfectly well call onesself - the line 2 lamp just starts flashing ;-)
16:24.07tzangercoppice: what TIFF format should I have my file to send in?
16:26.05ret28On another less controverisal note ... if I have SRV records for my SIP domain, but no A records, how many users would that mess about?
16:26.25ret28(as in, any common clients/PBXes lacking SRV support these days?)
16:27.05Uther_P... clients without srv support wouldn't be able to use it
16:27.24ret28But are there many of those?
16:27.36Uther_P*shrug*  not to my knowlege
16:27.43ret28Hmm, ok
16:27.47Uther_Pmost VoIP providers support srv these dayz
16:27.55Uther_Perrr VoIP devices rather
16:28.21*** join/#asterisk bgtroll (n=bgtroll@pirus.securax.be)
16:28.32lvpret: quite a lot
16:29.04lvpret: at least soft clients
16:30.00ret28Is it common for users to make calls straight from their phones without any interaction with or proxying through the provider?
16:30.37Uther_Pret28: you mean ip to ip calls?
16:30.43tzangerhttp://www.brehanbrand.com/images/lj/cameltoads.jpg
16:30.54ret28Uther_P: Mmm
16:31.32Uther_Ptzanger: bwahaha
16:31.42*** join/#asterisk n0where (n=kc@dsl001-136-136.lax1.dsl.speakeasy.net)
16:32.03tzangerI hope that busybody dropped dead from embarassment
16:32.10Uther_Phehe, hell yea
16:32.17Uther_Pthats what he/she gets for snooping
16:32.30ret28Hahaha, that's great
16:32.36ret28And yes, justice!
16:34.18Uther_P"please let me know what camel toads are and how I might be able to tell if he is smoking, taking, or licking them"... well... he'd probably only be licking, pounding or rubbing them.... a much different experiance to 'smoking the toad", but just as gratifying
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16:38.19ManxPowerUgh.  FedEx tracking is down
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16:39.25sigtermdoh
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16:52.46chidexdoes asterisk 1.0.9 have custom macros just like asterisk@home?
16:53.07*** join/#asterisk SpaceBass (n=sp@c-24-125-184-203.hsd1.va.comcast.net)
16:53.29ManxPowerchidex, Asterisk@Home IS 1.0.9
16:54.45chidexI thought asterisk@home had a little bit more done to it to help the novice
16:56.18ManxPowerchidex, Asterisk@Home does a lot of stuff for the user, but the Asterisk is the same as the one you can FVT or CVS
16:56.41tzafrir_laptopchidex, asterisk@home adds custom macros on top of asterisk. You can put yours instead.
16:57.22tzafrir_laptopasterisk@home (actually AMPortal) has a very complex dialplan. A default asterisk installation is something that is much easier to grasp
16:57.25SpaceBassbut the new version appears to have a bug with 2 or more zap channels in different contexts....
16:57.33*** join/#asterisk taec_ (n=phil@eventhorizon.hosting365.ie)
16:57.45taec_Hello, I've got a TE410P card from Digium, Asterisk and a PRI ISDN line. The Line is plugged into the first interface on the card, but I'm at a loss as to how to proceed any further. Google hasn't been kind! If anyone could provide any pointers I'd really appreciate it.
16:57.54SpaceBassall in all, it can let someone get up and running quickly
16:58.08ManxPower~docs
16:58.09jbotit has been said that 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
16:59.19tzafrir_laptopSpaceBass, the problem is how to customize it
16:59.24*** join/#asterisk sgorilla (n=tlp@cpe-24-160-119-179.houston.res.rr.com)
16:59.28taec_I've read a _lot_ of documentation, but I'm still unsure as to how to proceed with the ISDN line. Should I be setting it up with ISDN4Linux, or proceeding some other way? A small pointer in the right direction would be great.
17:00.02tzafrir_laptoptaec_, not an expert, but what card do you have, and what kernel version?
17:00.41*** join/#asterisk coppice (n=chatzill@190.196.17.210.dyn.pacific.net.hk)
17:01.02taec_TE410P and 2.6 kernel
17:01.03tzafrir_laptoptaec_, have you configured zaptel.conf and zapata.conf? for the PRI card?
17:01.30tzafrir_laptopISDN4Linux is for BRI ISDN
17:01.34ManxPowertaec_, ISDN4Linux is for ISDN BRI
17:03.09tzafrir_laptoptaec_, it is supported by zaptel (one of the t1 drivers). I wonder if my script would have configured it slightly correctly.
17:04.32taec_ok, thanks :) ... Yes, although I'm not sure if the values are correct.
17:05.04tzafrir_laptoptaec_, zap show channels
17:05.18tzafrir_laptopif it shows nothing, the cards are not configured
17:05.40tzafrir_laptoptaec_, is the module loaded? anything on /proc/zaptel/* ?
17:06.34Ariel_~seen shido6
17:06.43jbotshido6 is currently on #asterisk (1h 58m 34s)
17:07.03tzangercoppice: any idea why I am not getting any debug output at all with TxFax(/path/to/file.tiff,caller,debug) ??
17:07.04Ariel_shido6, you around? need to talk with you?
17:09.28leheltzanger: start asterisk with -cvvvddd
17:10.12coppicetzanger: no idea. which version are you using?
17:10.45ManxPowertzanger, Start Asterisk as "asterisk -cvvvddd" so you get STDERR to your console.
17:10.47taec_/proc/zaptel has 1 2 3 and 4 in it ... presuming they represent the 4 interfaces on the card
17:10.58taec_show zap channels gives a pseudo channel
17:11.02ManxPowerI don't believe you'll see STDERR/STDOUT if you don't do that
17:11.34ManxPowertzanger, I have to do the same thing when I debug AGI scripts
17:11.35*** join/#asterisk [hC] (n=hardcore@dsl001-136-136.lax1.dsl.speakeasy.net)
17:12.36*** join/#asterisk igori (n=Miranda@194.84.91.2)
17:13.37taec_zap show status gives 4 alarms ... on the channels.
17:14.29tzangercoppice: just trying to get it to work here... I converted the image to exactly 1728 pixels wide and it came across, although "stretched" a little beyonda page
17:14.32tzangerjust playing a bit now
17:14.49ManxPowertaec_, Then there is no active lines plugged into the card
17:15.31taec_The line that was plugged into our office PBX has been plugged out and brought straight into that card..
17:15.41tzangeryeah it seems to be a page width thing
17:15.44ManxPowertaec_, Then you have a wireing provlem
17:16.03ManxPowertaec_, RED Alarm means "I don't see a line"
17:16.22ManxPowertaec_, maybe you need a crossover T-1 cable rather than a straight thru T-1 cable
17:18.29*** join/#asterisk JerJer[mobile] (n=jj@dsl001-136-136.lax1.dsl.speakeasy.net)
17:19.38*** part/#asterisk bgtroll (n=bgtroll@pirus.securax.be)
17:20.12tzangerok a normal fax is 1728 pixels wide, how many pixels "long" is a regular 8.5x11" sheet?  It seems to be stretching it beyond a page length
17:20.28FuriousGeorgeanyevery installed a doorphone?
17:20.39*** part/#asterisk ret28 (i=rt@82-71-120-246.dsl.in-addr.zen.co.uk)
17:20.40*** join/#asterisk fugitivo (n=ajf@201.255.102.19)
17:21.19FuriousGeorgeim looking at those analog hookups.  it appears all i need is an fxo and the doorphpne
17:21.36tzangerFXS, no?  are you hooking the doorphone to *?
17:22.14ManxPower~fxofxs
17:22.15jbotfrom memory, fxofxs is An FXO port expects to receive dialtone and receive ring voltage.  An FXS port expects to provide dialtone and provide ring voltage.
17:22.24tzangerManxPower: yes I know that
17:22.33ManxPowertzanger, that was for FuriousGeorge
17:22.41tzangerheh
17:22.47*** join/#asterisk queuetue (n=queuetue@toronto-HSE-ppp4122670.sympatico.ca)
17:23.10distortionis there a way to send the "sip debug" output to a file?
17:23.13FuriousGeorgeyeah, but i heard that a doorphone wants an fxo
17:23.22queuetueDoes anyone use voipjet?  Are they currently down?
17:23.24ManxPowerdistortion, see /etc/asterisk/logger.conf
17:23.51FuriousGeorgeit doesnt want to dial anything, just ring the server
17:24.18ManxPowerFuriousGeorge, It would be unusual for a doorphone to not dial
17:24.57*** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com)
17:26.16distortionManxPower: thank you.
17:26.32FuriousGeorgeManxPower: http://lists.digium.com/pipermail/asterisk-biz/2005-July/007071.html
17:27.06FuriousGeorgemy understanding is, if they hit the "call:" button the fxo handles it like an incoming call, and thats why you use one of those
17:28.18*** join/#asterisk Tili (i=Tili@202-133-67-168-dialup.sat.net.pk)
17:28.26queuetueDoes anyone use voipjet?  Are they currently down?
17:28.38FuriousGeorgeit, the doorphone, doesnt need a dialtone or to ring.  the communication always goes the same way
17:28.46*** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1)
17:29.02*** join/#asterisk wrmem (n=monnin@monnin-win.cso.uiuc.edu)
17:29.57*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
17:30.23taec_ManxPower: the line almost definitely works. It's a straight through E-1 cable which was plugged originally into our PBX and now into our TE410P card.
17:30.34taec_ManxPower: I even went to the trouble of cable testing it there, just in case :
17:31.17*** join/#asterisk morale (n=russell@secure.deadbolt.ca)
17:31.21moralehas anyone had luck getting asterisk going with primus or vonage in canada?
17:32.56generalhandoes anyone know how to change the background on the FOP ? which file has to be edited ?
17:33.09taec_ManxPower: if the zaptel module is loaded and the cable is ok, would there be anything else that could be the problem?
17:33.13*** join/#asterisk toddf (n=toddf@wsip-70-182-74-104.ok.ok.cox.net)
17:34.07taec_Would there be a setting on the card I may have to configure for it to support E-1 lines?
17:34.09tzangercoppice: txfax is working as expected.  I had to adjust my scanned tiff image to account for the non-square pixels
17:34.29queuetueDoes anyone use voipjet?  Are they currently down?
17:35.39tzangerI don't use them
17:36.44jontowwoohoo
17:36.47jontow4 SNOM 320's arrived :)
17:37.24harryvvhow much per phone?
17:37.55harryvvmorale, hello?
17:38.01jontowhmm, i forget.. $210?
17:39.30moraleharryvv, eh?
17:39.37coppicetzanger: as expected? is that good or bad? :-)
17:41.45queuetueWho else do you use for outgoing?  (VOIPjet appears to be off the air...)
17:42.29tzangercoppice: so far good, now I need to see if I can get imagemagick ot create a 2-page tiff file
17:44.06moraleit looks like this inphonex.com company is the best deal.. they provide the SIP information
17:44.20*** join/#asterisk Seyr (n=Seyr@rrcs-24-227-133-226.sw.biz.rr.com)
17:44.26moraleInphonex - SIP Open - $ - prepaid available - Canadian DIDs available $8/month .. 8$ for a basic number.
17:44.41SeyrIs there any way to set the number of rings before * answers?
17:46.33tzangerSeyr: just wait()
17:47.23*** join/#asterisk scfrec (i=scfrec@scfrec.compic.ee)
17:47.35Seyrok, thanks :-)
17:48.22scfrectrying to call from voip to PSTN via asterisk. got echo, because analog phone on second site. reading manual don't help. can anyone point to right direction?
17:48.25ManxPowertaec_, there are jumpers on the board.
17:48.25*** join/#asterisk l1nux (n=moi@lns-bzn-4-82-250-119-89.adsl.proxad.net)
17:48.30l1nuxhi
17:48.38ManxPowerHowever, it should set the correct mode when you use a EU span= line.
17:49.05ManxPowerscfrec, Echo has to be removed at the VOIP/PSTN interface.
17:49.43*** join/#asterisk brainlight (n=zeldaxxx@dsl-du-83-173-249-189.cybernet.ch)
17:49.53scfrecVOIP interface - MOSA 3704B
17:49.57scfrecat analog - can't
17:50.02brainlightheya, anybody here has an eicon diva BRI-2M working on a 2.6 kernel with asterisk?
17:50.07ManxPowerscfrec, then that is the device that has to remove the echo
17:50.10scfrecif call go to GSM phone - no echo at all. only on analog
17:50.17generalhandoes anyone know how to change the background on the FOP ? which file has to be edited ?
17:50.31ManxPowerscfrec, correct.  the cell network has to cancel out echo in the PSTN/GSM interface.
17:50.35tzafrir_laptopgeneralhan, background.jpg , IIRC
17:50.50generalhani dont even see where that file is stored, or called apon
17:50.56*** part/#asterisk Seyr (n=Seyr@rrcs-24-227-133-226.sw.biz.rr.com)
17:50.57*** join/#asterisk igori (n=Miranda@194.84.91.2)
17:51.03l1nuxany one get call (over nat) success with voipuser ?
17:51.06generalhani made a background.jpg but i dont see where the old one was to write over it
17:51.15tzafrir_laptopat the same directory of thw .swf file. Check the logs of apache
17:51.28scfrecEcho Cancellation G.165/G.168 16ms - not enough ;)
17:51.48ManxPowergeneralhan, That issue is usually the browser caching the old image
17:52.13ManxPowerscfrec, It should be plenty, since the "16ms" is the latency of the PSTN part of the call, not the VoIP part of the call.
17:52.24generalhanwell if i go to the dir with the .swf file, there was never a background.jpg file there to bgin with
17:52.33l1nuxNon-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing)
17:52.34l1nux<PROTECTED>
17:52.35generalhanbut it worked anyway !
17:52.42generalhangood call guys thanks a lot !
17:52.58l1nuxwhat the codec need to use ?
17:52.58scfrecManxPower: problem i hear myself becouse phone on other side transmit my voice back
17:53.00ManxPowerl1nux, Um, Asterisk does not support G273.1
17:53.14l1nuxohh..
17:53.24ManxPowerscfrec, All analog phones do that, it's just that until VoIP came along you could not hear the echo because it happened so fast.
17:53.48harryvvHere comes the voip compitition to vancouver! http://www.canada.com/vancouver/vancouversun/news/business/story.html?id=f42b3c54-c400-4bb6-9305-40c849bd85b3
17:54.05harryvvshaw cable is now selling voip across its cable.
17:54.08ManxPowerscfrec, this is covered over and over and over again in the mailing list archives and there is much information on the Wiki
17:54.15ManxPower~mailinglist
17:54.17jbotmailinglist is probably Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
17:54.17ManxPower~docs
17:54.18jbotsomebody 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
17:54.19scfrecManxPower: echo return in 0.2 seconds.
17:54.42ManxPowerscfrec, you can believe me ot not.  I'm still right.
17:54.47scfrecManxPower: when i call from software phone - same problem - i hear myself
17:55.03scfrecso sound back to line on second side
17:55.14ManxPowerscfrec, that is expected.
17:55.25ManxPoweranywhere there is analog there will be echo.
17:55.28l1nuxManxPower, what the codec need to use with voipuser.org ?
17:55.40ManxPowerl1nux, I don't know.  What codecs do they support?
17:55.45scfrecManxPower: solution?
17:56.21ManxPowerscfrec, the solution is for your VoIP/PSTN gateway to cancel out the echo.
17:56.40ManxPowerSince you are using a 3rd party gateway, there  is NOTHING we can do to help.
17:57.09scfrecManxPower: it's possible todo on asterisk?
17:58.05ManxPowerscardinal, only if asterisk handles the PSTN/VoIP conversion.
17:59.31scfrecManxPower: schema is nexT: phone-gateway-asterisk(my)-asterisk(provider)-dont know-phone
18:00.20Ariel_JerJer[mobile], are you around? I have a biz question for you... 800 number service?
18:00.29*** part/#asterisk Ldr (n=Lode@194.206.157.226)
18:01.20InfraRed800 numbers
18:01.24InfraRedthanks for reminding me!
18:01.53tzangerfuck
18:01.55Ariel_InfraRed, that a way to do it.
18:02.11InfraRedAriel_: i have a more cunning plan :)
18:02.16tzangerI can't seem to get the return code from TxFax because the other side hangup is too fast
18:02.28ManxPowerscfrec, then complain to your provider since the provider is doing the PSTN/Voip conversion
18:02.29tzangerand there's no txfax (go on in context on hangup) :-)
18:03.04Ariel_There seems to be issues with the internet today... Some locations are down.
18:03.07Ariel_or very slow.
18:04.05harryvvhi Ariel looks like shaw cable is invading the voip market in vancouver BC.
18:04.31harryvvAriel, do you have a backbone web site that shows sites are down?
18:05.12l1nuxManxPower, from http://www.voipuser.org/forum_topic_330.html "g729 & gsm" ):
18:05.36Ariel_harryvv, no
18:05.49ManxPowerl1nux, H729 is a patented codec and requires a licensing fee to use
18:05.53ManxPowerso I guess you must use GSM
18:06.31*** part/#asterisk jcollie (n=jcollie@lt16586.campus.dmacc.edu)
18:06.51l1nuxManxPower, i dont have gsm in my ata device ):
18:08.08ManxPowerl1nux, then you'll need to use ulaw or alaw from your device to Asterisk, then from Asterisk to voipuser would be GSM
18:08.55l1nuxManxPower, ohh! is possible
18:09.12harryvvAriel, here is a snapshot of internet traffic.
18:09.15harryvvhttp://www.internettrafficreport.com/namerica.htm#graphs
18:09.16ManxPowerl1nux, Most people use ulaw or alaw (but not both) for the local network
18:10.17BrianR___ulaw on the lan - especially since many of the cheaper hardphones suck and fall over if you try to run too many simulataneous calls with compression :)
18:11.08harryvvIs there any current voip showsalers that can transfer my telus phone number as a did?
18:11.20harryvvwhosalers that is
18:11.29l1nuxok, thanks ManxPower :)
18:12.41Ariel_argh the wait for the zap to be realy is exten => _X.,1,Dial(Zap/g0,www/${EXTEN})
18:12.44vp7Hello! Could anyone tell me, if it's possible to run asterisk with H.323 support on FreeBSD? I'd like to run Asterisk as office PBX and our provider can give us only H.323
18:13.45*** join/#asterisk [hC] (n=hardcore@dsl001-136-136.lax1.dsl.speakeasy.net)
18:13.56ManxPowerAriel_, Actually  exten => _X.,1,Dial(Zap/g0/wwww${EXTEN})
18:14.02Ariel_vp7, asterisk can support h323 as an adon. I don't know about freebsd since I don't use it.
18:14.17Ariel_ManxPower, thanks my head is not on right today.
18:14.32ManxPowerAriel_, and as you know it only works on analog interfaces
18:15.09Ariel_only works on the tdm400 and the x100p not the t1 pri? or a channelized t1?
18:15.48ManxPowerAriel_, Correct
18:15.56ManxPowerit might work on T-1<->Channelbank
18:15.57Ariel_ManxPower, thanks...
18:16.01vp7ARiel_: And how about Linux? I tried to find valid versions of openh323&pwlib but didn't find any that can be compiled successfully with asterisk. Do you know where can i donwload it?
18:16.06Ariel_yes I know it works on a c/b
18:16.07*** join/#asterisk darkskiez (n=darkskie@host86-138-169-183.range86-138.btcentralplus.com)
18:16.14ManxPowerOh!  It WILL work on channelized T-1.
18:16.18Ariel_vp7, yes
18:16.26Ariel_linux works great I use CentOS
18:16.36ManxPowervp7, There are at least FOUR H323 drivers for Asterisk
18:17.34ManxPowerchan_h323 (NuFone, included with Asterisk).  asterisk-oh323 (3rd party download), chan_ooh323 (from asterisk-addons), and chan_woomera (openpbx.org I think)
18:17.53vp7<PROTECTED>
18:17.58*** join/#asterisk fugitivo (n=ajf@201.255.102.19)
18:18.13Ariel_vp7, gentoo works
18:18.22Ariel_~docs
18:18.24jbotrumour has it, 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
18:18.29l1nuxManxPower, no sound ):
18:18.31vp7ManxPower: Maybe i've done some wrong things, but these channels didn't wanted to compile :(
18:18.32fugitivogentoo rocks
18:18.33Ariel_vp7, see the wiki
18:18.51Ariel_vp7, I don't use h323 for any of my setups.
18:19.12vp7Ariel_: thnx, i'll try to search again. Hope this time will be more efficient :)
18:19.35l1nuxis only one way audio
18:20.20vp7Ariel_: Btw, our provider gives us H.323. Maybe the better way will be to convert it into SIP with external software? Do you know how is better?
18:20.37ManxPowerl1nux, sounds like a NAT problem
18:20.46brainlightheya, anybody here has an eicon diva BRI-2M working on a 2.6 kernel with asterisk?
18:21.11Ariel_vp7, no but there are other h323 channels that work.
18:21.30l1nuxManxPower, possible, but work fine with sipphone and others
18:22.02l1nuxone way audio, is only with voipuser ):
18:22.15harryvvwho has a track record of reliability and the infrastructor to keep up with demand?
18:22.19Ariel_argh where are the people from nufone when you need them.
18:22.39harryvvand has also some kind of customer service?
18:23.06*** join/#asterisk sjaak538 (n=sjaaknab@d5c53145.dsl.concepts.nl)
18:23.50harryvvohh common somone must know from experaince?
18:24.04l1nuxManxPower, howto debug it ?
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18:41.14SeyrIm using RealTime and having problems with MWI. I have rtcachefriends=yes and I can leave a message and retrieve them, just no light on the handset for my 7960.
18:41.52*** part/#asterisk reperire (n=nathan@wbs-146-168-56.telkomadsl.co.za)
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18:49.43*** join/#asterisk d33t (n=d33t@c-24-13-33-195.hsd1.il.comcast.net)
18:50.12d33tanyone around to answer a quick * question?
18:50.32Uther_Pjust ask... you don't need permission to ask
18:50.41d33theh, k, thanks.
18:51.00d33ti'm trying to connect my asterisk server to sipphone and i can't get it to see incoming calls
18:51.12Hmmhesaysanyone wanna buy a t1 card off me?
18:51.17Hmmhesaysnew
18:51.32*** join/#asterisk asterisk99 (n=dunoowhy@modemcable169.194-130-66.mc.videotron.ca)
18:51.36bjohnson$5
18:51.36Uther_Pd33t: are you registering?
18:51.37d33ti can make outbound calls to other sip networks and use my "gizmo out" minutes, but nothing comes
18:51.40d33t*in
18:51.41d33tyes
18:51.51d33tit syas registered on sipphone.com
18:52.02Jabroniguys lil' question, is the pickupexten parameter introduced in the asterisk 1.0.9 release??  I added pickupexten = *9 but when a phone dials it sip returns a 484
18:52.04d33t<PROTECTED>
18:52.16Uther_Pdoes the context towhich it's registering in exist, and does it have an 's' extension?
18:52.25asterisk99anyone know how to keep asterisk from chopping off 1st 1/4 second from sound files.... example "Welcome" comes out "Come" ... don't want that!!!
18:52.43Seyrasterisk99: add a Wait(1) before playing the file
18:52.44*** join/#asterisk _Thor (n=Christia@rrcs-24-39-156-126.nyc.biz.rr.com)
18:52.45d33tthe context exists, but i don't have an s extension
18:52.50d33theh...... i'm such a n00b
18:52.50Hmmhesaysi'd probably let it go for 4fiddy
18:53.27Uther_Pd33t:  actually, you probably only need an extension for the phone number that the calls would be comming in on
18:53.27d33ti thought i use the exten => SIPNUMBER,x,..... context
18:53.34_Thoranyone knows sangoma installation?
18:53.36bjohnsonasterisk99: add a wait before picking up
18:53.40d33ti have this.....
18:53.46asterisk99Seyr: You;d think that would work... but I still get COME - even with wait(2)
18:53.47d33texten => ${SIPPHONENUM},1,NoOp(Incoming call from SipPhone Account ${SIPPHONENUM})
18:53.51d33tand the variable is set
18:54.03d33tand of course, other lines follow
18:54.04d33theh
18:54.09bjohnsonI don't have any fiddy's
18:54.35Uther_Pd33t: try hard coding the number
18:54.40d33tk
18:54.42bjohnsonhow about 4fathers
18:55.08Uther_Pand add an 'i' extension
18:55.14Seyrasterisk99: no idea :-(
18:55.24_ThorSangoma T1 installation, anyone knows how to configure it?
18:55.30d33tstill nothing, right to sipphone vm system
18:55.43d33ti am running * in verbose mode and the call never seems to come in
18:56.07Uther_Pd33t: I'm going to assume you reloaded after changing the dialplan
18:56.12d33tyes
18:56.20d33theh, i'm not that new :)
18:56.28bjohnsonand verbose is level 5 or higher
18:56.40Uther_Pturn on sip debugging on your provider... see if they ever even send you an invite msg
18:56.46d33tusing 6 v's
18:56.56d33tasterisk -c -vvvvvv
18:57.44bjohnsoneasy there fella .. we just met
18:58.16Uther_Pjust friends, I swear
18:58.19asterisk99Seyr: There's a trick... answer() first... THEN Wait(1)
18:58.19Uther_Pheh
18:58.55d33tUther_P: any idea where i would turn on "sip debugging" at sipphone.com or gizmoproject.com?
18:59.01d33ti don't see anything like that
18:59.03Uther_Pasterisk99: I was going to suggest that... but didn't think it would have made a diff since you put the wait on, hrm
18:59.09Uther_Pd33t:  in the cli
18:59.34Uther_Pd33t:  sip debug ip (ip of the provider)   or  sip debug peer (peer name)
19:00.00Seyrasterisk99: ah, I do that by default :-)
19:00.03Uther_Pit auto completes... so if you just type  "sip debug peer " then push tab, it'll show you the peers to choose from a list
19:00.10asterisk99Uther_P:    Astwisk is vewy twicky  ;)
19:00.50SeyrAnyone know why I cannot get MWI when using Realtime with rtcachefriends=yes?
19:00.52asterisk99anywone know why the default 's' extension does not work for PRIs?
19:00.57d33ti can't tab the peer's name
19:00.58*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
19:01.05d33twhen i do sip show peers i get....
19:01.08Uther_Pd33t:  type  'sip show peers'
19:01.13d33tName/username    Host            Dyn Nat ACL Mask             Port     Status
19:01.13d33tdesktop/desktop  (Unspecified)    D   N      255.255.255.255  0        Unmonitored
19:01.13d33tcordless/cordle  192.168.1.50     D   N      255.255.255.255  5060     Unmonitored
19:01.14d33tproxy01.sipphon  198.65.166.131       N      255.255.255.255  5060     Unmonitored
19:01.15Uther_Peek
19:01.27Uther_Pdude,   http://pastebin.ca
19:01.54d33tthe desktop softphone is not running, the cordless (on a grandstream) is connected, and it seems sipphone is as well
19:02.04Uther_Pd33t:  sip debug peer proxy01.sipphone.com   or whatever that name is
19:02.13d33tyeah, that's right
19:02.36d33tok, debugging enabled...... try the call again?
19:02.40ManxPower<PROTECTED>
19:03.09Uther_Pyea.. his problem is the proxy entry
19:03.39d33tlooks like it came in
19:03.40Uther_Pd33t: yea, call it... if its comming back to you, it'll dump a bunch of sip messages
19:04.00d33tit did
19:04.22Jabronianyone uses the call pickup module ?
19:05.31d33tanything i should be looking for in there?
19:06.17Uther_Pd33t:  paste your sip.conf  on http://pastebin.ca
19:06.41d33tgive me sec, i need to strip out the comments
19:06.47lehelgoodbye all
19:07.20Uther_Pd33t:  cat sip.conf | grep -vE '/^;/'
19:07.59d33theh, gues that'd work well too
19:10.27*** join/#asterisk JerJer[mobile] (n=jj@dsl001-136-136.lax1.dsl.speakeasy.net)
19:11.44d33tuther_p: sent link
19:12.12*** join/#asterisk tsetane (n=tsetane@87.252.68.0)
19:13.15Uther_Pd33t:   register => user[:secret[:authuser]]@host[:port][/extension]  is the syntax... you aren't specifying an extension, thus incomming calls are dropped into the 's' extension
19:13.44d33tah, was never sure how that worked. like i said, n00b
19:13.53d33tso if i put my sipphone number in the extension it'll work?
19:14.24*** part/#asterisk tsetane (n=tsetane@87.252.68.0)
19:14.39d33tregister => 1747601xxxx:xxxx@proxy01.sipphone.com/1747601xxxx
19:14.41d33tlike that?
19:14.45Uther_Pd33t: yea
19:14.57d33tok, i'll give it a go
19:15.13d33tis there a link you can send me to that explains the s, r, t, etc extensions?
19:15.24Uther_Pbut you can put whatever extension you want at the end there... the last option specifies what local extension calls come into from that provider
19:15.35d33toh
19:15.49d33tso if i redo the extensions.conf to that ext it will use that
19:15.51d33ti see
19:15.58Uther_Pd33t:  s is default or 's'tart,  t is timeout,  i is invalid.... not sure about r
19:16.18Uther_Por you just put in an 's' extension
19:16.24d33ta lot of examples i read use r, i think it works like timeout but just proceeds to the next line
19:16.38d33tthat would also work i guess
19:17.04d33tbut i want to set it up to take calls in from FWD too
19:17.08*** join/#asterisk buddah (n=djbrianc@67.110.253.129)
19:17.19d33tand handle them differnetly
19:17.24buddahdoes anyone know how to fix the problem with cisco ata 186's second line not ringing
19:17.24buddah?
19:17.28d33tso i don't want to just use the default s
19:17.36*** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net)
19:17.38*** join/#asterisk Veto_laptop (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com)
19:17.54tzafrir_laptopwhich is the current "stable" spandsp? 0.2 or 0.3? is 0.3 "fluid" or the recommended version?
19:17.58Uther_Pthen either specify different extensions to dump the incomming calls
19:18.09d33tyeah, 'tis my plan
19:18.28d33tawesome!
19:18.32d33tworks liek a charm
19:18.34d33tthanks!
19:18.42Uther_Pd33t:  extensions can also be words, as long as they aren't reserved words
19:18.46Uther_Pno prob
19:18.58d33tah, that explains some of the other examples i saw
19:19.03d33tcan you also use contexts?
19:19.24d33tlike use /sipin and have a sipin context with an s extension?
19:19.35buddahanyone heard of pap2-na's having issues where the second line won't ring?
19:19.37Uther_Pe.g.  if you wanted to have FWD and SIPPHONE extensions... you end your registration with /FWD or /SIPPHONE
19:20.17d33tcan i use them as contexts too then, or just as extensions in the default context?
19:20.28Uther_Pno, those are extensions
19:20.32d33tok
19:20.51d33tsweet, you taught me more than you know, heh
19:20.52d33tthanks again
19:21.05Uther_Pthey are dumped in the context that is specified in your sip.conf... OR you can use a provider name by creating it as a peer
19:21.13Uther_Pthen putting that peer in a specific context
19:21.50d33ti think that's a bit over my head
19:21.57*** part/#asterisk TK9 (n=Administ@p54B28C56.dip0.t-ipconnect.de)
19:22.03d33ti'll stick with the default context
19:22.06d33tsince it works :)
19:22.16*** join/#asterisk Utah_Dave (n=boucha@0-1pool138-209.nas28.salt-lake-city1.ut.us.da.qwest.net)
19:22.17*** join/#asterisk n0where (n=kc@dsl001-136-136.lax1.dsl.speakeasy.net)
19:22.22Uther_Pd33t: much like you made the sip entries for you phones... you make one for your provider, and specify a context
19:22.58d33tyeah, i've seen it donw that way before too, but i couldn't get that working either
19:23.08d33talthough, now that i know a little more, i might be able to
19:23.28d33tbut, is there an advantage 1 way or the other, besides he config being a little more clear?
19:24.27SeyrAnyone know why I cannot get MWI when using Realtime with rtcachefriends=yes?
19:25.31Uther_Pd33t: for a simple config... no... its helpfull though, if you have many lines/providers and/or sets of extensions that must be handled a different way
19:26.25Uther_Pd33t:  especially if your provider will be sending calls to you where the target extension varies... then it enters the context at the extension of the number
19:26.33*** part/#asterisk n0where (n=kc@dsl001-136-136.lax1.dsl.speakeasy.net)
19:27.39d33tok
19:29.42*** join/#asterisk Tili (i=Tili@202-133-65-86-dialup.sat.net.pk)
19:31.25FuriousGeorgehas anyone ever installed a doorphone?
19:31.56FuriousGeorgespecifically an analog one.  like the some of the ones from viking
19:32.04*** join/#asterisk corne (n=corne@ndn-165-157-254.telkomadsl.co.za)
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19:35.36*** join/#asterisk pbd (n=plancomm@12.144.118.36)
19:35.44pbdGood afternoon, all.
19:35.52*** join/#asterisk jayk- (i=jayk@lasziv.reprehensible.net)
19:36.08jayk-what are the advantages to using chan_sccp over SIP for the Cisco 7960 phones? should i use SIP or chan_sccp?
19:36.22pbdJayk: Your Mileage May Vary.
19:36.45pbdI've got a 7960 on my desk right now running SCCP attached to Callmanager, and another one running SIP attached to Asterisk.
19:36.54jayk-ah.
19:37.17jayk-i have problems with the cisco phones dropping calls occasionally, getting hung up (where they won't dial) and need to be rebooted to have the problems fixed.
19:37.19pbdchan_sccp (or chan_skinny) have various strengths and weaknesses- most of which lie in the firmware revision.
19:37.25*** join/#asterisk zeedo (n=zeedo@80.68.92.188)
19:37.25jayk-and im wondering if it is because they are SIP
19:37.31pbdUnder SCCP, I've seen that problem- but never under SIP.
19:37.43pbdBut I've heard other people say they've seen it too.
19:37.50jayk-i'm running SIP7.5
19:37.54pbdSo am I.
19:37.56jayk-hrm
19:38.10jayk-sometimes i might pick up phone and dial, and it won't connect, but if i try it again, it will work
19:38.26pbdThat's something to check your asterisk console about.
19:38.40pbdI have seen cases where the console reports that the phone is lagged- I had to adjust for that.
19:38.52jayk-when it does that, it doesn't output anything on the console
19:38.58pbdI'm trying to track that one down- I *think* it's in relation to CDP and the switches I use.
19:39.01jayk-how did you adjust for that?
19:39.09jayk-i don't use CDP on the switches
19:39.09pbdDo you have verbose and debug turned up enough?
19:39.16jayk-i have it turned up to about 9. :)
19:39.22*** join/#asterisk fordvoice (n=chrisf0r@cpe-69-133-21-43.cinci.res.rr.com)
19:39.51pbdYou adjust the milliseconds of qualify  in the sip.conf entry.
19:39.54*** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net)
19:39.55pbdI find 500ms works well.
19:40.04Ariel_Just wonder why it's so hard to get good customer service in the Voip/Telecom buisness.
19:40.05jayk-the phones are all less than 10ms
19:40.24pbdI see this even when I can ping the phones from the server at <2ms .. but Asterisk (HEAD) reports it's TOO LAGGED, at >200ms.
19:40.43jayk-huh
19:40.43pbdSome folks have reported losing calls when the TOO LAGGED message comes across.. dropouts, that sort of thing.
19:40.46jayk-so 500ms works?
19:40.49Seyrqualify isnt ping
19:40.52pbdIt works better.
19:41.05jayk-qualify=500ms?
19:41.13jayk-or just 500 isnt it
19:41.15pbdNo, it's not ping- there's a lot of stuff there.. I'm seeing it as the phones are basically taking too long to respond to the headers.
19:41.24pbdYou don't need the 'ms' on the line
19:41.36jayk-ill try that
19:41.57jayk-so 500ms doesn't degrade from the voice quality?
19:41.59pbdSomeone here gave me a pointer that CDP, or the phone's attempt to work with it, may be slowing down the phone's response.
19:42.07*** join/#asterisk [_gordo_] (n=gordo@bl5-187-197.dsl.telepac.pt)
19:42.24pbdqualify doesn't have anything to do with the voice quality itself... merely how long it takes to answer the headers.
19:42.25jayk-i have tos=lowdelay
19:42.27jayk-could that be part of it?
19:43.07pbdDepends on your switch.  Are you running QOS?
19:43.10jayk-nope
19:43.19pbd(you should, btw, in almost all cases)
19:43.31pbdThen tos bits won't do anything one way or another.
19:43.32jayk-i've never used it before.
19:44.21jayk-how is it configured?
19:44.26pbdQOS is defined on your routers and switches- essentially, it moves certain traffic through before other traffic.  Even on a relatively uncongested lan segment, it may matter- it helps mitigate bursting.
19:44.55pbdIt's configured differently on each manufacturer's network equipment.
19:45.04jayk-i have all cisco switches and routers
19:45.15pbdIf you have a simple hub attached to your phones and the server, with no other devices on the hub, you wouldn't need it.
19:45.37pbdIf you're running them all as part of a LAN/WAN environment, with mixed loads, etc.. you probably need QOS running.
19:45.53jayk-k
19:45.56pbdemphasis on probably- your LAN may be different.
19:46.42pbdAnd it's not a panacea.. it merely helps.  In practice, it's a way of saying- look, switches/routers, please make sure the RTP for my phones goes through before the pr0n for the web browsers.
19:47.09jayk-k
19:47.11jayk-ill check into it
19:47.18pbdBut, if there's a lot of pr0n, and only a little RTP, it might not do much for you.
19:47.37pbdNow, I'll trade someone out there that longwinded answer for any experience in setting up caller-ID in Brazil?
19:48.15pbdI'll be testing it next week, but I'm hoping someone out there has some basic experience they can share- like 'I got it to work', or 'It will never work- you're on drugs'.
19:48.30*** join/#asterisk tsetane (n=tsetane@87.252.68.0)
19:48.35pbdAnd no, I don't have any extra to share. :)
19:49.13jayk-wow, interesting
19:49.17jayk-Oct 13 12:48:46 NOTICE[2216]: chan_sip.c:9598 handle_response_peerpoke: Peer '102' is now REACHABLE! (241ms / 500ms)
19:49.22jayk-i just turned on qualify and got that msg
19:49.43jayk-only a couple of phones did that
19:50.24pbdIf you keep watching, you'll see it for all the phones eventually.
19:50.35jayk-ok
19:50.41jayk-Oct 13 12:50:16 NOTICE[2216]: chan_sip.c:9604 handle_response_peerpoke: Peer '26201' is now TOO LAGGED! (501ms / 500ms)
19:50.43jayk-got that too
19:51.13*** join/#asterisk Gnurdux (n=gnurdux@69.251.241.119)
19:51.17jayk-i have callerid working here in the states. :)
19:51.19*** join/#asterisk obsidian-studios (n=obsidian@c-66-177-188-197.hsd1.fl.comcast.net)
19:51.26jayk-but that was pretty simple, after we converted to a PRI from a voice t1
19:51.31Gnurduxok.... im trying to setup asterisk to connect to fwd
19:51.47pbdYeah, caller ID in the US is a no brainer.
19:51.49Gnurduxi installed asterisk@home on an old box with no terminal
19:51.59Gnurduxand i followed the directions to setup FWD
19:52.07Gnurduxbut im not sure how to use it
19:52.16obsidian-studiosif using iax2 between * boxes what should I base the bandwidth on per line using ulaw? 80k? 20k? It's more about codec than protocol right?
19:52.33pbdJayk: if you do a sip show peers, it will show you what the lag time is currently.
19:52.50pbdobsidian: you are correct- it's more codec than protocll.
19:53.02pbdBut IAX takes a little smaller bite than SIP or H323..
19:53.11pbdwe're talking about 79K vs 80K, though.
19:53.19*** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn)
19:53.26jayk-oh yeah, look at that
19:53.35jayk-that didn't show up before (the lag time)
19:53.50pbdIt may only show up if you have qualfy on.
19:54.28obsidian-studiospbd: what's realistic for licensed codecs 20k or 40k?
19:54.38jayk-i think you are right
19:54.50jayk-thanks pbd. :)
19:54.52pbdWhich licensed codec?
19:55.22pbdIn theory, g.729 is about 8K.. GSM around 12K.
19:55.45pbdthere's a couple of good bandwidth calculators around- google for them, I don't have the urls' handy.
19:56.15Gnurduxcan someone help me?
19:56.22pbdKeep in mind, there's some overhead- that's just the RTP bandwidth.  I usually count around 20K for a compressed codec, and 80K for ulaw, regardless.
19:56.23obsidian-studiospbd: ok, just trying to get guestimates for now, no worries on exacts, is there a preferred licensed codec?
19:56.39tzafrir_laptopGnurdux, if you ask your question: maybe
19:56.52obsidian-studiospbd: tyring to stick with ulaw, but I think there are bandwidth limitations at one location
19:56.58pbdLicensed, I've seen people use 729 ($10 per channel), or 723 (ridiculous cost).. or GSM (preferred in Asterisk community- it's free and pretty good).
19:57.01jayk-w
19:57.36tzafrir_laptopspeex should give a good quality, but takes very much CPU
19:57.40obsidian-studiospbd: quality? pretty sure phones will run ulaw, might use different codec in phone if available to avoid transcoding
19:57.48obsidian-studiospbd: gsm is native to * right?
19:58.01*** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl)
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19:58.21tzafrir_laptopobsidian-studios, just as much as other codecs (ilbc, g726, etc.)
19:58.38obsidian-studiostzafrir_laptop: all modules so * does not care
19:59.10pbdI run 729, ulaw, and gsm.  They all have 'native' drivers in Asterisk.. including some I haven't mentioned.  To my ear, 729 and gsm sound about the same, ulaw a little better.  Don't try faxing over a compressed codec, as it narrows the frequency range.. but the human ear is pretty versitile.  If you have users with perfect pitch, they're going to complain at anything less than ulaw.
19:59.54Gnurduxtzafrir_laptop, the prob is that my question isnt that well-defined
19:59.55pbda PRI runs a form of ulaw/alaw- its what you'd get on a high quality digital line, since there is no compression.
20:00.12Gnurduxi setup FWD using the direction of the asterisk@home handbook
20:00.18pbdGnurdux: Since you have no well defined question, the best answer we can give is 'maybe'. :)
20:00.22Gnurduxand i dont know if i didnt get it setup properly
20:00.27Gnurduxor i dont know how to use it
20:00.43Gnurduxbut i cant call my old FWD contacts when i tell Kphone to connect to my asterisk box
20:01.36pbdGnurdux: Sounds like it's time to go into the asterisk console and do a little 'set verbose 4' and 'set debug 4'- and see what the messages say.
20:01.41obsidian-studiospbd: cool ty, one client is getting a PRI, but the other is a non-profit with limited budget
20:01.56Gnurduxhmm
20:02.11SeyrAnyone know why I cannot get MWI when using Realtime with rtcachefriends=yes?
20:02.13Gnurduxhow do you get to the asterisk console?
20:02.18tzafrir_laptopGnurdux, you can try to call the clock or the echo test on FWD (612, 613)
20:02.38tzafrir_laptopGnurdux, there's also an option in the web interface to send a call to your number
20:02.43Gnurduxoh
20:02.44Gnurduxok
20:02.56pbdGnurdux: Get onto the linux box where you installed asterisk, and type 'asterisk -r'.  It's all ugly from there. :)
20:03.21*** join/#asterisk X-Rob (n=rob@dsl-202-173-151-24.qld.westnet.com.au)
20:04.08sgorillahas anyone here messed around with access grid?
20:04.08Gnurduxtzafrir_laptop, how do i tell it to call me from the web interface?
20:04.20sgorillaor have a logitech quick cam 4000
20:04.49FuriousGeorgeanyone ever installed a door phone?  ive seen people say they hook to fxo which makes sense if they only have a call button, and ive heard other people say they hook to fxs, which makes sense if they can dial extensions and whatnot\
20:04.53FuriousGeorgeanyone know?
20:05.13FuriousGeorgei wouldnt wanna buy anything w/o being sure i got the right thing
20:05.21pbdFurious: Most door phones run FXS- they're essentially bat phones.
20:05.28fugitivoFuriousGeorge: you can get adapters for FXS
20:05.29Gnurdux612 and 613 say address incomplete btw
20:05.36pbdBut google for 'door phone', and see what you get.
20:05.51*** join/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net)
20:05.58*** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl)
20:06.05mizticFuriousGeorge, i'll be getting into that soon, the doorphone we have acts like a phone extention, so i expect ours to require an FXS connection
20:06.05FuriousGeorgepbd: ive been using the google search on the wiki, getting conflicting info
20:06.33pbdGeorge: In the end, it's going to depend on the model you pick.
20:06.41fugitivoFuriousGeorge: www.2n.cz, they have nice stuff
20:06.53mizticit'll get more fun when i have to activate the door contacts with a button on the phone
20:06.54pbdI can't imagine one running on FXO, since door phones typically don't have power to them.. but anything is possible.
20:07.00FuriousGeorgemiztic: ive hooked a POTS phone to an fxs before and hung it by the front door, but i need a real outside intercom for this one, weather resistant, etc
20:07.20mizticours came with the propriatary lucent system
20:07.24mcf3782I have a "Can Asterisk do this...." question. :)
20:07.25miztici plan to reuse it
20:07.26FuriousGeorgepbd: thats what i figured, it would be powered and it would ring the fxo when somene hit call
20:07.32fugitivoFuriousGeorge: www.2n.cz, check Communicators section
20:07.47FuriousGeorgefugitivo: checking
20:07.53Uther_Pits not uncommon for pbx systems to have functionality for door phones to buzz open the door
20:08.14pbdGeorge: It's a lot simpler to make a door phone unpowered- needing an FXS port, as the hardware is simpler and you don't need a separate power supply.  But I suppose you could make one loop start, like an alarm line, or something.
20:08.47mcf3782I have a single POTS line from BellSouth that has 3-way calling on it. To use the 3-way feature, you flash the switch hook, get a second dial tone, dial the other number, then flash again to get all 3 parties connected.
20:08.50pbdUther: True- but electric locks are generally higher amperage, 12v systems.. not compatible with phones.
20:09.18pbdSo it's more common to have the server talk to the lock through separate wire.  Added security advantage- you can't override the lock by isolating the phone.
20:09.41Uther_Ppbd:  yea, would be best to have another device for buzzing it open,  and use the door phone function as a logic gate to it
20:10.05mcf3782Can Asterisk put an inbound call "on hold", dial a second call, and then join them together on request when someone on the pots line does something like say dial "*2" (doesn't really matter to me what code I use).
20:10.26*** join/#asterisk fiber0pti (n=johndoe@207.114.199.98)
20:10.40Uther_Pmcf3782:  in the dialplan,,, you would want  to use  the flash application.. then senddtmf,  then flash
20:11.16Uther_Pflash,  wait(1), senddtmf(number), wait(1), flash
20:11.24mcf3782Ahh. cool. I figured the answer was probably "yes". Just wanted to verify that before I went off and started trying to figure out how to do it. :)
20:11.38Uther_Ptrying to bounce calls of your work? heh
20:12.35pbdUther: That only works if you have three way calling on your line.  Simpler if you have multiple lines on your system. :)
20:13.14mcf3782I want a way for my parents, who really don't like all my "gadgets" and stuff, to be able to just have Asterisk put them on hold, dial my cell phone, and then 3-way conference us together without them having to hang up and call my cell phone if I don't happen to be at home.
20:13.22Uther_Pmcf3782: remember, in this instance there is no dial app to get blocked on for the call... do unles you expect that your provider allows for unattended conference calls, then you'll have to put in a loop in the dialplan, simulating an indefinate wait so it doesn't establish the call, then hangup on the both of you
20:13.37*** part/#asterisk Gnurdux (n=gnurdux@69.251.241.119)
20:14.23mcf3782oh. ok. good to know. Thanks for that pointer, Uther_P.
20:14.34Uther_Pno problem.. i;ve done this before, heh
20:14.38pbdmcf: Then you're in fine shape.  I'd personally subscribe to a low cost outbound only VoIP provider for the outbound leg.. then simply execute a DIAL application out the VoIP provider to your cell phone.
20:15.15Uther_Pbah, no need for all that for just a 3 way
20:15.17pbd(it's simpler that way, and no three way calling charges).
20:15.22mcf3782That may certainly be an option.  I'm just not that far along yet. :)
20:15.36pbdI dunno about your phone company- but mine charges extra for three way calls.
20:15.39*** join/#asterisk Gnurdux (n=gnurdux@69.251.241.119)
20:15.49Gnurduxi had to change my dial rules
20:15.53FuriousGeorgefugitivo: that looks like exactly what i need but no distributors in the US
20:15.59Uther_Pmcf3782: btw, if it behaves anyway like it did on my provider, you might need to jackup the volume
20:16.01Gnurduxis there a way to make it forward ALL calls
20:16.14Uther_PGnurdux:    exten -
20:16.16Uther_Perr
20:16.27Gnurduxin dial patterns
20:16.29pbdexten => s,1,DIAL(FWD/#)
20:16.29Uther_Pexten => _X.,1,goto
20:16.30mcf3782BS charges me $.25 each time I use the 3-way-on-demand feature.
20:16.39fugitivoFuriousGeorge: I had a meeting with them last week, i can give you a contact if you want
20:17.30pbdAhh, BellSouth.  There's a happy company.. sitting out there next to the pond, ignoring everyone. :)(
20:17.41mcf3782pretty much
20:18.04mcf3782Anyone who isn't a BS customer care to guess what they want per-month for CallerID?
20:18.15mcf3782It's sad sad sad
20:18.29mcf3782$9.95
20:18.52Gnurduxis there a wildcard dial pattern?
20:19.27pbdGnurdux: 's' matches everything (default), or _X. ,which matches everything at least one digit long.
20:19.44Gnurduxwith the quotes or witout?
20:19.54pbdWithout.
20:20.10pbdexten => s,1,app_whatever
20:20.30mcf3782If I felt like paying that extortionist rate; then any time mom or dad called me; they'd just get routed over to the cell phone automagically without them having to do anything except just wait.  But I just refuse to pay over $10/month (by the time you add taxes) for the ability to get CID data.
20:20.54pbdmcf: So port your number to Vonage or some such.
20:21.15pbd$25/month flat, all services included.
20:21.18*** part/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net)
20:21.21Gnurduxjust s?
20:21.35pbd~voip-info
20:21.36jboti guess voip-info is the Voice Over IP wiki.  It is a community resource which will answer all of your questions, from Asterisk to ZTDummy.  You can find it over at http://www.voip-info.org - well worth bookmarking
20:21.53pbdThanks, jbot.
20:22.13Gnurduxso i should use that?
20:22.30mcf3782can't without moving my Internet connection from BellSouth.net to something else. I have a BellSouth.net DSL Internet connection. They won't sell me DSL without me having a voice line with them.
20:22.52pbdGnurdux: There's a million ways to write a dialplan.  My preferred method is the s extension- but it's your dialplan.
20:23.13Gnurduxi just want to have 1 line that forwards EVERYTHING
20:23.42pbdmcf: So get an 'emergency use only' line from them, with DSL on it.  If you push them hard enough, they may just drop your regular rates- especially if you tell them.. you know, I just got this cable modem offered in my area, and Vonage is looking fine..
20:23.42h3x0ri wish zaptel worked on freebsd 6
20:23.58Gnurduxtheres a freebsd 6?
20:24.03Gnurduxis it beta?
20:24.06h3x0rdude
20:24.08h3x0rthey are working on 7 now
20:24.25Gnurduxwait
20:24.29pbdmcf: That's the way SBC is working, in most cases.
20:24.29h3x0r6 is on beta #5 but its pretty damn close to release
20:24.35Gnurduxfreebsd 6 is stable?
20:24.39Gnurduxsee i use linux
20:24.39h3x0r7 is current for some time now
20:24.45*** join/#asterisk Uberbot (n=Uberbot@69.252.219.76)
20:24.46Gnurduxdont really follow the BSD world
20:24.59h3x0rwell their main web page is somewhat misleading
20:25.05h3x0ri mean yeah it says 5.whatever is stable
20:25.16h3x0rbut new hardware dosent fucking work with 5
20:25.23Gnurduxhehe
20:25.26mcf3782pbd - tried that already. :)  BS won't attach DSL to an "emergency use" class voice line.  Not "can't".. "won't".. they wouldn't make as much money that way. ;)
20:25.47pbdmcf: So go cable modem, and chuck BS into the wind. :)
20:25.48h3x0rlike my supermicro server's ethernet controller and sata ich7r controller
20:25.57h3x0rIt aint like linux dosen't have the same problem
20:26.08h3x0ryou can't time warp into the future to write device drivers for stuff that dosen't exist yet
20:26.21wunderkinmcf3782 what does changing internet providers have to do with using voip?
20:26.23Gnurduxyes i know
20:26.31h3x0ryou can backport to older kernels and stick it in the stable branch
20:26.35mcf3782I probably eventually will.  The cable provider here is finally getting their act together.
20:26.35Gnurduxbut i dont have supernew hardware either
20:26.37h3x0rbut that defeats the purpose of the stable branch
20:27.01h3x0rand that is 10x easier to do in *bsd than linux
20:27.04pbdHey, man- this is the telephony world.  We don't run anything that hasn't been in production for 2+ years.  And who needs sata for an Asterisk box anyway?
20:27.07h3x0rbecause theres one cvs respository for userland and kernel
20:27.26h3x0rUhm, who uses EIDE on a Asterisk box
20:27.30Kattywho wants to proof read my resume? :P
20:27.45Gnurduxwho can help me?
20:27.49h3x0rSATA deals with hard drive failures better, in conjunction with software raid 1
20:27.55pbdKatty: Having a good day, are we?
20:27.58h3x0ryou unplug a drive you ran atacontrol on with sata
20:28.01Kattypbd: mrow?
20:28.02h3x0rand the system keeps running
20:28.08h3x0ryou unplug a eide drive and the system dies.
20:28.34h3x0rwell CF is better
20:28.41h3x0rbut what about voicemail storage
20:28.55h3x0rand CDRs unless you are going to send those to another box.. but then again you STILL need a raid for that
20:28.55pbdKatty: You're putting out your resume- that's always a good sign.
20:28.58h3x0rsomewhere
20:28.59*** join/#asterisk JerJer[mobile] (n=jj@dsl001-136-136.lax1.dsl.speakeasy.net)
20:29.03Kattypbd: uh, no.
20:29.11h3x0rI am using 3ware controllers on database servers
20:29.16Kattypbd: i just want someone i know to proof read it for me
20:29.17h3x0rbut that would get really expensive if i put it on everything
20:29.28pbdh3x0r: Voicemail?  I use paper tape.  It stands the test of time.
20:30.06pbdKatty: And you know us here?  Boy, I feel honored. ;-)
20:30.06mcf3782Wonder how many people still know what that is? :)
20:30.16Kattypbd: i don't know you
20:30.17*** part/#asterisk Gnurdux (n=gnurdux@69.251.241.119)
20:30.27Kattypbd: i might let twisted or anthm proof read it for me though.
20:30.53pbdAhem.  So you asked in open forum who wants to read a resume, and now you're being picky.
20:31.00*** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka)
20:31.02Kattywell, duh
20:31.22Kattypbd: you've insaned. kthxbi
20:31.35pbdHrm.  Considering that I've been hiring people for companies for the last 10 years or so, I might have offered.  But now, fergeddaboudit.
20:32.05pbdFickle.  But I'd have guessed that from your nick.
20:32.15pbd;-)
20:32.53Kattyk
20:33.06Kattytwisted[asteria]: find me when you're done at that conference, kthx (=
20:33.50pbdKatty: I do have one question for you, that I started to ask you about a month ago, completely unrelated to Asterisk, but related to Linux, Samba, and 'click to print' under Windows.
20:34.19pbdI think you said you got it working.  I've had my MS and Linux guys working on it for weeks now, with only minimal success.  Did you say you had it working?
20:34.58KattyHrm. Considering that I've been using linux and samba for years now, I might have offered. But now, fergeddabouit.
20:35.20KattyFickle. But I'd have guessed that from your attitude.
20:35.21pbdYeah, yeah.  Same answer you started to give last month.  Figures. :)
20:35.32arp2get a room you two
20:35.41Kattyarp2: oh but I /do/ so love mocking people.
20:35.48Kattyarp2: and I shall, until they grow up.
20:35.53FuriousGeorgepbd: so i just checked a company called doorbellfon, and theirs hook up to "a spare trunk" so i assume that would be an fxo or something like that
20:36.16Kattypbd: why don't you /hire/ someone to fix that little issue of yours, hmm?
20:36.22*** join/#asterisk fulgas (n=fulgas@a81-84-116-219.cpe.netcabo.pt)
20:36.32cornelo ppl
20:36.45FuriousGeorgeits seems like there are two flavors of door phone, the one button kind which wants to go to an fxo, and the kind with a full dialpad which wants an FXS
20:37.11pbdKatty: I have, thanks. Unfortunately, the best answers in the world are 'it works sometimes'.  Then again, if I know someone who has gotten it to work successfully, I might even ask them. :)
20:37.36enderpbd: where is the 'click to print' dialog at?
20:37.50SeyrAnyone know why I cannot get MWI when using Realtime with rtcachefriends=yes?
20:38.00pbdFurious: Makes sense.  Maybe you can get someone to ship you a loaner?
20:38.05*** join/#asterisk e3g (i=ee@u15157627.onlinehome-server.com)
20:38.19pbdender:  Under file, run "//servername/printername"
20:38.22mcf3782'it works sometimes'.. isn't that pretty much a given with all things Microsoft? ;)
20:38.35e3ghi
20:38.37cornei have just tried to configure amp, but i made a mistake somewhere. amp doesn't show the status of asterisk(if a phone is ringing or call is active), but correctly indicates the extentions and trunks
20:38.49pbdThe idea is that a server can have all the drivers on it, and by installing the printer off the server, the drivers get sent to you, set up, and work.
20:39.03e3gcan we setup call back system with asterisk ?
20:39.19enderpbd: oh I see.  We don't share printers via samba.  We use cups so that Windows systems print to them via IPP
20:39.25pbdUnder samba- it's not so simple.  We've gotten it to send the drivers down, but they don't work in the end.  Frustrating, and barely documented.  Or worse- documents that say they work are either out of date or untested.
20:40.00pbdender: We can do that too- problem is, we print in Japanese, which the CUPS drivers need additional drivers for, which aren't available or don't work well.
20:40.06enderah
20:40.10cornecan you setup callbarring with asterisk?
20:40.27enderanybody know if it is possible to do handset 'paging' w/ *?  Something we used on our old phone system.
20:41.02vp7e3g: It's possible :)
20:41.15fiber0ptiender: if the phones you are using have an auto answer functionality
20:41.21e3gvp7: then what took you so long to answer me ?  ;)
20:41.23pbdender: Also, we have some copiers with a boatload of options, which need options that aren't supported on the stock CUPS driver.
20:41.39e3gvp7 : how ???? how asterisk can do automatic dialing?
20:41.40fiber0ptiender: otherwise you'd need a separate paging system
20:41.48vp7e3g: I'm newbie here, so was waiting for answer from GURU :)
20:42.02pbdender: Define handset paging.
20:42.06enderpbd: er... I don't follow.  Cups uses foomatic ppd files.  Samba would too.  how do you get around that?
20:42.34e3gvp7:  when all gurus Failed ...... newbie starts from there :)
20:42.35cornei have just tried to configure amp, but i made a mistake somewhere. amp doesn't show the status of asterisk(if a phone is ringing or call is active), but correctly indicates the extentions and trunks
20:42.43enderpbd: in our old system, we could pick up the phone, hit 'page' and select handsets and then speak.  All ahndsets would broadcast.
20:42.49pbdender:  If the stock ppds worked with the copier, it would print- but the options dialogs aren't there for all the other stuff.
20:43.02pbdender: Ahh.  Yes, it does support it, but in a backhanded sort of way.
20:43.19enderpbd: right, how does Samba get around that?  Also, when using IPP, you install a windows side driver for the printer, and the CUPS IPP is just the URI to the printer.
20:43.23e3gvp7 : shoot!!! your suggestion
20:43.29vp7e3g: You can create a file with calling parameters & asterisk will call where you want within several seconds.
20:43.37pbdThe common method is to open up a dynamic meetme bridge, then add in auto-answer extensions from all the phones you want to page to, (make sure they're added on mute), talk, then break apart the bridge.
20:44.01e3ghow asterisk can call without any event?
20:44.12pbdender: Theoretically, you don't need the foomatic stuff- you can do the direct rpcclient calls to add the entries into the samba tdbs, so it *looks* like a real Windows print queue.
20:44.14*** join/#asterisk nomazda (i=nyyankee@user-0c6tnqf.cable.mindspring.com)
20:44.25cornee3g, does e3g stands for egg? egg in you face!!
20:44.46e3gdont you know e3g?
20:44.55pbdOnce it looks like a real print queue, the driver is installed locally, and the file is printed up to the queue, which is defined by samba as pointing to a cups printer.. that's about as deep as I get with it personally.
20:45.17e3gvp7: ok I get the info in the file......now ??? how asterisk can do automatically DIAL() ?
20:45.25vp7e3g: You receive call. Came to conclusion, that call back should be made. Create a file with required parameters. Asterisk will find this file and make a call.
20:45.38corneno, but i now super man
20:45.57enderpbd: yeah, seems like you've put your thought into it.  At that point though, you have to wonder, why spend so much effort trying to emulate a windows print server, instead of just installing a windows print server.  THere are lesser evil things.
20:46.02vp7e3g: ok, let's me try to find what you have to do. wait for a bit...
20:46.41e3gvp7: after saving file.....I dont know how to let the asterisk know to dial :) ...ok I wait
20:46.42pbdender: Basically, the problem is that I have 12 remote offices, each with a small fileserver, better suited to a linux box than to a fullblown windows server (less costly, more reliable, once developed).
20:46.49pbdFor the main office, we still do run it as windows.
20:47.09e3gcorne: I make MEN ...Super man, Spider man, would you like to be skunk man ???
20:47.32pbdAlthough eventually, we'll axe the Win2K3 network in favor of samba PDCs and LDAP.
20:47.36corneheheeheheheheheheheheheheehhee
20:47.40cornebla bla bla
20:47.50vp7e3g: you save file in a special directory that is scanned by asterisk for new events. so you just have to save it. i'll try to find a name of this feature & this directory
20:48.12SeyrIm using Realtime and have rtcachefriends=yes, but I do not get MWI on my 7960 when I leave voicemail. Anyone have any experience with Realtime that might have any suggestions on what to check?
20:48.53pbdNow that the channel is a little more active, I'll go for a repeat.. anyone out there handling caller-id inbound in Brazil?
20:49.15*** join/#asterisk pardove (n=pardove@195.146.47.201)
20:49.19e3gvp7: I really appreciate your efforts buddy
20:49.21vp7e3g: I've got :) Just look into http://www.voip-info.org/tiki-index.php?page=Asterisk+AGI
20:49.33enderpbd: yeah.
20:49.44cornecan you setup callbarring with asterisk?
20:49.44enderpbd: hrm, what about using those embedded 'print server' devices?
20:49.50pardovewhen * attempts to detect CallerID on Zap Channels
20:49.57pardovewhen * attempts to detect CallerID on Zap Channels?
20:50.07pbdender: Thought about it- but most of them don't do the automatic driver downloads.  At least, I haven't found one that does.
20:50.07*** part/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net)
20:50.14enderpbd: ah.
20:50.25vp7e3g: Feature is called "Asterisk auto-dial out" and docs are here: http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
20:50.31enderpbd: w/ 12 people or so, is it really that difficult to put up a doc on how to do the driver install?
20:50.39pbdI smell a lot of Microsoft shenanigans there.. printing seems very poorly documented.
20:50.40vp7e3g: Is this an answer to your question?
20:51.10e3gvp7: do you love your girl friend ??? :)
20:51.17e3gvp7: hey ....thankssssss!!!
20:51.22pbdNow, Katty.. does that explain to you a little better why 'just hire someone' isn't really the right answer here?
20:51.27e3gvp7: U got me there :)
20:51.56pbdMoney isn't the issue- its finding someone who can actually do it, and repeat it, in our environment.
20:51.57e3gvp7: thanks man .....I appreciate that ....
20:52.13vp7e3g: not at all :))
20:52.28pardovewhen * attempts to detect CallerID on PSTN lines?
20:53.03Kattypbd: i haven't been paying attention
20:53.05*** join/#asterisk [hC] (n=hardcore@dsl001-136-136.lax1.dsl.speakeasy.net)
20:53.22pbdpardove: I'm sorry.. I think I missed the first part of your question, but perked up over caller-ID..
20:53.29Kattypbd: but that generally happens when you have an IT department to run
20:53.49Kattypbd: i hope you find your answer.
20:53.54cornevp7, can you setup callbarring with asterisk?
20:54.11pbdKatty: You are in a good mood today.  My group hasn't been bugging me- they're all waiting for the new laptops I ordered (133 of them) to arrive tomorrow.
20:54.25*** join/#asterisk Veto_laptop (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com)
20:54.28KattyI'm in a good mood every day.
20:55.20*** join/#asterisk Emrah (n=user@adslgva0491.worldcom.ch)
20:55.29pardovepbd: it seems that * attempts to detect caller-id just after the 1st ring. but i have a case that Telco sends the caller-id before the 1st ring, so * can't detect it :-(
20:55.37*** join/#asterisk Ash (i=aaron@outofband.org)
20:55.38SeyrAnyone know why I cannot get MWI on a 7960 using Realtime with rtcachefriends=yes?
20:55.39pbdKatty: Now, am I wasting air here, or can I now apologize for (I think) somehow ticking you off over your resume, and perhaps find out if you at all know of a resource, or have gotten click to print working under samba before?
20:55.41Ashyowsa
20:55.49EmrahHello!
20:55.51Ashhello there
20:56.08pbdpardove: ahh.. now I have to remember.. I've read on that one before.
20:56.19vp7corne: Explain plz what does "callbarring" means. I don't know such an english word (english is not my native)
20:56.20pbdpardove: Are you in the US?
20:56.27pardovepbd:: any solution?
20:56.30*** part/#asterisk obsidian-studios (n=obsidian@c-66-177-188-197.hsd1.fl.comcast.net)
20:56.34pbdpbd: And you're using POTS lines, not PRI, correct?
20:56.48pardovepbd: yes i'm using POTS lines
20:56.49Kattypbd: Yes, you did tick me off.
20:56.59Kattypbd: To be honest, I'm still rather ticked off about it too.
20:57.05Kattypbd: You're very presumptious.
20:57.10EmrahI've made some searches on the Internet, particularly in the Asterisks mailinglists, but I did not find any answer for my problem. I'm having a problem, perhaps with the motherboard, perhaps with the 4xdigium card... May some have have a look at this for me?
20:57.23AshHas anybody had an issue with their TDM400P not being recognized by asterisk?
20:57.24Kattypbd: And to be frank, you're wasting your breath with me.
20:57.26EmrahI will publish the error message in pastebin
20:57.27pbdKatty: Actually, I'm not- just busy.  But I'm sorry if it offended you.
20:57.40Kattypbd: let's hope it doesn't happen again, eh? (=
20:57.45syzygyBSDAsh: do you have the modules loaded?
20:57.51Ashztcfg -vvv shows the channels as being configured just fine, and there are no errors in the modprobe
20:58.01ender30~ IP-301s showed up today, so now I have to configure them alll.
20:58.06Uther_P*rolls eyes*
20:58.20pardovepbd: do you have any solution?
20:58.21Uther_Pever the drama major
20:58.23enderpeople see the phones w/ their names on it, and want them NOW.
20:58.25AshsyzygyBSD: I get this error:  WARNING[4607]: chan_zap.c:890 zt_open: Unable to specify channel 1: No such device
20:58.29KattyUther_P: naturally, i'm female!
20:58.30Ashand a few after that
20:58.36*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
20:58.37pbdpardove: You said POTS.. but are you in the US, or are we talking a non-US standard CID spec?
20:58.37KattyUther_P: it's /normal/ for me (=
20:58.44Uther_PI know man that dont do that, heh
20:58.45Ashit's a TDM400P with two FXO modules on it
20:58.47Uther_Perr many
20:59.16harryvvtraditionally what cable type is T-1 or T-3 normally?
20:59.17pardovepbd: i'm not in US i have this issue in germany
20:59.27Ashharryvv: T1's are usually RJ45
20:59.29pbdKatty: Erm, well- seems I could accuse you of also being at least one who's quick to judge- but hey.  If you don't know the answer, or aren't willing to help, so be it. :)
20:59.34*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
20:59.34cornevp7,call barring is when you want to restrict an extention to only be able to dail local numbers instead of national or international numbers
20:59.40harryvvash, rj45 is the connector.
20:59.54Ashharryvv: well yeah. I thought that's what you mean, sorry :)
20:59.54harryvvfor cat5 cable
21:00.08enderharryvv: not just cat5
21:00.17Ashall my T1's use UTP with RJ45 ends from where they come out of tthe demarc
21:00.22pbdpardove: Ok, then.  Now that I'm thinking about it, I think the issue I read up on was the converse- because they didn't have a wait() int the dialplan, it was not picking up the caller id, which came in on ring two.
21:00.26enderharryvv: rj45 is a connector type.  Any number of cables could be terminated into that connector type.
21:00.37harryvvI suspect its serial based cable such as those used to hook to the DTE DCE connectors on cisco routers.
21:00.40harryvvBut dont know.
21:00.58Ashharryvv: you can use a plain old ethernet cable for T1, dude.
21:01.03harryvvI see
21:01.24vp7corne: you want not to let some of your internal users to make international calls? it's easy - just set to them context that have no rules for international dialing :)
21:01.30pbdpardove: Your case is that it's coming in too early.. which is weird.  I'd more suspect a protocol incompatibility.  What is your PSTN interface? Digium card?
21:01.43*** join/#asterisk twisted|astricon (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted)
21:01.43*** mode/#asterisk [+o twisted|astricon] by ChanServ
21:02.06cornevp7, thanx
21:02.07pardovepdb: i'm using TDM400P
21:02.57pardovepdb: but a caller-id detection device catchs the caller-id on the this line
21:03.03EmrahMay someone have a look at this? http://pastebin.ca/25423
21:03.22syzygyBSDI think that is one of the cards I have in my system, also have a T100P and a TE100P, they all work fine
21:05.06pbdpardove: Right, but it's possible the digium card isn't configured to read your signalling method.
21:05.06pbd(or the tdm400 doesn't have the ability to read it)
21:05.07*** join/#asterisk folsson (n=filip@lund-meje-sr0-vl101-249.perspektivbredband.net)
21:05.46pardovepdb: i've also tested it with TE110P and TA750. no success!
21:06.17EmrahMay someone have a look at this? http://pastebin.ca/25423
21:06.34pardovepdb: and also tested with all cidsignallings
21:06.35*** join/#asterisk [hC] (n=hardcore@dsl001-136-136.lax1.dsl.speakeasy.net)
21:06.54pbdpardove: That's indicative.  Did the 110P show anything for caller ID when you debugged the span?
21:06.59*** join/#asterisk n0where (n=kc@dsl001-136-136.lax1.dsl.speakeasy.net)
21:07.33pbdIf the 750 wasn't able to decode it, it's possible that your telco is using a signalling method that's too unique for the cards to handle.
21:08.03*** join/#asterisk fifer (n=sirfifer@207.202.227.161)
21:08.24pardovepdb: i didn't do that. as i know channelbank just passthroughs everything, am i right?
21:09.25*** join/#asterisk zotz (n=zotz@24.231.36.100)
21:09.56pbdPardove: Well.. maybe.  Keep in mind, the 750 is going to hand you ISDN PRI, which communicates all the stuff digitally- it uses IE messages to pass things like caller-id.  An analog POTS line doesn't work that way.. its more like a cheapo modem sending the signal in a frequency range that's hard to hear, between the rings.  So the 750 has to catch it one way, and re-encode it to the other.
21:09.57Samoiedhello all
21:10.18SamoiedI have a problem with authentication i asterisk
21:10.41EmrahAnyone has an idea to help me finding where is the problem comming from?
21:10.54SamoiedI want to use one userid <From> and other authid <Authorization>
21:11.13denondoes anyone know if there's a changelog or other details on adtran 750 firmware updates?
21:11.16Samoiedbut   the asterisk always use From header, not auth
21:11.17denonI thought there used to be
21:11.19pbdEmrah: It can come from all sorts of places- mostly bugs in Asterisk.
21:11.53EmrahBut the card has just crashed yesterday and now it's really strange
21:12.00*** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
21:12.13pbdEmrah:  I see it occasionally on my h323 channels, if the gatekeeper wigs out in *just* the right way.
21:12.15EmrahIs there anyone to catch how is the connection between the card and Asterisk?
21:12.23Emrahis there any way*
21:12.49AshSo has anybody had an issue with the TDM400 channels not showing up?
21:12.55Ashlog output is here: http://pastebin.com/392749
21:13.00pbdEmrah: I'm not sure what you mean by 'catch the connection'.  You can look for interrupt conflicts, etc, by doing a 'cat /proc/interrupts'.. or check your dmesg log.
21:13.02Ashthis is with the kernel modules loaded
21:13.15pardovepdb: when i connect the caller-id detector to the line, it shows the caller-id just before the first ring. does it give any idea? does * or CB try to detect CID before 1st ring? if CID detector shows cid so telco CID sig. is known, isn't it?
21:13.56pbdpardove: It's known to the makers of your CID detector device- but I'll bet that a similar device from the US won't catch anything.
21:13.59pardovepdb: what is the difference between cid detector and *?
21:14.08pbdpardove: Signalling, mostly.
21:14.53pbdThere's a lot of different ways to pass CID.. timing, protocol, frequency ranges, etc.  And there's a few different standards.  In the US, we use 'bellcore' standards.. but that's not true everywhere.
21:14.56pardoveso we can add more cid sig. standards to *? is there any ref. for all known cid standards?
21:15.12EmrahThanks a lot pbd , I will try
21:15.27*** join/#asterisk nagl (n=nagl@213.235.241.6)
21:15.56pbdpardove: Big maybe there.  Depends on the card drivers as to whether it will catch it at all.  You'd be best off talking to Digium or Sangnoma directly on that one.
21:16.32*** part/#asterisk sgorilla (n=tlp@cpe-24-160-119-179.houston.res.rr.com)
21:16.51SeyrAnyone know why I cannot get MWI on a 7960 using Realtime with rtcachefriends=yes?
21:17.33pbdPardove: the sipura-spa3000 I'm playing with now supports 8 different CID standards.. some of which send before 1st ring.
21:17.53pbdI'd be surprised if the Digium hardware didn't support the same standards.. but not *Very* surprised.
21:17.54pardovepbd: to me it seems that it is a generic problem to chan_zap or ... because i have the same prblem with TDM400P, TE110P TE405P and A104...
21:18.21pbdpardove: Not necessarily chan_zap- it depends on what the card passes up to it.
21:18.24*** join/#asterisk pifiu (n=myassisb@208.205.181.170)
21:19.50pardovepbd: do you have any ref./link to CID signalling types?
21:19.52pbdpardove: Check this page out, it might help you. http://artofhacking.com/files/callerid/CLI_FAQ.HTM
21:21.20*** join/#asterisk oej (n=Olle@dsl001-136-136.lax1.dsl.speakeasy.net)
21:21.38pardovepbd: thanks
21:24.16*** join/#asterisk bongfrog (n=winston@dsl001-136-136.lax1.dsl.speakeasy.net)
21:24.29HiltonTlove the nick  :)
21:25.38*** join/#asterisk Trionnis (i=fsn@12-215-249-177.client.mchsi.com)
21:26.06Trionnisso does nufone ever answer their sales email ?
21:26.12Trionnis:)
21:27.34pardovepdb: just to make sure. does * really support these 3 cid sigs: bel202, v23, dtmf?
21:28.09FuriousGeorgeso how many telephones worth of voltage do these ATA devices support?
21:28.43FuriousGeorgepbd: i went with the viking k-1700, and a c-1000 to unlock the doors from extensions
21:30.01*** join/#asterisk Veto_laptop (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com)
21:30.40*** join/#asterisk bjohnson (n=bjohnson@i216-58-62-82.cybersurf.com)
21:31.23Kattyanyone want to quote me a server?
21:31.29Ash"server"
21:31.33Kattya /windows/ server
21:31.37Katty;)
21:31.39syzygyBSDlol..
21:31.46denonsure, 1 nice quad opteron for $25k
21:31.51syzygyBSDKatty: what is it going to be used for?
21:31.59KattysyzygyBSD: well it's windows.
21:32.02*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)
21:32.02KattysyzygyBSD: so obviously a paperweight.
21:32.05Trionnisnot a lot then?
21:32.08syzygyBSDhow about you save your time and just burn the money now
21:32.11Trionnisdang, beat me to it
21:32.17Katty;)
21:32.22Kattyi'll take that as a no
21:32.25Kattykthxbi.
21:32.38syzygyBSDlol.. i run a couple of them, just prefer linux
21:33.32FuriousGeorgedualcore athlon64, 1 gig of ram, 4 x 250 gb sata 1,300 or so
21:33.49FuriousGeorgethats with a highend mb
21:33.56enderKatty: what specs?
21:34.15Kattyender: pretty and black.
21:34.15enderKatty: Pogo Linux makes great hardware and they can ship/support Windows too.  We bought our * servers from Pogo.  Very satisfied.
21:34.22enderKatty: and they're black.
21:34.25denonI wouldnt recommend an ath64 on a server ..
21:34.28denonthat's what opterons are for
21:34.29denonof xeons
21:34.32Kattyender: hot.
21:34.58Kattyi'm sure i'll build a xeon server.
21:35.05denonwe sell a ton of em
21:35.08Kattyi was just teasing about the quote. heh
21:35.09enderdenon: dual-core Opterons are quite nice.
21:35.10Assidhow much are opterons going for now adays
21:35.10denonway more than opteron
21:35.17fiferWe have been going with Dell 1850 recmount 1u servers. Under $2k
21:35.27Katty:<
21:35.33FuriousGeorgedenon: i just happened to have that quote for a network cam server
21:35.35Kattythose are noisy.
21:35.52syzygyBSDget water cooling
21:35.54Kattyevery single 1u i see sounds like an airplane.
21:35.54denonFuriousGeorge: what kinda gear?
21:36.11enderKatty: hehe, our 1us aren't bad.  But they're just P4 Prescotts w/ 2x Sata disks.
21:36.12denonKatty: not so much like a plane .. but more like a jet engine :)
21:36.29FuriousGeorgedenon: snc-df70 sony cameras
21:36.38SeyrAnyone know why I cannot get MWI on a 7960 using Realtime with rtcachefriends=yes?
21:36.40denonFuriousGeorge: eth-attached?
21:36.43FuriousGeorge1,800 with dual 17" lcd display
21:36.50FuriousGeorgedenon: yeah all PoE
21:36.57denonFuriousGeorge: 1800 cams?
21:36.57FuriousGeorgeno electrician required
21:36.58*** join/#asterisk loick (n=loic@APuteaux-151-1-46-19.w82-124.abo.wanadoo.fr)
21:37.00FuriousGeorgeno
21:37.15FuriousGeorgecams cost me 850 each, the server for 8 was what i quoted before
21:37.20denonah
21:37.23denonwhat software?
21:37.34FuriousGeorgeadd 1400 for sony realshot
21:37.48denonbeen looking for good applications, open source would be nice, to handle commercial security cam installs
21:37.52Assidwhats better? opteron 242 or   athlon 64 3400
21:38.02denonAssid: depends what you're doing with it
21:38.05Kattywhat's better?
21:38.07Kattya nice salad.
21:38.10Kattyand hugs.
21:38.20denondepends how cute she is ..
21:38.23denonthe one you're hugging
21:38.24FuriousGeorgedenon: there are two that ive found that work with sony in mpeg4, sony software (of course) and d3data, which is actualy more $
21:38.30Kattydenon: heh.
21:38.50FuriousGeorgeKatty: whats ur pretty black server gonna serv?
21:38.56Assiddenon: video encoding.. gaming.. everyday shit..
21:38.56KattyFuriousGeorge: hot chocolate, obviously.
21:38.57denonFuriousGeorge: i dont really care what transport/codec/etc they use .. usb cams would be nice, but not necessary .. and not practical anyway I guess
21:39.06denonAssid: think I'd go opteron then
21:39.09KattyFuriousGeorge: it's just gonna be a file server
21:39.17denonAssid: though the ath has some advantages .. you wont need 64 anyway
21:39.21Assidhow do they differ?
21:39.35FuriousGeorgedenon: i dont think theres a such thing as commercial usb camera security systems
21:39.38denoncache and instructions
21:39.43denonFuriousGeorge: nope, not really
21:40.09KattyFuriousGeorge: with 2 300 gig hds and a raid1
21:40.10denonFuriousGeorge: sony's app any good? it comes with the cams?
21:40.11FuriousGeorgedenon: mpeg4 is good b/c you can get good quality at 2kB/s which makes recording for days (i.e. commercial application) feasable on less space
21:40.32*** part/#asterisk Seyr (n=Seyr@rrcs-24-227-133-226.sw.biz.rr.com)
21:40.33generalhananyone know of some really good IVR software that i can use with linux ?
21:40.42FuriousGeorgeKatty: how much fileserving will it do at peak hours
21:40.44denongeneralhan: uh, asterisk?
21:40.50KattyFuriousGeorge: haha
21:40.53shido6anyone know how to install Universal.pm ?
21:40.54KattyFuriousGeorge: hahahahaha
21:40.57KattyFuriousGeorge: peak?
21:40.57generalhandennon: LOL ! lemme explain
21:41.00KattyFuriousGeorge: 10 people!
21:41.01FuriousGeorgedenon: sony sw that comes with a camera just drops file on an ftp
21:41.15FuriousGeorge10 people downloading 10 gigs or 10 megs?
21:41.23Kattymore like 5 megs.
21:41.23denonFuriousGeorge: and you can easily view lots of cameras, navigate archive footages, etc right from an app?
21:41.33HiltonTthen any box will do - its way small
21:41.42*** part/#asterisk Trionnis (i=fsn@12-215-249-177.client.mchsi.com)
21:41.46generalhanmy boss wants, instead of people leaving messages, wants to set up a VR system that will prompt the user for their name and phone number then log those responses to a database, or even just a spreadsheet
21:41.54FuriousGeorgedenon: for "easily manage multiple cameras" ur gonna want the $1400 real surveilance recording sw
21:41.54HiltonTcheap HP or Dell box, depends on what u prefer
21:42.03HiltonTor a whitebox (build it yourself) unit
21:42.15denonFuriousGeorge: from d3data? or from sony?
21:42.22FuriousGeorgedenon: other cameras are compatible with cheaper sw, but they dont do PTZ outside, with only PoE for less than a grand, if at all
21:42.34FuriousGeorgedenon: havent used either
21:42.46HiltonTI like the LOOK of the panasonic "HAL" camera, not seen it in action, tho
21:42.47denonFuriousGeorge: well, I mean which is the 1400
21:43.16FuriousGeorgeKatty: piii1ghz 256 mb of ram, on 100baseT network.  that might be overdoing it i dunno
21:43.19pardovejust to make sure. does * really have full support of these 3 cid sigs: bell202, v23, dtmf?
21:43.42HiltonTnot the 256MB - that's not enough for Windows
21:43.48FuriousGeorgesony=1400 for 8 camera licence.  d3data > 1700 for 10 licence (less is a 5 licence)
21:44.05pardovejust to make sure. does * really have full support of these 3 cid sigs: bell202, v23, dtmf?
21:44.10denonhmm .. gets pretty steeap for only 10
21:44.20denon-e
21:44.44FuriousGeorgedenon: if ur cameras are gonna be indoor maybe you can get cheaper cameras which are compatible with cheaper software
21:44.44denonboth can control robotics cams?
21:45.00*** join/#asterisk queuetue (n=queuetue@toronto-HSE-ppp4122670.sympatico.ca)
21:45.10FuriousGeorgedenon: i believe so, but the onboard httpd of the cam probably can too
21:45.18harryvvFuriousGeorge what ccd cameras do you run?
21:45.26denonFuriousGeorge: yeah, just idiot-proofing for endusers here ..
21:45.51FuriousGeorgeharryvv: i just quoted a guy for snc-df70 outdoor.  i think theyre .3 MP
21:45.58FuriousGeorgeremains to be seen if hes serious
21:46.01*** join/#asterisk syle2 (n=blag@unaffiliated/syle)
21:46.22denonFuriousGeorge: so do they both give a nice mult-cam interface for remote login?
21:46.30HiltonTFuriousGeorge; seen the Panasonic cams that look like HAL (2001)?  Any good?
21:46.42FuriousGeorgedenon: having never used/bought/seen either sw, i cant tell you much.
21:46.44harryvvOnce I buy a BTV cctv capture card going to interface my sony cctv to asterisk. if there is any motion in the backyard when im not here it will triger a script in asterisk to call me on my cell phone.
21:46.56denonFuriousGeorge: ah. . what'd you say you use? or nothing, just dump to disk?
21:47.06FuriousGeorgei can tell you ive heard sony support is terrible, and u'd think being 3rd party and so steep, support would be d3data's angle
21:47.25HiltonTI don't like Sony's non-Professional kit
21:47.37harryvvFur, this is a clone sony brand. Runs about 100 dollars for the camera. I dont worry about it to much.
21:47.38HiltonTand their Pro gear is not in the same ballpark as what we're discussing here
21:47.46HiltonTI mean pro as in broadcast quality
21:47.46*** join/#asterisk syle2 (n=blag@unaffiliated/syle)
21:47.46denon"scalable (i.e., a companie with locations " .. it'd be nice to buy from a company that can spell company...
21:48.06FuriousGeorgeharryvv: i was asking someone about streaming the rtp from the camera to * and using libraries to make it oh323 or something
21:48.13FuriousGeorgenot that i could do that
21:48.22HiltonTwe use a lot of Sony Pro gear in TV and it is good.  Their consumer gear is not what it used to be 7+ year back
21:48.28FuriousGeorgeHiltonT: never heard of those cams, btw
21:48.35HiltonTI'll find a link
21:48.36harryvvi seee
21:48.45FuriousGeorgeHiltonT: or at least they werent what i was looking for when i was looking
21:48.48denonFuriousGeorge: so do you have suggestions on a cheaper, but relatively secure internal cam?
21:49.01FuriousGeorgesecure as in vandleproof?
21:49.06harryvvcam that fits inside the pc?
21:49.06FuriousGeorgevandal proof
21:49.16denonwell .. in a school, but not like inner-city chicago
21:49.16FuriousGeorgelol
21:49.24denonwell, in lots of schools actually ..
21:49.25denonbut yeah
21:49.25harryvvdenon secure cam?
21:49.32FuriousGeorgewell lit area?
21:49.38denonrelatively
21:49.41denonhalls and such
21:49.53*** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com)
21:49.56harryvvi think its sad to put cams in highschools.
21:49.57iCEBrkr${UNIQUEID:1:-3}
21:50.06harryvvWe never had any
21:50.06FuriousGeorgei dunno, if ur lighting is good you could use the dlink 5300
21:50.07iCEBrkrShouldn't that grab all but the last 3 characters?
21:50.23Corydon76-homeACK, no... Do not use a negative length
21:50.30denonharryvv: yeah .. they're gettin a grant for it .. and they're hoping it'll keep kids from pullin crap .. little bit of an intimidation factor
21:50.38Corydon76-homeNegative offset is fine, though
21:50.48Corydon76-home${UNIQUEID:-3:3}
21:50.50harryvvdenon, what do you mean by pulling crap
21:50.50*** join/#asterisk KranZ (n=user@216.16.193.200)
21:50.52HiltonTa couple here, http://www.smarthome.com/971302.html but not the one I was after...
21:51.01FuriousGeorgedenon: what about dome cameras, arent they all drop ceiling
21:51.03FuriousGeorges
21:51.12denonharryvv: pranks, breaking into lockers .. noogying the little guy .. whatever, I dunno
21:51.22Supaplexthey have crap on a string now? what's next, wagging tails on dogs?
21:51.25denonFuriousGeorge: yeah, that could be a good option
21:51.26KranZpoop
21:51.27Corydon76-homeiCEBrkr: why are you trying to break off the last 3 chars of UNIQUEID, anyway?
21:51.39iCEBrkrCorydon76-home: cuz I don't need the decimal place
21:51.43KranZcus he's the ice breaker
21:51.45harryvvdenon i can see for breaking into lockers. But what are the locks for?
21:51.48Corydon76-homeiCEBrkr: that's not a decimal
21:51.51FuriousGeorgedenon: isnt  the sony snc-df40 an indoor dome camera with PoE
21:51.52*** join/#asterisk brent21 (n=Brent21@70.88.149.221)
21:51.59Corydon76-homeiCEBrkr: that's how it goes unique
21:52.09denonharryvv: Im a tech guy, I dont care what they want to use the tech for at their schools..
21:52.13iCEBrkrexten => setup,2,SetVar(sequence_id=${UNIQUEID:1:-3})
21:52.17iCEBrkrThat's what I need to do.
21:52.18denonFuriousGeorge: no clue
21:52.19Corydon76-homeThey are two numbers, together they are unique
21:52.21iCEBrkrI don't need that shit...
21:52.22*** part/#asterisk brent21 (n=Brent21@70.88.149.221)
21:52.31Corydon76-homeEither one is NOT guaranteed unique
21:52.38iCEBrkrah
21:52.42harryvvdenon, same here. been a tech guy since age 6 :)
21:52.48Corydon76-homeiCEBrkr: the first one is just the same as ${EPOCH}
21:52.49FuriousGeorgedenon: if it is PoE you dont need to do electrical work, thats a big savings some times
21:52.55HiltonToh, yeah
21:52.59iCEBrkrCorydon76-home: I was wondering why that looked familiar.
21:53.03denonFuriousGeorge: looks like a nice cam
21:53.18Corydon76-homeiCEBrkr: it's EPOCH and an incrementing counter
21:53.20HiltonTeven if not PoE, you can use those power injectors and break it out at the camera end
21:53.29denonFuriousGeorge: dosnt say anything about PiE ..
21:53.32Corydon76-homeiCEBrkr: together, they're unique
21:53.38denonPoE
21:53.48KranZwhat is a compression codec which is cpu friendly
21:53.52*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)
21:53.59iCEBrkrCorydon76-home: agreed. I understand.
21:54.17iCEBrkrBut for what I'm using it for ( for now ) I kinda need to get ride of th .
21:54.21Corydon76-homeiCEBrkr: however, if you really, really want to break off everything after the period, use CUT()
21:54.21iCEBrkrerr rid
21:54.23denonKranZ: ulaw
21:54.23SupaplexKranZ: a dedicated codec expansion card ;)
21:54.43KranZdenon: < 64k
21:54.46Corydon76-homei.e. ${CUT(UNIQUEID,.,1)}
21:54.54KranZSupaplex: can u even get those working with asterisk?
21:54.56denonKranZ: well-configured speex
21:54.59Corydon76-homebecause the second portion is variable length
21:55.02denonFuriousGeorge: Power requirements AC 24 V 50/60 Hz, DC 12 V,POE
21:55.14denonso guess it is
21:55.26SupaplexKranZ: no idea, but if I had a free one, I'm likely to try :)
21:55.39KranZword
21:55.40denonFuriousGeorge: cant seem to find pricing on em though
21:55.44*** join/#asterisk Dougnaka (n=Doug@66.236.77.194.ptr.us.xo.net)
21:55.49KranZdenon: on what
21:55.55denonsnc-df40
21:56.01*** join/#asterisk DanielArndt (n=DanielAr@reverse-82-141-57-74.dialin.kamp-dsl.de)
21:56.04*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)
21:56.09Supaplexwb lilo
21:56.21denonlilo's come to chew us out for using too much bandwidth again :)
21:56.26Corydon76-homeActually, both portions of UNIQUEID are variable length... the first portion has just already gotten to its maximum
21:56.45Supaplexla la la here's more bw down the drain. ;o)
21:57.01KranZdenon: well the price for the ceiling mount of a snc-df70n is $109
21:57.02denonFuriousGeorge: oh. looks like ~$700
21:57.12KranZso its gonna be pricey
21:57.42*** join/#asterisk [hC] (n=hardcore@dsl001-136-136.lax1.dsl.speakeasy.net)
21:58.08denondunno if that would fly price-wise
21:58.10KranZthe df70n is around $800
21:58.31*** join/#asterisk pifiu (n=myassisb@208.205.181.170)
21:58.34KranZgivin the higher model number, the df40 might cost less
21:58.37DanielArndthi, does anyone here own a German T-Com Eeumex 820 Lan ? and knows how to connect it with two ISDN (cologne) cards in NT mode, so that 4 lines can be used at onces ?
21:59.09denonprobably, df40 is fixed
21:59.52KranZdenon: there's a snc-df40n for $627
21:59.58FuriousGeorgedenon: theres a sony scn-df series pdf out there, says all the models are PoE
22:00.17denonFuriousGeorge: doesnt it seem like something like this should happen for sub-400 bucks?
22:00.30KranZhttp://froogle.google.com/froogle?q=snc-df40n&scoring=p&sa=N&start=50
22:00.31FuriousGeorgealso says the df40 doesnt have night vision, and is indoor
22:00.35denonsome of those axis ones are fairly cheap
22:00.47*** join/#asterisk Avero (n=no@dsl001-136-136.lax1.dsl.speakeasy.net)
22:01.13FuriousGeorgedenon: you gotta factor into the cost of the camera the fact that you dont need to submit plans to the city and get a licened electrician to do the wiring
22:01.27FuriousGeorgeand when theres a power outage one battery backup keeps everything going
22:01.30denonFuriousGeorge: yeah ..
22:01.32FuriousGeorgewhile you find a gas gnerator
22:01.39*** part/#asterisk Utah_Dave (n=boucha@0-1pool138-209.nas28.salt-lake-city1.ut.us.da.qwest.net)
22:01.51denonFuriousGeorge: these arent real high-security areas, so they dont really care much about the important things :)
22:02.13FuriousGeorgeyeah, but im tlaking about the bottom line too
22:02.21denonnod
22:02.25denonthey have a staff electrician I believe ..
22:02.32denonso he'd be the guy dragging cat5 too
22:02.34denonwouldnt matter much
22:02.38denonbut its still less hassle
22:02.52FuriousGeorgeyeah, he can go back to changing lightbulbs faster this way :)
22:02.56denonheheh
22:03.23denonso 1700 + (10*650) would get them a 10 install setup
22:03.32FuriousGeorgeplus you can tell the school: "look you not gonna sit here and watchem, just dumpem on disk and watchem if you need to
22:03.34denonplus a server
22:03.52*** join/#asterisk X-Rob (n=rob@dsl-202-173-151-24.qld.westnet.com.au)
22:03.52FuriousGeorgedenon: they need an ftpd to dump the files on
22:03.56denonFuriousGeorge: so with mpeg4, is it really very i/o intensive?
22:04.07denonftpd? figured they stream right to the software. .
22:04.40FuriousGeorgedenon: i think its just a matter of 13 kB/frame with jpg vs ~2kB with mpeg4
22:05.02*** join/#asterisk Damin_PDA (n=pocketir@19.sub-70-209-70.myvzw.com)
22:05.08Damin_PDAwerd...
22:05.12denonFuriousGeorge: yeah, but 10-20 cams doing that .. would I need to even consider array speeds?
22:05.29denondoesnt seem like that's much traffic
22:05.33FuriousGeorgedenon: the cameras are like little computers, they got firmware you log into thats free, that firmware has the ability to send ftp streams of video for free too.  its your job to catch it
22:05.59endercan * play aiff files?
22:06.02FuriousGeorgedenon: no, but if ur recording for security you may want redundancy.  liability is expensive
22:06.02denonFuriousGeorge: but I assume an app like we were talking about before lets you easily see realtime stuff quicker
22:06.22denonFuriousGeorge: yeah, offsite mirror maybe .. compress and shuffle
22:06.25GXTia striping array would be a plus
22:06.30GXTior full raid
22:06.37denon0+1 would be nice .. but ..
22:06.38FuriousGeorgedenon: exactamundo, control all the cameras, bring up archived video, monitor, from one interface
22:06.42denontheyd kick and scream
22:06.46GXTiyeah
22:06.47GXTibit pricey
22:06.52GXTijust a _bit_ ;p
22:07.00denonFuriousGeorge: so do these apps still hook ftp? or do they just do an ip stream
22:07.10FuriousGeorgeboth
22:07.11denonGXTi: well .. its not pricey if you want your whole array to be 40 meg ..
22:07.21GXTidenon: flash drive array ;)
22:07.28FuriousGeorgethe ftp is optional, their job is to ipstream
22:07.31denonFuriousGeorge: both? watching an ftp doesnt seem very realtime
22:07.33denonah ic
22:07.40FuriousGeorgeyou gotta connect something to it (video server)
22:07.42GXTidenon: you could tail the videofile on the ftp server
22:07.51FuriousGeorgeto record
22:08.01denonGXTi: oh yeah, that's real platform-independant :)
22:08.01FuriousGeorgeto view, the client is IE6
22:08.16FuriousGeorgedont know about ff/mozilla support
22:08.17GXTidenon: actually its more of a problem with the player software
22:08.24denonGXTi: nod
22:09.50FuriousGeorgebut of course, if ur paying 1400 for sw, it can view the cameras, too
22:14.22syle2anyone use SER?
22:14.33file[laptop]yes no maybe so
22:14.57syle2i been trying to figure out if it does cdr entries like cdr_mysql from mysql-addons
22:15.13syle2how you suppose to know duration of calls
22:15.17file[laptop]it doesne't.
22:15.27file[laptop]the accounting module records all the SIP messaging for the calls
22:15.34file[laptop]then you have to use outside stuff to construct that information
22:15.53syle2how do you get duration of call?
22:16.01file[laptop]you use outside stuff to figure it out
22:16.10syle2ie ?
22:16.21file[laptop]I'm not going to find the software for you
22:16.26file[laptop]as I don't know it off the top of my head
22:17.22syle2i;d rather code my own, just look for references to what variables to use
22:17.43file[laptop]variables? where?
22:18.18enderwhat file format should I make my voice prompts for * to use?
22:21.06*** join/#asterisk CrazyYoss (n=nobody@adsl-69-236-44-222.dsl.pltn13.pacbell.net)
22:21.23CrazyYosswhat is a "rate center"?
22:21.44*** part/#asterisk wrmem (n=monnin@monnin-win.cso.uiuc.edu)
22:21.52AsteriskNoobrate center is usually a city or area in regards to billing
22:22.11CrazyYossthank you
22:22.18*** join/#asterisk bongfrog (n=winston@206.165.75.198)
22:23.06AsteriskNoobrates in boise idaho are different from rates in new york.... yeah it just defines the rate for an area
22:23.09AsteriskNoob:) yw
22:25.29*** join/#asterisk Avero (n=no@dsl001-136-136.lax1.dsl.speakeasy.net)
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22:27.45*** mode/#asterisk [+o twisted] by ChanServ
22:29.30KranZgsm
22:33.27develword, people.  i looked on the voip-info wiki for a "voicemail reference" (like an end user reference) and came up with nothing.... is there something?
22:34.16*** join/#asterisk oej (n=Olle@dsl001-136-136.lax1.dsl.speakeasy.net)
22:36.46SupaplexHow can I find the bug int he bug tracking system for this message? http://www.archivum.info/asterisk-dev@lists.digium.com/2005-02/msg00122.html  I'm having the same issue.
22:37.01Supaplexint he/in the/
22:37.44*** join/#asterisk longtail (n=longtail@a213-22-84-129.cpe.netcabo.pt)
22:43.58*** join/#asterisk kb1_kanobe (n=krisbout@h24-207-96-50.cst.dccnet.com)
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22:49.15syzygyBSDWhat? boise?
22:50.46syzygyBSDdevel: is it something like http://voip-info.org/tiki-index.php?page=Asterisk+voicemail... or did you want more of a client's guide to options?
22:52.21syzygyBSDdevel: something closer to http://voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMailMain
22:52.44develright, syzygyBSD, that second one is the stuff.
22:52.59develthanks, syzygyBSD (not sure why i didn't find that....)
22:53.39syzygyBSDprobably didn't look up the right voicemail command, ended up with callers being directed to leave a voicemail instead of checking theirs
22:54.56develsyzygyBSD, nah, i think i just didn't think to look at the command reference for the behaviour.  it's been a long day :)
22:57.15*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
23:02.28*** join/#asterisk Starmaker (n=magnus@85.8.2.169)
23:05.56*** join/#asterisk drbrown (n=chatzill@72.9.47.17)
23:11.24*** join/#asterisk RazorJack (n=RazorJac@205.211.153.20)
23:12.18RazorJackHey Guys, having a problem with jitter..... When someone phones in zaptel -> sip.... I hear them fine on my PAP2-NA, but they get jitter hearing me.... I have no clue where to start debugging it other than the various adjustments I have made in the zap*.conf files
23:12.23*** join/#asterisk gambolputty (n=gambolpu@72.240.241.108)
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23:16.02drbrownhas anyone had any probs w/ asterisk T1's & gsm????
23:16.17drbrownspecifically in version 1.2beta1
23:16.25RazorJackasterisk t1's?
23:16.38RazorJacknope
23:16.43RazorJackother than my jitter problem
23:16.45drbrownasterisk + T1 + gsm
23:16.57drbrowngsm as in voicemail, promtps and codecs
23:17.09RazorJacknope, but not using 1.2beta1 yet
23:17.29drbrowneverything gsm is distorted
23:17.41drbrownonly over the T1 though
23:18.02drbrownit's a te110p & a Rhino channel bank
23:18.15*** join/#asterisk wcj (n=wcj@wsip-68-110-129-70.ga.at.cox.net)
23:18.23*** join/#asterisk SpaceBass (n=SpaceBas@c-24-125-184-203.hsd1.va.comcast.net)
23:18.28wcjhello
23:18.31RazorJackhey
23:18.39SpaceBassok... fyi.. #mtyhtv-users ... they are assholes...  just had to vent
23:18.48RazorJacklol spacebass
23:18.57RazorJackI agree.
23:18.59SpaceBassevery single possable topic is banned
23:19.16RazorJackI bit the bullet, and bought a wicked media center pc
23:19.20wcjwill someone in here answer some basic questions about pbx's ?
23:19.23Ariel_SpaceBass, wow nice hello
23:19.23RazorJackits awesome, will never go back to mythtv
23:19.26SpaceBassF that... my HD tivo works just fine... so what if I it doesnt integrate with my * box... not worth being in that #
23:19.33SpaceBassAriel_: sorry :) just venting :)
23:19.35RazorJackwcj, you just joined, didnt see a question from you.
23:19.48Ariel_SpaceBass, it's ok I understand
23:19.51wcjlol, well that was my first question
23:20.06wcjI don't know anything about pbx's or how to use them
23:20.10Ariel_what would you like to know about pbx?
23:20.13wcjbut I was thinking about setting one up
23:20.14Ariel_~docs
23:20.15jboti guess 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
23:20.21RazorJackAriel_; you a jitter fixing guru?
23:20.22wcjwell hold on
23:20.27wcjjust some basic questions here
23:20.27Ariel_wcj, what do you want to do with it?
23:20.38Ariel_RazorJack, no I am no guru
23:20.50RazorJackAriel_; hehe, me neither....
23:20.50wcjcan I create a public voip network?
23:21.04Ariel_wcj, great you have come to the right place.
23:21.05RazorJackwcj; FREE CALLING! WOOT!
23:21.14snittlol
23:21.17Ariel_RazorJack, what is the problem your having with the jitter?
23:21.26wcjlol
23:21.32SpaceBassAriel_:  i was looking for you earlier... had an amp question
23:21.35RazorJackzap -> sip.... I hear them PERFECTLY... no echo, they get jitter on the PSTN/ZAP side
23:21.41wcjso this is possible to do then? with only an internet connection?
23:21.51RazorJacknot sure where to start
23:22.14wcjI think razorjack said it best, I myself am not sure where to start
23:22.31RazorJackwcj; asterisk@home or amp.voxbox.ca
23:22.34RazorJackwcj; two good starts
23:23.08wcjwhat is asterisk@home ?
23:23.14RazorJack1sec, ill google it for ya
23:23.20wcjdon't worry
23:23.22wcjI can
23:23.23RazorJacklive linux mini cd
23:23.29RazorJackwith asterisk and amp installed on it
23:23.34RazorJackhttp://asteriskathome.sourceforge.net/
23:24.12Ariel_it's a self installing asterisk setup not a live cd. It will erase your drive.
23:24.25Ariel_but it's the fastest way to get asterisk and a great gui up and running.
23:24.28RazorJackAriel_; sorry, ya just saw that
23:24.39SpaceBassAriel_:  how can I enable call waiting at the user level? seems to be at the device level
23:24.41RazorJackIt was a live cd at one point wasnt it?
23:24.51RazorJackbased on knoppix
23:24.51Ariel_SpaceBass, *70
23:24.53RazorJackI thought
23:25.16SpaceBassAriel_:  I did that, and in the CLI it showed the device, not the user
23:25.17Ariel_RazorJack, there is a rapid cd which was based on Debian
23:25.30RazorJackmaybe thats it, too many linux distros
23:25.34Ariel_ahh you have the device separete from the extensions.
23:25.41wcjso basically all I have to do is install asterisk and configure it?
23:25.54wcjthen purchase hardware that can interface to it?
23:26.02RazorJackwcj; installing is easy... configuring... dont call me, ill call you :)
23:26.07wcjlol
23:26.09wcjso I've heard
23:26.12*** join/#asterisk oej (n=Olle@dsl001-136-136.lax1.dsl.speakeasy.net)
23:26.15wcjbut I like a challenge ;)
23:26.16*** join/#asterisk xyharley (n=jets@dsl001-136-136.lax1.dsl.speakeasy.net)
23:26.33RazorJackwcj; DONT get a clone wildcard x100, biggeste mistake I made learning asterisk
23:27.00wcjI'll keep that in mind
23:27.00RazorJackwcj; or a grandstream phone
23:27.05RazorJackwcj; they belong in the lake
23:27.26wcjwell i might go for a swim if I can find some free ones down there
23:27.31SpaceBassAriel_:  well I have a device with an extension (6000) and a default user (6010)...
23:27.35wcj(long as no ameaba attack me)
23:27.54wcjanyways
23:27.59RazorJackwcj; if you wanna tweak, sure go for it, but I use my asterisk not for just testing
23:28.06wcjso I don't have to have any kind of link to a phone company?
23:28.16wcjI just install and set it up?
23:28.17RazorJackwcj; http://amp.voxbox.ca/ and http://asteriskathome.sourceforge.net/
23:28.23RazorJackwcj; ummm you wanna call ppl right?
23:28.28wcjyes
23:28.37RazorJackwcj; skype....
23:28.49RazorJackwcj; free world dialup
23:29.01HiltonTschype
23:29.06RazorJackwcj; vonage lite
23:29.12wcjyeah, but then I learn onthing and have nothing new to put on a resume
23:29.12RazorJackI dunno, depends on your preference
23:29.28HiltonTif only they complied with standards, maybe eBay's $2.6bn would have made sense
23:29.41RazorJacklol
23:30.10HiltonT$2.6bn - I mean, welcome back to the Tech issues a few years ago in the stockmarket!
23:31.12RazorJackProgramming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the Universe trying to produce bigger and better idiots. So far, the Universe is winning. "
23:31.25HiltonTit always will!
23:31.46RazorJackHiltonT; you know anything about jitter?
23:32.01wcjwell thank you guys for your input
23:32.06RazorJackwcj; good luck
23:32.21wcjty
23:32.28HiltonTRazorJack; I've only started playing with * in the past 24 hours  :)
23:32.35RazorJackgive asterisk@home a try
23:32.46RazorJackHiltonT; I've been way longer
23:33.10wcjwill do
23:33.12RazorJackHiltonT; no clue where this jitter came from, it just started a month ago after no upgrades
23:33.50wcji'm sure i'll see you guys in here again
23:33.50wcjbye for now
23:33.51SpaceBassthe latest aah seems to have more jitter for me... lot more... but I think its b/c it has more overhead and I have a slow box
23:33.55Ariel_RazorJack, your getting a sound that is jitter?
23:34.04RazorJackWell thats the thing, how do you calculate your overhead....
23:34.21Ariel_RazorJack, what codec are you using how many users?
23:34.22HiltonTmaybe local network bandwidth is less than it was
23:34.44RazorJackulaw, 4 users
23:35.06*** join/#asterisk Avero (n=no@dsl001-136-136.lax1.dsl.speakeasy.net)
23:35.08Ariel_SpaceBass, I need to change my setup sometime to device different then extensions to find do some test. But I kinda like the extension being like the devices.
23:35.25Ariel_RazorJack, what service do you have dsl/t1 data cable?
23:35.36RazorJackI just got the PAP2-NA firmware updates from cisco today.... havent upgraded the adapter yet... but it WAS working fine
23:35.48SpaceBassAriel_:  I just did it to play around... I can easily go back to devices and extensions being the same
23:36.00RazorJackAriel_; sip extension 200 -> pap2-na -> asterisk -> zap tdm400 -> bell
23:40.38RazorJackhttp://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM400P thats the card im using
23:40.55RazorJackwonder if theres a firmware update or something
23:45.12*** join/#asterisk SpaceBass (n=SpaceBas@c-24-125-184-203.hsd1.va.comcast.net)
23:46.10*** part/#asterisk Avero (n=no@dsl001-136-136.lax1.dsl.speakeasy.net)
23:46.59Ariel_RazorJack, are you using stable or cvs head?
23:46.59*** join/#asterisk ben_d (n=ben@cpe-66-66-209-96.rochester.res.rr.com)
23:47.02*** part/#asterisk IzNoGooD (n=marc@iznogood.demon.nl)
23:47.26RazorJackAriel_; stable
23:47.42RazorJackjust removed couple lines from my zapata.conf
23:47.49Ariel_you can use the zaptel drivers from cvs head which have a better echo routing.
23:47.51RazorJackcontext=from-pstn
23:47.51RazorJacksignalling=fxs_ks
23:47.51RazorJackgroup=0
23:47.51RazorJackchannel=4
23:48.10RazorJackthis aint echo, its jitter, or am I missing something?
23:48.29RazorJack1.0.9 asterisk
23:49.14*** join/#asterisk bweschke (n=bweschke@dsl001-136-136.lax1.dsl.speakeasy.net)
23:50.05kb1_kanobeRazorJack: if you originate the call from the opposite side does it still display that behaviour?
23:50.11RazorJackyep
23:50.25kb1_kanobehave you tried dialing into app_milliwatt() from both sides?
23:50.39RazorJackwhoa, no, how do I do that?
23:50.47*** join/#asterisk Aquiles (n=Aquiles@63.245.80.243)
23:51.06RazorJackjust make an extension to app_miliwatt?
23:51.25Ariel_RazorJack, like a small buzz
23:52.07kb1_kanobeRazorJack: yes. It will generate a constant 1004hz signal. If the sip side shows the defect but the zap doesn't, so there is a problem on that path etc.
23:52.17RazorJackfrom the sip side, its not coming down to me, its going back jitter
23:52.24*** join/#asterisk carrar (i=tim@osburn.com)
23:52.26carrarw00t!
23:52.32kb1_kanobetry zap into app_milliwatt.
23:52.43RazorJack1s, ill do it right now
23:54.06RazorJackexten => *07,1,app_milliwatt() ?
23:54.34RazorJackim missing something :P
23:55.00kb1_kanobetry just milliwatt()
23:55.08RazorJackdont I have to load it in modules.conf ?
23:55.22RazorJackload => app_milliwatt.so ?
23:55.25*** join/#asterisk yogurt2ungue (n=yogurt2u@44-170-114-200.fibertel.com.ar)
23:55.43*** join/#asterisk wcj (n=wcj@wsip-68-110-129-70.ga.at.cox.net)
23:56.16RazorJackspeaker phone on my sip extension 200
23:56.20kb1_kanobehere we are: http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Milliwatt
23:56.21RazorJackdialed *07, constant tone
23:56.27RazorJackno jitter
23:56.40*** part/#asterisk Aquiles (n=Aquiles@63.245.80.243)
23:56.42*** join/#asterisk Aquiles (n=Aquiles@63.245.80.243)
23:56.44RazorJackgonna phone in on lanline, 1 sec
23:56.53*** part/#asterisk Aquiles (n=Aquiles@63.245.80.243)
23:57.41RazorJackweird, just dialed from bell line, to my asterisk server and dialed *07, constant 1004 tone
23:57.43RazorJackno jitter
23:58.13*** join/#asterisk apardo (n=w0w0@1.Red-83-46-192.dynamicIP.rima-tde.net)
23:58.14kb1_kanobeinteresting.
23:58.47wcjcan anyone tell me what asterisk's capabilities are with a single lan line?
23:59.28fiferFull SIP based phone server

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