IRC log for #asterisk on 20070915

00:00.21dan__tI remember the #postfix days, I used to hang out there a lot.
00:02.52rob0But codefreeze ... you CANNOT pass up this ONCE-IN-A-LIFETIME opportunity to own a classic and very important landmark!
00:03.40rob0Oh I'm still in #postfix, answering questions for those who won't RTFM. :)
00:04.30*** join/#asterisk el_critter (n=chatzill@190.74.100.35)
00:04.56el_critterhi
00:05.52el_critterCan I use asterisk as a main-door intercomunicator for my building?
00:06.43hmmhesayswhy not
00:07.35tzafrir_homerob0, asterisk starts (forks into background) . It may hang a bit on modules loading
00:07.47Poehaliyou mean like a doorbell?
00:07.48tzafrir_homeHence: don't use safe_asterisk and all will be well...
00:08.15rob0And then chan_sip or chan_iax barfs and it dies?
00:08.16el_critterThe intercom model is one with a dialpad an a mic/speak, without a handset. You select your department and push "*" to dial. Thas must act a phone, I think, I'm right?
00:08.35rob0Actually I do have just "asterisk" in the startup.
00:08.43tzafrir_homeit shouldn't. Why should it just die?
00:09.03rob0I guess I can pull the Internet plug and test it.
00:09.12rob0but not right now ;)
00:10.23rob0I was thinking it might be cool to set up a special extension which I could use as a sort of a root prompt. Insecure as hell, but who's going to know to try it? :)
00:11.02rob0but, if the machine is online, I don't need it, and if * failed to start, I don't have it.
00:13.31*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)
00:13.47tzafrir_homerob0, I was about to try some real-world examples of people who broke into systems using such extensions, so
00:14.04tzafrir_homeI looked up in google "rooting a pbx"
00:14.07tzafrir_homefirst hit is:
00:14.26tzafrir_homeLeast cost rooting system - US Patent 5764741: http://www.patentstorm.us/patents/5764741-claims.html
00:14.46rob0lol
00:14.49tzafrir_homeWow: people have actually automated it. And made sure it is efficient. Scary!
00:18.46*** join/#asterisk zotz (n=zotz@24.244.163.157)
00:18.57*** join/#asterisk Barmal (n=info@c-24-30-126-164.hsd1.ca.comcast.net)
00:20.19Barmalwhen I receive a call and answer my server transmites invites and on 6th invite it gives up and hangs up the channel. It takes about 20 secs and audio works fine but call gets disconnected them. ANY IDEAS???
00:24.36Barmalanybody please any ideas? I just stuck without any idea where to go next....  sip debug at http://www.trixbox.org/forums/trixbox-forums/help/help-call-gets-disconnected-pleaseeee
00:27.25tzafrir_homeBarmal, please pastebin the output from sip debug
00:27.47tzafrir_home(which means: I don't have anything more intelligent to say)
00:29.16Barmalhere it goes http://pastebin.com/d52e6e7e2
00:29.23*** join/#asterisk bintut (n=bintut@cm47.gamma178.maxonline.com.sg)
00:30.20Barmalwhat do not make sense it works on the other server fine.
00:31.46Barmalwhy my end tryes to send invite to a server where the call came from?
00:33.48*** join/#asterisk coppice (n=chatzill@234.155.17.210.dyn.pacific.net.hk)
00:35.58Barmaland can it disable it somehow?
00:39.30*** join/#asterisk roberted (n=roberted@c-76-17-219-22.hsd1.mn.comcast.net)
00:51.50tzafrir_homeBarmal, I might be missing something, but I don't see any reply from the remote side there
00:52.27Barmalreply on invite?
00:56.31tzafrir_homeis it you connecting, or someone connecting to you?
00:57.36BarmalI am calling sip provider did and it calls my server and ivr answers
00:58.09*** join/#asterisk |dennis| (n=dennis@200.32.233.84)
00:58.27robertedhi, is aterisk really hard to interface with a t1 interface?
00:58.37tzafrir_homeno
00:58.47tzafrir_homeyou do need a T1 card, though
00:59.27robertedI have done it with pots lines, but I am really scared to jump the gun to a t1 because I have never dealt with the voice side
00:59.51robertedof a t1 that is
01:06.11JTit's easier, imho
01:06.22JTpots is a massive pain in the arse
01:06.26JTdigital is nice :)
01:06.47coppiceCome to the voice side, Luke
01:07.50roberted:)
01:08.22robertedIs there much echo with going the t1 route as opposed to a tdm card?
01:08.43JTnot if you get a card with hardware echo cancellation
01:08.52coppiceit elminates local echo, but you still get the remote one
01:09.12robertedwhich card do you recommend?
01:09.50*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
01:12.28robertedI have been eyeing one of these: http://www.8774e4voip.com/ProductDetails.asp?ProductCode=Digium+TE120P
01:17.06*** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net)
01:19.58*** join/#asterisk saftsack (n=saftsack@pD9E042FB.dip.t-dialin.net)
01:20.30BarmalTzafrir_home: look at this one it more readable: http://pastebin.com/d6d2346de
01:20.52saftsackare 128mb ram enough for installing asterisk without special arguments? no transcoding etc. is done. voicemail ... yes
01:21.18tzafrir_homeit will do
01:22.42Barmalmy question is what am I trying to retransit those 6 times before the call gets hanged up??????
01:22.58coppicesaftsack: people squeeze simple configs into little 16MB routers
01:23.03saftsacki plan a system with up to 10 user. should this be possible with 733hz (intel) and 128mb-ram?
01:23.09Barmaland can i turn off this transmission?
01:23.26saftsackcoppice: yes i know that. i have an openwrt system too but they are compiled without zaptel and with enable memory optimize
01:23.41tzafrir_homewhy do you think that the problem lies with the retransmission?
01:23.51coppiceand this is somehow not possible with 128MB?
01:25.00saftsackim asking if this is easily possible. my openwrt system breaks down at about 3 users. voicemail isnt possible. so the point which wasnt known by me is, if asterisks voicemailpart and its memory consumption with more than 5 users is hardly rising
01:26.25Barmaltzafrir_home, I am not sure but thats where I see it. every time after I make 6 retransmission this incomming call gets disconnected.
01:27.54Barmalwhere else should I look for a problem. If I register this trunk with my other server or with sip phone directly it works fine, so I know the problem is on my end....
01:30.03Barmalas I understand is like the other end establishes session and I have call connected and my end is thinking that no session is beeing established and it hang up the call.
01:33.13*** join/#asterisk ManxPower (n=manxpowe@38.sub-75-203-142.myvzw.com)
01:38.56Barmalhow is it asterisknow gui compared with freepbx does it have more features less or it's totally different product?
01:46.06*** join/#asterisk Kernel_Core (n=I@217.218.80.156)
01:46.58Kernel_Corehi all
01:52.11tzafrir_homefar less features. And this *is* a completely different pbx
01:53.14Barmalwhat you mean with different pbx? they both based on top of asterisk
01:55.17coppiceLondon is based on top of the old Roman Londinium, but Latin won't get you far far there :-)
01:57.01Kernel_CoreI am looking for a solution to limit all ZAP channels not to dial-out more than 6 hours/day ... ?
01:57.04Kernel_Coreis there  ? :)
01:57.28Barmalman that was too much to get it :)
01:57.36*** join/#asterisk sevard (n=sev@192.235.0.85)
02:05.37robertedanyone here used an unlocked dlink vta-vr with asterisk?
02:09.03*** join/#asterisk brunner (n=chris@75-143-105-41.dhcp.aubn.al.charter.com)
02:09.23brunnerCorydon!
02:09.41watchyhowdy
02:09.54brunnerhi
02:10.20watchyim heading out guys talk to you when i get home
02:10.35brunnerI'm trying to get in touch with Corydon76
02:16.52*** join/#asterisk Maxxed (i=foobar@65.59.245.122)
02:17.06Maxxedhey, can asterisk emulate modems?
02:17.14Maxxedlike, i can plug in a pri line card
02:17.29Maxxedand have ppl dial in to the pbx via the pri with 56k modems?
02:17.29[hC]you can use a digium or sangoma pri card yes.
02:17.41[hC]and do what?
02:17.52[hC]get internet access?
02:17.55[hC]you want a router dude, not a pbx.
02:18.00Maxxedhah
02:18.20Maxxedim looking at this little number http://www.patton.com/products/pe_products.asp?category=22&tab=ri&MiDAS_SessionID=a6ba98032daf4249801c9cf6639f93d6
02:18.26Maxxedmodem bank/server
02:18.40Maxxedplug in a t1 pri and have 23 modems for dial in/out
02:18.55[hC]sure, but that has nothing to do with asterisk man
02:19.05Maxxedok ok, i was just thinking of something weird
02:19.09Maxxedthx :D
02:19.24Maxxedwow, now that i think about it..
02:19.25[hC]np
02:19.28Maxxedwhat a dumb ass idea :p
02:22.46*** part/#asterisk brunner (n=chris@75-143-105-41.dhcp.aubn.al.charter.com)
02:25.21*** join/#asterisk Maxxed (i=foobar@65.59.245.122)
02:25.25Maxxedwait a tic
02:25.29MaxxedZapRAS
02:25.32Maxxedhttp://www.voip-info.org/wiki/view/Asterisk+cmd+ZapRAS
02:25.47J4k3if you don't need V.92
02:25.48Maxxeddialup internet access via Asterisk with a PRI
02:26.01Maxxednah, i dont know wtf v.92 really is anyway
02:26.01J4k3you can get a v.90 capable access server capable of handling two PRIs worth of traffic for under $150
02:26.09Maxxedno lie!?
02:26.17J4k3Lucent Portmaster 3
02:26.18J4k3ebay
02:26.21Maxxedthe ones iv seen are in the 2k range
02:26.29Maxxedoh, well im taking like still vendor supported
02:26.31Maxxednot end of life
02:26.36J4k3Ascend 6000-series (6096)
02:26.39J4k3why bother?
02:26.48Maxxedthe deal is, im supporting some old legacy dial out point of sale junk
02:26.55Maxxedand its a heaping mess right now
02:27.01J4k3yeah, which means you probably want hardware that you know all the bugs in already
02:27.12Maxxed30 plain ol analog lines, a dozen modems, a serial card in a linux box
02:27.13Maxxedbleh
02:27.25J4k3ack, analog is the poop
02:27.30Maxxedyeah
02:27.43J4k3the day I got two CT1s installed... was a very happy day
02:27.50J4k3fired up my Portmaster 3
02:27.54J4k3and my customers got a LOT happier
02:27.57Maxxednice
02:28.13ectospasmT1s are the shit
02:28.17Maxxedim looking at 3 devices
02:28.18Maxxedhttp://www.patton.com/products/pe_products.asp?category=22&tab=ri&MiDAS_SessionID=a6ba98032daf4249801c9cf6639f93d6
02:28.22Maxxedhttp://www.multitech.com/PRODUCTS/Families/MultiAccess/
02:28.32Maxxedand ciscos http://www.multitech.com/PRODUCTS/Families/MultiAccess/
02:28.42J4k3you realize what modern access gear costs new, right?
02:28.42Maxxedthese 3 all look to be able to do the crap i need
02:28.49Maxxedyeah, like 2k+
02:28.53Maxxedbase
02:28.54J4k3and you do realize it *Does* fail, and you get to pull your hair out figuring out why
02:29.01J4k3$150.  trust me.  just do it.
02:29.10J4k3if you need support, jake messenger will sell you some
02:29.16J4k3(no, I'm not jake messenger)
02:29.16Maxxedmmm, and find support where?
02:29.22J4k3www.portmasters.com
02:29.22Maxxedhah :p
02:29.28J4k3he'll sell you gauranteed units too
02:29.33Maxxedmmmm
02:29.35J4k3he's in houston, I bought my 2nd PM from him
02:29.39Maxxedno lie, hell im in houston
02:29.43J4k3haha
02:29.48Maxxedwell shit, this might be the way to go
02:29.52J4k3dude, call him
02:29.54J4k3he's a good guy
02:30.23Maxxedwith these will i have the ability to hookem up to a linux box
02:30.27J4k3yes
02:30.28Maxxedand see the modems in the /dev/
02:30.32Maxxedno joke?
02:30.32J4k3they'll do telnet and rlogin
02:30.32Nuggettelnet is eeeeeeevil!
02:30.35J4k3you can dial in/out
02:30.39Maxxeddamn!
02:30.39J4k3where are you getting your T1s from?
02:30.47J4k3SBC?  colo?
02:31.02Maxxedso i can see the modems like /dev/ttymodem1 2 3, etc
02:31.03Maxxednice
02:31.12Maxxedi havent priced em out yet
02:31.17J4k3if the world was less evil, I'd still be colo'd at 5959 corporate, and I'd be selling you colo and a $300/mo PRI :)
02:31.26*** join/#asterisk Strom_M (n=strom@216.64.24.250)
02:31.33Maxxedil jus need 1pri for now, but i expect growth
02:31.43Maxxed5959 corp?
02:31.46J4k3yeah
02:32.00J4k3focal -> broadwing -> (bankrupt) colo house
02:32.02Maxxedi have a few cabs at level3 in the gunspoint hood, and a cage at XO off kirby
02:32.09J4k3also verizon wireless's CO for this LATA
02:32.19Maxxedcolo house?
02:32.31J4k3kinda... its a bunch of random stuff in there
02:32.35*** join/#asterisk sadmin (i=sadmin@203.81.208.243)
02:32.38J4k3vzw does ts out of there
02:32.40Maxxedwheres the joint at?
02:32.42J4k3its in the hood tho
02:32.50J4k3a block off bw8, a block south of harwin
02:32.51Maxxedshit i live in the hood :p
02:32.57J4k3right by directron/axiontech's new office
02:33.02Maxxedah, yeah thats hood allright :p
02:33.13Maxxedhow much for a cab with 20a of power?
02:33.21Maxxedtha nickle
02:33.22Maxxedheh
02:33.52Maxxedhomstead
02:34.01Maxxedi wint to sam houston sr
02:34.06J4k3ahh
02:34.12Maxxedlowest test scores in the STATE
02:34.12J4k3I grew up about 6 blocks south of aldine high school
02:34.14J4k3on sweetwater
02:34.17J4k3block off I45
02:34.18MaxxedNO SHIT
02:34.21J4k3I'm amazed I don't have lung cancer
02:34.21Maxxedfuck man, gulfbank
02:34.25J4k3word
02:34.28Maxxedi know that hood fa sho
02:34.40Maxxedbuncha my buds stay out that way
02:34.46Maxxedheather glen n shit
02:34.50Maxxedacers
02:35.01J4k3I went to bethune for 5th grade
02:35.09J4k3all up in acres homes
02:35.17Maxxeddamn, what are the odds finding a hood nigga in a nurdy ass asterisk channel
02:35.18Maxxedheh
02:35.40J4k3haha...  random people on the internet
02:35.44Maxxedwhat kinda hook up u got on colo?
02:35.49J4k3everybody's gotta hustle one way or another
02:36.01J4k3nothing anymore.  I got a t1 to my house in the woods :|
02:36.06J4k3running a smalltime wisp
02:36.07Maxxedim looking for a full cab with 20amps, carrier nutral
02:36.14Maxxedcool beans
02:36.28Maxxedwhat kind of coverage u got going?
02:37.26J4k3pretty small....  I've got a coverage map somewhere...
02:37.51Maxxedyou using proxim gear or bootleging 802.11
02:37.54Maxxedheh
02:38.05J4k3bootlegged 802.11, of course.
02:38.07J4k3;)
02:38.08Maxxednice
02:38.20Maxxedkick it up a few hundred watts
02:38.21Maxxedhaha
02:38.23J4k3hey, as ubiquiti networks says...  802.11 is the working alternative to wimax! :)
02:38.27J4k3no need
02:38.41Maxxedu got some fine tuned arials n shit
02:38.49Maxxedpringal cans
02:38.51J4k3900 is quiet around here... the ubiquiti sr9 has treated me quite well
02:38.52Maxxedcantenas
02:38.52Maxxedheh
02:38.57Maxxedsweet
02:39.13J4k3nah... too much hassle doing it the ghetto way, too cheap to do it right
02:39.25Maxxedheh
02:39.30Maxxedi guess if ur ballin like that
02:39.55J4k3$250 installs aren't a hard sale...  sat installers here won't move for less than $300
02:40.03J4k3and usually its like $500-700 before they leave... its insane
02:40.06Maxxedya thats not bad
02:40.12Maxxedwhats the cost per month
02:40.17J4k3$29-49
02:40.28Maxxedwhat kind of bandwith u have at hq/base
02:40.29J4k3but the rates suck so far... 384-768
02:40.36J4k32xT1
02:40.39Maxxednice
02:40.45Maxxeddamn dewd how many customers u have?
02:40.45J4k3working on something better, quick
02:40.52Maxxedsounds like a sweet lil biz
02:40.55J4k3about 80 so far
02:41.00J4k3just got to find a spot thats just right
02:41.10Maxxeddamn, that some nice hussle
02:41.13J4k3not dense enough to give the cable co/telco a hard-on
02:41.27J4k3not really, its growing though
02:41.31J4k3I'm lazy, its an issue.
02:41.38Maxxedhaha, story of my life
02:41.46Maxxeddoor to door sales? word of mouth?
02:42.11J4k3wom mostly...  converting over dialup customers
02:42.18Maxxednice
02:42.52J4k3about to do some d2d in easy-install places....  gotta get a flyer out, but my graphics designer appears to keep himself overmedicated.
02:42.55J4k3:(
02:43.03J4k3so I think I need to find a new one sooner or later
02:43.17Maxxedheh
02:43.34Maxxedhow old r ya?
02:44.08J4k329
02:44.14Maxxeddamn old school
02:44.16Maxxedheh
02:45.39osirisdo you guys support trixbox at all ?
02:45.53osirisor freepbx for that matter
02:46.12J4k3I think its considered beyond the scope of #asterisk
02:46.35osirisim trying to learn trixbox, on a VERY limited timeframe.
02:46.54osirisi work for a voip provider, and want to get one working
02:47.13J4k3its pretty simple
02:47.44JTthe voip provider wants to use trixbox? :p
02:47.48J4k3you set up extensions (phones), you set up your provider... then you use incoming and outgoing routes to decide how calls should be routed in and out
02:47.53Maxxedwell im dipin out for grub
02:48.01Maxxedyo J4k3 it was cool chatin with ya
02:48.01osirisno, i want to learn it to provide it as a solution for customers
02:48.06J4k3haha, a good texan... eating at 10pm
02:48.20Maxxedil have to talk u out for a beer
02:48.34Maxxedsound like ur full of hook ups and information :D
02:48.37Maxxedhalla!
02:48.39J4k3haha
02:49.02osirisi have an old sunrocket innomedia ATA, and a x-ten softphone that register with the trixbox, and the trunk registers
02:49.50osirisi either get about 30 seconds of ringback, with the softswitch intercept, or i get instant "call can not be completed as dialed"
02:50.15osirisboth are intercepts from the softswitch im registering to
02:50.48J4k3can you call voicemail, 611, etc?
02:50.54J4k3on the local box?
02:51.24osirisnot sure.  like i said.  i know about 0 about trixbox, and im learning telephony
02:51.44osiris611 is the VM dial code ?
02:52.53J4k3*98
02:52.54osirisi deal with pap2's, mediatrix 2102, innomedia MTA's, epygi PBX's, etc.
02:52.55J4k3is the default
02:53.05osirisi just know nothing about this system
02:53.09J4k3611 gets weather
02:53.57osiris*98#  does get me asterisk VM mesage
02:54.14osirisfrom the inno
02:54.58osirisim sure my dialmaps are wrong and what not, but im just trying to get asterisk to pickup/intercept the call on inbound right now
02:55.31*** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net)
02:55.39osirissorry if im flooding or breaking basic n00b rules.  i havent ACTUALLY got on irc in a while
02:56.23J4k3dunno, its pretty quiet tonight
02:56.55J4k3being 'nice' on IRC is like being nice on a CB radio...   its all fun and games til somebody sticks a push-pin in your coax.
02:57.19*** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
02:58.58osirisheh.  and im fresh out of butt-splices
02:59.02osiris;)
02:59.29osirisformer pro home theater installer too.  nice coax joke
03:00.20osirisits funny.  your nick reminds me of one of my favorite customers
03:00.27osirishis names jake
03:01.16osirisi would laugh if you were actaully him.  small world'ing the internet and all.
03:01.51J4k3haha
03:02.11*** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga)
03:03.10osirisIOS tech ?
03:06.40J4k3nope
03:07.50*** join/#asterisk docelmo (n=vircuser@c-76-99-153-242.hsd1.de.comcast.net)
03:15.25*** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn)
03:18.56*** join/#asterisk tzafrir (n=tzafrir@62.90.10.53)
03:38.01ectospasmanyone here have the DCAP?
03:38.34russellbi wish i did :)
03:38.59ectospasmI took the DCAP this afternoon
03:39.04russellbMontreal?
03:39.05ectospasmI got a 95 on the practical
03:39.08ectospasmHuntsville
03:39.14russellboh, nice
03:39.28ectospasmthey haven't graded the written yet
03:39.35russellbgotcha ...
03:39.40ectospasmwhich is what I'm nervous about that
03:39.47russellbI need to go take it one of the days it's being done in Huntsville
03:39.50russellb(i work for digium)
03:40.04russellboh, ha
03:40.14ectospasmyeah, I kinda figured that
03:40.19russellb:p
03:40.24ectospasmyou came to my birthday suarez
03:40.40ectospasmwith your fiance
03:40.45codefreezelove the nick; not ectoplasm, but ectospasm. Cool.
03:40.51*** join/#asterisk d3wayne (n=deeewayn@c-71-228-186-75.hsd1.al.comcast.net)
03:40.51*** mode/#asterisk [+o d3wayne] by ChanServ
03:41.43russellbectospasm: yep, i remember :)
03:42.03ectospasmcodefreeze:  my license plate says 'ECTO'
03:42.22ectospasmsince most people just abbreviate my nick to ecto anyway
03:42.56codefreezeI picture a black 'stealth' sort of vehicle...
03:43.10ectospasmnah, I ain't quite that cool
03:43.38ectospasma lot of people ask me,"Is that from Ghostbusters?"  "No, that was the ECTO1."
03:44.16codefreezebetter than "Staypuf"
03:44.20JunK-Yectospasm: dcap in montreal?
03:44.33ectospasmHuntsville
03:44.34JunK-Yha, sorry, :)
03:44.37ectospasmheheh
03:44.59JunK-Ycause there was dcap in montreal, too
03:45.17JunK-Yrussellb: do we know how many dcap ppl world-wide exists?
03:45.28codefreezeIf your license plate said "staypuf", I'd picture an old VW van, light colored, of course...
03:46.16russellbJunK-Y: i don't know
03:46.37JunK-Ythat would be great to know, just for curiosity.
03:47.38fileJunK-Y: [TK]D-Fender and Sascha went off to find a pub, you should have gone
03:47.58JunK-Ytoo bad, im already at home.
03:48.05d3wayneO.o
03:48.13JunK-Yjulie is sick, she left the meeting after you, since she was feeling really bad.
03:48.35ectospasmJunK-Y:  I heard a figure that put it around 400 today, but I have no clue how accurate that is
03:48.59JunK-Y400 what?
03:49.23JunK-Y[TK]D-Fender: STOP DRINKING!
03:51.44ectospasm400 worldwide DCAPs
03:52.00ectospasmbut I have no idea how accurate that is
03:52.11ectospasmAnd 400 may not have been the number I heard
03:52.14ectospasmIt may be like 200
03:52.14JunK-Yme neither
03:52.28ectospasmAll I know is Digium has a large concentration of them
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03:53.03NivexDrunken Concentration of Asterisk Programmers!
03:56.10russellbwheeeeeeeeeee
03:57.12ectospasmI'm supposedly gonna go meet up with a couple of dcappers in a bit
03:59.11Corydon76-digSeems low
04:00.46JT~phones
04:00.46jbotmethinks phones is http://bani.anime.net/phones/.  While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows:  Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else.  Do not consider Grandstream phones.  Ever.
04:00.49JunK-Yfile: if you find a quick solution for the .version, let me know.
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04:02.29rob0DCAP-itation
04:03.00Corydon76-digectospasm: irony is when the reseller agreement forces a developer to get a dCAP...
04:04.34ectospasmdo they get a discount on the cost?
04:04.44J4k3~gs
04:04.45jbotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
04:04.53J4k3~jt
04:04.54jbotTemplate to compose LaTeX jewel case CD inserts. URL: http://www-stud.enst.fr/~michon/realisations.html
04:05.00J4k3weaksauce
04:05.40Corydon76-digectospasm: Yes, I did
04:06.01russellbdo they still give out plaques?
04:06.10Corydon76-digpretty much because the SW guys thought it was ridiculous
04:06.18ectospasmyou gotta buy 'em, but yes russellb
04:06.43JunK-Ywe can buy them? how much?
04:06.55ectospasmI dunno
04:06.58J4k3I hear down in phoenix, they give out the plague.
04:07.02russellbgive me a million and i'll get you one by Sunday
04:07.03ectospasmouch
04:07.15ectospasmI wouldn't want the plague
04:07.19ectospasm:-D
04:07.20J4k3http://www.kpho.com/news/14107744/detail.html
04:07.22J4k3^ plague
04:07.23JunK-Y1 million monopoly dollars sounds cool?
04:07.30russellbno!
04:07.37ectospasmJ4k3:  you can't win tonight, can you?
04:07.39rob0Actually unless you get the septicemic variety, it's quite controllable.
04:07.41JunK-Yrussellb: when you do arriving in phoenix btw?
04:07.45russellbi'll even take 1 million canadian dollars, just because
04:07.49russellbJunK-Y: don't remember
04:07.55russellbmonday sometime
04:08.04JunK-Ycdn ~ us since few months
04:08.07rob0but yes, plague is no fun, even if not life-threatening.
04:08.10Corydon76-digrussellb: that's worth $2.86 US, right?
04:08.31JunK-Y1 Canadian dollar = 0.966277 U.S. dollars
04:08.50russellbthat is dangerously close
04:09.18JunK-Yfinancial canadian experts said we gonna beat us dollars by the end of nov.
04:09.36russellbcanadian financial experts?  oxymoron?
04:09.44JunK-Yask jsmith, he got a canadian dollar :)
04:09.52JunK-Yhahaha
04:10.11Corydon76-digUm, yeah, but anybody making predictions on the financial markets are gypsy fortunetellers.
04:10.21Corydon76-digActually, I think the gypsies do a better job
04:10.30ectospasmno one can see the end of this sub-prime mess
04:10.35russellbwhat about jipsies
04:10.46rob0tramps and thieves
04:10.59russellbi know, blame everything on the sub-prime market
04:11.06russellbin fact, i think the sub-prime market caused some asterisk bugs
04:11.14ectospasmheheheh
04:11.33ectospasmI heard terrorists and drug cartels are using Asterisk
04:11.49coppicethere were plenty of prime subs on the market when the cold war ended
04:11.53russellbterrorist cells have IT departments?
04:12.03ectospasmOh, yeah!
04:12.19russellbcoppice: nice
04:12.27ectospasmWho do you think attacked the root DNS servers a couple of years ago?
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04:16.05Corydon76-digSo much for "greed is good"
04:16.29coppicethey didn't actually say what it was good for
04:17.23coppicea friend's brother in law visits the mercedes dealer at every minor economic downturn - he's a lawyer dealing with insolvency
04:19.22russellbf/wi3
04:19.32russellbs/f\/wi3//
04:19.57ectospasmheh
04:25.35Teln1100Awhat is a subprime mess?
04:26.12russellbsomething totally unrelated to asterisk :)
04:26.14coppicewhen someone who lost all him money lands after jumping from the 40th floor
04:26.38Teln1100Ais it banks lending at really low rates?
04:26.46Teln1100Aor people defaulting on loans
04:27.36coppiceIts "Savings and Loans II, the Sequel"
04:28.09Corydon76-digIt's banks lending to people who have subprime credit ratings
04:28.43Teln1100Adoes it also mean : slowing down economy
04:29.02coppiceI might mean serious economic collapse
04:29.17Corydon76-digBanks can get away with a small percentage of defaults, but when something like 50% of the subprime market is expected to default, that's what makes banks go belly-up
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04:29.44Teln1100Aso thats happening in Canada or US?
04:30.49coppicewhat happens in the US always hurts everyone
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04:31.07coppicewhen the US economy sneezes, europe gets flu
04:32.34coppicee.g. http://news.bbc.co.uk/2/hi/business/6994160.stm
04:35.22Teln1100Awhats the deal with ABN? is it being sold and if so why has it been in the news for so long
04:36.14Corydon76-digBoth
04:40.48QwellWhat's ABN?
04:41.05coppicea right bunch of bankers
04:41.05Teln1100AABn Amro Bank, again nothing to do with Asterisk
04:41.11Qwelloh
04:45.30ectospasmI've run this by angler and Strom_M, and I can't figure it out
04:46.49ectospasmI've got an IAX trunk and an IAXy, and the only way I can get incoming calls over the IAX trunk to work is if IAXy password is the same as the trunk password
04:47.32ectospasmIf the passwords are different, I get "<IAX trunk IP> is unable to authenticate as iaxyuser" in the CLI
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05:04.40dan__tWhat kind of things should I be looking at/for if I wanted to do something like accept some sort of input over the line, like a series of key presses or if I wanted to get tricky, some voice recognition to determine the input?
05:04.59dan__tlike the person is prompted to give some input, and then have * execute some command or whatever
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05:10.18ectospasmdan__t:  the Read() app can do basic input (I *think*), but an AGI script could do more
05:10.30dan__tAGI?
05:10.36dan__tSorry, still kinda new heh.
05:10.49ectospasmAsterisk Gateway Interface
05:11.04dan__tAh
05:11.56ectospasmhttp://www.voip-info.org/wiki/view/Asterisk+cmd+Read
05:12.28dan__tYep, reading that one right now.
05:12.28ectospasmhttp://www.voip-info.org/wiki-Asterisk+AGI
05:12.29dan__tThank you.
05:12.41ectospasmAGI can do a lot of powerful stuff
05:14.50dan__tLooks like it.
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05:20.28dan__tThanks
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05:42.49iPod-nanoAnybody know anything about the DVG-1120S ATA?
05:47.55ectospasmnope
05:48.04ectospasmor I should say,"Not I"
05:54.14dan__twtf?
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06:05.16_10nix_i have a dvg-120s
06:05.16_10nix_er 1120s
06:05.16_10nix_what do you need to know?
06:05.31dan__toh heh
06:05.31dan__tsorry wrong window
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06:42.00[TK]D-FenderJunK-Y, file : Just got in.  Rain sucked for driving, but the food was good where we went.  Anyways, great meet-up.  Will have to schedule again soon.
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07:02.14nickzxcvhi. so is there a device that would be powered by the fxs at the central office and if it detects no dialtone from another fxs (say from asterisk) switches the other side of the asterisk fxs line to be directly connected to the central office?
07:10.19pkunkrawonder if there's a way to short out his stereo....
07:11.05pkunkratoo bad EMP devices aren't directional.....
07:11.41nickzxcvsort of like this:
07:11.44nickzxcv<PROTECTED>
07:11.44nickzxcv<PROTECTED>
07:11.44nickzxcvFXO--|---|- /-|----FXS at the central office
07:11.44nickzxcv*box |   |  | |
07:11.44nickzxcvFXS--|---|- \-|----FXS powered phone set
07:11.46nickzxcv<PROTECTED>
07:11.49nickzxcv<PROTECTED>
07:11.51nickzxcv<PROTECTED>
07:11.54nickzxcv<PROTECTED>
07:11.56nickzxcv<PROTECTED>
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07:23.54Mavvieis euhm... is a 1GHz VIA CPU fast enough to convert from SIP to PRI and from PRI to SIP ?
07:29.16dukihello all.
07:30.03dukiwhen running asterisk -vvvc, I got this warning:
07:30.06dukiloader.c:360 load_dynamic_module: Error loading module 'pbx_functions.so': /usr/lib/asterisk/modules/pbx_functions.so: cannot open shared object file: No such file or directory
07:30.39dukiand using find, I didn't dind it.
07:30.56dukiI am under Linux (archlinux).
07:31.07dukiAny idea?
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09:10.08Kernel_Corehi all
09:10.18Kernel_Coreis it possible to use Speex with IAX2 Trunk ?
09:10.44Kernel_Coredoes it decrease the performance ?
09:12.54JerJeryou can run the speex codec
09:13.05JerJerperformance is a subjective thing
09:13.51Kernel_Coreops
09:13.58Kernel_CoreI mean Quality JerJer
09:14.09JerJerMavvie: If you can get the PRI card to play nice with the VIA PCI bridge - early mini-itx MBs used a really crappy pci bridge
09:14.46Kernel_CoreJerJer: I have another question.... I have between 5-15% Loss Packet , and I want to get g729 like quality ... is it possible with Speex ?
09:15.02JerJerKernel_Core:  again that is subjective  - some say they can hear the compression - others have no clue
09:15.28JerJerKernel_Core:  only thing I can say is to test it in your specific environment
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09:16.08JerJerif you have a bunch of DJs and audiophiles as customers, they are going to hear the compression of any compressed voice codec
09:16.29JerJerif you have a bunch of teenagers who are used to cell phones, they won't have any idea a compressed codec is getting used
09:18.24JerJeri believe speex has an adaptive packet loss process, so speex may very well do better in a packet loss environment
09:18.51JerJerI have one project that does voice over wireless (wifi) and speex does perform much better than any other codec
09:19.54Kernel_CoreJerJer: what about iLBC?
09:20.07Kernel_CoreI heard it handles 3-4% Loss Packet with MOS 4
09:20.31JerJerat the time we were testing, we could not use iLBC
09:20.56JerJeragain you need to test it for your specific environment
09:21.34JerJerone thing many people do not fully comprehend about VoIP is testing is paramount
09:22.00Kernel_CoreJerJer: as you experienced with Speex ... does complexity 10 effect on Loss Packet ?
09:22.11JerJerone specific configuration may work for one project but would totally not be acceptable for another project
09:23.03Kernel_CoreOK!...
09:23.09Kernel_Corethank you for clue...
09:23.28Kernel_Coreand I have a question about speex CPU usage ...
09:24.16Kernel_Corewith P4 2.8 ( just translates speex to g711)
09:24.27Strom_Mgood god, why am I awake at this hour?
09:24.37Kernel_Corehow many channels can handle ? :)
09:24.40Kernel_Core30 ?
09:24.42Kernel_Coreor more ?
09:24.46JerJerspeex is going to use more cpu than G.729
09:24.54Kernel_Corehow much ?
09:24.56Kernel_Coretwice ?
09:25.10JerJeresp a DSP based implementation - like the TC card from Digium
09:25.27JerJerdepends on your CPU - do a show translation in your asterisk CLI
09:26.03Strom_MKernel_Core: also, try running calls through your system while you're running "top" and see how it affects your load
09:26.15JerJeron one of my lower end systems I have  24 for speex and 15 for g.729
09:26.28Kernel_Corespeex ----> uLaw 23
09:27.24Kernel_CoreStrom_M: if it uses more than 70% CPU does it affect the quality ?
09:27.47Strom_MKernel_Core: you'll have to test it.
09:28.03JerJeryeah what Strom_M said
09:28.34Strom_Mlike JerJer said, a lot of this is subjective and you can't put it together solely based on numbers on a chart or spreadsheet or whatnot
09:28.49Strom_Myou have to put it together and then actually try to use it
09:29.27Kernel_CoreStrom_M: Thank you ....
09:29.30JerJeryep
09:30.14Kernel_CoreI will test and later write the results on voip-info site :)
09:31.25Strom_Mwelcome to #asterisk-and-terrible-english
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09:31.56Strom_Mbrought to you by rice-a-roni for no readily apparent reason
09:36.05dan__tAnyone care to lend me a hand with using an IAX2 provider for outbound calls?  I'm having quite a bit of trouble pinning it down.
09:36.15dan__tI've got inbound calls working fine, that's cool - just the outbound calls that are bugging me.
09:36.56Strom_Mdan__t: sure
09:37.17Strom_Mpastebin the console output of a call setup attempt
09:37.23dan__tThank you, I'd appreciate it.
09:37.24dan__tOk, hold on
09:37.31dan__tEr, should I have any particular debug options on?
09:37.51Strom_Mjust set verbose 10 or whatever
09:38.33dan__tWell, here's the cool part - I don't see any activity in * when I attempt an outbound call.
09:38.43dan__tI'm thinking I simply have the extension set up improperly.
09:38.48Strom_Mare you calling from a sip phone?
09:38.52dan__tI am.
09:38.56dan__tA PolyCom 601
09:39.07Strom_Mdid you include the outbound context in the context the phone lives in?
09:40.29dan__tI'm sorry, I'm not sure on that one.
09:40.42dan__ter, wait
09:41.00dan__tI have a context= declaration in sip.conf for the actual phone, in a [dan] context - being my phone
09:41.11dan__tCan I declare multiple context= lines in a sip.conf context entry?
09:41.14Strom_Mno no
09:41.41Strom_Myou want to make sure [dan] includes whatever context contains the extension which calls your ITSP
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09:42.05dan__tBut [dan] still belongs in sip.conf, yes?
09:42.20Strom_Myes
09:42.30Strom_Mcontext= whatever
09:42.48Strom_Mhere, just pastebin your sip.conf and extensions.conf and i can tell you exactly what you need to do
09:42.58dan__tSure, give me a second please.
09:45.34dan__thttp://pastebin.ca/698393
09:45.43dan__tI'm sooo close heh.
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09:46.28Strom_Myour outbound dialing rule should not be in an "incoming" context
09:47.04Strom_Mgive it a separate "outbound" context, and then in your "internal" context, add the line "include => outbound"
09:47.44Strom_Mfinally, take that silly 30,tr off your outbound Dial() statement
09:47.55dan__tDo I need to make outbound-local and outbound-long-distance separate contexts right now?
09:48.04Strom_Mnot unless you want to
09:49.05dan__tOk, outbound dialing rule should not be in an incoming context... so line 32 is in the wrong place?
09:49.34Strom_Mcorrect
09:53.14dan__tOk... I think I got it fixed, removed a lot of the other BS, as well
09:53.18dan__tStill have the same results, however
09:53.30Strom_Mpastebin what you have now
09:53.36dan__tSure, one second please.
09:54.40dan__thttp://pastebin.ca/698395
09:55.25Strom_Mwhy did you obliterate your internal context and then create a new one in sip.conf?
09:57.57dan__tWow, because I obviously wasn't thinking.
09:58.53dan__tAnd I totally just gave myself a high five for not making a backup.
09:59.23dan__tMaybe I should do this when I've had more sleep
09:59.27Strom_Mheh
09:59.30Strom_Mit's not hard man
09:59.36Strom_Myou've got a "backup" on pastebin
09:59.41dan__tI was looking for it
10:00.11dan__tgot it, hold on a sec.
10:02.29dan__tOk, the [internal] context still belongs in sip.conf, yes?
10:02.35Strom_Mno
10:02.40dan__tand that's where I include a reference to the outgoing context
10:02.43dan__toh
10:02.52Strom_Mthe context= line in your sip.conf entry for your phone references a context in extensions.conf
10:03.00dan__tYes.
10:05.17dan__tAlright, you know what.
10:05.27dan__tI'm going to re-visit this tomorrow.  No sense in wasting anyone else's time on this.
10:07.15Strom_Mhow is this wasting my time?
10:14.04Strom_Mi guess we shall never know
10:14.10dan__ter sorry heh
10:14.40dan__tbecause we'll backtrack forever on account of my drowsiness, and I can't "use up" good help on account of that :)
10:15.15Strom_Mdude, you're like two steps away from a solution
10:15.22Strom_Mjust do what I tell you
10:15.52Strom_Mput your old "internal" context in extensions.conf rather than sip.conf, and add include => outbound
10:15.52Strom_Msave, reload, done
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10:20.15*** join/#asterisk ThoMe (n=tm@tm.muc.de)
10:20.21ThoMehello
10:20.36ThoMeif i like for out-calls press a 0 is this ok?: exten => _*XXXXX.
10:20.37ThoMe?
10:21.52dan__tI think something is up with my phone now, it's not booting.
10:21.53dan__tAwesome.
10:21.59Strom_Myour question makes no sense
10:22.25Strom_Mdan__t: did you reboot it?
10:22.47dan__tYep, these phones take a bit to reboot anyway.
10:22.56Strom_Mwhy did you reboot it?
10:23.14dan__tBecause it's ugly.
10:23.36dan__tNo, honestly, I moved it.
10:23.41Strom_Moh ok
10:23.50dan__tbetter than reaching across the desk
10:24.16MavvieJerJer: in that case I'm not going to suggest it as a solution
10:26.32dan__tAnd I was hoping to make it use dhcp.  BUt it's not working.  So that's a problem for later.
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10:29.45ZenJabbahaving trouble compiling asterisk on a XEN box, specifically the zap part.
10:30.08ZenJabbait is unable to find my sources for the linux version, but I have it all in the right place, any ideas on where I am going wrong?
10:31.02ZenJabbaand is asterisk 1.4.11 easier to compile :)
10:33.04tzafrir_homeZenJabba, what distro is that?
10:33.34dan__tOk, ended up with this, Strom_M -> http://pastebin.ca/698411
10:34.50Strom_Mdan__t: in theory that should work
10:34.58dan__thrm
10:35.23ZenJabbaUbuntu 7.04 with Xen extentions
10:35.32dan__tI get that fast tone
10:37.22Strom_Mdan__t: did you reload your configs?
10:37.26dan__tI did.
10:37.34Strom_MZenJabba: do you have the linux-headers package installed?
10:38.17Strom_Mdan__t: do you get any console output now?
10:38.26dan__tNot when placing an outbound call, no.
10:38.31dan__tInbound, yes.
10:39.11Strom_Mdo me a favor...save your files and type "reload" on the console
10:39.33dan__tThey're not open, and I did.
10:39.51dan__teven looked at the filenames just to make sure heheh
10:40.12*** join/#asterisk michael-i (n=michael-@Wa877.w.pppool.de)
10:40.27ZenJabbaStrom_M: Sorry, child screaming in my ear.. I don't have a standard ubuntu kernel...
10:40.39ZenJabbaI have /usr/src/linux-2.6.16.29-xen3-U
10:40.54ZenJabbaand below that a include directory, which according to my pea brain, is the right way to have it setup
10:41.06ZenJabbauname -r returns 2.6.16.29-xen3-U-pae
10:41.13dan__tI see debug during reload, about adding extensions 100,611 to internal, and including context outgoing into context internal
10:41.38Strom_Mand you're dialing 1+NPA+NXX+XXXX right?
10:41.48dan__tSure am.
10:42.02Strom_Mtype "set verbose 10" at the console
10:42.05Strom_Mthen try another call
10:42.28dan__tNot a thing.
10:43.10Strom_Mdo calls to 611 and 100 work>
10:43.11Strom_M?
10:43.40dan__tNo.  I'll be damned.
10:44.36Strom_Mare you sure the phone is set up correctly?
10:45.06dan__tNope.
10:45.09dan__tJust went through it again.
10:45.10ZenJabbayou don't have a dialplan on your phone do you dan__t that is failing the call before getting to the asterisk box?
10:45.34dan__tI was not completely aware that the phone itself needed a dialplan.
10:45.52dan__tI am booting a generic firmware-polycom package with the phone, and it is snagging the defaults
10:45.55michael-idoes anyone have experience with concurrent Dial() targets (SIP/101&ZAP/3/1234...etc) stopping after one of the ZAP channels starts ringing? My setup rings a SIP phone and two Zaptel channels, once the one Zap channel starts ringing, all others stop.
10:46.01dan__tI just adjusted some SIP credentials on the phone, let's try this.
10:47.40Strom_Mmichael-i: what's on the zaptel channel?
10:48.12dan__tThat must be it, I guess.
10:48.16dan__tSomething is up with the phone.
10:48.21michael-iStrom_M, one is connected to my internal phone and the problematic one dials out through my provider to reach my cell phone
10:48.58Strom_Mmichael-i: ah, that's the thing - FXO ports don't have full pass-through of answer supervision, so they just assume the call is answered as soon as they finish outpulsing and cut through.
10:49.15ZenJabbaon my cisco phones, I can push a dialplan down the line via TFTP and if a number I dial doesn't match a dialplan, it never gets sent to asterisk
10:49.29dan__tD'oh.
10:49.31ZenJabbaie, if I don't have a NXXX and I dial a 4 digit extention, it never even makes it to asterisk
10:49.44dan__tOk, well, dissecting the Polycom dialplans looks like hell.
10:49.47dan__tMight do that tomorrow.
10:49.53dan__tBut that would make perfect sense.
10:49.55michael-iStrom_M, oh wonderful :( is there any workaround for that or is too deep to get around
10:50.14ZenJabbaglad I could help, now back to stupid fsck'n zaptel
10:50.19dan__thaha thanks.
10:50.43Strom_Mmichael-i: yeah, don't use an FXO port in that situation
10:51.03Strom_Muse an ITSP or an ISDN circuit or something that gives you proper answer supervision
10:51.08michael-iStrom_M i.e. dial the cell phone through a ITSP
10:51.09michael-igotcha
10:51.40michael-ifor now i'll just make it two ring groups, ring internal, wait a bit, ring cell
10:51.48dan__t<digitmap dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT" dialplan.digitmap.timeOut="3"/>
10:51.52michael-iStrom_M, thanks for the insight....this was really bothering me
10:52.36ZenJabbabasically that dialplan dan, lets you prent to be a normal phone
10:53.19michael-iI need to go, wanted to get forwarding working here before I leave for the day. Thanks again!
10:53.20dan__thrmmrmrmm.... ok.
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10:54.58ZenJabbawhen will they invent another timer for asterisk, so we can leave zaptel
10:57.30dan__tAs soon as I get around to it.
10:57.32dan__tSorry it's taking so long.
10:57.43ZenJabbayou'll live :)
10:58.06dan__tSo does PolyCom make utilities to make these phone config files for?
10:58.16dan__ter wtf that made no sense.
10:58.50dan__tIf I ever figured out what all the options were, I suppose it would be neat/easy to make a php-based configuration wizard
10:59.09Strom_Mdan__t: you can download the polycom admin guide...
10:59.19dan__tYep.
10:59.20dan__tI've got it.
11:02.35*** join/#asterisk PepOSX (n=pepOSX@190.72.148.113)
11:02.56dan__t%@!^%!@^#&#@
11:04.52Strom_M?
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11:07.17dan__tSo there's nothing wrong with the Asterisk setup, I just need to fix these phones.
11:07.27dan__ter, the dialplans on the phones
11:14.45dan__tI like how PolyCom has a "Publicly available" software section that links to their Extranet.
11:16.10ZenJabbadan__t: life was't meant to be easy
11:16.17dan__tSure wasn't :)
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11:26.26tzafrir_homeZenJabba, apt-get install linux-headers-`uname -r`
11:26.43ZenJabbadoesn't work, cannot find -xen extentions
11:26.45tzafrir_homethat said, ztdummy still won't work under Xen
11:26.51ZenJabbaI'm about to give up on Xen :)
11:26.54ZenJabbavmware seems to work
11:27.08tzafrir_homeZenJabba, the above works under Debian
11:27.21ZenJabbaI have it working perfectly on a non-virtual machine
11:27.29ZenJabbaI'm trying to get it working under virtual machines, failing badly
11:27.42ZenJabbaanybody know any good virtual machine hosting services in the states
11:27.45tzafrir_homeZenJabba, do you really need Xen? why not go for linux-vserver?
11:27.58ZenJabbaI'm using a virtual hosting service
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11:53.15dan__tDamnit, Strom left heh.
11:53.19dan__tWell, I got the phone to work.
11:53.26dan__tI can see data in the asterisk console now.
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11:57.37dan__tAnyone seen this one before - [Sep 15 04:57:05] WARNING[3895]: chan_sip.c:8272 check_auth: username mismatch, have <dan>, digest has <>
11:57.44dan__tGoogle suggests that it's actually a bug
12:05.11dan__tHere's the error in full - http://pastebin.ca/698449
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12:17.42ZenJabbaline 33 does it for me dan...
12:17.58ZenJabbait isn't passing a username
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12:18.18ZenJabbanow my apache is seg faulting
12:18.21ZenJabbafucken thin
12:18.55JunK-Yapache segfaulted, wow, pretty rare.
12:19.10ZenJabbayea, no idea what I did :)
12:19.22JunK-Yu served too much porn!
12:19.32ZenJabbaI serv one file.. proxy.pac
12:20.18dan__tZenJabba, alright, does that mean I need to configure line 1 in the phone?
12:20.25dan__ter, are you familiar with the Polycom 601's?
12:21.05ZenJabbano, I cannot deal with polycom
12:21.09dan__thaha
12:21.14dan__tI'm starting to completely understand why.
12:21.16ZenJabbabut what is happening on line 33, its telling you that it isn't getting username
12:21.30dan__tyeah, just was wondering in which part
12:21.53dan__tLooks like you have to configure each line independently
12:22.12dan__tAnd that I have to use a SIP proxy, that's why, from earlier, I was seeing nothing being sent to *
12:23.18ZenJabbawhy would you use a sip proxy internally?
12:23.25GodseyI thought my extension had been trying to dial out but now I'm not sure
12:23.44dan__tThat's the only way that I can see any kind of connection attempts against * from the phone.
12:23.50dan__tRemember earlier, I was being told to look at the debug?
12:23.58dan__tI just did some trial and error stuff and saw that.
12:24.01Godseyit seems when I don't answer and it goes to voicemail, cdr logs the remote caller id with NO ANSWER in the outgoing context because that's the context the friend is in
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12:28.16dan__tYeah, see, I take the proxy out of the equation and I don't see any outbound activity in the console.
12:28.38dan__tThat's nuts.
12:28.54ZenJabbaso the polycom needs a proxy to register
12:30.18dan__tSounds about right.
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12:36.30Godseyis the Aastra 9112i an ok phone?
12:37.00GodseyI want a low cost sip phone w/ speaker phone
12:37.12ZenJabbaI've used the Linksys 941 and Cisco 7960
12:37.31yannj_frgodsay: look at globalsources.com
12:39.18ZenJabbaXen is basically a piece of crapola
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12:39.32GodseyZenJabba: the hypervisor?
12:39.37ZenJabbayup
12:39.42Godseywhy don't you like it?
12:39.51ZenJabbaI just cannot get asterisk compiled init
12:39.57Godseyoh :)
12:40.07ZenJabbagetting timing errors
12:41.03Godseyyou using hardware?
12:41.04coppiceyou put something real time on a shared system, and you get timing error? who'da thought? :-)
12:41.25GodseyI had xen and asterisk happy
12:41.45Godseybut, I didn't use any hardware or meetme
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12:44.33GodseyI used app_conference I think
12:44.46Godseybut now it looks like freeswitch may work better *shrug*
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12:48.05Godseysorry callweaver not freeswitch
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12:54.55dan__tWhich port does a SIP proxy run on?
12:55.45dan__thrm guess it's still 560
12:55.47dan__t5060, too
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12:56.48dijungalhi... if i wanna run a IAX trunk between 2 * servers on g729, do i need only 1 license per server from digium?
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12:58.06sniper[FOO]dijungal: I don't think you'll need a license if you can avoid audio transcoding
12:58.37sniper[FOO]hi all
12:59.15voipnet-techmorning all, has anyone done paging/intercom with Aastra phones (auto-answer via SIP message) ?
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13:00.49dijungalsniper[foo]: i wanna use g729 between the boxes because they're in different locations and i want to conserve on bandwidth
13:01.35sniper[FOO]dijungal: IC, but are you going to use any other vocoders?
13:03.06sniper[FOO]carrier-grade traffic uses g.723 and g.729, so you probably don't want to use other codecs
13:03.23sniper[FOO]question here
13:03.32*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
13:04.07sniper[FOO]how can I bridge the legs of a call and still be able to execute applications?
13:05.07sniper[FOO]talking about a dialplan, I'd need to connect media streams and then execute stuff after a given period of time
13:05.55sniper[FOO]but Dial() passes control after at least one of the channels is disc'd
13:06.30sniper[FOO]I could use a macro passed to Dial() as a parameter, but how can I connect audio, then?
13:06.43*** join/#asterisk jeebusroxors (n=jeebusro@cpe-66-74-106-65.dc.res.rr.com)
13:06.44sniper[FOO]inside the macro called by Dial()
13:08.23dijungal:|
13:08.39dijungali just wanna send calls between locations
13:08.54dijungalwe're currently using cisco -[g729] - cisco
13:09.05sniper[FOO]ouch
13:09.12sniper[FOO]that must be a pain in the ass
13:09.19dijungalwanna change that to asterisk -[IAX]-asterisk
13:09.25sniper[FOO]?
13:09.29dijungalactually it works well....
13:09.46dijungalwe're just looking to open a new location and reduce cost...
13:10.20dijungalso we need calls to go between the locations and i was readying g729 on asterisk allows 103 channels per MB
13:10.42dijungalwhile gsm would be 68 per mb
13:10.44sniper[FOO]thought you wanna use cisco > * >>>>>>>>>>> * > cisco
13:10.50dijungalif u calculate all the overhead etc.
13:10.58dijungalnope
13:11.05dijungal*<<>>>*
13:11.39dijungalthen we have sip or iax phones connected on one send and pstn or sip phones on the other
13:12.06sniper[FOO]you are using SIP-compatible IP phones in the recently opened location, right?
13:12.19rob0((X>>> gunnery sgt
13:12.22dijungalnow i know its possible... i'm just trying to figure out if i need a per channel license or would an IAX trunk act as 1 channel?
13:12.43sniper[FOO]listen
13:13.03sniper[FOO]if both endpoints use g.729 as the preferred vocoder
13:13.05dijungalall ears
13:13.31sniper[FOO]then you won't have to recode audio inbetween
13:13.48sniper[FOO]not even if you use a trunk
13:15.19sniper[FOO]you only need licenses if somewhere between the endpoints, including the endpoints themselves, you can't use complex codecs or you want to open the stream and manipulate it
13:15.42dijungalunderstand... but the endpoints would be using GSM
13:15.47sniper[FOO]why
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13:16.36dijungal<PROTECTED>
13:16.55dijungaland let asterisk do transcoding from g729 - GSM
13:17.56sniper[FOO]you will have to calculate with $10 * the number of desired parallel calls and a huge investment on hardware if you're dealing with more than 60 channels
13:18.57dijungalok so yuh suggestion is g729 on the endpoints... get phones that already have g729 on them, and use * as a pass through
13:19.19dijungalsince * will not be doing any transcoding of the audio stream, i will not need to buy a license
13:21.18sniper[FOO]this would be the preferred solution, I wouldn't go with a phone without g.723.1/g.729
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13:22.10masushi all
13:22.36masusFrom address missing 'sip:' more detail is here --> http://pastebin.ca/698483
13:24.25masushave anybody an idea ?
13:25.07dijungalok
13:25.14dijungalall our phones have g729
13:26.16rob0All your phones are belong to us. Somebody set up us the codec.
13:26.45riddleboxyou know what sucks, I bought a TDM card for 141, from a website, and aftwards was telling my boss, and he said I couldnt have gotten it from one of our distributors for 102
13:27.16dijungalthanks sniper
13:27.31sniper[FOO]you're welcome
13:28.23sniper[FOO]anyone here?
13:29.05dijungali am
13:29.06dijungal:|
13:29.08sniper[FOO]:)
13:29.13dijungalthe others have hangovers
13:29.19sniper[FOO]IC
13:29.28sniper[FOO]yep, it's saturday
13:29.35sniper[FOO]reasonable :)
13:29.49dijungalrussellb, qwell.... they were prolly out at the Bosses ranch again....
13:29.50dijungallol
13:30.54masusdijungal: did u ask me with g729
13:30.56rob0don't TYPE so loud in here!
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13:33.41dijungalmasus???
13:34.31masusFrom address missing 'sip:' more detail is here --> http://pastebin.ca/698483
13:38.39ThoMehello?
13:39.02ThoMeif i try exten => _0.,n,Chanisavail(misdn/1&misdn/2)
13:39.02ThoMeexten => _0.,n,Dial(mISDN/${AVAILCHAN}/${EXTEN:1})
13:39.16ThoMethen try misdn use channel "0" i have only 1-4
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14:24.19alexpeHello, i have an tdm400p with fxs and fxo and when there is an incoming call on my fxo i would like asterisk to display the caller id on my analog phone befor answering the call
14:26.25rob0You want the PBX (a menu or something) to pick up the call, but only after showing CID on the phone on FXS?
14:27.05rob0I don't know, but I doubt that is possible.
14:27.32rob0If you just mean to require caller ID, that's pretty easy.
14:27.41alexpeyes, coz some people calling me from foreign country just want to leave their number and then i call them back
14:28.02russellbthat's how it will work by default
14:28.11rob0My FXOs don't have caller ID because the stupid telcos charge extra for it.
14:28.19alexpe[Incoming]
14:28.19alexpeexten => s,1,Answer
14:28.19alexpeexten => s,2,Dial(zap/g1&iax2/alex,15,rt)
14:28.24russellbunless you have something in the dialplan that answers the incoming call, asterisk will not answer it until you answer your phone when you do a Dial()
14:28.46russellbyeah, remove that Answer
14:29.03alexpeimmediate is set to no in zapata.conf
14:29.27alexpei just remove the answer command?
14:29.31russellbyes
14:29.47rob0and renumber the priorities of course :)
14:29.52russellbright :)
14:29.52alexpeok, so i don't need to answer the call???
14:30.00alexpeok
14:30.11russellbnot immediately you dont
14:30.21russellbDial() will do it automatically once someone answers the phone
14:31.25alexpe[Incoming]
14:31.25alexpeexten => s,1,Dial(zap/g1&iax2/alex,15,rt)
14:31.26alexpeexten => s,2,Answer
14:31.26alexpeexten => s,3,Playback(vm-nobodyavail)
14:31.26alexpeexten => s,4,Voicemail(u1000)
14:31.26alexpeexten => s,103,Voicemail(b1000)
14:31.36alexpeis that okM
14:31.38alexpe?
14:32.05alexpesorry i should have voicemail at 3
14:32.17alexpei should NOT have voicemail at 3
14:32.30alexpethanks, i am going to try that
14:32.33russellbwelllll ...
14:32.44russellbyou should really shouldn't be using that priority jump
14:33.02russellbtake a look at configs/extensions.conf.sample and look for the stdexten macro
14:33.12russellbfor an example of doing it with the DIALSTATUS variable
14:36.41alexpeLooks more flexible with the DIALSTATUS variable, thanks a lot ;)
14:36.49russellbno problem
14:38.32alexpeexten => 1234,n,Macro(stdexten,1234,${CONSOLE})
14:38.52alexpethis will expand Macro at priority n?
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14:39.34russellbyeah
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15:54.06shido6mic check
15:57.27shido6exchange?
15:57.31shido6what are you up to?
15:59.19hypa7iaUDP is still leaking back in somehow
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16:04.33dmcnhi - i get the following error when trying to register to freecall: Sep 15 17:35:04 NOTICE[29086]: chan_sip.c:5431 sip_reg_timeout:    -- Registration for 'mcnally4242@sip.voiparound.com' timed out, trying again (Attempt #2)
16:04.52dmcni have no problem pinging the address outside of asterisk - what could be the problem?
16:05.02hypa7iadmcn: can you telnet to it on 5060?
16:06.55apturaA few issues that my system has such as vm not playing the message but will say the phone number calling. When I hit option 3 in advanced options it plays vm. Anyone seen these issues? searched for the issue and have not seen a resolve.
16:09.56shido6recompile :)
16:10.01hypa7iaaptura: that's odd, have you looked at the console when it's recording a vmail?
16:10.29apturayes
16:11.51dmcnhypa7ia: doesn't look like it, no
16:11.56shido6one way audio?
16:12.01shido6is there nat involved?
16:12.20hypa7iadmcn: i suggest resolving that first :)
16:12.34hypa7iashido6: dmcn can't telnet to it on 5060
16:12.39hypa7iathat won't even get you one-way
16:13.01dmcnhypa7ia: my best guess is they changed the host, i have other connections on port 5060 to an other provider :|
16:13.12*** join/#asterisk saftsack (n=saftsack@pD9E06F1D.dip.t-dialin.net)
16:14.56dmcnanybody else using freecall.com who could give me info on what server they connect to? ;)
16:16.31hypa7iadmcn: i can't telnet to it either
16:17.12_x86_anyone ever setup a directory server for polycom video conferencing units (i.e. VSX7000s)?
16:18.13russellbtelnet will only check that it's listening for tcp connections, won't help with udp
16:19.18russellbhypa7ia: if you can get a sip debug trace of what is getting sent with udp, i can try to take a look sometime
16:20.30shido6shouldnt be using tcp for sip anyway :)
16:20.52russellbheh, call microsoft and tell them that
16:20.56hypa7ialol
16:21.03*** join/#asterisk linagee (n=linagee@about/linux/staff/linagee)
16:21.04russellbi'm sure they'll say "Oh, you're right!  We'll fix that right away."
16:21.07hypa7iahahahaha
16:21.09hypa7iano wai
16:21.18hypa7iarussellb: more like "fuck off"
16:21.34russellbhypa7ia: more like ...     *crickets*
16:22.11hypa7iano seriously, they've said on technet that there's no interest in implementing it
16:24.48russellbi wonder if asterisk had any part of that decision
16:25.01russellbthat's probably just conspiracy theory ... but who knows
16:26.00hypa7iarussellb: nah, it impacts a ton of sip-based stuff
16:26.08hypa7iaNortel BCM for example
16:26.21russellbthat's udp only, too?
16:26.26hypa7iayup
16:26.30russellbheh ...
16:26.34hypa7iano intention of supporting TCP
16:26.40russellbweird
16:27.10hypa7iayeah
16:27.20russellbsounds like both admitting shortcomings as a decision, and intentional mis-feature ...
16:27.35hypa7iathey don't see it as a shortcoming
16:27.54*** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
16:27.54*** mode/#asterisk [+o blitzrage] by ChanServ
16:27.58russellbright, that's what i mean
16:28.11russellbor at least on the marketing front, they don't :)
16:28.40hypa7iahehe
16:28.59russellbwell hopefully we'll have this code finalized and merged by the end of the year ....
16:29.14JunK-Yhypa7ia: hey.
16:29.15*** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
16:29.15*** mode/#asterisk [+o blitzrage] by ChanServ
16:29.32russellbjust have to find someone to spend the time to finish it up
16:29.40russellbi have so many projects on my plate :-/
16:29.50blitzrageit's true
16:29.59hypa7iahey JunK-Y
16:30.10apturawhat kind of projects?
16:30.20yannj_frrusselb : You will have to stop sleeping
16:30.20hypa7iait's seriously almost workign
16:30.32hypa7iait's probably a config issue on my part
16:30.33russellbhypa7ia: good to hear :)
16:31.23*** join/#asterisk Strom_M (n=strom@216.64.24.250)
16:33.32hypa7iaclass is out, gotta run to lunch
16:34.17*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
16:39.38apturahypa7ia still here? just read the cli and no errors poped up saying why the vm was not played.
16:43.45dmcnis anybody in here currently using freecall.com/sipdiscount.com/voipstunt.com (looks like the same company) without problems? my asterisk refuses to connect to any of the hosts, so it appears they might have shut something down?
16:44.05apturalots of these serices are shutting down
16:46.06dmcni can imagine :|
16:46.28dmcnit's just odd that i can't find any info about it in any SIP-forums
16:46.41sniper[FOO]hi there again
16:47.29sniper[FOO]I was just wondering if an application is faulty or maybe I'm missing something more trivial
16:47.33sniper[FOO]http://pastebin.com/d3bc3633c
16:47.43sniper[FOO]this is my extensions.conf
16:48.03sniper[FOO][audio-clear] is included within the default context
16:49.03*** join/#asterisk yang (i=yang@static-ip-62-75-255-124.inaddr.intergenia.de)
16:49.05sniper[FOO]I'm just curious if the NVLineDetect app has the chance to get hold on to the audio stream
16:49.18shido6take i tout
16:49.22shido6and try it
16:49.42sniper[FOO]I see the console messages and the app gets called
16:49.46shido6what does the CLI say?
16:50.27sniper[FOO]just a moment
16:51.30hypa7iaaptura: no idea, sorry
16:51.45giesenanyone here really good with callerid?
16:51.51giesenI have a really weird problem
16:52.25Strom_Mdmcn: my experience with those services is that they are so universally terrible that you don't want to even bother trying to get them working
16:52.41shido6whats the problem, giesen?
16:52.42giesenI'm dialing into DISA with a cell phone
16:52.46giesenand then making outgoing calls
16:52.54hypa7iaStrom_M: well put
16:52.57shido6yep
16:52.58giesenand I set out going caller ID
16:53.07giesenif I block caller id on my cell phone
16:53.08shido6i dont use disa for mine :)
16:53.13giesenthen asterisk's callerid is blocked as well
16:53.20shido6ok so set one
16:53.24shido6force it
16:53.24shido6:)
16:53.25giesenyeah I did
16:53.28Strom_Mgiesen: you also want to Set the caller ID presentation flag
16:53.31shido6to whatever you have legal rights to use
16:53.37Strom_Mlook at the SetCallerPres() application
16:53.38giesenStrom_M: how do I do that?
16:54.05giesenStrom_M: you just might be my hero
16:54.12Strom_Myay
16:54.17Strom_Mbuy me a beer at astricon :D
16:56.15giesenYOU ARE MY HERO
16:56.20sniper[FOO]shido6: http://pastebin.com/d512d2bcc
16:56.25giesenit was the weirdest shit
16:56.33giesenand I figured there must be something to caller ID block
16:56.42Strom_Mgiesen: yes
16:56.43giesenrather than it just being blank
16:56.57Strom_Mgiesen: read this document
16:56.58Strom_M~101
16:56.58jboti heard 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony.  You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf
16:57.18Strom_Mveeerrrryyy useful :)
16:57.27giesenwill do
16:57.35giesenI love this stuff =)
16:57.44giesenI just setup DISA using voicemail passwords
16:57.56giesenwith proper outbound caller ID on a per-user basis
16:58.36apturadisa is good if you have your own WORKING 1800 number and call into your system. point is our out of cell range or area does not cover your cell call into your box and call out
16:58.39sniper[FOO]anyone having experience with the nvlinedetect app?
16:59.55dmcnStrom_M: i had them working, my girlfriend saved ~1000 euro in about a year using it :)
17:00.09dmcnbut then i reinstalled my server from scratch and since then it's been broken :(
17:00.18*** join/#asterisk klictel (n=klictel@modemcable159.7-200-24.mc.videotron.ca)
17:00.33sniper[FOO]or, alternatively, any sort of line detection stuff in *?
17:00.59giesenaptura: yeah, I'm going to use that for my personal asterisk box
17:01.15sniper[FOO]gotta detect a UK ringing tone, but as someone on this channel pointed out, I'm only getting frustrated :S
17:01.58*** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org)
17:04.31sniper[FOO]anyone? please...
17:05.14apturaanyone here have working exeraince with a good 25 pair bix tester?
17:05.28Strom_Msniper[FOO]: use circuits with actual supervision passthrough :)
17:06.55luke-jrIs DUNDi dead?
17:07.28Qwellthe technology?
17:07.46sniper[FOO]Strom_M: I actually made this app work... got the source, realized that it's for some near-1.0 CVS version, ported it to 1.2.24, made it compile flawlessly... now it doesn't work and I really gotta find out why :S
17:09.07Strom_Msniper[FOO]: use circuits with actual supervision passthrough :)
17:09.29sniper[FOO]noticed the smiley at the end of the line :)
17:09.54Strom_Mthe rest are all ugly hacks
17:10.00sniper[FOO]is it a deliberate attempt?
17:10.12Strom_Ma deliberate attempt at what?
17:10.59sniper[FOO]...at detecting a sine wave lasting for 400 ms in HQ PCM audio
17:12.07Strom_Mwhat is this "it" you're referring to?
17:12.19Strom_Mthe smiley?  nvlinedetect?  something else?
17:12.24_x86_what's it mean when a polycom IP501 says that it failed to load MACADDRESS.cfg?
17:12.35_x86_where MACADDRESS is the mac address of the phone ;-)
17:12.42Qwell_x86_: means it failed to load MACADDRESS.cfg
17:12.47sniper[FOO]erm... my English is actually not the best :)
17:12.49_x86_i'm not provisioning the phone, just want to use a local config
17:12.54Strom_M_x86_: means it failed to load MACADDRESS.cfg
17:12.55_x86_Qwell: how do i reset it?
17:13.01_x86_Strom_M: ?
17:13.06Strom_Mhi
17:13.18_x86_can i clear the existing MACADDRESS.cfg and start over?
17:13.21Strom_MQwell: wanna grab lunch quickly before I leave?
17:13.35_x86_Strom_M: hold on a sec ;)
17:13.37QwellStrom_M: ...I just nuked a hot pocket like 3 minutes ago
17:13.41Strom_Mhah
17:13.42Strom_Mok
17:13.43sniper[FOO]Strom_M: ...at detecting a sine wave lasting for 400 ms in HQ PCM audio
17:13.47Qwellsorry ;/
17:13.50Strom_Mchick-fil-a it is
17:13.52Strom_Mno worries
17:13.52_x86_Qwell: any ideas?
17:14.01Strom_Mwe can lunch at Phoneix
17:14.08Qwellyeah, in and out FTW
17:14.10*** join/#asterisk ectospasm (n=ectospas@c-68-62-214-17.hsd1.al.comcast.net)
17:14.21Strom_Mactually, there's in-n-out AND waffle house in phoenix
17:14.26Qwellnice
17:14.31Strom_Mso we could theoretically eat both in the same day
17:14.32Qwelloh, do you know if they have AMPM there?
17:14.34_x86_die hookers ;)
17:14.38Strom_MQwell: IIRC yes
17:14.40QwellI've been seriously craving some AMPM
17:14.50sniper[FOO]Strom_M: any ideas?
17:14.58Qwell$0.08 cheeseburgers++
17:14.59Strom_Msniper[FOO]: use circuits with actual supervision passthrough
17:15.11*** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org)
17:15.50luke-jrQwell: the 'normal' group
17:16.06Qwellwhat 'normal' group?  it's p2p...
17:16.23Strom_Mok, time to check out of the hotel
17:16.25sniper[FOO]Strom_M: could you explain in a greater detail?
17:16.25Strom_Mlatarz
17:16.38luke-jrQwell: yeah, it's peered p2p
17:16.48Qwellpeered p2p?
17:16.51*** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt)
17:17.40Strom_Msniper[FOO]:
17:17.41Strom_M~101
17:17.41jbotextra, extra, read all about it, 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony.  You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf
17:17.41luke-jryou need to establish arrangements to access it
17:17.58Qwellluke-jr: no you don't.  set it up internally...  hence the p2p
17:18.22luke-jrinterally is useless
17:18.28luke-jrthere's no global network?
17:18.43sniper[FOO]I'm aware of the terms you were speaking about, I know what circuit-switched telephony is but I need a solution implemented in software
17:19.06luke-jreg, one the GPA was written for
17:19.08Qwellluke-jr: there are probably many "global" networks
17:20.41luke-jr...
17:20.54*** join/#asterisk Ebola (n=Ebola@host86-143-7-120.range86-143.btcentralplus.com)
17:21.16luke-jrQwell: so how about helping instead of picking on trivial terminology?
17:21.35hmmhesaysi need a new mobile phone
17:21.39hmmhesaysany recommendations?
17:22.00luke-jrhmmhesays: Neo1973?
17:22.16Qwellluke-jr: try asking a real question...  dundi is a distributed protocol...  there is no "central authority"
17:22.32sniper[FOO]hey guys, could someone explain what could the answer "use circuits with actual supervision passthrough"
17:22.45luke-jrQwell: there is a "central network"
17:22.52luke-jrwhether there is an authority or not
17:22.54QwellNo there is not.
17:22.59QwellIt's completely distributed
17:23.03apturahmmhesays so many variables like what city do you live and what are your intentions with the phone. I am going with a industrial mic phone by telus if my supervisor does not get a phone back from another employee.
17:23.08hmmhesaysi'm looking at this moto Q
17:23.14hypa7iaQwell: there can be a network with the most hosts though
17:23.14QwellI don't need to contact anybody if I want to setup dundi
17:23.32giesenaptura: I also do it so people can dial in and create conference bridges
17:23.51*** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq)
17:23.52luke-jrQwell: if you want to make a call to anyone else with Dundi, you do
17:24.05Qwellyou don't send calls over dundi
17:24.16luke-jrI know
17:24.20luke-jrI didn't say you did
17:24.25_x86_gah, i just reset the polycom (4+6+8+*), and it's still failing :(
17:24.36QwellI have dundi.  You don't need to do anything to call me.
17:24.52luke-jrexcept identify your IAX/SIP address
17:24.55luke-jrwhich uses Dundi
17:25.05luke-jrand won't work unless I can peer with someone on the same network
17:27.41Qwellsounds like what you want is enum
17:27.51*** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu)
17:34.32drakook all in the sudden my iax trunk stoped work....
17:34.41drakoit says respond to slow
17:46.00hmmhesayscan ayone recommend a good place I can rent a dedicated server?
17:46.12hmmhesaysgoogling this kind of stuff is near useless
17:47.27*** join/#asterisk bintut (n=bintut@cm47.gamma178.maxonline.com.sg)
17:47.32Qwellhmmhesays: I use meganetserve (or something), and I've been happy with them
17:47.38Qwellfile told me about them
17:48.45*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
17:49.53blitzrageQwell: speaking of DUNDi.... any idea how hard it would be to backport pbx_dundi.c from trunk to 1.4... ?
17:50.04Qwellprobably not hard
17:50.33blitzrageif you had some time... would you try for some beers at AstriCon... ?
17:50.39Qwelltake it from just before Tilghman made the change for config mods on reload, and it should be trivial :D
17:50.43*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
17:50.44Qwellblitzrage: I don't drink :p
17:50.52blitzrageQwell: hrmmm... for food? :)
17:51.10QwellI'll do it because you rock
17:51.16blitzrageQwell: you rock more so
17:51.38blitzragebecause DUNDi now has two small, but very important features for clustering
17:51.44fileblitzrage: he's going to ask you for something later!
17:51.51blitzragefile: and I'll be at his mercy to do it... :D
17:52.11Qwellwell, why not just take those two small but important patches from trunk?
17:52.21blitzragebecause my coding skills are weak
17:52.35blitzragelol
17:52.37blitzrageit's true
17:53.01fileO.o naked?
17:53.02blitzrageat AstriCon I'll be waking up with a girl next to me though.. w00t
17:53.05Qwell...no
17:53.08blitzragefile: I was FULLY clothed :)
17:53.13Qwellabove the covers :P
17:53.16fileuh huh
17:53.27Qwelland there was a Kielhofner on the floor
17:53.33blitzrageoh ya! hehehe
17:53.43blitzragebut ya... a backport of DUNDi would be pretty leet....
17:53.51blitzrageand would be useful in my presentation
17:53.57QwellI can't imagine it would be that hard
17:54.10Qwellmodule stuff hasn't changed too dramatically
17:54.11blitzrageI'd hope not... I can't imagine there have been that many core changes to that file
17:54.37Qwellabout the only thing I can think of is the stuff Tilghman did recently
17:56.11JunK-YQwell: use the force and do it :)
17:57.05JunK-Yi will focus my energy on cli filtering :)
18:00.24*** join/#asterisk strepsils (n=strepsil@bon31-2-89-80-46-186.dsl.club-internet.fr)
18:00.30tzafriranybody with a spare 2.6.22 to test ztdummy changes?
18:01.05tzafrirI hate testing kernel code on my laptop
18:01.51blitzrageQwell: back soon -- gotta run to the gym and bank
18:02.00blitzragethen I'll be working on my presentation, so I'll be around
18:02.14blitzrageif you happen to get a chance to look into DUNDi, that'd kick ass :D
18:03.09hmmhesaysthis grand central page looks suspiciously like it was made with drupal
18:06.28*** join/#asterisk Shadowfire_ (n=jeff@rrcs-67-79-144-150.se.biz.rr.com)
18:06.47Shadowfire_Anyone here familar with HPEC?
18:07.37hypa7iahmmhesays: it sure does
18:08.53russellbblitzrage: backport of dundi to what?
18:09.09Qwell1.4
18:09.14Qwellstuff that's in trunk
18:09.17russellbohhhh
18:09.18Shadowfire_I have set it up on my box... for 2 channels... and it looks like it's all compiled correctly with zaptel devices, and it shows it registered... but I have made some calls in and now its not picking up
18:09.26Qwellprobably trivial, eh?
18:09.29russellbblitzrage: if you email me, i'll backport those changes and put them in my svncommunity repo
18:09.33Shadowfire_it's 1.4 on a 64bit
18:09.38russellbblitzrage: an email will remind me and make sure i do it
18:09.41russellb... back to mario kart!
18:10.20Shadowfire_guys... any chance I could get your help?
18:16.10*** part/#asterisk strepsils (n=strepsil@bon31-2-89-80-46-186.dsl.club-internet.fr)
18:18.07luke-jrwhat does the 'nopartial' flag do on DUNDi mappings?
18:18.33*** join/#asterisk ommm (i=grabli@ommm.ru)
18:22.35Shadowfire_it's amazing that there is 275 people on here and no one to help...
18:23.16JunK-Yon what Shadowfire_ asking a specific question would be a start.
18:23.23Shadowfire_I did...
18:23.42luke-jr...
18:23.45luke-jrJunK-Y: I did too
18:23.51Shadowfire_I guess it depends on the way you want me to ask it...
18:23.58Shadowfire_ok...
18:24.03Shadowfire_let me see
18:24.36Shadowfire_Has anyone had HPEC kill there SIP connects
18:24.37Shadowfire_?
18:24.46sniper[FOO]did you post the relevant configuration file entries?
18:24.57*** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq)
18:25.07luke-jrblitzrage: what's the difference between 1.4 DUNDi and pbx_dundi?
18:25.30sniper[FOO]it's the first thing people start asking when you start asking :)
18:26.01Shadowfire_I guess I am different... I like to know what is happening before I get to far in to it...
18:26.07Shadowfire_makes more sense to me...
18:26.12Shadowfire_but that is me
18:26.22Shadowfire_so I know what i am getting into...
18:26.45luke-jr...
18:26.52sniper[FOO]in the meantime, anyone having experience with ringing tone detection?
18:28.29Shadowfire_Let me ask a more direct question... Has anyone had experience with HPEC?
18:29.14sniper[FOO]see, we're having the same problem
18:29.22sniper[FOO]noone answers ;)
18:29.29Shadowfire_lol
18:29.50sniper[FOO]btw what is hpec?
18:30.00sniper[FOO]too lazy to google for it
18:30.06Shadowfire_I am use to a more active Q & A in places like this...
18:30.22Shadowfire_High Performance Echo Cancellation...
18:30.27sniper[FOO]wow
18:30.59fileit's a G.168 tested echo cancellation implementation for zaptel
18:31.52sniper[FOO]IC
18:32.01fileeliminates echo very very nicely
18:32.01Shadowfire_It's supposedly great... and I have heard some great reviews... I have just setup 2 channels with it...
18:32.21Shadowfire_I haven't been able to test it because it has shutdown my incoming SIP connections
18:32.29luke-jrwhy does extensions.ael talk about priorities?
18:32.40luke-jrdidn't AEL get rid of those?
18:32.43filethe HPEC stuff does not touch Asterisk, there is no reason that would happen
18:32.59luke-jr(hide them, I mean)
18:33.06[TK]D-FenderShadowfire_, why do you believe HPEC is in fact responsible for your problem?
18:33.11Shadowfire_It does touch asterisk...
18:33.12*** part/#asterisk ommm (i=grabli@ommm.ru)
18:33.18fileit touches zaptel
18:33.27Shadowfire_in zapatel,conf
18:34.06Shadowfire_asterisk uses zapatel.conf ...that is one of it's config fies
18:34.25[TK]D-FenderShadowfire_, why do you believe HPEC is in fact responsible for your problem? <-- again
18:35.11luke-jrzapAtel? :)
18:35.25luke-jr[TK]D-Fender: any ideas on DUNDi?
18:35.29*** join/#asterisk mihi (n=Michi@p549BA786.dip0.t-ipconnect.de)
18:35.34Shadowfire_Why... Well... I never said it is solely responsible... more like it is in the mix.... only because I install HPEC and now SIP ext are not receiving calls...
18:35.43[TK]D-Fenderluke-jr, Yeah, Paul Hogan is great!
18:35.53luke-jr[TK]D-Fender: who?
18:36.02[TK]D-Fenderluke-jr, think on it :p
18:36.04rob0Crocodile DUNdi
18:36.05luke-jrO.o
18:36.09luke-jro
18:36.14luke-jrso on a serious note? :)
18:36.25luke-jr'nopartial' means what exactly?
18:36.55[TK]D-FenderShadowfire_, Go look at the rest of the mix.  You're shooting blind and not showing SIP debug of failed calls or anything of use.  Randomly pointing the finger like that is extremely counter-productive
18:37.50sniper[FOO]so I ask again :)
18:38.10sniper[FOO]I wanna detect UK ringtones in a g.711 media stream
18:38.27sniper[FOO]someone pointed me to use freeswitch
18:38.35Shadowfire_I am not pointing fingers at anything right now... I am asking for help dude?  I am not sure why it is a problem to help... but I am starting to think this is a rather unfriendly place...
18:39.12Shadowfire_or at least some people here...
18:39.35sniper[FOO]unfortunately, my former colleague who asked for help, insists on *, because it'll perform some other functions
18:39.43[TK]D-FenderShadowfire_, Go show us something that will actually help pinpoint and correct your problem.
18:39.48sniper[FOO]pastebin'd code to come:
18:40.08sniper[FOO]http://pastebin.com/d512d2bcc
18:40.20luke-jrsniper[FOO]: what else would you use if not *?
18:40.24sniper[FOO]my extensions.conf and the session output
18:40.31Shadowfire_So the blind is supose to lead the guru's?????  OK.... never mind... I work it out... thanks anyway
18:40.40sniper[FOO]freeswitch I've been told does this OOB
18:40.52[TK]D-FenderShadowfire_, You have not shown us ANYTHING.  Do you think we're psychic?
18:41.24Shadowfire_I realize I haven't shown you anything... but what have you asked for...
18:41.35Shadowfire_don't assume I know what you need,,,
18:41.54luke-jrsniper[FOO]: extensions.conf is obsolete :)
18:41.55[TK]D-Fender<[TK]D-Fender> Shadowfire_, Go look at the rest of the mix.  You're shooting blind and not showing SIP debug of failed calls or anything of use.  Randomly pointing the finger like that is extremely counter-productive <---- I already asked you for SIP DEBUG FOR YOUR FAILED CALL
18:41.56sniper[FOO]Shadowfire_: configuration files, relevant data, output
18:41.56Shadowfire_A highway is a two way road... like communication
18:42.05sniper[FOO]it's for 1.2
18:42.19[TK]D-FenderShadowfire_, pay attention.
18:42.25Shadowfire_nice
18:42.56sniper[FOO]I ported app_nv_linedetect.c to 1.2 'cause I didn't want to dig deep into restructuring the code and so on
18:43.31[TK]D-Fendersniper[FOO], the CNG warnings might be throwing things off...
18:44.05sniper[FOO]erm... what is CNG?
18:44.19sniper[FOO]comfort noise
18:44.22sniper[FOO]IC
18:44.58sniper[FOO]does this apply if I really hear the audio and it's decent quality?
18:45.07*** join/#asterisk n0n4m3 (n=NoName@noname.rula.net)
18:45.12n0n4m3evening!
18:45.19[TK]D-Fendersniper[FOO], if you're getting CNG warnings, that means some audio is being masked....
18:45.24sniper[FOO]IC
18:45.34sniper[FOO]I'll go ahead and disable CNG
18:45.49[TK]D-Fendersniper[FOO], perhaps the audio you are loking for is too weak to register and gets filtered at source
18:49.10sniper[FOO]didn't find the respective option in the SJPhone client :S
18:49.39*** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ma.comcast.net)
18:49.47xpotanyone able to help with a non-root asterisk implementation? errors are here: http://pastebin.com/m6a8f2b9f
18:50.07sniper[FOO]* is not meant to be run as root
18:50.10sniper[FOO]IIRC
18:50.49xpotaccording to my errors it doesn't like the fact that root is not running it, that is why I need assitance
18:50.51sniper[FOO]in fact, IIRC, the 1.4 releases won't let you run it as root, and it's hardcoded in the binary
18:51.51sniper[FOO]or am I wrong?
18:52.04sniper[FOO]nope
18:52.15sniper[FOO]you gotta run the init scripts as root
18:52.24xpotare you able to explain my error messages then?
18:52.34xpotI am not sure what is going on
18:52.39sniper[FOO]look
18:52.58sniper[FOO]there's a monolithic binary, called /usr/sbin/asterisk on most systems
18:53.09xpotyes and safe_asterisk
18:53.17sniper[FOO]that expects some parameters
18:53.35sniper[FOO]and then there's an initialization script provided by ubuntu
18:53.53sniper[FOO]which passes the right parameters to this binary
18:54.20xpotperhaps I have the wrong script? It looks like I am using debian script
18:54.21[TK]D-Fenderxpot, don't start it with safe_asterisk.  Start it as "asterisk -gvvvvc" under the user its supposed to run as so you can see what its crashing on.
18:54.22sniper[FOO]the init script must be run as root as it needs uid=0 access to some files
18:54.27sniper[FOO]nope.
18:54.40sniper[FOO]just type sudo /etc/init.d/asterisk start
18:54.54xpotsniper: I will try that
18:55.04[TK]D-Fenderxpot, One classic cause of an error like that is not having zaptel loaded before * when channels need to be initialized
18:55.20[TK]D-Fenderxpot, run is STRAIGHT, not through an daemon init script.
18:55.43rob0Yes, the init script is preventing you from debugging.
18:56.22sniper[FOO]you can pass any parameter to * via the standard Debian defaults interface
18:56.28rob0If zaptel is in use, the udev rules need to have the right ownership of the devices.
18:56.57xpotI did not instal zaptel since I do not have any zaptel devices, is zaptel required regardless?
18:57.12sniper[FOO]absolutely not
18:57.26rob0possibly wanted for dummy, but no
18:58.43[TK]D-Fenderxpot, Still if you want to see whats going on FORGET the scrips and run it directly
18:58.58xpotwhen I run as [TK]D-Fender suggested I get the CLI and seems to run fine
18:59.18rob0So then fix or replace the init script.
18:59.30[TK]D-Fenderxpot, as the user * runs as?
19:00.06sniper[FOO][TK]D-Fender, I can't find comfort noise generation anywhere in the SJPhone client :S
19:00.08xpotI am not sure what needs to be fixed, based off of my error I am assuming that my ast user does not have the appropriate rights to an unknown location, else why the "Oops, Im not root" error?
19:00.09rob0[TK]D-Fender: is it worth the trouble to set up * for running non-root, IYO?
19:00.53[TK]D-Fenderrob0, I never have and I'm not the best qualified to answer.  Its a question of "security"
19:00.55xpot[TK]D-Fender: ast
19:00.55sniper[FOO]xpot: your error came because you called an init script without root privs
19:01.30xpotI performed sudo /etc/init.d/asterisk start --> same results as in pastebin
19:01.56*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
19:02.22sniper[FOO]btw, I bet the debian init script gives you the chance to run * as any user
19:03.42sniper[FOO]xpot: did you change the root password or left the root account unused?
19:05.41xpotleft root unused
19:06.14xpothere is my init.d script: http://pastebin.com/m1cafcbad
19:07.19*** join/#asterisk ectospasm (n=ectospas@c-68-62-214-17.hsd1.al.comcast.net)
19:10.11sniper[FOO]xpot: please confirm that 'sudo /etc/init.d/asterisk start' didn't work
19:10.44xpotsure
19:11.59ectospasmstandard Asterisk installs run as root, IIRC
19:12.38ectospasmif not, you'd have to make sure a bunch of directories in /var and /usr are writable as the asterisk user
19:12.48ectospasmamong other things
19:12.58xpotsudo fail: http://pastebin.com/m3e20127a
19:13.00rob0(or set the right values in asterisk.conf)
19:13.01*** join/#asterisk Strom_M (n=strom@206.166.206.34)
19:13.20sniper[FOO]just the control socket, pid file and /var/log/asterisk
19:14.14sniper[FOO]OK then
19:14.43sniper[FOO]please post the output of 'strace sudo /etc/init.d/asterisk start'
19:15.12xpotok coming up, rob0: asterisk.conf http://pastebin.com/m46c6bf60
19:15.17sniper[FOO]look for your password in your post
19:15.41xpotok
19:15.56sniper[FOO]and remove it if necessary
19:16.51sniper[FOO]gonna be lengthy :)
19:16.55*** join/#asterisk Cresl1n (i=matt@nat/digium/x-fdaa602b6240f97a)
19:16.55*** mode/#asterisk [+o Cresl1n] by ChanServ
19:17.16*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)
19:18.10Strom_Mhi Cresl1n
19:18.19Cresl1nStrom!
19:18.37Strom_Mhow ya doin?
19:18.41Cresl1ngood
19:18.44Cresl1nkinda quiet in here
19:18.45Strom_Myay
19:18.49Strom_Mare you at the office?
19:18.55Cresl1nyeppers
19:19.01Strom_Mah cool cool
19:19.05Strom_Mi'm at HSV
19:19.13Cresl1noh wow
19:19.22Cresl1nwhat brings you to this part of the country?
19:19.25fileyou should visit Digigraph and tackle Cresl1n
19:19.30Cresl1nheh
19:19.37Strom_Mfour passengers, myself, and the Gevalia Coffee attendant are sitting around half-paying attention to CNN
19:19.54Cresl1noh, you're for real in HSV then
19:19.55Strom_MCresl1n: I taught another course this week
19:20.00Cresl1nah....
19:20.03xpotsniper[FOO]: here it is http://pastebin.com/m8c1cc4c
19:22.09sniper[FOO]xpot: do you insist on disabling the root account?
19:22.34sniper[FOO]I realized that tracing won't really work like this
19:23.04xpotsniper: not entirely, but I would like to if possible
19:23.44xpot* starts fine when I perform asterisk -gvvvvc
19:24.11ectospasmStrom_M:  I enjoyed what little bit of your class I sat in yesterday
19:24.17Strom_Mectospasm: :D
19:24.21xpotI just wanted it to load automatically on boot, and I now assume the problem is in the init.d script somewhere...
19:24.26sniper[FOO]nope
19:24.40sniper[FOO]it's gonna start on bootup
19:25.16xpotwithout error?
19:25.54Maxxedwhats the typical unlimited long distance pri cost?
19:26.03Maxxed2-300 a month?
19:26.04sniper[FOO]could you check for a symlink in /etc/rc2.d, it's called Snnasterisk
19:26.14xpotsure
19:26.26Strom_Mis there even such a thing as an "unlimited long distance PRI"?
19:26.38docelmoSay does anyone know if there is a way to force an AGI script to keep running until its over even if the channel is hung up?
19:26.39Maxxedwell, iv used time warner before
19:26.53Maxxedand i could call anywere in the usa
19:26.55Maxxedno long dist
19:27.54xpotsniper: S20asterisk
19:27.58sniper[FOO]great
19:28.02Maxxedsprint i know offers a unlimited usa long dist pri
19:28.06Maxxedbut whats the going rate?
19:28.20sniper[FOO]xpot: type 'sudo init 2'
19:28.21xpotsymlinks to ../init.d/asterisk
19:28.25sniper[FOO]nice
19:28.26xpotok
19:29.02Strom_MMaxxed: well I can guarantee it probably is going to be a touch pricey
19:29.07xpotsniper: done
19:29.12Maxxed300 a month or more?
19:29.18Strom_Mand there are probably limitations on what they consider "unlimited"
19:29.28Strom_Moh, i'd expect at least $700 if not $1000-$1200
19:29.51Maxxedgotcha
19:30.03Maxxedthats all i wanted to know, thx :)
19:30.03voipnet-techhas anyone ever built a system that boots linux from a USB flash card reader?
19:30.06sniper[FOO]xpot: what does 'netstat -upln | grep 5060' say?
19:30.23voipnet-techi've never seen boot options to look for usb drives to boot
19:30.48sniper[FOO]voipnet-tech: yes, recent mobos add this option
19:30.58apturaWatching one of Marks keynote speaches on you tube but do not know what city this was in.
19:31.34xpotsniper: udp 0 0 0.0.0.0:5060 0.0.0.0:*
19:31.41voipnet-techsniper[FOO], i'm looking at this mobo: http://www.newegg.com/Product/Product.asp?item=N82E16813128043
19:32.44sniper[FOO]xpot: great, you have * up and running
19:32.53apturaokay its in Dallas :)
19:33.19*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)
19:33.29xpotexcellent
19:33.46xpotsniper & all who assisted: thank you for your help
19:34.16*** join/#asterisk jmacz (n=jmacz@190.24.98.133)
19:34.27sniper[FOO]xpot: you can tweak the settings in /etc/init.d/asterisk and maybe /etc/default/asterisk
19:37.00xpotsniper: ok I will try it out
19:43.03thansen|laptopwhat permissions does asterisk need on /dev/zap/* in order to function properly with MeetMe?
19:46.06*** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53)
19:53.53sniper[FOO]hi all
19:54.12sniper[FOO]can't get rid of this 'process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP:' stuff
19:54.50Strom_Msniper[FOO]: yes you can
19:55.00sniper[FOO]I don't seem to be able to switch off the relevant option in any of the softphones I'm using
19:55.01Strom_Mturn off CNG on your endpoints :)
19:55.14sniper[FOO]sure, I did, and the message is still here
19:55.28Strom_Mare you calling softphone-to-softphone?
19:55.37sniper[FOO]nope, via a GSM gateway :)
19:55.50sniper[FOO]started a request to switch it off
19:55.51Strom_Mhave you considered that you need to turn CNG off on the gateway too?
19:55.59sniper[FOO]on the g/w too
19:57.09sniper[FOO]do you think it's CNG what makes any sort of tone/noise detection apps malfunctioning?
19:57.22sniper[FOO]not functioning, actually
19:57.46Strom_Msniper[FOO]: tone detection works even worse on GSM calls
19:57.54sniper[FOO]greeeeat
19:58.01Strom_Myou should be getting answer supervision information on those calls
19:58.57sniper[FOO]I only get an immediate OK after my invite and I should be able to determine if the call's connected or what
19:59.21Strom_Mdo you understand what answer supervision is?
20:00.44sniper[FOO]sure, but that's what I'm not getting
20:00.57Strom_Mwhat path is your call taking from you to the PSTN?
20:01.33sniper[FOO]just me and the g/w
20:01.50Strom_Mand the gateway is connected to the PSTN...how?
20:02.09sniper[FOO]it's a GSM gateway
20:02.24sniper[FOO]has lots of cellular engines in it
20:02.50Strom_Mhow does the GSM gateway talk to asterisk?
20:02.55sniper[FOO]and controls them on their serial interface
20:02.58Strom_Mis it your gateway?
20:02.58sniper[FOO]SIP
20:03.18Strom_Mtime to board my flight
20:03.20Strom_Mlater
20:03.30sniper[FOO]look, there's a box with an RJ-45 connector on it, and lots of blades, 8 engines in them
20:07.00dmcnhah! my problems with freecall and similar is solved - i commented out srvlookup=yes in my sip.conf and now it works
20:07.16luke-jrHow can I determine if something is a valid voicemail box?
20:08.23sniper[FOO]executing the system app and looking for .gsm files?
20:08.55sniper[FOO]luke-jr: from inside * or from the system?
20:11.30blitzragerussellb: hope you're enjoying mario kart! Just sent you that email FYI
20:12.51n0n4m3i've got a little ol' question... i have a couple of belco bcip-300 phones and an asterisk server. i've disabled all the codecs except alaw and ulaw (both on ip phones and asterisk) and i can't seem to talk between them :S
20:13.50dukihello,
20:14.00n0n4m3in case i call myself on my celular over voipdiscount, that's registered in asterisk as a peer, i can hear everything in belco but nothing in my gsm
20:14.28dukiis it possible to encrypt the passwords in *.conf?
20:15.19blitzrageduki: some of them allow you to use md5secret so you can put a hash there instead
20:15.45blitzragehttp://www.voip-info.org/wiki/index.php?page=Asterisk+sip+md5secret
20:16.42dukiblitzrage: thank you very much, it is really what I need .
20:17.48*** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com)
20:20.16n0n4m3where else should i look
20:29.04luke-jrsniper[FOO]: from dialplan
20:32.14*** join/#asterisk ManxPower (n=manxpowe@61.sub-75-201-47.myvzw.com)
20:35.42*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)
20:35.55*** join/#asterisk SA007 (n=sa007@ip565e0006.direct-adsl.nl)
20:36.15SA007anyone here any experience with 3com hardphones?
20:36.32*** join/#asterisk CrazyTux[m] (n=CrazyTux@h460ed1d7.area3.spcsdns.net)
20:38.03*** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg)
20:45.09*** part/#asterisk nickzxcv (i=nick@schmalenberger.us)
20:46.46sniper[FOO]luke-jr: you may get adequate results by looking for .gsm files in the appropriate directory
20:48.08sniper[FOO]e.g. 'ls -l *.gsm | wc -l '
20:48.59sniper[FOO]if the reported number is greater than 0, you have a voicemail dir
20:49.17sniper[FOO](or at least a directory with at least one .gsm file :) )
20:49.45sniper[FOO]^^ and you can execute that via the System() app
20:53.47*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)
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21:14.24ManxPowerSA007: what specific model?
21:14.49SA0073com 3101
21:16.06ManxPowerResults 1 - 10 of about 502 English pages for 3com 3101 Asterisk AND "too lazy to search google"
21:16.41SA007no, i'me not, been searching all evening
21:16.55ManxPoweror even better: Results 1 - 10 of 26 English pages from lists.digium.com for 3com 3101 Asterisk.  (0.35 seconds) 
21:17.26ManxPowerSA007: so you know that they do not run real SIP and won't work with Asterisk.
21:18.17SA007i know, but maybe there is some dort of VCX to SIP bridge or something
21:18.31ManxPowerThere isn't.
21:20.06SA007:( so i;ve got a useless hardphone
21:23.16*** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net)
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21:37.18CCFL_Man2wonder what kind of ringer boxes would be used with the WE 202 imperal models
21:37.28CCFL_Man2anyone know?
21:37.36CCFL_Man2i'd guess the 685A
21:40.20CCFL_Man2no western electric nerds here?
21:40.22*** join/#asterisk shido6 (n=shido6@74-130-56-203.dhcp.insightbb.com)
21:40.23CCFL_Man2:P
21:41.00ManxPowerCCFL_Man2: the only one I know of is Strom
21:42.53*** join/#asterisk klictel (n=klictel@modemcable159.7-200-24.mc.videotron.ca)
21:43.55CCFL_Man2ahh, and he's not in yet
21:43.58CCFL_Man2no big deal
21:44.19CCFL_Man2i just want authenticity with my 50s model green 202
21:44.42CCFL_Man2or as Paul Sr. said in the caddyshack bike episode, orthentic
21:45.15*** join/#asterisk Skaag (n=skaag@212.199.202.142.static.012.net.il)
21:56.41shido6hrmm
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22:33.13docelmoDoes anyone know where I can find a spreadsheet that lists all of the area codes by state?
22:34.23dijungalgoogle.com
22:35.07Corydon76-dighttp://www.localcallingguide.com/
22:35.12rob0nanpa maybe
22:35.17dijungalprolly nanpa
22:35.24docelmoCorydon76-dig do they are a CSV or something I can download?
22:35.37Corydon76-digThat is what you really want... it's a list of NPA-NXX's that are local to you
22:36.20Corydon76-digplus lists of area codes.  No, it's not on a spreadsheet
22:36.27docelmook thanks!
22:37.57Corydon76-digIn fact, you're probably not going to find any place that's going to give you that information in a database-importable format without paying for it
22:37.58dijungalis there anyways to compare the area code to a list... for example XXX in (721,467,123,456) ?
22:38.34Corydon76-digdijungal: yes, you can use func_odbc to lookup the number in a database
22:39.24dijungalfrom asterisk dialplan?
22:39.35Corydon76-digYes, that's why func_odbc exists
22:40.24sniper[FOO]hello again :)
22:40.36Corydon76-digIt's so you can do simple database queries
22:40.54dijungalk thanks
22:41.07dijungalvery nice :)
22:41.13sniper[FOO]still noone to help me out with the ringtone detection issue?
22:41.29Corydon76-digdijungal: thank you.  I think so, too
22:41.46dan__thrm
22:41.57dan__tWonder if they make a DOCSIS WIC module for Cisco products...
22:42.18*** part/#asterisk sifusam (n=sifusam@nat-vlan0200.sat4.rackspace.com)
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22:47.21*** join/#asterisk Yourname`` (n=IM@unaffiliated/yourname/x-837320)
22:47.57Yourname``Hello. Is there something in Asterisk that will let me dial a number, and then whatever I see will be heard on speaker on all other agents phones?
22:49.06shido6whatever I see?
22:49.14shido6what do you mean by whatever you see?
22:49.30Yourname``lol, sorry whatever I "say".
22:49.50shido6do you want them to respond?
22:49.56shido6and have you hear their response?
22:50.00shido6like a conference room?
22:50.01Yourname``Maybe, maybe not.
22:50.24shido6so u want to call them, dump them into a conf room and have other ppl join and
22:50.31shido6say or just listen in
22:51.15Yourname``That sounds complicated. How about just a scenario where I dial *1, say "All hands on deck!", and hangup. And "all hands on deck" can be heard on all phones.
22:54.15ManxPowerThere is an application called Page in 1.2.x+
22:55.03Yourname``Ah, is that what it is?
22:55.19Yourname``Basically like an announcement service for all phones connected to the pbx.
22:56.01*** join/#asterisk Skaag (n=skaag@87.70.7.186)
23:08.34blitzrageYourname``: yes, but the phone also has to support the Alert-Info string. Polycom's work for sure.
23:08.46blitzragebut you also have to configure the phone to listen for it
23:09.22*** join/#asterisk Zyl0ne (n=zylone@ip-112-136.atvci.net)
23:10.00Zyl0neanyone know what the deal is with the speech codec file not available for download on intel's site for g729??
23:11.02mogwe dont talk about the intel g729 codec in this channel
23:11.04Yourname``blitzrage : Someone made it work with Linksys SPA941s before. I think Trixbox was used. Wondering how we can do it on vanilla Asterisk.
23:11.05*** join/#asterisk DrukenLPY (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com)
23:11.09Zyl0newhy is that
23:11.14DrukenLPYevening everyone....
23:11.21mogits illegal in the majority of the world
23:11.26Zyl0neoh
23:11.30Zyl0newell poo
23:11.37DrukenLPYanyone know if someone has done a CRM with asterisk yet?
23:11.39Zyl0neI am in america! heh
23:12.10DrukenLPYmog: what's illegal in most of the world? :)
23:12.17Yourname``DRUGS!
23:12.48mogthe intel g729 codec
23:12.56DrukenLPYoh
23:13.01mogas you arent paying proper royalities
23:13.18mogand dont have permission by them the patent controllers to do so
23:13.33Zyl0neumm
23:13.37Zyl0neI am not using it for business use
23:13.39Zyl0neso it is fair
23:13.46ManxPowerZyl0ne: cite your source.
23:13.56Zyl0neI have read that all over the net
23:14.00Zyl0nemaybe I am wrong
23:14.08ManxPowerSo it's OK for me to use MS Office without paying for it -- since it's just for personal use.
23:14.09Zyl0neor they changed it
23:14.12DrukenLPYlaws are ment to be broken... and only enforceable when caught breaking them
23:14.15Zyl0nedude
23:14.22Zyl0neI am just going off what I have read on the net
23:14.28mogno Zyl0ne
23:14.30Zyl0neI can care less about paying 10 bucks a license
23:14.32mogthats not the way it works
23:14.35Zyl0nedang
23:15.09Zyl0nehttp://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
23:15.13Zyl0nethere..
23:15.19mogim aware of it
23:15.28Zyl0newell.. that is my only knowledge of it
23:15.30Zyl0neexcuse me
23:15.37mogno problem
23:15.47Zyl0neit had a link directly to intel's site
23:15.51Zyl0neand said to register on intel's site
23:15.57Zyl0neapparently it has changed
23:16.38Zyl0neas a matter of fact I have already bought 1 license through digium
23:16.48mogi could be wrong, because i often am, but it is my understanding that it is illegal to use
23:16.50Zyl0neand guess I will have to buy some more
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23:17.04Zyl0neok, maybe so =)
23:17.27mogsee legal stuff
23:17.30mogat the bottom
23:17.47Zyl0neya
23:17.52mogpatent law says you have to pay sipro
23:18.06mogand sipro wont give you a license unless you want to buy million or so worth of license
23:18.17Zyl0neguess people are twisting the truth
23:18.29luke-jrsniper[FOO]: ...
23:18.33mogwell most people dont care about violating ip
23:18.36Zyl0neno biggie, I will go grab the credit card and order a few more =)
23:18.41Zyl0neyeah I hear ya
23:18.59Zyl0neheck since I am here... I got a question
23:19.16Zyl0nethe disallow=all and then allow=g729, etc
23:19.24Zyl0nedoes it matter what order they are in, in the conf file?
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23:19.31mogyes
23:19.38mogcall disallow=all then allow
23:19.51Zyl0neso whatever allow is first has priority
23:20.21mogya
23:20.32Zyl0necool
23:20.40Zyl0newell
23:21.09Zyl0neI setup g729, and then a few others.. and it would go with ulaw, but if I only allowed g729, it would use it
23:21.13Zyl0nekinda weird
23:21.29Zyl0neand I had g729 as the first allow
23:21.53mogwell hte other end has some say in it to , whats at the other end?
23:22.27Zyl0neg729 and ulaw
23:22.47Zyl0neand I think alaw
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23:59.22Zyl0nethere we go.. I bought a few more
23:59.26Zyl0neworking great

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