irclog2html for #asterisk on 20060502

00:04.12*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
00:07.50mog_workit builds here MRH2
00:07.53mog_worktrunk does not though
00:07.55mog_workat the moment
00:16.11*** join/#asterisk IceManRISK (n=kart@200.138.147.142)
00:16.23MRH2ok this time it worked how wierd
00:16.47MRH2(weird)
00:19.42*** join/#asterisk kamileon (n=kamileon@68.62.190.253)
00:23.21*** part/#asterisk CoffeeIV_ (n=CoffeeIV@64.149.168.97)
00:25.45websaeheloo, is this thing on?
00:25.47websae*hello
00:26.01websaeanyone getting these RTP packets...
00:26.01mog_workno
00:26.05websaedidn't think so
00:27.03mog_workthis being irc
00:27.06mog_workits difficult
00:27.28websaechuckles
00:27.42websaemog_work where are you from?
00:28.03mog_workhuntsville alabama
00:28.19websaethat's right
00:28.33orlokok, i want to kill the fucker esponsible for sorbs
00:29.55MRH2ok definately all working must have been something from the prev install it didn't like
00:30.02MRH2thanx mog
00:30.46*** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)
00:32.07xachenHeya there aleph :)
00:32.31alephcomgreetings xachen
00:32.32xachenorlok: What did SORBS do to you?
00:33.44xachen:)
00:34.29xachenAnybody know if Rackmountetc boxes are any good?
00:36.06*** join/#asterisk demigod2k (n=joey@cpe-24-210-97-162.twmi.res.rr.com)
00:36.16demigod2khi
00:36.27*** part/#asterisk willt (i=wt@wifi-napanet-static-206-81-99-68.napanet.net)
00:37.24*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
00:38.14orlokxachen: Hmm. Listed our static ranges
00:38.25orlokxachen: then, they list our ISP's SMTP server
00:38.55xachenheh
00:39.05xachenI do my part and refuse to use BL's
00:39.09orlokyeah
00:39.15orlokone of the big isp's here uses sorbs though
00:39.27*** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com)
00:39.29orlokits funny, the guy that started sorbs actuall works for somebody we do business with
00:39.31xacheninstead I just use Spamassassin/Dspam
00:39.33xachenworks good ;)
00:39.35orlokyeah
00:39.38orlokand greylisting
00:39.53orlokdo some smart/funky stuff on the secondary mx's too
00:39.53xachenI block a few fraud countires
00:39.53xachenbut thats it
00:39.55alephcomxachen:  Do you still like Telus? :-)
00:39.58xachenMalaysia, Romania
00:39.58demigod2kthose BLs cause me nothing but trouble. I publish SPF records and that's about it
00:40.01xachenalephcom: I hate it
00:40.21demigod2kthe store-and-forward thing looks like it'd be worthwhile, but other than that just too many rejections
00:40.27alephcomThey changed a bunch of "static" addresses last week with absolutely no warning.
00:40.33xachenoh nice
00:40.47xachenthe ISP i'm getting NATs its customers onto like 3 public IPs
00:40.49demigod2kthose turds. they should use a DNS record
00:41.07*** join/#asterisk e-milio (n=emilio@pmr.pmrtechnologies.com)
00:42.02e-miliohello everybody
00:42.04*** join/#asterisk websae (n=websae@CPE-24-167-206-22.wi.res.rr.com)
00:42.28e-miliohave a quick question with eagi perl sayunixtime
00:42.51xachenanyways I'm going for dinner
00:42.52xachenbbiab
00:43.41e-miliothis is not working:  $AGI->exec('SayUnixTime', sprintf("%s||ABdY \'digits/at\' HM", UnixDate($data, "%s")));
00:44.03alephcomxachen:  They check sites you visit too?
00:45.03e-milioif i use default format works
00:45.34e-miliobut if i tried to use dif from default dont work
00:45.48e-milioanbody can elighten ??
00:49.23*** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com)
00:49.26*** join/#asterisk terrapen (n=cjs@166.70.183.109)
00:50.54terrapenanybody seen this before?
00:50.56terrapenhttp://pastebin.com/693403
00:51.12terrapenAs soon as I dial an extension on the PBX at the other side of this PRI, it hangs up
00:51.27terrapenfunny thing is, I had it working this morning.  Not sure how I broke it
00:53.50terrapensigh
00:54.29*** join/#asterisk zagaya971 (n=almeli@APointe-a-Pitre-102-1-3-9.w81-248.abo.wanadoo.fr)
01:01.57*** join/#asterisk surfdue (n=tyler@unaffiliated/surfdue)
01:01.58surfduehey
01:02.10surfduewhy would asterisk be playing a no service announcement ?
01:02.14surfduewhats a possiblity
01:02.22*** join/#asterisk moprilo (n=jjohn@201.198.78.23)
01:02.55Strom_Csurfdue: I'm going to guess there's no serice
01:02.58Strom_Cer, serice
01:03.02Strom_Cservice
01:03.03moprilohi, i've been using asterisk for a while now, i have xorcom on my main servers, but asterisk@home on the small offices, I'm looking for an alternative to asterisk@home
01:03.04surfdueStrom_C, how?
01:03.05mopriloany ideas?
01:03.14surfdueStrom_C, i can see in asterisk -r that it connects and all
01:03.16Strom_Csurfdue: what is the text of the announcement?
01:03.22surfdueStrom_C, asterisk is just playing a message lol.
01:03.29surfduethere is no service.
01:03.39surfdue<PROTECTED>
01:03.40Strom_Csurfdue: yes, please transcribe exactly what the recording is saying
01:03.41surfdue:P
01:03.55Strom_Cah ok
01:03.57Strom_C*shrug*
01:05.09*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
01:09.29*** join/#asterisk trbldwine (i=trbldwin@c-71-194-161-170.hsd1.il.comcast.net)
01:11.48orloknemith: wireless isp?
01:11.55orlokman i've dealt with some shit wisp's
01:12.09*** join/#asterisk MacWeenie (n=chatzill@82-35-73-28.cable.ubr02.dals.blueyonder.co.uk)
01:13.47e-milioHello
01:14.13e-milioI and trying to sayunixtime
01:14.26e-miliobut it always says one hourearlier
01:14.34e-miliothan the time that i give to it
01:14.54e-milioplease, anyboyd has anyidea ?
01:22.17dlynestainted-, qwell: seems like the problem might be the machine for that error it's spitting up, not the card
01:23.25dlynese-milio: what's your time zone?
01:24.38dlynese-milio: EDT, right?
01:26.36e-milioyes
01:26.54e-miliodlynes: I have just changed it to ESt
01:27.05e-miliodlynes: but same problem
01:27.55dlynese-milio: um....you should be EDT
01:28.04dlynese-milio: EST was about a month ago
01:28.35dlynese-milio: what does it say when you type 'date' at a linux prompt?
01:29.57e-milioMon May  1 21:33:45 EDT 2006
01:30.22e-milioso it is right now
01:31.24e-miliobut still
01:31.36e-miliodlynes: $AGI->exec('SayUnixTime', sprintf("%s", UnixDate($data, "%s")));
01:31.53IceManRISKboa garoto
01:32.33dlynesummm
01:32.46dlynesnvm
01:33.04e-milio?
01:34.54dlynese-milio: in agi, can you specify parameters to 'SayUnixTime'?
01:35.15e-miliodlynes: I tried but doesn't seem so
01:35.44e-miliodlynes: i just rejects the date and 'says' today
01:36.11dlynese-milio: like maybe $AGI->exec('SayUnixTime(,EDT)', sprintf("%s", UnixDate($date, "%s"))) ;
01:36.22dlynese-milio: btw...what's "UnixDate"?
01:36.53e-milioit returns the num of mins from epoch
01:37.00*** join/#asterisk bjohnson (n=bjohnson@i216-58-62-76.cybersurf.com)
01:37.06e-milioseems standard
01:37.34dlynese-milio: in perl?
01:37.38dlynese-milio: or C?
01:37.47dlynese-milio: or?
01:37.50e-milioat least in AGI, on all examples
01:38.05dlynese-milio: what language are you writing AGI in?
01:38.10e-milioperl
01:38.14dlynesah
01:41.42e-miliodlynes: didnt work
01:41.59e-miliodlynes: sure about syntax ?
01:42.02gaupee-milio: number of seconds
01:42.24e-miliogaupe: yes, your are right
01:43.37dlynese-milio: Nope, I am not
01:43.42demigod2kopinion. buy all polycom 301s or all gxp2000s ?
01:43.58Strom_Cwhy in god's name would you even consider grandstream sets? :P
01:44.00dlynese-milio: but I've never used the UnixDate function in perl, either
01:44.11demigod2kthe gxp has tons of buttons to assign to features
01:44.19dlynesIf it was a choice between grandstreams and polycoms
01:44.19demigod2kalso the 301 has 2 line appearances while the gxp has 4
01:44.28dlynesI wouldn't even think about the grandstreams
01:44.38Strom_Calso the grandstream will fall apart if you sneeze on it
01:44.53demigod2kanything else in that pricerange is fine too but I just own those two to compare so far
01:44.56dlynesUnfortunately, for a lot of my customers, polycoms aren't even an option; they're too cheap to pay for them
01:45.09demigod2kya the gxp does sort of feel like crap
01:45.16demigod2kI also really, really hate the binary configuration file
01:45.30dlynesdemigod2k: sipura's same crap
01:45.49e-milioI have had very good experiences with polycom
01:45.58e-miliomany callcenters working for many time
01:46.19demigod2kI have one polycom for testing. it seems ok so far, nice provisioning files, nice ringtones, just the fewest buttons and line appearances
01:46.33e-milioi agree
01:46.35demigod2kand the manual claims no paging
01:46.39demigod2kI havent tried though
01:48.14harryvvdam door bell hold on
01:48.17*** part/#asterisk surfdue (n=tyler@unaffiliated/surfdue)
01:51.21e-miliodlynes:thanks anyway
01:51.56*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
01:54.22SplasPooddemigod2k: define "paging" ... You can set the polycom up to autoanswer on a given ring type sip header
01:54.37demigod2kthat would be my definition
01:54.54SplasPoodyea works fine.. howto on voip-info.org
01:55.17demigod2ksaid in the manual book that it wouldnt do it, but I had heard it in here once before
01:55.33demigod2kthe manual may be phone-to-phone without a server, or something
02:03.02*** join/#asterisk surfdue (n=tyler@unaffiliated/surfdue)
02:03.04surfduehi
02:08.27*** join/#asterisk Eggplant (i=No@dsl-332.cascadeaccess.com)
02:09.59mog_workw
02:11.49*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
02:16.54DoktorGregok, any suggestions on which mobo i need to get 100% in zttest?
02:18.32DoktorGregpri + musiconhold = dropped call at 98.75% zttest results....
02:18.34surfduehaving problem with linksys pap2 im getting the signal from asterisk its gooing in and out for some reason ? I have a oruter but the ports are unblocked :|
02:19.09demigod2kbummer. I get around 98% or so
02:19.21demigod2kI figured that was natural
02:19.24DoktorGregdocs say i should shoot for 100
02:19.34DoktorGregpri is bridging just fine
02:19.47DoktorGregsip phones hit all the features just fine
02:19.57demigod2kmakes sense. I bought an off-the-shelf box, no clue what mobo. mine gets that 98% or so
02:20.01DoktorGregpri cant hit the features
02:20.06surfdueoh disabled firewall and it works :|
02:20.11surfduenot safe though lol oh well.
02:20.59DoktorGregDo i just need to burn the cash and get smp for that last 1.25%?
02:21.11DoktorGregand
02:21.22DoktorGregdo i need actual smp mobo?
02:21.33DoktorGregor can i get one of the fancy new multi core cpus?
02:21.49DoktorGregoh
02:21.53DoktorGregand there blows my budger
02:21.58DoktorGregbudget
02:23.13demigod2kno idea. mine is only about 4 lines
02:23.16demigod2kand a 1.0 ghz machine
02:23.24*** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)
02:23.24xachenalephcom: As far as I know they are known to do that
02:23.26xachenblah
02:23.27xachenhaha
02:23.34xachentalk about timing
02:24.07DoktorGreglol, i must need different pci chipset
02:24.24DoktorGregcpu is amd is 2200 thingie
02:24.47demigod2kor different PCI cards on your bus or something
02:24.50demigod2kmaybe something doesn't play fairly
02:24.58DoktorGregand i have shut everything off, except for on mobo lan
02:24.59*** join/#asterisk Strom_C (n=strom@gateway.digium.com)
02:25.12DoktorGregi even took out the graphics card and ran it headless
02:25.26demigod2ktheres a possibility too, check if you have a shit chipset
02:25.52DoktorGregiirc nforce2
02:26.06DoktorGregmsi motherboard
02:26.39DoktorGregi wonder if i should just build a 4 way server
02:27.00DoktorGregnaw
02:27.09demigod2kI doubt it's necessary, the processor isnt that huge a deal really
02:27.30demigod2ksomething with clean interrupts seems most important from what I've read. although my system is tiny compared to others
02:27.31DoktorGregwell digium does suggest smp may be needed
02:28.05demigod2kcould be I've never had that large a configuration. I just imagine you hit the point of diminishing returns
02:28.27*** join/#asterisk brockj49464 (n=brockj49@63.87.56.236)
02:28.39DoktorGregwell generally on big servers the idea is
02:28.53DoktorGregyou can run more threads
02:29.09DoktorGregand server software is generally multi threaded
02:29.10demigod2kassuming the linux multiprocessor code is good which is a big question
02:29.27demigod2kI'd just make damn sure that every single card on the bus has the worlds best driver first
02:29.41demigod2kthe fastest processor in the world won't make up for something eating up that 33 mhz bus
02:29.42DoktorGregI have had better results with linux on servers than i have had with windows...
02:30.00demigod2kI wouldn't say windows does any better with multiprocessor
02:30.07demigod2knothing does well enough to justify the cost if you ask me
02:30.50DoktorGregwell i have a dual 1 ghz p3 laying around
02:30.58DoktorGregI could try it on that
02:31.20demigod2keh maybe. I bet replacing that nforce2 based NIC with something decent would do more
02:31.38DoktorGregyah...
02:31.43demigod2kI wouldnt be surprised if you ran into more problems than ever with a dual processor
02:32.11DoktorGregi consider this a bump in the road to be solved
02:33.10DoktorGregso tomorrow i try different nic
02:33.21DoktorGregat least i figured out my problem with my ata today
02:33.23demigod2kya its a start
02:33.46DoktorGreglol, problem with my ata was me assuming i knew better than default settings
02:34.46DoktorGregi had messed with a bunch of things in the nat config on the ata
02:35.02DoktorGregwhen all i had to do was say, nat=yes
02:35.14DoktorGregstun server = my stun server
02:35.21DoktorGreguse stun = yes
02:35.36dlynesAnyone know where the Makefile.patch file is for spandsp 0.0.3?
02:36.25DoktorGregcan i get a mobo with multiple pci busses?
02:36.28*** join/#asterisk Alystair (n=bob@CPE001109c15241-CM00407b8794db.cpe.net.cable.rogers.com)
02:36.38AlystairAnyone here heard of Thirdlane?
02:36.39dlynesDoktorGreg: you mean like pci-x and pci?
02:37.37DoktorGregum kinda
02:37.56*** join/#asterisk linlin (n=linlin@c-67-184-230-198.hsd1.il.comcast.net)
02:38.07DoktorGregcan i get 2x+ pci busses on the same mobo?
02:38.28*** join/#asterisk linlin (n=linlin@c-67-184-230-198.hsd1.il.comcast.net)
02:40.58DoktorGregok here i go, found some more stuff to tru
02:41.01DoktorGregtry
02:41.07DoktorGregscarry
02:41.13DoktorGregim gonna build a kernel remotely
02:41.49xachenI do that all the time
02:42.02xachennothing scary about it.... except when you get it perfect except build the wrong NIC module
02:43.30DoktorGregwell im gonna use old config file
02:43.56DoktorGregofficial docs call for 2.4.20 kernel
02:44.13*** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca)
02:44.13DoktorGregand i have a 2.4.27 kernel
02:44.24*** join/#asterisk angom_h (n=angom@red-corp-200.79.134.173.telnor.net)
02:44.39DoktorGregso my first thing to do is an apt get
02:45.11DoktorGreger rather wget
02:49.35*** join/#asterisk speedracer (n=collin@24.96.142.189)
02:50.16*** join/#asterisk fami (n=fami@unaffiliated/pmai)
02:51.11tainted-anyone need colocation
02:51.21speedracerfree?
02:51.30speedracerlol
02:51.32speedracerhey file
02:51.36filehi!
02:52.03xachentainted-: TEll ya what, I'll talk colocation if you pay me $50/mo
02:52.26speedracerlol
02:52.34xachenI think thats fair
02:52.48*** join/#asterisk trig_hm (i=jason@home.monkeypr0n.org)
02:53.58Netgeeksanyone ever seen a SIP/2.0 603 Declined message before from asterisk?
02:54.04speedracerOkay, so I'm a FC nub...as well as an Asterisk nub I guess, and I'm trying to compile Asterisk.  I'm starting with Zaptel, and `make clean` works fine, but `make` exits with an error about the sources for my kernel not being installed...but at the same time, yum tells me they are...
02:54.24speedracerany ideas?
02:54.47Netgeeksspeedracer there is a text file in the zaptel distribution that tells you what you need to do to fix that
02:55.15Netgeeksdon't remember off the top of my head what it's called but it's there
02:55.27filespeedracer: you should call support tomorrow ^_^
03:00.10speedracerfile: when?  I'll be at work at the same time they are...
03:00.14speedracerno good
03:00.47filegrab someone from support and offer to buy them lunch if they get your stuff working :P
03:00.57speedracerlol
03:01.00speedracerif only it were that easy
03:01.01speedracer:-\
03:01.13speedraceri figured I could get some help in here
03:01.16filedooooooooooo it
03:04.38russellbfile: shush you
03:05.21filerussellb: do I have to? :(
03:05.27russellbfile: YES
03:05.38russellband go get me something to drink!
03:05.47fileyou're not my boss! :P
03:06.05russellbgooood!
03:06.08russellbthat would be weird
03:06.17fileor what if I was YOUR boss...
03:06.23russellbNEVAR
03:06.39*** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka)
03:06.43orlokbloody. sorbs.
03:06.48*** join/#asterisk tessier_ (n=treed@adsl-75-5-99-178.dsl.sndg02.sbcglobal.net)
03:06.49dlynesDoes anyone have spandsp 0.0.3 working with asterisk 1.2.7.1?
03:07.33dlynesI've decided to try giving it another go, but the Makefile.patch doesn't seem to exist in either
03:07.58dlynesOr the app_txfax.so and app_rxfax.so, for that matter
03:08.03dlyneserm .c i mean
03:09.50*** join/#asterisk dahunter3 (n=dahunter@pool-71-110-89-49.lsanca.dsl-w.verizon.net)
03:15.08*** join/#asterisk OloBola (n=not@netblock-68-183-67-158.dslextreme.com)
03:15.52AlystairWhen a phone is single line does that mean you cannot do a call waiting kinda setup?
03:16.14Alystair(like pressing flash to switch conversations on normal phone when you get another call)
03:18.50dlynesAlystair: sure you can, if you have call waiting subscriber service
03:19.08dlynesAlystair: it all depends on what line subscription services you have on that line
03:20.15Alystairyou mean from the DID provider
03:20.34Alystairor an internal asterisk setup kinda thing
03:23.45*** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net)
03:31.39*** join/#asterisk Iaxy (n=Iaxy@modemcable236.55-131-66.mc.videotron.ca)
03:31.52IaxyHi all
03:32.37IaxyI just installed asterisk for buisness and there is no /etc/zaptel
03:32.47IaxyDid they omit that?
03:33.00CunningPikeIaxy: I have to ask - did you install zaptel?
03:34.43Iaxyit is a binary version of Asterisk, and there is one install script. I will check the book again. but thats what I read
03:38.02*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
03:38.05CunningPikeIaxy: I'm afraid I've never installed ABE
03:38.29CunningPikeBut I would check to see if zaptel is included
03:38.50CunningPikeIt's /etc/zaptel.conf you're looking for, right?
03:39.02IaxyCunningPike: yup. its not there
03:39.28CunningPikeHmm
03:39.48*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org)
03:39.54CunningPikeDouble-check the docs - I don't want to advise you install the OSS version of zaptel if it will break your suppoirt
03:40.36Iaxyhehe what good is ABE if there is no zaptel and I got a quad T card in it...:-)
03:41.56*** part/#asterisk speedracer (n=collin@24.96.142.189)
03:42.04Iaxyit says to uninstall Asterisk run rpm -e asterisk libtonezone libpri
03:42.20Iaxywhere the hell is zaptel?
03:42.59*** join/#asterisk junbug (i=junya@67.191.62.53)
03:43.01*** join/#asterisk _x0r (n=dusty@12-219-148-217.client.mchsi.com)
03:43.31*** join/#asterisk salviadud (n=dude@dsl-201-129-86-188.prod-infinitum.com.mx)
03:43.47salviaduddammit, nobody's from brazil
03:44.35CunningPikeIaxy: This is ABE, right?
03:45.11IaxyCunningPike:  yes Asterisk Buisness Edition
03:45.31CunningPikeWell, I would contact Digium - you've paid for it :)
03:45.47salviadudyeah man, who pays for asterisk anyways?
03:45.51salviadudi thought it was free...
03:46.03salviadudi'm using it... it tastes like it's free
03:46.07IaxyCompanies that feel better when they pay.
03:46.26salviadudwell, you get what you pay for
03:46.32salviadudcall the dudes at digium
03:46.49salviadudthey outta be nicer than the guys at broadvoice
03:46.52salviadudor so i've heard
03:47.21IaxyThis is a bitch....
03:47.42salviadudsomegeek, what's wrong eh?
03:47.53salviadudoh.. damn xchat
03:48.01russellbIaxy: zaptel isn't installed as an RPM on ABE
03:48.02salviadudi meant to say, so, whats wrong eh?
03:48.19russellbsince it's very specific to the kernel you have, it has to be built on the system
03:48.44russellbIaxy: however, the source is included on the CD, and the install script will install it for you automatically.
03:48.53*** part/#asterisk _x0r (n=dusty@12-219-148-217.client.mchsi.com)
03:49.30russellbor at least it is supposed to.
03:49.48russellbIaxy: if you would like, I could log in to your system and fix it up for you ...
03:49.51Iaxythe install script did not install it.
03:49.51*** part/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
03:50.15IaxyI don't think I put the kernel sources in the dist install
03:50.18Iaxyhmmm...
03:50.35russellbso maybe the script tried to install it and failed
03:50.49russellbit's supposed to handle dependency checking, but i haven't touched that script in about a year
03:50.58IaxyIt should have told me.
03:51.08*** join/#asterisk bmg505 (n=leon@c1-175-1.rndf.isadsl.co.za)
03:51.19IaxyLet me look at the script
03:51.28*** join/#asterisk TheCops (i=nobody@got.securebinary.com)
03:51.40russellbIaxy: just untar the source that's on the cd and install it manually
03:55.02Iaxyit supposed to have installed it but it didn't.
03:55.45CunningPikeIaxy: If you do what russellb says, you might get an error you can work with
03:56.57IaxyI just installed bear minimum. Thats alot of stuff I am going to have to do to get it to compile.
03:57.47russellbwell, you can contact Digium support if you would like help doing it.
03:57.47CunningPikeProbably why the install didn't work off the CD then
03:57.57CunningPikeWhat he saisd
03:58.33*** join/#asterisk Johnnie (n=jdlewis@dynamic-acs-24-154-91-195.zoominternet.net)
03:58.42Iaxyyup....
04:01.30IaxyI added all the requirements.
04:03.09Iaxytar: zaptel-be/zconfig.h: time stamp 2005-09-13 16:34:50 is 7380521 s in the future
04:03.09Iaxyzaptel-be/zonedata.c
04:03.10Iaxytar: zaptel-be/zconfig.h: time stamp 2005-09-13 16:34:50 is 7380521 s in the future
04:03.10Iaxyzaptel-be/zonedata.c
04:03.11Iaxytar: zaptel-be/zconfig.h: time stamp 2005-09-13 16:34:50 is 7380521 s in the future
04:03.11Iaxyzaptel-be/zonedata.c
04:03.12Iaxyhahaha, it didn't compile because blabla timestamp is in the future.... hmmm
04:03.46russellbwell fix your clock
04:03.51russellb2005 is obviously not the future
04:03.54Iaxynow thats funny
04:04.38CunningPikeWhat does 'date' say
04:05.05Iaxyjun 20 2005....:-)
04:08.38*** join/#asterisk kmilitzer (n=km@office-gw.westend.com)
04:09.00*** join/#asterisk xtr (n=94752345@S0106000c41ed11e1.vf.shawcable.net)
04:16.53Iaxyoo those are ugly errors
04:20.05CunningPikeIaxy: ntp is your friend ;)
04:20.39*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
04:21.07Iaxyit has ugly compile errors
04:22.30IaxyIt says required dists. Redhat enterprise 3 or fedora Core 3.
04:23.07IaxyI used redhat enterprise 4
04:23.25russellbshould still work
04:24.22Iaxyit shows RHEL 4 in the install script. so yeah it should work, anything other than normal is it  is smp, dual cpu's.
04:25.10asterboy@russellb, whats it take to setup a jbot in a channel?
04:25.11*** join/#asterisk carrar (i=tim@osburn.com)
04:25.14carrarhi
04:25.17*** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com)
04:25.19asterboyhigh
04:25.23carrarAnyone use the unistim driver?
04:25.38asterboy~unistim
04:26.13russellbasterboy: i have no idea
04:26.28asterboywho setup jbot here so I can poll them?
04:26.38russellbdon't know
04:26.42russellbit has been here longer than i have
04:26.50carrarI can't get a unistim phone to call a cisco sip phone
04:26.52asterboyinteresting
04:27.01carrarcisco can call unistim though
04:27.12asterboy~jbot
04:27.14jbotfrom memory, jbot is only marginally useful at best,  He got a C- on his Turing Test
04:27.18carrarI've tried forcing ulaw and alaw
04:27.25russellbjbot: who is your daddy
04:27.26jbotYOU are, Mr. Sexy Pants
04:27.29asterboylol
04:27.32russellb:)
04:27.51asterboyjbot: who is your real daddy
04:27.54jbotasterboy: what are you talking about?
04:28.03russellbjbot: who is your owner
04:28.05jbotI think you lost me on that one, russellb
04:28.09russellboh well :)
04:28.16asterboyhuh
04:28.27Iaxymake -C /lib/modules/2.6.9-34.ELsmp/build SUBDIRS=/tmp/zaptel/zaptel-be modules
04:28.27Iaxymake[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-smp-i686'
04:28.28Iaxy<PROTECTED>
04:28.28Iaxymake -C /lib/modules/2.6.9-34.ELsmp/build SUBDIRS=/tmp/zaptel/zaptel-be modules
04:28.29Iaxymake[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-smp-i686'
04:28.29Iaxy<PROTECTED>
04:28.30Iaxy.
04:28.30asterboyI want to get jbot in #mythtv
04:28.54russellbIaxy: those aren't errors
04:28.58russellband please use pastebin ...
04:29.15distortion./k Iaxy spammah!
04:29.23asterboylol, not only is it not an error, but its spam
04:29.27Iaxyyeah sure.... sorry
04:30.15*** join/#asterisk L|NUX (n=linux@202.5.145.58)
04:30.20Iaxythe errors didn't make the paste
04:30.26asterboyah
04:30.50asterboythat sucks...maybe a 2>&1 at the end of the command
04:30.59russellbasterboy: mythtv rocks :)
04:31.19asterboyI'll say, I'm can't wait to get my box up and running.
04:32.03Iaxyhttp://pastebin.com/693589
04:32.10Iaxythere are the errors...
04:33.08asterboyJust picked up a P4 1.8GHz, 40Gb, 256Mb box for $140 CDN.
04:33.25asterboyThat will be my Myth box with 3 PVR-250s
04:34.03asterboyFrontend will be a MediaMVP with mvpmc
04:34.15russellbeep, you need more drive space than that
04:34.23russellbbut i guess this is the wrong channel for that discussion :)
04:34.50*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)
04:34.53asterboyoh ya...I'm adding a 120Gb Sata but would be nice to have much more.
04:34.55Iaxyrussellb: did you take a look at the compile errors?
04:35.19russellb~centosbug
04:35.20jbothmm... centosbug is a problem with the latest Centos kernel (4.2 and 4.3).  To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package.
04:35.41russellbsooo ... those errors look just like that bug
04:36.34russellbyes, that's what it is
04:36.36Iaxyhmmm...
04:36.53russellbI would be happy to fix that for you if you would like.
04:37.53distortionhmm, i think im going to grab the dell 2405 monitor
04:38.02distortionits so big and sexy
04:39.27*** join/#asterisk gursikh (n=guriskh1@adsl-68-95-82-50.dsl.hstntx.swbell.net)
04:40.53*** part/#asterisk ghost99 (n=neville@222-152-219-77.jetstream.xtra.co.nz)
04:41.06Iaxyrussellb: you da man...
04:41.20blitzragehail!
04:41.32*** join/#asterisk yxa (n=diablo@58.185.90.101)
04:41.33russellbIaxy: that will be $50.
04:41.45blitzragerussellb: you're cheap!
04:41.50russellbi know!
04:41.57Iaxy$50 for typing ~centosbug
04:41.58blitzrageI could afford you at that rate :)
04:42.20Iaxyhow about a supper
04:42.28russellbIaxy: I'm just kidding ... consider that your Digium support
04:42.38blitzrageIaxy: what is your time worth, and how long would you have spent searching for the answer? :)
04:42.45Qwellrussellb: "That'll be $125".
04:43.16JunK-Yblitzrage: your mother is 10$ :P
04:43.22Iaxyhehehe.. I know....
04:43.23russellbbut since I helped you in the middle of the night ... you owe me $50.  :D
04:43.56blitzrageJunK-Y: your mother owes *me* $10
04:46.07russellbI'm trying to update my resume, and it's very hard to explain what I do in words
04:46.33blitzragerussellb: it should just say, "What *don't* I do?"
04:46.39Iaxyrussellb: need a reference? :-)
04:46.50russellbIaxy: ha, no thanks :)
04:46.54Qwellheh, I think russellb has plenty of references
04:46.59blitzrageaye
04:47.17russellbIaxy: you can email my boss and tell him how grateful you are that I helped you :-p
04:47.22blitzrageok, I think I need to go read for 30 mins, then finally go to bed :)
04:47.47Iaxywill do.
04:48.00russellbblitzrage: g'night!
04:48.04Iaxyshopuld I tell him you charged me? ....hehe
04:48.15russellbbut I didn't!
04:48.21blitzragerussellb: night!
04:48.22Iaxy$50
04:48.28russellbi was kidding!
04:48.31blitzragelol
04:48.47*** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)
04:48.48Iaxyand so was I.
04:48.50russellbhehe
04:49.02MrDigitalrussellb: what do you do?
04:49.23russellbhonestly, I saw that you were using ABE, so I wanted to make sure you got everything resolved ASAP
04:49.29russellbotherwise, i probably would have been working on code :)
04:49.47blitzragewhat the heck... my asterisk seems like its ignoring modules.conf noload statements
04:49.48russellbMrDigital: I'm a full time student, but I also work for Digium
04:50.18IaxyI appreciate that. Its very noble of you.... thanks you
04:50.27CunningPikerussellb is who the users of 1.0 owe a tremendous debt of gratitude
04:50.36QwellCunningPike: and 1.2
04:50.37russellbCunningPike: :D
04:50.55CunningPike;) I've not forgotten, russellb
04:51.49CunningPike:)
04:51.52russellbit's cool, a lot of people still run the 1.0 series
04:52.12CunningPikeWe're not using it any more, but we did right up to 1.2.1
04:52.26russellbI don't touch it anymore, either
04:52.28CunningPikeAnd we are eternally grateful to you
04:52.37russellbthank you, very much.  I appreciate that
04:52.54QwellCunningPike: he likes free beer
04:53.03russellbQwell: :)
04:53.09CunningPikeNow we're on 1.2, you can esad ;)
04:53.13CunningPikej/k
04:53.14Qwellof course...he and blitzrage turned MINE down, but.. :P
04:54.46*** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)
04:54.46russellbCunningPike: when we started 1.2, we had a discussion and decided to make it a policy that bugs are fixed in 1.2 first, before the trunk
04:55.00russellbthat way, it's not just me coming behind and pulling them over by hand
04:55.19russellbthat has worked out really well, and now the release branch is a group maintained thing for the most part
04:55.27CunningPikeThat's great - and I think the quality shows through - we're loving 1.2, even more than 1.0
04:56.12russellbyeah, and i didn't have to stay up all night to catch up on commits at any point :/
04:56.20russellbI had a few of those nights last year ...
04:56.58dlynesrussellb: there's one bug fixed in trunk that's not fixed in release, afaik :)
04:57.04russellbdlynes: d'oh
04:57.14dlynesrussellb: but not asterisk....zaptel and libpri
04:57.25russellbwell that's not really my fault, heh
04:57.26russellbbut which one
04:57.28dlynesasterisk 1.2.7.1 for the most part is pretty stable
04:57.37dlynesI don't know who all it affects
04:57.45dlynesI would imagine not too many people
04:57.57russellbyou like that super-minor release number?  :)
04:58.21russellbthat means ... really refined ... or something
04:58.42dlynesBut for me, when I've got incoming calls on the pri, sometimes, the pri will hang, giving all incoming calls a busy signal, and giving all outgoing calls a busy signal
04:58.57dlynesI just figured why bother upgrading to 1.2.7, if 1.2.7.1 was otu
04:58.58russellbweird
04:59.11dlyneslibpri-trunk and zaptel-trunk from apr 24 fixed that problem for me
04:59.13*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
04:59.16russellband this doesn't occur running trunk?
04:59.30russellblibpri and zaptel trunk and 1.2 don't have many differences.
04:59.34russellbi actually thought they were still the same
04:59.47dlynesi'm too scared to try a newer trunk, for fear it might blow up my system even more than zaptel 1.2.5 did
04:59.59dlyneswell zaptel-1.2.5 with libpri 1.2.2
05:00.24russellbah, well there was a fix that went in very recently, yes
05:00.30russellbthat went into both the trunk and 1.2 branch
05:00.35russellbbut hasn't made it into a release tarball yet
05:01.07dlynesWell, i don't know if the problem was with zaptel, libpri, or both
05:01.23dlynesI just know at first the problem would hit me about once a week
05:01.28dlynesthen it was every couple of days
05:01.33dlynesand then it got to be once a day
05:01.49dlynesI couldn't deal with it anymore, so someone suggested grabbing trunk to me
05:05.14*** join/#asterisk LasaK (n=mypain@202.158.79.151)
05:05.23LasaKhi all
05:11.13*** part/#asterisk gursikh (n=guriskh1@adsl-68-95-82-50.dsl.hstntx.swbell.net)
05:11.54*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
05:14.47Iaxygotta run, Thanks again russellb .
05:16.01*** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr)
05:25.31*** join/#asterisk parag7732 (n=root@de2-b1835.alshamil.net.ae)
05:25.57parag7732I am a new bie to Free PBX...Can anybody let me know that Free PBX supports all the features of Digium Asterisk
05:28.29dlynesparag7732: FreePBX afaik, is Asterisk, with some management tools included
05:29.06dlynesparag7732: The most significant of which is AMP (which is now called freepbx)
05:29.44*** join/#asterisk Pageus (n=FreePBX2@ip70-190-19-6.ph.ph.cox.net)
05:30.14PageusHowdy all
05:30.47QwellNo, FreePBX *IS* AMP.
05:30.59Qwellnothing more, nothing less
05:31.17parag7732There are two features are given in the digium site Call Recording and Call snooping. What is the differnece ?
05:31.40Qwellparag7732: recording records it to disk, snooping, or ChanSpy, allows you to listen in on live calls
05:36.00parag7732what does " Blind Transfer " means
05:36.00parag7732??
05:36.16parag7732This is again a feature of asterisk
05:36.18Qwellwhen you transfer...blindly
05:36.21jqlit means you transfer the call without first talking to the person you're transfering to
05:36.31jql"attended" transfer would be when you do
05:36.40QwellI would have called it a deaf transfer, but that's just me
05:36.52*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)
05:38.08MikeJ[Laptop]the dumb transfer is transfering tothe guy sitting next to you :P
05:38.08*** join/#asterisk sergeus (n=s@195.112.98.13)
05:38.17dlynesblind transfer
05:38.33dlynesthat's what it's called in most pbxes and keysystems
05:38.35Qwellno, you know what would be cool...
05:38.37Qwellmute transfer
05:38.46Qwelltransfer a call, but force the callee to be muted. :P
05:38.53MikeJ[Laptop]hehe
05:39.21jqlsometimes, you need a pimp-slap transfer
05:39.45MikeJ[Laptop]I like when people call wrong number into our conference bridge, you ask the number, they tell you, you tell them, that's this number, but I am not who you were trying to reach... and they CALL BACK
05:41.20*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
05:42.09parag7732what does Call Queuing means???
05:42.21parag7732i mean in asterisk how it works??
05:42.39Qwellsame as any other PBX
05:43.41*** part/#asterisk parag7732 (n=root@de2-b1835.alshamil.net.ae)
05:43.56*** join/#asterisk parag7732 (n=root@de2-b1835.alshamil.net.ae)
05:44.14Qwell~root
05:44.15jboti guess root is not a Good Thing to use when using IRC. Please use a different account. You will probably not be able to speak until change your user account.
05:44.32MikeJ[Laptop]??
05:44.51*** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be)
05:45.07LasaKi had problem to get asterisk works with NAT issue
05:45.18dlynesMikeJ[Laptop]: parag's ircing as root
05:45.35MikeJ[Laptop]ahh
05:45.39MikeJ[Laptop]now I see it
05:46.13MikeJ[Laptop]wow.. state your question, then immdeiately leave
05:46.31roothmmm
05:46.57russellb/kick root
05:47.00russellbdang, just missed
05:47.03russellb:-p
05:47.17MikeJ[Laptop]heh
05:47.26MikeJ[Laptop]what, no one likes root?
05:47.35MikeJ[Laptop]I thought everyone like root
05:47.36Qwellreal men use...
05:48.16jqlsudo bash
05:48.24Qwellsudo su -
05:48.55dlynesNow that's just sudo cool
05:49.04*** join/#asterisk austinnichols101 (i=austinni@dsl-10-169.cofs.net)
05:49.11luke-jr_sudo ./x --login
05:49.12russellbsudo -s
05:49.36Qwellrussellb: yeah, that works too
05:49.48Qwelland you keep cwd
05:49.56russellb:)
05:50.02Qwellexcept, no root $PATH, right?
05:50.04luke-jr_sudo visudo
05:50.11*** join/#asterisk downunder33 (n=robert@219.95.158.235)
05:50.15luke-jr_Qwell: erm, bash --login shouldn't change cwd ...
05:50.17Qwellhmm, I guess you do get path
05:50.36Qwellrussellb: neat
05:50.37luke-jr_tsurukikun postfix # bash --login
05:50.37luke-jr_tsurukikun postfix #
05:51.12Qwell/bin/ls: /bin/ls: cannot execute binary file
05:51.13Qwellheh
05:51.15Qwellsudo -s ls
05:52.33downunder33hi all.  Is there an announce list for asterisk security related bulletins?  thx
05:52.56Qwelldownunder33: closest thing, is the svn-commits or asterisk-dev list
05:53.16*** part/#asterisk Alystair (n=bob@CPE001109c15241-CM00407b8794db.cpe.net.cable.rogers.com)
05:54.12parag7732I have just installed asterisk@home 2.8 with free pbx......But it dosn't support GUI. So can i install all the rpms of Cent OS 4.3 now. So that i can get the GUI
05:54.27downunder33thx qwell.  So, there is no standard protocol for announcing security related issues or patches?
05:54.49Qwelldownunder33: pretty much just the -dev list, or they're sent straight (privately) to Digium
05:55.15dlynesdownunder33: http://search.securityfocus.com/swsearch?sbm=%2F&metaname=alldoc&query=asterisk&x=0&y=0
05:55.25russellbwe'd prefer private communication first ...
05:57.17CunningPikerussellb: Have you ever seen really high size-32 usage with * running?
05:57.45russellbi believe i'm too tired to think right now
05:58.08Qwellrussellb: sneak off to bed - I won't tell
05:58.59*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
05:59.25PageusHey guys, I'm having some echo trouble.  It seems to echo a bit when a call comes through on my T1.. but my other connections are fine..
05:59.25dlynesHas anyone been able to get spandsp 0.0.3 to work with asterisk 1.2.x?
05:59.42Pageusi have echo canceling turned on.. and the echotraining set to 400..
06:05.18CunningPikePageus - which EC are you using?
06:05.32Pageusdigium t1000p
06:05.49Pageussoftware i think.. since it's not built into the card
06:06.01Pageuscouldn't get the additional funds for the better card
06:07.05*** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be)
06:07.29Pageust100p
06:07.30Pageussorry
06:07.34Pageuslong weekend.
06:10.16CunningPikePageus: Which software EC - look in zconfig.h
06:12.09Pageussec
06:14.21dlynesCunningPike: cdr pivottable reports, as in Excel?
06:14.26CunningPikeYes
06:14.31CunningPikeMy head hurts :)
06:14.36dlynesah
06:14.56CunningPikeWe import our CDR into MSSQL and join it with our employee database
06:15.13CunningPikeWorks out OK, actually
06:15.17dlynesah...and then spit out the data from mssql into excel for a report?
06:15.49CunningPikeYes - I create a PivotTable with the MSSQL view as an external data source
06:16.24CunningPikeLets people slice and dice themselves without me having to create upmteen reports
06:16.50CunningPikeI got your email, thanks - reply will be forthcoming shortly :)
06:16.58dlynesnod
06:17.36CunningPikeYou still there, Pag
06:17.39CunningPikePageus:
06:17.58*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
06:18.40Pageusyeah was looking through the file
06:18.53Pageusit looks like all the echo canceling is commented out
06:18.55Pageushow odd
06:19.06*** join/#asterisk lorinc (n=ang@caracas-2758.adsl.interware.hu)
06:19.23Pageusoh wait
06:19.34Pageus#define ECHO_CAN_KB1
06:19.37Pageuswas burried
06:19.38Pageuslol
06:20.27CunningPikePageus: OK, try MG2 - we had better results with it
06:20.37Pageusok
06:20.41CunningPikeIf that doesn't work, consider an external hardware EC
06:20.46Pageusso uncomment that one comment the other one..
06:20.59CunningPikeYes - make clean; make; make install
06:21.06Pageusyeah when i use my usr USB skype phone it seems to work fine
06:21.15*** join/#asterisk rdgzt (n=joakim@201.137.86.15)
06:21.22CunningPikeIs that on your PRI?
06:21.24Pageusbut that isn't an option for the rest of my office
06:21.26Pageusno
06:21.28Pageusno pri
06:21.28CunningPikeOK
06:21.31PageusT1 voice..
06:21.37PageusE&m Wink
06:21.40CunningPikeOh
06:21.45dlynesSo did you ever get it to work properly, Pageus?
06:21.55rdgztLoading the wctdm and wcfxo modules don't really seem to find my TDM400P card.
06:22.02rdgztEven though it's showing up in lspci.
06:22.12dlynesrdgzt: wctdm would be the one you need, not wcfxo
06:22.34dlynesrdgzt: make sure you've got your card defined properly in zaptel.conf, too
06:22.59dlynesrdgzt: how many fxo ports, and how many fxs ports do you have?
06:23.20Pageusyeah.. the system came online this morning with no issues.. as long as i was running hard phones i had no echo issues
06:23.23rdgztdlynes: I have 4 FXOs.
06:23.30_VileHI
06:23.44dlynesrdgzt: make sure you have a line like:  fxsks=4 in your zaptel.conf file then
06:24.00dlynesrdgzt: fxo ports use fxs signalling, not fxo signalling
06:24.04Pageuscept they sent my fax did to my system and i didn't have it setup for it.. so it caused my * to loop uncontrolled which kept anything else from coming in
06:24.17CunningPikeGood night all
06:24.19Pageusok now for the stupid question..
06:24.19dlynescool
06:24.22dlynesNight, cp
06:24.23_Vilegoing to get some food finally ttyl
06:24.32Pageushow do i recompile that part of the system
06:24.33Pageuslol
06:24.35CunningPikeNight, dlynes
06:24.57dlynesPageus: recompile?  what for?
06:25.04Pageusok so i don't have to?
06:25.08dlynesPageus: That sounds like an extensions.conf misconfiguration to me
06:25.41dlynesPageus: Check your log to see if it's dropping an incoming call into a non-existent extension
06:25.45Pageusno.. the system wasn't setup for fax in the extensions.. but it was setup in the main system.. it was never supposed to take faxes.. we have an alternative efax service for it
06:25.49Pageusyeah
06:26.16rdgztdlynes: Now I get ZT_CHANCONFIG failed on channel 1: No such device or address (6)
06:26.24rdgztWhen I modprobe wctdm.
06:26.27luke-jr_Pageus: so rm the fax module?
06:26.41dlynesSo, if you have a fax come in on that number, just do a Dial(Zap/g1/xxxxxxxx) where xxxxxxxx is the number for your efax server
06:26.43Pageusnever installed it
06:26.44Pageuslol
06:27.15dlynesPageus: Asterisk doesn't have any problems bridging two zaptel channels for delivering faxes
06:27.27Pageusconsidering the PC handles it all.. i wasn;t worried.. took them all of 2 mins to change it
06:27.29rdgztdlynes: That means it doesn't find the card, I assume.
06:27.33dlynesIt's not the best way to deliver faxes, tying up two lines
06:27.40dlynesbut, at least your faxes get through
06:27.48*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
06:27.55luke-jr_are PRI T1s the same price range as data T1s?
06:28.01Pageusok so when i made the change in the header file it should automaticly take
06:28.58dlynesrdgzt: Yeah...I'm getting weird ass sh*t like that, too, with an x100p card
06:29.23dlynesI'm going to be trying it in a different machine
06:29.25rdgztdlynes: It's worth noting that this used to work with older drivers. Maybe I should downgrade?
06:29.39dlynesrdgzt: which version are you using?
06:29.46rdgztThis is 1.2.5.
06:30.26*** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be)
06:30.33*** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-224-92.claranet.co.uk)
06:30.57dlynesyeah...that's the one i'm having problems with, too
06:31.00rdgztdlynes: I tried with the drivers that come packaged with Ubuntu, which are 1.2.1, a few days ago, and those work.
06:31.04rdgztWell, worked.
06:31.10dlynessame crap happens with zaptel-trunk
06:31.18rdgztI'm considering going to the svn version, seeing if that works better.
06:31.24dlynesnope :)
06:31.29dlyneslike i said...i was trying that :)
06:31.31dlynessame problem
06:31.37rdgztHeh, ok, I'm going to try downgrading, then.
06:31.57rdgztI'll go one version at a time back until it works, let's see...
06:33.51*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
06:34.16rdgztOk, 1.2.4 still fails the same way, it seems.
06:34.24dlynesMaybe 1.2.1? :)
06:34.37*** join/#asterisk MGSsancho (n=user@adsl-67-125-156-130.dsl.irvnca.pacbell.net)
06:35.05rdgztAs I said, I'll go one version at a time backwards.
06:35.34rdgztAnd we'll see if I don't end up with 1.2.1 anyway. :)
06:35.39*** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek)
06:35.49dlynesLemme know how you make out :)
06:35.57dlynesThis problem is pissing me off, too
06:36.10dlynesI was just going to reformat and reinstall on a different machine to see if that fixed it
06:36.23dlynesThe driver was working in that machine for about 2 days
06:36.24rdgztYeah, it's pretty quick testing this, I'll keep you updated. :)
06:36.29dlynesand then i shut it off for about 12 hours
06:36.31dlynesturned it back on
06:36.36dlynesand all hell broke loose
06:37.01rdgztWhoa.
06:37.06rdgztThat sounds like fun.
06:37.33dlynesah...i take it you have the driver versions all downloaded and compiled already?
06:37.48rdgztI'm downloading and compiling as I test.
06:37.52dlynesah
06:37.53rdgzt1.2.3 was no good either.
06:37.53dlyneshehe
06:38.15*** join/#asterisk Givur (i=anwi73@p54BC8EFB.dip0.t-ipconnect.de)
06:38.34rdgztDo specific versions of Asterisk require specific versions of Zaptel?
06:38.42Zeeekya
06:38.52dlynesrdgzt: sometimes
06:38.56dlynesrdgzt: but not always
06:38.57rdgztBecause I'd like to run the most recent asterisk version, to solve a problem I had, will I be able to do that even if I run, say, zaptel 1.2.1?
06:39.12dlynesrdgzt: try it and see
06:39.15rdgztI guess we'll see. ;0
06:39.18rdgztUm, :)
06:39.25dlynesrdgzt: that's the only way you'll know
06:39.34Zeeekrdgzt the only caveat is that if you have problems, it's one more possible hidden cause
06:39.45dlynesrdgzt: certain versions of asterisk will only work with certain zaptel versions
06:39.52dlynesbut not every asterisk version is like that
06:40.03Zeeekbut certain others will work with certain others ;)
06:40.42rdgztZeeek: Well, given that newer versions of zaptel don't seem to detect my hardware at all, I don't have many options.
06:40.50Zeeekwhat hw ?
06:41.10rdgztTDM400P
06:41.14rdgztWith 4 FXOs.
06:41.29Zeeekthat's extremely odd!
06:41.38Zeeekwhat does Digium say?
06:41.39rdgztI agree.
06:41.42rdgztI have no idea.
06:41.55Zeeekyou didn't contact them?
06:42.03dlynesZeeek: I've got the same problem with an x100p card
06:42.15dlynesrdgzt: actually
06:42.17Zeeekthere must be thousands of people that have it then
06:42.22rdgztZeeek: No, I don't really have the time to go through customer support stuff right now.
06:42.29dlynesrdgzt: do you get the same problem when you try to load the ztdummy driver?
06:42.31rdgztI just need this to work.
06:42.36rdgztdlynes: Good question, I haven't tried.
06:42.40Zeeekhow can anyone fix the problem if they don't get input?
06:42.46dlynesIf you do, you've got the same damned problem I do
06:43.02dlynesZeeek: I havne't reported the problem yet, because I don't know what the problem is yet
06:43.29rdgztOk, confirmed.
06:43.35rdgzt1.2.1 works, later versions don't.
06:43.44dlynesAnd besides...I'm running a clone card which digium probably isn't too keen on making  run with asterisk
06:43.54Zeeekdlynes you should call Dig or email with the symptoms. THey may have a workaround
06:44.00rdgztOr, wait, maybe not, let me see...
06:44.09Zeeekoh, well then you can't call or write them
06:44.28dlynesZeeek: well, i've got another problem, too
06:44.29ZeeekI'm running 1.2 so I can't help you
06:44.38ZeeekI have two X100P and a TDM400P
06:44.39dlynesZeeek: and it's with a genuine digium x100p card
06:44.43rdgztNope, 1.2.1 doesn't work either.
06:44.50dlynespiece of crap doesn't get the caller id some fo the time
06:45.03rdgztThis is very weird.
06:45.24Zeeekcallerid can vary accoring to a lot of stuff both inside and outside the hw
06:45.34dlynesZeeek: how so?
06:45.38Zeeekbut I assume this problem JUST started
06:45.48russellbrdgzt: would you like me to take a look?
06:45.51dlynesnah...caller id problem was there all along
06:46.10dlynesthe problem with the driver not loading is something that started recently
06:46.13Zeeekwell then it can depend on your telco, the dialplan and you connections
06:46.19rdgztrussellb: Possibly, who are you? :)
06:46.22russellbrdgzt: i work for digium, by the way, not in support, though.
06:46.31Zeeekrussellb heh he doesn't have time for support ;)
06:46.39dlyneslol
06:46.42Zeeekbugmonger
06:46.44russellbrdgzt: i'm an asterisk developer ...
06:46.45rdgztrussellb: If you could, that'd be great.
06:46.47russellbthat should be sleeping
06:46.53russellbbut i'm not, so i might as well be useful
06:47.05russellbrdgzt: sure, just msg me the login info
06:47.05dlynesYeah...maybe if he's able to solve rdgzt's problem
06:47.10dlynesIt's probably the same as my problem
06:47.12rdgztI just need to figure out how to make you able to log in through the fw.
06:47.18rdgztHold on a sec.
06:47.21russellbk
06:47.48russellbi'm really not all that experienced with zaptel :)
06:48.00Zeeekhardware sucks anyway
06:48.01russellbbut ... i should be able to figure out the common stuff, heh
06:49.26*** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be)
06:50.16Zeeekhow's it going in general russellb? Life treating you ok?
06:50.20dlynesHeh...looks like JerJer's awake now
06:50.33ZeeekI won't ask him than question ;)
06:50.39russellbZeeek: just fine, thanks for asking.  :)  I'm in the middle of final exams, actually
06:50.42*** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie)
06:50.59*** join/#asterisk joelsolanki (n=joelsola@202.160.163.144)
06:51.28dlynesEvening, solanki
06:51.48joelsolankiGood evening daniel
06:51.49joelsolanki:)
06:53.43*** part/#asterisk joelsolanki (n=joelsola@202.160.163.144)
06:53.48dlynesfine!
06:53.49*** join/#asterisk joelsolanki (n=joelsola@202.160.163.144)
06:53.52dlynesbe that way, joel
06:54.12joelsolankihehe. i m on asteris-unregistered :(
06:54.26dlynesand so am i
06:54.35dlynesI didn't realize #asterisk existed?
06:54.41joelsolankibut y ..i m already registed on #asterisk
06:55.32joelsolankiNo probs. atleast i m able to talk
06:55.43*** join/#asterisk bzbw (n=wlwzhang@68-190-223-129.dhcp.mtpk.ca.charter.com)
06:55.53dlynesi dunno
06:56.06dlynesi join #asterisk and it always throws me into #asterisk-unregistered
06:56.32Zeeekur in asterisk now
06:56.37dlynesit seems #asterisk and #asterisk-unregistered are the same thing
06:56.47Zeeekfunny
06:57.10joelsolankioh ok.
06:57.12dlynesX-Chat [2.0.9]: dlynes @ herbert.freenode.net / #asterisk (+tncrf #asterisk-unregistered)
06:57.43joelsolankiis this due to xchat ?
06:57.49Zeeekreminds of a chat I once ran. I announced that every member had their own chat room. When they started inviting each other in to their private rooms, they figurered out it was all the same room, only the name  displayed was fifferent :)
06:58.20ZeeekAre you recognized before you join? If not that's why
06:58.32dlynesYeah, i'm recognized before i join
06:58.50dlynesMy nick is registered, and i autoregister on login
06:58.52Zeeekoh well
06:58.57dlynesafter i log in, i join asterisk
06:58.59*** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz)
06:59.13joelsolankiassword accepted - you are now recognized
06:59.14joelsolanki* services. sets mode +e joelsolanki
06:59.14joelsolanki-MemoServ- You have no new memos
06:59.20joelsolankime too same.
06:59.31dlynesi dunno
06:59.39dlynesif i was the server, i wouldn't recognize you
06:59.43joelsolankii guess me and dlynes both are using xchat in linux
06:59.52joelsolankiheheh :)
06:59.54dlynesi'm sure most people are, joel
07:00.34iceyphey guys, i'm trying to connect 2 asterisk systems together, all appears fine in show iax2 peers, but when trying to call between them i get an error like: chan_iax2.c:6786 socket_read: Rejected connect attempt from 60.234.x.x, who was trying to reach '99705580@'
07:00.40dlynesWell, Zeeek's using icechat
07:01.11dlynesiceyp: the context that you're trying to go into doesn't exist
07:01.27dlynesiceyp: I suspect it's an error in your dial command
07:01.28iceypbut it does ;/, do i need to add an exten => s
07:01.28iceyp?
07:01.29ZeeekOMG everyone will know I use Windows now!!!
07:01.44dlynesicechat's a windows client?
07:01.46dlynesnever knew
07:01.49dlynesdidn't care, either :)
07:02.06dlynesonly two windows clients i know are mirc and pirch
07:02.08iceyp[sip_incoming]
07:02.08iceypexten => 99705580,1,Dial(IAX2/dhodd@voip.unix.co.nz/0508888802,60,r)
07:02.18ZeeekWell it might run on Mac in Windoze mode
07:02.34dlynesyou mean in windows on a mac?
07:02.39Zeeekyeah
07:02.42dlynesdon't need to emulate windows anymore :)
07:03.13iceypdlynes can both sides be peer?
07:03.22*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
07:03.23Strom_Ciceyp: no
07:03.29Strom_Ciceyp: peer is for ourbound only
07:03.31Strom_Cer
07:03.34Strom_Coutbound
07:03.42Strom_Cuser == inbound
07:03.47Strom_Cfriend == both
07:04.04iceypahhh ok
07:04.09iceypthats probably my issue
07:04.55Strom_Cchange them both to friend and see what happens
07:05.00iceypMay  2 19:01:13 NOTICE[63559]: chan_iax2.c:6786 socket_read: Rejected connect attempt from 60.234.68.100, who was trying to reach '99705580@'
07:05.03iceypsame thing
07:05.47iceyp<PROTECTED>
07:05.52iceypfrom iax debug
07:05.57*** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be)
07:07.54joelsolankiI need some information regarding the pstn card. my requirement is i need 4 outgoing lines and 16 to 20 internal extensions ..so can any body recommend any card or solution ?
07:08.21Strom_Cjoelsolanki: if you want analog sets internally, use a tdm2400p
07:08.37iceypany ideas?
07:09.00Strom_Ciceyp: show me the relevant sections of iax.conf and extensions.conf
07:09.13Strom_Cuse pastebin
07:09.16Strom_C~pb
07:09.17jbotextra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
07:09.18iceypyep
07:10.33joelsolankiStrom_C: yes means i have 4 pstn lines from my telephone company and i want to setup 16 to 20 analog extensions. so is the tdm2400p accurate for it ?
07:10.52Strom_Cyes
07:10.57Strom_Cit's perfect
07:11.12iceyphttp://pastebin.ca/index.php
07:11.12Strom_Cyou can do a quad-FXO module and five quad-FXS modules
07:11.24Strom_Ciceyp: um, thats not the link to your code ;)
07:11.32iceypthe http://pastebin.ca/52797
07:11.34iceypsorry
07:12.24*** join/#asterisk CpuID2 (n=nathan@gentoo/contributor/cpuid)
07:12.32Strom_Ciceyp: and what's the dial line you're using on the second system?
07:12.50CpuID2anyone aware of if theres anything in zaptel-trunk that would make it incompatible with asterisk-1.2.x in any way?
07:13.12CpuID2its working fine atm, just wanted some feedback from others that have tried the same zap/ast combination
07:13.36iceypStrom_C that part appears to be working ... exten => 99705580,1,Dial(IAX2/dhodd@littledan/${EXTEN},60,r)
07:13.37Strom_Ciceyp: so you're dialing from the first system to the second system?
07:13.46iceypyes
07:13.46joelsolankiOk let me go through this tdm2400p card
07:13.50carrarjoelsoanki
07:13.55carrarTDM2451B PCI Card
07:14.03carrar20 fsx, 4 fxo
07:14.21Strom_Ciceyp: the string in brackets is what defines the username
07:14.43Strom_Cso for calls to the second box, you need to call littledan@(second_box_url)
07:14.50dlynesCpuID2: nah...i've been running the two together for about 2 months now
07:14.55CpuID2cool, np
07:14.58CpuID2no weird crashes or anything?
07:15.02CpuID2pretty stable?
07:15.05joelsolankicarrar: oh is that digium card ?
07:15.10dlynesCpuID2: on the contrary...for me, it was more stable than zaptel 1.2.5
07:15.14CpuID2ive only been using it maybe 3 days or so, with 2 tdm400p's
07:15.16iceypStrom_C littledan@HOSTNAME ?
07:15.19CpuID2nice :)
07:15.22CpuID2cool, good to hear that
07:15.36dlynesCpuID2: but that's just me....zaptel 1.2.5 was extremely unstable for me
07:15.40CpuID2i think im gonna switch another installation to it actually
07:15.40Strom_Ciceyp: yes, because your second box's IAX2 config is called [littledan]
07:15.50CpuID2zaptel-1.2.5 was usable here, but i think ive found trunk better
07:16.08dlynesCpuID2: yeah, for me, zaptel-1.2.5 was putting my pri card in an unusable state
07:16.14CpuID2from what i heard, a lot better echo cancellation in trunk atm (with the default cancellation)
07:16.15iceypumm so ...  exten => 99705580,1,Dial(IAX2/littledan@60.234.x.x/${EXTEN},60,r)
07:16.16CpuID2ick
07:16.17dlynesCpuID2: the only way i was able to fix it was to reboot asterisk
07:16.25CpuID2thats a PITA
07:16.34CpuID2hmm i really should test out my iaxY again
07:16.40CpuID2see if i did actually kill it last time lol
07:16.47dlynesCpuID2: the default cancellation for trunk is still available in zaptel 1.2.5 as well...it's just not the default in 1.2.5
07:16.47Strom_Ciceyp: well, you have an entry in the firsy box's iax.conf already
07:16.54iceypbut its actually looking at iax config to get the host currently cuz it is getting to the system
07:17.01Strom_Cso it would be littledan@voip.unix.co.nz
07:17.08CpuID2dlynes, ah np, i wasnt sure if it was a new canceller or not
07:17.09Strom_Cor even better
07:17.13Strom_Cjust change user= to littledan
07:17.23Strom_Cand then remove the part before the @ in extensions.conf
07:17.42Strom_Cso you just dial IAX2/voip.unix.co.nz/whatever
07:17.50iceypStrom_C  exten => 99705580,1,Dial(IAX2/littledan/${EXTEN},60,r)
07:17.50Strom_Cand have the username in the iax.conf
07:17.51dlynesCpuID2: nah...don't know if it was in zaptel 1.2.4 or not...never checked
07:17.55dlynesbut it's definitely in 1.2.5
07:17.59Strom_Ciceyp: you're not listening to me
07:18.02CpuID2np
07:18.24iceypit's littledan not accepting the call not voip.unix.co.nz
07:18.34iceypvoip.unix.co.nz is passing the cll to littledan
07:18.43Strom_Ciceyp: yes I know
07:19.28Strom_Cwhen you have a string such as Dial(IAX2/dogballs/555), what happens is that your PBX looks up a section called "dogballs" in your iax.conf and dials 555 into whatever box that section specifies
07:21.52iceypStrom_C thanks, all working :)
07:22.09*** join/#asterisk shiznatix (n=shiznati@213-35-237-38-dsl.end.estpak.ee)
07:22.23dlynesWhat happens when it encounters monkeyballs?
07:22.27Strom_Cand with that I think I'm going for a bike ride
07:22.32*** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be)
07:22.42Strom_Cdlynes: well then it plays "The monkeys!  THE MONKEYS!"
07:22.52dlynesheh
07:29.30dlynes~seen coppice
07:29.35jbotcoppice <n=chatzill@153.192.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 16h 35m 59s ago, saying: 'the real downside is you can't keep doing it. the occassional dead meter is one thing, but failing between every reading it quite another'.
07:30.26rdgztIs there a flag somewhere in the Makefile to optimize asterisk for AMD K8?
07:30.38dlynesdoubtful
07:31.07dlynesbut you came back rather late
07:31.26dlynesI think russellb got tired of waiting for you :)
07:31.32rdgztNo, he helped me out.
07:31.35dlynesah
07:31.40dlynesso did you get the driver loading, then?
07:31.40rdgztMy problem was pretty simple, I just had to power-cycle the box.
07:31.45dlynesoh
07:31.52dlynesyeah...my problem is bigger than that
07:31.58dlynespower cycle didn't help
07:32.00rdgztThe card didn't like having the drivers changed from underneath it several times without power cycling, it seemed.
07:32.04dlynesi've power cycled like 5 or 6 times
07:32.15rdgztOk, probably not the same thing, then.
07:32.38rdgztI'm wondering if I can use the PPro optimization flag on K8, maybe.
07:40.02joelsolankidoes echo cancellation required while buying tdm2400 ?
07:41.47joelsolankiany idea
07:41.54*** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be)
07:42.20dlynesyou can get hardware cancellation for it?
07:42.39joelsolankiuhh whats that ?
07:43.35dlynesWell, you asked if echo cancellation was required when buying a tdm2400
07:43.49dlynesYou didn't ask about after you buy it...you asked when you buy it
07:46.06joelsolankiok :(
07:47.48dlynesI've never heard of a hardware canceller for the tdm2400, but htat doesn't mean there isn't one
07:47.52dlynesit's a pretty new card
07:48.20dlynesbut if there is a hardware canceller for it, that'd be a better bet than using hte software one built into zaptel
07:50.16Strom_Cdlynes: there is a cancellation module for the 2400
07:50.28dlynesjoelsolanki: there ya go
07:50.35Strom_Cjoelsolanki: how far away are you from the CO?
07:53.59joelsolankiCO ?
07:54.03dlynesCentral Office
07:54.03joelsolankiwhat is CO ?
07:54.06dlynesswitching station
07:54.28joelsolankiI dont understand u.
07:54.33Strom_Cjoelsolanki: the CO is the telephone company's switch
07:54.52Strom_Cjoelsolanki: it's where your dial tone comes from
07:54.55dlynesWhere all the telco company's switches are centralized for your phone number prefeix
07:54.58joelsolankiwe have been provided 4 analog pstn lines.
07:55.09Strom_Cjoelsolanki: how long are those lines?
07:55.36joelsolankii can extend it anywhere.
07:55.41Strom_Cno no no
07:55.59joelsolankilong means ?
07:56.00Strom_Chow many feet of copper wire are between your premises and the telephone company's switching equipment?
07:56.28dlynesjoelsolanki: long means what length?
07:56.35joelsolankioh it is in my building only.
07:56.40joelsolankimeans its very near.
07:56.46dlynesStrom_C: English isn't his native language; he's in India
07:56.50Strom_Cjoelsolanki: the telephone company's switch is downstairs?
07:57.11joelsolankiyes it in downstairs
07:57.31Strom_Cyou live in the telephome company building? :O
07:57.47joelsolankino
07:58.01Strom_Cjoelsolanki: where is the telephone company building?
07:58.26joelsolankiits far way appx 9 kms
07:58.36joelsolankiy are u asking this questions ?
07:58.43*** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be)
07:58.49x869km?!
07:58.55Strom_Cjoelsolanki: if you have a long telephone line, you will have a greater need for hardware echo cancellation
07:58.56joelsolankiyes
07:59.11x869km is very far ;)
07:59.11Strom_Cjoelsolanki: is that the building where the telephone company's equipment is located?
07:59.15joelsolankioh ok got it.
07:59.19x86i bet your LBO is crazy ;)
07:59.23joelsolankiyes
07:59.33Strom_Cjoelsolanki: so the switch is not downstairs
07:59.36joelsolankii got your point Strom_C
07:59.38Strom_Cjoelsolanki: the switch is 9km away
07:59.51joelsolankithey have keep some switch in the downstair of our building.
08:00.07joelsolankifrom there we have got our 4 pstn lines
08:00.11Strom_Cwhat kind of switch?
08:00.37joelsolankioh not familiar with it. it has lot of wires and thing in it :)
08:00.47Strom_Cjoelsolanki: is it just wire?
08:00.53Strom_Cis it just a room with lots of wire in it?
08:01.05joelsolankino wires connect to some switch.
08:01.27dlynesby switch, do you mean a white board with a lot of terminals sticking out of it?
08:01.34*** join/#asterisk MstlyHrmls (n=mh@melbourne.mostly-harmless.ca)
08:01.51joelsolankihang on. let me ask someone and give u clear information.
08:01.53rdgztAny ideas why, when registering a softphone via SIP, I get May  2 03:01:48 NOTICE[14256]: chan_sip.c:10886 handle_request_register: Registration from 'john <sip:john@10.0.254.35>' failed for '10.0.2.2' - Username/auth name mismatch
08:02.08dlynesbecause your username and password don't match
08:02.26Strom_Crdgzt: it helps if you read the error message
08:03.00rdgztI'm reading it, but I've seen before that if the password is actually wrong, I'll get a more specific message.
08:03.19x86uh
08:03.19x86no
08:03.20x86;)
08:03.40dlynesnah...that's the message you'll get
08:03.47dlynesit's pretty clear
08:04.05joelsolankiIn india we called it DP. junction box.
08:04.16joelsolankiwhere all pysical cables gets connected.
08:04.17rdgztOk, so what if I know that the password is correct?
08:04.20dlynesYeah...that's not a switch, joel
08:04.28joelsolankioh
08:04.29Strom_Cjoelsolanki: junction box is just a splice point
08:04.32dlynesThat's just called demarc
08:04.38joelsolankiok
08:04.40Strom_Ca switch is an actual piece of electronic equipment
08:04.46*** join/#asterisk darkskiez (n=darkskie@194.247.78.146)
08:04.52Strom_Ca fairly large one at that
08:05.02joelsolankihmm ok. got your point
08:05.13dlynesUsually a specialized computer specifically for handling tens of thousands of conversations at a time
08:05.16joelsolankiso i assume i need echo cancellation.
08:05.18rdgztBecause I've checked and double-checked the password on both ends.
08:05.24rdgztSo I know I'm entering it correctly.
08:05.34rdgztAny other reasons I might be getting that error message?
08:05.54dlynesYour username?
08:06.21dlynesYou're trying to authenticate using an auth?
08:06.57dlynesrdgzt: What kind of sip device is this that you're trying to connect?
08:07.05rdgztI don't know what "trying to authenticate using an auth" means.
08:07.07rdgztIt's a softphone.
08:07.12rdgztWell, I've tried two, both do the same thing.
08:07.26joelsolankiok going from lunch. will be back in 20 mins.
08:07.50dlynesSome softphones and/or sip phones ask you for a username, password, authorization name
08:08.02dlynesYou don't need to specify the authorization name for asterisk
08:08.06dlynesIt just confuses things
08:08.10rdgztOk...
08:08.33x86i always specify both with asterisk
08:08.40x86and asterisk does handle both of them
08:08.40rdgztOk, I removed that, I had it set, doesn't seem to make a difference, though.
08:08.48dlynesDoes your sip context look like [john]?
08:09.02rdgztYeah.
08:09.13dlynesrdgzt: do you have a username= set?
08:09.19rdgztIt's a very basic one, I'm following the Asterisk: Future of Telephony book.
08:09.24rdgztI did, but removing it makes no difference either.
08:09.47dlynesrdgzt: and do you have a secret= set?
08:09.48rdgztIt just has the secret now, no diff.
08:09.49rdgztYes.
08:10.04rdgztAnd I've confirmed that I typed the secret correctly in the client.
08:10.22dlynesand do you have auth=md5?
08:10.57rdgztIn the sip context?
08:11.02dlynescorrect
08:11.03rdgztNo?
08:11.06dlynesgood
08:11.06rdgztDo I need that?
08:11.09rdgztOk.
08:11.10dlynesno
08:11.20dlynesbut that could mess things up
08:11.33rdgztIt contains just:
08:11.34rdgzttype=friend
08:11.34rdgztsecret=welcome
08:11.34rdgztqualify=yes      ; Qualify peer is no more than 2000 ms away
08:11.34rdgztnat=no           ; This phone is not natted
08:11.34rdgzthost=dynamic     ; This device registers with us
08:11.38rdgztcanreinvite=no   ; Asterisk by default tries to redirect
08:11.40rdgztcontext=internal ; the internal context controls what we can do
08:11.49dlynes~pb
08:11.50jbothmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
08:12.12rdgztYeah, I know, figured it was short enough that it wouldn't matter much.
08:12.26rdgztNot like there's a ton of activity right now, but sorry if that was out of line.
08:12.40dlyneswell, this shouldn't affect the authentication
08:12.58dlynesbut qualify is usually set to a numeric value
08:13.01dlynessuch as 2000
08:13.25rdgztYeah, according to the book, yes means the same as 2000, as it's the default.
08:13.29rdgztBut I can set it to 2000.
08:13.40dlynesbut, as for authentication, it's probably a username or password mismatch
08:13.51dlynesif you turn on sip debug
08:13.58dlynesyou can at least see if your username is correct
08:14.07*** join/#asterisk saftsack (n=saftsack@p54A7FC22.dip.t-dialin.net)
08:14.09dlynesthe password will be encrypted, though
08:14.39rdgztFrom: <sip:john@10.0.254.35>;tag=1032e54d-21d8-da11-8234-0011098d4c7d
08:14.42rdgztEtc.?
08:14.48dlynescorrect
08:14.52dlynesyou'll get an error there somewhere
08:14.57dlynestake a look at the error
08:15.04dlynesand look at the context that it's within
08:15.11dlynesi.e. look at the surrounding text
08:15.19dlynesYou'll probably see the answer to your problems
08:15.32rdgztWell, it seems it gets that request, and sends a response.
08:15.38dlynesexactly
08:15.41*** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be)
08:15.43dlyneslook at your request
08:15.48dlynesand then look at the response
08:15.50rdgztSIP/2.0 404 Not found
08:15.53rdgztWhat's up with that?
08:15.55dlynesthere ya go
08:16.01dlynesyou probably forgot to do sip reload
08:16.24rdgztNope, I've actually restarted asterisk completely several times.
08:16.38dlynesuse pastebin, and paste your sip log, then
08:16.40rdgztHowever, can I list the sip contexts on the asterisk console with settings, to see that it's loaded?
08:16.55dlynessip show peer john
08:17.10*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
08:17.10rdgztPeer john not found.
08:17.11rdgztInteresting.
08:17.13dlynesthere ya go
08:17.15dlynesnot loaded
08:17.25dlynesimagine that :)
08:17.46rdgztSo, it's in my sip.conf, and I've restarted asterisk, is asterisk loading its config from somewhere different than I think, I wonder?
08:17.58dlynes/etc/asterisk/sip.conf?
08:18.17rdgztYeah, although I think I just figured out what's going on.
08:18.32rdgztAsterisk doesn't handle install prefixes like autoconf programs do.
08:18.38*** join/#asterisk janekm (n=janek@host81-157-239-71.range81-157.btcentralplus.com)
08:18.49x86err
08:18.52dlyneslook at /etc/asterisk/asterisk.conf
08:18.53x86asterisk uses autoconf...
08:18.58x86wtf are you talking about? :P
08:19.05dlynesx86: asterisk-trunk uses autoconf
08:19.06rdgztIf you give it /opt/asterisk as a install prefix, it puts its config in /opt/asterisk/etc/
08:19.10dlynesand it's a bastardized autoconf
08:19.27rdgztUnlike autoconf programs that use config_prefix (I think) for that.
08:19.28dlynesrdgzt: correct
08:19.30rdgztAnyway, let me see.
08:19.38x86rdgzt: "prefix" and "sysconfdir" are completely different though ;)
08:19.39rdgztThat probably just fixed my problem.
08:19.41rdgztRight, sysconfdir.
08:19.58dlynesx86: Yeah, prefix sets INSTALL_PREFIX atm
08:19.59rdgztx86: Non-autoconf asterisk thinks the install prefix is for everything.
08:20.18dlynesx86: russellb's aware of the problem, and he's working on a solution
08:20.50*** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135)
08:20.53x86dlynes: right, and SYSCONFDIR is appended to INSTALL_PREFIX... standard way of life with autoconf
08:21.03dlynesx86: Yeah, but that's not standard
08:21.12x86sure it is
08:21.26x86the standard behaviour of autoconf, anyway
08:21.29dlynesbecause with standard autoconf, i can do make install INSTALL_PREFIX=/usr/local/src/staging, and it will actually stage
08:21.41dlyneswith asterisk, it totally fubars everything
08:21.43janekmHi everyone, I've been playing with originating calls from asterisk (using the call file interface).
08:21.52x86dlynes: i see what you're saying
08:22.07janekmFor my experiments I've been connecting the calls to the demo extension...
08:22.10dlynesIf you're not doing staging, the asterisk autoconf is fine, though
08:22.17x86dlynes: i thought you were talking about an autoconf issue, not automake ;)
08:22.33janekmBut I haven't figured out a way to have asterisk drop its side of the call if the called party hangs up?
08:22.50janekmHaven't had much luck googling it but then perhaps I don't know the right terms yet...
08:22.51dlynesYeah, but that's just it...asterisk doesn't do a standard autoconf...you have to use ./bootloader.sh or whatever it was called again
08:23.12x86there is no configure.in and configure.am ?
08:23.34dlynesx86: You try generating configure from those, and the install fubars, unless russellb's fixed it
08:23.44*** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81)
08:24.03dlynesit goes into the never-ending configure loop of hell
08:24.07x86dlynes: works fine with gentoo's patches ;)
08:24.22dlynesx86: are you talking about asterisk-trunk, or asterisk-1.2.7.1?
08:24.39x86asterisk 1.0.8 - 1.2.7.1
08:24.42dlynesah
08:24.47dlynesthen that's a gentoo thing
08:24.53dlynesasterisk doesn't come with autoconf
08:25.08x86hmm
08:25.08dlynes1.2.8 or 1.3 will be the first to have it
08:25.25dlynesand it's not gnu autoconf compliant, either
08:26.09dlynesjanekm: it should do it automatically
08:26.24x86you're right... it is a gentoo thing
08:26.25x86heh
08:26.30x86never knew that :P
08:26.47dlynesI use slackware, so I always compile the original sources
08:27.04dlynesi just compile them once, and then deploy the binary packages everywhere
08:27.05x86gentoo compiles the original sources too, but adds some cool patches ;)
08:27.23x86same here... compile once, deploy binary
08:27.48dlynesgentoo doesn't use some horrid packaging method like rpm does it?
08:27.55snittasterisk on gentoo? comeon, 1.0.8 is the stable afaik
08:27.56*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
08:28.02x86no way, it's source-based, like BSD's ports
08:28.13dlynesyou said you deployed binaries
08:28.16x86snitt: they have 1.2.7.1 in overlay... that's what i'm running
08:28.18dlynesbinaries aren't source
08:28.19janekmdlynes: it hasn't done so far... It would keep the connection open (the called party was able to pick the phone off the hook again and still had it running)
08:28.24x86dlynes: uh, right ;)
08:28.34janekmperhaps some of the changes I made to the demo extension could have screwed it up?
08:28.35x86dlynes: you compile the source and get a binary ;)
08:28.36dlynesso what format are the deployed binaries?
08:28.43x86dlynes: ah... tar/bz2
08:28.51dlynesah...so gentoo doesn't support packages, then, per se
08:28.52snittx86: i've compiled 1.2.7.1 for gentoo by hand
08:29.02snittfor overlay..
08:29.05snitti dunno
08:29.10snitti dont trust it simply
08:29.24dlynesjanekm: no idea...but then again, i've only used call files for faxing
08:29.25x86dlynes: sure it does
08:29.26*** join/#asterisk mikl (n=mikl@pdpc/supporter/active/mikl)
08:29.28*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
08:29.42x86dlynes: all an RPM is, is a tar/gz with a preinstall and postinstall script ;)
08:30.01miklerr, when connected to an asterisk server in ekiga, how do I call out to "real" numbers?
08:30.03dlynesx86: then why is it I can never find any documentation that tells me how to make one?  only how to install one?
08:30.20x86dlynes: because you've never dug into the man page of 'rpm' ?
08:30.50rdgztOk, so now I have the sip context loaded and whatnot. Now, asterisk says: May  2 03:30:10 NOTICE[14353]: chan_sip.c:10886 handle_request_register: Registration from '<sip:john@10.0.254.35>' failed for '10.0.2.2' - Wrong password
08:30.50dlynesummm
08:30.51dlynesyeah i have
08:30.56dlynesit's not terribly useful
08:31.07x86dlynes: it explains how to create an RPM heh
08:31.10dlynesit mentions rpmbuild
08:31.13dlynesand that's about it
08:31.14x86(check out -b, for example)
08:31.19x86--build
08:31.28dlynespattern not found
08:31.43rdgztWhich is even more weird, since I'm again pretty sure the password is correct on both ends.
08:32.00dlynesYeah...no --build or -b switch
08:32.16dlynesrdgzt: Your password is incorrect
08:32.21janekmdlynes: I guess with faxes asterisk does know for sure when the transmission finished though... I guess I'll have to try it with sip debug sometime to see if asterisk actually gets any notification that the called party hung up
08:32.30rdgztNow, in the SIP request, there's a mention of algorithm=md5, is that what's messing me up?
08:32.32Zeeekyou gotta love a SIP phone that reboots in 5 seconds!
08:32.37rdgztdlynes: No, it's really not.
08:32.57dlynesrdgzt: No, algorithm=md5 is normal for sip...you just don't specify it in your sip.conf file
08:33.06*** join/#asterisk opus_ (n=opus@68.216.187.60)
08:33.08*** part/#asterisk opus_ (n=opus@68.216.187.60)
08:33.19dlynesopus must be getting bored
08:33.24dlynesnobody talking to him in asterisk-dev
08:33.47rdgztI'm 100% sure the password in the client matches the secret in sip.conf.
08:33.51rdgztSo what else could be tripping me up?
08:33.59dlynesrdgzt: sip debug
08:34.14x86dlynes: dont you love repeating yourself? :P
08:34.26rdgztNot that useful, SIP/2.0 403 Forbidden (Bad auth)
08:34.29rdgztIs the response.
08:34.31dlynesx86: Well, it's pretty obvious what the problem is
08:34.43dlynesrdgzt: Your password is incorrect
08:34.55dlynesrdgzt: Try retyping your password into your sip client
08:34.55rdgztI've retyped the secret in sip.conf, and in the client, restarted asterisk, repeat 3 times.
08:35.02rdgztSo no, the password is *not* incorrect.
08:35.04x86dlynes: hehe, you told him like 50 times and he still asks ;)
08:35.06rdgztI'm not an idiot.
08:35.07x86dlynes: it's kinda funny ;)
08:35.18dlynesWell, that's the only explanation for that error message
08:35.34dlynesIf it's not the truth, there's something fubared with your install
08:35.59dlynesAnd one can only guess what's wrong then
08:36.11rdgztThat's interesting, but doesn't really get me that far.
08:36.15x86rdgzt: look at your extconfig.conf... it's not pointing to mysql for anything is it?
08:36.47rdgztIt's the standard example file, looks like everything's commented out.
08:36.49dlynesx86: i doubt it...he's gotten this far...he's not getting a userid/password mismatch anymore...it's just a password mismatch now
08:37.11x86true
08:37.35dlynesi suspect he's probably including one context into another
08:37.42rdgztThing is, this exact problem happened to me two days ago, and was why I went from the prebuilt binaries on Ubuntu to installing from source, since I wasn't at the latest version.
08:37.47rdgztdlynes: How do I tell?
08:37.50dlynesand not realizing it because a include => is covered up by a #include
08:37.55dlynesor something similar
08:38.07rdgztAre we talking extensions.conf here?
08:38.15dlynesi was giving an example
08:38.24dlynesfor sip.conf, the only thing you could be doing is doing a #include
08:38.41dlynesand in that case, you probably have a sip context defined where you're not expecting one
08:39.23dlynesIf all of your sip clients are defined in one big sip.conf file, there's no guessing
08:39.36dlynesbut otoh, it makes it harder to manage then
08:40.02rdgztLet me try to cut down my sip.conf, it's basically just the sample file with this one context added at the end.
08:40.16x86realtime > sip.conf, imho
08:40.18dlynesyeah...get rid of all the sample cruft
08:40.38dlynesx86: yeah, but if you're having troubles with sip.conf, you're going to have real troubles with realtime, i woudl imagine
08:41.14dlynesbetter to get the simple case working first
08:41.20dlynesand then gradually go more advanced
08:43.02rdgztOk, so with a clean sip.conf, the same thing happens.
08:43.06*** join/#asterisk oej (n=oej@apollo.webway.se)
08:43.19rdgztClean defined as a general section and then my context.
08:44.40dlynessip show peers
08:44.43dlynesWhat do you get?
08:44.53rdgztHm, wait, I think I solved it.
08:45.19rdgztRealm defaults to "asterisk", while the client was trying to use the server's IP as the realm.
08:45.31rdgztNot the most obvious thing to get a "wrong password" error in that case, perhaps.
08:46.09dlynesheh
08:46.15dlynesfirst time i've heard of that problem
08:46.25dlynesevery client i've tried using just ignores the realm
08:46.38rdgztI'd argue that that error message is plain wrong. But well, I got it working, which is the important thing.
08:46.39miklargh, I can't dial out :(
08:46.49*** join/#asterisk |cleric| (n=dacleric@p54821028.dip0.t-ipconnect.de)
08:46.53miklhow do you configure Ekiga to work with asterisk?
08:46.56*** join/#asterisk Tangent (n=Tangent@connerdata.plus.com)
08:47.13dlyneswould help if you could explain what ekiga is
08:47.20saftsacknew gnome meeting
08:47.28dlynesah yeah...i forgot
08:47.35dlyneswhy'd they change the name, anyways?
08:47.39dlynesJust to confuse us?
08:47.44saftsacki think so ^^
08:48.02mikldlynes: no idea - guess gnomemeeting wasn't sexy enough :)
08:48.11dlynesmikl: don't you just configure it as a sip client?
08:48.28saftsackyes i would guess too
08:49.15mikldlynes: yes, I can connect to the server, but I can't dial out - If i try to enter a normal phone number in the field, I just get "Abnormal call termination"
08:49.32dlynesI would check the asterisk logs
08:49.39dlynesthe client isn't gonna tell you why
08:52.10*** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it)
08:52.13miklmeh :(
08:52.28miklthe server isn't mine, so i can't really check the logs :(
08:53.12dlynestell them to fix their server then
08:53.28dlynesthe only other problem it could be probably is a codec mismatch
08:53.42dlynesthey might be trying to get you to use a codec you don't have
08:53.46dlynessuch as g729
08:53.51miklah :/
08:53.52*** join/#asterisk bintut (n=bintut@202.164.162.222)
08:54.05dlyneswhich in that case
08:54.12dlynesis not a very good error message
08:54.29mikltrue
08:56.02janekmmikl: just to check the obvious, do you know if that server can dial out to the same number using a different client?
08:56.21mikljanekm: I've tried in twinkle, that works fine
08:56.50*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no)
08:57.12dlynesmikl: yeah, so it's probably either your username/password, or your codec settings then
08:58.11dlynesanyways...back to coding
09:05.27*** join/#asterisk littlejohn (n=little@host146-255.pool8289.interbusiness.it)
09:07.50*** join/#asterisk KPax (n=ask@droid.chaosmedia.org)
09:08.47*** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135)
09:09.43*** join/#asterisk Assaf (n=Drake@62.90.49.96)
09:11.40*** join/#asterisk Assaf (n=Drake@62.90.49.96)
09:13.11dlynesman...just when i want to get something done on the server
09:13.22dlynessome clown logs in wanting to read three weeks worth of voicemail
09:13.26janekmmikl: I imagine that the folks in the ekiga channel might know a bit more about those sorts of things, have you tried there?
09:13.34janekmdlynes: heh ;)
09:14.34shiznatixwhen a phone call comes in on a zapata card how can you tell what number that phone line has...like the number that the caller dialed. ${CALLERID(num)} does not work
09:14.41dlyneswhat's he doing in at his office at 2am, anyways?
09:15.06dlynes${CALLERIDNUM}
09:15.16dlynesif neither one of those work,
09:15.26dlynesa) that caller is 'unknown caller'
09:15.38dlynesb) your line doesn't have caller id subscription service
09:15.46shiznatixno i dont want the caller's number, i want my number
09:15.56dlynesoh yeah...nvm
09:16.00dlynes${DNID}
09:16.06dlynesor ${EXTEN}
09:16.09dlynesheh
09:16.13dlynestoo late
09:16.24shiznatixi have 4 lines coming in to a zapata card all on the same context. they are all fax lines. i want to organize my faxes based off number which is easy but i just need the number of the line to do that
09:17.21*** join/#asterisk L|NUX (n=linux@202.5.145.56)
09:17.22*** join/#asterisk RoyK (n=roy@80.239.107.70)
09:17.23dlynesexten => 6042663001,1,... exten => 6042663002,1,... exten => 6042663003,1,... exten => 6042663004,1,....
09:17.40dlynesshiznatix: is that not how you're handling them?
09:17.55jqlyou want the actual channel number? ${CHANNEL}?
09:18.11dlynesyeah, or what jql suggested
09:18.18RoyKtag
09:18.46shiznatixdlynes, no im doing it like this
09:18.57shiznatixdlynes, s,1,Dial(...
09:19.33shiznatixand it has to stay like that because people are idiots and this is being shipped to many different places so the numbers will change
09:19.47dlynesshiznatix: try jql's suggestion then
09:20.11jqlyeah, people do tend to be idiots
09:20.16dlynesshiznatix: but i think it might actually be more like ${DESTCHANNEL} or something
09:20.50OloBolawhy is callerID not always supported?
09:20.56dlynesok, or not
09:21.14jqlbecause callerid isn't required to complete a call. sadly
09:21.16dlynesseems kinda silly destchannel, but it's not specifying whether it's the called channel, or the calling channel
09:21.26KPaxlo all, is it okay to ask a few basic questions ? i've read some docs and installation manuals but there's a few things i'd like to clear up
09:21.30janekmOloBola: Some phone companies like to charge extra to send that info to your line...
09:21.39dlynesKPax: don't ask if it's ok to ask, just ask
09:21.56OloBolasucks butt
09:22.29janekm(or, of course, some people just don't want you to know their number so you can't complain about their agressive sales tactics...)
09:23.12dlynesor their aggressive collections tactics
09:24.39KPaxk, well i'd like to know if asterisk config can be stored in a database (mysql would be better), i've read that mysql support was removed for license reasons but it was still available thru some addons, unfortunately i can't really find a clear tutorial on how to enable mysql support.. asterisk install from source worked nice on my debian sarge.
09:24.59*** join/#asterisk salviadud (n=noyb@dsl-201-129-86-188.prod-infinitum.com.mx)
09:25.12salviadud!seen _paulo_
09:25.28bintutwhat are the basic features for an asterisk voip and pbx that are easy to configure to beginners?  :)
09:25.42salviadudsip and iax chans
09:25.53salviadudand um... a default context that dials out
09:26.01salviadudto FWD for example
09:26.07salviadudthat's what i did as a newbie
09:26.10dlynes~seen _paulo_
09:26.16jbot_paulo_ <n=pirch@201-13-16-73.dsl.telesp.net.br> was last seen on IRC in channel #asterisk, 13d 13h 40m 48s ago, saying: 'Why dont Borland made Delphi with C++?'.
09:27.25salviaduddlynes, you know paulo?
09:27.25dlynesnope
09:27.26*** join/#asterisk v3rmap (n=puser@unaffiliated/v3rmap)
09:27.26dlynesi was just showing you it's '~seen', not '!seen'
09:27.26KPaxwhen i look for mysql stuff i usually end up on some AMP tutorials, should i start from there ?
09:27.26salviadudi got a private message with !seen
09:27.26salviadudlook
09:27.26salviadud[04:25] -GerbilWrk- [LAST SEEN] "_paulo_" - n=pirch@201-13-16-73.dsl.telesp.net.br Quit 1wk 6days 11hrs 54mins 54secs ago Last Note: private/secret channel Quit: (freenode) Remote closed the connection
09:27.30dlynesah
09:27.40salviadudlots of ways i guess
09:28.10salviadudi need to find that dude.
09:28.26salviadudmy prank calls depend on him
09:28.48salviadudi feel so proud, i do my pranks with asterisk
09:28.58v3rmapMy asterisk installation on Ubuntu is not listening on port 5060, so sip phones can't login. Any suggestions on what could be wrong with my installation??
09:29.12salviadudv3rmap, iptables?
09:29.19salviadudbtw, i hate ubuntu...
09:29.23dlynesKPax: http://www.google.com/custom?tk=d86a5ba922fc092a368c&domains=www.voip-info.org&q=mysql&sitesearch=www.voip-info.org&sa=Google+Search&client=pub-6210650267389726&forid=1&ie=ISO-8859-1&oe=ISO-8859-1&cof=GALT%3A%23008000%3BGL%3A1%3BDIV%3A%23336699%3BVLC%3A663399%3BAH%3Acenter%3BBGC%3AE9ECEF%3BLBGC%3AFFFFFF%3BALC%3A0000FF%3BLC%3A0000FF%3BT%3A000000%3BGFNT%3A0000FF%3BGIMP%3A0000FF%3BLH%3A20%3BLW%3A100%3BL%3Ahttp%3A%2F%2Fwww.voip-info.org%2Fimages%2FVOI
09:29.24dlynesP-info.jpg%3BS%3Ahttp%3A%2F%2Fwww.voip-info.org%3BLP%3A1%3BFORID%3A1%3B&hl=en
09:29.27dlynesack
09:30.02dlynesanyways...append the 'P-info.jpg' to the '2FVOI', KPax
09:30.52v3rmapsalviadud: iptables --list shows this: http://pastebin.com/693802 I hope this means I have no active firewall?
09:30.58dlynesv3rmap: edit your /etc/asterisk/modules.conf file so that you have a line:  load => chan_sip.so
09:31.13janekmdlynes: the first part of the URL seems to work too
09:31.24dlynesv3rmap: it has nothing to do with your firewall
09:31.40v3rmapdlynes I'll check that ...just a min.
09:31.43dlynescool
09:31.44salviadudyeah, looks like you're open like a pornstar
09:32.07dlynesnah...even pornstars aren't that open
09:32.16v3rmaphehe
09:32.53salviadudyou know, back in the day... i used to trade pr0n
09:33.01salviadudbut NOOOOW
09:33.12dlynesnow you keep it all to yourself
09:33.19salviadudthey got their fingle fangled kazaa's and emules
09:33.26janekmnow you spend your time prank calling people with asterisk instead? ;)
09:33.45salviadudand here on irc, no more trades, just leech
09:33.53salviadudi dunno if that's good or bad
09:33.57dlynesyou suck
09:34.19salviadudugh? elaborate
09:34.25dlynesyou said you were a leech
09:34.27dlynesleeches suck
09:34.44salviadudof course, i suck the pr0n off the fserves
09:35.02dlynesi'd hate to see what else you suck \;)
09:35.07salviadudi miss the trades
09:35.29salviadudi suck suculent women, which i also prank call
09:35.54janekmsalviadud: have you tried the ratio-based bittorrent sites? or directconnect? Might satisfy some of the nostalgic feelings...
09:36.28salviadudit's not the same... i liked the social engineering bit
09:36.34salviadudasking around
09:37.18janekmthat's true, the personal connection to the pusher is really lacking...
09:37.26*** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it)
09:37.57salviadudyeah, you see, if i found out some dude had lots of... fat girl vids
09:38.06salviadudthen, i would have this really big conversation
09:38.20salviadudlike "dude, they're really big... why would you like 'em big?"
09:38.40salviadudand he would go "cause i'm a skinny ass dude, i need some meat"
09:38.40janekmperhaps #empornium might amuse?
09:38.45salviadudyou know, funny stuff
09:38.49janekmtrue
09:39.07salviadudempornium? is that here on freenode?
09:39.24janekmirc.whatnet.org apparently
09:39.41salviadudi consider freenode to be a very subtle network, full of smart people and devs
09:39.57janekmhaven't been there actually, just imagine it might be amusing since that's a fairly community-oriented torrent site...
09:40.15janekmhence you probably won't find as much pr0n amusement on freenode ;)
09:40.32v3rmapdlynes, I added "load => chan_sip.so" in modules.conf and restarted asterisk. The login from the sip phone still timeouts and "netstat -a | grep 5060" returns nothing. Anything else I could try?
09:40.40JohnnieYou can always start.
09:40.49JohnnieA little T&A never hurt.
09:41.02salviadudyou see, it's not about the pr0n, broadband has made leeching cheap
09:41.04JohnnieYou can liberate FreeNode.
09:41.33salviadudT&A?
09:41.40salviadudyou mean R&R?
09:41.45janekmhttp://www.urbandictionary.com/define.php?term=t+%26+a
09:41.59janekmhad to look it up too ;)
09:42.40Johnniehahaha
09:43.31salviadudanybody else like to prankcall?
09:43.42*** join/#asterisk faljse (n=martin@213.235.245.210)
09:43.43salviadudi speak 3 different languages
09:43.51salviadudand 2 kinds of spanish
09:43.52janekmlooks like they're playing trivia on #empornium so maybe it's not that amusing after all ;)
09:44.24faljseis there a way i can limit the call duration(just the billsec time...)?
09:44.27janekmmexican and, eh, spanish spanish?
09:44.43salviadudyeah, castellano
09:44.46salviadudmaybe i can do more
09:44.51salviadudargentinian spanish
09:44.53salviaduddominican spanish
09:45.03salviadudit's freakin easy
09:45.09salviadudi used to be an international operator
09:45.13janekmah, right
09:45.24salviadudi used to be a monkey, taking calls all day
09:45.43janekmsalviadud: I was sort of prank calling people yesterday trying to get originate to work ;)
09:45.59salviadudyeah, that's the spirit
09:46.09salviadudyou prank while you test, win/win
09:46.43janekmwas at a friend's place while I'm waiting for BT to fix my landline...
09:47.28*** part/#asterisk bintut (n=bintut@202.164.162.222)
09:47.28janekmthey were meant to enable ADSL on the line and instead cross connected my incoming number to some little old scottish lady...
09:47.53salviadudwhere are you from janekm?
09:47.57janekmwell I suppose I don't know if she's little that's just my imagination...
09:48.03janekmIn Scotland
09:48.16salviadudi can do british english too
09:48.22salviadudi find it impossible to do pikey
09:48.23janekmoh, sorry I'm from Germany but have lived in Scotland for 10 years so I guess I get confused ;)
09:48.35salviadudgermany, very interesting
09:49.02salviadudi suppose i'll have to speak german sooner o later
09:49.29salviadudi can say basic stuff
09:49.43salviadudich bin studentin
09:49.57salviadudich heisse Salviadude
09:50.08janekmit's a horrible language to try and learn... I always get really lost when friends ask me about something grammatical and I can't coherently explain how it works ;)
09:50.16salviadudheir ist der schlangemann
09:50.36salviadudwell, i got this book, learn german in 40 lessons
09:50.43*** join/#asterisk psk (n=psk@golia.caltanet.it)
09:50.45salviadudi found it very english-like
09:50.54salviadudlots of similarities
09:51.14salviadudyet, i was only reading it, i would need to prank call more germans to get the hang of it
09:51.40salviadudwell, i'm gonna go to sleep
09:51.45salviadudgot work in the mornin'
09:51.45janekmthere are lots of similarities, but those are what trips people up with more complex things where it's different...
09:51.53salviadudnice talking to you, janekm
09:51.57janekmyou too
09:52.11*** part/#asterisk salviadud (n=noyb@dsl-201-129-86-188.prod-infinitum.com.mx)
09:54.30*** join/#asterisk apardo (n=apardo@87.217.145.111)
09:55.50*** join/#asterisk Greek-Boy (n=grb@193.220.93.162)
09:55.58*** join/#asterisk X-Gen (n=X-Gen@dsl-145-220-183.telkomadsl.co.za)
09:56.17X-Genhey freaks
09:56.44Greek-Boyany voip software for Nokia 6680?
10:04.09*** join/#asterisk niZon (n=ilt@S010600080db4ab60.wp.shawcable.net)
10:06.52Zeeekvas ist das?
10:09.13*** join/#asterisk MrEntropy (n=entropy@ppp142-239.lns2.adl2.internode.on.net)
10:09.13MrEntropyyo
10:10.20MrEntropycan i configure a hunt group in asterisk? i dont want to call all phones at once and whichever one picks up first wins, i want the first not-busy phone to ring exclusively from a list of phones.
10:10.58AhrimanesMrEntropy: hm queue with round robin strategy?
10:11.45Greek-Boydoesn't anybody know voip software that will work with symbian OS on 6680?
10:12.15AhrimanesGreek-Boy: hm i havent found any yet.. not for qtek 8310 either..
10:12.18MrEntropyAhrimanes: what config file is that configured in?
10:12.31X-GenGreek-Boy, write some, i dont think there is any
10:12.52starleinGreek-Boy: http://www.forum.nokia.com/main/0,6566,034-561,00.html
10:12.59AhrimanesMrEntropy: queues.conf
10:13.02starleingo and develope your own
10:14.03Greek-Boythanks
10:14.06Greek-Boylol
10:15.08dlynesX-Gen: there is some...they just suck
10:15.33Greek-Boydlynes how u doing
10:15.43dlynestired
10:15.45Ahrimanesbut nokia are launching their own voip clients on some of the new phones
10:15.45Greek-Boydo they work, even though they suck?
10:16.05dlyneswell, i woudl guess if they charge money for them, they must work
10:16.47Ahrimanesdont bet too much on that
10:17.01Greek-Boylol
10:17.03X-GenAhrimanes, SUCKER !!!
10:17.09AhrimanesX-Gen: :P
10:17.35Ahrimanesif only my homebanking software and such would work under freebsd..
10:17.58X-Geni suppose writing a SIP client in Java will really suck on a mobile device
10:19.32janekmI would expect so...
10:19.56janekmbut on the S60 mobiles you should be able to write it using the Psion API too I guess
10:19.58Ahrimaness/on a mobile device//
10:20.35janekmso you should be able to write it in C++
10:22.28janekmah I just read the link starlein posted, looks like Nokia already released a SIP plugin for the SDK
10:22.35v3rmapppl, I added "load => chan_sip.so" in modules.conf and restarted asterisk. The login from the sip phone still timeouts and "netstat -a | grep 5060" returns nothing. Anything else I could try?
10:22.57janekmso that should make it really easy, you'd have thought someone would have released a SIP phone for S60 based on that by now?
10:23.10v3rmap^^ That means asterisk is not listening on port 5060, any ideas whay that could be so?
10:24.23*** join/#asterisk apardo (n=apardo@87.217.145.111)
10:25.24*** join/#asterisk syle (n=blah@unaffiliated/syle)
10:25.26*** join/#asterisk backblue (n=igor@82.102.1.42)
10:25.33backbluemorning all
10:26.06Skid__question: is there a way to "ping" a sip phone, using some sip command?
10:26.28Skid__sip notify?
10:26.38Ahrimanessip options ?
10:26.39*** join/#asterisk r0d3nt|m (i=r0d3nt@tinfoilhat.net)
10:26.59Skid__I want to see if the phone is alive
10:27.39AhrimanesSkid__: hm for what purpose? are you using qualify=yes in asterisk?
10:29.17Skid__Ahrimanes: I want to make a call-forward if a phone is nologer reachable.
10:29.41*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
10:29.45Skid__the phone is a member of a queue
10:30.19Skid__and I dont want to use call agents
10:30.58AhrimanesSkid__: http://www.voip-info.org/wiki-Asterisk+call+forwarding <- *61* should be useful for you
10:34.32backblueppl, anyonw know something about dialing in block, or digit by digit, some convencional pbx (alcatel,siemens) do that... they dial digit by digit, how can i put asterisk receiving digit by digit?
10:34.33Skid__Ahrimanes: not that simple..
10:34.53AhrimanesSkid__: could be done with that
10:35.37Skid__Ahrimanes: I dont think you have the whole picture..
10:36.19AhrimanesSkid__: phone is in the queue, if it's not available, as in unreachable or offline, you want to forward the call somewhere else?
10:37.07syleif you use canreinvite, do you loose cdr record for talk time on that call?
10:37.43Skid__Ahrimanes: If both phones in the queue ar  not available or not reachable.. I want to forward all calls.. and not use the queue anymore
10:39.05AhrimanesSkid__: then add exten => 1234,N,Dial(SIP/otherphone) after exten => 1234,N,Queue()... if noone in the queue can answer it will jump to the next priority
10:39.32Skid__Ahrimanes: It dont work.
10:39.51Skid__becase I dont use agents
10:40.08AhrimanesSkid__: i use dynamic members in the queues and it works for me
10:41.30Skid__Ahrimanes: well..  Is it possible to autamatic logon an sip client to the queue?
10:42.31*** join/#asterisk ivanfm (n=ivanfm@201-27-67-81.dsl.telesp.net.br)
10:42.34AhrimanesSkid__: yes i have setup a button on a snom phone that dials an agi that uses cmd add queue member SIP/1234 to queue queue1 fx
10:43.50Skid__Skid__: that is not automatic enough :(
10:43.58Skid__oups
10:44.16AhrimanesSkid__: you want to add the sip client as soon as it registers?
10:44.25shiznatixim trying to get the phone number of the zapata line that the call is coming in on. i have tried these ${DNID}-${CHANNEL}-${DIALEDPEERNUMBER} but none of them are the number of the line. how can i get the number of the line?
10:44.40Skid__Ahrimanes: rigtht.
10:45.02AhrimanesSkid__: i dont believe this is currently possible.. but i have some ideas on how to do it.. will require some code though
10:45.45ZeeekBeer is currently possible though
10:46.33Skid__Ahrimanes: for the moment I have a static entry for every phene in the queue.
10:46.39AhrimanesZeeek: hehe
10:46.42*** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be)
10:47.10*** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com)
10:47.10AhrimanesSkid__: hm ok, did you look at levewhenempty and related options in queues.conf ?
10:47.21Skid__Ahrimanes: yeah.
10:47.45Skid__Ahrimanes: "If you have any static queue members defined as "Member => Agent/XXXX" in your Queue definiation, Asterisk will considered the Queue in-use"
10:48.38AhrimanesSkid__: ah yes
10:48.59*** join/#asterisk rkr245 (n=ravi@office.callsat-telecom.com)
10:49.19AhrimanesSkid__: my idea on the automatic way: listen on the manager interface for register, unreachable and logoff events and have a script add and remove queue members based on those events
10:49.49Ahrimanesusing add queue member and remove queue member
10:50.43Skid__Ahrimanes: Is there possible to run a script extensions.conf ?
10:51.07AhrimanesSkid__: hm no, but you can run commands via the manager interface
10:51.08Skid__that check status for any phone in the queue
10:51.39AhrimanesSkid__: i think reading up on the manager interface would help you
10:51.53Skid__Ahrimanes: looks like so.:(
10:52.29AhrimanesSkid__: it should be doable in something like 100 lines of perl
10:52.52backbluehow can i put asterisk waiting for X number of digits, when some zap channel it's dialing?
10:53.06Skid__Ahrimanes: but.. to many ways of things going wrong:(
10:53.25AhrimanesSkid__: hm how?
10:56.07AhrimanesSkid__: i've been doing perl interaction with the manager interface for a long time now, only problems i had were hardware related
10:57.11Skid__Ahrimanes: well, I havent.. and It must work at 100% today.
10:58.55AhrimanesSkid__: hm, then i'd contact digium and have a large amount of money at hand
10:59.54Skid__Ahrimanes: well I dont:)
11:00.00*** join/#asterisk AsteriskAlbania (n=info@217.24.244.130)
11:00.19AsteriskAlbaniaasterisk <-> Radius ? any help please
11:00.31Skid__Ahrimanes: tanx anyway.
11:00.42Ahrimanesnp
11:00.59AhrimanesAsteriskAlbania: what info do you want to get from/save to radius?
11:02.38AsteriskAlbaniaI need to nake asterisk talk to radius
11:02.44AsteriskAlbaniamake sorry
11:02.58AhrimanesAsteriskAlbania: something like http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth ?
11:03.19AsteriskAlbaniaAhrimanes: I have tried it but there is a patch for 1.0.9
11:03.42syleradius unreliable, why use it
11:04.04Ahrimanesradius is quite reliable if setup correctly like most things
11:04.07AsteriskAlbaniaI have a billing system with alepo
11:04.09sylefunny
11:04.25AsteriskAlbaniawhat do you think is the best way for billing
11:04.36sylei won;t even begin listing all the problems using radius
11:04.45sylemiss calls if you want
11:05.17sylebest is to embed directly into asterisk
11:05.32AsteriskAlbaniasyle: which is the best way for doing billing with asterisk
11:05.49Ahrimaneshm, so the current asterisk radius implementations are faulty.. but that's not radius' fault
11:05.50syleuse the existing c source and code c functions to do what you need
11:06.18AsteriskAlbaniasyle: I am not good at programming myself
11:07.12sylewell that sucks, well use what you can i guess
11:07.17janekmyikes the nokia n80 is one expensive phone...
11:07.27AhrimanesAsteriskAlbania: http://www.paskambink.lt/mcc/index.php?option=com_content&task=view&id=77&Itemid=1 works for me
11:07.41AsteriskAlbanialet me check it
11:14.39*** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be)
11:15.09shiznatixim trying to get the phone number of the zapata line that the call is coming in on. i have tried these ${DNID}-${CHANNEL}-${DIALEDPEERNUMBER} but none of them are the number of the line. how can i get the number of the line?
11:16.10Ahrimanesshiznatix: you're matching the call on an exten => XX ?
11:16.42Skid__Ahrimanes: Hey.. It will work.. I connected to the manager interface and removed the static members of a queue..:)
11:17.09shiznatixAhrimanes, no, its all on the s extension. 4 lines share the same context and I need to split them up by the number of that phone line
11:17.22AhrimanesSkid__: cool :)
11:17.30AsteriskAlbaniaAhrimanes: Have you tested MCC
11:17.51AhrimanesAsteriskAlbania: yes, am currently running it on a test server and doing code changes to fit it to my needs
11:19.12Ahrimanesshiznatix: hm ok, usually that's what you use exten => for.. maybe ${EXTEN} will tell you the number though
11:19.15*** join/#asterisk razu_ (n=razu@tln-kontor.norby.ee)
11:20.01I-MODshiznatix: you should probably just make a macro that gets the number from the channel
11:30.46joelsolankiHi all. planning to buy tdm244E from digium. but i want to know which cable they will provide which i have to join on tdm244E card ? are they going to ship the cable too ?
11:32.36joelsolankisyle: u there /
11:32.56*** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be)
11:33.03*** join/#asterisk apardo (n=apardo@87.217.144.132)
11:33.23Ahrimanesjoelsolanki: call up digium and ask :)
11:33.47joelsolankihehe :)  i m poor with cards so dont know much about it. so thought to ask on list.
11:34.02Ahrimaneshehe
11:34.16Ahrimaneswell i'm sure digium can answer any and all questions about their products :)
11:35.10joelsolankiAnybody on list can answer my small question :(
11:37.22*** join/#asterisk coppice (n=chatzill@153.192.17.210.dyn.pacific.net.hk)
11:38.16MrChimpyjoel: if your question made sense you might get an answer
11:39.00Zeeekthe question is: what is small?
11:39.18AhrimanesZeeek: dont get me started
11:40.48joelsolankii m planning to buy tdm244E which has 4 FXS module and 2 FXO module. means i can have 16 FXS ports and 8 FXO ports.
11:41.31MrChimpycan't find 244E on digium's site
11:41.32joelsolankinow i want to know where do i plug the pstn lines. ?
11:42.20MrChimpyRTFM, dude
11:42.21Ahrimanesjoelsolanki: TDM2400P ?
11:42.30Zeeekbest to just call them
11:42.32joelsolankiyes
11:42.33joelsolankihttp://www.voipsupply.com/product_info.php?products_id=1152
11:42.39joelsolankiabove is the link.
11:42.41MrChimpyFXO/FXS stuff is all explained in the docs
11:42.45Zeeekcall voipsupply - or email Corey
11:42.55Zeeekhe's good about giving the answers
11:43.20joelsolankiwho is the email id of corey ?
11:43.35MrChimpyi'm sure corey will be pleased
11:44.02MrChimpyprobably want to get the product code right before asking
11:44.21joelsolankican anybody give me the email id of corey ?
11:45.44Zeeekthere's the interent - that's what it's for!
11:46.22MrChimpyhttp://www.voipsupply.com/product_info.php?&products_id=1164
11:46.27MrChimpyi suspect that is your answer
11:46.58Zeeekbe sure to buy the tech support option
11:47.00tzangerwhy on earth would you use one of those things
11:47.03MrChimpyhorrid idea using those things though
11:47.09tzangerterminate to BIX and be happy
11:47.25tzangerany office with more than a half dozen phones will have terminated to BIX anyway
11:47.26MrChimpyhorrid idea using fat analogue cards for starters
11:47.33joelsolankiYes i founded that :) ;)
11:47.43joelsolankiMrChimpy: thanks :)
11:48.10tzangerI mean hell they make pre-punched D50->BIX cables
11:48.18tzangerlike $3
11:52.18AsteriskAlbaniahave any installed asterisk on FC5
11:52.28AsteriskAlbaniaI mean DIGIUM CARDS
11:52.34Greek-Boywhat is BIX?
11:52.40AsteriskAlbaniaany problem ?
11:52.52ZeeekBIX was the name of one of my dad's dogs
11:53.05Greek-Boyseriously
11:53.10Greek-Boyfor us newbies
11:53.12Greek-Boywhat is BIX
11:55.25*** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br)
11:56.49Greek-Boyso whats the alternative to horid analogue cards?
11:58.05tzangeranalogue cards aren'tbad
11:58.21tzangerif you're going high density though take alook at direct digital (i.e. CAS T1 or PRI)
11:58.49mutman we used to be 'lax' in our adding late fees to customers
11:58.57mutever since we started doing it, it's like a cash machine
11:59.16muti can't believe how many ppl are past due all the time
11:59.46*** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it)
11:59.50Greek-Boywhat if the premises only has analogue lines available? what good will a digital card be?
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12:00.24coppice2 analogue pair == 1 E1/T1 :-)
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12:03.13Greek-Boylol
12:03.17Greek-Boythats not efficient
12:11.15MrChimpyi'd sooner go voip or use something else to do analogue->t1 than have buttloads of analogue lines coming into my asterisk box
12:11.57MrChimpyat least a T1/E1 card is more use later
12:13.31*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)
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12:18.35tzangerGreek-Boy: I'd still pull in everything into a good channel bank and go to the Asterisk box over a T1
12:18.49tzangeri.e. POTS -> Adit600 FXO -> T1 -> TE110P
12:19.18tzangerI like using the channel banks because they're a proven technoplogy. Just don't use CarrierAccess Access Bank I or II for FXO, they do not have CPD
12:19.24Greek-Boyhmmm
12:19.28Greek-Boyeven for a few lines?
12:19.53*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
12:20.00tzangerdefine 'few'
12:20.01*** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com)
12:20.48Greek-Boy4 lines
12:20.55tzangerGreek-Boy: use a TDM400 then
12:21.08tzanger4 lines is fine, you don't need to spend a pile on a TDM2400 for 4 lines
12:21.21*** join/#asterisk Ariel_ (n=Ariel@70.46.87.158)
12:23.44[TK]D-FenderGreek-Boy : How many lines can you forsee upgrading to?
12:23.51Greek-Boy16
12:24.20coppicesome bought a MiG21 for $24,730 on E-Bay. I guess that is a slight discount from the new price :-)
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12:24.43tzangeryeah see once you getabove about 8 lines I start recommending a channel bank
12:25.08[TK]D-FenderGreek-Boy : if you think of 8+ I'd suggest  trying to get a partial PRI.
12:25.19alib80hi all does anyone know how to get multiple instances of asterisk on different servers to write to the same cdr
12:25.43tzangeralib80: use a database
12:25.48tzangerI have 6 asterisk boxes dumping into one DB
12:25.57[TK]D-Fendercoppice : I'm Looking for an AH-64 for "recreational" use, keep an eye out for me, ok? ;)
12:26.01Greek-Boyok
12:26.09tzanger# select count(*) as "Total Calls",host as "Host" from cdr group by host;
12:26.09tzanger<PROTECTED>
12:26.09tzanger-------------+----------
12:26.09tzanger<PROTECTED>
12:26.09tzanger<PROTECTED>
12:26.10[TK]D-FenderGreek-Boy : Where are you located again?
12:26.11tzanger<PROTECTED>
12:26.13tzanger<PROTECTED>
12:26.16tzanger<PROTECTED>
12:26.18tzanger(5 rows)
12:26.21tzangersorry 5 boxes not 6
12:26.22[TK]D-FenderSPAM!!!!!
12:26.29alib80tzanger: ia m running mysql but2nd instance won't write to it
12:26.31[TK]D-FenderuNF!
12:26.35*** part/#asterisk pigpen2 (n=mark@fw.seamans.cc)
12:26.37tzanger[TK]D-Fender: that's flood, not spam :-)
12:26.49[TK]D-FenderSame shit, different marketing!
12:26.50alib80no error messages either
12:27.00coppice[TK]D-Fender I used to fly a radio controlled model of one :-)
12:27.01tzangeralib80: I use postgres,and have a trigger on the cdr table that injects the hostname
12:27.20Greek-Boyanother question; If i have 4 analogue lines but I want one number to always be available  how would i do that? Ie, one number is for four lines and if first one is busy it goes to second line. if second is busy it goes to third line, etc, etc.
12:27.31Greek-Boy[TK]D-Fender i'm from east africa :(
12:27.32Greek-Boylol
12:28.18alib80tzanger: would one be able to do the same with mysql. I must admit my db understanding is quite simple
12:28.18[TK]D-FenderGreek-Boy : What you are describing is called "line-hunting" and is offered by your telco as a base service across the lines you have with them.
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12:28.25tzangeralib80: I don't know, I refuse to support mysql
12:28.48coppiceline-hunting in east africa might have a different meaning :-)
12:28.57tzangerthat's *terrible*, coppice
12:29.20Greek-Boy[TK]D-Fender i suppose its never done on the pbx side?
12:29.25alib80tzanger: does the trigger basically mimic the same server or distinguish each server entry?
12:29.33coppiceit wasn't intended as a joke. have you tried to get lines in places liek that?
12:29.37tzangeryep
12:29.42tzangeralib80: it distinguishes the server
12:29.44tzangerlet me show you what I have
12:29.54alib80thanks
12:30.10[TK]D-FenderGreek-Boy : You can't MAKE line 1 fall over to line 2 when busy :)
12:30.20[TK]D-FenderGreek-Boy : Its a telco feature
12:30.53[TK]D-FenderGreek-Boy : And another advantage to PRI.  Another alternative would be to find a VoIP carrier that services your area.
12:31.23coppiceyou are more likely to be able to get MFC/R2 than PRI in east africa
12:32.46shiznatixare there any SIP fax machines?
12:33.44[TK]D-Fendercoppice : E1 varient?  I've seen that acronym before but never followed up on non-North American tech...
12:35.08[TK]D-Fendercoppice : What is the identifying factors & advantages to MFC/R2 over "standard" E1?
12:35.46coppicewhat is standard E1?
12:36.36[TK]D-Fendercoppice : Hmmm I suppose that would eb grey... ok, what is the TYPICALY signalling put over E1 (As PRI has become for T1)
12:37.21coppiceshiznatix: while T.38 is designed to talk directly to FAX machines, most machines with an RJ-45 do strange proprietary things
12:37.50coppicetypical signalling over T1 would be a list of about 20 things. same with E1
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12:38.56parag7732Linux people don't get a good job...then why they strugle too much
12:39.39coppicethe commonest things over E1 these days are EuroISDN and MFC/R2. In many countries ISDN is almost impossible to get, and you have to use MFC/R2
12:39.54Greek-Boywhat is MFC/R2?
12:40.00Greek-Boyanalogue line hunting?
12:40.20coppiceone nice thing about E1s is apart from a couple of weird places, like the UK, pretty much everywhere uses the same variant of ISDN.
12:40.33coppiceMFC/R2 is a signaling system for E1s
12:42.03Ariel_so why is there so many different signalling. MFR/R2 SS7 etc..
12:42.27tzangerthat's the great thing about standards... there's just so many to choose from!
12:42.40coppicewhy are there so many different OSes, linux, unix, window os/400, etc.?
12:43.09Ariel_I know it's just would have been allot easyer if we had only 1 or 2.
12:43.58coppicewell, things change over time. MFC/R2 dates back to the mid 50s, when ISDN was not viable
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12:45.23Greek-Boyso if the telco providers line hunting and I phone out on any of those lines will it show the main number on caller ID?
12:45.38tzangerok that's it... I'm registering the german pastebin.. pastenderbin.de
12:45.38[TK]D-FenderGreek-Boy : THAT is another feature
12:45.46coppicecaller ID varies from telco to telco
12:46.07tzangeralib80: http://pastebin.ca/52862 that's the trigger and function
12:46.22alib80thanks
12:46.53parag7732Okk i have one query
12:47.13parag7732Our company is running a traditional PBX system
12:47.34parag7732they want Voice mail and auto attender facility
12:47.40parag7732I suggested to go for asterisk
12:47.47alib80tzanger: did u have to setup postgres specifically to allow writing from different servers at the same time or does it do this by default.
12:47.48parag7732but they don't want to change
12:47.50[TK]D-Fenderparag7732 : How big is your current setup?
12:48.06tzangeralib80: they just access the DB as regular users with insert access on the table
12:48.08parag7732arround 300
12:48.10tzangernohting fancy at all
12:48.10parag7732users
12:48.28tzanger300 users is a fairly hefty investment in terms of hardware
12:48.36parag7732yaa thats why
12:48.42parag7732can i do one thing
12:48.54tzangerwhat are you suggesting they replace it with?
12:48.58[TK]D-Fenderparag7732 : You can do it in many case, but transferring calls back out of there can be challenging.  You'd want to like them by wat of a T1/E1 trunk
12:49.05alib80this is what stumps me is that i get no error msg's but when my sencond server tries to write nothing happens even though i tested it on another db
12:49.05parag7732No i suggested that I can integrate it
12:49.18parag7732with asterisk
12:49.18tzangerintegration is tricky
12:49.18tzangermost PBXes have no desire to do so
12:50.01tzangerI can get calls IN to my norstar very easily (assign a DID for each extension, then call that DID), and calls out are trivial, howeverhaving the norstar treat that call out as an extension call is next to impossible.
12:50.20tzangerthere is a protocol called MCDN which is used to tie together meridian systems but of course it's proprietary
12:50.29tzangerand q.sig does not work on the norstars in north america
12:50.54tzangerbasically if you can get a PRI connection into your PBX and use Q.Sig you're a great ways ahead ofthe game
12:51.26coppice1st law of telecoms: if the telcos didn't spec it, its proprietary :-)
12:52.56*** join/#asterisk pointer (i=pointer@aj.catt.com)
12:53.13parag7732in TDM400P with 1 fxo and 1 fxs.....Can i integrate 1 fxo with one incomming no. and out put 1 fxs to traditional pbx
12:53.28pointerparag7732: yes
12:53.56pointerparag7732: we do something similar to our pbx, but we use an FXO between * and the definity
12:55.24pointerparag7732: we're changing our config a bit now though... pstn pri-> * -> analog ports + pri -> definity -> another pri -> pstn
12:55.46*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)
12:56.37pointerafter poking around a bit, I have been unable to find a provider for free 800 termination that doesn't state that they have call time limits, known reliability issues, or service "holes" as it were...does anyone else know of a SIP/IAX provider that does free 800 termination?
12:57.50Hmmhesays3 days down, 2 to go
12:58.15Greek-Boyso with a PRI/E1 the call is digital all the way?
12:58.20Greek-Boyor still goes over analogue
12:58.31[TK]D-FenderGreek-Boy : E1 = digital
12:59.11Greek-Boyand PRI T1
12:59.12Greek-Boy?
13:00.00starleinthe same!
13:00.57pointerno, 8 extra ports!
13:01.05pointer24/32
13:01.12starleinyes but digital too
13:01.16pointerGreek-Boy: they're both digital
13:01.51Greek-BoyE1 is mostly used in europe, right?
13:01.58pointerPRI/T1 == 24;E1 = 32 digital channels, but in both cases you have a control channel, so it's actually 23 and 31 lines
13:02.07pointeryup, E1 -> .eu
13:02.42pointerthey've also got some pretty cool calling features that .us doesn't have, like overlapdial
13:03.00*** join/#asterisk Sonderblade (n=muh@host-213.131.147.169.addr.tdcsong.se)
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13:07.00Hmmhesaysoh I miss my guitar
13:07.18Hmmhesaysi woke up last night and I was playing air guitar
13:08.40MrChimpysure you weren't just playing your banjo string?
13:09.07Hmmhesaysno because I was also saying the words to backwater by the meat puppets
13:09.17Kattyhi lads.
13:09.41HmmhesaysKatty
13:09.56coppicepointer only 30 voice lines on an ISDN E1
13:10.57HmmhesaysOh hell yeah! check this out http://today.reuters.com/news/articlenews.aspx?type=filmNews&storyid=2006-05-02T083541Z_01_N02260638_RTRIDST_0_FILM-NIGHTMARE-DC.XML
13:13.00coppiceHmmhesays: Chicken Little could only possibly do good business because of the novelty of 3D. it really really sucked
13:13.16HmmhesaysI was talking about nightmare before christmas, that was a fantastic movie
13:13.59coppiceI was referring to the article. it says nothing about the potential for 3D audiences, because they tried such as lousy movie
13:14.26*** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135)
13:14.38starleinexactly there are 30 B channels + 2 D channels
13:15.07coppicestarlein: I wish people wouldn't make this stuff up as they go along. its 30 B + 1 D
13:15.12HmmhesaysAre there any decent hardware video phones out there?
13:15.13pointercoppice oh, is it 31 and not 32?
13:15.16starleinsorry thats true
13:15.23pointercoppice I stand corrected
13:15.35pointerthat's interesting
13:15.41pointerwhy 2 control channels?
13:16.00pointerI should finish reading before I type
13:16.22coppicethere aren't 2 control channels. there's one time slot used to create framing,; one is signalling; the other 30 are voice
13:16.45pointergotcha
13:16.46MikeJ[Laptop]coppice. or 31 if you are not using isdn
13:16.52MikeJ[Laptop];)
13:17.07pointerI've only worked with PRI here in the us, personally
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13:17.33[TK]D-Fendercoppice : Entirely agree on your movie critique
13:17.37coppice31 is rarely available. only things like SS7 ever use it, and then only rarely. when the channel isn't used as a D channel it is used to provide 30 CAS channels
13:17.38pointerand ds3/oc3,12,48
13:18.11Hmmhesayshas the gxv-3000 come out yet?
13:18.20MikeJ[Laptop]coppice, yah.. I was thinking of wasim's setup
13:18.42coppiceDisney got out of 2D animation, because their audiences sucked. this is entirely because their stories sucked, and now they are making 3D animation that sucks as badly
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13:19.41pointercoppice: do you expect anything good out of disney?
13:19.55pointercoppice: other than theme parks that kids like, that is
13:20.07coppiceHayao Miyazaki still gets *huge* audiences for 2D animation. maybe it because he's a story teller
13:20.20Hmmhesaysnightmware before christmas was a good movie
13:20.28Hmmhesaysbut it was all because of tim burton
13:20.43coppicewithout successful movies the theme parks are doomed. they feed entirely for the movies
13:20.48[TK]D-FenderTim Burton = GOLD
13:21.03coppiceTim Burton == quirky
13:21.03MrChimpyyeah. until you see corpse bride.
13:21.17coppicehe's made more stinkers than successes
13:21.23[TK]D-Fendercoppice : A sorely need trait in this homogenized world..
13:21.25HmmhesaysHim and quinten tarrention should make a movie
13:21.44[TK]D-FenderHmmhesays : Hard to picture that...
13:21.47pointerHmmhesays: ROTFL
13:21.50coppicecorpse bride isn't too bad. seen charlie and the chocolate factory? :-)
13:21.54HmmhesaysIt would be dark and weird
13:21.58MrChimpycharlie was ok!
13:21.59pointerHmmhesays: QT is my hero
13:22.12Hmmhesaysa vampires with scissor hands
13:22.14[TK]D-Fendercoppice : I loved that one personally, then again, its Depp & Burton together that made his successes
13:22.26coppicecharlie was terrible, compared the gene wilder one
13:22.46Hmmhesayshe would never surpass gene wilder
13:22.52Hmmhesaysno one would
13:23.07coppicethat guy looks sooooo old now
13:24.01MrChimpyis it normal to see occasional loads of the following on a E1 card (TE411 in this case)
13:24.07MrChimpy<PROTECTED>
13:24.07MrChimpy<PROTECTED>
13:24.07MrChimpy<PROTECTED>
13:24.09MrChimpyetc etc
13:24.10MrChimpy?#
13:24.11coppice(I have small kids - I only see kids movies there days :-) )
13:24.30MrChimpylooks like somethings renegociating. work though...
13:24.32*** join/#asterisk iulius (n=iulius@mail1.technologieshq.com)
13:24.32[TK]D-FenderMrChimpy : Yeah sometimes telcose do a forced bchan reset on interval/idle
13:24.38*** part/#asterisk pointer (i=pointer@aj.catt.com)
13:24.52MrChimpyah, cool. that'd be our switch then :)
13:25.43tzangerMrChimpy: asterisk is doing that
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13:25.47tzangerMrChimpy: it's totally normal
13:26.02tzangerMrChimpy: if you don't want it or if it's causing issues, play with resetinterval in zapata.conf
13:26.02MrChimpygood, good. I thought something was potentially broken :)
13:26.25MrChimpynot bothered as long as nothing is borken. thanks tz/tk
13:26.34[TK]D-FenderMrChimpy : First few times I saw it here scared the shit outta me given the problems we'd encountered earlier
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13:31.32sumaIs there is any IAX Wifi Phone Commercially available in the market?
13:31.59sumaor even firmware upgradable ?
13:32.57tzangersuma: no.
13:33.01[TK]D-Fendersuma : I don't think I've heard of an IAX2 Wi-Fi phone ANYWHERE....
13:33.19tzangerI want a wifi phone with bluetooth
13:33.24tzangerhaven't found one of those yet, either
13:33.34[TK]D-Fendertzafrir : UTStarcom PPC6700
13:33.40[TK]D-Fendertzanger rather...
13:34.04tzanger[TK]D-Fender: any better than their F1000Gs in terms of feel and flimsiness?
13:34.13[TK]D-Fendertzafrir : Wifi / CDMA / BT / PPC (WinCE 2003)
13:34.45tzanger[TK]D-Fender: I don't think those are actually shipping yet either
13:35.02tzangerand CDMA is *useless* if the carriers don't support it
13:35.02filethe 6700? yes it is
13:35.24tzangerI don't need a damn smartphone, just a damn wifi phone with BT.  no PPC, no fucking camera.. ugh  (general rant)
13:35.32bkw__haha
13:36.18blitzragetzanger: haven't seen you here in a while (was just thinking that last night :))
13:37.21sumaWill any wifi SIP Phone works behind NAT and the asterisk in public IP ?
13:37.49tzangerblitzrage: :-)
13:37.59tzangersuma: depends
13:38.07tzangersymmetric RTP is a glorious thing
13:38.33sumawithout symmetric RTP, i guess
13:38.51sumai can use IAX without any configuration or port forwarding
13:39.01[TK]D-Fendertzanger : Yeah, they're shipping, Telus is pimping them to me.
13:39.14tzanger[TK]D-Fender: telus is, really
13:39.33[TK]D-Fendertzanger : yup.  I was going to get one to eval here
13:42.11*** join/#asterisk pif (n=ldm@zenon.apartia.fr)
13:44.19blitzrage[TK]D-Fender: send me one too :)
13:45.52[TK]D-Fenderblitzrage : ! ! !
13:46.07blitzrageheheheheheheheheheheheheehehe
13:46.34[TK]D-Fenderblitzrage : Sorry, I'm putting them off for now since Bell may be able to offer me the same and we're with them already and there is a factor in having to change salesmen's phone #'s.  This project is "on hold" for a while/
13:47.18fileyeah... I wish we had number porting
13:48.50tzangeryeah
13:49.00tzangerapril 2007 is when LNP comes to mobile networks in Canada
13:49.59[TK]D-FenderMontreal and the surrounding region hits forced 10-digit dialing next month....
13:50.01blitzragew00t
13:50.13blitzrage[TK]D-Fender: welcome to Toronto :)
13:50.34[TK]D-FenderFollowing all of Toronto's biggest mistakes including forced mega-mergers
13:50.38*** join/#asterisk BadPacket (n=root@unaffiliated/badpacket)
13:51.25[TK]D-FenderI should move to BC.... little snow, and still far enough north :)
13:51.46[TK]D-FenderUnfortunately I still love this place for other reasons...
13:53.38fileit's the chinese food
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13:53.45fileit's poisoned your mind
13:53.52*** join/#asterisk gr0mit (n=w10277@dhcp4.zuk40.mot-tools.co.uk)
13:55.33[TK]D-Fenderfile : Then again, BC has a huge asian population and is better for imports also being closer to CA....
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13:57.22vader--have any of you guys had to provision a bulk of phones at one time?
13:57.40vader--i was looking to program a script to write the conf files and stuff but im a vb programmer not perl
13:57.52vader--wondering if there is any already writen utilities
13:58.56tzangervader--: polycom makes it nice
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14:01.18[TK]D-Fendervader-- : Easily scriptable.  Make a template and parse and extensions list and you can generate sip.conf & the provisioning files for them based on MAC
14:02.13[TK]D-Fendervader-- : What kind if "bulk" are we talking about anyways?
14:04.35Hmmhesayssnap into a slim jim
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14:06.26pifdoes a jitterbuffer make a big quality difference with iax?
14:07.18Hmmhesaysif you have a lot of jitter it sure does
14:07.48coppiceand if you have packet loss, PLC makes a big difference too
14:08.25pif"jitter" meaning variations in latency ?
14:09.21coppicewell, if doesn't mean a caller with a stutter
14:09.41Kattyunless it's natural stutter.
14:09.47Kattyin which case, they just need a speech therapist.
14:10.08Ahrimanesw-w-w-w-w-ha-a-t-s t-t-t-tha-t-t ?
14:10.23*** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.63.Dial1.SanJose1.Level3.net)
14:10.26Hmmhesaysgreat a huge thorn in my side is back
14:10.37*** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.63.Dial1.SanJose1.Level3.net)
14:10.40pifhi coppice, and what is the tradeoff? increased latency?
14:12.13coppicewell, a really good jitterbuffer balances packet loss and latency to maximise perceived quality. the one in *'s IAX is not as good as it could be. however, if you have jitter it is still a huge improvement
14:12.52*** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com)
14:13.25*** part/#asterisk parag7732 (n=root@de2-b15868.alshamil.net.ae)
14:13.39piffor instance a DSL connection going from ping 12ms to 100ms all the time, would be a good candidate?
14:13.54coppicesounds like it
14:14.15pifthanks, and google says you wrote the PLC code
14:14.28pifis that included in the jitterbuffer=yes ?
14:14.43coppiceyes
14:14.54pifthanks thanks thanks
14:16.15Kattyi could use a new car.
14:16.20sevard<PROTECTED>
14:16.40Ahrimanesi've had a 1001 that did it
14:16.51sevardwhat the hell is up with that
14:16.58Kattysevard: it needs a hug.
14:17.04sevardi'm just sitting here and it scares the crap out of me by giving me one ring
14:17.09Kattysevard: possibly a firmware update.
14:17.09MrChimpydunno, but it's a good way to pretend to be popular
14:17.13sevardKatty: i've tried hugging/raping it... doesn't work.
14:17.25Kattysevard: your hugs must suck then.
14:17.41sevardthey can't, i won a contest.
14:17.49Kattyfor strangling to death?
14:17.50MrChimpyi'm not sure if a rapey hug counts
14:18.06sevardKatty: for hugs, i haven't been entered into a raping contest although if you're up for it
14:18.06coppiceisn't sucking while hugging pretty normal?
14:18.35Kattysevard: you shouldn't joke about things like that.
14:18.39Kattysevard: you have no idea what people go through
14:18.43sevardKatty: the world needs jokes.
14:18.54MrChimpywhy not? 8 mile was a whole movie about raping contests
14:19.00MrChimpy:p
14:19.01sevard:)
14:19.05Kattysevard: but it's not a joke if it isn't funny.
14:19.19Kattysevard: if i were you, i'd never joke about raping a female.
14:19.20sevardKatty: only to you dear, I found it funny.
14:19.29Kattyit's just not polite.
14:20.00sevardKatty: I might have a hissy fit about irish catholic jokes but whenever somebody lays one down it gets me in stiches
14:20.19Kattywe're not all you, sevard
14:20.22MrChimpyand if it's ok to joke about male rape, well, that's just SEXIST AND WRONG
14:20.28sevardhow is that different than rape or black jokes? it's not.
14:20.41sevardKatty: you can't please them all honey.
14:20.45Kattywhy are we arguging about this?
14:20.49Kattywe have better things to argue about
14:21.07sevardthere's always somebody somewhere who is going to be offended by something. it's unavoidable.
14:21.13sevardKatty: not sure why you brought it up then.
14:21.20MrChimpysve: how dare you say that!
14:21.34Kattysevard: let's just move on.
14:21.35Nivexsevard: yeah, but there are some doozies that are pretty well known that can generally be avoided.
14:21.38Kattysevard: if you want the last word, then have it.
14:21.43sevardHeh
14:21.59*** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu)
14:22.03sevardLast words: cooooooooooooooooooooooooooooooooookie chrisp!
14:23.06Kattyright.
14:23.12Kattynow, i'd like to know something.
14:23.17*** join/#asterisk psk (n=psk@golia.caltanet.it)
14:23.27Kattywhy do lamborghinis always break down after a month if you're not driving them over 70mph all the time?
14:23.40Kattythere's something terribly wrong with this.
14:23.47sevardI'm not sure if that's true
14:23.47MrChimpybecause they're made by italians
14:23.56MrChimpythey go vroom, or not at all
14:24.05Kattysevard: it's true all right.
14:24.05MrChimpyit's the same with ducattis
14:24.12Kattyand they lag on the shirt
14:24.14Kattyi mean shift
14:24.31syzygybsdKatty: it sounds like there is an easy fix for that problem
14:24.34Kattytorque converter, i think.
14:24.37coppiceits better than BMWs. they break down every month whatever you do with them
14:24.51sevardI think sports cars run just fine at normal speeds people just want to drive them fast.
14:24.52Kattysyzygybsd: yeah, get soemthing else.
14:25.04syzygybsdI was thinking drive them faster
14:25.09Kattynot that i have a lamborghini.
14:25.18coppicewankel engines only work properly when driven fast
14:25.34Kattyactually, i'm trying to decide what my next car will be.
14:26.05Kattya coupe of some sort.
14:26.05sevardKatty: get a cadillac 16
14:26.09coppiceand you've discounted the lambourgini, due to reliability?
14:26.10Kattyhell no
14:26.17Kattycoppice: that and price
14:26.23sevardthe Cadillac Sixteen is the best car ever made
14:26.36Kattyi disagree.
14:26.44sevardHave you ever driven in one?
14:26.45coppiceGM and good don't seem like a matching combination
14:26.51Kattythere are too many factors to say Best Car
14:26.53Kattybest car for what?
14:26.54MrChimpyforget the ticket price. then look at the service costs. then say goodbye to comfortable retirement.
14:27.01sevardKatty: it's simplistic math
14:27.04sevardit _is_ the best car
14:27.06Kattycoppice: my dad works for GM/chevrolet
14:27.16Kattycoppice: they seem to be ok vehicles.
14:27.22Kattycoppice: tho, i'm kinda partial to pontiac myself.
14:27.25Hmmhesaysanyway ever heard of "virtual line"?
14:27.34sevards/way/one
14:27.41sevardoh, you can't sed eachother's lines.
14:27.42MrChimpythey sell some chevrolet thing over here. it looks bloody horrible
14:27.48coppicethe european GMs really are the worst crap made in europe
14:27.52*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
14:28.07Kattycoppice: european cars don't lag as much on the gear shift.
14:28.11coppiceand some of the US GMs are like death traps
14:28.22Kattyi think i'm going to stick with pontiac
14:28.24MrChimpyoh, maybe not
14:28.40sevardi had a pontiac for a couple years, i hated it.
14:28.44Kattycoppice: i've had a cavalier (first car), a sunfire, and a trans am.
14:28.46MrChimpythe chevys on .co.uk look like everything else
14:29.01MrChimpychevys are sold as vauxhall over here, or were.
14:29.11MrChimpynow they sell under chevy brand
14:29.15MrChimpystill look dull
14:29.18Kattycoppice: this G6 coupe is starting to look real nice
14:29.20[TK]D-FenderKatty : I'm just waiting to ditch my 1987 Camaro and get something about a decade newer....
14:29.58coppicethe nastiest thing I ever drove was a Chevy. Can't remember the name, but it had plenty of power to go, and no brakes to stop.
14:30.03Katty[TK]D-Fender: that's a cute little car i bet
14:30.31Katty[TK]D-Fender: they kinda remind me of corvettes for some reason
14:30.59chapeaurougehow can i start * to have more debug messages in /var/log/asterisk/messages ?
14:31.09Kattycurvy little bubble corvette
14:31.56*** join/#asterisk ibob63 (n=hp@bb-87-82-7-89.ukonline.co.uk)
14:32.07sevardI'm telling you man, the Cadillac Sixteen
14:32.07vader--do you guys know off the top of your heads what color the FXS modules are suppose to be on the TDM2400P cards?
14:32.28sevardFXO = red FSX= green
14:32.35Kattysevard: they're not pretty!
14:32.35[TK]D-FenderKatty : I'm looking at getting likely either a Cavalier or Sunfire, 1995-1997
14:32.44vader--thanks
14:32.45sevards/FSX/FXS/g
14:32.46Hmmhesayscamaro what
14:32.46Katty[TK]D-Fender: they're reliable little things, i'll say that.
14:32.58Katty[TK]D-Fender: i think my cavalier handled a little better than the sunfire.
14:33.07ibob63Occasionally the my office asterisk server mysteriously can't register with the gateway. Is there a way I can get asterisk to email me when it fails to register with the gateway?
14:33.08[TK]D-FenderKatty : Maybe more where you are... lets say that Quebec winter is remarkably UNKIND to cars....
14:33.09Katty[TK]D-Fender: but that's probably cause the cavalier was my first. and you always remember your first
14:33.16Katty[TK]D-Fender: yeah.
14:33.25Katty[TK]D-Fender: if it was snowing, that trans am wasn't going /anywhere/
14:33.37Ahrimanesibob63: monitor the log files and have a script email you when it's logged?
14:33.39Katty[TK]D-Fender: which is why i'm back to driving a sunfire again
14:33.47Katty[TK]D-Fender: i loved that trans am...
14:33.51coppicei've just looked up the cadillac 16. I see a lot of those in china. probably the world's ugliest car (which is saying something, considering the current BMWs)
14:33.55[TK]D-FenderKatty : I'm going to miss my 0 degree turn radius :D
14:34.04Katty[TK]D-Fender: i miss driving a manual.
14:34.09Katty[TK]D-Fender: and i miss the purr :<
14:34.14sevardcoppice: 16 cyl man.  best car.
14:34.22[TK]D-Fendercoppice : Doubt it beats Renaut, Hugo, or Lada for that...
14:34.28*** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr)
14:34.32Katty[TK]D-Fender: *shift*, *exhaust*, *swoon*
14:34.34[TK]D-FenderI'm regretably stuck on automatics.
14:34.41ibob63Ahrimanes: so there isn't a built in function for emailing when an error is logged?
14:34.46Katty[TK]D-Fender: don't take corners at 60mph, your ears will pop :P
14:35.16*** join/#asterisk apardo (n=apardo@87.217.145.29)
14:35.17Ahrimanesibob63: no, really isnt a job for asterisk.. more a job for a monitoring system
14:35.21coppicea lada looks elegant compared to that cadillac
14:36.17coppicethough whole car industry seems to be going through a butt ugly phase
14:36.23Kattyyeah, it is.
14:36.28Kattyespecially the suvs
14:37.06[TK]D-FenderKatty : Yeah, the Honda Element is FUGLY.
14:37.15[TK]D-FenderLike a brick with wheels...
14:37.17coppicedunno. the only reasonable looking thing BMW make now is their SUV, and the porsche one isn't too bad
14:37.31[TK]D-FenderAnd the ever gay Aztec...
14:37.41sevardhonda element is one ugly POS but not quite as ugly as the PT Cruiser
14:37.44HmmhesaysI miss driving
14:37.45Kattythe pontiac solistice......i dunno what they were thinking.
14:37.58Hmmhesaysjamie, I wish you a happy life with your fiance
14:38.01Hmmhesaysbwhahaha
14:38.11Kattyit looks like a very seriously deformed coupe of some sort.
14:39.00Kattysevard: the pt curisers remind me of old lady cars.
14:39.13Kattysevard: i saw one with a little old lady and wood paneling on it ;)
14:39.20Kattysevard: after that, i was ruined.
14:39.35sevardI used to work with this jesus freak who had a gold PT cruiser
14:39.42sevardapparently it made him closer to god
14:40.08Hmmhesaysand I always that f@#$#@ you like an animal brought you closer to god
14:40.31[TK]D-FenderKatty : Even worse was Chevy's attempt to market a 3-speed automatic veriosn of the Nova to South America.  Keeping in mind that in Spanish "Nova" means "Doesn't Go" :D
14:40.37sevardHmmhesays: combine that with a heavy dose of acid
14:41.24Katty[TK]D-Fender: 3 speed automatic?
14:41.33Katty[TK]D-Fender: 1,2, and ohgodoverload?
14:41.35[TK]D-FenderKatty : Special transmission...
14:41.46Katty3 speed automatic is confusing
14:41.52Kattyi can't even picture that
14:41.54[TK]D-FenderHmmhesays : Occasionally I like NIN too...
14:42.01Katty1,2 and reverse.....
14:42.04Katty1,2 and 'splode
14:42.05coppiceall automatics were 3 speed just a few years ago
14:42.14Hmmhesaysa local band here used to a do a kickass rock cover of "head like a hole"
14:42.17Kattycoppice: i've never /seen/ a 3 speed automatic before
14:42.28Kattycoppice: how many is a 'few years'
14:42.33sevard40ish
14:42.54*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
14:43.24coppice20
14:43.42coppicein the mid 80s only the high end had more than 3 speeds
14:44.15sevardcoppice: 20 years != few years
14:44.27*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
14:44.31coppiceyou must be rather young
14:45.09coppicethe last time I bought a car was 1985. that might colour my thinking
14:45.14Kattycoppice: i'm only one year over 20 :P
14:45.44coppiceI haven't owned a car for 15 years
14:46.33tzangerinconcievable!
14:47.04NuggetI own a car but I rarely drive it.
14:47.27NuggetI've put less than 5,000 miles on it in the past year.
14:47.46tzangerwow..> I typically put about 20-25k on mine a year
14:47.49tzanger(km not miles)
14:47.55BadPackethas anyone seen JerJer?
14:48.01tzangerI don't consider thata lot of driving though
14:48.17NuggetMy mileage was 23,619 on 6-Apr-2005 and 27,514 on 3-Apr-2006.
14:48.26*** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com)
14:48.41Nuggetdunno what it is today, but it can't be more than 27,800 since I haven't filled it up since 3 Apr.
14:48.52KattyNugget: what do you drive?
14:49.00[TK]D-FenderI do around 8-10k on mine...
14:49.05Nuggeta bmw m roadster.
14:49.09Kattyah, k
14:49.12Kattydon't want a bmw :)
14:49.16Nuggethttp://slacker.com/~nugget/mroadster/
14:49.23Nugget(car stats.  I love stats)
14:49.33[TK]D-FenderBreak My Window?  Bimbette Motor Weapon?  Bus Metro Walk?
14:49.42Nuggetbroke my wallet.
14:50.03sevardbuggered my mom
14:50.14*** join/#asterisk jhava (n=icechat5@200.58.26.21)
14:50.18[hC]This might sound weird, but has anyone experienced a problem on a PRI where 9 times out of 10, you can place a call, but occasionally you get channel unavailable for a number that you can normally dial (and since i dial it via another provider right away,and it goes thru, theres nothing wrong with the number itself)? Is it possible that theres something wrong on my end, or is this likely my provider's fault?
14:52.33Hmmhesayswhat was the website where you can share a whiteboard
14:53.46noname32netmeeting
14:54.01*** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net)
14:55.19jhavahello, has anyone configured two step dialing when connecting Asterisk to an ISDN PABX (PABX dialing to Asterisk)?
14:55.43sevardhahah
14:56.15sevardHmmhesays: was it a website recently bought by google?
14:56.43Hmmhesaysaren't they all?
14:57.01sevardI forgot Internets == GOOG
14:57.02syzygybsdcan someone direct me somewhere to learn about call routing by callerid?
14:57.16HmmhesaysI put up an example on the digium forums
14:57.17coppiceif google buys something released, does it immediately go back to being beta? :-)
14:57.28MrChimpycop: heheh
14:57.29mutyes
14:57.39syzygybsdthanks
14:57.47Hmmhesayshttp://forums.digium.com/viewtopic.php?t=6171&highlight=
14:58.17*** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it)
14:58.24Hmmhesaysshould be pretty straightforward
14:59.19sevardHmmhesays: www.writeboard.com ?
14:59.53[TK]D-Fendersyzygybsd : Read up on "gotoif" and "asterisk functions" on the WIKI.  Everything you should need is in those 2 references
15:00.20syzygybsd[TK]D-Fender: thanks now that you mention that function it all makes sense...
15:00.31Hmmhesayssyzygybsd: did you even click that link?
15:00.45Hmmhesays[TK]D-Fender hogging all the credit
15:00.46sevardHmmhesays: the one google acquired was Writely
15:00.51syzygybsdoh.. not yet, was off looking for what you said
15:00.57Hmmhesayshttp://forums.digium.com/viewtopic.php?t=6171&highlight=
15:01.00Hmmhesayshaha
15:01.07Hmmhesaysgoogle ownz joo
15:01.07syzygybsdthanks
15:01.14syzygybsdjah
15:01.34filesweet
15:01.38sevardI forgot google bought sketchup, i haven't tried it yet
15:01.40*** join/#asterisk gursikh (n=gursikh@158.135.0.125)
15:01.40noname32hey guys is there any good howtos for upgrading asterisk on a centos box?
15:01.46filehigh speed data service is deployed here :D
15:01.49filefrom Rogers
15:01.53Hmmhesayssame way you would do it on a none centos box
15:02.00MrChimpy1) back up
15:02.02MrChimpy2) build
15:02.08*** join/#asterisk SplasPood (n=jwb@206.252.198.101)
15:02.08Hmmhesays3) sammichs
15:02.12MrChimpy3) sacrifice chicken
15:02.15sevard3) profit!
15:02.28noname32lol i dont have a chicken will a dog do?
15:02.29MrChimpy4) ?
15:02.30Hmmhesaysno 2) is always ??? if 3 is profit
15:02.37sevardright, you did 2 wrong
15:02.44syzygybsdI am going to build it anyway, but is there a script that will call a number, and return whether the call was sucessfull or not (read answered, possible digits returned)
15:02.50sevard1) download new copy
15:02.52sevard2) ????
15:02.54sevard3) PROFIT!
15:03.28sevardsyzygybsd: what you're thinking of is a 'phone'
15:03.44syzygybsdhow many people know where that is from?  It has been used so much I don't think lots of people do
15:03.59sevardsyzygybsd: the underpants gnomes in a south park episode
15:04.12MrChimpysev: seen the new ones?
15:04.14syzygybsdsevard: I want an automated job so it can alert someone if there is a problem
15:04.27MrChimpyoprah winfrey's lower regions going psychotic?
15:04.27sevardMrChimpy: I haven't had cable television in over a year so I'm a bit behind ;/
15:04.28syzygybsdnew one tomorrow right?
15:04.38syzygybsdoprah's minge
15:04.45MrChimpysev: but you have t'internet. www.mrtwig.net
15:04.53[TK]D-Fendersevard : Been 2 for me, and thats immeditaely following a purchase of a 52" HDTV :D
15:05.02Sonderbladeanyone know from where i can get a backport of asterisk 1.2.6 or greater for debian sarge?
15:05.22codebreakerSonderblade: which one is at bpo?
15:05.23syzygybsdSonderblade: just download adn compile the source
15:05.24sevardMrChimpy:  I know. I don't have any stable HDDs at the moment.  I needs moola to get disks so I can keep content.
15:05.45codebreakerokay 1.2.1
15:05.48syzygybsdthat is what I am running on sarge
15:06.03MrChimpywow. you must be poor. i'd throw you a dime if my throwing arm could manage the atlantic :)
15:06.25*** join/#asterisk nahirean (n=nahirean@unaffiliated/nahirean)
15:06.26sevardMrChimpy: poor atm :) i havn't found any nice 10 gbs in the trash yet
15:06.28MrChimpyi keep choking at how cheap HDs are :)
15:06.46Sonderbladecodebreaker: 1:1.2.1.dfsg-2bpo1 which is to old i think
15:06.49gursikhI love how cheap they have gotten, and hope they go cheaper
15:06.52MrChimpyi have a bunch of 40s and stuff doing nowt.
15:06.56syzygybsdheh, I remember having a 10MB hd when I was younger
15:06.59sevardit's okay being poor though, you wait till you have money then what you wanted back then is 100x cheaper now
15:07.04syzygybsdthen before that a computer without a HD
15:07.12*** join/#asterisk SuperLag (n=aaron@gentoo/developer/SuperLag)
15:07.14codebreakerSonderblade: it depends on waht features you need
15:07.17sevardsyzygybsd: I paid $150 for 10 gb back in the day
15:07.26Sonderbladesyzygybsd: cool, it would be nice of you to upload it to the backports site
15:07.30syzygybsdsevard: that wasn't that long ago
15:07.41sevardabout 9-10 years ago
15:07.47Sonderbladecodebreaker: i need >= 1.2.6
15:07.52MrChimpyyes, we all had big HDs with tiny capacity. 10meg full height SASI. etc etc :)
15:07.59sevardhahaha
15:08.15*** join/#asterisk wunderkin (n=kev@mmds-216-19-40-108.mm.az.commspeed.net)
15:08.16sevardBack when I was your age we had these huge fucking cabnits that stored a character a piece
15:08.28MrChimpyST506, now that was a interface
15:08.35syzygybsdSonderblade: it is just the source... no packages
15:08.39sevardit only took 3 square blocks of network equiptments to map out a paragraph!
15:08.50coppiceSMD. now that was an interface
15:08.53sevardthat was back when computars were fast
15:09.00sevardcomputers*
15:09.13coppice1kW per drive. worse than a pentium 4 :-)
15:09.19MrChimpypppft. computers are MODERN
15:09.25Sonderbladesyzygybsd: oh, for my purpouses i need to have a sarge compatible binary package
15:09.29MrChimpyI used a DIFFERENCE ENGINE and LIKED IT.
15:09.56codebreakerSonderblade: backport it yourselve?
15:10.21Sonderbladecodebreaker: am trying but its hard
15:10.47[TK]D-FenderSonderblade : Why binary?
15:10.56[TK]D-FenderSonderblade : Use the Source Luke!
15:11.10codebreakerSonderblade: now its normaly not hard. fetch source do uupdate and dpkg-buildpackage
15:11.27sevardI have an uncle who sells electronics, one of his clients were buying all this old shit, tubes from the 70's, switches from the 60's
15:11.51sevardhe got curious and made a visit to the client, found out it was a bank under another name with 40 year old machines
15:11.56*** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk)
15:12.07codebreakerSonderblade: i have no time now. but if you can wait a day or two. ask me again
15:12.51MRH2hi can someone point me in the direction of what the "new sip tranfer code" is all about?
15:13.22pifwhat is the default value when using jitterbuffer=yes ?
15:13.22*** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)
15:14.25*** join/#asterisk lzhang (n=rjrae@67.95.13.46)
15:14.25MRH2can't u save to 1 db and replicate?
15:15.35Sonderblade[TK]D-Fender: to be able to distribute it
15:16.08*** join/#asterisk adker (n=adker@67-136-210-63.dsl1.glv.ny.frontiernet.net)
15:16.43*** join/#asterisk inv_arp[work] (i=junya@c-67-191-62-53.hsd1.fl.comcast.net)
15:18.42*** join/#asterisk zaf (n=zaf@wsip-68-228-9-79.br.br.cox.net)
15:18.49codebreakerMRH2: yes/no i think about storing to one host and then replicating to many(more then 2) and then accessing the mbox directly from there. but this is know to be sometimes a little bit tricky when doing deletes
15:18.50[TK]D-FenderSonderblade : If there is already a binary package, wouldn't it ALREADY be in distribution?
15:19.07chapeaurougeanyone using Junghanns QuadBRI card?
15:19.28codebreakerMRH2: and one db is a SPOF
15:20.41stoffellyes chapeaurouge
15:20.48MRH2i mean replication and  the ''traditional' failover type stuff
15:21.25chapeaurougestoffell, im gonna paste my config on some site... will give you a link... if you could take a look quickly and tell me if you see anything wrong, it'd be great
15:21.40*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
15:21.42stoffellit's not working then
15:21.43stoffell?
15:22.28chapeaurougei guess it is. i need to plug some isdn lines in :) but i want to see if everything is normal so far. i cant unplug the isdn lines in the middle of the day :)
15:23.20MRH2anyone know what new stuff the  the new sip transfer code does? (/team/oej/siptransfer)
15:23.25Zeeeksome can... replicant
15:23.53oejMRH2: No, haven't got a clue :-)
15:24.02oejSorry
15:24.03MRH2lol
15:24.12oejIt implements REFER in a better way.
15:24.14Ahrimaneshehe
15:24.22oejIt's gradually moving into SVN trunk
15:24.27chapeaurougestoffell, http://pastebin.ca/52888
15:24.36Zeeekoej you should change the method to REEFER
15:24.39oejWill continue as soon as all my branches are back
15:24.44oejzeek: :-)
15:24.47Cresl1noej!!!!!!
15:24.53oejCresl1n: Hi!!!
15:25.00Zeeekoej Paris soon?
15:25.16chapeaurougestoffell, note i dont know if i need the span stuff in zaptel.conf... kinda copied-pasted ;)
15:25.23oejzeeek: Paris was cancelled
15:25.29Zeeekah. sorry
15:25.31oejzeeek: Brussels and London this week
15:25.38oejYes, I would have loved going to Paris now
15:25.52Zeeekwell, break a leg as we say. A call leg obviously in this case!
15:26.06MrChimpylondon/paris as in astricon?
15:26.09coppiceparis often gets cancelled. they keep closing down the country
15:26.10MRH2thanx
15:26.11ZeeekWeather finally got decent and I was all set to buy you a glass of fine wine
15:26.13*** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net)
15:26.39oejMrChimpy: No, meetasterisk.com. I am no part of Astricon and won't be there
15:26.41Zeeekcoppice nah, that's just video we put out to the press to keep the tourists away
15:26.56MrChimpyparis is a craphole. I went to biarritz in south of france last week. much better. gorgeous girls.
15:27.11Zeeekbiarritz is really great
15:27.12coppicethe only time I tried to go to paris the whole bloody country shut down
15:27.17MRH2is it to make it more compatible or are there changes from the POV of us end users?
15:27.18Hmmhesaysparis almost got destroyed in the last book i read
15:27.23Zeeekthey knew of you in advance coppice
15:27.37MrChimpyso is astricon some unholy thing?
15:27.45Ahrimanesno astricon is good
15:27.52Ahrimaneswell was last year anyways
15:27.57MrChimpygoodo. i shall attend the london one
15:28.23oejThis year's Astricon has nothing todo with previous Astricons
15:28.23MrChimpyI was trying to blag the stockholm one, then kind of undermined myself by implementing what we needed from the book :)
15:28.31*** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr)
15:29.20MrChimpynot got much reason to go as yet, other than skiving for a day. everything i've done in asterisk has just "worked".
15:29.27*** join/#asterisk DoktorGreg (n=Greg@70.91.121.89)
15:29.38MrChimpypast a bit of trickiness getting the E1s talking
15:29.58Ahrimanesoej: oh?
15:30.29*** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler)
15:30.31stoffellchapeaurouge, looks good
15:31.07ZeeekAhrimanes oej ins't doing this one. Just for info
15:31.22AhrimanesZeeek: ok
15:31.26ZeeekMrChimpy yes, but there's the beer and girls, not nec.  in that order
15:31.42Zeeekand if your thing is boys, there's even more of that :)
15:31.48MrChimpypppfftft.
15:31.48*** join/#asterisk salviadud (n=ralfalfa@dsl-200-78-64-10.prod-infinitum.com.mx)
15:32.01Zeeekmeeting other geeks in meat space is a trip!
15:32.12salviadudmeat space?
15:32.16MrChimpyi'm sure it'll be chock full of sexy women. just like linux today.
15:32.21Zeeekhahahaha
15:32.23vader--do you guys know the difference between the s400M revision B and revision B2?
15:32.32Zeeekwell, there were a few handing out t-shirts
15:33.00MrChimpyi spent about 30 mins in that show. it started getting depressing very quickly :)
15:33.39rpmis there something im missing if i connect to port 5038 (astmanager) and type 'Action: SIPpeers' i want to get a list of SIPpeers
15:33.59MrChimpybest trade show I ever went to was CabSat. there were porno channels trying to sell content with these hookers trotting round in next to nothing, then an irish tv station had a free guinness bar.
15:35.10ZeeekI went to a linux one in L.A. once years ago and the little LED mouse keychain light they gave out *still* works!
15:35.33MrChimpyyeah, crap freebies at linux today too.
15:35.44ZeeekI think it was worth the time and money to meet some of the majors last year (including oej)
15:36.09Zeeeksince Mark is in Paris several time a year we don't need Astricon for that
15:36.23SplasPoodhrm.. I have a context which does a Background() (IVR menu) and lets people either dial a single digit extension from the local context, or a direct extension which is included ...  seems sometimes people miss the first digit of the exten .. 2002 and instead dial 002...   problem is that gets matched the _0, exten rather than going to invalid...  any thoughts?
15:36.44SplasPoodor rather my exten is simply '0' not _0
15:37.39MrChimpyyeah, i'm not about to bat around with the big-wigs :)  at the moment this is just my job, so unless work want something super thrilling...
15:37.54SplasPoodseems as soon as it gets that first 0 it dials rather than waiting for the timeout...
15:38.22sevardMrChimpy: DUDE, cabsat sounds awesome
15:38.26MrChimpythings will probably get hairy when i get some investment in the project and we start scaling up
15:38.57MrChimpysev: yeah, it was. but then that was 1998 I think. I was drunk when I got there too thanks to a pub lunch.
15:39.10sevardthat sounds like one awesome show.
15:39.14DoktorGregAny digium guys around?
15:39.47noname32hey when compiling asterisk-addons do you do that before or after the core ?
15:40.40*** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane)
15:40.59chapeaurougestoffell, cool thx. What about these May  2 18:12:56 WARNING[11306] chan_zap.c: Detected alarm on channel 10: No Alarm
15:40.59chapeaurougeMay  2 18:12:56 WARNING[11306] chan_zap.c: Unable to disable echo cancellation on channel 10
15:41.07chapeaurougenormal behavior?
15:41.23asterboyyep
15:41.29fileDoktorGreg: maybe, wazzup?
15:41.30SplasPoodHrm.. why does background abort immediately on a single digit 0, but on 165 (for example) it waits for all digits?
15:41.58vader--voipsupply.com sent me a TDM2400P board with 5 s400M Rev B2 modules and 1 s400M Rev B module
15:42.04vader--wonder if i should bitch
15:42.15ZeeekSplasPood have you looked at the extension evaluation order on the wiki?
15:42.50vader--the board they sent me didn't work so i RMAed it and they sent out another board with FXO modules on it
15:43.07DoktorGregI need to know what kind of motherboard to buy to get zttest score closer to 100%
15:43.19DoktorGregfile
15:43.45filewhat is it at now?
15:43.48DoktorGregright now i am at 98.75%
15:43.59fileusing what, actual zaptel hardware?
15:44.08DoktorGreg405
15:44.12SplasPoodZeeek: Yes I'm looking at it now, although I'm not sure how it applies to my issue
15:44.13DoktorGreger 205
15:44.20DoktorGregsip phones work fine
15:44.26jhavaHello all, quick question: is there a configuration in ISDN PRI to accept digit by digit dialing from a PABX using a TE110P?
15:44.28DoktorGreghowever when i hit music on hold with
15:44.31SplasPoodif the exten => 0,1,NoOp()
15:44.34DoktorGregone of the pri lines
15:44.36SplasPoodand you dial 011
15:44.36*** join/#asterisk mog_work (n=mogorman@gateway.digium.com)
15:44.37SplasPoodfor example
15:44.39DoktorGregthe call drops
15:44.42SplasPoodshouldn't that NOT match?
15:45.26fileDoktorGreg: only when it goes through the 205?
15:45.29*** join/#asterisk bkw_ (n=brian@adsl-70-143-63-171.dsl.tul2ok.sbcglobal.net)
15:45.32DoktorGregyah...
15:45.41DoktorGregthe 205 is bridging calls just fine
15:45.50fileinteresting
15:46.18DoktorGregand zttest reports that i drop to 97.5% when the call drops
15:46.42eric_hillDoktorGreg: What motherboard/system type do you have now?
15:47.02DoktorGregits a MSI mobo, AMD nforce2 thingie
15:47.09ZeeekSplasPood your issue is that the evaluation is "premature" is it not?
15:47.11*** join/#asterisk Xacau (i=TCHE@201-35-189-247.smace701.dsl.brasiltelecom.net.br)
15:47.15*** join/#asterisk hrhrhr (n=c1@87.127.7.210)
15:47.16eric_hillCan you pastebin the output of lspci?
15:47.21DoktorGregI am gonna try a different notwork
15:47.22SplasPoodZeeek: Correct
15:47.23DoktorGregsure
15:47.28DoktorGregone sec
15:47.34hrhrhrhello :)
15:47.49eric_hillAlso, an "lspci -tv" please.
15:47.58ZeeekSplasPood and that page is about the order in which the exten are evaluated, is it not?
15:48.29SplasPoodZeeek: Yes, but based upon the page I don't see what's wrong with my setup.    I'm not trying to argue here.. maybe you could show me where I'm going wrong..
15:48.43DoktorGreghttp://pastebin.ca/52890
15:48.48ZeeekWhat URL are you looking at?
15:49.27Xacaucan i disc to anyone using asterik?
15:49.28DoktorGreghttp://pastebin.ca/52891
15:49.31Xacau*call
15:49.42Xacauincluding cellphone?
15:49.54ZeeekSplasPood as far as the wrong first digit, all you can do is prevent it by limiting what is evaluated
15:50.15SplasPoodZeeek: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf+sorting
15:50.25Zeeekyeah that's the one
15:50.29DoktorGregafk getting hot coffee
15:50.32DoktorGregbrb
15:50.47Zeeekdon't let any extension be dialed wihout checking it first would be my suggestion
15:51.38SplasPoodZeeek: What do you mean checking it first?
15:51.53eric_hillDoktorGreg: any idea what the "unknown device" is?  Also, are you using a Wildcard?
15:52.12*** join/#asterisk marl (n=matt@albacom.plus.com)
15:52.12DoktorGregthe unknown device is the 205
15:52.48ZeeekSplaspod,  I reread what you wrote higher up. Background is falling thru after a single zero, is that it?
15:53.19hrhrhrcould you point me in the direction of a guide to getting a generic wildcard fxo card working with asterisk (built from src)? :)
15:53.22SplasPoodZeeek: Correct
15:54.03Zeeekand beneath Background() there is a check for a few digits and then you dial anything that doesn't meet the few single digits you allow?
15:54.07marlhi, can anyone point me in the rite direction for this, ive search google etc. and not found anything, i have an TDM400 card with an 2 x fxo and 1 x fxs boards on it, i find that any calls through the fxs phone tend to be very quite at my end, hoiw can i turn the gain up on this board? ive used fxotune to tune the fxo cards, but cant find the equiv for fxs, any one any pointers?
15:54.32Zeeekmarl look at zaptel.conf
15:54.46Zeeekand serach google for zaptel gain or something like that
15:55.06SplasPoodZeeek: No, I define 4 extens... 0, 1, 2, 3  and have an 'i' exten that plays 'pbx-invalid' then goto's back to the background()
15:55.21SplasPoodZeeek: I also include another context which handles my direct dial extens...  _20XX and _21XX
15:55.24ZeeekSplasPood maybe you want to pb that
15:55.34SplasPood~pb
15:55.35jboti heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
15:55.51Zeeekheh, I'm getting lazy in my advanced age
15:56.39marlthanks Zeeek ill have a look there
15:56.52Zeeekyeah I don't recall the gain lines in my head
15:57.06Zeeektxgain? rxgain?
15:57.42Zeeekzapata.conf, not zaptel, sorry
15:57.58Zeeekand it is rxgain= and txgain=
15:59.12hrhrhri've followed this so far http://forums.digium.com/viewtopic.php?t=6151&highlight=wildcard+x100p
15:59.26hrhrhrbut looking at the asterisk console whilst dialling that pstn number doesn't give any output
15:59.45hrhrhri guess i've missed something huge out somewhere...
16:00.48*** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at)
16:02.02vader--heh dude at voipsupply.com screwed up my RMA
16:02.16vader--sent me a 2406E instead of a 2460E which is what he sent me first
16:02.28salviadudvoipsupply suck
16:02.36salviadudthey wouldn't even sell to me
16:02.41vader--really why?
16:02.42salviadudcause i'm international
16:02.47vader--gotcha
16:02.50salviadudi used voxilla
16:02.54salviadudthey rock
16:02.55vader--they probably can't or something
16:03.06salviadudvoipsupply is based in NY
16:03.13salviadudthey're a bunch of sissies
16:03.15vader--some US companies won't send international
16:03.19Zeeekwhere are you salviadud?
16:03.22vader--because of US laws or what not
16:03.23salviadudmexico
16:03.29salviadudbecause they are idiots
16:03.31Zeeekfunny they sold to me
16:03.45salviadudi have bought some good stuff from the USA, me being mexican
16:03.47vader--heh tomorrow i will have 3 TDM2400P cards on my desk
16:03.51salviadudthe smart companies, those are the ones that sell
16:03.52vader--1 which doesn't work
16:03.57vader--1 which has the wrong modules
16:04.07vader--and 1 which hopefully will have the right modules and works
16:04.08*** part/#asterisk downunder33 (n=robert@219.95.158.235)
16:04.20salviadudso, you'll have to return 2
16:04.23vader--ya
16:04.23vader--hehe
16:04.25salviadudthat's a bummer
16:04.52vader--ya it sucks because the first one they sent has the right modules but the card doesn't work
16:05.00vader--and the second one works but doens't have the right modules
16:05.02brookshirewhat's wrong with the tdm2400p?
16:05.10vader--and i asked if i can just switch the modules he said no
16:05.15vader--they will send a whole new card
16:05.24brookshirethat's efficient, lol
16:05.36vader--brookshire no matter what computer i put the card into it doesn't recognize
16:05.41SplasPoodbrookshire: any idea if asterisk treats '0' during a Background() differently than any other digits?
16:05.43eric_hillDoktorGreg: Based on the lspci, try moving the 205 to a different slot and see what happens.
16:06.01Hmmhesaysgod there is a lot of shiat broken in the latest a@h
16:06.02SplasPoodbrookshire: it seems to be taking the 0 and immediately executing rather than waiting for more digits
16:06.02brookshirespals: is this during voicemail?
16:06.03DoktorGregeric_hill, thats where i was headed next
16:06.10SplasPoodbrookshire: nope, during a Background()
16:06.13ZeeekSplasPood it doesn't in 1.2 because I use it there with 0
16:06.39salviadudsplashpood, do you have an i extension?
16:06.45*** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br)
16:06.45brookshiresplas: you mean like 0XXXX
16:06.48SplasPoodsalviadud: yes
16:06.58salviadudi was playing with background just last night
16:07.00SplasPoodI have an i exten and an exten => 0,1,NoOp()
16:07.06SplasPoodif someone dials 0123
16:07.11SplasPoodI expect it to hit the 'i' exten
16:07.14SplasPoodinstead it goes to 0
16:07.16SplasPoodimmediately
16:07.37Zeeeknormal
16:07.39brookshiremake sure the 0123 pattern is before the 0,1,NoOp()
16:07.56brookshirei don't know why that would matter, but!
16:07.57salviadudyou see, that pattern does not exist
16:08.03salviadudhehe
16:08.12SplasPoodright
16:08.14SplasPoodits invalid
16:08.25SplasPoodthere is no 0123 pattern
16:08.25brookshire1.2.7.1
16:08.28SplasPoodyes
16:08.43salviadudmmmm
16:08.48salviadudtake out that extension?
16:08.49ZeeekSplasPood show us the context in pb
16:08.54salviadudit doesn't seem to be useful
16:09.11salviadudor just change the 0 for something else
16:09.26SplasPoodZeeek: I did, I msg'd it to you when you asked the first time :P
16:09.43salviadudsplaspood
16:09.43Zeeekoh. I don't get messages, sorry
16:09.45salviadudi want to see it
16:09.46SplasPoodsalviadud: 0?   Well 0 rings the operator...   so I'd say thats useful :)
16:09.52*** join/#asterisk fu3 (n=kaa@234-200-29-134.hcc.mnscu.edu)
16:09.59fu3hello lads
16:10.06brookshiresplaspod: works for me
16:10.20ZeeekSplasPood what's the big secret, show it to the world
16:10.27fu3Is there any way to expand the line number on a Polycom 301?  It only shows "...900" as line1  and i'd like it to show the entire number.
16:10.33brookshireexten => 0123,1,Answer
16:10.33brookshireexten => 0123,2,MusiconHold(native-random)
16:11.14ZeeekSplasPood by the way if you hit '*' or '7' does it go to i ?
16:11.45brettnemfu3: no
16:12.12fu3no? really? theres all that space next to it!
16:12.26fu3so i'll only ever be able to see the last three digits of an extension?
16:12.43SplasPoodZeeek: 7 does
16:12.50SplasPoodZeeek: as does *
16:12.59SplasPoodimmediately too
16:13.02Zeeekok
16:13.02brettnemfu3: You can't change the display on the polycom
16:13.14salviadudthat's gotta be the weirdest ivr i've ever seen
16:13.19fu3weak.. why would they design it so as to only show the last three digits.. argh
16:13.33SplasPoodsalviadud: what's weird about it?
16:13.40SplasPoodseems pretty straight forward to me...
16:14.02*** part/#asterisk ibob63 (n=hp@bb-87-82-7-89.ukonline.co.uk)
16:14.21salviadudwell, it's not that it's weird, it's a different style
16:14.31*** join/#asterisk dos000 (n=dos000@wsp05974758wss.cr.net.cable.rogers.com)
16:14.51SplasPoodsalviadud: What specifically?
16:14.58dos000is there a command to read from arbitrary fields in the mysql db ?
16:15.49salviadudthey way you start the backgroudn app
16:16.12salviadudi usually just do exten => s,1,Bacground...
16:16.26noname32any ideas on what i cant record with *1? i have it enabled in features and the dial and when i press it this is what shows in the log res_features.c: Feature interpret: chan=SIP/200-3650, peer=SIP/201-42b5, sense=1, features=16 res_features.c: Set time limit to 500 res_features.c: Timed out for feature!
16:16.43noname32i just cant get it to work for the life of me
16:17.11*** join/#asterisk Gamercjm (n=chris@pool-71-254-177-36.lsanca.fios.verizon.net)
16:17.16salviadudwell, there's no real error here...
16:17.26salviadudif they dial an initial 0
16:17.30SplasPoodsalviadud: Thats exactly how I'm doing it...
16:17.32salviadudit will always go to the operator
16:17.42SplasPoodyes
16:17.44SplasPoodI don't want that
16:17.49SplasPoodonly if they dial 0 on it's own
16:17.53SplasPoodsince it's not _0.
16:17.56SplasPoodits just 0
16:18.07coppicethe weirdest ivr i've ever seen says "press 1 for english" in cantonese and "press 3 for cantonese" in english :-)
16:18.31*** join/#asterisk Assid (n=assid@203.115.64.12)
16:18.36Assidyoza
16:18.40salviadudthe guy who made that. must be a prankster
16:18.49salviadudgotta admire the ingenuity
16:19.49salviadudsplaspood, you are not giving * any reason to wait for another digit
16:20.00salviadudall your extensions are single numbers
16:20.13salviadudyou would need to have a 02 extension
16:20.16salviadudfor example
16:20.41SplasPoodsalviadud: what's TIMEOUT(digit) for then
16:20.49SplasPoodyea I setup exten => _0X.,1,Background(invalid) and that fixes it
16:21.14salviadudthat variable is the time between digits
16:21.23[TK]D-FenderSplasPood : Just pastebin the whole thing so we can steer you right in 1 shot....
16:21.25salviadudbut since, they are all single digits...
16:21.30salviadudit's kinda pointless
16:21.32Zeeek<PROTECTED>
16:22.02SplasPood[TK]D-Fender: I did pastebin it and gave it to the 3 people I was talking to :P
16:22.17*** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)
16:22.18Zeeekif you did it publicly you'd have 240 possible helpers
16:22.26[TK]D-FenderSplasPood : If its still being discussed maybe you might want to ask someone else :)
16:22.31Zeeekheh
16:22.37Assidsup Zeeek
16:22.45Zeeeknada Assid
16:23.09SplasPood[TK]D-Fender: Nah I think i've "fixed it" ...
16:23.28[TK]D-FenderSplasPood : "think"? :)
16:23.53salviadudwell, it seems like it could work splaspood, give it a try
16:23.56salviaduddebug the thing
16:24.15SplasPoodit does work
16:24.17SplasPoodI have tried it
16:24.21SplasPoodI just don't like the solution :)
16:24.32salviadudwhy, the code is ugly?
16:24.34[TK]D-FenderSplasPood : C'mon show me....
16:25.01salviadudd-fender, when is your b-day?
16:25.15[TK]D-Fendersalviadud : March 7th
16:25.59[TK]D-Fendersalviadud : Why do you ask?
16:26.14*** join/#asterisk lzhang (n=rjrae@67.95.13.46)
16:26.39salviadudi was wondering about your astrological sign
16:26.52[TK]D-Fender"Octagon... as in STOP" :D
16:26.53lzhangI'm trying to make a call using the asterisk manager to have the system dial a number and just play a wav file
16:26.59salviadudyou are a water element, that's nice
16:27.08LostFrogDamn, [TK]D-Fender, you stole my joke.
16:27.09lzhangI'm looking at originate, but I'm not sure what I put for the channel
16:27.33[TK]D-FenderLostFrog : Depends, how old are you?
16:27.34*** join/#asterisk MacDome (n=eseidel@A17-255-100-181.apple.com)
16:27.55LostFrog[TK]D-Fender: Old enough to know better, but young enough not to care.
16:28.12SplasPoodsalviadud: Just kludgy
16:28.28[TK]D-FenderLostFrog : "Better than whom" :)
16:28.32*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
16:28.41salviadudwell, i would worry if it works, rather than if it's kludgy
16:28.59LostFrogI would worry if it *didn't* work.
16:29.03[TK]D-FenderSplasPood : Just pastbin it... I'm sure theres something we can do to clean it up.
16:29.37SplasPood[TK]D-Fender: Forget it man.. I'm convinced this is by design and kludging it with dummy patterns is the only fix
16:30.25*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
16:30.43salviadudright, if it didn't
16:30.50salviadudO_o
16:30.53[TK]D-FenderSplasPood : Your call... though I doubt it....
16:31.30salviadudSplasPood, i think you fixed it.
16:32.31*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
16:32.46[TK]D-Fender[Airwolf] : *boing*
16:33.00*** join/#asterisk MacDome (n=eseidel@A17-255-100-181.apple.com)
16:33.01[TK]D-FenderThats the best part of my God complex.... no peer pressure :)
16:33.14*** join/#asterisk Lino` (n=Lino@i577BC81E.versanet.de)
16:34.17coppiceI have a superiority complex without equal :-)
16:35.37noname32any ideas why one touch recording throughs this error Timed out for feature! and never works :/
16:35.53[TK]D-Fendernoname32 : Taking too long between the * and 1 maybe?
16:36.09SplasPood[TK]D-Fender: You're the only dissenting voice of 4
16:36.10*** join/#asterisk oej (n=oej@apollo.webway.se)
16:36.30Zeeekwhat 4? I didn't vote yet
16:36.39noname32i dont think so cause soon as the call is connect i press it and soon as i do it it outputs that to the log
16:36.44*** join/#asterisk Moradinn (n=H0t-5auc@213.122.19.209.transedge.com)
16:36.57noname32but if i change the command in features to ** it will work
16:37.14ZeeekSplasPood would say, "change it to **"
16:37.25SplasPood** ?
16:37.25noname32the features cmds dont work for me as long as they are * followed by a number
16:37.41Zeeekcheck your phone
16:37.53froguzi'm trying to get realtime to work (sipusers and sipeers in realtime mode) but i'm getting : res_config_mysql.c: MySQL RealTime: Failed to connect database server realtime on localhost.
16:38.10CunningPikeGood morning
16:38.30noname32well i can use the other stuff like *61 *43 *98 ect
16:38.50froguzmy res_mysql.conf file :
16:38.52froguz[general]
16:38.52froguzdbhost = localhost
16:38.52froguzdbname = realtime
16:38.52froguzdbuser = root
16:38.52froguzdbpass = pass
16:38.53froguzdbport = 3306
16:38.55froguzdbsock = /tmp/mysql.sock
16:38.59noname32it seems obnly when the call is active i cant do like *0 to hangup or *2 att transfer
16:39.07froguzsorry about the flooding
16:39.14noname32i have tryed with x-lite and a cisco ata 186
16:40.19*** join/#asterisk pythos (i=pythos@unaffiliated/pythos)
16:41.08noname32it says i need to turn on dynamic features for this right so in extensions.conf i put [globals]
16:41.08noname32Set(DYNAMIC_FEATURES=automon)
16:41.08noname32<PROTECTED>
16:44.38*** join/#asterisk mko-025 (n=korpim@p54989A8C.dip0.t-ipconnect.de)
16:44.39froguzi've created static_table, sip_table, user 200 and 300 in sip_table, i've edited extconfig.conf to the database settings, but it still can't connect
16:45.28froguzi've edited the res_mysql.conf too
16:47.08froguzam i missing something
16:47.10froguz?
16:48.45*** join/#asterisk esculapio_ (i=elvyn@200.88.44.66)
16:48.56*** join/#asterisk Abydos313 (i=abydos31@adsl-71-129-52-80.dsl.irvnca.pacbell.net)
16:48.57esculapio_hola quien habla espanol?
16:49.30*** join/#asterisk suma (n=suma@cm69.gamma29.maxonline.com.sg)
16:49.41sumawhich is the best calling card app for asterisk?
16:49.54sumawith web user interface
16:50.00*** part/#asterisk Moradinn (n=H0t-5auc@213.122.19.209.transedge.com)
16:50.06esculapio_tecnico, Hi, Hola
16:50.28froguzesculapio_, yo hablo español ¿qué necesitas?
16:50.53*** join/#asterisk razu (n=razu@dhcp-84-52-1-207.cable.infonet.ee)
16:50.55esculapio_froguz, me puedes ayudar con la coneccion de dos asterisk
16:51.22esculapio_s/s/x
16:51.22sumaam i in the wrong channel !, something other than english is going on here
16:51.45mog_workno your in the irght channel
16:51.58sumamog_work: thanks
16:52.08[TK]D-Fender<mog_work> no your in the irght channel <- no we have American in here too ;)
16:52.22tainted-lol
16:52.23mog_workheh
16:52.32tainted-I speak well Aelmican
16:52.49[TK]D-Fendertainted- : I read that like "pellican" on first clance :D
16:52.49Nivex"Lady, I only speak two languages: English and bad English."
16:52.53sumame to, i speak well almecidnassan
16:53.17sumawhich is the best calling card app for asterisk with web user interface?
16:53.37[hC]a2billing seems popular
16:53.42tainted-u speak delicatessen?
16:53.49[TK]D-Fendertainted- : YUM!
16:54.03[TK]D-FenderI speak in tongues... just ask my ex :)
16:56.56pythos<PROTECTED>
16:57.37esculapio_froguz, como es mejor por que solo estoy empezando
16:58.02esculapio_froguz, para empezar el que tu me digas\
16:58.20froguzesculapio_, read the private message window, ppl don't like any other lang. tha  english here
17:00.28esculapio_froguz, no tengo problemas con leer el ingles muchas gracias en esto momento me encuento en el link que me diste
17:01.51froguzok
17:02.30Hmmhesaysis froguz from france?
17:02.56coppicewaak je kui dei m'se ying man
17:04.47*** join/#asterisk zagaya971 (n=almeli@APointe-a-Pitre-102-1-3-9.w81-248.abo.wanadoo.fr)
17:05.41*** join/#asterisk saftsack (n=saftsack@p54A7F975.dip.t-dialin.net)
17:05.49pythoswho is psychic here? What do I need now :-)
17:06.59*** join/#asterisk markus99 (n=markus@165.154.121.219)
17:07.30pythosok... that was sillyness, now then: I got the hardware al figured out for my tdm400p, but I am not sure what to use for a web based management product, suggestions?
17:08.10Hmmhesaysdon't you wish your girlfriend was hot like me
17:08.27pythosare you a female?
17:08.37Hmmhesaysno but I like them
17:09.39pythoshmm, well then, I don't wish my girlfriend was hot like you
17:10.31[TK]D-Fenderpythos : Sorry, we don't have any spare clues to give you :)
17:11.03pythos[TK]D-Fender: hmm. well Im looking at sourceforge... there are a TON, just wanted to know where to start
17:11.22froguzHmmhesays, nop. is froguz from chile
17:11.30[TK]D-Fenderpythos : All web interfaces for * suck, just some less than others.  Why do you think you want/need it, and describe the size and type of system its expected to help you administer.
17:11.38HmmhesaysDo they hold carnivale in chile?
17:11.52tainted-froguz chilean women are HOT!@#!@$
17:12.08coppicewhat do you expect in that climate
17:12.13froguzhahaha tainted- yes they are
17:12.14Hmmhesaysthere are beautiful women everywhere
17:12.32froguzthat's true too
17:12.39coppiceI have two in this house :-)
17:13.52froguzi can't get asterosk to connect to the realtime database... i think there's something wrong with my res_mysql.conf but i can't find what is wrong
17:14.02froguzmaybe the socket name
17:14.21froguzis /tmp/mysql.sock an standard to all distros??
17:15.22Nuggetno.
17:15.43froguzmmmm... that may be the problem
17:16.58tainted-froguz cafe w/ legs :D
17:18.04froguztainted-, hahaha are you chilean?
17:18.21tainted-i wish
17:18.24tainted-hahaha
17:18.30froguzyou surely have been here
17:18.39froguzcafé con piernas rocks
17:18.57tainted-are u there now?
17:19.46froguzin a cafe??? i wish i could, but i'm at the office, here in santiago
17:22.20sevardcan anyone help me with a zap option I can't find
17:22.47sevardI know how to direct all calls from ZAP to my IVR, but I can't figure out how to direct individual ports to different sip lines
17:25.58froguztainted-, were are you from?
17:26.08froguzwhere*
17:26.15*** join/#asterisk MrDigital (n=VBDIGITA@pool-72-81-113-227.phlapa.east.verizon.net)
17:26.26tainted-froguz i'm in Los Angeles currently
17:26.34tainted-(not by choice)
17:26.36tainted-haha
17:27.16vader--is it normal for there to be no ringing when dialing an sip extension that has the phone off or not connected?
17:27.53syzygybsdon an incomming call on a registered sip connection, should it be asking me to reregister?
17:28.15vader--when there is no phone connected and you try to dial that sip extension the phone dialing just sits silent
17:29.52froguznot by choice!!! wow! don't you like to be there?
17:31.07Hmmhesaysgeebus it's like eleventy billion degrees in here
17:33.08HmmhesaysI love it when clients of clients try to contact me
17:33.34Hmmhesaysthey think they can just bypass the chain of command
17:34.08*** join/#asterisk Astinus- (n=abba@213.167.111.138)
17:34.12syzygybsdlol @ Hmmhesays what about providers of your clients?
17:34.15salviadudso, what do you do?
17:34.31salviadudsend em' to your clients?
17:34.34Hmmhesaysbingo
17:35.00salviadudyeah, that's lovely
17:35.25salviadudask Bob, cause it's not my problem yet
17:35.36froguzwiki says "You can keep any sip users in the flatfile AND use RealTime. How cool is that?" but in other page of the wiki says : "If you store sip.conf in the RealTime database, you need to rename/remove the text file otherwise the text file will superceed RealTime." wich one is right??
17:35.57Hmmhesaysi also dislike cheap bastards
17:36.11salviadudwhat kinda cheap are we talking about here?
17:37.54*** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk)
17:38.18Hmmhesaysthe kind of cheap that thinks they should get something for nothing
17:38.58Zeeekentropy is against something for nothing. It won't allow it, ever.
17:39.14ZeeekYou can only *think* it's for nothing.
17:39.30sevardanyone? zap help?
17:39.42*** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br)
17:39.48Zeeekno thanks
17:39.57sevardHmmhesays: btw thanks for something for next to nothing.
17:40.06Hmmhesaysyou paid
17:40.12Hmmhesaystherefore your legs aren't broken
17:40.24*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net)
17:40.58Zeeeksevard have you asked your question yet?
17:41.12sevardZeeek: yeah
17:41.19sevardI don't like reposting, but if you'd like
17:41.21Zeeekand still no answer?
17:41.29sevardIt was about a half hour ago, but no.
17:41.32Zeeekhow about re-stating?
17:41.34sevardI know how to direct all calls from ZAP to my IVR, but I can't figure out how to direct individual ports to different sip lines
17:41.46Zeeekwhat do you mean by ports?
17:41.56sevarda TDM400P has four ports
17:42.02Zeeekchannels
17:42.15Zeeekyou have 4 FXS?
17:42.15sevardI'm pretty sure that 'port' isn't the techincal term but that's what I've been using
17:42.17sevardokay, channels.
17:42.27sevardI have four FXO i believe, four red ones :)
17:42.38Zeeekwell otherwise we'd be forwarding 5060 to one place, 5061 to another
17:42.52Zeeekok you have four incoming ZAP channels
17:43.06sevardYup, outgoing works great
17:43.23sevardI have round robin on one and two and long distance on 3, four isn't hooked up yet
17:43.37*** join/#asterisk drfoomod2 (i=DrFooMod@ool-43501d9f.dyn.optonline.net)
17:43.44drfoomod2can you get info in the sip.conf file from the management API?
17:43.47*** join/#asterisk markus99 (n=markus@165.154.121.219)
17:43.53sevardI have all zap incoming directed to my ivr and the question arose how do I direct channel 1 to a sip line directly and the rest to the ivr
17:44.14sevardI already know how to direct a IAX trunk directly to a SIP line
17:44.19Zeeekwait I now have to deal with a totally incompetent person who is crossing my emails with incredibly silly questions about an order
17:44.24Kattyto all of you guys that have offered to get someone a drink when they're having a really crappy, i'd like to thank you obsessively.
17:44.25myiagyusing different contexts?
17:44.32Kattys/crappy/crappy day/
17:44.43Kattythank you thank you thank you
17:44.45*** join/#asterisk gammacoder (n=chatzill@64-132-192-33.gen.twtelecom.net)
17:44.48sevard<PROTECTED>
17:44.51myiagysevard if i'm not wrong, you can use a different context for each channel.. you define them in the zapata.conf file
17:44.55Kattyno, i'm being serious
17:45.06Kattyi was /just/ at the gas station, and the guy offered to get my tea for me
17:45.13Kattyhe must have been able to tell i was having a really shitty day
17:45.14*** join/#asterisk _-Jon-_ (n=jon@CPE000d8861e8f7-CM00080d290642.cpe.net.cable.rogers.com)
17:45.21Kattyjust remember that boys!
17:45.22sevardKatty: or thought you were pretty.
17:45.27Kattyif she looks down, offer to get her tea.
17:45.34_-Jon-_hey everyone
17:45.38Kattyeven if she doesn't take you up on it, she'll feel better.
17:45.41sevardKatty: in kung foo you hit when they look away
17:46.00Kattysevard: would you shut up and take me seriously, you goofball.
17:46.12ZeeekKatty I am so resisting answering the above statement about "looking down"
17:46.29Kattyoh dear.
17:46.40_-Jon-_I'm having some slight issues with my Asterisk setup and wondering if there's any way to pinpoint the problem.   Basically when a call comes in, 9/10 times it works, but sometimes it doesn't.  And I'm not sure if it's Asterisk, BroadVoice, or what
17:46.40Kattyfor /once/ don't twist what i say :P
17:46.46Zeeekand you weren't speaking in the first person so I could have
17:46.54*** join/#asterisk Seggy (i=rbutler@tsss.org)
17:46.59Zeeekbut as a gentlement... didn't
17:47.02Kattygood job.
17:47.07_-Jon-_I can sit here all day and place test calls and they'll all work, but sometimes other people call and it won't come through
17:47.08Zeeekagain!
17:47.14Kattythe Good Girls like gentleman.
17:47.28Kattyif you're not a gentleman, we're not going to waste our time
17:47.33Kattyor at least i wouldn't.
17:47.50Kattybut then again i'm insanely picky :P
17:47.54ZeeekI can't believe how stupid some people are! This woman says "just email me your credit card info"
17:48.06Zeeekyeah right, let me send my carys with that too
17:48.17Zeeek<PROTECTED>
17:48.26*** join/#asterisk schirpich (n=kvirc@ip21.farheap.net)
17:48.28Kattyif she's asking you for the credit card number, and she's not your wife or a sales rep....
17:48.32Kattythen don't waste your time
17:48.49Zeeekno it's legit, just a stupid person with the title "Vice Presifent"
17:49.10Kattythere are still good people out there :P
17:49.11Zeeekproving that women *can* be as stupid as men sometimes :)
17:49.30Kattyyeah, we can all be stupid sometimes though
17:49.34Kattywhether intentionally or not.
17:49.40*** join/#asterisk Lord_Drachenblut (n=Lord@12-210-117-62.client.insightBB.com)
17:49.42Zeeekplus all my tollfree are down and I can't reach her for what would take 1 min on the phone
17:50.07KattyHmmhesays: you around?
17:50.25schirpichIm kinda green when it comes too asterisk, but i was wondering if anyone could point me in the right direction.  I wanna have a SIP polycom 301 phone connect to my work asterisk server through our vpn,  how would this be done?
17:50.52schirpichdo i use something like a sipproxy?
17:51.00_-Jon-_Is there any way to have the SIP debugging information saved to a file?
17:51.58_-Jon-_schirpich, just tell your phone to connect to IP of the Asterisk server that goes through the tunnel
17:53.12schirpichi get what your sayin, but the phone would still have to be connected to the vpn some way.  i wanna make it so one of these phones can be plugged into an internet connection anywhere and connect up to my asterisk server
17:53.22HmmhesaysKatty: aye
17:53.25Hmmhesaysi'm avoiding sevard
17:53.51KattyHmmhesays: how often would you say you get a drink for a girl?
17:54.01Hmmhesaysa girl that i'm trying to woooo?
17:54.08_-Jon-_schirpich, hmm, i don't know if that's possible.  I have mine connected through a vpn, but the VPN tunnel is done with my router
17:54.09KattyHmmhesays: a girl in general.
17:54.15KattyHmmhesays: whether you're trying to woo or not
17:54.19KattyHmmhesays: your mom does not count
17:54.20schirpichwhat kind of router?
17:54.21Hmmhesaysit's different in each case
17:54.31KattyHmmhesays: 3 or 4 times a week?
17:54.36_-Jon-_schirpich, Linksys WRT54G running openwrt
17:54.55HmmhesaysI don't buy girls drinks unless they are 1. a good friend or 2. a girlfriend
17:55.38schirpich-jon- that a modded router right?
17:55.54_-Jon-_schirpich, yup.  flashed with openwrt firmware
17:55.59KattyHmmhesays: not even if they look really sad?
17:56.14schirpich-jon- are there any home/residental routers out there that have support for connecting to a vpn?
17:56.18Hmmhesayswhy I need to buy them a drink?
17:56.18KattyHmmhesays: you mean to tell me if i was moping in the corner somewhere, you wouldn't offer to get me a drink?
17:56.30HmmhesaysI don't see that at the bars I go to
17:57.01Kattyk
17:57.13_-Jon-_schirpich, i think so.  they might be a little bit more expensive.  but you can buy a ton of different routers and run openwrt on them, not just Linksys
17:57.14Hmmhesayswhy do you ask?
17:57.47KattyHmmhesays: if you ever /do/ see some mopey chick at a bar.
17:57.52KattyHmmhesays: get her a drink, k?
17:57.55Hmmhesayswhy?
17:57.57schirpich-jon- i'll have to look into that
17:58.04KattyHmmhesays: because it will turn her entire day around
17:58.07Hmmhesaysok this guy is really getting on my nerves
17:58.12Hmmhesaysunless she wants to be alone
17:58.23KattyHmmhesays: she can be alone.
17:58.27_-Jon-_brb
17:58.36KattyHmmhesays: but the offer is still a really great gesture.
17:58.54KattyHmmhesays: even if she won't take you up on it :P
17:59.14*** join/#asterisk C4T3l (n=rcall01@216.54.143.2)
17:59.56C4T3lhowdy asterisk buddies!
18:00.13Zeeekare we still on that? After the call to the Vice President, I can now relax
18:00.30KattyZeeek: yes, we are.
18:00.31HmmhesaysC4T3l: hi Dr. Nick?
18:00.32KattyZeeek: it's important.
18:00.38KattyZeeek: also, coupes are too
18:00.39*** join/#asterisk MacDome (n=eseidel@A17-255-100-181.apple.com)
18:00.46KattyHmmhesays: are you driving?
18:00.56ZeeekWell, if I offer something to a woman to make her feel better, she should take it!
18:01.08*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)
18:01.11HmmhesaysKatty: I got thrown in jail
18:01.11KattyZeeek: sometimes it not polite to accept
18:01.13ZeeekI'm talking about herbal tea, obviously
18:01.18KattyHmmhesays: still no driving then :<
18:01.24Hmmhesays1 year
18:01.25KattyHmmhesays: you can still help me pick out a car!
18:01.46KattyHmmhesays: are you goign to be alright?
18:02.01HmmhesaysKatty: do I have a choice?
18:02.03KattyHmmhesays: getting attached to cars can be emotional :<
18:02.11KattyHmmhesays: sarcasm, dear ;)
18:02.19ZeeekQuestion to the world: now that most companies use voIP, is it possible that some calls will not be properly routed and won't work between some carriers?
18:02.22syzygybsdI am having trouble with my sip.conf file.  I would like to be able to make incomming and outgoing calls on the same sip connection, however don't you need a register => and a context?
18:02.26HmmhesaysI'll be alright when I get the fuck outta this town
18:02.39Kattyyes, you're doomed there.
18:02.47justinuLA?
18:02.52Hmmhesayssyzygybsd: what an impossibly vague question
18:02.55C4T3lsyzygy
18:03.05*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
18:03.06C4T3lmore info plz
18:03.08KattyHmmhesays: i've decided i want a manual coupe of some sort.
18:03.12Zeeekno, vague is "Newbie-pls help"
18:03.22Hmmhesaysspelled poorly
18:03.28Hmmhesaysbetter make sure you can drive a manual
18:03.35justinuanyone can drive a manual
18:03.39Hmmhesaysnot true
18:03.43Kattymom can't
18:03.45[TK]D-Fendersyzygybsd : No, you don't necessarily need to register
18:03.48Hmmhesaysthis is the USofA buddy
18:03.49Kattyi can sorta drive a manual
18:03.51justinuthey just didn't try hard enough
18:03.55Kattybut i've not had a lot of 'sperience with it
18:04.03justinuif my wife can drive a manual, anyone can
18:04.05HmmhesaysAutomatics are the transmission of choice up here
18:04.18syzygybsdfunny, I tried to give enough.  Ok, simple sip connection to another server.  if I want to accept incomming calls, I need a register =>
18:04.26Hmmhesayshow many clutches you go through a year
18:04.31*** join/#asterisk Danett (i=none@a144029.upc-a.chello.nl)
18:04.31KattyHmmhesays: i think it really comes down to I Love Exhaust.
18:04.31justinunone
18:04.32Zeeekmakes talking on your cell while driving so much easier
18:04.32Danettheya
18:04.47syzygybsdif I want to make outgoing calls, I need a context in sip.conf with all the info so I can include that in my dial() statement
18:04.48[TK]D-Fendersyzygybsd : You only need to register if the other side doesn't know your IP
18:05.06[TK]D-Fendersyzygybsd : Thats the reason for its existance.
18:05.08DanettWhen you run a asterisk box, wich is attached to a domain, is it possible to dial to sip://user@domain.tld?
18:05.23syzygybsd[TK]D-Fender: I don't think they do...
18:05.37[TK]D-Fendersyzygybsd : Thats why then.
18:05.46justinui live in LA, i know all about people who buy automatics so they can put on makeup/eat/read/bucketmouth/beat their children/etc
18:06.23[TK]D-Fenderjustinu : Seen Pink's video for "Stupid Girls"? :)
18:07.05justinuno
18:07.11C4T3lIs the X-lite soft phone inherently choppy when it comes to the audio stream?
18:07.19justinupop culture usually eludes me somehow
18:07.42*** join/#asterisk dwildes2 (n=dwildes@209.164.237.195)
18:07.44justinui watched some guy crash his vette into a center divider because he was too busy smaking his kid
18:08.16*** join/#asterisk cytrak (n=kvirc@adelphi.geofocus.com)
18:08.21[TK]D-Fenderjustinu : Go watch!
18:08.49DanettHow can i setup asterisk so it can relay sip calls made to user@domain.tld?
18:09.20[TK]D-FenderDanett : * is not a SSIP Proxy.  Go look at SER
18:09.42C4T3lDanett: I think SER is better for that.
18:09.50DanettI am registere with x-lite to my asterisk box which sits on a certain domain. I want other agents to be able to call to user@domain.tld and reach my softphone
18:09.54Danetthmm ser is an option
18:09.56justinuyou can terminate them pretty easily, by defining an exten => user
18:10.57C4T3lDanett: Have them register with yer *
18:11.05Danettwell
18:11.09Danetti don't want that really
18:11.17cytrakis it ok to have two TE110P cards in one box ?
18:11.17Danetti want them to be registered at *some* provider
18:11.44[TK]D-FenderDanett : Does domain.tld point to your * box direct?
18:11.55Danettyes it does
18:12.04Danetton standard ports
18:12.05[TK]D-Fendercytrak : Cure, though You're better off with a single multi-port card
18:12.13Danettit points to voice.pbxservices.nl
18:12.16[TK]D-FenderDanett : Then you can already do that.
18:12.27DanettI am not able to reach it...
18:12.43C4T3lDanett: is sip reg failing?
18:12.50Danettdoesn't show any debug
18:12.55[TK]D-FenderDanett : pastebin your sip.conf
18:12.57Danettit think x-lite cannot local loop
18:13.33Danetthttp://pastebin.com/694602
18:13.42cytrakI've been trying to setup my * to talk to a siemens PBX using a single span card and it works fine for internal calls between soft<->hard phones but the soft phone can't get a way out to the world
18:14.29cytrakweek configuration wouldn't work with me
18:14.35cytraki mean wink
18:15.36[TK]D-FenderDanett : that is not SIP.CONF.
18:15.44Danettsorry
18:15.46Danettpasted the wrong one
18:15.49Danett;)
18:16.03Danettwhich part do you want?
18:16.29[TK]D-FenderDanett : In you domain.tld server you'll need to set allowguest=yes, and set up a context.  in extensions.conf you'll create that context containing the names you with to allow and then tell * what to do with it.
18:16.42[TK]D-FenderDanett : Just follow that and let us know after.
18:17.06Danettok
18:17.08Danettthanks
18:17.52*** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane)
18:18.06Hmmhesays2 more days in the hole
18:18.07Hmmhesaysugh
18:18.26C4T3l2 days??
18:18.33Danett; By default, all domains are accepted and sent to the default context or the
18:18.33Danett; context associated with the user/peer placing the call.
18:18.36C4T3lwhat you mean?
18:18.51*** part/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)
18:23.26*** join/#asterisk stoffell_h (n=stoffell@81.83.249.224)
18:24.02Danett[TK]D-Fender
18:24.05Danettdone all that
18:24.10Danettcan you try it for me?
18:24.18syzygybsdhow can I just accept all calls over sip?  It is complaining that the authentication is bad because of how dial is different then register (ie it thinks the phonenumber being called is the user not extension)
18:25.27Danettyeah!
18:25.30Danettworking baby
18:25.32Danettthat's nice
18:27.51_-Jon-_is it possible to log debug messages to a file?
18:28.44*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
18:31.07*** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie)
18:31.19*** join/#asterisk Sixes (n=Cliff@73.Red-217-125-3.staticIP.rima-tde.net)
18:31.29Danettis there a way to set a maximum amount of minutes/ per month / per peer?
18:32.16Hmmhesaysthere is always a way
18:32.28Danett;)
18:32.34Hmmhesayswhether you want to write it, or pay to have it done is another question
18:32.47Danettmaking it is not the problem
18:32.48stoffell_hhm, is there no #asterisk irc log any more?
18:32.57DanettI just thought there was a native way
18:33.01Danetthave to go
18:33.02Danettbbl
18:33.52*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
18:34.39C4T3lanybody know what the cause of jittery audio streams through a voip connection.  I'll post my sip.conf if needbe.
18:34.54C4T3ltwo x-lite phones
18:35.04rpmls
18:35.08rpmerm
18:35.29rpmare they both over a wan connection?
18:35.40rpmor on a local network?
18:35.43Lord_DrachenblutC4T3l, what does the ping time between them look like
18:36.23C4T3lOoh, hold on... they are con over a WAN
18:37.17C4T3li beleive that icmp-echos are dropped on the other end!!
18:37.32rpmC4T3l: from what i remember the X-ten X-Lite client does not support much besides g723, alaw, and ulaw.. it will take at least a constant 64kbps on both ends. you can use the jitterbuffer in the sip.conf though and tweak it.
18:39.35[TK]D-FenderDanett : Sorry, no subdomain routing.  if it hits your * it all goes to the same place.  Not context control.
18:40.37*** join/#asterisk MacDome (n=eseidel@A17-255-100-181.apple.com)
18:40.55[TK]D-FenderDanett : You want something more, get a SIP proxy and multiple other * servers.
18:42.16C4T3lrpm: I cant find jitterbuffer in my sip.conf is there a standard value that should be set???
18:42.32vader--are any of you guys pulling your conf files from mysql?
18:42.40[TK]D-FenderC4T3l : there IS no jitterbuffer for SIP in *
18:43.00lunkC4T3l: is the machine slower?
18:43.02C4T3lI thought that jitterbuffer was an IAX thing
18:43.08[TK]D-FenderI believe there is a flakey experimental one in the works, but nothing worth trying.
18:43.10tzangerC4T3l: nope
18:43.15lunkC4T3l: i have a p2-300 i do testing on, and it's sound quality is soo crappy
18:43.21tzangerthe sip jitter buffer is the same as the IAX2 one
18:44.12justinu64kbps is just the g711 payload, you need more like 80kbps to account for ip/udp/rtp overhead
18:44.38C4T3lthe machine is a celeron 2.5 GHz with a gig of ram just sitting at my house
18:44.47C4T3lnothing really runs on it
18:45.29C4T3li'm using a high-speed cable connection and so is the far end
18:46.02*** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV)
18:46.45C4T3lvader--: at work we pull our sip.conf from mysql
18:47.07[TK]D-Fendertzanger : * needs to add support for preferred codec settings on a per-peer basis like "allow=gsm:40" for 40 MS frams, ets
18:47.07tzangeryes
18:47.23justinuwouldn't that be something
18:47.29[TK]D-Fendertzanger : Major savings ther...
18:47.34[TK]D-Fenderjustinu : it would...
18:48.14[TK]D-Fenderjustinu : Also nifty for IAX2 as well.... if possible
18:48.24starleinis that a correct sql structure of the cdr table: ALTER TABLE `cdr` ADD INDEX ( `uniqueid` ) ?
18:48.28[TK]D-Fenderjustinu : Depending who gets to negociate.
18:48.33starleinbecause on voip-info i saw an INDEX on dst,calldate,accountcode ?
18:49.17*** join/#asterisk Gamercjm (n=chris@pool-71-254-177-36.lsanca.fios.verizon.net)
18:49.21justinud-fender: i won't hold my breath
18:49.32starleinah i think it should be FULLTEXT
18:50.57[TK]D-Fenderjustinu : Whats scary is I don't actually see this is that big a problem, maybe its just me.... but that doesn't necessarily change my odds.
18:51.43justinuit seems like a lot of "little things" get held up like that
18:52.04*** join/#asterisk caloi (n=caloi@nat-66-218-1-139.usadatanet.com)
18:53.36[TK]D-Fenderjustinu : Besides I have other priorities like some Queue basics (bug fixes), SIP-B, and a prayer for Polycom ACD support.
18:54.17starlein720 rows in set (2 min 17.79 sec)
18:54.17[TK]D-Fenderjustinu : PRI 2BT is up there for a lot of people.  The sort of stuff we should concentrate on.
18:54.19starleinoh damn
18:54.32justinui thought it did support 2bct?
18:54.36[TK]D-FenderWhere database performance is anything but!  Whe!
18:54.52tzangerI'd LOVE to use 2BCT
18:54.56[TK]D-Fenderjustinu : Maybe I'm just a little out of the loop, or its on different tech
18:55.22justinui remember hearing that at astricon, but i have no experience trying to make it happen
18:55.46tzangerour telco doesn't support it
18:55.50tzangerat least not to my knowledge
18:56.03justinudo they support NI2?
18:56.13tzangeryes
18:56.19tzangerthat's what my switchtype emulation is set for
18:56.19justinu2bct is part of NI2, but has to be "provisioned"
18:56.30[TK]D-FenderWhat else does NI2 offer over NI1?
18:56.45justinuum, stuff like ANI-II delivery, iirc
18:56.50Lord_Drachenblutany good suggestions of a distro of linux to run asterisk on
18:57.02justinui can't remember all the differences anymore
18:57.16justinui sorta turned my back on legacy telco stuff :)
18:57.29*** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com)
18:57.35tzangerjustinu: hmm... does Asterisk provide proper CDR with 2BCT?  Or is it similar to the IAX2 media path minimization where there is no callback or even attempt to callback to update CDR?
18:58.19CunningPikeLord_Drachenblut: Take your pick, really - I think it runs on almost anything. My advice is go with the distro you are most comfortable with
18:58.38justinutzanger: proper CDR? lol
18:58.39brif8besides the book "Asterisk the future of telephony"  what other good source material is there? I have played around with * but when I read the book it made a lot of puzzle pieces fall in place
18:59.09justinutzanger: actually, i have no idea... but I have little faith in CDRs WRT asterisk
18:59.19CunningPikebrif8: The Book, The Wiki and The List and This Channel
18:59.23justinu~docs
18:59.24jboti guess docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
18:59.54Lord_DrachenblutCunningPike, bout the answer i was expecting.... was just wanting to make sure that there was one to avoid completly
19:00.29CunningPikeLord_Drachenblut: Others may chip in here, but I believe SuSE can be a pain, but I have no direct evidence
19:00.34[TK]D-Fenderbrif8 : What specific aspects?
19:02.44Hmmhesayshouse is such a freaking good show
19:03.10drfoomod2is anyone using Asterisk RealTime?
19:03.43Lord_Drachenblutwhat is asterisk realtime
19:04.06drfoomod2http://www.voip-info.org/wiki/view/Asterisk+RealTime
19:04.19Lord_Drachenblutthere now
19:04.19Lord_Drachenbluttell me your version
19:04.23NuggetLord_Drachenblut: asterisk with a database backend instead of flatfile config files, and a really confusing name.
19:05.15C4T3lDoesn't Asterisk Realtime allow you to change exensions.conf on-the-fly?
19:05.31*** join/#asterisk pif (n=ldm@zenon.apartia.fr)
19:05.33Nuggetplain asterisk lets you do that too, just less efficiently.
19:05.53pifI need some guidance on the new jitterbuffer ...
19:06.02tzangerpif: hunt down oej
19:06.16g__Question for anthm, if you're around.. the wiki suggests we avoid using the 'r' flag on Dial().. is that still true?
19:06.22oejHunt down zoa instead
19:06.25oejIt's his code, not mine
19:06.26pifoki
19:06.28tzangerg__: you NEVER use the 'r' option unless you know WHY you need it
19:06.38anthmr to generate ring ?
19:06.39tzangeroej: :-)
19:06.41tzangerbut you run faster
19:06.44g__tzanger: because it doesn't work unless I use it?
19:06.45oej:-)
19:06.48g__<PROTECTED>
19:06.52pifis jitterbuffer=yes enough or must I adjust other paramerters?
19:06.53tzangerg__: explain "doesn't work"
19:07.02tzangeranthm: yes.  basically replace early audio with ring
19:07.10tzangerwhich is only infrequently what you actually want to do
19:07.10filetzanger: is it working...
19:07.24tzangerfile: not a crash yet...  <knocks on file's head>
19:07.28filesweet
19:07.34anthmif you are calling a circuit for instance
19:07.37tzangerfile: so what precisely did you do?
19:07.37g__I'm dail()ing a sip channel and a ZAP channel with a 30 second timeout.  There's no ringing sound unless I use 'r'.
19:07.41anthmreal early media is better
19:07.46anthmcos you have realtime ring
19:07.53tzangeranthm: exactly
19:07.57anthmyou can trick ppl with r if it ends up being busy
19:08.06anthmso you get RRRING RRING BUSY SIGNAL
19:08.13SixesCan someone help me with a DTMF problem please?  On the Wiki page for the ZyXEL P200W, smeone's written "DMTF relay: outband did the trick" .... I have no clue what he means. :(
19:08.15tzangeranthm: well your PBX just starts to sound like ass if you hear RING RIBEEP BEEP BEEP
19:08.18tzangeroh you beat me
19:08.20tzangerheh
19:08.27*** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk)
19:08.42websaesixes: i have that phone
19:09.02Sixeswebsae: can you make DTMF work?
19:09.11websaemine works
19:09.12tzangerg__: what's your Dial command?
19:09.13g__Ok, I hear you both tzanger and anthm.  so the next question is how do I figure out how why it isn't ringing by itself.
19:09.28tzangerdialing zap should always give you early audio, even if it's just the zaptel driver giving it to you
19:09.29anthmwhat is the path of the call
19:09.32marcus2has anyone heard any updates about nufones 888 service?
19:09.47tzangermarcus2: not I, no
19:09.53marcus2sigh
19:10.08tzangerI've been using my asterlink termination since this happened, haven't turned up nufone's yet
19:10.19marcus2now my 888 is in limbo :(
19:10.25tzangermarcus2: that sucks :-(
19:10.37tzangerI've always been leery of origination for that very reason
19:10.43Sixeswebsae: what setting do you use for DTMF in sip.conf?  rfc2833 ?
19:10.46marcus2i see that voicepulse connect is offering inbound 888 now
19:10.49tzangertermination I can flip to whomever by dialing a few digits
19:10.56marcus2maybe i should see if i can port it to them
19:11.16g__It's basically Dial("SIP/itd371hw&Local/6135554848@itd01-out,30,r")..
19:11.16brad_msswi've had the best luck with junctionnetworks ...
19:11.25g__Err, without the 'r'.
19:11.26brad_msswit takes about 2 days to port an 800/866/888/877 number
19:11.46tzangerok so it's SIP and Local/ not Zap/
19:11.56brad_msswglad we didn't go with nufone here :)
19:11.57anthmwhat is the path like phone->sip->ast->zap etc
19:12.01g__Local/ does eventually call Zap/ though.
19:12.03*** join/#asterisk ringhals (i=fwuser@firewall.drgutah.com)
19:13.19g__The path is ast->sip and ast->Local/6135554848@itd01-out->Zap/g1/6135554848
19:13.35ringhalsOk .. new to the irc thing.. is this a channel that I can ask questions about configuration etc in?
19:13.37anthmwhat about into ast
19:13.39anthmwhat is that
19:14.12g__anthm: you mean the channel that placed the call?  It's a SIP client.  so sip->ast.
19:14.17lzhanganybody ever use app_machinedetect?
19:14.18justinu~suggestions
19:14.20jbotit has been said that suggestions is 1) Don't ask to ask. Just say your problem, 2) Don't repeat until 5 mins after, 3) Read and re-read the docs first, then admit it if you REALLY don't understand. You're wasting your time and ours if you haven't at least tried. 4) If your problem ain't solved, come back in 12 hrs or 24 hrs later. We're very international. 5) Be ...
19:15.01anthmif you make an exten that you can call with the sip phone that just goes to zap as a test does that work?
19:15.45g__Yes.  Actually, the scenario I described above works for some people but not for others.
19:17.05anthmit may be because of the &'ed together channels
19:17.10*** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie)
19:17.15anthmcos in that case you are not one on one with 1 channel
19:17.23anthmyou are waiting for one of them to answer first
19:17.32syzygybsdSOB!
19:17.45g__So what is the expected behaviour for multiple channels?
19:17.52syzygybsdMay  2 13:08:31 NOTICE[24932]: chan_sip.c:8727 reload_config: Unable to load config sip.conf, SIP disabled
19:17.55CunningPikeringhals: Shoot
19:18.07syzygybsdwhen I get that line, it works, when I fix the configuration it doesnt'
19:18.08anthmsit there in silence till one or the other answers
19:18.45g__And the call progress information--just don't pass it on at all?
19:18.46anthmthat would for sure be a case where you have no choice but to provide ringback or something or other
19:18.48*** join/#asterisk gursikh (n=guriskh1@adsl-68-93-88-61.dsl.hstntx.swbell.net)
19:18.52g__(Because sometimes it does.)
19:19.51anthmwhat is the reason for calling sip and zap at the same time just for optional answering in 2 locations
19:20.01*** join/#asterisk jjwx (n=samuel@levinux.UQAR.UQUEBEC.CA)
19:20.03jjwxhi
19:20.26ringhalsOk.. I have been trying to get chanspy to work for me with little success. I have a number of devices set up and working. For example I have an iax device and a sip device set up. I can place and receive calls from both. I also have the extension exten => 8102,1,Chanspy(SIP|g) however I get no audio and the Chanspy commands do not seem to work. So to clarify I place a call from the sip device. then from the iax device dial the spy extension, if I
19:20.42g__anthm: basically.  It's a personal extension with a hard phone and a cell phone.
19:20.51anthmmaybe you can use the "please hold while i try that extension" file
19:20.55g__(err, mobile phone--depending on which English you speak.)
19:21.00anthmor the m opt for hold music
19:21.18anthmdoing r will seem ok but it will make no sense if you end up with a busy tone
19:21.23jjwxis it possible to do a regex in asterisk? I mean, something like Set(BLEH = /SIP:(.?)@.*/)
19:21.32*** join/#asterisk gr0mit_home (n=wendolen@extrt.txrx.org.uk)
19:21.58g__Ok, I'll be sure not to send them a busy tone then.  I've been opting for a "no one is available to take you call at the moment.. please try your call again later" for extensions without voicemail.
19:22.02anthmi recall making the regex diaplan function dont know if it's still there
19:22.43jjwxshit man, regex is a must...
19:22.45g__Thanks anthm.  I'll add a couple of notes to the voip-wiki documentation about this.
19:23.48anthmyah i agree I made a perl compatable regex mod for another project i am working on
19:23.59schirpichwhat would the correct terminology be for the recording you hear on the IVR when you call into a queue
19:24.15Zodiacalanyone know how i can create a .call file to use for testing? i tried to make one, and it works, but it hangs up after like 20 seconds.. anyone know why it doesn't stay connected longer? http://pastebin.com/694770
19:24.24justinug__: what are you gonna do when your cell phone voicemail picks up?
19:24.58anthmyah listen to justin i think he is well aware of this snafu =p
19:25.03justinu;)
19:25.22justinui had to write a patch to AMI, and a manager app to make it do what I want
19:25.36dpryoIs it possible to detect the "referer" from a remotely transfered call? (Say, detect that the call has been transfered from my cellphone-number)
19:25.46*** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)
19:25.49generalhanwhats up all !
19:26.16justinudpyro: if you have PRI, it should be available
19:26.36dpryojustinu: Do you know which parameter? I have an E1.
19:26.36anthmand if it's within the pbx you can save it in a variable
19:26.40justinuRDNIS?
19:27.01justinuhttp://www.voip-info.org/wiki/view/RDNIS
19:27.01dpryoOkay, thanks. I'll look into it :)
19:27.06dpryoYeah, looking at it :)
19:27.20vader--can you pass an extension into the voicemailmain app?
19:27.21Zodiacalany ideas?
19:27.22dpryoWould be cool to use it for a voicemail solution.
19:27.27vader--so you don't have to enter your extension
19:27.33justinuof course, vader
19:27.43noname32anyone here use the Grandstream 2000?
19:27.45justinuVoiceMailMain(${CALLERIDNUM})
19:28.24drfoomod2has anyone suggested using SQLite>
19:28.24drfoomod2?
19:29.05drfoomod2oh?>
19:29.14vader--justinu thanks
19:29.18justinunp
19:29.20HmmhesaysWhat is more annoying that people that don't follow your precise instructions?
19:29.23pif_must_ I use trunktimestamps for jitterbuffer to work?
19:29.44HmmhesaysPeople that fail to follow your precise instructions repeatedly
19:29.51drfoomod2anthm: what is being stored in/with sqlite>?
19:30.14*** join/#asterisk PakiPenguin (n=Junaid@linuxpakistan/admin/pakipenguin)
19:30.24PakiPenguin~seen areski
19:30.33jbotareski <n=areski@polar.es6.egwn.net> was last seen on IRC in channel #asterisk, 40d 9h 24m 8s ago, saying: 'the real billable seconds'.
19:30.36g__justinu: we don't have voicemail on the company cell phones
19:30.48justinuso they just ring forever?
19:31.05anthmit can store config, be a dialplan switch, umm i forget what else
19:31.10anthman app i think
19:31.18anthmto do sql st
19:31.21g__Yup.  For values of "forever" that are more than 30 seconds, anyways.
19:31.25anthmanother one to select a row in to vars
19:31.35justinug__: lucky you then
19:31.50PakiPenguinjustinu, there's a "No Answer" tone after a certain number of secods
19:31.54PakiPenguinwhich actually shows the
19:32.03PakiPenguin"no answer" text on your cell screen
19:32.05g__But on my personal voicemail, I've been using the zap "push-# after picking up the phone" feature.
19:32.06justinuhere the crooks answer the call regardless
19:32.15ringhals~seen jsmith
19:32.17jbotjsmith is currently on #asterisk-doc #utah. Has said a total of 48 messages. Is idling for 8m 51s, last said: 'Wow... looks like all this talk about documentation has more people in here!'.
19:32.23justinug__: what's that?
19:33.09anthmand cdr i think
19:33.10g__justinu, look up option 'c' in http://www.voip-info.org/wiki/index.php?page=Asterisk+Zap+channels
19:33.52justinug__: interesting... didn't know about that... doesn't help me tho, since I don't use zap channels
19:33.59g__PakiPenguin: is that only when you place the call from your cell phone?
19:34.24PakiPenguinnope , from the pots line , i get a no answer tone
19:34.30PakiPenguinwhich is different from everything else
19:34.37PakiPenguinbut i think its a exchange dependent thingy
19:34.38g__justinu: yeah, I'm going to run into trouble if we switch from zap channels.  It'd be nice to build it into a Local/ or Dial() instead.
19:34.51justinug__: won't be easy
19:35.00g__I'm not volunteering..
19:35.06g__not now, anyways.
19:35.18justinui'm just warning you
19:36.05g__But if you want to do it for me.. I'd be more than happy to help you test it.
19:36.21[TK]D-Fendersemi-lazy question : Can the AMI be used to place calls in a similar manner to .call files?
19:36.30justinui've already done it in a technology agnostic way, but it's not very elegent
19:36.33*** part/#asterisk Sixes (n=Cliff@73.Red-217-125-3.staticIP.rima-tde.net)
19:36.41justinu[TK]D-Fender: yes
19:37.07*** join/#asterisk ph|ber (i=phiber@slackwaresupport.com)
19:37.12[TK]D-Fenderjustinu : Neato.... going to have to start playing around with this stuff...
19:37.34ph|beranyone have an example of multiple DID's that route to different extensions?? i cant seem to get this one working.
19:37.48justinuthe originate action is somewhat flawed, imo, but I fixed that too
19:38.44[TK]D-Fenderph|ber : What are the calls coming in on? (interface)
19:38.47anthmalong with the concept of actions and the entire manager interface itself.
19:38.54justinuheh
19:39.27ph|berzap
19:39.37justinupots line?
19:39.48justinuanthm: i can only imagine
19:39.58*** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com)
19:40.06anthmi did manage to add some patches to it eventually
19:40.25justinuwhy isn't the simplest of all apps "app_bridge" in the standard asterisk disto?
19:40.26anthmlike the one that lets you turn events off so you can have a command only pipe
19:40.35justinuisn't that the first thing you write when you write a switch?
19:41.24dwildes2ph|ber   for Zap lines, you'll have to create a context for each incoming channel
19:42.09ringhalsdoes anyone know of a good write up on the chanspy app? I have been over http://www.voip-info.org/wiki-Asterisk+cmd+ChanSpy and the info from show application chanspy but am still a bit lost.. not as to setting up the extension but as to how to debug the fact I get no sound when using it.
19:42.10anthmi can only guess but i'd say the first thing would be a solid core
19:42.19[TK]D-Fenderph|ber : Can you be a little more precise...
19:42.20ph|berwait, this is a iax connection
19:42.20ph|bersorry
19:42.43[TK]D-Fenderph|ber : Depends on how your provider works.
19:43.01anthmringhals at one point I could have since I invented it but i have no idea what they have done to it by now
19:43.02[TK]D-Fenderph|ber : Its possible you could use a single registration for multiple DID's or that each could be its own.
19:43.20*** join/#asterisk Renacor (n=kvirc@ip21.farheap.net)
19:43.23ph|beratm it is dialing all extensions with exten => s,1,dial(SIP/0960&SIP/0961&SIP/0962&SIP/0963&SIP/0978|20|tr)
19:43.27*** join/#asterisk R3DB0x (i=nobody@66.142.28.36)
19:43.37ringhalsbummer ... it is kicking my butt
19:43.41dwildes2ph|ber, are they passing the DID?
19:43.55ph|berthe extensions are relative to the last 4 of the #.. ie 555-555-1234 should go to extension 1234
19:43.58*** join/#asterisk fjean (n=fjean@201.29.130.118)
19:44.11ph|beryes.
19:44.16justinuhow, pri?
19:44.20ph|berand i tried exten => 18043430978,2,Dial(SIP/0978,20,m)
19:44.23ph|berdoesnt work.
19:44.40justinuthey might be sending you 4 digit dnis only
19:44.43justinuor 10 digit
19:44.43justinuor 7
19:45.00*** join/#asterisk MacDome (n=eseidel@A17-255-100-181.apple.com)
19:45.04ph|ber<PROTECTED>
19:45.08ph|berahh
19:45.11dwildes2try putting a underscore:
19:45.16dwildes2_18043430978
19:45.46justinuso they're sending you 7 digit dnis
19:45.49justinuthere's your answer
19:45.50dwildes2you can remove the 's' extension and see if you get a reject.  the reject should have the DID as they are passing it to you
19:46.08justinuexten => 3430978 ...
19:46.41*** join/#asterisk MacDome (n=eseidel@A17-255-100-181.apple.com)
19:46.42dwildes2justinu, yeah - looks that way  :)
19:46.49ph|berthis is what happens.
19:47.04ph|berhttp://pastebin.com/694842
19:47.59justinufucking amp
19:48.05justinu~amp
19:48.07jboti guess amp is "amp is, like, NOT supported here! people using it should join #freepbx (the new name of amp)"
19:48.07anthmringhals, if they still preserved the basic functionality you shuld be able to make exten => 1000,Chanspy and when you call 1000 amd every time you press * you will cycle active channels
19:48.55[TK]D-Fenderjustinu : Hello Mr. Bile!
19:48.59justinu;)
19:49.05justinujust following the trend
19:49.21justinuph|ber: the reason it doesn't work is because of the goto(menu,s,1)
19:49.43ph|ber?
19:49.56fjeanhi everybody , anyone can give me a hint on troubleshooting zaptel / ztdummy installation....
19:50.19ringhalsanthm thats basically what I have done. however for whatever reason there is no audio. I have open sip and iax channels but unless I ommit the |g I get nothing at all
19:50.58anthmwhat do you have in the g()
19:51.27ph|ber;exten => s,1,Goto(menu|s|1)
19:51.28ph|berthat ?
19:51.39ringhalslooking
19:51.41justinuyeah, it's throwing out the DID info
19:51.47ph|berit is commented out.
19:51.55ph|ber;
19:52.00justinuso then why do lines 13 and 14 print out in your pastebin?
19:52.19[TK]D-Fenderjustinu : I try to leave off the colourful expletives... its poorly received news as it is...
19:52.20ph|berafter it does not answer it rings all extensions
19:52.40anthmif you add g(wazzup) to it then you need Set(SPYGROUP=wazzup) at the top of every extension you want to spy on
19:52.50*** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk)
19:53.38ph|berexten => _X.,7,SetVar(FROMEXTEN=${DID${EXTEN}})
19:53.38ph|berexten => _X.,8,Goto(extensions|${DID${EXTEN}}|1)
19:53.38ph|berexten => _X.,10,Goto(menu|s|1)
19:53.54*** join/#asterisk IceManRISK (n=kart@201-40-207-74.mganm702.dsl.brasiltelecom.net.br)
19:53.56ringhalsI think I missed that actually..I did the g(29) but not the SPYGROUP
19:54.08brookshire_X. is bad!
19:54.11ringhalsI missunderstood what I read LOL
19:54.18anthmyah you need to set the var
19:54.36ph|berim walking into a previously install
19:54.51anthmand you can make it like _88XX to match g(${EXTEN:2}) so you can dial 8829 to spy on group 29
19:54.52justinuph|ber: ask on #freepbx, they're more familiar with that stuff
19:55.32ringhalsI will try that
19:55.36ph|berhrm this is 1.2.4 how old is that
19:55.38*** join/#asterisk ToTo (n=ToTo@host128-207.pool879.interbusiness.it)
19:55.42justinurecent enough
19:55.48ph|bernm. found it.. 2/06
19:56.02[TK]D-Fenderanthm : We need multiple spy/ring/other groups pe entry!
19:56.23justinuph|ber: problem is that AMP/FreePBX created that dialplan
19:56.35ph|berahh.
19:56.44ph|beri think it has AMP installed.
19:56.45anthmwould be not too hard
19:56.48*** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net)
19:57.00ph|berwhats the default uname and pass to AMP??
19:57.03ph|beradmin and amp111
19:57.17justinu~amp
19:57.18jbotrumour has it, amp is "amp is, like, NOT supported here! people using it should join #freepbx (the new name of amp)"
19:57.32ph|beryea yea
19:59.26*** join/#asterisk mtaht3 (n=m@reserve-64-79-114-30.wiline.com)
19:59.26fjeanany reasons why udev would not create the zap devices in /dev.... ?  with no hardware or such
19:59.37*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
20:00.36ringhalsso I tried to make it more simple.. I took out the |g etc and set my extension to Chanspy(SIP) placed the call from my sipphone and then dialed the extension for Chanspy from the iax device.. still no audio and a show channels does not show me bridging to anything.. any suggestions?
20:01.43anthm(SIP) means only spy on sip channels
20:01.55anthmjust put nothing in the ()
20:02.01anthmand you will get everything
20:02.11ringhalskk I will try that next
20:02.41*** join/#asterisk brockj49464 (n=brockj49@41.105.dhcp.hope.edu)
20:02.44*** join/#asterisk techie (n=gus@antibala.com)
20:03.04justinuanthm: you taking a break or something? not used to seeing you in these parts ;)
20:03.26Aursfjean: checked README.udev?
20:04.10fjeanAurs:  yes...I did it the same way for the two files, permissions.d and rules.d
20:05.03Aursfjean: ok
20:05.05fjeanAurs: I am using mandrake 10.1, I did the installation already four times on different computers using the same distro
20:05.32fjeanAurs: but this time, it does not want to  :- )
20:06.37Aurssounds nice ;)
20:06.57*** join/#asterisk xunil (n=wkurdzio@office1.visionpointsystems.com)
20:07.15fjeanAurs: hey I noticed my computer is a 686...dual CPU but actually just one
20:07.27*** join/#asterisk ToTo (n=ToTo@host128-207.pool879.interbusiness.it)
20:08.05justinu1337
20:08.43brif8~docs
20:08.44jbothmm... docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
20:10.38stoffell_his it allowed to use irclog to log the channel to a webpage?
20:10.40*** join/#asterisk Tangent (n=Tangent@connerdata.plus.com)
20:11.03dlynes_zaptel seems to autconfigure udev under zaptel 2.6.13 and higher, anyways...2.6.12 and earlier it seems you have to manually configure it
20:11.22dlynes_or, something to that effect
20:11.39dlynes_I've seen a number of people with 2.6.9 running into troubles with udev and zaptel
20:11.59*** join/#asterisk Eight (n=blake@12-227-169-99.client.mchsi.com)
20:12.06fjeandlynes: ok, cool
20:13.11sevardRiddle me this * gurus.  I have an Aastra 480i CT. I could make calls out but no audio.  I restarted *, no go, I restarted *, no go -- desperation, I restarted *, everything magically works.
20:13.29fjeandlynes: i'll take a look
20:13.34sevardAm I safe to assume * reliability == windows or would this be a configuration issue
20:13.50dlynes_fjean: which version of asterisk are you using?
20:13.59dlynes_erm
20:14.04dlynes_sevard, i mean
20:14.18sevarddlynes_ Asterisk 1.2.5
20:14.26gursikhJust wondering, what distro do you guys recomend to run * on?
20:14.30dlynes_sevard: upgrade to 1.2.7 or higher
20:14.40websaefedora core here :)
20:14.40dlynes_sevard: You'll find the sip subscriptions work infinitely better
20:14.41sevardgursikh: I run on slackware.  I recommend it for everything.
20:14.50dlynes_Slackware here, too
20:14.51websaedebian works nice as well
20:15.07gursikhoh websae, is that what you would recomend?
20:15.11sevarddlynes_: When I upgrade is there any sort of configuration changes between these versions?
20:15.19dlynes_From what I hear though, Gentoo has some special tweaks for Asterisk, too
20:15.35dlynes_but I'm not into running source code based distros for a telephony server
20:15.46*** join/#asterisk clive- (n=pirch@dsl-145-52-204.telkomadsl.co.za)
20:15.50*** join/#asterisk Ariel_ (n=Ariel@70.46.87.158)
20:15.52dlynes_don't want to have to do compiles on my main telephony server, degrading call quality for the users
20:15.56websaedlynes: I had heard that as well
20:16.03dlynes_sevard: no
20:16.17tzangerdlynes_: ever hear of 'nice' :-)
20:16.17dlynes_websae: yeah...gentoo also has a gnu autoconfigure script for asterisk
20:16.18Ikarusdlynes_: "special tweaks" ofcourse, the same ones like -funroll-loops ?
20:16.26Hmmhesayswho can we get on the case, we need peerrrrryyyyy maaaaasssssssson
20:16.26sevardAm I safe to assume upgrades are fairly painless as long as I put my builds into packages?
20:16.36DoktorGregI recently switched from slack to debian
20:16.42Hmmhesaysfor you sevard? no
20:16.44*** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com)
20:16.45dlynes_tzanger: still...i don't like the possibility :)
20:16.48*** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net)
20:16.57dlynes_sevard: extremely painless then
20:17.19dlynes_Hmmhesays: eh?  I use slackware, put everything for asterisk into packages...it's super easy
20:18.13dlynes_sevard: You might want to try installing a 2.6 kernel too, if you haven't done so already
20:18.47dlynes_sevard: 2.6 has low latency optimizations and ztdummy is automatically compiled for it
20:19.19Hmmhesaysdylnes i was using something not common in the geek wordl
20:19.21Hmmhesayshumour
20:20.04tzangerHmmhesays: ?
20:20.09Hmmhesayshaha
20:20.26fjeancould msec have an impact on the installation ?
20:20.41DanettDoes anyone know a provider which sells sip up?
20:20.44clive-fjean hardly imho
20:20.49fjeank
20:20.56Danettlike a local telefone number which endpoint is a sip user agent
20:21.05sevarddlynes_: I'm alreadying running 2.6 and I hav ea zaptel device
20:21.40blitzragehrmmm... still can't seem to get qualify to work as expected -- peers keep going REACHABLE then UNREACHABLE
20:21.51dlynes_Hmmhesays: Well, it usually helps if it's obvious
20:22.07dlynes_blitzrage: that's normal
20:22.14dlynes_blitzrage: as long as they become reachable again
20:22.33blitzragedlynes_: but when it goes unreachable, asterisk doesn't even attempt to place a call to that end point
20:22.50blitzrageI'd hardly say that is ideal
20:23.04dlynes_blitzrage: so your sip devices don't become reachable again?
20:23.04blitzrageand I realize its "normal" its been doing it since I started using Asterisk 3 years ago :)
20:23.19filewhat are you babbling about now blitzrage
20:23.19blitzragedlynes_: yes, they do -- but during that interum you can't send calls to that device
20:23.25blitzragefile: qualify! :)
20:23.32filewhat about it1
20:23.33dlynes_blitzrage: how long are they unreachable for?
20:23.33file!
20:23.45sevarddlynes_: Do you ever have any issues with DTMF and cellphones?
20:23.48syzygybsddoes anyone have a working configuration for connecting 2 asterisk boxes?  Anytime I try to send calls between them I get a "Failed to authenticate user ${phonenumberdialed}"
20:24.10syzygybsd*sip configuration
20:24.17blitzragefile: peers go REACHABLE then UNREACHABLE and back again -- causing some calls to be sent to voicemail when the peer is actualyl reachable (although asterisk doesn't see it that way :))
20:24.17dlynes_sevard: don't know offhand
20:24.25dlynes_sevard: define 'issues'
20:24.43fileblitzrage: ah, stupid SIP device that gives lesser priority to OPTIONS so it doesn't respond in time and thus Asterisk thinks it isn't there?
20:24.54sevard* thinks I'm entering 501 or 01 when I try entering 5001
20:24.55blitzragefile: seems that way --- polycom's mostly
20:25.10blitzragefile: although it happens with other Astersk boxes too :)
20:25.13filecan't say my Polycom does that option
20:25.15fileer often
20:25.37dlynes_sevard: Do you answer, then wait a bit, then do the ivr?
20:25.37filetherefore you must be a witch
20:25.49sevarddlynes_: of course.
20:26.04blitzragefile: :D
20:26.28sevarddlynes_: answer, wait 2, ivr asks for input, you enter in 5001, it says Sorry, that's an invalid extension.
20:27.11dlynes_sevard: weird...never had that problem
20:27.33dlynes_sevard: I think I might have had someone complain about that happening once or twice
20:27.39dlynes_sevard: but it's very rare
20:27.56sevarddlynes_: I have sipura 2002 ATAs going into a TDM400P that might have something to do with it
20:28.02dlynes_sevard: is this on a zaptel channel, or a sip channel?
20:28.09*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
20:28.42sevarddlynes_: a zaptel channel
20:29.45*** part/#asterisk Lord_Drachenblut (n=Lord@12-210-117-62.client.insightBB.com)
20:31.02*** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr)
20:31.57dlynes_sevard: try adjusting your gain
20:33.57*** join/#asterisk Skarmeth (n=Skarmeth@201009033182.user.veloxzone.com.br)
20:34.01Skarmethhi all
20:34.10*** join/#asterisk flop110 (n=ozverenm@162.27.103-84.rev.gaoland.net)
20:34.14flop110hello all
20:34.31flop110i have question about DACS modes on zaptel cards
20:35.21SkarmethI'm in trouble with a TE110P and a TDM04B. When I had only the TDM04B, everything worked fine, I was able to receive and make calls. But after installing a TE110P, I cannot receive calls or make calls anymore
20:36.12SkarmethI've a span configured for TE110P as 1-10 for b channels and 16 for d channel, and TDM04B card configured from 32-35 as fxsks
20:36.30OloBolawhat do you think a good per minute price is for 800 origination? 3ish?
20:36.52Skarmethat zapata.conf, all needed parameters, but when I make a call to a line pluged at TDM04B, I can't see no action on the console screen
20:37.36Skarmethany idea? /proc/zaptel/1 tell me it is a TE110P (as I put 1-10 for it) and 2 as a TDM04B (as I put 32-35 for it)
20:39.17syzygybsdwow, that took me way too long to figure out
20:39.20*** join/#asterisk heison (n=heison@ns.somanetworks.com)
20:40.00syzygybsdif you have a register => in sip.conf, don't have type = friend for that same connection, they will conflict and not allow incomming calls
20:40.28*** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com)
20:40.45Hmmhesayswhy is brian setzer such a damn good guitarist
20:41.11*** join/#asterisk StevenL (n=steve@216.62.85.65)
20:41.15clive-prolly because he works hard ata it:)
20:42.47*** join/#asterisk RoyK (n=roy@ti211310a080-13754.bb.online.no)
20:42.49sevarddlynes_: the gain on my zaptel card?
20:43.07dlynes_correct
20:43.11sevardclive-: you've apparently typed ata too many times
20:43.21sevarddlynes_: is that in zaptel.conf or in zapata.conf
20:43.33dlynes_sevard: zapata.conf
20:43.46dlynes_sevard: rxgain=, txgain=
20:43.56sevardshould I play with my gain or do you have suggested values
20:43.58dlynes_sevard: the values can be between -7 (minimum) and +7 maximimum
20:44.08dlynes_sevard: I'd suggest starting with 0, and adjust it from there
20:44.18sevardreally
20:44.24sevardHmmhesays: a maximum of 7
20:44.31dlynes_s/maximimum/maximum
20:44.32sevardHmmhesays: not 12.0
20:44.32*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)
20:44.40Hmmhesayswtf are you talking about
20:44.50sevardrxgain=12.0
20:44.50sevardtxgain=3.0
20:44.55DanettAnyone knows a international carrier which has sip connectivity?
20:45.01Hmmhesayshaha thats right, quintums gains are max 12
20:45.02Hmmhesaysmy bad
20:45.06Danettlike a reliable one for professional use
20:45.12Hmmhesayssevard I found you a russian bride
20:45.16Hmmhesayson www.bride.ru
20:45.17sevardHmmhesays: sweet!
20:45.24Hmmhesaysshe kind of has a mustache
20:45.31sevardsweet!
20:45.49dlynes_sevard: that's why your dtmf tones aren't working
20:45.57dlynes_sevard: the dtmf is all distorted like shit :)
20:46.20Hmmhesaysand her head is disproportionate to her body
20:46.26Hmmhesayshttp://www.bride.ru/ph/htcgi/ladies/461/461229P1.html
20:46.27Dr-Linuxhi folks
20:46.50Dr-Linuxanybody expert with "call by name directory" ?
20:47.15NuggetI turned it on and it worked.  Does that make me an expert?
20:47.30dlynes_Nugget: no doubt...never had any problems myself, either :)
20:47.40Nuggethow about if I'm not an expert but I do know the answer to the question you haven't asked yet.  Would that help?
20:47.45Nuggetor would you rather wait for an expert?
20:47.46HmmhesaysDr-Linux
20:47.58Hmmhesayscmd directory()
20:48.09Dr-Linuxopss
20:48.32Dr-Linuxwell, im thinking how can i ask my question, bcoz of damn english
20:48.33Dr-Linux:P
20:48.35Dr-Linuxwell,
20:48.57HmmhesaysI don't think i'm going to find a sugar momma on this site
20:49.18Dr-Linuxi want, when call routes to the caller, system should say " call is going to abc at extension xyz" ?
20:49.19DrukenHmmhesays: http://www.bride.ru/ph/htcgi/ladies/461/461655P1.html
20:49.22Hmmhesayshttp://bride.ru/ph/phcgi/ladies/457/457037P1.jpg this one kinda looks like she shiat herself
20:49.22Dr-Linuxsomething like that
20:49.37HmmhesaysDr-Linux: check out the directory app
20:49.39noname32anyone here use the polycom phones?
20:49.40Hmmhesaysdoes exactly that
20:50.06Dr-LinuxHmmhesays: what's the best way to have this feature?
20:50.19Hmmhesaysgo to the wiki and read about "cmd directory"
20:50.30HmmhesaysDruken: she's got kind of a chubby face for being 106lbs
20:50.39DrukenDr-Linux: when your people record their name, it should be <name> <extension>
20:50.43sevardAwesome
20:51.05DrukenHmmhesays: very true, but i'd still give her something to put in it :)
20:51.13Hmmhesaysyeah a freaking sammich
20:51.14znoGchan_sip.c:1210 retrans_pkt: Maximum retries exceeded on transmission 50f2fa9234600011222e38ec220cad6f@192.168.136.67 for seqno 102 (Critical Request)
20:51.16sevardDTMF works flawlessly
20:51.21flop110on a PRI link, if i place one double E1 between my PBX and my telco and if i read audio buffer in userland and
20:51.22znoGanyone know what would cause that?
20:51.24sevardI love all of you.
20:51.29znoGpackets being sent to 192.168.136.67 but not getting back?
20:51.39*** part/#asterisk StevenL (n=steve@216.62.85.65)
20:51.49*** join/#asterisk riksta (n=rick@62.6.163.87)
20:52.06flop110write these audio buffers on HDLC drivers, its will have a repercution on performances ?
20:53.27Hmmhesaysgeebus i swear these girls don't eat
20:54.17dlynes_znoG: probably a conversation that's long since ended, but for whatever reason asterisk still insists on sending sip packets to ti
20:54.22*** join/#asterisk papo (n=mathias@adsl-177-161-fixip.tiscali.ch)
20:54.39dlynes_znoG: I see that regularly in the debug logs
20:55.38znoGdlynes_: whats happening is that when I call in via Zap, it should dial SIP/3000 and on the calling end, i hear a busy tone. On the asterisk side, i see that maximum transmission message that I just pasted.
20:56.08dlynes_znoG: ah...there's probably other errors as well
20:56.26dlynes_znoG: but you're probably getting spammed so much with the retrans_pkt messages that you don't see them
20:56.50znoGthats the thing, I debugged the peer IP where the SIP extensions sits and there's no clear error message or stuff like that
20:57.11znoGand if i call SIP->SIP, its fine... only when coming in via Zap doesn't work
20:57.15dlynes_znoG: what's your logger.conf setting for that log file?
20:57.32dlynes_znoG: I get the feeling it's your dialplan for zap
20:57.52znoGdlynes_: but i can see in the asterisk console its trying to dial the SIP extension just fine
20:58.02dlynes_znoG: Oh
20:58.07znoGand as soon as it starts to ring, the SIP extension keeps ringing and the Zap channel drops
20:58.14*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)
20:58.31Dr-Linuxsorry, i got d/c
20:58.39Dr-Linuxsorry my question was about Dirctory
20:58.47dlynes_znoG: so you're not getting a SIP 40x error message?
20:58.55Dr-LinuxHmmhesays: any clue, maybe i missed your answer
20:59.17HmmhesaysYou want people to dial by name right?
20:59.29vader--do you guys know what this line will do
20:59.30vader--exten => s,1,Playback(transfer,skip)
20:59.36papoHi. I set up my asterisk-server with sip. It's not behind NAT. When I call in with a client which is behind nat using a sip provider, I don't hear anything. Though I see in the log that the files I indicated in the Playback() application are being played
21:00.03*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
21:00.11Dr-LinuxHmmhesays: that's already working fine, but i want as the call routes to the callee, system should announce extension
21:00.24Dr-Linuxso the caller should know at what extension he called.
21:00.41Dr-Linuxi don't know why asterisk doesn't provide this feature.
21:00.44HmmhesaysSo you want to complicate things
21:00.51papoI have to mention that the client is behind the NAT of which the server asterisk is running on is the gateway, but that shouldn't matter theoretically
21:01.01Dr-LinuxHmmhesays: yes maybe
21:01.31Dr-LinuxHmmhesays: by default call by name app is very easy
21:01.46Dr-Linuxbut i want that option which i described
21:02.06HmmhesaysI could probably write you a patch, if motivated by monitary compensation
21:02.26dlynes_Dr-Linux: The feature you're asking for is 'Call Announce'
21:02.35znoGdlynes_: nope, just a temporarily not available
21:02.54Dr-LinuxHmmhesays: i didn't understand your last words?
21:02.55dlynes_znoG: Yeah...that phone might have do not disturb enabled, then
21:03.10Dr-Linuxmotivated by monitary compensation? :S
21:03.14dlynes_Dr-Linux: he's saying he can add that feature to asterisk if you pay him
21:03.22dlynes_s/monitary/monetary
21:03.31Hmmhesayshaha my typing skillz are bad today
21:03.52Hmmhesaysgood call dlynes_
21:04.01dlynes_Dr-Linux: The grandstream 102's have that feature to announce the number coming in
21:04.02znoGdlynes_: nope, it's not that as i can ring that extension directly
21:04.09Dr-Linuxi'd love to pay, if i was in US or CA,
21:04.10*** join/#asterisk nagl (n=nagl@86.59.54.237)
21:04.11dlynes_Dr-Linux: However, they don't announce the name of the person
21:04.21Hmmhesayspakistan doesn't have paypal do they
21:04.36dlynes_Dr-Linux: OTOH, the Dial application(?) has a call screening feature
21:04.40Dr-LinuxHmmhesays: yes, but not that much money i have
21:04.52Hmmhesaysyeah no one wants to pay
21:05.01dlynes_znoG: perhaps the zaptel channel is trying to ring a different account on the sip device than the other sip devices are
21:05.10Dr-Linuxdlynes_: do you understand what i want?
21:05.24dlynes_Dr-Linux: Yeah...and Asterisk already has that feature
21:05.41znoGahh i fixed it!
21:05.43HmmhesaysDoes directory announce the extension its calling?
21:05.46dlynes_I think it's called the privacy manager
21:05.46Dr-Linuxdlynes_: can you guide me how can i do that?
21:05.53znoGdlynes_: the whole problem is me trying to use the latest version... asterisk 1.2.4 works fine
21:06.01dlynes_Dr-Linux: Use the force, use the search engine on voip-info.org
21:06.15dlynes_znoG: which phones are you using?
21:06.30Dr-Linuxdlynes_: yes, but what should i search in voip-info.org?
21:06.44dlynes_Dr-Linux: or try the applications section under asterisk
21:06.50dlynes_Dr-Linux: Call Privacy Manager
21:07.13dlynes_Dr-Linux: also try call screening, or call announce
21:07.18Dr-Linuxdlynes_: i tried much but still no luck
21:07.28dlynes_Dr-Linux: watch...i'll get it first try :)
21:07.47Dr-Linuxdlynes_: that's fine, but i need this option in "call by name directory"
21:07.58Dr-Linuxdlynes_: ok
21:08.19dlynes_I have no idea about call by name directory, but if it works, it works
21:08.46Dr-Linuxdlynes_: it's not a feature .. that's why i'm asking
21:08.51sevardHmmhesays: sorry for humping you
21:08.57Dr-Linuxbut asterisk should have this feature.
21:09.04sevardactually sorry for the delayed response, i couldn't see and had to get some visine
21:09.24dlynes_http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+PrivacyManager
21:09.25dlynes_first try
21:09.53anthmforgot who asked but here you go http://bugs.digium.com/view.php?id=7072
21:10.05anthmchanspy with mutiple groups
21:10.19Dr-Linuxsmacks? :P
21:10.36dlynes_anthm: heh...nice plug for cluecon :)
21:11.40*** join/#asterisk hans (n=fugalh@dhcp21.cs.nmsu.edu)
21:12.05dlynes_Dr-Linux: Have you read that link yet?
21:12.09dlynes_Dr-Linux: read it!
21:12.23*** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net)
21:12.31*** part/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it)
21:13.02Dr-Linuxdlynes_: wait my friend
21:13.05Dr-Linuxreading
21:13.25dlynes_yeah...you got another answer in asterisk-dev, too
21:13.56*** join/#asterisk viperdude (n=viperdud@84-45-168-60.no-dns-yet.enta.net)
21:14.12*** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-147.claranet.co.uk)
21:15.19znoGdlynes_: all PAP2-NA
21:15.25*** join/#asterisk MacDome (n=eseidel@A17-255-100-181.apple.com)
21:16.36hanstrying to get realtime to look up voicemail users in the db (still storing vm on the filesystem). it's making the odbc connection but never querying it
21:16.47hansI followed the instructions on the wiki very carefully, but nada
21:17.26dlynes_znoG: damnit
21:17.50dlynes_znoG: Do all your pap2-na's have a nat between them and asterisk?
21:21.42*** join/#asterisk LBJ- (n=no@0x5551a675.adsl.cybercity.dk)
21:22.15Dr-Linuxdlynes_: heh
21:23.56*** part/#asterisk fjean (n=fjean@201.29.130.118)
21:28.55*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
21:30.55*** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)
21:32.17Dr-Linuxwhat's the best way to apply a patch into the asterisk?
21:32.47Damincd /usr/src/asterisk ; patch < patchfile.txt
21:34.03*** join/#asterisk Marquis42 (i=HydraIRC@c-24-56-217-74.chrlmi.cablespeed.com)
21:34.27Dr-LinuxDamin: i my case it will be " cd /usr/src/asterisk ; app_dicrectory.c < patchfile.txt ?
21:34.39DaminNo.
21:34.40Damincd /usr/src/asterisk ; patch < patchfile.txt
21:35.49DaminDr-Linux: And with that, I must leave. If you can't figure it out, try "man patch"
21:35.50*** part/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-147.claranet.co.uk)
21:35.50*** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it)
21:35.50Dr-LinuxDamin: okey thanks, so after doing that, do i need to recompile?
21:35.58ravenpiDr-Linux: Yes.
21:36.03russellbDr-Linux: there is a patch howto on http://www.asterisk.org/developers/
21:36.22Qwell[]howtos are for people who can't RTFM :P
21:36.23Dr-Linuxrussellb: that's would be helpful thanks
21:36.27ravenpiDr-Linux: And re-install.  Note that you probably do NOT want to do a "make samples", 'cause your config files will get overwritten.
21:36.48russellbQwell[]: it's only a few commands :-p ... svn diff and patch
21:36.50russellbok, 2 commands
21:37.07Qwell[]"make oops-I-screwed-up-and-overwrote-my-configs"
21:37.12Dr-Linuxravenpi: yeah that i know, but what about modules? they will not bother?
21:37.15russellbmake uninstall-all
21:37.25LBJ-Anyone know if there is a problem with callstatus returning busy when the called phone i disconnected? It should return unavail. asterisk 1.2.5
21:37.33Qwell[]well, I was thinking a little less malicious :)
21:37.37ravenpiDr-Linux: They'll install fine with "make install".
21:37.42Qwell[]like...replacing the samples with the backups
21:38.03Dr-Linuxi have backup
21:38.21Dr-Linuxheh, i have very good expereince with such thing
21:38.35*** join/#asterisk kaz0358 (n=kaz@kazg5.telecom.ksu.edu)
21:38.39ravenpiSpeaking of overwriting files, what's The Right version control system for stuff like /etc files?  CVS?  Subversion?
21:38.56Dr-Linuxyou know, once mistakenly i did rm -rf /etc/asterisk
21:39.05Dr-Linuxi had no back for all
21:39.10Dr-Linuxit was peak time
21:39.19Dr-Linuxbut i checked, everything was working fine :P
21:39.24CunningPikeravenpi: Whatever works for you - I'm using Visual Sourcesafe
21:39.26Marquis42I personally use a central Subversion repository for all of my configs, both /etc/asterisk and others.
21:39.35Dr-Linuxso i hold my head and was thinking what to do now :P
21:39.38Marquis42VSS!?!  Ewww.... :P
21:39.46CunningPikeravenpi: But I'm planning to switch to Subversion
21:40.07CunningPikeMarquis42: Actually, VSS is not that bad - I like it
21:40.14ravenpiCunningPike: <grin>  I guess so, based on Marquis' reaction.  Never heard of VSS, m'self.
21:40.37*** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)
21:40.46CunningPikeravenpi: It works for me :)
21:41.11ravenpiDr-Linux: Yeah, deleting Asterisk's /etc hierarchy is a Bad Thing, but, so long as you don't terminate the program, or do a reload, it'll keep chugging.  Just hope you can get your restores back quickly.
21:41.25Marquis42I'm just razzing you.  It's not entirely bad, though there are certainly issues that exist.  My biggest one is any attempt at remote repository access over a WAN that isn't capacious
21:41.44CunningPikeMarquis42: True enough
21:42.28*** join/#asterisk hads (n=hads@203.109.245.87)
21:42.53BadPacketdoes this syntax work for anyone:  Dial(SIP/user:pass@${EXTEN}@gateway.itsp.com) ??
21:42.54CunningPikeAnd the other issue, and the reason I'm considering a move to SVN is that it deploys DOS formatted files, regardless of the format of the original. This has no effect for .conf files, but buggers up init scripts and the liek
21:43.26ravenpiCunningPike: Oh, wow.  Yeah, that would suck.
21:43.26Marquis42CunningPike:  I didn't know that, I've never used VSS outside a Windows environment.  Ouch!
21:43.37CunningPikeYes - it's real dumb
21:43.43Marquis42I agree.
21:43.44[TK]D-FenderBadPacket : that be better done as : Dial(SIP/user:pass@gateway.itsp.com/${EXTEN})
21:43.56ravenpiCunningPoke: You'd think, I dunno... that it'd give a binary-complete restore of the file.  That's just silly.
21:44.02BadPacket[TK]D-Fender: thanks
21:44.05CunningPikeIt only matters when transferring files between servers - it doesn't mess up the original
21:44.24CunningPikeravenpi: You'd think. MS doesn't
21:44.31BadPacket[TK]D-Fender: ...the example sip.conf in the /configs directory is wrong then...
21:44.43[TK]D-FenderBadPacket : Although typically bad idea.  Better to make a peer entry in sip.conf so you don't have to filter out password in pastebin'ing your diaplan for help, or repeating that long string in many places.
21:44.53ravenpiCunningPike: Oooooohhh.  Didn't realize it was an MS product.  *nods*  All clear, now.
21:45.01[TK]D-FenderBadPacket : Never ask me about "whats wrong", I might never stop....
21:45.21BadPacket[TK]D-Fender: heh
21:45.30CunningPikeravenpi: Not all MS products are bad
21:46.11BadPacket[TK]D-Fender: hmm... still not working
21:46.19[TK]D-FenderCunningPike : yeah MS had a money market simulator game back around 1980 in BASIC that I loved!
21:46.24*** join/#asterisk eric_hill (i=EricHill@204.94.175.11)
21:46.24*** part/#asterisk gr0mit_home (n=wendolen@extrt.txrx.org.uk)
21:46.35CunningPike[TK]D-Fender: Heh heh
21:46.50*** join/#asterisk jsaunders (n=root@216.86.121.58)
21:46.51Hmmhesaysyep i was -2 in 1980
21:46.55tainted-QBASIC?
21:47.00[TK]D-FenderActually I think it was stock market, not specifically curreny...
21:47.09jsaundersI still have an ol RPG I wrote in QBasic years ago.  Heh.  It rules.
21:47.12[TK]D-Fendertainted- : No, GW at best...
21:47.15tainted-nibbles was great
21:47.19jsaundersNibbles was fun.
21:47.19tainted-jsaunders send send!
21:47.21[TK]D-FenderQBASIC is for NEWBS!
21:47.23tainted-haha
21:47.23nahireangorilla owned nibbles
21:47.28jsaundersgorilla was kew too
21:47.36CunningPikeOMG - what have I done.........
21:47.37tainted-was that the velocity, trajectory game
21:47.49nahireanya.. two apes chucking fruit at one another
21:48.03jsaunderstainted:  Will do...  check www.northworld.ca for "what it became"  :)  Ported it to Pascal, but send me an email @ jsaunders@devarus.ca and I'll send you QBasic code.
21:48.26CunningPikejsaunders: Another Canuck?
21:48.32jsaundersyessir, vancouver bc
21:48.36*** join/#asterisk squinky86 (n=squinky8@gentoo/developer/squinky86)
21:48.39CunningPikeMe too :D
21:48.45[TK]D-FenderI remember working on a real-time 3-D (calculated, not drawn) version of Trade Wars 2002 I was working on in BASIC back in the day....
21:48.55*** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net)
21:48.57jsaundersWell, technically I'm in Chilliwack although I'm at our company office in Langley atm.
21:48.57tainted-jsaunders that's awesome!
21:49.01CunningPikeMaybe we should set up #asterisk-canada
21:49.05jsaundersHeheh
21:49.12jsaundersPerhaps ya'll could help me w/ a problem.
21:49.17CunningPikeLeave this channel for those who speak only Americanish ;)
21:49.19jsaunders[ext-did] has following line....
21:49.34Zodiacalanyone know of a good sound file to play that represents break times? e.g. over a loud speaker.
21:49.42jsaundersexten => s,1,Set(FROM-DID=s)
21:49.52*** join/#asterisk techman97_andy (n=me@70-98-31-249.dsl1.rsm.mn.frontiernet.net)
21:49.59jsaundersWhich is an amp thing yes... but it's still valid * so I figure I can ask in here.
21:49.59[TK]D-FenderZodiacal : Watch a hockey game... plenty of inspiration there....
21:50.23jsaundersI'm assuming FROM-DID=s should replace s w/ DID dialed.
21:50.26[TK]D-Fenderjsaunders : So far yet... and your point is...?? :)
21:50.37[TK]D-Fender(keep going so we have something to roast you for)
21:50.46jsaundersHeheh,  k.   :)
21:51.03jsaundersBut when it gets to inbound routing...  it's looking for extension 's', not populating w/ DID dialed.
21:51.06jsaundersugh
21:51.24[TK]D-Fenderjsaunders : What is the incoming interface?
21:51.29jsaunderszap or sip
21:51.44[TK]D-Fenderjsaunders : And DID's don't LAND on s.  they hit an EXTEN in the target context.
21:52.02*** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net)
21:52.13jsaundersWell, I can get it to catch S if I put a "catch all" route in there.
21:52.31jsaundersUgh, I'm confused.  Heh.  Excuse my newbness.
21:52.34CunningPikejsaunders: Wouldn't it be exten => s,1,Set(FROM-DID=${s}), or am I am idiot - warning, both may be true
21:52.41jsaundersThat's what I was thinking....
21:52.48[TK]D-Fenderjsaunders : "s" is when the target is UNKNOWN, which is exactly the opposite of that a DID is.
21:52.50Dr-Linuxi just wanna make sure will this patch work on my 1.2.0 version? http://www.delink.net/software.php
21:53.02[TK]D-Fenderjsaunders : as a Catch-all it'd be more like _X.
21:53.12jsaundersAh, very useful info.
21:53.16*** part/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu)
21:53.17jsaundersI wonder why ZAP doesn't know number dialed.
21:53.19jsaundersweird
21:53.30Dr-Linuxthe Directory() patch.
21:53.46[TK]D-Fenderjsaunders : What kind of Zap?
21:53.50jsaunderstdm2432e
21:53.57CunningPikejsaunders: What I do with DIDs is this: exten => _604990XXXX,1,Set(DID-FROM=${EXTEN:-4})
21:54.08[TK]D-Fenderjsaunders : Analog doesn't support the concept of DID/
21:54.13[TK]D-Fenderjsaunders : Thats why
21:54.23[TK]D-Fenderjsaunders : Analog lines "just ring".
21:54.41jsaundershmm
21:54.55[TK]D-Fenderjsaunders : only digital interfaces support DID signalling typically.
21:54.56CunningPikejsaunders: What type of service do you have?
21:55.10[TK]D-FenderCunningPike : Looking like POTS to me....
21:55.22[TK]D-FenderCunningPike : do look at the card he referenced
21:55.39jsaundersI'm trying to do following...  have fax come in on zap chan 1 and get routed to zap chan 13.
21:55.45jsaundersUsing freepbx / amp.
21:56.01CunningPikeAh, but there are flavors of POTS, no? More than one phone line a single POTS line can have
21:56.21[TK]D-Fenderjsaunders : Can't help you with the AMP part of that, but * can do taht easily enough.
21:56.29jsaundersIndeed.  :)  Hmm.
21:56.40LBJ-Hi can someone help me with a vm problem ?
21:56.56[TK]D-FenderCunningPike : If you're referring to distictive ring, perhaps, but thats only questionably reliable.
21:57.12justinupots is pots
21:57.13Dr-Linuxdoes recompile asterisk effect of /var/lib/asterisk/agi-bin/ dir?
21:57.19brodiemmaybe you're thinking of centrex
21:57.19justinuif you had distinctive ring, it wouldn't be pots anymore, would it
21:57.36[TK]D-Fenderjustinu : Um, yeah, thats part of POTS in the boring analog world...
21:57.51jsaundersSure is a nice card though...  thanks Digium.
21:57.58[TK]D-Fenderjustinu : Simple cadence to the ring pulse.
21:58.18[TK]D-Fenderjsaunders : Not to fit it in my MiniITX EPIA case.... hmmm!
21:58.18justinubut it's not plain old, anymore
21:58.27jsaundersHeheh
21:58.32jsaundersYeah, she's a beast.
21:58.44*** join/#asterisk lzhang (n=rjrae@67.95.13.46)
21:58.55jsaundersAtleast it doesn't take up multiple pci slots like Sangoma.
21:59.22[TK]D-Fenderjsaunders : That IS a minus in certain cases.  Each have their benifits
21:59.26lzhangIf I make a call from the system to an external number, is it possible to pass that call to an extension locally? (through the asterisk manager)
21:59.41jsaundersAlthough I heard the sangoma architecture is supposed to be very good.
21:59.53[TK]D-Fenderlzhang : take a look at the "m" parameter of app_dial....
22:00.26[TK]D-Fenderjsaunders : it is.  MUCH kinder on IRQ calls, and no know incompatabilities with any standard hardware.
22:00.27kaz0358is anyone getting involved with ISN? anyone have thoughts on the idea? http://www.freenum.org/ descibes ISN
22:01.07[TK]D-Fenderjsaunders : And can support a serious EC DSP.
22:01.12websaeanyone done any T.38 faxing?
22:01.28jsaundersGlad to hear Fender...  I heard talk about the irq thing.  Dunno the full deal though.
22:01.34jsaundersSoo much to learn.
22:01.56jsaundersI kinda like the backplane idea...  kewl.
22:03.06*** join/#asterisk zippp (n=zip@63.99.9.4)
22:03.11CunningPikeLBJ-: What was your vm question?
22:04.02zipppcan anyone tell me how to set the caller id to xxx-xxx-xxxx if the current channel CID number isn't in that format?
22:04.15kaz0358there are already about 100 univerities and companies involved in ISN.. it seems like a better solution than having a centralized location like sipbroker.com provide all the 3-5 digit provider gateway numbers.. and there is a better seperation between the subscriber number and provider number
22:05.16[TK]D-Fenderjsaunders : I can explain better later as I own 2 of their card, and support several with the Analog flavour.
22:05.26[TK]D-FenderBut for now, I'm off to class, later all.
22:05.54CunningPikezippp: Set(CallerID())?
22:06.20*** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net)
22:06.26zipppI know that part, I need to logic to say if callerid != fmt(xxx-xxx-xxxx) then setcallerid(xxx-xxx-xxxx)
22:06.33[TK]D-Fender|AFKzippp : CallerID is plain numberic and doesn't include non-digits like "-".
22:06.35CunningPikeDr-Linux: What's the question
22:06.39jsaundersLater Fender.  :D
22:06.45[TK]D-Fender|AFKanyways, I'm off
22:07.01[TK]D-Fender|AFKCunningPike : CALLERID() function is case sensitive and all-caps BTW
22:07.21CunningPike[TK]D-Fender|AFK: Thanks :)
22:07.21Dr-LinuxCunningPike: 2 questions
22:07.27Dr-Linuxi just wanna make sure will this patch work on my 1.2.0 version? http://www.delink.net/software.php
22:07.38*** join/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.hssx.sasknet.sk.ca)
22:07.41Dr-Linuxdoes recompile asterisk effect of /var/lib/asterisk/agi-bin/ dir?
22:08.09Tall-guyOk, before I rant...anyone using Eyebeam 1.5 with asterisk? (just released last week)
22:08.14CunningPike"If you need to add it to a production version of Asterisk, here is a patch to Asterisk 1.2.6" - I would take that to mean that you need 1.2.6 for it
22:09.07Dr-LinuxCunningPike: but i have asterisk 1.2.0
22:09.16Dr-Linuxbut i need that feature
22:09.21CunningPikeDr-Linux: Then you'll likely need to upgrade to 1.2.6
22:09.28CunningPikeYou probably should anyway
22:09.53CunningPikeNeat little patch, btw
22:10.06Dr-Linuxyeah :S
22:12.27lzhangAnybody know how to use the manager command Redirect? documentation is a little sparse...
22:12.38Tall-guyeyebeam...anyone?....Beuhler??? Beuhler?? :)
22:12.58lzhangI'm trying to pass a call that the system made to an external number to a sip phone locally
22:13.07*** join/#asterisk SuperLag (n=aaron@gentoo/developer/SuperLag)
22:13.07Dr-LinuxTall-guy: i use eyeBeam but don't know what version
22:13.10*** join/#asterisk kainam (n=Jake@202.137.160.110)
22:13.24Tall-guydr-linux: problay 1.1 if you don't know  (cause they just released a new version that is all f*KED up in my opinion)
22:13.39*** join/#asterisk tdmoc (n=jghj@VA1-1B-u-0044.mc.onolab.com)
22:14.02tdmochola
22:14.58*** part/#asterisk ph|ber (i=phiber@slackwaresupport.com)
22:14.59Dr-LinuxTall-guy: lolz i have eyeBeam for unlimited users, but still i don't use it
22:15.05Dr-Linuxi love SJphone
22:15.18*** part/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it)
22:17.28*** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com)
22:18.18brif8anyone using astertext?  I've compiled and now trying to restart * is failes  "app_securax_serverload.so: undefined symbol: ast_load
22:18.18brif8"
22:18.22brif8any ideas ?
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22:19.40*** part/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.hssx.sasknet.sk.ca)
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22:26.53jsaundersI'm diggin' idefisk these days.  Works decent.  No probs *that I know of*.
22:27.38jsaundersSJPhone is a sip classic...  but I'm enjoying ease of cutting thru nat w/ IAX.
22:29.16syleiax hmmm
22:29.28sylethere a iax proxy you guys use?
22:29.42jsaunderssure, it's called asterisk.  Heh.
22:29.48justinuiax proxy? uh
22:30.00sylesomething like ser for iax maybe
22:30.10jsaundersFor?
22:30.21sylevoip provider
22:31.18justinuit would be asterisk, like jsaunders said
22:31.28jsaundersasterisk & yate are only switches I know of that deal w/ IAX.  Yate apparently is not doing a very good job of it, although that's just hearsay.
22:31.37jsaundersI'm sure there are others but I am not aware of 'em.
22:31.59*** join/#asterisk r0d3nt|m (i=r0d3nt@tinfoilhat.net)
22:32.09sylei guess the biggest question is does the iax protocal support redirects
22:32.37jsaundersIt facilities something similar, yes.  google iax.
22:32.40[hC]it supports native transfer, which eliminates middle-man hops that are unnecessary if thats what you want.
22:32.58*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
22:32.59sylebut if that happens do you loose accounting info
22:34.54MikeJ[Laptop]syle, iax is media and signaling all in one
22:35.05MikeJ[Laptop]so if you do a native transfer your out of the loop
22:35.33MikeJ[Laptop]to do what I think you want is tricky, I think file is playing with ideas on how to do it
22:35.44jsaundersGo file.
22:35.46sylei have been wondering say with iax native transfer and sip's canreinvites if you loose cdr for not handling SDP stream
22:35.59jsaundersrtp
22:37.02*** join/#asterisk docelm0 (n=docelmo@55-65.126-70.tampabay.res.rr.com)
22:37.07jsaundersTo stay in signalling path when forwarding rtp you need a proper b2bua...  only one I know of that handles *questionably* properly is Yate.
22:37.47sylei see what mikej is saying, if your signalling is lost then your foobarred then
22:37.52sylewhat about sip
22:38.40jsaundersLast comment was regarding sip.
22:38.50jsaundersiax does not use rtp
22:38.53*** join/#asterisk r0d3nt|m (i=r0d3nt@tinfoilhat.net)
22:39.27syleis it possible for a provider to avoid being the man in the middle but still bill his customers for a call
22:39.40jsaundersw/ sip & h323, yes.
22:39.43dlynes_~seen coppice
22:39.45jbotcoppice <n=chatzill@153.192.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 5h 27m 6s ago, saying: 'I have two in this house :-)'.
22:40.09dlynes_damnit...keep missing him :)
22:40.34sylewhy h323
22:41.08jsaundersWe have a commercial yate switch that handles just signalling for tens of thousands of calls daily....  forwards audio between endpoints w/out first having to start rtp and the "reinvite" like asterisk does.
22:41.26jsaundersthe = then
22:41.28*** join/#asterisk danilom (n=danilom@201.209.43.241)
22:41.50*** join/#asterisk MstlyHrmls (n=mh@melbourne.mostly-harmless.ca)
22:41.55*** join/#asterisk ptiggerdine (n=ptiggerd@router-bne.action-international.com)
22:42.05syledo you have to pull billing info off the yate switch then?
22:42.17justinujsaunders: for a n00b, you sure seem to understand this stuff
22:42.23jsaundersCan be annoying when trying to make sure endpoints have firewalls configured properly.  Ugh.
22:42.34jsaundersjustinu  :D
22:42.38jsaundersI try.
22:42.44jsaundersI'm weak at * scripting though.  :(
22:43.32sylei'm trying to think of a way for asterisk to handle signalling to bill for call but toss the RTP stream
22:43.35jsaunderssyle:  Yate provides all call related info and passes via sql to database.
22:43.47jsaundersAsterisk will always start audio (rtp) stream.
22:43.57jsaunderscanrinvite will simple move audio after 8 or so seconds.
22:44.01jsaunderssimple = simply
22:44.08dlynes_jsaunders: freeswitch does iax as well, but it's far from being a finished product
22:44.13jsaundersfreeswitch pwns
22:44.18jsaundersI can't wait to hop onboard.
22:44.40jsaundersanthm & MikeJ and the crew are doing a great job from what I can see.
22:44.43jsaundersLove the community.
22:44.49dlynes_Yeah, they're doing a great job
22:44.56dlynes_I'm just saying it's far from being finished :)
22:44.57justinu8 seconds?
22:45.05justinumy asterisk systems reinvite the call as soon as it answers
22:45.22jsaundersCannot verify that justinu...  I read somewhere it was something like 6-8 seconds.
22:45.35jsaundersAnd testing seemed to confirm.
22:45.40mog_workheh i got  you beat justinu mine does is it before the cal is answered !!!!! ^_^
22:45.47jsaundersHeheh
22:45.50justinuwell, i'm running early media
22:45.54justinufor call progress tones, etc.
22:45.58justinuso that isn't possible.
22:46.05mog_workmmmhmm
22:46.18mog_workman telephony would be so much simpler
22:46.24jsaundersword mog
22:46.25mog_workif everyone just charged when you went off hook
22:46.25sylei guess i;ll test later this week if i can;t get a direct answer with tcpdump lol, thx anyways
22:46.38mog_workearlymedia is just retarded in my opinion
22:47.04justinulol, that statement is kinda retarded in my opinion
22:47.13jsaundersheheh
22:47.18mog_worksure
22:47.22justinusome of us need to generate call progress tones
22:47.27justinuor messages
22:47.34justinuand we don't want people to pay for that call
22:47.35mog_workjust think how much simpler all the protocols would be if when you dialed it was considered answered
22:47.41mog_workand you just got charged for busy tone
22:47.44mog_workor whatever
22:47.48mog_workit would all even out in the end
22:47.50justinusay what?
22:48.09mog_workwhen you called to hear that busy tone it costs the provider to connect
22:48.12*** join/#asterisk kainam (n=Jake@202.137.160.110)
22:48.14mog_workand everything
22:48.19mog_worki mean the cost is still their
22:48.22mog_workerr there
22:48.31syleyeah and think how kewl it woudl be if everytime you had sex you never orgasmed!
22:48.38justinulol
22:48.40justinuyou guys are whack
22:48.54jsaundersahhh....  syle...  lets not get carried away here.
22:49.09dlynes_early media has its uses
22:49.14jsaundersHeh
22:49.18dlynes_such as announcing the caller
22:49.25jsaundersIsn't that the whole point of sex?   Heh
22:49.32dlynes_then the other end can refuse the call if it's that bastard collection agency
22:49.38justinui use it to announce number changes, etc.
22:49.40justinutypical telco crap
22:49.44*** part/#asterisk websae (n=websae@CPE-24-167-206-22.wi.res.rr.com)
22:50.19mog_worksyle what the hell does that have to do with anything?
22:50.41mog_workbut why shouldnt person x pay for that
22:50.47mog_workyou still used telco services etc
22:50.55justinubecause they didn't actually talk to the person they called
22:51.09dlynes_people don't want to have to pay for dialing the wrong number
22:51.09mog_workbut you were connected
22:51.15mog_workyou arent paying for the conversation
22:51.18mog_workbut the connection]
22:51.19justinuconnected to a media server
22:51.25justinunot the destination
22:51.36dlynes_even when they dial the wrong number on a typical telco, and they realize that, a lot of people will call up the telco to refute the charges
22:51.44mog_workyeah i know
22:51.47justinuback in the way old days, all calls were person-to-person, i think
22:51.55mog_workas you have used their services
22:52.05mog_workit shouldnt matter that you didnt get result x
22:52.11jsaundersthru the ol switchboard operators?  Heh
22:52.11mog_workservices were still rendered
22:52.12dlynes_so work that into your cost of doing business
22:52.18dlynes_deal with it
22:52.21mog_workwhy not just have it simple
22:52.24mog_workand average it out
22:52.25dlynes_people are people...you can't change the way they think
22:52.31dlynes_they don't think like computer programs
22:52.32mog_workand just charge people for using services?
22:52.36mog_workit makes it so much simpler
22:52.40jsaundersI like mog's thinkin' outside the box though...  can't knock him for that.
22:52.55justinui dunno, i'm too programmed
22:53.01justinui'm always concerned about answer sup
22:53.09dlynes_jsaunders: sounds more like a lazy programmer that doesn't want to have to do the extra programming to allow for that :)
22:53.09justinutoo many years in telco, i guess
22:53.16mog_worksee in my system you wouldnt have to deal with it
22:53.19mog_worklol
22:53.22CunningPikeHow about making it even easier - flat rate, like internet ;)
22:53.40mog_workas i make maybe 3 dollars worth of calls a month
22:53.47jsaundersA lazy programmer @ Digium?  Heh
22:53.54DoktorGregthe best model i have found for the small to no cap startup is a ma-pa enterprise
22:54.02justinujsaunders: lol
22:54.09jsaundersI think they're doin' a good job over there.  But, that's just my opinion.
22:54.24dlynes_yeah...russellb rocks
22:54.26jsaundersThat wasn't a burn mog.  Heh.  Was actually a question.
22:54.41mog_workahh okies, feelings less hurt
22:54.51dlynes_He's a full time student, and still finds time to manage trunk
22:54.57jsaundersHow could I burn ya...  you helped me w/ my tdm2400.
22:55.07jsaundersYou pwn in my books mang.
22:55.07dlynes_plus help everyone out on #asterisk/#asterisk-dev
22:55.16mog_workreally /me doesnt remember i help lot of people out
22:55.19mog_worksorries
22:55.36jsaundersYep, just under 2 weeks ago.  I've chatted ya up a few times...  no worries, I'm sure your busy.
22:56.23mog_workoh i know i have talked with you few times
22:56.31mog_workbut dont always remember what i fix
22:56.37jsaundersheheh, I hear ya.
22:56.43jsaundersI can't remember what I fixed 5 minutes ago.
22:56.58dlynes_btw
22:57.00jsaundersJames, fix this....  James, fix that.... ugh.
22:57.22dlynes_Anyone know why Caller ID wouldn't work on either x100p.com cards, or a genuine Digium X100P card?
22:57.33dlynes_It works on one out of every ten calls or so
22:57.40dlynes_but that's just a statistic, not a hard and fast rule
22:58.03mog_workwhere you located dlynes
22:58.06dlynes_The rest of the time, the caller id shows 'asterisk'
22:58.06jsaundersa guesstimate?
22:58.08jsaunders:D
22:58.13dlynes_mog_work: Canada (Vancouver)
22:58.16CunningPikedlynes: I'm sure you're doing the obvious Wait(2)
22:58.34dlynes_probably
22:58.37mog_workshould be fine
22:58.49mog_workyou have the caller id configged correctly in zapata.conf?
22:59.06CunningPikedlynes: We have very inconsistent incoming CID, too - most of our calls have number but no name
22:59.28justinubitch at your telco about that
22:59.30dlynes_Yeah, I've done Wait(1), Wait(2), Wait(3)
22:59.35dlynes_it doesn't seem to make a difference
22:59.36*** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)
22:59.51dlynes_It works fine with a Nortel CICS keysystem, but not with asterisk
22:59.52CunningPikeTelus and Allstream just mumble about CO and network transfer hand-offs
23:00.16justinudlynes: rxgain too high?
23:00.23CunningPikeAnd yet, every call I get at home has something in the name
23:00.37dlynes_justinu: rxgain/txgain are both set at 1.5
23:00.50justinuis that tuned, or arbitrary?
23:00.52jsaundersYou should be able to get a doctorate in telco...  people who hax0r this all day deserve a flashy title.
23:00.55*** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)
23:01.01dlynes_justinu: does rxgain/txgain affect callerid?  I thought it only affected dtmf and faxing?
23:01.10justinucallerid is delivered via FSK
23:01.15justinumodem tones
23:01.17dlynes_You've lost me
23:01.19dlynes_ah
23:01.23justinuso if gain is too high, it'll be distorted
23:01.29justinuand the decoder says "wtf is this?"
23:01.32dlynes_Yeah, it was just arbitrary
23:01.52dlynes_So the only way you can check it is by playing with it continuously until the error rate settles down?
23:02.02justinuok, so start going down in increments of 4 until you can't hear people well enough
23:02.03*** join/#asterisk Eggplant (i=No@dsl-332.cascadeaccess.com)
23:02.14justinumake it as low as you can without being too soft
23:02.22dlynes_increments of 4?  I thought the range was only -7 to +7?
23:02.32dlynes_or you mean increments of .4?
23:02.57justinuno, it's +/- 100
23:03.00dlynes_oh
23:03.01dlynes_ok
23:03.06justinuand those increments aren't dB
23:03.24dlynes_what are they?  just some arbitrary logarithmic values?
23:03.27justinuthey're something like "100%" of what the card is capable of attenuating/boosting
23:03.40justinui can't find any better spec than that
23:03.44dlynes_ah
23:07.09*** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net)
23:09.27dlynes_Yeah, that definitely seems to be affecting it
23:09.30dlynes_Thanks, justinu
23:09.35justinunp
23:09.41*** join/#asterisk Abydos313 (i=abydos31@adsl-71-129-52-80.dsl.irvnca.pacbell.net)
23:10.04jsaundersquit
23:10.07jsaundersoops  :)
23:10.36dlynes_Hrm
23:10.44dlynes_But now it doesn't seem to be recognizing hangups
23:10.52dlynes_When it kicks into Background()
23:11.12dlynes_Or maybe it is...just extremely delayed
23:11.58justinuusing kewlstart?
23:12.02dlynes_correct
23:12.11dlynes_fxsks
23:12.17justinucan't imagine how gains would affect that
23:12.19dlynes_both zaptel.conf and zapata.conf
23:12.30dlynes_Maybe it's just the damned cellphone then
23:12.40justinusince kewlstart is looking for a loop current drop as a disconnect supervisory
23:12.57dlynes_Nortel CICS also has problems on line 1 when you call into it with a cell phone
23:13.36Dr-Linuxwhen my rxgain is set to 2.0 , caller hear very low, when rxgain is 3.0 then caller gets echo
23:13.44justinuouch
23:13.45Dr-Linuxnot sure where to go :S
23:13.47justinu2.5?
23:15.21dlynes_It only happens on line 1, and it only happens when you hang up; the Nortel cics often doesn't recognize the hangup
23:15.25*** part/#asterisk BadPacket (n=root@unaffiliated/badpacket)
23:15.34dlynes_So, I'm wondering if maybe asterisk has the same problem
23:15.58justinuand it's only when you call with a cell phone?
23:16.04dlynes_I've noticed Nortel CICS and Asterisk also both suffer from the same problem if the caller is a voip caller
23:16.15dlynes_It doesn't happen every time though...just most times
23:16.36dlynes_justinu: cell phone or voip phone
23:16.55dlynes_justinu: in my case, it would be ACT/Azatel/GVC SIP phones
23:17.08justinuthat phone line originates on what piece of equipment?
23:17.24dlynes_justinu: it doesn't seem to happen for asterisk originated calls
23:17.59justinui'd have to know more topology to understand it
23:18.08dlynes_justinu: The cell phone problem always happens on Fido (Microcell Solutions)
23:18.16fileRogers.
23:18.18dlynes_justinu: the voip phone problem doesn't happen every time though
23:18.33dlynes_file: Don't know about Rogers, but fido had that issue long before Rogers bought them
23:18.51justinuso you have a phone line coming from an ILEC?
23:18.52dlynes_it could affect other cell phones too, for all i know
23:18.56dlynes_but most people i know are using fido
23:19.10dlynes_justinu: yes...Telus
23:19.27dlynes_justinu: the ones it seems to work on for cell phone hangups are allstream lines
23:19.37dlynes_justinu: the ones it doesn't are telus lines
23:19.38justinuand when you call that PSTN number from certain types of phones, asterisk doesn't detect the hangup?
23:19.59dlynes_justinu: correct, but the problem manifests itself on nortel cics keysystems too...not just asterisk
23:20.00justinuor call it from certain networks?
23:20.24justinuthat's bizzare
23:20.24dlynes_justinu: i don't know if it's all networks or not...the only networks i can test it from are fido
23:20.42dlynes_but the problem only seems to exist on telus lines
23:20.42justinudoes an IVR answer the line?
23:20.50dlynes_For the asterisk box, yes
23:20.53dlynes_for the nortel box, no
23:20.56justinuk
23:21.24justinui dunno what fido or telus are... canadian ilecs?
23:21.38dlynes_doesn't seem to matter whether i'm using an x100p card, or a sipura 3000, either
23:21.48dlynes_fido is a clec, telus is an ilec, allstream is a clec
23:22.01*** join/#asterisk bzbw (i=bwz@ip67-153-142-109.z142-153-67.customer.algx.net)
23:22.05dlynes_all canadian, yes
23:22.19dlynes_Fido is also known as Rogers
23:22.24dlynes_Telus is part of Stentor
23:22.30dlynes_Allstream is owned by AT&T
23:23.04justinuit almost sounds like your ilec doesn't have disco supervision with some of the networks, but that seems really hard to believe
23:23.55dlynes_stranger things have happened
23:24.26dlynes_one night I couldn't place any calls to certain co's on the fido network because fido's peering with group telecom was down
23:24.38dlynes_that was totally fubar
23:24.44justinuwow
23:25.09dlynes_604628xxxx was working, but not 604484xxxx
23:25.14dlynes_it was really strange
23:25.25dlynes_I'm a customer of both companies, so I was able to bitch at both of them :)
23:25.35*** join/#asterisk phonic (i=phonic@antisocial.nu)
23:26.25justinuso bitch at them about their shitty disco sup
23:26.58*** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net)
23:27.54phonichow do i disable or enable a port in my tele-pci card?
23:28.01dlynes_yeah..never had a problem with group telecom in taht respect...their equipment works great
23:28.17dlynes_but, yeah...i'll talk to telus about it i guess then
23:28.41phonici will make sure asterisk is using port 1
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23:31.47sevardOH SHIT OH SHIT ON SHIT OH SHOT
23:31.50sevardOH FIUCK
23:32.23sevardi just upgraded asterisk and I didn't make make samples but all of my configs are gone
23:32.25sevardmy configs were massive
23:32.43sevardwhy would it do that
23:32.44sevardwhy
23:32.45sevardwhy
23:32.59dlynes_why would sevard not make backups?
23:33.10sevardI thought I did
23:33.12sevardoi thought i did
23:33.15sevardi thought i did
23:33.35dlynes_and why would you blindly type 'make samples'?
23:33.44dlynes_without knowing what it would do to a live install?
23:33.45CunningPikesevard: For next time, 'OVERWRITE=no' in your Makefile
23:33.52sevardi NOT type make samples
23:33.54dlynes_oh
23:34.02dlynes_you installed from a binary package?
23:34.04sevardyup
23:34.12dlynes_damn
23:34.13sevardFUCK
23:34.23dlynes_for slackware?
23:34.28sevardyes
23:34.29dlynes_was it your own binary package?
23:34.31sevardyes
23:34.44sevardi removepkg the old one and installpkg the new one
23:35.01sevardi thought i made a fucking back up :(
23:35.01dlynes_sevard: a hint for next time
23:35.07dlynes_sevard: when you make the staging directory
23:35.38dlynes_sevard: rename all the /staging/etc/asterisk/*.* to *.*-sample
23:37.07dlynes_sevard: also, make a backup copy of your /var/spool/asterisk and /var/lib/asterisk/keys, /var/lib/asterisk/licenses/ directories
23:37.24sevardthere is no way i can recover this can i
23:37.26sevardi'm so fucked
23:37.29sevardi'm fucked more than ever
23:38.14dlynes_do you have any old backups?
23:38.52*** join/#asterisk Gamercjm (n=chris@pool-71-254-177-36.lsanca.fios.verizon.net)
23:38.52dlynes_guess not
23:38.53*** join/#asterisk elg (n=fugalh@falcon.fugal.net)
23:39.14justinulol
23:39.35dlynes_I'm sure he doesn't think it's so funny
23:39.37*** join/#asterisk ghost99 (n=neville@222-152-219-77.jetstream.xtra.co.nz)
23:40.04justinui'm allowed to laugh, having been in similar situations
23:40.07dlynes_CunningPike: besides...he did a removepkg
23:40.08VoicePulseDid you pastebin any of your configs?
23:40.19dlynes_CunningPike: OVERWRITE=no won't solve that problem
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23:40.40dlynes_CunningPike: because he had all the /etc/asterisk/*.* with the original filenames
23:40.47CunningPikedlynes_: I guess not - at the outset, I thought he was installing from source
23:41.01dlynes_CunningPike: He staged from source, and installed from binary package
23:41.04sevardwish reiserfs has an undelete
23:41.06dlynes_CunningPike: It's the same way I do it, too
23:41.23dlynes_CunningPike: But, I do a test on the package before I actually try deploying it any where
23:41.55CunningPikedlynes_: I install from source - on a test server first
23:42.16dlynes_CunningPike: exactly...staging is almost the same, but there's one more step
23:42.45dlynes_CunningPike: I just build everything into /usr/local/src/staging first
23:42.48sevardlooks like reiserfs 'undelete' only works when the directory you wish to undelete is a partition
23:43.04dlynes_CunningPike: then edit /usr/local/src/staging/etc/asterisk/asterisk.conf
23:43.17dlynes_CunningPike: then edit /usr/local/src/staging/usr/sbin/safe_asterisk
23:43.26dlynes_CunningPike: check all symbolic links to make sure they're relative
23:43.40dlynes_CunningPike: and then make a package
23:43.58dlynes_after i've renamed the files in the /etc/asterisk directory in the staging directory, that is
23:44.29dlynes_It's a bit more work than compiling from scratch
23:44.41dlynes_but i don't need to put a build load on the target machine then
23:45.06dlynes_and i can easily advance/revert version numbers
23:45.49CunningPikedlynes_: Makes sense - especially from a loading perspective as our production server gets busier
23:46.02dlynes_Yeah...it doesn't make any difference to me now
23:46.09dlynes_but when the server is at capacity
23:46.12dlynes_it makes a huge difference
23:46.17dlynes_I'd rather be prepared ahead of time
23:46.27CunningPikedlynes_: Good thinking
23:46.48*** join/#asterisk Weezey (n=ohno@206.210.109.235)
23:47.03dlynes_George went home already, Weezey.
23:47.12Weezeythat ass
23:47.48Weezeywhat's the cheapest way to get a zaptel timer in a box?
23:48.36dlynes_modprobe ztdummy
23:48.52*** join/#asterisk terrapen (n=cjs@166.70.183.109)
23:48.57dlynes_Assuming you're using a 2.6 kernel
23:49.02*** join/#asterisk xcoyote (n=farfan@201.135.194.118)
23:49.03terrapendoes anybody have b2llc's new ftp server login?
23:49.35xcoyotequestion: does exists api documentation for the asterisk source code ?
23:49.53dlynes_xcoyote: make docs
23:50.02terrapen(voipsupply)
23:51.15znoGdlynes_: nope, direct to Asterisk
23:51.22Weezeyterrapen: they changed it?
23:51.56dlynes_znoG: huh?  that response was so delayed, i don't even remember what i was talking to you about
23:52.08Weezeyterrapen: damn, they changed it.  call them.
23:52.18znoGdlynes_: you asked me if I had to NAT between the ATAs and Asterisk
23:52.33dlynes_ah
23:52.49dlynes_can't remember what the problem was, either :)
23:53.06dlynes_just that it was something to do with sip and pap2-na's
23:54.06dlynes_sounds like a lot of the digium servers are going down now :)
23:54.10terrapenweezey, i just tried
23:54.13terrapennobody answering
23:54.59terrapeni really need some 501 firmw4rez
23:55.18terrapenwhy the hell doesn't polycom just provide it?
23:55.26terrapenit's not like i didn't buy an assload of their phones
23:55.42dlynes_terrapen: they do, but to their channel partners, not end consumers
23:56.08dlynes_terrapen: if you buy that many phones, why not just become a channel partner?
23:56.23terrapenmaybe i should.
23:56.35terrapendo you have the latest IP501 fw/br?
23:56.47dlynes_nope...i'm not a channel partner though, either
23:57.00terrapencertainly someone here must have some
23:57.01dlynes_We've filled out all the necessary paperwork already though
23:57.05CunningPikeBootrom:  http://support.gafana.com/polycom/bootrom_313.zip
23:57.05CunningPikeSIP: http://support.gafana.com/polycom/SoundPoint_IP_SIP_1_6_5.zip
23:57.08terrapenhow many do you have to buy?
23:57.11dlynes_now we need to write all the online tests
23:57.20dlynes_not really a limit
23:57.24terrapenis that fairly recent, cunning?
23:57.28CunningPikeFrom a recent posting to asterisk-users
23:57.32terrapensweet
23:57.48CunningPike3.1.3 is the current bootrom - Sip 1.6.6 just came out
23:57.58CunningPikeBut 1.6.5 is current
23:58.15terrapeni have BootROM 2.6.2 and sip 1.6.3
23:58.29terrapenhrmm...shouldn't the IP601 bootrom work for the IP501, too?
23:59.17*** join/#asterisk ringhals (i=fwuser@firewall.drgutah.com)
23:59.26CunningPiketerrapen: Yes - we have it on both
23:59.33CunningPikeIt's the same
23:59.46terrapenok, maybe my configs are just wrong...for some reason this new 501 was saying invalid bootrom
23:59.53terrapenand the bootrom is there and works just fine on the 601
23:59.59CunningPikeWhat is your current bootrom

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