irclog2html for #asterisk on 20051024

00:01.12DrukenTUplink: iaxtel.com
00:01.33TUplinkwhen you do a iax2 show registery
00:02.02TUplink69.73.19.178:4569     xxxx  <Unregistered>             60  Timeout
00:02.14TUplinkthe 69.73.19.178
00:02.26TUplinkis the ip the same
00:02.31DrukenHost                  Username    Perceived             Refresh  State
00:02.32Druken65.39.205.121:4569    632421      <Unregistered>             60  Request Sent
00:02.47TUplinkthats fwd
00:02.59TUplinkno its not
00:03.00*** join/#asterisk |Vulture| (n=V@c-69-180-67-53.hsd1.fl.comcast.net)
00:03.51tzangerReD-MaN: well you would have entered it into the dialplan
00:04.00infinity1tzanger: i'm trying to install your zaptel packages... having a little hickup.
00:04.04infinity1<PROTECTED>
00:04.18tzanger... my zaptel packages??
00:04.39JASON-0Is asterisk supposed to bring up the D-Channel on my PRI ?
00:04.48tzangerJASON-0: if you tell it to, yes
00:05.01infinity1tzanger: the ones from rapid.dotsrc.org
00:05.07tzanger... ??
00:05.16JASON-0tzanger: Zaptel is installed, but the D-Channel isn't coming up. How do I configure Asterisk to bring it up?
00:05.37tzangerJASON-0: have you told zapata.conf it's on a PRI
00:05.46JASON-0yes
00:06.04JASON-0maybe I did it wrong though :P  But I think i did it right
00:06.11JASON-0switchtype = 5ess
00:06.11JASON-0signalling = pri_cpe
00:06.11JASON-0group = 0
00:06.11JASON-0channel => 1-23
00:06.17infinity1tzanger: oh ..:) ..tzafrir's ..not yours.
00:06.33tzangerJASON-0: looks good.  zaptel.conf has bchan=1-23, dchan=24 ?
00:06.39JASON-0tzanger: yes
00:06.47*** join/#asterisk e-Hernick (n=ncc@modemcable120.39-131-66.mc.videotron.ca)
00:06.50JASON-0span=1,0,0,esf,b8zs
00:06.50JASON-0bchan=1-23
00:06.50JASON-0dchan=24
00:06.51tzangeris the link up?
00:06.57JASON-0the link is up yes
00:06.58tzangeryou will want that 1,1,0 too
00:07.01JASON-0but a d alarm
00:07.10tzangerthere's no such thing as D alarm
00:07.29TUplinkmine is trying the same ip that yours is working on
00:07.47JASON-0tzanger: There is a DChannel Alarm
00:07.58tzangerJASON-0: there is no such thing as a D channel alarm
00:08.00JASON-0On my Telco switch
00:08.04DrukenTUplink: can you dial out iaxtel ? or is that the problem?
00:08.15TUplinkboth dont work
00:08.21tzangerJASON-0: you have a green light on the card?
00:08.23TUplinki can dialout fwd out
00:08.25JASON-0tzanger: the telco side is showing the
00:08.32JASON-0tzanger: yes, green light on card
00:08.34Drukentry to call 700-725-9124 over iaxtel
00:08.41TUplinkit dont register
00:08.46tzangerJASON-0: plug a loopback in to the smartjack, do they see it looped?
00:08.49TUplinkgive me a min
00:09.04Drukendon't need to register to dial numbers, need to register to receive calls
00:09.34JASON-0tzanger: its a sangoma card.. where is the smartjack?
00:09.45*** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de)
00:09.56tzangerJASON-0: the smartjack is on the wall, it's what the telco puts there for you to plug in to
00:11.41TUplinkDrunken its trying to call
00:12.05*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
00:12.31TUplink<PROTECTED>
00:12.31TUplink<PROTECTED>
00:12.31TUplinkOct 23 20:11:55 WARNING[527]: chan_iax2.c:1477 attempt_transmit: Max retries exceeded to host 69.73.19.178 on IAX2/Iaxtel/2 (type = 6, subclass = 1, ts=3, seqno=0)
00:12.31TUplink<PROTECTED>
00:12.31TUplink<PROTECTED>
00:12.36JASON-0tzanger: I have not tried that but the T1 works with other T1 equipment. When I connect it to asterisk the link comes up but the d channel does not come up.
00:12.56tzangerJASON-0: that's odd.  I'd call Sangoma support in the a.m.
00:13.03tzangerthey should just come up
00:13.04tzangermine did
00:13.07tzangermy digium cards do too
00:13.08DrukenTUplink: i'm begining to think iaxtel is foobared
00:13.12tzangeroh wait
00:13.17tzangeryou said switchtype's 5ess
00:13.20tzangerwhat is it emulating
00:13.20TUplinkbut explain FWD to
00:13.28tzangerVERY RARELY is a 5ESS actually signaling using 5ESS
00:13.29TUplinkFWD dose the same thing
00:13.39tzanger99% of the time it's dms100 or ni2 or something
00:13.48tzangerask them what their switch is emulating, not what the switch is
00:13.51DrukenTUplink: let me get my FWD number...
00:13.55JASON-0ok I'll check that
00:13.57tzangerwhat is their switch signaling
00:14.04JASON-0give me a few mins
00:14.34TUplinkFWD should be send out 1800 calls use to
00:14.36DrukenTUplink: 632421
00:15.15*** join/#asterisk __Elf (n=irc@rivendell.glassfish.net)
00:15.23tzangerdms100's kind of old but still used a lot
00:15.24TUplinkDrunkin just ringing
00:15.33tzangerBell Canada typically uses ni2 (national)
00:15.58DrukenTUplink: it's not hittin my server then
00:16.01tzangeras does telus, group and UUnet (but that's a LONG time ago)
00:16.02JASON-0tzanger: to reload zapata.conf what do I do? Just to make sure it's reloading
00:16.12tzangerrestart asterisk
00:16.19DrukenUUnet... haven't seen that in a while...
00:16.21tzangerswitchtype is in asterisk not zaptel so you don't need to reload the kenrel module or execute zapate.conf
00:16.25ReD-MaNtzanger: is this what you were talking about?
00:16.25tzangerDruken: yeah no shit
00:16.27ReD-MaN[outrt-001-9_outside]
00:16.27ReD-MaNinclude => outrt-001-9_outside-custom
00:16.27ReD-MaNexten => _9.,1,Macro(dialout-trunk,1,${EXTEN:1},)
00:16.27ReD-MaNexten => _9.,2,Macro(outisbusy); No available circuits
00:16.37tzangerour first ever T1 was a 40km loop length from uunet
00:16.37JASON-0zapata.conf is asterisk isnt it ?
00:16.40tzangercost us a fucking fortune
00:16.43tzangerJASON-0: yes.
00:16.49JASON-0just reload asteris ?
00:16.54tzangerReD-MaN: ok that's the macro that dials it, now go into that macro, it will eventually get to a Dial() command
00:17.01tzangerJASON-0: restart, not reload
00:17.21JASON-0ok
00:17.34TUplinkDruken why not?
00:17.53TUplinkdo you have guest for you IAX setup
00:18.05TUplinkso that i could try to IAX you
00:18.07DrukenTUplink: yes sir
00:18.29TUplinkill add a regester line in my IAX
00:18.34TUplinkand c if that registers
00:18.58Drukenwell, you'd just dial 7057259124@fishy.abss.ca
00:19.18filesounds a little fishy
00:19.31TUplinkbut that dont test te reg
00:19.47Drukenha ha ha file... hehe
00:20.21DrukenTUplink: well, no.. you wouldn't register to a guest... a guest is for stray incoming calls
00:20.33TUplinkok....
00:20.44TUplinkwell.... i dont know what to do?
00:20.57Drukensomeone just called....
00:20.59TUplinkim going to creat another FWD account and c if that dose any thing
00:21.15Drukenwas that you file?? :)
00:21.19fileno
00:21.22IronHelixfwd over iax has been having trouble lately
00:21.26fileI'm on my cell calling my ISP
00:21.33e-Hernickhey
00:21.35ReD-MaNtzanger: exten => s,14,Dial(${OUT_${ARG1}}/${OUTNUM})  ; Regular Trunk Dial
00:21.35e-Hernickwhat's up with AEL
00:21.49tzangerReD-MaN: where are you getting this dialplan from?
00:21.50JASON-0tzanger: i changed it to national and its the same.. I'm not seeing anything about it looking at zapata.conf in the full logs.. Is there an asterisk command that will tell me if it sees it?
00:22.05tzangerthere's a lot of abstraction there and it looks like you're missing something in setting it up
00:22.27ReD-MaNdunno.. just found that in my extensions.conf file
00:22.47ReD-MaNunder the macro-dialout-trunk
00:23.24Drukenhmm.... tommy huff....
00:23.26ReD-MaNI haven't modified a lot, just using what genzaptelconf made, with some modifications in AMP
00:24.46*** join/#asterisk _DAW (n=_DAW@adsl-6-122-39.msy.bellsouth.net)
00:25.04TUplinkDruken ???
00:25.10TUplinkTommy Would be me
00:25.41ReD-MaNi also see this in the logs: Executing Dial("SIP/2600-72e8", "ZAP/g1/905741xxxx") in new stack
00:26.58DrukenTUplink: oh... ok, well then you called me...
00:27.31Drukenhey... a toronto guy
00:27.40*** join/#asterisk Chuji (i=Chuji@pcp09930052pcs.tulipgrove.tn.nash.comcast.net)
00:28.07ReD-MaNwho's a toronto guy
00:28.07*** join/#asterisk orbi (n=orbiwork@65-86-47-114.client.dsl.net)
00:28.19DrukenReD-MaN: edit your zapata.conf file and make sure your zap device belongs to group 1
00:28.40__Elfis iaxtel working properly at the moment?
00:28.41DrukenReD-MaN: well... perhaps not toronto... but close enough
00:28.55ReD-MaNsignalling=fxs_ks
00:28.55ReD-MaNrxwink=300
00:29.02ReD-MaNis that right for Canada? lol
00:29.12Drukenshould be fine
00:29.33*** join/#asterisk Inv_Arp (i=junya@adsl-144-134-75.mia.bellsouth.net)
00:29.42ReD-MaN; Span 1: WCFXO/0 "Generic Clone Board 1"
00:29.43ReD-MaNsignalling=fxs_ks
00:29.43ReD-MaN; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 1
00:29.43ReD-MaNcontext=from-pstn
00:29.43ReD-MaNgroup=0
00:29.46ReD-MaNchannel => 1
00:29.59ReD-MaNDruken: Hamilton :)
00:30.10ReD-MaNWork in Mississauga
00:30.11Inv_Arphell0 to all my fellow floridians
00:30.15DrukenReD-MaN: change that to group=1
00:30.18tzangerReD-MaN: that's all fine and working if your incoming calls work
00:30.20tzangeryou have a dialplan issue
00:30.21tzangerthat's all
00:30.39tzangeryou need to dig through and find out what the OUT_* vars are if they're set right
00:30.42Drukentzanger: nah.. he's dialing g1, but no devices in group 1
00:30.51tzangerDruken: oh?
00:30.52ReD-MaNwell incoming calls get answered, but if you try to dial an SIP extension, it says the person is busy.. goes right to vm
00:30.58tzangerDruken: I missed that part
00:31.02Druken:)
00:31.04tzangerahh group=0
00:31.05tzangerI see it now
00:31.15Drukenthat's why we all look over the information
00:32.29TUplinkDruken ou know whats wierd it dosent even tell me my pass is wrong
00:33.09ReD-MaNomg... it worked! :)
00:33.18Drukenhow about that... :)
00:33.23Chujiheh
00:33.26ReD-MaNchanged it back to ZAP/g0
00:33.31ReD-MaNand then it actually dialed out
00:33.45ReD-MaNnow, one more question.. is there a way to change after how many rings before it answers?
00:33.59TUplinkWait()
00:34.07ReD-MaNmy wife wants to keep using an analog set alongside this box..
00:34.21tzangerReD-MaN: you just Wait() before answering
00:34.44DrukenReD-MaN: uhmm.... are you going to use an ivr?
00:35.38JASON-0tzanger : I had a noload = chan_zap
00:35.39ReD-MaNwell what I was hoping to do, was not have the asterisk box answer the line right away.. all I have right now is a softphone.. don't have any ATA's yet
00:35.44tzangerJASON-0: hahahaha
00:35.48tzangerJASON-0: that'll do it
00:35.53JASON-0tzanger: removed that and everything is fine now.. thanks for the help
00:35.54JASON-0:)
00:36.06ReD-MaNso when my wife is home, she can still use a normal phone to answer the line.
00:36.11DrukenReD-MaN: get an ata as soon as possible :)
00:36.18tzangerJASON-0: I'd still verify the dchan switchtype emulation, it's unusual to actually use 5ess
00:36.29ReD-MaNDruken: I would, but they seem so expensive lol
00:36.43ReD-MaNI would only need one for my four handset cordless phone system tho
00:36.46Drukenexpensive? no more expensive than a telephone... 100 bux...
00:37.12tzangerI typically recommend the TDM11B or some variant instead of X101Ps and ATAs
00:37.39tzangerbut tha'ts just me
00:37.47Drukeni wouldn't reccomend the use of a TDM ever....
00:37.55ReD-MaNDruken: expensive when you normally don't have to pay for stuff ;) mebbe I will sell off one of my extra Cisco switches and buy a decent ATA or something
00:37.59Drukeni have a TDM and it's a peice of crap
00:38.11tzangerDruken: mine work *great*
00:38.20Drukenwhat kinda cisco switchs?
00:38.35ReD-MaN3548
00:38.49ReD-MaNor a 2950
00:38.51Drukentzanger: mine craps out all the time... only thing i use it for is my fax machine... otherwise i use ata;s
00:38.59tzangerDruken: using CVS HEAD?
00:39.05Drukenyes sir
00:39.12tzangerhmm
00:39.17tzangerI don't know what to tell ya then
00:39.23tzangermine work *great*
00:39.27tzangerand it's on an old dell P2
00:39.29Drukenjust that i got a lemon
00:39.31Druken:)
00:39.35tzangerRMA the fucker
00:39.56Drukencan't be bothered.... the ata's work just fine...
00:40.01tzangerheh
00:40.16Drukeni just won't buy any other TDM's
00:40.53ReD-MaNhmm.. dialin gets answered, accepts the extension, but goes right away to saying the person at extension 2600 is unavailable.. weird
00:42.12*** join/#asterisk digime (n=drooth@ip68-111-235-172.sd.sd.cox.net)
00:42.39DrukenReD-MaN: is that a 48 port switch ?
00:42.49ReD-MaNthe 3548 yup
00:42.53ReD-MaN2950 is a 24 port
00:43.30Drukennice... :)
00:43.42Drukeni had a small cisco fry on my today... :(
00:43.56ReD-MaNswitch?
00:44.42ReD-MaNok this is weird.. I can call from my X-Lite softphone, call out no problem.. try calling into it, and says it is busy
00:45.01ReD-MaNalmost like * doesn't know I am connected
00:45.06moralew
00:45.34moralehere can i find a decent iax/sip provider in alberta? areacode 403
00:45.39Drukenyeah a switch... just a lil 5 port... but it still added to my day of my entire network crashing
00:46.02*** join/#asterisk shakuhashi (n=shaku@200.163.5.67)
00:46.23ReD-MaNDruken: I hear ya. I now have 7 cisco's.. always gotta have a spare or two ;)
00:48.07Drukenwell, i had 3 servers crash... the hydro dropped out last night at the office, and the switch kicked the bucket
00:48.21Drukenkicker is... one of the servers that crashed was 4 provinces over in alberta
00:48.37ReD-MaNouch
00:49.07tzangerfour provinces over? you're an easterner then
00:49.13tzangerNB?
00:49.18Drukenno... i just can't count :)
00:49.28tzangerneither can I
00:49.34tzangerfour provinces over is Quebec
00:49.43tzangerI forgot Saskatchewan
00:50.05tzangeryeah I know you're out there :-)
00:50.14ReD-MaNON here
00:50.15tzangeralthough I thought you were newf
00:50.18tzangerON here too
00:50.33tzangerMoc's PQ
00:50.35ReD-MaNomg.. calling my cell phone just caused it to reboot.. nice work Palm
00:50.37Drukeni'm from good old barrie ontario :)
00:50.39tzangersivana's ON
00:50.46tzangerReD-MaN: what phone?
00:50.51ReD-MaNTreo 600
00:50.58tzangerbarrie?  you're not far from sivana then, he's in NB
00:51.06tzangerI'm 1.5hrs WNW of Pearson
00:51.15tzangerReD-MaN: I'm waiting for the Treo650
00:51.19file[laptop]Pearson, I know that airport well
00:51.20Drukenhehehe if you drive like 200km/h
00:51.23ReD-MaNmy work is 5 mins from Pearson lol
00:51.31ReD-MaNthe plains fly over our office
00:51.47tzangerDruken: well he does drive quickly
00:51.54tzangeryou're not really far from me either, probably 2h
00:51.55ReD-MaNtzanger: work phone.. we bought over 200 of them.. before we knew how crap they were
00:51.56Drukenhehehe so i've heard
00:52.04ReD-MaNnow Rogers is going to upgrade us to 650s
00:52.14tzangerReD-MaN: yeah the 600s have their bugs but I thought most of those were corrected in the 650, with the addition of a few other bugs :-)
00:52.22tzangerReD-MaN: yeah I'm with Telus and Bell
00:52.26tzangerneither have the 650 yet
00:52.26ReD-MaNespecially now that RIM and Palm have signed to have Blackberry services on Treo 650 +
00:52.37tzangerReD-MaN: yeah, I have always HATED the blackberries
00:52.40file[laptop]we have no Mike out here  :D
00:52.43tzangerproprietary, slowass pieces of shit
00:52.54tzangernow bring some of that tech over to Palm and we got something
00:52.56ReD-MaNI have Telus for personal service, Rogers for work
00:53.30ReD-MaNwe call BB's Crackberries at work
00:53.38ReD-MaNcause every one who has one is ALWAYS doing something on it
00:54.07file[laptop]I remember when I was waiting at an airport, guy beside me was playing with his... trying to get it hooked up for data service to his laptop
00:54.10file[laptop]failed horribly
00:54.23tzanger:-)
00:54.25tzangerI hate them
00:54.29tzangernice transflective screens though
00:54.33tzangerbut they need a touch screen
00:54.36tzangerholy hell do they need touch screen
00:54.53ReD-MaNheh.. one guy at work.. every time he presses his space bar, the thing reboots.. so he has to send emails with a - instead of a space
00:55.03file[laptop]LOL
00:55.08rkingthat-sucks.
00:55.44Drukeni just use webmail... where can you go these days where a computer isn't already at?
00:56.10wunderkina cardboard box
00:56.10file[laptop]laptop+cellphone=goodness
00:56.50Drukenlaptop + aircard == heavenly
00:57.15file[laptop]Druken: bluetooth capable cellphone and laptop  :D
00:57.33*** join/#asterisk tehdely (n=delysiid@home.teambarry.org)
00:57.56Drukenthat works too... but ya still can't get calls and be online at the same time
00:58.18Drukenwith an air card, ya may be able to use voip with g729
00:58.20tehdelynews at 11: for anyone around the other night when i was trying to solve my popping/crackling problem
00:58.26tehdelyupgrading to bleeding-edge zaptel from cvs fixed it
00:58.29tehdelyoh yes it most certainly did
00:58.29Drukenmight be a lil shitty, but i bet it would work
00:59.20*** join/#asterisk Inv_arp (i=junya@adsl-144-134-75.mia.bellsouth.net)
01:00.12orlokI wonder how many people have been killed by Cisco 7940/7960's
01:00.14Drukenok, all these canadians in here, who has a 1u keyboard 19" rolling shelf they want to sell me for cheap ?
01:00.39*** join/#asterisk file (n=jcolp@mctnnbsa31w-142166113136.nb.aliant.net)
01:00.45filebleh
01:00.53fileinternet connection be b0rken
01:01.51ReD-MaN1u keyboard 19" rolling shelf?
01:02.01ReD-MaNI have some, dunno if they are 19" though
01:02.24ReD-MaNthey were from some weird rack at work.. thought they would fit my 19" rack, but they are too wide
01:03.17Drukenwell, if they won't fit your 19" i doubt they would fit mine...
01:04.06*** join/#asterisk orbi (n=dantate@pcp08696782pcs.500ash01.tn.comcast.net)
01:05.10orbiI just upgraded a Cisco 7960 phone to the latest SIP firmware, made a typo and got a Protocol Application Invalid error.    I cant find any definitive answers on ye olde google, but can someone tell me if i just made a perfectly good phone into a pretty paperweight?
01:05.41Qwellorbi: just reboot it, and it should get the right firmware, no?
01:06.04orbi<PROTECTED>
01:06.16orbibut if i watch the TFTP server, it never attempts to pull any of the right files
01:06.25orbinever tries to pull OS79XX.TXT for example
01:06.26Qwell"right files" or "any files"?
01:06.42orbiit tries to pull its own config, and a .tlv file, but thats it.
01:06.50ReD-MaNmmmm Cisco voip phone.. what I wouldn't give for one of those
01:07.01QwellReD-MaN: I'll sell you mine :p
01:07.03*** join/#asterisk ejo1 (n=ejo1@209.32.147.246)
01:07.10ReD-MaNalmost had Cisco to deploy at work... until they decided to go with Mitel yuck
01:08.01orbiim just now starting to play with Asterisk
01:08.19orbiwe use Cisco Callmangler at work, and resell it
01:08.27orbibut they're interested in asterisk for smaller offices
01:08.41orbii'ev got a ton of newbie questions, but i'm not going to espouse them here.
01:08.45*** join/#asterisk file[laptop] (n=jcolp@mctnnbsa31w-142166113136.nb.aliant.net)
01:08.54orbiThats what bugging twisted is for.
01:08.54orbi;)
01:09.11tehdelywe have a cisco phone at work
01:09.14tehdelybut it has the SCCP image
01:09.21tehdelyand i do not feel like dicking with chan_sccp
01:09.35tehdelyi think i'll go spend $200 on a support contract to get the SIP image
01:09.36tehdely;)
01:09.59ReD-MaNheh.. I have access to all Cisco downloads.. but not much to put them on lol
01:10.05Augheyhow many X100P cards can I put in a machine reliabily?  I can get 4 X100P cards for much less than a single tdm04 card setup.
01:10.45Drukeni managed to get 3 working properly
01:10.46Qwellreliably?  0
01:11.02Drukenbut that's 2 x100p's and 1 TDM
01:11.12orbiI was going to use sccp or skinny, but people swear up and down that SIP is better
01:11.17Augheydid you try 3 x100p?
01:11.17orbido you guys find that true?
01:11.42tehdelyorbi: i don't know enough about the implementation of either to say
01:11.48tehdelybut SIP is definitely a better choice if you're working w/ open systems
01:11.50DrukenAughey: nope... but if i can use the 3 cards like i do, i'm sure you could do 3 x100p's
01:11.54tehdelylike asterisk, or any other SIP IP PBX
01:11.57Drukenthey all use an IRQ
01:11.58tehdelySCCP is proprietary
01:12.05tehdelyand there is no open reference implentation, or really a reference at all
01:12.12tehdelyso everyone's impl except for cisco's is bound to be half-assed
01:12.15tehdelywhich i hear chan_sccp is :/
01:12.33Drukennight
01:13.28tehdelynn
01:13.42ReD-MaNhmmm. I wonder if Mitel phones do SIP
01:13.57QwellReD-MaN: are they even IP phones?
01:15.22ReD-MaNyup
01:15.32ReD-MaNin the middle of deploying em at work
01:15.53ReD-MaNI was given their crappy softphone to use since I am mobile user
01:16.27tehdelyanother crazy boogyman protocol from the crypt: unistim
01:16.32tehdelyanyone here ever buggered around with * and nortel phones
01:16.35tehdelyi'd like to hear how that went
01:18.06JASON-0I am using the AMP portal and i've setup an inbound DID but when I call it I get the following error in the logs..  - Extension '(did here)' in context 'default' from '(clid here)' does not exist.  Rejecting call on channel 0/14, span 1
01:22.34JASON-0anyone have any idea?
01:24.55Augheydo you have a default context in your extensions.conf file?
01:24.59tehdelyJASON-0: probably want to create an extension for it then ;)
01:25.01tehdely[globals]
01:25.05tehdelyMYCOOLDID=3125551212
01:25.13tehdelyexten => ${MYCOOLDID},1,DoStuff
01:25.55tehdelyjust guessing though
01:25.59tehdelyi am actually teh n0b
01:26.01tehdely^0
01:27.16*** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com)
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02:13.26*** join/#asterisk Goshen (n=Goshen@c-67-172-238-57.hsd1.ut.comcast.net)
02:13.33GoshenWhere does monitor store its files?
02:13.47Goshenseems like it was /var something
02:14.14Goshenahh, /var/spool/asterisk/monitor
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02:18.34lunkw
02:19.35|Vulture|Goshen: you can specify
02:20.15Goshen|Vulture|: Thank you, I just wanted to back up some recordings
02:20.50*** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au)
02:21.01|Vulture|Goshen: we use the monitor() function on all of our calls and it works well, make sure you specify the mux string as a global
02:21.45*** join/#asterisk Katty (n=katrina@68-112-15-110.dhcp.cpgr.mo.charter.com)
02:21.49Kattyhihi.
02:24.52*** part/#asterisk squeegy (n=mike@cpe-24-33-74-234.cinci.res.rr.com)
02:38.06Vcoany pointers on where i can unfux0ur this?    "Everyone is busy/congested at this time (1:1/0/0)"
02:38.12Vcoincoming SIP DID..
02:38.14Vcono joy
02:39.04X-RobVco - about 10 lines back is the actual error.
02:39.22X-Robusually 'no extension xxx in context yyy'
02:39.42*** join/#asterisk shido6 (n=shido6@d221-68-210.commercial.cgocable.net)
02:39.42Vcowell..thats the thing
02:39.46shido6boink...
02:39.51Vcothe phone is ringing..
02:39.56Vcoi answer it and it goes busy
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02:39.59shido6answer it
02:40.00Vcoon the phone
02:40.07Vcothe incomgin call still rings
02:40.30X-Robsip debug
02:42.02paxr0i correct this line? exten => s,28,NoOp(${CHANNEL STATUS ZAP/1})
02:42.21paxr0y have a FXO in Zap/1-1
02:45.43Vcoyay...miles of debug output
02:45.53phrog123if I put context=something in zapata.conf in the channel for ZAP/1 and in extensions.conf, I do not create a context called something, does this guarantee that * will not answer zap/1?
02:46.04phrog123if I forego of doing context= in zapata.conf, then this is as if context=default?
02:46.10phrog123finally, if I put context=something after channel=>1, then it is as-if context=default
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02:50.12*** join/#asterisk JASON-0 (n=jason@jason.unitz.ca)
02:50.38JASON-0Hello, I have AMP and I've added extensions busy when I call them, they are all busy and I get forwarded to vmail.. I'm not sure why..
02:51.10shido6brb
02:51.18shido6Jason dont freak out - brb
02:51.22shido6i'll help ya out in a minute
02:51.26JASON-0thank you :D
02:53.55hugo-v6does someone know if there exists a better solution than suEXEC from apache to wrap cgis?
02:54.51Inv_arpAriel_:  around?
02:58.35*** join/#asterisk hcir (n=hcir@rdbck-static-532.palmer.mtaonline.net)
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03:04.02JASON-0Hi shido6, did you still want to help me :P
03:05.35shido6sure
03:05.37shido6whats up?
03:06.38JASON-0Every extention I dial I get busy or vmail.  I am able to dial out on trucks but no extensions. I have used AMP to add the extension.
03:06.58hugo-v6hmmm dan 5am and no sleep at all.
03:07.04hugo-v6at least its my bd.
03:07.14hugo-v6s/dan/damn/
03:10.21glm2kbd?
03:10.30glm2khappy birthday then?
03:11.16blitzrageJunK-Y: lol -- you have a Junky tshirt!
03:11.28JASON-0shido6: Any idea?
03:12.04file[laptop]blitzrage: !!!
03:12.08X-RobJASON-0 - your dialparties.agi is broken.
03:12.17file[laptop]blitzrage: did you make it okay?
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03:12.32X-RobJASON-0 - also, AMP support is on #amportal (hint)
03:12.44JASON-0X-Rob: sorry
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03:13.19shido6ok
03:13.21shido6what phones?
03:13.25X-Robyou're, uh, meant to join the #amportal channel.
03:13.25shido6ip phones or analog
03:13.35JASON-0Cisco 7960
03:13.38shido6nice
03:13.43shido6sip show peers shows them registered?
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03:14.13JASON-0yes, sure does
03:14.28shido6any "x" 's next to the line names on the criscos?
03:14.43blitzragefile[laptop]: nope
03:14.54file[laptop]blitzrage: so they really turned you away?
03:14.59blitzragefile[laptop]: yep
03:15.25JASON-0shido6: no
03:15.32file[laptop]blitzrage: I prefer you without 'da beard
03:15.42hugo-v6thanx glm2k
03:15.57blitzragefile[laptop]: that was kind of the point :)
03:16.21Corydon76-homepretty pretty blue eyes
03:16.42shido6kewl
03:16.48shido6now follow the CLI error
03:16.53shido6and paste them at pastebin.ca
03:16.57shido6erro(s)
03:17.01JASON-0http://pastebin.ca/26437
03:17.15JASON-0I think I found the problem.. X-Rob was right
03:17.27X-Robshido6 - it's AMP, can't call extensions, dialparties.agi is broken.
03:17.31shido6ooooh busy phones
03:17.33X-Robthat's always the problem 8)
03:17.46shido6u want my dialparties.agi?
03:17.48JASON-0Can't locate Asterisk/AGI.pm
03:17.56shido6erf
03:17.57X-Robshido6 - it's missing a prerequisite.
03:18.00JASON-0I'm midding Asterisk::AGI
03:18.04JASON-0oops.. missing
03:18.04shido6asterisk perl?
03:18.05Qwellblitzrage: The tartan room!  heh
03:18.07orlokwoo
03:18.11JASON-0yes :P
03:18.13shido6then download asterisk perl
03:18.14JASON-0thanks guys
03:18.14orlokgot the sip images onto this here 7940
03:18.15shido6dagnabbit
03:18.16orlok:)
03:18.17shido6and install it
03:18.20shido6perl Makefile.PL
03:18.23JASON-0thank you shido.. I will :P
03:18.23shido6make install
03:18.32JASON-0Thanks X-Rob
03:19.14X-Robw00
03:19.23shido6my site will be up shortly
03:19.25X-RobOpteron?
03:19.27shido6no
03:19.28shido6X2
03:19.31X-Robbah
03:19.38blitzrageQwell: hehehehe :)
03:19.39X-Robthe one that takes 400w at idle?
03:19.49X-RobI'll pass on that. I like my power bill only being 2 digits long, thanks.
03:19.59orlokman
03:20.08orlokimagine upgrading a whole server room to those suckers
03:20.14orlokimagine the aircon upgrade you would need
03:20.19shido6fine pass it up :) give it back
03:20.24orlokwould probably cost more than the CPU's :)
03:20.35X-RobBut I'll have a dual core opteron anytime, thanks 8)
03:20.43shido6thanks
03:20.46JASON-0It works now :D :D :D
03:20.48JASON-0thanks guys
03:21.11shido6I will ship my first one next week
03:21.29shido6i have 4600+'s working
03:22.08Qwellblitzrage: so, where was his shoe?
03:23.23Vcook..
03:23.35Vcoi'm thining this DID issue is something else..
03:24.02Vcohave some UK ones that work fine..same config...it's 2 DID's for Tokyo from 2 different providers...
03:29.51*** part/#asterisk mcadory (n=mcadory@208.149.64.28)
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03:36.40Inv_arpAriel_: around?
03:38.26blitzrageQwell: I think it was in a bush somewhere :)
03:41.29hypa7iablitzrage: the unidentified person in the photos with you and mark and allison is qwell
03:41.44JunK-Yblitzrage: u were able to decipher my shit btw?
03:42.46Qwellhypa7ia: where are those?
03:43.10hypa7iahttp://leifmadsen.com/gallery/astricon_2005
03:43.16Qwellahh
03:44.09JunK-Ysome are at
03:44.12JunK-Y~astricon2005
03:44.13jbotrumour has it, astricon2005 is at http://www.midsouthmarketplace.com/~krice/gallery/view_album.php?set_albumName=album02
03:44.18*** join/#asterisk dudes (n=dudes@12-215-32-62.client.mchsi.com)
03:48.12FuriousGeorgeyou know what bothers me.  when you a dial(sip/user,30) followed by a voicemail(u${USEREXTEN}), what if the call is answered and the answering party hangs up first.  why shouldnt it still go on to the next priority
03:48.37FuriousGeorgeiow, the calling party would still go to voicemail.  seems like it should
03:50.05IronHelixdial(sip/user,30,g) will do that
03:50.10IronHelixg: When the called party hangs up, exit to execute more commands in the current context.
03:55.23FuriousGeorgeIronHelix: i recind my complaint
03:55.28IronHelix:)
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03:58.58*** part/#asterisk SplasPood (n=sp@brooklyn.paravolve.net)
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04:01.50encodeooh, lots of pretty astricon pictures
04:02.07encodewhat happens at astricon? obviously its a conference about asterisk
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04:02.14Qwellencode: lots of drinking
04:02.20encodelol
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04:05.21blitzragehypa7ia: oh nooooooo way!!!
04:05.30blitzrageQwell: holy shit, I just realized *now* who you are!!!!!!
04:05.33Qwellhaha
04:05.52Qwellyeah, I don't think I ever actually intoduced myself...
04:05.53blitzragenext Astricon, I'm making sure there is a spot to put IRC names on the badges
04:05.59Qwellblitzrage: yes, thats important
04:06.04blitzrageQwell: you probably did, but I'm terrible with names
04:06.36Qwellblitzrage: makes sense why I bought you a drink now, eh? :p
04:06.57blitzrageQwell: uhhhhh yah :)
04:07.02blitzrageQwell: I feel like a fool now :)
04:07.09Qwellblitzrage: its all good, heh
04:07.24QwellI thought you were one of the Matts when you first got to the tartan room...
04:07.32blitzrageQwell: lol
04:07.38Qwellin my defense, I was a bit drunk by that time, heh
04:07.50blitzrageQwell: hehehehehe, who wasn't? :)
04:07.53file[laptop]blitzrage: your name is cooler then Matt though
04:08.02blitzragefile[laptop]: well yah, thats true :)
04:09.16blitzragebut I *am* a r0ck star... so it makes sense
04:09.55Vcothat doesn't look like your arm tho
04:10.11Corydon76-homeThe Leif doesn't fall far from the (Asterisk) tree
04:11.08blitzrageCorydon76-home: I don't really know what that means... :)
04:11.19blitzrageCorydon76-home: oh I get it now if I say my name wrong
04:11.24blitzragelol
04:11.28blitzrageLeif == Life, not Leaf
04:11.48Corydon76-homeblitzrage: it's meant to confuse you while I take other measures to seduce you... :-P
04:12.25blitzrageCorydon76-home: gah!
04:12.44Corydon76-homeEverybody can be seduced...
04:13.10Corydon76-homeThe only question is, how difficult?
04:13.11blitzrageCorydon76-home: not true
04:13.33Corydon76-homeblitzrage: there's always chloroform... :-P
04:14.10blitzrageCorydon76-home: I suppose thats true... not much you can do to fight that
04:14.57Qwelleh...chloroform isn't difficult to avoid.
04:15.14Qwelllearn the smell of it, and if you ever smell it, just hold your breath :p
04:15.27Corydon76-homefile[laptop]: only mildly?
04:15.33file[laptop]only mildly.
04:15.35file[laptop]try harder next time
04:16.09Corydon76-homefile[laptop]: it doesn't get much harder than it is now...
04:16.19file[laptop]that's sad
04:16.57Corydon76-homefile[laptop]: Ask Coil about it sometime...
04:17.02blitzragecan any else get to creativecommons.org ?
04:17.56Qwellno, but I get an incredibly low ping to them
04:18.05Qwell9ms on cable?
04:18.08encodepage doesnt seem tobe loading
04:18.14*** join/#asterisk denon (i=denon@synapse.subneural.net)
04:18.14*** mode/#asterisk [+o denon] by ChanServ
04:18.26file[laptop]omg denon
04:18.27encodei get "operation timed out" from our proxy
04:18.30file[laptop]nini all
04:18.37Corydon76-homeg'night, file
04:18.43encodecya file
04:19.01Qwelloh, I'm on my server still...wtf
04:19.50blitzrageyah, doesn't load for me, must be down or something
04:21.07blitzrage~seen jerjer
04:21.10jbotjerjer <n=JerJer@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #asterisk, 1d 11h 1m 39s ago, saying: 'but playback will answer the channel if its not already up'.
04:21.33Qwellblitzrage: working now
04:21.35blitzrageanyone have Jeremy's IM? (if he uses IM that is)
04:21.37blitzrageQwell: thanks
04:26.54Corydon76-homeblitzrage: AIM:Corydon76-home is his IM...
04:27.31Corydon76-homeOr maybe just AIM:Corydon76  ;-)
04:28.48blitzrageheh
04:29.14blitzrageplus I'm a bit disillusioned with the USA today
04:29.18Corydon76-homeGee, I dunno why...
04:29.43Corydon76-homeIt's not like Canada doesn't also watch its citizens...
04:29.58wunderkinblitzrage: you'll probably end up talkin to some 70 year old pervert :D
04:30.19Qwellwunderkin: in the internet age, that is inevitable
04:30.25Corydon76-homewunderkin: No, that's file.  He only pretends to be 19 online...
04:31.00blitzragelol
04:31.15wunderkin.oh maybe hes 76 :D
04:31.22Corydon76-homeUh huh
04:32.04Icemaann76?
04:32.05Icemaannlol
04:32.15wasimdon't forget silly c888
04:33.36Corydon76-homeIcemaann: bitch.  :-P
04:33.57glm2k76 going 77? or 75 going 76?
04:34.18Icemaanni have impressive ping times to sellvoip (the ips listed on their site), but ive never used them
04:34.33QwellIcemaann: What is impressive?
04:34.39Corydon76-homeBorn in 76.  Which would make me one thousand, nine hundred, twenty nine years of age...
04:35.12glm2khmmm, that means you're either a ponty eared, blue skinned elf or a warlock
04:35.34Corydon76-homeglm2k: and I have a bridge I'd like to sell you...
04:35.44glm2khehehe
04:35.54IcemaannQwell: 16 ms
04:36.23glm2kIcemaann: wow. i get 76 ms
04:37.27Icemaannglm2k: did u look at the ips in the FAQ? i get 76 to 1 and only 16 to the other
04:38.37*** join/#asterisk LostFrog (n=reallyno@dsl093-100-093.wdc2.dsl.speakeasy.net)
04:38.43Icemaann7 hops for me
04:39.21LostFrogkram: may I just say thank you for asterisk and gaim.. you are a god.
04:41.35glm2k13 hops for one and 18 for the other
04:41.50Vcoanyone else here use voxbone?
04:42.13Qwellvoxbone sounds like a really bad porno or something
04:42.17Vcoyea..i know
04:42.32glm2khehe
04:42.37VcoVoip Over Interracial Porno
04:42.43Vcoor some shit
04:43.24LostFrogVery Overrated Internet Porn
04:43.39*** join/#asterisk wjn78 (n=wjn78@adsl-68-123-232-96.dsl.irvnca.pacbell.net)
04:44.38wjn78did I miss anything?
04:44.39*** join/#asterisk Jabroni (n=Hercules@red-corp-200.76.249.142.telnor.net)
04:45.24Vcoyea, Microsoft Just bought Digium
04:45.55*** join/#asterisk denon (i=denon@synapse.subneural.net)
04:45.55*** mode/#asterisk [+o denon] by ChanServ
04:46.04wjn78lol that would be bad
04:46.07wjn78I like Asterisk
04:46.53Jabronianyone knows if there a way to get the response time for IAX2/SIP peers via the manager api ??
04:47.13LostFrogAck.. those are fighting words, Vco.
04:47.29wjn78LostFrog its a joke because I use to work at Microsoft
04:48.18Icemaannthis conversation about goiax on -users is interesting.
04:53.43*** part/#asterisk Hobbes` (n=calvin@59.92.143.37)
05:04.04Vcobah...verbose cli display and sip debug just blur together now....
05:09.46wjn78Vco how long have you been working on Asterisk?
05:21.49*** join/#asterisk santiago (n=santiago@208.195.215.158)
05:22.42Vco**shrug**, maybe year or 2 now...more seriously for the last 6-8 months
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05:46.27genmudescribzz what part of phoenix you in?
05:46.49escribzzGenmud: North Phoenix
05:46.56genmudCoo
05:47.00escribzzWhere are you at?
05:47.03genmudI am like 13th and Union Hills
05:47.35escribzzI'm at 23rd & Happy Valley lol just down the road
05:47.42genmudyea
05:47.44genmudthat is funny
05:47.56genmudwhere do you work for?
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05:55.27Qwelloej: morning
05:58.15paxr0i have a question, im tryng to use DIALSTATUS before Originate, the print of NoOp(${DIALSTATUS}), why i can know if the called answer , or this variable it only por use in AGI ? anybody can suggest me ?
05:58.37paxr0the print of NoOp(${DIALSTATUS}) before Originate is ""
05:59.13paxr0im tryng to use DIALSTATUS in dialplan for outcall....
05:59.22blitzrageoej: morning!
05:59.39Qwellpaxr0: What version of asterisk?
05:59.44QwellDIALSTATUS is fairly new, isn't it?
06:00.31paxr0hummm
06:00.39paxr01.0.9
06:00.40QwellDon't quote me on that
06:00.58Qwellpaxr0: read doc/README.variables in the asterisk source for 1.0.9
06:01.10blitzragenight all
06:01.12paxr0ok thanks
06:01.14paxr0night
06:01.50infinity1wow. fun day updating voip-info
06:07.16littleballhello, any relationship between extension h and T?
06:07.28Qwelllittleball: just that they happen after certain events
06:09.00littleballQwell, because i need to trigger DeadAGI at the end of call. It seems sometimes, DeadAGI is triggerd by T and other times, it is triggered by h
06:09.22Qwellt is timeout.  If its waiting for something, and it never happens, thats where it goes
06:09.22littleballQwell, i need to make sure the DeadAGI only run onetime
06:09.28Qwellif the call is hung up, it goes to h
06:09.51littleballafter T, does it to to h?
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06:15.10wunderkinT is absolute timeout
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06:16.57LostFrogtest it by putting a NoOp in each priority.
06:17.04LostFrogAnd watching the log.
06:19.04infinity1tzafrir_laptop: thanks!
06:19.15infinity1tzafrir_laptop: -Brendon
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06:37.20LostFrogI should be able to provide settings to a snom phone using tftp, shouldn't I?
06:39.23Inv_arpLostFrog: is that what the directs say?
06:39.33Inv_arpdirections*
06:40.07*** join/#asterisk e-Hernick (n=ncc@modemcable120.39-131-66.mc.videotron.ca)
06:44.05LostFrogI think I found what I needed.
06:44.07*** join/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com)
06:49.52*** join/#asterisk af_ (n=af@ip-142-250.sn1.eutelia.it)
07:00.27*** join/#asterisk argos73 (i=1000@jason.argos.org)
07:00.51argos73any echo gurus around?  (not a usual "why does my x100p have echo" question)
07:04.32Guggemandargos73 try asking what you wanna know, maybe soneone have the answer :)
07:04.43argos73heh - that makes too much sense... :)
07:04.48argos73here ya go
07:06.38argos73given a T1 going into an adtran 750 channel bank, with some channels passed to the FT1 port to a digium T1 card, echo is still a problem.  phones are SIP hardphones.  can the same be expected if I was to bring a PRI directly into the digium card?
07:12.26*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
07:13.05*** join/#asterisk mkl1525 (n=daniel@84.19.198.194)
07:14.59*** join/#asterisk giesen (i=giesen@CPE0040f404c559-CM00122546417a.cpe.net.cable.rogers.com)
07:15.18giesenanyone alive that could lend a hand and perhaps a thought?
07:15.33giesenI get - wrong password on authentication for INVITE
07:15.46giesenwhen I dial between to cisco 7940's
07:16.01giesenbut I can dial between an Xlite soft phone and one of the 7940's no problem.
07:16.25*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
07:16.30mkl1525Hi, trying to setup asterisk with a hfc-s card. When starting with asterisk -vvv -c it quits with "chan_modem.c:394 modem_setup: Unable to autodetect modem." so it seems that it tries to open some modem is this the right way for a zapata card or is there some misconfiguration?
07:17.27X-Robmkl1525 - you're not using chan_modem
07:17.31X-Robthat's not the error you care about.
07:18.23*** join/#asterisk cj- (n=jrocha@CPE0040ca7bd059-CM001225db967c.cpe.net.cable.rogers.com)
07:19.14cj-Hey guys, is Asterisk seg faulting on rxfax a known issue?
07:19.23X-Robyes 8)
07:19.27X-Robusually because you haven't run ldconfig
07:19.45X-Robcheck where spandsp.so is, and make sure it's in /etc/ld.so.conf and then run ldconfig
07:20.35cj-Already done :(
07:20.43X-Robcheck /var/log/asterisk/full
07:21.46cj-Do I have to run in debug or soemthing to get that log?
07:21.56X-Robenable it in /etc/asterisk/logger.conf
07:22.26cj-aha... Brilliant... LEt's see if I get anything now.. (reproduces crash)
07:24.06mkl1525X-Rob, this means in my configuration I wouldn't need chan_modem? Tried to comment chan_modem out in modules.conf but starting asterisk still fails with "WARNING[4542]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_modem_bestdata.so: undefined symbol:ast_unregister_modem_driver"
07:25.04X-Robmkl1525 - put noload=>chan_modem.so in modules.conf
07:25.31X-Robchan_modem_bestdata.so?
07:25.33X-Robjust delete it
07:25.43X-Robrm /usr/lib/asterisk/modules/chan_modem_bestdata.so
07:25.56X-Robit's some old module that's been left lying around
07:26.43cj-Nothing useful in full :(
07:27.16cj-Essentially it seems to be related to large tif files... I think. Although it's nothing I'd call 'massive'.. It's 400k
07:27.19*** join/#asterisk shido6 (n=shido6@d221-68-210.commercial.cgocable.net)
07:27.28X-Robcj- oh. I thought it was an onload crash
07:27.29cj-(I'm using txfax to send it back to myself)
07:27.33X-Robmake sure you're using the correct libtiff
07:27.37X-Robthat causes that problem
07:28.00cj-Which one is the correct one? :) I'm running 3.7.3
07:28.10X-Robit's on the wiki. voip-info.org
07:28.13af_what may I put as "Application" in originate? (manager interface)
07:28.24cj-*checks*
07:28.39X-Robaf - on the command line, 'show applications'
07:29.06*** join/#asterisk paxr0 (n=P@200-85-200-140.bk3-dsl.surnet.cl)
07:29.14af_I see, so any of the cli commands?
07:29.21af_that's cool
07:29.34af_mhhh, what could be used for?
07:31.08cj-oh shit I'm a moron
07:31.17cj-I updated my libtiff during a recent emerge :(
07:32.25*** join/#asterisk eivindtr (n=wingnut-@194.248.208.94)
07:33.17X-Robooh, I know this one. Gentoo sucks!
07:33.47Igbothom_IIIrofl
07:34.25Igbothom_IIIvacuum cleaners do, as to their salespeople
07:35.01orlokHey
07:35.07orlokis anybody using a cisco 7912?
07:35.31orlokI've just managed to upgrade a 7940 to a sip image, but i forget now how the hell i made it get the new image!
07:35.41Igbothom_IIItftp
07:36.19cj-lol I've actually only been using Gentoo for like 2 weeks now (in a larger role that I previously used it for)... Since I'm a FreeBSD guy I sort of like it... I'm chalking this one up to me being careless :)
07:36.36Igbothom_IIIFreeBSD is nice
07:36.47mkl1525X-Rob, thanks for the help but no it complains about chan_modem_aopen.so and if I move this "/usr/lib/asterisk/modules/chan_modem_i4l.so: undefined symbol: ast_unregister_modem_driver" so it seems as there is still a reference to chan_modem although I inserted  noload => chan_modem.so in modules.conf - any further suggestions?
07:36.51cj-Okay, I just compiled 3.5.7... now to test :)
07:37.09X-Robmkl1525 - you can happily delete any chan_modem* files
07:37.11X-Robyou don't need 'em.
07:37.23orlokIgbothom_III: yes, i know that bit.. i'm wondering how i make the phone reqyest the image?
07:37.30orlokIgbothom_III: or do i specify a file via dhcp?
07:41.39cj-hmm, that seems to have fixed the crashing... It still doesn't accept the transmission though :(
07:42.28*** join/#asterisk wjn78 (n=wjn78@adsl-68-123-233-181.dsl.irvnca.pacbell.net)
07:42.41cj-Although I'm doing some weird things with my fax
07:43.31cj-I'm basically working on a email -> fax gateway... I have tif's working properly, but now I got greedy and want gifs. So I'm converting the gifs to tifs but ... oh it crashed again...  when it gets the large tif it craps out :( (the whole server dies)
07:43.48orlokwhat gif-tiff util?
07:43.49Igbothom_IIIpdf is so much nice
07:43.52Igbothom_IIInicer
07:43.54orlokimagemagic?
07:43.57orlokyes, PDF's rock
07:44.19Igbothom_IIIlol
07:44.20cj-I tried a few... I used the gif2tiff util but that wouldn't detect any gifs properly
07:44.35argos73orlok: IIRC, my 7960 requests OS79XX.TXT from tftp, which contains the requested software version (POS3-06-1-00).  if the phone is not running that version, it grabs it via tftp and updates.
07:44.38cj-so I used a program called a2ping.pl which is a front end to several utils
07:44.47Igbothom_IIII'm not wearing my "deCSS" t-shirt, but I *did* support the guy
07:44.50orlokargos73: yeha, thats what the 7940 did
07:44.57cj-I have my incoming faxes convert from tif to pdf already
07:45.02cj-I'm trying to do the opposite
07:45.05orloktcpdumping tftp, havent seen any requests for it
07:45.08cj-My outgoing have to end up as a tif :)
07:45.22orlokIgbothom_III: i have to wear this. Work.
07:45.29Igbothom_IIItif is just sooooooo big
07:45.35argos73orlok: "for it" - OS79XX.TXT or POS3.... ?
07:45.39Igbothom_IIIwhere ewe work?
07:45.58cj-txfax won't send in anything other than tif though
07:46.01cj-(TMK?)
07:46.17*** join/#asterisk kks (n=kks@202.73.8.130)
07:46.57orlokargos73: i get requests for SEP000E833CB763.cnf.xml
07:47.35cj-Oh well, time for bed :(
07:47.36orlokoh gno, you know my mac address, dont maxxor me!
07:47.37orlok;)
07:47.44orlokhaxxor, even
07:47.47argos73hehe
07:48.09orlok7940 is working well
07:48.13orlokbut my boss took it home
07:48.36orlokIgbothom_III: place that specialises in stuff for the graphic arts industry
07:49.03argos73the xml request should happen after the image request..
07:49.17argos73i think
07:49.18orlokargos73: yeah, i thought so
07:49.23argos73lemme look
07:49.29Igbothom_IIIorlok; cool, what drugs in particular?
07:49.40orlokheheh
07:50.03orlokhmm.. hye, another aussie ;)
07:50.09X-Robwhat?
07:50.09X-RobwherE?
07:50.10argos73my SIP...xml file does have a line in it - "image_version: P0S3-06-1-00"
07:50.15orlokhey, wtf, you are on #centos as well! :)
07:50.29Igbothom_IIIyup
07:50.43argos73random memories tell me that the "image_version" line is important
07:50.43X-Robwoo aussies!
07:50.45X-Robwoo!
07:50.49X-Robwe r0xx0r.
07:50.56Igbothom_IIII'm an Australyanmate!
07:50.56orlokbeer oclock
07:50.59orlokand we all hit irc
07:51.05Igbothom_IIIyup
07:51.18X-RobI've been here all damn day
07:51.30Igbothom_IIIyou hear that "beer o'clock" is now in the Macquarie Dictionary?  :)
07:51.42Igbothom_IIII've been onsite for half of it
07:51.51Igbothom_IIIoffice for the rest
07:52.16*** join/#asterisk smurf (n=smurf@debian/developer/smurf)
07:54.33paxr0asterisk not load app_dial.so, i think i need this for DIALSTATUS anybody can suggest ?
07:54.44X-Robyou need app_dial
07:54.48X-Robthat's the most important application
07:55.01paxr0:(
07:55.13Igbothom_IIIanyone else having issues with their IAX connexion to fwd?  Mine remains <unregistered>
07:55.57paxr0asterisk send errors to the log whe i try to load
07:56.12X-Robwhich errors?
07:56.15X-Rob~pb
07:56.16jbotpastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca/
07:56.20X-Rob^^^ put them there
07:57.37paxr0Oct 24 09:22:15 WARNING[3443]: /usr/lib/asterisk/modules/app_dial.so:   undefinedsymbol: ast_bridge_call      Oct 24 09:22:15 WARNING[3443]: Loading module app_dial.so failed!
07:58.00X-Robpaxr0 - recompile everything.
07:58.06X-Robyour modules are out of date.
07:58.29paxr0ok thanks :)
07:58.42X-Robpaxr0  - btw, use CVS HEAD
07:58.47X-Robit's much better than 1.0
07:59.00*** join/#asterisk rene- (n=rene-@201.144.60.6)
08:00.08paxr0ok thanks, really i will follow ur suggest
08:01.03*** join/#asterisk Guest^DJ (i=me@60.48.51.109)
08:01.21Guest^DJhi all, when a user call into *, how can i give the user a dial tone ?
08:01.38X-RobGuest^DJ - the magic command is DISA
08:02.12Guest^DJX-Rob: DISA? where can i find more info ?
08:02.35X-Rob'show application DISA'
08:02.44X-Robon your asterisk console
08:03.17littleballhi, is the default context must be defined in the extensions.conf ?
08:03.30X-Roblittleball - try again?
08:03.34Guest^DJX-Rob: thanks
08:03.38*** join/#asterisk Miggidy (i=user@dsl-202-72-180-171.wa.westnet.com.au)
08:04.02littleballX-Rob, must i define the default context in the extensions.conf?
08:04.25X-Roblittleball - no. The default context is defined in your sip.conf or iax.conf or zapata.conf
08:04.28X-Robuh
08:04.30X-Roblet me rephrase that
08:04.35X-Robthe _context_ of the client is..
08:04.44X-Robif there's no context specified, it uses [default] in extensions.conf
08:05.13littleballX-Rob, what i mean is the [default] compulsary?
08:05.55shido6how do you replace all occurances of a string in vi
08:05.56shido6?
08:05.57X-Robwell, no
08:06.10X-Robbut if you dont' specify a context and there's no default, I'm not sure what happens
08:06.18X-Robshido: ':%s/string1/string2/g'
08:06.27X-Robthat's through the entire file
08:06.33X-Robdrop the % for just that line
08:06.38littleballX-Rob, if [default] is not in the extensions.conf, and if there is no context specified in the sip.conf etc, what will happen
08:06.45X-RobI don't know.
08:06.46X-Robdon't do it 8)
08:07.05littleballX-Rob, yes, i just drop it.
08:07.07littleballthanks
08:07.23littleball[default]
08:07.23littleballexten=>i,1,Hangup
08:07.26littleballis this enouth?
08:07.27glm2kIgbothom_III: iax2 thru fwd status UNREACHABLE :(
08:07.35littleballs/enouth/enough
08:07.47Igbothom_IIIglm2k; yeah, seems they are having issues
08:07.53Igbothom_IIImentioned on their forum too
08:08.06shido6X-rob
08:08.19shido6I want to replace /nagios/cgi-bin with /cgi-bin
08:08.20shido6in vi
08:08.21X-Roblittleball - um. should be exten => _X.,1,Playback(weasels) then _X.,2,Hangup
08:08.29shido6so do i use the '/' you mentioned?
08:08.37X-Robso if anyone ever mentions hearing 'weasles have eaten our phone system!' you know there's a problem
08:09.16X-Robshido6 - ':%s/nagios\///g'
08:09.20littleballok
08:09.24littleballthanks
08:10.51littleballX-Rob, weasels is already included in the system?
08:11.21X-RobI think so
08:11.26X-Robcheck /var/lib/asterisk/sounds
08:11.35X-Robyou may need to download asterisk-sounds from cvs
08:11.37*** join/#asterisk psk (n=psk@golia.caltanet.it)
08:12.46littleballX-Rob, weasels is not there. but there is beep
08:13.00X-Robwhatever you want 8)
08:13.13*** join/#asterisk darkskiez (n=darkskie@194.247.78.146)
08:13.19*** join/#asterisk TMirage (n=mirage@cust.12.229.adsl.cistron.nl)
08:13.25kksif two clients in the same LAN registered at a public asterisk server and want to call each other. Will the asterisk server detect those clients are from same LAN, like MSN do???
08:13.57X-Robkks - it will try to bridge the connection between the phones no matter what.
08:14.14X-Rob(unless you specify no briding, by using tT or something els eon the dial string, or various other reasons)
08:17.39glm2kargos73: lol. yep. just a few minutes alright.
08:17.54kksX-Rob, thanks
08:18.11glm2kargos73: use the 24 bucks for something else
08:18.30argos73glm2k: it's a pair of connectors with a capacitor in the middle....  $25 is crazy
08:18.37glm2kaye
08:19.27glm2kargos73: you's paying for the robot's labor hehe
08:19.39glm2ker, you're
08:19.40argos73of course, i'll need to pick up a $3 SV to SV cable from walmart to hack apart... :)
08:19.43orlokand nicer connectors
08:19.45orlokand a clean job
08:19.51orlokand not worrying about breaking solder join
08:19.53orloks
08:20.01argos73heh - my soldering is picture-perfect... :)
08:20.04glm2kheh, it all depends on your soldering skill.
08:22.23*** join/#asterisk newl (n=newlook@203-59-214-216.dyn.iinet.net.au)
08:26.15mkl1525X-Rob, thanks for the help so far after moving all chan_modem modules asterisk is starting again and works. But I have another problem: my configuration is sip-phones <-> asterisk <-> elmeg pbx <-> ISDN provider, I'm able to call from my sip phones to the outside world but asterisk doesn't react on incoming calls so I'll have to assign a msn is this done in capi.conf or another file?
08:26.54*** join/#asterisk Goshen (n=Goshen@c-67-172-238-57.hsd1.ut.comcast.net)
08:28.18*** join/#asterisk brasco (n=root@83.137.128.7)
08:28.35X-Robmkl1525 - sorry, out of my league there.
08:30.14mkl1525X-Rob, don't worry - you helped me a lot thanks for this!
08:37.22paxr0X-Rob, u think an update to asterisk using CVS and make update work fine?
08:37.39X-Robpaxr0 - yep.
08:38.00paxr0thanks a lot
08:40.36encodehi - i'm trying to get asterisk to register with a voip provider (engin in australia), and i'm getting in my log the following message: "SIP response 423 "Interval Too Brief""
08:40.39encodeany ideas?
08:41.28X-Robencode - odd.
08:41.57littleballhello, what is the function of the context field in zapata.conf?
08:42.08mkl1525So does anybody know where I have to configure the msn on which asterisk should listen if I use a hfc-s card in combination with the bristuff drivers?
08:43.18littleballhi, mkl1525, what is hfc-scard?
08:43.55*** join/#asterisk christo (n=chris@brezhnev.spiration.co.uk)
08:44.01christomorning all
08:44.30festr__hello, i've problem with sending calleridpresentation over IAX2 channel, is it possible this? anyone is using this?
08:44.45mkl1525littleball, it's a isdn card using a hfc-s chip to process the isdn data
08:45.31littleballmkl1525, like digium card?
08:46.05encodeX-Rob: got any more insight than just odd?
08:48.43X-Robencode, I think it's saying 'you're trying to log in too rapidly'
08:48.57X-Robbut I don't use engin
08:48.58X-Robsorry 8)
08:49.04X-Robfaktortel is the best I've found
08:49.09X-Robas they actively support asterisk
08:49.19encodebut...how can i be trying to log on too fast??
08:49.24encodenever mind...
08:49.27encodetheres an engin forum
08:49.37encodewhich would be great, if i could sign up
08:49.56encodestupid website wont let me log on with either my phone number of my account number
08:50.02encodeboth of which are supposed to work
08:50.07encodeanyway, bbl
08:50.29kllhow do I get asterisk going with some form of zaptel dummy on sparc/linux? zaptel won't compile properly.. any pointers?
08:50.37X-Robkll - you can't.
08:50.49X-Robztdummy only works on x86
08:50.59mkl1525littleball, don't know if this is comparable to the digium cards
08:51.52kllX-Rob: so any corresponding for sparc? I've heard of zaprtc but I'm unable to find much more info on it..
08:54.29X-Robrtc only works on intel, too
08:54.37X-Robunfortunately, you're outta luck for timing on sparc
08:54.46X-Robyou got a PCI bus in the spac?
08:54.48X-Robsparc?
08:58.02christoI have a media gateway receiving a call over E1. It then routes the call over IAX to an application server which runs an AGI. For some reason the call just cuts off in the middle and I can't work out why
08:58.06christocan anybod think of a reason for this?
08:58.32christoIt seems to happen randomly - sometimes not at all and always at different places
08:59.59christocould it be the quality of the streamed sound file? I mean if it's too quiet, or if there is a gap, the IAX connection might just drop it
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09:06.26kllX-Rob: yupp, got one PCI...
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09:49.08MuppetMasterHello
09:49.17MuppetMasterAnyone had a chance to download and test http://www.astjab.org?
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10:08.05mkl1525because I don't get any notice of incoming isdn calls, is there a way to look at the isdn channels, which data is transmitted on them?
10:08.13mkl1525using zaphfc driver
10:08.52jbensonHi all.  Does anyone use the ZyXEL P662HW ADSL router please, with the bandwidth management option?  We are trying to convince ZyXEL that the SIP support just doesn't work.
10:12.51*** join/#asterisk cinzas (n=ashes@62.48.254.104)
10:12.53cinzasHowdy
10:13.34MuppetMasterHowdy
10:18.35*** join/#asterisk ful|work (n=fulgas@209.8.233.227)
10:19.00*** join/#asterisk mmmToop (n=chatzill@196.31.11.194)
10:19.06|baby|alguien habla español? xD
10:19.13*** join/#asterisk dfordivam (n=dfordiva@59.176.52.35)
10:19.16MuppetMastersolo un poquito
10:19.24MuppetMasterComprende mucho, hablo poquito
10:19.27*** part/#asterisk dfordivam (n=dfordiva@59.176.52.35)
10:19.27cinzas|baby|: Portugues. Pero te comprendo
10:19.50|baby|jeje, gracias!
10:19.53cinzasI'm having some strange behavior with asterisk 1.0.8
10:20.36|baby|mientas estoy transfiriendo una llamada usando "atxfer" como puedo cancelarla y recuperar la llamada?
10:21.48X-Robsi
10:22.03cinzasThis is my diaplan
10:22.05cinzasexten => s,1,Answer()
10:22.05cinzasexten => s,2,SetLanguage(pt)
10:22.05cinzasexten => s,4,SayDigits(${LINHA})
10:22.05cinzasexten => s,5,wait(2)
10:22.05cinzasexten => s,6,Playback(linha)
10:22.06cinzasexten => s,7,SayDigits(${LINHA})
10:22.24cinzasAnd this is what happens
10:22.27cinzas-- Executing SayDigits("Local/1234@testes-ea9a,1", "100") in new stack
10:22.27cinzas<PROTECTED>
10:22.27cinzas<PROTECTED>
10:22.27cinzas<PROTECTED>
10:22.36MuppetMaster|babu|:  Have a look here (as I have never used atxfer myself) http://www.voip-info.org/wiki-Asterisk+config+features.conf
10:22.56MuppetMasterI do that with a flash hook on an analog line on a Sipura ATA, works the same.
10:22.58cinzasSometimes it stays correct, but sometimes "asterisk" starts playing the messages in english
10:23.33*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
10:23.37cinzas|baby|: No te puedo ayudar, pues nunca trabajei com atxfer
10:23.56|baby|cinzas no pasa nada ;)
10:26.34cinzasAnyone knows why it changes my Language from pt to en ?
10:27.27*** part/#asterisk MuppetMaster (n=MuppetMa@62.37.170.15)
10:28.30paxr0i have recompiled asterisk for a upgrade from CVS, all appear good ....but has not created chan_zap.so anybody can suggest me why my system haven t chan_zap.c?
10:28.45|baby|cinzas haz un ln a "digits" en ingles al pt... y listo
10:29.25X-Robpaxr0 - you need to download zaptel (and libpri, if needed) from cvs and install that too
10:29.52cinzas|baby|: Si. Esso as una possibilidad. Pero my sistema tiene que funcionar en varias lenguas.
10:30.08cinzas|baby|: y me gustaria saber lo que passa :(
10:30.10paxr0ok X-Rob thanks a lot again :)
10:30.45cinzasI just doesnt understande why it changes the Language ...
10:31.04|baby|cinzas si t sirve de consuelo a mi tambien me ocurre lo mismo que a ti...
10:31.18cinzasHeHe
10:32.38cinzasReturn Code
10:32.38cinzasReturns -1 if the channel was hung up, or if the file does not exist. Returns 0 otherwise.
10:32.46cinzas<PROTECTED>
10:33.12cinzasThe file exists ... and the channel exists too
10:36.51*** join/#asterisk RoyK (n=roy@80.239.107.70)
10:41.32shido6any nagios nuts
10:48.42*** join/#asterisk Hotfurry (n=bharatsa@210.211.246.47)
10:48.50Hotfurryhello there
10:49.29Hotfurryi am facing the problem of relocation error when installing the mysql support to the Asterisk..
10:49.57Hotfurrydoes anybody know about the installing the mysql support for Asterisk
10:50.03Hotfurryplease tell me
10:50.48puppetits just to compile it
10:51.01puppetof couse u need the mysql libs
10:51.05wasimHotfurry: asterisk_addons
10:53.01paxr0# apt-cache search asterisk|grep sql
10:53.02paxr0asterisk-sqlite - sqlite support for the Asterisk PBX
10:53.06paxr0:)
10:55.16*** join/#asterisk tessier (n=treed@wsip-68-15-4-13.sd.sd.cox.net)
10:56.23cinzasHotfurry: what error ?
10:58.37puppet~pastebin
10:58.39jbotextra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/
10:58.49puppetHotfurry: paste your error there
11:09.42Hotfurryok guys
11:09.47Hotfurrylet me paste the error
11:10.08Hotfurryasterisk: relocation error: /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: ast_copy_string
11:10.16Hotfurrythis is the error
11:10.18Hotfurryi am getting
11:10.51Hotfurrycan anybody help me figure out whats the error because of
11:12.18Ahrimanesseems it's unable to find the asterisk include files?
11:16.19cinzashotfurry when it happens ?
11:18.32*** join/#asterisk wmandra (i=wmandra@pcp04943183pcs.verona01.nj.comcast.net)
11:19.59Hotfurrycinzas: this happens when the asterisk is getting loaded..
11:20.44Hotfurrythe res_config_mysql.so is present in the modules directory...
11:20.52*** join/#asterisk maik (n=maik@bfs.cs.uni-sb.de)
11:21.08Hotfurryi installed it from the CVS...
11:21.12*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no)
11:23.00cinzasdo a make clean
11:23.04cinzasand recompile it
11:23.23cinzaslunch time
11:26.33Hotfurryi tried doing that
11:26.42Hotfurrybut still I am getting the same errot
11:26.44Hotfurryerror
11:27.18*** join/#asterisk zotz (n=zotz@24.231.36.100)
11:27.23*** join/#asterisk Inkubot (n=inkubot@200.75.4.7)
11:27.30drumkillawhat version of asterisk are y0ou using
11:27.35Inkubotgood morning
11:31.38HotfurryAsterisk CVS-HEAD-05/12/05-02:17:19, Copyright (C) 1999 - 2005 Digium.
11:35.39drumkillaast_copy_string probably came after 5/12/05
11:35.42drumkillayou need to update asterisk.
11:39.45*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
11:40.01*** join/#asterisk oej (n=Olle@apollo.webway.se)
11:40.02puzzledmorning all
11:46.03*** join/#asterisk scubasteve (n=steve@cpe-071-065-212-199.nc.res.rr.com)
11:46.33mkl1525can anybody give me the name of the voicemail box tool that added the boxes on the filesystem?
11:46.46scubasteveHey guys!  I need some help troubleshooting a SIP registration problem.  I've got two phones outside of my network (one at a family member's house and one in my camper outside, bridged wireless) and both are having issues.
11:47.04oejOutside of a NAT, scubasteve?
11:47.10scubasteveThis started last week when I noticed the Linksys adapter at my mom's was causing stale nonce errors every few seconds.
11:47.25scubasteveoej, yes, but I have a linux firewall that's been fine and unchanged for quite some time
11:47.45scubasteveMy SIP softphone at work was doing the same thing last week... gave up after a while and poof it started working.
11:47.53scubasteveInstalled HEAD over the weekend and it's doing it agian.
11:48.07scubasteveOne adapter gets stale nonce errors all day long, the other doesn't cause * to complain at all.
11:48.15scubasteveIt just registers and registers over and over ...
11:49.01scubasteveI'd like to dump the sip debug to pastebin and see what you folks think
11:50.22scubastevehttp://pastebin.ca/26472
11:54.05*** join/#asterisk paxr0 (n=C@200-85-216-219.bk4-dsl.surnet.cl)
11:59.49scubasteveName/username              Host            Dyn Nat ACL Port     Status
11:59.49scubasteveanybody?
12:01.25drumkillaok, well, you can probably safely ignore the stale nonce messages ...
12:01.34drumkillaas for registering over and over, that is what is supposed to do
12:01.44scubastevedrumkilla, well... I would if the linksys at my mom's house would register.
12:01.46drumkillait re-registers within the negotiated registration timeout
12:02.11scubasteveDrumkilla, I see nonce warnings every 10 seconds or so.
12:02.12drumkillaah, well I didn't catch that part :)
12:02.18drumkillathat it won't reg ...
12:02.23drumkillasilly SIP
12:02.32scubasteveAnd the one on pastebin... doesn't reg at all and causes no warnings or anything on the console.
12:03.48*** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka)
12:04.05scubasteveIt looks like the linksys at mom's house isn't doing the nonce warnings this morning...it is doing the same as the other adapter I put on pastebin
12:04.52*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
12:05.42*** join/#asterisk kshumard (n=root@gateway.digium.com)
12:07.46*** join/#asterisk Lathos42 (n=Lathos42@adsl-69-208-243-229.dsl.lgtpmi.ameritech.net)
12:09.56wmandramorning all
12:10.05paxr0morning
12:10.07scubasteveHey, does the latest CVS HEAD have the new iax2 jb in it?
12:10.11scubasteveOr is that 1.2 ?
12:13.26Guggemandi would guess the latest CVS HEAD is somewhere after 1.2 beta1
12:16.21*** join/#asterisk docE (n=docE@66.237.242.41.ptr.us.xo.net)
12:16.34docEYA BUDDY How's the hurricane?!?!?!
12:17.54scubastevecool.  thanks.
12:18.40docEYour not even here..   go sit in a corner..
12:23.01Inkubotanyone use flash operator panel ?
12:23.11*** join/#asterisk hatems (n=hatems@196.203.15.218)
12:23.12docEnot here..
12:23.17hatemshi all
12:23.24docEhola
12:24.00*** join/#asterisk |baby| (n=jrp@62-43-207-229.user.ono.com)
12:24.00X-RobInkubot - try #amportal - AMP contains FOP
12:24.05mkl1525there is CALLERID and CALLERIDNUM are there also variables like CALLEDID or CALLEDIDNUM?
12:24.21X-Robmkl1525 - CALLEDID == EXTEN
12:24.30X-RobYou're matching that.
12:24.42docEya more or less..
12:24.43InkubotX-Rob: thnks
12:24.59docEUnless your coding a c app then you get all the variables you need..
12:25.21|baby|transfering calls, no ringing sent to caller
12:25.21|baby|I'm transfering an extension or number to,
12:25.21|baby|doesn't hear any ringing. When
12:25.33|baby|Anybody have any ideas how I can resolve this?
12:25.53docEadd a r to your dialplan
12:26.14docEDIAL(Tech/${EXTEN},timeout,r)
12:26.46tzangerI'm pretty sure you shouldn't need 'r' in that case...  what technology?
12:27.05docEYou shouldnt but he asked how to make it produce a ring..  :)
12:27.17tzangerthat's not a very helpful answer...
12:27.52|baby|-- Executing Dial("Local/100@from-internal-3598,2", "SIP/100|50|trR") in new stack
12:27.52|baby|-- Called 100
12:27.52|baby|-- SIP/100-3a02 is ringing
12:28.00tzangerok it's sip
12:28.32tzangerwhat's the 'R' for?
12:28.42tzangerI don't see an 'R' flag in my dial help
12:28.53docEya I dont remember that one either
12:29.11tzangerunless he meant Ttr
12:29.18tzangerinstead of trR
12:29.20|baby|this position the r... and does not generate tone when transferring using to atxfer
12:30.10docETrue..   would allow transfers and make it ring..   but if its not then I dunno..
12:30.19X-Robbaby - that's not how attended transfers work.
12:30.20tzanger|baby|: what is the 'R' for
12:30.28docEMan is it howling outside..
12:30.34tzangerdocE: where are you?  FL?
12:30.38X-Robthey'll be listening to music on hold while you're transferring.
12:30.39docEHurricane party in Tampa!
12:30.51docEI thought that was m
12:31.26tzangeryeah that is m
12:31.33X-Robno, but he's doing an attended transfer.
12:31.37X-Rob(this is what he said)
12:31.48tzangeryeah attended transfer I thought was different
12:31.55|baby|http://archives.free.net.ph/message/20050930.235856.7d2bcce3.en.html
12:32.01|baby|it happens to me just like this person, you excuse my english
12:32.29X-Rob|baby| - your musiconhold is broken.
12:32.57docEMy MOH sucks..   elevator music..
12:32.59tzangerI have not used atxfr myself yet
12:33.03docEIm putting in some light trance..
12:33.11tzangerI think it was blitzrage who had spanish flea for his MOH
12:33.38docEfigures..  :)    Hes crazy like that..  T were you at the conference?
12:33.49tzangerno I wasn't :-(
12:33.56docEman did you miss out..
12:34.00tzangeryeah I bet
12:34.16docEMost of the IRC crew hung out..  Was a real good experience to meet and greet
12:34.19X-Rob|baby|  - attended transfer does not ring. It puts the caller on hold. This is what attended transfer does. fix your musiconhold.
12:34.40tzangerawesome
12:34.51tzangerI go to the toronto asterisk user's meetings on occassion
12:35.11docEWell Im off to work on my companies new PBX I finally talked them into asterisk after a year..
12:36.14|baby|X-Rob how it's possible that a ringtone doesn't sound when I pull an extension number?
12:36.29tzanger|baby|: you're not listening
12:36.46X-Robtzanger - english as a third language.
12:36.47X-Robbe nice.
12:36.50tzanger|baby|: attended transfers put the person you're transferring on hold, and they should hear your music on hold
12:36.53tzangerI am being nice
12:36.56X-Robyou are
12:37.02X-RobI thought you were going to go off at him
12:37.02tzanger|baby|: you need to set up your music on hold
12:37.02X-Rob8)
12:37.08tzangerX-Rob: :-)
12:38.22|baby|X-Rob speak spanish? noo?:(
12:38.35|baby|tzanger ? speak spanish? :P
12:38.40*** join/#asterisk fourcheeze (n=rich@82.153.215.21)
12:38.47X-Rob|baby| -- I'm in Australia. No habla espanol.
12:38.58|baby|ok
12:39.01X-Rob(???)
12:39.01|baby|no problem
12:39.05fourcheezegot interesting router problems
12:39.10cinzas|baby|: If you want i can trasnlate
12:39.16fourcheezetook a couple of Snoms out to a customer's place last week
12:39.18X-Robcinzas - go go!
12:39.24|baby|cinzas bien!
12:39.27fourcheezethey register fine
12:39.29paxr0muy bien
12:39.34paxr0jeje
12:39.38fourcheezebut when I place a call the router drops the ADSL line
12:39.46fourcheezeanyone else had anything like that?
12:39.50cinzas¤0.5  for letter
12:39.51cinzashehehe
12:39.51tzanger|baby|: unfortunately not.  I am learning Romanian though
12:40.05X-Robfourcheeze - yeah. Buggy firmware in the router.
12:40.24fourcheezeX-Rob: ok, firmware is supposed to be up to date, router is a fairly old Vigor
12:40.38fourcheezeis the only option a new router?
12:40.42|baby|cinzas diles que cuando yo tecleo # (atxfer key) el usuario escucha la musica (musiconhold) pero yo al marcar una extension para transferir la llamada no escucho el tono de llamada, ni es escucho nada! en cambio si suena la extension a la que llamo
12:41.12cinzasOk
12:41.52fourcheezeis this an issue of SIP getting through the router or would a different phone help things at all (can't see how it would myself)
12:42.38X-Robfourcheeze - get a linksys WRT54GS with the adsl modem thing built in
12:42.41cinzasAfeter pressing the # key (atxfer key) the user listens to music on hold, but when he starts dialing the extension, so he can transfer the call, there is no dial tone
12:42.42X-Robthey are _the_ best.
12:43.01|baby|cinzas graciasss!! :D
12:43.05fourcheezeX-Rob: is that wireless? They have wireless already on a different router
12:43.06X-Robcinzas - that's correct?
12:43.06cinzas|baby| se marcas bien la extension, la llamada es transferida ?
12:43.22fourcheezethe strange thing is that a similar Vigor router in the office works fine
12:43.26cinzasX-Rob: yeap
12:43.27fourcheezehas anyone else had problems with Vigor?
12:43.43X-Robcinzas - that's the way dialling works. As soon as you start dialing, the dialtone goes away
12:44.06*** join/#asterisk [ASK]ithrynn (n=nick@host217-34-132-179.in-addr.btopenworld.com)
12:44.13|baby|cinzas si, puedo hablar con la extension remota, pero si no consigo contactar con la extension se cuelga la llamada.
12:44.22cinzasX-Rob: No theres is no translation error. He say's that there is no dial tone.
12:44.23[ASK]ithrynnlo
12:44.40X-Robwell that's right? Does he want the dial tone to keep going while he's dialling?
12:44.41cinzasX-Rob: Wait. i think he is confused :)
12:44.44fourcheezeX-Rob: if I set up a vpn from the customer's office to our asterisk server would that likely remove the problem?
12:44.52X-Robfourcheeze - possibly.
12:45.04X-RobBut you may find that the same thing happens
12:45.14X-Roba whole pile of UDP traffic comes through, and the router/modem has a hernia.
12:45.26*** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62)
12:45.35fourcheezeso sending that via vpn would disguise it all
12:45.45cinzasX-Rob: Ok. He saying that if the user doesn't answer the phone, he can't go back and retrieve the call
12:45.47fourcheezeor at least should
12:46.11fourcheezeI suppose it might depend on the vpn technology
12:46.13cinzasMaybe he's doing a blind transfer
12:46.27X-Robcinzas - aaah. OK. I don't know how you do that. I thought that if you hang up the call, the attended xfer should call you back. I don't know. Usually, it's better to use the phone's transfer.
12:46.38wiizardwhat would cause my extensions to report they are busy when i try to dial them
12:46.45cinzasOk x-rob
12:46.57|baby|cinzas k dice? :P
12:47.10X-Robwiizard - are you using AMP?
12:47.13wiizardyep
12:47.14cinzas|baby|: A ver ..
12:47.14X-Robor Asterisk@Home?
12:47.20X-RobYour dialparties.agi is broken
12:47.20wiizardamp
12:47.22wiizardasterisk
12:47.27wiizardsimilar thing ent it?
12:47.36wiizardwhats the crack with asterisk@home
12:47.39cinzas|baby|: Estas usando SIP o H323 ? Los telefonos estan conectados a asterisk como SIP users ?
12:47.48X-Robwiizard - I just committed a fix to dialparties.agi to tell you exactly what _is_ broken
12:47.55|baby|cinzas, SI, son SIP concretamente PAP2 de Linksys
12:47.56X-Robbut that'll be in AMP ..10
12:48.00X-Robwhich is due out in a couple of days
12:48.08cinzas|baby|: Los telefonos no tienen la tecla de transfer ?
12:48.20cinzas|baby|: O lo tenes que hacer com la # ?
12:48.20|baby|tecla d transfer k es? la R ?
12:48.25cinzas
12:48.29|baby|es que la R no funciona
12:48.33X-Robuntil then, you'll need to actually run /var/lib/asterisk/agi-bin/dialparties.agi and see what the error is
12:48.34|baby|la pulso y no hace nada
12:48.35cinzashmmm
12:48.56|baby|hay que activar algo en la configuracion para que funcione la R?
12:49.09nfi|ermesif i would like to take a call arrived to another extension(phone), for example dialing *57, sould be enough an "Answer" command ?
12:49.16cinzasLo que quieres es hacer la transferencia de la llamada. Y si no hai nadien en el otro lado, queires voltar a hablar con el utilizador
12:49.19cinzases esso ?
12:49.32|baby|asi es
12:49.38|baby|transferencia asistida
12:49.39mutilator<PROTECTED>
12:49.56cinzasPues, todoavia no ententei. Da-me um minuto para que le vea
12:50.03|baby|pulsar R o # o lo que sea, marcar extension, hablar con la extension y colgar para transferir. en caso de que no me conteste, poder volver a la llamada
12:50.06X-Robnfi|ermes -- 'show applcation pickup' on your asterisk console
12:50.15|baby|ok
12:50.17|baby|gracias!!
12:52.10nfi|ermesX-Rob, i found very few informationm in the consolle, but thanks for your hel; now i ll go to study PickUp command
12:52.21*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
12:52.23nfi|ermeshel=help
12:52.28X-RobI usually use **
12:52.39nfi|ermesok
12:52.43X-Robexten => _**.,1,Pickup(${EXTEN:2})
12:52.47tzangercinzas: what was the problem?
12:52.55nfi|ermesthx so much
12:54.40synthetiqif u dial ** that will pick up every extension you dialed beginign with the first 2 letters?
12:55.47X-RobNo.
12:55.52*** join/#asterisk wolfson (n=hehe@usr-kdh-208-6-58-26.beachlink.com)
12:56.24synthetiqim interested into finding out how i can get devstate and call pick up to work by pressing the flashign button on snom 360 is someone is on the line
12:56.58X-Robcan't get call pickup yet
12:57.12X-Robyou can get state with CVS and 'HINT'
12:57.25synthetiqyea but uspposedly there is a way to do it with a directed call pick up
12:57.37X-Robwith an ancient version of chan_sip
13:01.00*** join/#asterisk jake1932 (n=jake1932@pool-68-236-60-180.phil.east.verizon.net)
13:02.18cinzasX-Rob: Something like this. I've answered a call. i want to transfer the call. I press de  # key
13:02.41cinzasX-Rob: But i've dialed the wrong number.
13:03.00cinzasX-Rob: How can i pickup the call again ?
13:03.08*** join/#asterisk Meaty (n=cp_simbu@office.abi.ca)
13:04.15|baby|transfer assisted (atxfer)
13:04.24brad_msswIf a number i've dialed a remote number, and they want you to hit the '#' sign after dialing say a conference room number, how do you do that with asterisk.  Hitting # initiates the internal blind transfer feature
13:04.29brad_msswof asterisk
13:07.39*** join/#asterisk ejo1 (n=ejo1@209.32.147.246)
13:08.25encodewhat settings should i check if incoming calls (via DID) dont seem to go anywhere? i just get entries in my log about congestion and absolute timeout
13:10.43docEAnyone know the CVS tag for 1.0.9 stable?
13:11.29JamesDotComnot v1-0-9 or something obvious?
13:12.02docEI tried 1.0.9 and got nothing..   I tried 1-0-9 and said it was an invalid format
13:12.20ReD-MaNanyone ever encountered where if you dial another extension is goes direct to vm saying the person is on the phone?
13:12.23JamesDotComtry what i said then
13:12.43*** join/#asterisk iCEBrkr (n=icebrkr@242858hfc41.tampabay.res.rr.com)
13:13.27Kattymorning.
13:14.26wiizardright ive got a question
13:14.36wiizardwhich is more reliable amp or asterisk@home
13:14.42wiizardor are they infact the same
13:14.48docEsame
13:14.48brad_msswdocE: Just go to the CVSweb interface at : http://cvsweb.digium.com/index.cgi/asterisk/   Then drop down the box with 'All tags / default branch' .. it shows _all_ the CVS tags for asterisk!
13:14.55wiizardyeah thought as much
13:15.04docEthanks brad.
13:15.39brad_msswbut v1-0-9 was correct ...
13:16.04encodeSpawn extension (from-internal, *43, 3) exited non-zero on 'SIP/203-d102' <-- is it bad or ok that spawn extension exited non-zero?
13:16.17jake1932wiizard: they are not the same - AMP is a web interface included in @home
13:16.30docEya forgot the v
13:16.39wiizardok
13:16.41jake1932<PROTECTED>
13:16.42wiizardso if i have asterisk
13:17.05wiizardand if ive installed the Asterisk management portal i technically have asterisk@home
13:17.17docENow if I could find a reliable H323 for stable..   Then I would be set!
13:17.42jake1932wiizard: no - you'd have asterisk + AMP
13:18.06jake1932wiizard: i'm pretty sure asterisk@home has a page that tells you what's included
13:18.35brad_msswdocE: thought openh323 was fairly stable ... just didn't support pass-thru or something, so it would transcode communicating  h323->h323
13:18.36SlickRickhello everybody
13:18.51jake1932http://asteriskathome.sourceforge.net/features.html
13:19.24SlickRickCan I somehow strip some digits from the beginning and from the end of a number?
13:19.27wiizardhmmm looks like asterisk@home isnt as feature packed
13:20.05SlickRickI tried something like ${EXTEN:1:-4} but that didn't work.
13:20.09docEHonestly IMO A@H sucks..
13:20.21docENice for beginners..   but thats bout it.
13:20.50wiizardi
13:20.53jake1932i spent way to much time on @home before realizing I need to understand what's going on
13:20.55wiizardshame AMp is broke
13:21.06*** join/#asterisk paxr0 (n=c@216-155-78-128.bk1-dsl.surnet.cl)
13:21.43Kattyjake1932: beep!
13:22.15jake1932Katty: beep beep
13:24.31Kattyjake1932: you set off my hilight.
13:24.56jake1932Katty: thank you
13:26.47jake1932Katty: good morning
13:27.02Katty^_-
13:27.28tzangerKatty: he thinks you want him.  :-p
13:27.36tzangerwel all know you love me though
13:27.43Kattyoh, that.
13:27.56Kattyyou've both clearly insaned.
13:28.25tzangerDRUNK LIKE A FOX!!
13:28.45docEIm almost waterlogged like one
13:29.35jake1932late sunday night session at the local pub?
13:30.44tzangerheh
13:30.55*** join/#asterisk marc324 (n=marc3234@206-248-133-144.dsl.teksavvy.com)
13:31.58mkl1525when I get a call from external the leading 0 is missing in the shown number and so I can't use the recall function of my sip phone is there an option that causes this behavior (or anything to avoid this)?
13:36.24*** join/#asterisk paxr0 (n=c@216-155-71-234.bk1-dsl.surnet.cl)
13:39.07*** join/#asterisk Tili (n=Tili@211.147.234.5)
13:39.27synthetiqi have a end;ess loop of goto's going how do i kill it?
13:39.55IronHelixmkl125- edit the context whereby calls come in to make the first command SetCIDNum(0${CALLIERIDNUM})
13:40.04IronHelixsoft hangup the channel?
13:41.43*** join/#asterisk SERGEUS (i=sergey@195.112.98.13)
13:42.06SERGEUSHi! is there expirienced AGI users? :)
13:42.16*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
13:42.40SERGEUSI have a problem with AGI, - probably my missunderstanding..
13:43.23*** join/#asterisk _T3_ (n=rposada@53.228.uio.satnet.net)
13:45.11*** join/#asterisk iCEBrkr (n=icebrkr@242858hfc41.tampabay.res.rr.com)
13:45.12jake1932SERGEUS: ask
13:46.16SERGEUSwhen i perfom command "WAIT FOR DIGIT -1"
13:46.22SERGEUSeverything is ok
13:46.47SERGEUS* blocks and wait till i push any button
13:47.11SERGEUSbut when
13:47.46*** join/#asterisk Teeli (n=Tili@211.147.234.5)
13:47.54SERGEUSWAIT FOR DIGIT 10
13:49.13jake1932that's 10 milliseconds
13:49.48Delvardo you want to do a read digits loop?
13:51.21*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
13:53.11encodeok, asterisk is really starting to bug me...i have a sipura 3000 and a softphone, both of them registered to my asterisk box - the softphone can dial out fine, the handset connected to the sipura gets the congested message
13:53.25encodethey're both trying to dial the same number, even in the logs
13:53.46encodewhat on earth is going on?
13:54.12jake1932encode: are they both using the same context?
13:54.37SERGEUSpardon
13:54.43SERGEUSohhhh
13:54.47SERGEUSi got it
13:54.52encodejake1932: yes, from-internal
13:54.55SERGEUSi thougt
13:55.02SERGEUSit was 10 seconds
13:55.20encodeso the sipura phone always gets "Got SIP response 404 "Not Found" back from 202.61.13.40" when dialling
13:55.38SERGEUSthat's why * returned control immediateley...
13:55.38bjohnsonencode: you're just talking about the fxs on the spa correct?
13:55.40jake1932encode: try sip debug?
13:55.55encodebjohnson: yes, analog handset connected to the fxs
13:55.58encodejake1932: ok
13:56.04jake1932encode:  then you'll get more detail as to what's going on
13:56.04SERGEUSjake1932: THANK YOU VERY MUCH :)
13:56.12jake1932SERGEUS: np
13:56.22SlickRickCould I ask again?
13:56.28encodejake1932: what should i look for?
13:56.47SlickRickIs there any way how to strip the last 4 digits from a number?
13:56.50jake1932encode: differences between the two scenerios
13:56.59bjohnsonencode: what does sip show registry display?
13:57.24mkl1525has anybody a snom (360) phone and got the mwi (message waiting indicator) working? when I press the key asterisk doesn't note anything - any help?
13:58.19jake1932SlickRick: should be ${var:-4}
13:58.21*** join/#asterisk aRJAy (n=aRJAy@adsl-130-112.swiftdsl.com.au)
13:58.31jake1932no
13:58.34jake1932that's wrong
13:58.42SlickRickjake1932: no, that returns the only the last 4 digits.
13:58.49bjohnsonencode: nm the registry thing.  sip show peers
13:58.57SlickRickbut I want everything except the last 4 digits.
13:58.58bjohnsonencode: it's likely an authentication issue
13:59.19bjohnsonSlickRick: read the varieable wiki page
13:59.48bjohnsonSlickRick: it's something like $var:0:4
13:59.54encodebjohnson: ok to paste here? or is there a pastebin somewhere
14:00.02aRJAyany fellow aussies out there? :)
14:00.04jake1932~pastebin
14:00.06jbotfrom memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/
14:00.38bjohnsonSlickRick: I can't rmember if the second parameter is the length to take or the number off the right end
14:01.08*** join/#asterisk evangelion (n=manzy_ze@213.199.26.99)
14:01.13evangelionhi all =)
14:01.26jake1932it's the length from the offset
14:01.28encodebjohnson: http://pastebin.ca/26480
14:01.50encodeextension 203 is sipura, 204 is softphone
14:01.51evangeliondoes any of you use asterisk-realtime?
14:02.26festr__i've problem with sending calleridpresentation over IAX2 channel, is it possible this? anyone is using this?
14:02.28*** join/#asterisk azzie (n=az@azzie.net)
14:02.51aRJAyI'm looking to hook myself up with VOIP. Am aware of Asterisk and have only just heard of Asterisk@Home. What's Asterisk@Home like?
14:02.58aRJAyWorth the effort?
14:03.02jake1932no
14:03.15aRJAyjake... why so?
14:03.33SlickRickjake1932: I got that already, but is there no way to get everything except the last 4 digits? in the variable wiki page I found only examples with fixed string lengths.
14:03.36jake1932do you know a linux distro already?
14:03.42aRJAynope
14:03.55SlickRickand ${EXTEN:0:4} doesn't work with realtime.
14:03.57aRJAyfresh machine completely
14:04.09paxr0any asterisk@home user? i need know zaptel modules precompiled in this distro
14:04.20*** join/#asterisk aor (n=bob@209-220.246.81.adsl.skynet.be)
14:04.22jake1932aRJAy: xorcom.com
14:04.31aorHi everybody
14:04.48jake1932aRJAy: and check the docs
14:04.50jake1932~docs
14:04.51jbothmm... docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
14:04.51aordoes someone know if it is possible to do an include with realtime ?
14:04.52paxr0jake1932,  rapid dont have dial and chan modules precompiled?
14:05.15bjohnsonencode: increase verbosity and watch the spa for authentication errors
14:05.17jake1932paxr0: are you sure?
14:05.24aori need this do to dynamicaly dial plan changes according to date/time
14:05.28*** join/#asterisk Toadee (n=pixie@wblv-146-199-121.telkomadsl.co.za)
14:05.35Toadeegreets
14:05.41aRJAyjake1932: will check out that site now..
14:05.42aRJAytah
14:05.49jake1932paxr0: it worked great for me - and was simple to install
14:05.49Toadeequick (complicated?) question
14:05.59SlickRickoh, I take it back, it works with fixed string lengths.
14:06.04paxr0yeah i think , and debian style
14:06.13paxr0but i have problems with modules
14:06.56jake1932paxr0: only reason i switched to debian + asterisk was I was ready to try an install myself on a new machine
14:07.07jake1932paxr0: but rapid worked fine
14:07.07encodebjohnson: the softphone seems to be using NAT and some other stuff (which i havent filled out in the settings) wheras the sipura isnt
14:07.15Toadeei have a factory and showroom connected by a wireless network and a pabx in the factory, what is the preferred method to connect the showroom to the pabx
14:07.47mutilatora wire?
14:07.57Toadeeerm no!
14:08.11jake1932heh - cans and string were abandoned a while ago
14:08.23evangeliondoes any of you use asterisk-realtime?
14:08.24mutilatornot where i work
14:08.43*** join/#asterisk olivier_ (n=olivier_@obs92-4-82-239-116-113.fbx.proxad.net)
14:09.08*** join/#asterisk mmmToop (n=chatzill@196.31.11.194)
14:09.24jake1932mutilator: are you using chan_can_and_wire?
14:09.54jake1932uck - getting worse
14:10.04jake1932and katty didn't even beep me for that
14:10.04aorevangelion, yes I do, well I try :-)
14:10.05FuriousGeorgedoes anyone use sipphone (gizmo) accounts with *
14:10.21mutilatorjust so everyone knows
14:10.22FuriousGeorgei find that mine tend to not want to register
14:10.30mutilatori'm depressed right now
14:10.32mutilator:'(
14:10.33FuriousGeorgebut then they do and its ok for a while
14:10.42mutilatori fscking slept under my office desk last night
14:10.45bjohnsonencode: is the SPA going through NAT to get to the * server?
14:10.46mutilatorgot about 2 hrs of sleep
14:11.04Kattybeep!
14:11.10jake1932haha
14:11.11rocketToadee: not that I have done it with asterisk .. but you could get a pair of wrt54g's and install custom firmware on them that lets you bridge the two together .... this is if you have a small installation of phones you need connected back ..
14:11.37Toadeeyeah that's it
14:11.40*** join/#asterisk gabb0 (n=gabb0@131.202.90.23)
14:11.50Toadeethanks for the nudge in the right direction
14:11.59evangelionaor: how does it works?
14:12.03Kattyhow does one relocate?
14:12.07bjohnsonToadee: another idea
14:12.11Kattythere seems to be this big loop of doom
14:12.21jake1932just do it
14:12.25Kattyjust..how?
14:12.27mutilatorover wifi you need atleast 60% link quality to get one good call
14:12.28rocketToadee: look at dd-wrt for the firmware
14:12.38jake1932different town?
14:12.38Kattymishehu: how does one relocate to chicago!
14:12.44bjohnsonToadee: make a * server just for the showroom.  allows you to keep local calls between showrooms extensions local.  Then trunk between * servers
14:12.57aorevangelion well it does not work too bad, but some features are not available (or I wasn't able to let them work)
14:13.04FuriousGeorgeKatty: depends how big the things you need to take with you are
14:13.08bjohnsonToadee: wifi will limit how many concurrent calls you can make
14:13.16rocketToadee: a wired connection will definately be better though
14:13.23KattyFuriousGeorge: i'll need a moving truck, for sure.
14:13.35FuriousGeorgebigger than a u-haul
14:13.37mutilatori have about 100 phones on wifi right now
14:13.47KattyFuriousGeorge: i'm more caught up in the problem of how do you get a job out there so you can move out there
14:13.54bjohnsonmutilator: on one AP?
14:13.57Toadeebjohn, ta
14:13.57KattyFuriousGeorge: except you need to be there for an interview..
14:13.58mutilatorand in my experience i tell customers no more than 2 lines period and 2 lines are iffy
14:14.04evangelionaor: wich features (i know i'm buggy)?
14:14.05mutilatorbjohnson: network covering nothern michigan
14:14.07evangelion=)
14:14.10FuriousGeorgemutilator: you know 100 people who can cycle a wireless connection all by themselves!
14:14.23bjohnsonmutilator: so not really equivalent to his showroom and factory example
14:14.24mutilatornot ap technology
14:14.31blitzrageKatty: don't do it!
14:14.33mutilatorwell 2 are
14:14.35FuriousGeorgeKatty: offer video conferencing when you send your resume
14:14.46Kattyblitzrage: why not?
14:14.55KattyFuriousGeorge: hmm.
14:14.59blitzrageKatty: not sure... I've been moving a lot in the last 5 years and I'm sick of it :)
14:15.03KattyFuriousGeorge: i'd rather not video conference an interview
14:15.04aorevangelion : well for example, for pickups, it seems that the application don't look in the db to retrive the pu group so you can't do pickups
14:15.32mutilatorbjohnson: i've also setup in office configurations
14:15.34aorevangelion : also some parts must be done in static files, like the zap configuration
14:15.35FuriousGeorgeKatty: shoot, if your trying to get a job in telecom it would probably look good to have your own toll free video conferencing DID
14:15.48blitzragelol
14:15.50KattyFuriousGeorge: i'm not trying to get a job in telecom silly
14:15.51blitzrageFuriousGeorge: true :)
14:15.55evangelionaor: thanks so much
14:15.58blitzragelol
14:16.05mutilatorit's 'reliable' with a great connection
14:16.33FuriousGeorgeKatty: i tend to stereotype people in #asterisk.  what are you getting a job in
14:16.37Kattyif anything i've the most experience in regular windows tech stuff and web design.
14:16.42Kattymcse stuffs.
14:17.23jake1932it's proabaly a little difficult to get a job with the number of MCSEs today then it were a few years back
14:17.23blitzrageKatty: should be able to find a job for that no problem :)
14:17.38blitzragejake1932: really? stuff like that I see jobs for ALLLLLLL the time at places
14:17.42*** join/#asterisk riksta (n=rick@62.6.163.90)
14:17.45blitzrageat least in Canada
14:17.51FuriousGeorgei dunno if chicago is like newark, but here those types of people are everywhere.  ive heard people say that you "lift a rock and find 10" :)  but that just shows theyre making a living
14:17.54aRJAyjake1932: interesting site: www.xorcon.com
14:18.25jake1932xorcom
14:18.41Kattyjake1932: i will go to chicago if i want to. end of story
14:18.41docEI got an interesting question for someone..   :)     I am setting up a PBX for my office.  When I call another extension it works fine.  But when I try to dial outside the office it gives me 484 Proxy Auth Required.
14:18.42jake1932Katty: you can do it!
14:18.43paxr0very easy way to make work debian & asterisk :)
14:18.45docEAny ideas?
14:19.17FuriousGeorgeKatty: i was encouraging you.  im saying there is demand and a lot of people to hook up with, in the IT sense
14:19.26*** join/#asterisk marv[work] (n=timr@border0hsv.asterisksgi.com)
14:19.31KattyFuriousGeorge: yeah, i just can't find them.
14:19.53KattyFuriousGeorge: hopelessly unqualified :<
14:20.23blitzragedocE: that should be normal, then your PBX should try to send authorization based on the information (nonce) that is sent in the Proxy Auth
14:20.42FuriousGeorgethe unqualified paradox:  you cant get qualified till you get a job, and you cant get a job unless youre qualified
14:20.52encodeok, after reading through those debug results
14:21.03encodethe sipura is just getting a 404 not found for the number
14:21.16encodebut the softphone is gettin 407 proxy authentication required
14:21.17docEblitz yes..  But when I call extensions -> extension in the office its fine.  When I call extension -> SIP PSTN it says 484..   Its in the same context..   I dont get it.
14:21.22KattyFuriousGeorge: and throw the whole can't do an interview unless you're in the area and can't be in the area unless you have an apartment and can't have an apartment unless you make enough money to pay for one
14:21.44FuriousGeorgelife sucks sometimes :)
14:22.03Kattymishehu will help me!
14:22.07aRJAyjake1932: is that what you're using?
14:22.08Kattyor maybe hug me
14:22.12aRJAyxorcom ?
14:22.17jake1932<PROTECTED>
14:22.21tzangerxorcom.com doesn't resolve for me
14:22.27tzangerer xorcon
14:22.35tzangerxorcom does :-)
14:22.39encodehow can i get asterisk to send the proxy authentication instead of the handset?
14:22.54jake1932aRJAy: using debian + asterisk HEAD now
14:23.02aRJAyKatty: and what happened? :)
14:23.11aRJAyasterisk HEAD? :)
14:23.12KattyaRJAy: i imploded.
14:23.18jake1932aRJAy: yes
14:23.24*** join/#asterisk ejo1 (n=ejo1@209.98.205.179)
14:23.25aRJAyKatty :)
14:23.35KattyaRJAy: but probably not knowing what the hell i was doing had something to do with that ;)
14:23.47Kattylinux? what's linux?!
14:23.50aRJAyKatty: OIC :)
14:23.53n0rf-HEAD.. the version that broke so many of the former
14:23.59aRJAyLine Ucks
14:24.04bjohnsonencode: you can't
14:24.33bjohnsonencode: why doesn't the spa find the server?
14:24.56aRJAywell.. I have a 'little' experience with linux, but certainly not a confident user
14:25.22aRJAyIs Asterisk 'that' hard to set up??
14:25.27bjohnsonno
14:25.39Toadeeta for the help
14:25.39jake1932aRJAy: especially on debian - really simple
14:25.43bjohnsonespecially if you use atome
14:25.46aRJAyDoes it really need a 3rd party to come along and make a special installer for it and rebrand it?
14:26.04jake1932apt-get install asterisk
14:26.08bjohnsonI don't understand the question
14:26.11Kattyjake1932: eww!
14:26.15Kattyjake1932: don't do that!
14:26.20jake1932(of course that's not a current version)
14:26.21Kattyjake1932: you'll get an insanely old version!
14:26.29encodebjohnson: i've no idea - the sipura was working fine for a different voip provider
14:26.46bjohnsonencode: are they on the same subnet?
14:26.52jake1932but at home isn't the most current either
14:27.00encodebjohnson: yes
14:27.24aRJAyjake1932: if the Asterisk install is simple, why would one deviate from the core s/w and install process?
14:27.36bjohnsonread the wiki page for a spa 2000 and it links to a howto.  the fxs on the 3000 sets up exactly the same as the fxs on 2000
14:27.44bjohnsonencode: you've got some setting wrong
14:27.48jake1932aRJAy: use the recommended way on the digium site - that's the best way
14:27.48Vco'make install' isnt' exactly a ball buster
14:28.09NuggetIt can be tricky!  On some systems you have to type gmake install!  :)
14:28.14encodebut some setting in asterisk? or sipura?
14:28.25tzanger"not exactly a ball-buster"  haaaaaaaaaaaahahahahaha
14:28.42n0rf-aRJAy: just don't use HEAD if you're not that comfy eventually ripping it all back out and reverting to a release version.. which might happen after you searched for a fault on your side that was in fact just bad luck :)
14:28.56aRJAyjake1932: so, your suggestion is now to just get the normal Asterisk instead of xorcom?
14:28.59ejo1Any digium employees online here now?
14:29.09jake1932aRJAy: either
14:29.11docEERIC!
14:29.31ejo1Hey!,  I can't get through to support since 3pm on Friday
14:29.34jake1932aRJAy: xorcom is 1-2-3 simple
14:29.34VcoNo, they're just hitting the local bars as they open PST
14:29.35evangeliondoes res_data project die?
14:29.40aRJAyn0rf-: sounds like it's a bit finiky.
14:29.41Vcoactually that would be 7.30 am....
14:29.52Vcothat is a little early..
14:29.56jake1932aRJAy: the asterisk install over a distro could take a little patience
14:30.04aRJAyjake1932: k... xorcom is well supported? upgrade/update path, etc. ??
14:30.13ejo1I get dead air, on hold? for 40 minutes, then disconnected
14:30.16bjohnsonencode: if no NAT involved, likely a SPA setting issue
14:30.28docEtheir screwing with you
14:30.30jake1932aRJAy: once it's on your system, you should be able to upgrade to a newer version
14:30.33encodeok, got it
14:30.33aRJAyjake1932: I get your point :)
14:30.35ejo1Must be
14:30.42docEThey know its you calling
14:30.46docE:)
14:30.48*** join/#asterisk kippi (n=kippi@host86-133-85-206.range86-133.btcentralplus.com)
14:30.50kippihey,
14:30.54aRJAySo, xorcom is not a 'fork' as such...
14:30.59encodei needed to add "fromuser=<user here>"
14:31.02encodeto the outgoing section
14:31.16encodethanks for your help
14:31.35jake1932aRJAy: rapid (xorcom) includes the debian and asterisk (and a few other tools) - i don't think of it as a fork
14:31.44ejo1Installing a TE410P in an SMP machine.  I get the message in the linux console about "putting 1 in register 31 on span 1"
14:31.47ejo1<PROTECTED>
14:31.53kippiwhen I have a voicemail message and I do *97 to pick up my voicemail it asks for my password. Once I have put my password in it says login in correct. Anyone got any ideas on this?
14:31.58bjohnsonxorcom is a packaging to make install on a new server faster
14:32.23jake1932kippi: check voicemail.conf - do you have the proper password?
14:32.42encodenow i only have my digital receptionist not responding to keypresses when dialed from DID issue
14:32.46encodebut that can wait
14:32.48bjohnsonencode: weird.  I've never needed that
14:33.07aRJAyjake1932, bjohnson: k... I guess when I see that they have their own hardware coming out (not sure if it's available at this point, is it?) then it makes me a little dubious..
14:33.13encodeits bedtime - 12:35 here
14:33.15aRJAyinstinct
14:33.17encodenite folks
14:33.27aRJAyencode: say hi to ya brother for us
14:33.33kippijake1932: I can logon using the web interface
14:33.38encodewhich brother would that be?
14:33.44bjohnsonaRJAy: kind of like Lindows?
14:33.45aRJAydecode :P
14:33.50encodeoh lol
14:33.52Vcoyour brother from anotha mother
14:33.56paxr0rofl
14:34.10encodei dont have a brother whose name starts with de
14:34.15encodebut mine starts with en
14:34.16aRJAytogether, they make codec :)
14:34.18mutilatoror sister from another mister!
14:34.24aRJAyhow sweet ;)
14:34.24encode(well, D and N, but yeah)
14:34.49Vcoi met his cousin vocode once...
14:34.57aRJAyVco: heh
14:34.59*** join/#asterisk fiber0pti (n=johndoe@pcp01876618pcs.sandia01.nm.comcast.net)
14:34.59Vcocouldn't understand a word he said
14:35.20n0rf-ah, the voder family
14:35.20aRJAySo then.. this voip stuff is really new to me.. just been reading up on it a little..
14:35.25aRJAyheh
14:35.35jake1932aRJAy: you can use regular asterisk hardware/software with rapid
14:36.07aRJAyI have an ADSL connection, a spare machine (P3 550) and I wanna have some fun..
14:36.24kippivoicemail.conf seems to have the user login on there
14:36.31jake1932aRJAy: asterisk is not the recommended method
14:36.40vader-wrkdoes anyone in here use asterisk with IP faxing?
14:36.51aRJAyjake1932: why so ?
14:37.15jake1932aRJAy: maybe we have different ideas of fun
14:37.16vader-wrkim looking to hook my main fax line in so it goes through ip to my ip capable fax machine
14:37.21bjohnsonaRJAy: for a home system, there are easier methods
14:37.36mutilatorerm
14:37.44bjohnsonaRJAy: * is more suited a business style phone system
14:37.53mutilatortrying to compile zaptel 109
14:38.07mutilatortells me i need my kernel source installed
14:38.10mutilatorbut its there
14:38.11bjohnsonaRJAy: most people using * at home are tying into other remote systems
14:38.13mutilator/usr/src/linux]
14:38.55aRJAyjake1932: perhaps we do have a different definition of fun...
14:39.19aRJAyI like the idea of having growing flexibility...
14:39.29n0rf-bjohnson: or they're just going for UMS the geeky way :))
14:39.33vader-wrkdoes anyone have their asterisk system tied into a paging system of any sort or is there paging capabilities to asterisk?
14:39.38aRJAySo, I tend to get something that's more than what I "need"
14:41.07hypa7iavader-wrk: http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom
14:41.13aRJAyi believe that there are relatively cost-effective ways of getting into VOIP with existing home telephones, etc...
14:42.24mutilatorgrr
14:42.28evangelionwhere can i find some docs on tables structure to use with realtime?
14:42.28mutilatordamn zaptel won't compile
14:44.14aRJAyI have an ADSL connection (512/128), one phoneline coming into the house with 3 sockets around the house... I want all incoming calls to come through either the regular telephone number which woud then route to the regular telephones... Calls going out I'd like to go through my ADSL line (or drop down to POTS if the system fails, etc..
14:44.37*** join/#asterisk Defraz (n=t0tal@24-119-12-238.cpe.cableone.net)
14:45.18aRJAyVoice mailbox, web-browser interface, ease of use... are all important..
14:45.31aRJAynot sure of the hardware I need... Sipura?
14:45.46aRJAyI need an ATA (I believe)..
14:45.53aRJAyI already have the computer
14:46.14Qwellevangelion: You just add a column for each config setting
14:46.19DefrazSipura is a good ATA
14:46.29Defrazmakes a good ATA
14:46.43aRJAythe 3000 ?
14:46.45{zombie}aRJAy: use a sipura3000 to handle the PSTN line, and hang one of your 3 phones off
14:47.04jake1932can't get rid of the echo on my 3000
14:47.08aRJAy{zombie}: tah
14:47.09{zombie}and either rewire the other two to hang off the back of the sipura, or get another sipura 2000/pap2 to handle those two
14:47.12wmandraarjay: or you could just get a TDM-400P or dev kit from digium
14:47.23{zombie}or ignore those sockets and use real IP phones, or whatever
14:47.28Vcoof course if you're goign to run pstn adn a few phones, you may as well get a TDM
14:47.30evangelionQwell: thank you
14:47.33*** join/#asterisk MuppetMaster (n=MuppetMa@62.37.170.15)
14:48.14MuppetMasterHello
14:48.20MuppetMasterAnyone run Asterisk on AMD 64-bit?
14:48.23aRJAyI don't want to go out and buy new phones.. use existiing phones to start off with.. cost is an issue..
14:49.08marc324keep what you have.
14:49.26file[laptop]MuppetMaster: yes works fine
14:49.31*** join/#asterisk stefanro (n=stefan@ip51cf43f8.direct-adsl.nl)
14:50.55aRJAyI found a SPA3000 for AUD$153 and Digium TDM400b for AUD$110
14:50.56MuppetMasterIs Asterisk optimized for 64-bit?
14:51.09QwellI wonder how many calls a 4 way X2 Opteron could handle...
14:51.30marc324is there a reason why most use xeon as servers?
14:51.44{zombie}aRJAy: does that price for the tdm include any modules?
14:51.45AugheyMM: It's mostly your compiler that will generate more optimized code
14:51.47{zombie}or is that the bare card?
14:52.00*** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com)
14:52.17Qwell{zombie}: for 4 modules, it'll be between 290 and 330 or so, depending on which modules you get
14:52.25aRJAy{zombie}: well.. that's another thing I don't know... what modules would I need with my setup? and why?
14:52.36AugheyMM: The algorithms could be written to take advantage of special processors, but for the most part it won't matter much just dealing with audio
14:52.42wmandraarjay, 1 fxo & 1 fxs
14:52.44MuppetMasterOk, thanks!
14:52.46QwellaRJAy: 1 FXO, and 1-3 FXS.
14:52.58marc324a x100p and a tdm400p
14:52.58blitzragemarc324: probably because Asterisk is usually a CPU intensive application (especially with transcoding) -- so thats probably a good reason to have Xeons if you want to try and put a number of calls on one server
14:53.01Qwell1 if you want all 3 phones to be the same "line", 3 otherwise
14:53.13Vcowhy would you futz it up with a x100p?
14:53.14aRJAymarc324?
14:53.32mishehuugh.  this is annoying...  a system of mine can't even seem to keep one single g729 channel open without dropping packets...   and I only have about 47 ms of ping between the two...  (and the other end is on a frac ds3 circuit)
14:53.33*** part/#asterisk cinzas (n=ashes@62.48.254.104)
14:53.41*** part/#asterisk MuppetMaster (n=MuppetMa@62.37.170.15)
14:53.49wmandraarjay: as long as you don't mind all of the anaolog phones being on the same extension you can get away with 1 fxs, if you want each phone to have it's own extension you need 1 fxs per analog phone
14:54.07marc324blitzrange-- most amd processors outperform xeons.
14:54.24rocketare sipura3000 normally locked .. or if you find em on ebay etc. will they be able to be easily reprogrammed?
14:54.32iCEBrkryeah yeah yeah.. and OS/2 out performs Windows..
14:54.34iCEBrkrBlah blah blah
14:54.37aRJAySo, I'd need a 4 port unit with 1 x FXO and 3 x FXS (one of which could be the fax)
14:54.55blitzragemarc324: yah... but you'll get less problems with Intel chipsets (and I love AMD stuff)
14:55.16marc324oldmyth.
14:55.18aRJAyIs that about riht?
14:55.21aRJAyright
14:55.28QwellaRJAy: pretty much
14:55.28wmandraarjay: right
14:55.38aRJAywmandra: thank you..
14:55.44blitzragemarc324: well, tzanger says he has problems with AMD stuff
14:55.46aRJAyQwell: tah :)
14:56.01blitzragemarc324: and I think he's a smart cookie :)
14:56.01Qwelltzanger just has problems. ;)
14:56.11mishehublitzrage: could it just be the amd motherboards most people use?  I saw some comments about this on saturday's /. article about the new dual core overly power hungry xeons...
14:56.26blitzragemishehu: it is entirely possible
14:56.26aRJAySo then, it's now a matter of chosing between the SPA3000 and Digium 400b/p
14:56.36blitzrageQwell: lol
14:56.41blitzrageQwell: no comment :)
14:56.43marc324marketing does work
14:56.45QwellaRJAy: with the sipura, you'd only have 1 fxs
14:56.58mishehublitzrage: one guy was saying that it is more crucial to follow the memory recommendations of the mobo manufacturer with amd opteron equipment than xeon equipment.
14:57.05wmandraarjay: I would start with the developers kit from digium, it gives you 1 fxo and 1 fxs
14:57.19wmandraarjay: you can always add additional fxs modules later
14:57.21aRJAyQwell: oic :)  that makes the choice slimmer
14:57.22aRJAyheh
14:57.24aRJAyk
14:57.27VcoOR fxo
14:57.30Vco;)
14:57.36rocketwmandra: does the devel kit allow you to pass through the call to the pstn if the server is down?
14:57.51iCEBrkrI've always had CPU issues with AMD.  Intels just work.
14:58.05mkl1525Hi, I get the follow warning message quite often (every 5 seconds) " chan_zap.c:7545 zt_pri_error: PRI: !! Facility message shorter than 14 bytes" anybody knows what goes wrong?
14:58.25iCEBrkrOf course, I have more experience with matching chipsets with Intel CPU's than I do with AMD, so that may have something to do with it
14:58.29*** join/#asterisk Little-L (n=daniel@0x50a471db.svgnxx1.adsl-dhcp.tele.dk)
14:58.46aRJAyiCEBrkr: I've never had any issue with AMD CPUs. I've always stayed away from them for the same reason. 99.99% of software/solutions are developed on Intel
14:58.48marc324icebrkr-- can you be more specific. what os/app was causing problem?
14:58.56wmandrarocket: no it doesn't
14:59.04n0rf-yay, amd vs intel jihad \o/
14:59.05QwellaRJAy: that statement is very untrue
14:59.12aRJAyQwell: oh really?
14:59.13blitzragejihad! :)
14:59.15aRJAylol
14:59.21rocketwmandra: that would make the WAF much better if it did ...
14:59.33aRJAyblitzrage: I'll tip a bucket of holy oil on you ;)
14:59.39*** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net)
14:59.49Vcoscrew the cpu religion crap, it's 10am
14:59.51Vcoand it's monday
14:59.52iCEBrkrmarc324: Um, windows.. In general. Things won't boot consistantly.  They randomly reboot.. THey lock up for no apparent reason.. The list just goes on and on.
14:59.53power1Hey all, ive got a strange problem, I have just set  up and outgoing sip trunk using inphonex as the voip provider, I can make calls and can connect to the party and can chat for about 20 secs, then we are cut off abruptly...any ideas?
14:59.56blitzrageaRJAy: hrmmm... if you're a hot chick, then maybe we can work something out
14:59.57aRJAyblitzrage: it'll be US$54.4/barrel ;P
15:00.09blitzrageaRJAy: but I'm going to assume not since we're in #asterisk :)
15:00.20blitzrageaRJAy: woh... thats cheap / barrel :)
15:00.21iCEBrkraRJAy: I try to keep an open mind.. But any time I venture down the road of AMD, it sucks.  So I go with what works for me. :P
15:00.35wmandrarocket: all the dev kit is, is a TDM400P with 1 fxo and 1 fxs daughter card installed
15:00.35aRJAyblitzrage: let's just stick to a handshake and a *cough* on that
15:00.42iCEBrkrI'm the same way with Seagate vs. Maxstor.  I'm a Segate person.
15:00.48*** join/#asterisk b0xii (i=b0xii@pool-70-110-83-70.dfw.dsl-w.verizon.net)
15:00.56blitzrageaRJAy: fair enough :)
15:01.01aRJAyheh
15:01.14rocketwmandra: ok .. thanks for the info
15:01.17wmandrarocket: i don't even think the cisco vic cards will allow you to do any king of pass-though if the server is down either
15:01.50rocketwmandra: was just looking that the sipura3000 lets ya do that ..
15:02.49aRJAymuch diff between the 400b and the 400p?
15:03.30wmandrarocket: yes true, but it only gives you 1 fxs port
15:03.52rockettrue .. but I guess in my case that is not currently an issue
15:03.57*** join/#asterisk KriS83 (n=KriS@212.202.141.92)
15:04.04KriS83Hi,...
15:04.33rocketwmandra: as I only have one phone in my house currently .. and would probably pick up an sip phone for other extensions later ..
15:04.37mmmToopHi...quick question...can an agent dial when logged in with the AgentLogin() cmd?
15:04.48wmandraarjay: the tdm-40b has 4 fxs modules installed, the 400p is a blank card
15:05.00rocketwmandra: or an iax based phone if there are any that are good
15:05.07KriS83If I needed a 4 Port ISDN Card, could anybody give a suggestion?
15:05.19aRJAyThere is a friend of mine who has a company with one advertised telephone number and a fax number. He's got 5 lines coming into the office to cope with more calls going out/in. How would VOIP help him?
15:05.28docEIs there anyway to turn of digest authentication?
15:05.41aRJAywmandra: ahhh... k :)
15:05.49wmandraarjay: a tdm-31b would give you 1 fxo port and 3 fxs ports
15:06.13*** part/#asterisk Crontibs (n=Cronpars@office-nat.choopa.net)
15:07.35*** join/#asterisk klasstek (n=nunyobiz@sta-206-168-218-202.rockynet.com)
15:08.22Vco4 port isdn?
15:08.35*** join/#asterisk Uther_P (n=uther_p@66.180.120.82)
15:09.34KriS83Vco, Yes.. well all we want to be ablt to do is handle 4 ISDN Lines
15:09.37Vcohttp://www.westbaseuk.com/products.php?cat=24&SID=
15:09.54Vcojust happen ed to see today..never used tho
15:09.56*** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
15:10.07*** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
15:10.54*** join/#asterisk garthvs (n=chatzill@196.31.11.194)
15:10.59KriS83Well we use a AVM B1 already.. but that card can only handle 2 Lines
15:11.17Vcothey have the quad too
15:11.21Vcowhere are you using it?
15:11.27aRJAywmandra: I can't seem to see this tdm-31b you speak of
15:11.28Vco(globally speaking)
15:11.29KriS83Germany ;)
15:11.55Vcoah...wondering what works or doesnt' work in japan
15:11.59Vco:|
15:12.20Augheyarjay: it's a convienent part number.  It's the base tdm card with the combination of fxo and fxs ports you want
15:12.23KriS83anyone have an idea what card I could use with zaptel?
15:12.39KriS83I mean the AVM Cards work with CAPI
15:12.56aRJAyAughey: ahhh.. I see now...
15:12.57aRJAy:)
15:13.03KriS83but from what I have heared zaptal supported cards are better
15:13.57aRJAyok.. sleeeep time here.... nice to chat to y'all
15:13.59wmandraarjay: here is a good link that will help you in getting started with ast... http://www.automated.it/guidetoasterisk.htm
15:14.09tzangerQwell: I'm having problems with what?
15:14.13aRJAyI'm sure I'll be back wit more questions..
15:14.26aRJAywmandra: thanks.. I'll bookmark that now.
15:14.34wmandraarjay: yw
15:15.30*** join/#asterisk mkrufky (n=mk@68.160.103.77)
15:17.00KriS83noone that could suggest me one?
15:17.08aRJAyright... cheers all ~~
15:17.10aRJAy:)
15:17.14aRJAy-=out=-
15:17.31Qwelltzanger: nothing :)
15:20.10*** join/#asterisk AgiNamu (n=AgiNamu@8.7.80.195)
15:20.30mmmToopbye
15:24.05*** join/#asterisk myiagy (n=myiagy@200.138.215.78)
15:25.17*** join/#asterisk brookshire (n=pfffft@gateway.digium.com)
15:25.54AgiNamuHey, anyone have any experience with Cornet or Dataprobe T1 fail switches?
15:26.52Vcook..the word corn and probe just arent meant to go in same sentence
15:29.16*** join/#asterisk Prego (n=dante_em@66.237.242.41.ptr.us.xo.net)
15:29.34AgiNamulol
15:29.42AgiNamuhey, if it keeps my T1 up...
15:29.49Vcohttp://news.cincypost.com/apps/pbcs.dll/article?AID=/20050930/EDIT/509300303/1003
15:29.59Vcosorry i had been reading this the other day and having a good laugh
15:30.14docEPrego its IN there!
15:30.42AgiNamuheh, i lived in guatemala for a few years
15:30.55AgiNamuno fuycking way -- Daniel Todd wrote that!!!! lol
15:30.57AgiNamui know that guy
15:31.21*** join/#asterisk paxr0 (n=jmardoin@200-85-216-205.bk4-dsl.surnet.cl)
15:31.21AgiNamushit what a small world.
15:31.24*** join/#asterisk damned (n=vpol@damned.vpol.org.ru)
15:32.00*** join/#asterisk bweschke (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net)
15:32.13blitzragemorning Mr. Weschke
15:32.30bweschkegood morning
15:33.38AgiNamuvco thats hilarious. made my morning.
15:34.11*** join/#asterisk eivindtr (n=wingnut-@062016241059.customer.alfanett.no)
15:34.12bjohnsonaRJAy: how many people work at your friend's 5 line office and do any of them travel?
15:34.54*** join/#asterisk santiago (n=santiago@208.195.215.158)
15:38.31KriS83What would be to competative of digium to a AVM C4 Card?
15:39.00wiizardX-Rob u still about?
15:39.13*** join/#asterisk kiwnix (n=egarcia@82.158.153.207)
15:44.46blitzrage~thebook
15:44.47jbotsomebody said thebook was Asterisk: The Future of Telephony by Jim Van Meggelen, Jared Smith & Leif Madsen, published by O'Reilly Media. It can be found at http://www.oreilly.com/catalog/asterisk
15:45.10evangelioncan i load sip users on start and then switch them realtime?
15:46.05*** join/#asterisk Igbothom (n=HiltonT@static-84.217.240.220.dsl.comindico.com.au)
15:46.12synthetiqis it possible to hold sip phone registrations in realtime db and not on an asterisk server  (not the sip.conf but the actual phone reg) ???
15:46.23blitzrage~thebook
15:46.24jbotwell, thebook is Asterisk: The Future of Telephony by Jim Van Meggelen, Jared Smith & Leif Madsen, published by O'Reilly Media. It can be found at http://www.oreilly.com/catalog/asterisk for purchase, or FREELY AVAILABLE under the Creative Commons license at http://www.asteriskdocs.org
15:47.00docEsynth, yes..  If you populate your information for your peers/users in the db then all of your registrations will be kept there
15:48.34*** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.cust.bezeqint.net)
15:53.07KriS83Which ISDN card would suite for doing the following: incomming call -> IVR -> forward to local MSN Don't need more than 4 - 6 Lines
15:53.49JunK-Y~astricon2005
15:53.50jbotsomebody said astricon2005 was at http://www.midsouthmarketplace.com/~krice/gallery/view_album.php?set_albumName=album02
15:54.55*** join/#asterisk svenna_ (n=svenna@p548D1D75.dip0.t-ipconnect.de)
15:55.04synthetiqyes docE but how do i go about doing that so far to my knowledge that functioanlity has not been implemented
15:56.15jake1932KriS83: http://www.junghanns.net/en/produkte.html
15:56.45mkl1525When a call comes in via zap interface and I do a NoOp(${CALLERIDNUM}) I get 1753754444 although normally there is a leading zero (01753754444) have tried with and without nationalprefix=0 in zapata.conf but that didn't help - any suggestions where to get the 0 in front of the number?
15:57.13jake1932use setvar with  0${var}
15:58.45*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfl9e.dialup.mindspring.com)
15:58.51AgiNamuanyone here have experience with T1 failover switches?
16:01.36RoyKhm. just a thought... if not having a jitterbuffer for SIP, will forwarding the call through iax2 to a separate box with iax2 jb enabled have any effect on the jitter?
16:01.54mkl1525jake1932, thanks wil try it
16:02.00jake1932np
16:02.14evangelionCan i load realtime data on start\reload or i must wait for a new registration from the ata?
16:03.05*** join/#asterisk viLeR (i=1000@66.128.47.232)
16:03.26jake1932mkl1525: should actually be Setvar(CALLERIDNUM=0${CALLERIDNUM})
16:05.01mkl1525jake1932, have tried it with SetCIDNum(0${CALLERIDNUM} and seems to work, but would that work with foreign callers too that have a 00?
16:05.39jake1932mkl1525: it will append the zero to everything that passes that line
16:06.06jake1932mkl1525: if you want to filter international calls, you'll need to check for that in an expression
16:07.01mkl1525ok thanks for your help!
16:07.23*** join/#asterisk kiwnix (n=egarcia@82.158.153.207)
16:10.58*** join/#asterisk ManxPower (n=ewieling@stirprop-s4-0-0-21.ndcr2.datasync.net)
16:12.29*** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn)
16:14.00nfi|ermes<PROTECTED>
16:14.00nfi|ermes<PROTECTED>
16:14.20nfi|ermeswhere does "128" comes out ?
16:14.25*** join/#asterisk stevek (n=stevek@slim-eth0.horizonlive.net)
16:18.21*** join/#asterisk lmergen (n=Leon@grib.btcnet.nl)
16:18.51lmergenhello, I'm trying to get asterisk to recognize my newly bought TDM400P card
16:19.02lmergenI configured zaptel drivers as should, I think
16:19.15lmergenFXO ports to the outside, so I used fxsks (kickstart) as protocol
16:19.34lmergenno startup problems, kernel message logs goes fine
16:19.40lmergenztcfg -vvv shows 4 channels
16:19.41Augheyhave you run ztcfg -vv?
16:19.45lmergenyes
16:20.04lmergenanyway, in my zapata.conf, under [channels] I configured the 4 channels I want to use
16:20.08lmergensignalling = fxs_ks
16:20.12lmergenand well
16:20.16*** join/#asterisk ian_k (n=ian@gateway.digium.com)
16:20.16*** join/#asterisk fugitivo (n=ajf@209.13.241.231)
16:20.16lmergenwhen starting up asterisk
16:20.18fugitivohello
16:20.23lmergenit just doesn't register the channel "Zap"
16:20.39Augheydid you compile asterisk from source?
16:20.42*** part/#asterisk SERGEUS (i=sergey@195.112.98.13)
16:20.43marc324ne1 can recommend a athlon64 board?
16:20.44lmergenyes I did
16:21.14Augheydid you compile and install the zaptel libraries before compiling asterisk?
16:21.20lmergenno I didn't
16:21.24Augheydo that.  That'll fix it
16:21.30lmergen\o/
16:21.30lmergentx
16:21.33lmergenwill see :)
16:21.35Augheycompile and install zaptel.  then recompile asterisk
16:21.39Aughey(and install)
16:21.59AugheyI had that exact same problem.
16:22.33VcoAsterisk is bascially wondering "What is this 'Zaptel' you speak of?"
16:26.08lmergenmyeah i understand that now
16:26.16lmergenbit since asterisk lacks a `configure` script
16:26.22*** join/#asterisk pif (n=ldm@zenon.apartia.fr)
16:26.22lmergeni figured it always installed everything
16:26.34Augheyit installed everything it knew of
16:26.42fugitivoanyone using asterisk+h323?
16:28.49pifhi, I'd like to have the telco dialtone, how should I Dial() ?
16:29.03pifI'm using bristuff
16:29.49malverian[work]When I do a Dial(SIP/123&SIP/456) it should ring both of them even if one is not available right?
16:29.58malverian[work]As long as one is available it should ring?
16:30.49*** part/#asterisk Primer (n=primus@sh.nu)
16:31.04pifno, if one is busy voicemail shortcut it
16:32.36drumkillamalverian[work]: yes, that is correct
16:32.54malverian[work]pif, Yeah, that's what I'm seeing, but it seems like bad behavior...
16:33.09malverian[work]pif, It wasn't always like that I don't think.. is there some way to circumvent?
16:33.23malverian[work]drumkilla, I'm seeing the behavior pif mentioned.
16:33.57drumkillawell if that's true, then it is a bug
16:34.00malverian[work]For example.. here I know SIP/128 is available.. I can dial it manually and it works fine.
16:34.02drumkillacvs head?
16:34.19malverian[work]drumkilla, From 10/19
16:34.45drumkillatry the latest
16:34.47malverian[work]Anyway, if I do a Dial(SIP/128&SIP/119) and 119 is in "dnd" it says "All channels are busy/congested"
16:34.52drumkillathen, if it still doesn't work, please file a bug report
16:34.56*** join/#asterisk niZx (n=ilt@S0106deadbeefbeef.wp.shawcable.net)
16:34.58malverian[work]Will do.
16:35.03drumkillathanks
16:35.08pifmalverian[work] : use queues
16:35.28pifthey ignore redirects and busies
16:35.42*** part/#asterisk santiago (n=santiago@208.195.215.158)
16:36.17ManxPower*shivver*  *shivver*
16:36.22ManxPowerwinter has come to the south
16:36.51pifwhat arg should I pass to Dial to hear the ISDN dialtone?
16:36.53tzangerManxPower: you just admitted in -users that you used -HEAD and -1.2beta
16:36.58ManxPowermalcolmd, Are you SURE 128 IS available.
16:37.12ManxPowertzanger, I confess!  I confess!
16:37.16tzangerhahaha
16:37.31Renacoranybody know where I can find an app that has reports for calls/queues, etc for asterisk?
16:37.34ManxPowertzanger, We tried it out after the hurricane to fix some call dropping problems.  Only using it on 2 systems
16:37.48tzangerwhat's your impression?
16:37.54ManxPowertzanger, It's beta.
16:38.00ManxPowerSeveral annoying things/issues.
16:38.22ManxPowerI think they have been fixed in CVS-HEAD.
16:38.28VxJasonxVRenacor, voip-info.org has a list with a ton of CDR (and more) analyzers
16:38.47tzangerManxPower: what do you do to test these systems?  I'm just curious, I just run them and see what crops up
16:38.48ManxPowerSpecifially priorityjumping was off by defauilt, also in CVs-HEAD enumlookup REQUIRES a + prefix on the number to look up.
16:39.18tzangeryeah that priorityjumping thing was a mistake commit
16:39.35ManxPowertzanger, I usually do this 1) deploy to home Asterisk server, 2) deploy to the developement Asterisk server, 3) deploy on production servers IN ORDER OF SIZE of the server.
16:39.56tzangerI hear ya
16:40.02tzangerso nothing specific for call generation/load testing?
16:40.10ManxPoweri.e. the 4 person office gets upgraded before the 60 person office.
16:40.14VxJasonxVRenacor, http://voip-info.org/wiki/view/Asterisk+GUI
16:40.15*** part/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV)
16:40.19ManxPowertzanger, not really.  We don't have load issues.
16:40.22*** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV)
16:40.23VxJasonxVwhoops
16:40.28VxJasonxVwrong application to ^W :P
16:41.19lmergenAughey, worked like a charm, tx ;)
16:41.26Augheyno problem
16:41.29tzangerok
16:41.57ManxPowertzanger, we don't run huge systems.
16:42.06tzangerme either
16:42.17ManxPowerReal Estate companies tend to have many small offices, rather than few large offices.
16:42.23malverian[work]Hmm.. nothing changed in CVS since the 19th that would affect this.
16:42.48malverian[work]Just some changes to the Dial() documentation (and the addition of a spelling error "defiend")
16:43.27ManxPowermalcolmd, have you confirmed you can call both phones individually?
16:44.42ManxPower..er... malverian[work]  have you confirmed you can call both phones individually?
16:44.50malcolmdoi :)
16:45.08ManxPowerSomehow I suspect malcolmd would not have that problem 8-)
16:45.21*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
16:46.52malverian[work]ManxPower, One I can, one I can't.
16:47.19malverian[work]ManxPower, The strange thing is that sip debug on the available peer shows nothing being sent to it.
16:47.26malverian[work]ManxPower, Other than OPTIONS for qualify.
16:47.31malverian[work]ManxPower, Nothing when I try to dial it.
16:48.27malverian[work]"sip show peer 128" shows --  Status       : OK (41 ms)
16:48.34ManxPowerWell, if you try calling SIP/123 and it's not registered and SIP/456 and it's in DND, then all channels will be congested.
16:48.46*** join/#asterisk mutilator (n=animenod@65.111.201.79)
16:49.00ManxPowerfile[laptop], thanks!
16:49.05malverian[work]Okay, I'll use the real numbers..
16:49.06ManxPowerIt was 90F when I left TX
16:49.10malverian[work]SIP/128 is registered, as you can see.
16:49.22malverian[work]SIP/112 is in DND
16:49.34ManxPowerCan you Dial(SIP/128)?
16:49.36malverian[work]SIP/310 is not registered
16:49.38malverian[work]ManxPower, Yes.
16:49.46ManxPowerweird
16:49.57malverian[work]ManxPower, But when I do Dial(SIP/128&SIP/112&SIP/310) it doesn't pickup.
16:50.09malverian[work]Says all are congested.
16:50.28malverian[work]I even rebooted the phone (Snom) and it reregistered.
16:51.05malverian[work]ManxPower, the actual dial I'm using has about 30 numbers in it.. It's possible that Asterisk is not doing bounds checking on arguments passed to Dial
16:51.19malverian[work]And it's somehow screwing up the stack.
16:51.25harryvvmalverian[work] what does your extension.conf looks like.
16:51.32Vco<malverian[work]>  is this sip phone to sip phone?
16:51.40ManxPowermalverian[work], Possible, I guess, but I doubt it.  What happens if you dial a shorter list?
16:51.46Vcoi'm having similar problem with an incoming DID
16:51.49Vcowell..2 DID's
16:51.56ManxPowerAnd why not use a queue?
16:52.12ManxPoweryou can do a ringall for queues
16:52.20Vcois a queue resolving the issue or just ignoring a problem?
16:52.23malverian[work]ManxPower, I haven't looked into them actually.. maybe I should ;)
16:52.32malverian[work]Vco, Really?
16:52.39ManxPowermalverian[work], most of the docs for queues are complicated and annoting.
16:52.54ManxPowerBut a simple queue can be very simple to set up
16:53.15pifstroke your own queue and be happy
16:53.27tzangerstroke it to the left
16:53.30tzangerstroke it to the right
16:53.43pifmake it shiny
16:54.12*** join/#asterisk msw (n=msw@rdu-nat.rpath.com)
16:54.27malverian[work]Vco, I'm doing mine through a special macro.
16:54.35rocketI was just curious how many people are using asterisk in their homes?  and what they mainly use if for?  I know it can do a ton .. I am just trying to see what the most useful features are
16:54.59puppetwoot i got bluetooth working ;D
16:55.01ManxPowertzanger, it does appear that forwarding voicemails is broken in 1.2beta
16:55.01*** join/#asterisk justinu (n=j2@72.18.13.48)
16:55.03malverian[work]Vco, Basically I'm defining ring groups in an included configuration file and then using a special "groupring" macro that allows you to specify overflow extension and eventually what voicemail to go to if no one ever answers.
16:55.06puppetits hot ;D
16:55.15puppetno more cellphone taxes ;D
16:56.50malverian[work]Vco, And yes.. I'm using SNOM phones for them, but it also occurs through our PRI.
16:57.03Vcohmm..mines different i guess...
16:57.09Vcobut equally annoying
16:57.30puppetanyone know if its possible to get DTMF to work threw GSM?
16:57.44*** join/#asterisk myiagy (n=myiagy@200.138.215.78)
16:58.02filepuppet: sure, use an out of band dtmf method
16:58.09filelike rfc2833 or info on SIP
16:58.32ManxPowerpuppet, yes, but not inband DTMF
16:59.16puppetManxPower: is inband im talking about
16:59.22puppetManxPower: outband works ok
16:59.42puppetbut i want inband :/ so i dont have to specify each damn number i should be able to call "enter phonenumber to call"
16:59.50ManxPowerpuppet, you cannot get reliable DTMF over ANY compressed codec.  This is not an Asterisk issues, this is just the way codecs are designed.
17:00.08puppetHow does banks etc. work?
17:00.11ManxPowerpuppet, you can do that.
17:00.14puppetyou can call thoose form cellphone
17:00.24ManxPowerpuppet, Um, cell phones use out of band DTMF
17:00.39puppetah like file said :P
17:00.53filecrazy eh?
17:00.54ManxPowerthe carrier converts the OOB DTMF from the phone to DTMF audio when it passes it off the PSTN
17:01.00puppetyes ;P
17:01.04ManxPowerJust like Asterisk does.
17:01.14ManxPowerand all voIP carriers do
17:01.52ManxPowersome carriers do it wrong, but.....
17:02.35*** join/#asterisk orbi (n=orbiwork@65-86-47-114.client.dsl.net)
17:02.51puppetbut is there anyway to reach the out of band dtmf? if chan_bluetooth is recoded some?
17:02.51orbiany idea on how to generate a CTLFile.tlv for a Cisco 7960 phone?
17:03.34Vcoi get incoming DID via SIP, call hits the server, "exten => tokyo1,1,goto(business,s,1)" it goes to the right context and starts the steps...
17:04.05*** join/#asterisk cfrank (n=cfrank@bi01p1.co.us.ibm.com)
17:04.08Vcobut just cycles around and around in the context, dying off and respawning..
17:05.55Vcoif i direct the call to a SIP phone, call comes in , i answer it on the phone, i hear busy on the handset, but the calling party still hears ringing..
17:06.04Vcoi hang up and it starts to ring again..
17:06.54*** join/#asterisk oej (n=Olle@apollo.webway.se)
17:07.16Vcobut according to debug, - Got SIP response 486 "Busy"
17:07.39tzangergod damn P4 2.8GHz is expensive
17:07.56VcoEveryone is busy/congested at this time (1:1/0/0), but this is the line that was ringing..
17:08.00*** join/#asterisk asteriskmonkey (n=phil@HSE-Windsor-ppp211407.sympatico.ca)
17:08.12asteriskmonkeyhey all
17:09.13asteriskmonkeywith a single pri is there a way of splitting of the channels and assigning them context id's?
17:09.20tzangerasteriskmonkey: huh?
17:09.48asteriskmonkeynm: just figured it out :)
17:10.26asteriskmonkeyanother quest5ion though for those that have experince with the cdr system, how do you get it to record accountcodes on incoming calls
17:11.24malverian[work]Can you include one queue in the other and add to it?
17:14.55*** part/#asterisk orbi (n=orbiwork@65-86-47-114.client.dsl.net)
17:15.05*** join/#asterisk jalsot (n=tamas@abacus.eworldcom.hu)
17:15.08jalsothi
17:15.35jalsotdoes anybody use muxmon? for me it doesn't want to do it's job
17:15.40malverian[work]Eg.. "foo" has members SIP/110 and SIP/112. I want "bar" to have whatever "foo" has but also SIP/114
17:16.51*** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com)
17:16.57hardwirefo to the mo
17:17.59Vcocdr is for wussies
17:18.01VcoALT-PRTSCR
17:18.04Vcoya baby
17:19.27lmergenhey, if I want to associate some external functionality (say, manipulating a database) with Asterisk when a caller presses 2 for example, would there be built-in support in asterisk in some way, or shall I start writing my own extension ? :)
17:20.03brookshireagi or realtime
17:20.22lmergenrealtime
17:20.34brookshirewell.. both have database support
17:21.01brookshirerealtime basically puts the extensions.conf into a database
17:21.25brookshireagi lets you extend asterisk so that you can write applications in perl or php much like cgi
17:21.33*** join/#asterisk jets (n=jets@guardian.pmt.org)
17:21.37lmergenoh, i guess we misunderstood
17:21.58lmergenwhat I want is use the extensions.conf, and the for example bind a simple command to a certain action
17:21.59b0xiianyone here having issues with the goiax DIDs?
17:22.11b0xiior is it just me?
17:22.23lmergenand I'm browsing the internet right now (yeah baby!) and only now notice the System() command for dialplans - I assume that will pretty much do
17:22.24lmergen?
17:22.35lmergen(when I connect a proper CGI script to it)
17:22.47brookshire:)
17:23.01brookshirei would find a nice agi tutorial :)
17:23.42syle2got a question for you VOIp PRI , sip termination dudes, if you got a route that is say 10/6 , does that mean calculate 6 second increments after the 10 original seconds or with the 10 original seconds? cause if duration of call was say 36 with 10 would be 36 but without 10 would be 40 in my calculations
17:24.14lmergenbrookshire, k will do that :)
17:24.23lmergenbrookshire, just checking whether it's possible at all :)
17:24.46harryvvHas anyone tried out this wireless voip phone before? looks alot like the samsung wirless flip cell phone.
17:24.48brookshirelmergen: it's just software, of course you can do it
17:24.48harryvvhttp://www.voipsupply.com/product_info.php?products_id=1067
17:24.49brookshire;)
17:25.21lmergeni know, but i was doubting between writing my own extension or that it was already supported... glad to see I can call external CGI scripts, which is just fine by me
17:26.16brookshireit's AGI in asterisk ;)
17:26.19brookshireCGI is http
17:27.21blitzrageAGI tutorial
17:27.26blitzrage~thebook
17:27.27jbot[thebook] Asterisk: The Future of Telephony by Jim Van Meggelen, Jared Smith & Leif Madsen, published by O'Reilly Media. It can be found at http://www.oreilly.com/catalog/asterisk for purchase, or FREELY AVAILABLE under the Creative Commons license at http://www.asteriskdocs.org
17:27.35*** join/#asterisk escribzz (n=escribzz@71.36.229.227)
17:27.40blitzragecheck chapter... 7 I think
17:29.35*** join/#asterisk jtodd (n=jtodd@ti.fox-den.com)
17:29.48blitzragejtodd: Mr. John! (err... Todd :))
17:30.09justinueverytime I hear John Todd, I think of Ferrari's Jean Todt.
17:32.51malverian[work]Anyone?
17:37.20AugheyHere's a quick question: I'm using a PSTN line, and I'd like incoming calls to first ring the secretary phone for 10 seconds before being answered by the automated system.  But I don't want Asterisk to actually answer the channel until the secretary picks up or it gets answered by the automated system.  If I just Dial(Sip/200) without doing an Answer beforhand, will that work?
17:37.48*** join/#asterisk toddf (n=toddf@ns0.fries.net)
17:38.39*** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net)
17:39.20fugitivoAughey: Dial(Sip,200,10) that's 10 seconds
17:40.01*** join/#asterisk greekman (n=alex@host254.209.113.199.conversent.net)
17:42.52*** join/#asterisk areski (n=areski@polar.es6.egwn.net)
17:43.14trelanemark's over a day idle
17:43.23trelaneand my iaxy is toast and noone in customer service is responding
17:43.43brookshiretrelane: did you call support?
17:43.49trelanewith what?
17:43.53trelanemy iaxy=my phone line
17:44.01brookshireahh.. email support@digium.com
17:44.01fugitivosoftphone?
17:44.02trelanewell it's the extension that rings and is dialable anyway
17:44.05trelanebrookshire, already did
17:44.15trelanefugitivo, soundcard isn't set up for it
17:44.25*** join/#asterisk scubasteve (n=steve@cpe-071-065-212-199.nc.res.rr.com)
17:44.38fugitivocellphone?
17:44.45scubasteveDoes anyone know how to make * not native bridge for SIP?
17:45.04fileyou mean not reinvite?
17:45.11trelanefugitivo, don't have one.  Trust me if there was a great solution I'd have it
17:45.15scubasteveI just upgraded to head.. everything appears fine .. but calls between SIP devices have no audio
17:45.16trelanefile, want to do me a favor?
17:45.23filetrelane: like?
17:45.26jetsBROOKS!
17:45.28scubastevefile:     -- Attempting native bridge of SIP/ScubaSteve-20c6 and SIP/Tiffany-5076
17:45.36filescubasteve: that's normal.
17:45.48scubastevefile: I have no audio on calls between SIP devices.
17:45.58malverian[work]app_queue doesn't work how it's intended I don't think...
17:46.00fugitivoscubasteve: nat between them??
17:46.03filescubasteve: separate problem
17:46.05scubastevefile: I have audio on outbound calls through * and calls to AGI and vm..
17:46.07*** join/#asterisk durex (n=ironman@weber.anpa.org.br)
17:46.18scubastevefile: No nat between them.  Nothing has changed except for a new HEAD (today)
17:46.36file"now the days are gone, now you're on your own"
17:46.37scubastevefile: I want to see if I can tell * to stay in the media path
17:46.39*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
17:46.43filescubasteve: canreinvite=no
17:46.47scubastevehm
17:46.50*** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net)
17:47.09durexhi. in this context: exten => _5561.,1,Dial(SIP/${EXTEN}@bdf_out-bsb,30) , the number dialled will be 5561XXXXXXXXX@bdf_out-bsb . How to dial JUST XXXXXXXXX@bdf_out-bsb, and not the 5561?
17:47.20filedurex: ${EXTEN:3}
17:47.22fileer
17:47.24filedurex: ${EXTEN:4}
17:47.30durexfile thank u
17:47.32scubastevefile: I set canreinvite=no and it still bridges..
17:47.32durexwhere I found doc about it?
17:47.39filescubasteve: bridging is normal
17:47.44filescubasteve: repeat after me, "BRIDGING IS NORMAL"
17:47.56filecanreinvite just stops asterisk from going directly
17:47.59scubastevefile: no sound is not normal :-)
17:48.16filertp debug
17:48.36scubastevertp debug returns nothing.
17:48.50fileyou put canreinvite=no in each peer/friend entry?
17:48.52fileand did a sip reload?
17:49.02scubastevefile: yes
17:49.13filedo a sip debug and pastebin it
17:49.22Kattysave me from the windows servers!
17:49.23Kattysave me!
17:50.32malverian[work]!!!!
17:50.38file!!!!!!!!
17:50.45malverian[work]From "show application queue"
17:50.47malverian[work]<PROTECTED>
17:50.58malverian[work]However, instead of doing what it says.. it just NEVER retries, and NEVER exits..
17:51.12*** join/#asterisk shmaltz (n=chatzill@69.28.255.210)
17:51.47*** join/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com)
17:51.54malverian[work]I also set retry=0 timeout=5 in queues.conf and it it just retries infinitely
17:51.56shmaltzanybody here know where I can check the status of CallForwarding on Zap channels (using the default built in extensions of *72 on the zap channel)
17:52.50b0xiiif anyone here has a goiax account can you please test my setup by calling 878201001165    I'd appreciate it.
17:53.49bjohnsonthe LAN equivalent to rebooting
17:54.16Kattybjohnson: :<
17:55.20malverian[work]Seriously.. am I doing something wrong?
17:55.39*** join/#asterisk w0w0 (n=w0w0@130.Red-83-44-178.dynamicIP.rima-tde.net)
17:56.09malverian[work]I have a queue "support" with members SIP/112 and SIP/114, i set timeout to 5 seconds and retry to 0
17:56.23malverian[work]I run Queue(support,t) and the application never returns.
17:58.51malverian[work]And if I run Queue(support,nt) it also never returns.
17:58.55malverian[work]Anyone else have this issue with CVS HEAD?
17:59.59malverian[work]Asterisk does print this out:    -- Nobody picked up in 5000 ms
18:00.07malverian[work]But it still never returns from the Queue() application.
18:01.40malverian[work]Please, I'm about to lose my fucking mind...
18:02.07Kattymalverian[work]: gosh.
18:02.16Kattymalverian[work]: are the color metaphores really required?
18:03.02malverian[work]It's extremely frustrating that I'm having to use this because of some stupid bug in asterisk, and when I try to use this, the documentation is not accurate.
18:05.22*** join/#asterisk evangelion (n=manzy_ze@3ffe:80ee:1e96:c:20e:35ff:fe28:c2b1)
18:05.54malverian[work]I'm really beginning to regret recommending using Asterisk to my employer.
18:06.04malverian[work]We should have just shelled out the 5,000 for a comdial machine.
18:06.59*** join/#asterisk JASON-0 (n=jason@jason.unitz.ca)
18:07.23JASON-0Hi, I'm just wondering if its normal for the sound quality to be a little choppy sometimes in the voicemail and in conversations..
18:07.24mutilatorwell for one.. why are you using head for your pbx?
18:07.28trelanemalverian[work], I've got comdial at work, if this is the only issue you have...
18:08.08*** join/#asterisk loick (n=loick@APuteaux-151-1-54-57.w82-120.abo.wanadoo.fr)
18:08.10malverian[work]trelane, This isn't the only one.
18:08.26malverian[work]trelane, It's things piled on top of things piled yet on top of others.
18:10.20malverian[work]I've had to edit far more code than I'd have prefered to, just to get things as simple as call parking working in a sane manner.
18:10.54asteriskmonkeysilly question if anyone can help... my voicemail keeps writing the files as root how do i change it back to the asterisk user?
18:10.56trelanemalverian[work], the comdial transfers calls at random, puts voicemails on phones which are not even mentioned in the voicemail routing system, the time is never right, voicemail messages disappear, and then there are the silent calls of death
18:11.22*** join/#asterisk macTijn (i=martijn@linda.net.insecure.nl)
18:12.22malverian[work]We have a 15 year old comdial system that is working flawlessly if it weren't for hardware failure..
18:12.55trelanethis is one of the new ones, installed last year
18:12.56malverian[work]It's so old they don't even have replacement parts anymore. Our sync card died so we get frame slips all the time now.
18:13.05malverian[work]Which is why we're replacing it.
18:13.27malverian[work]The point is, things that work out of the box on comdial have taken silly hacks to get working at all.
18:13.38trelanehave you tried 1.0 stable?
18:13.41asteriskmonkeyANYONE KNOW WHY a comedian mail system would be writing the files as root and how to fix it?
18:13.51malverian[work]trelane, Yes.
18:14.00trelanebroken in stable and in head?
18:14.18malverian[work]trelane, On that, I can't even get auto-answer with intercoms working on my SNOM phones. I had to write some code and submit to CVS to get it to work.
18:14.33malverian[work]Luckily they accepted my patch and it's in the trunk now.
18:14.39malverian[work]But it's just one thing after another.
18:14.55asteriskmonkeytwisted , drumkilla anyone of you around?
18:15.09trelanemalverian[work], keep writing code and submitting bugs
18:15.34malverian[work]trelane, I don't have time for that. Our phone server is failing and I need a _working_ machine.
18:15.43shmaltzanybody here know where I can check the status of call forwarding for zap channels that have been forwarded using zaptel and not the dialplan?
18:16.03malverian[work]trelane, Now to be fair, Asterisk is a nice piece of software. And I know I'm using CVS, so bugs are to be expected.
18:16.48trelanemalverian[work], I'm not necessarily defending asterisk, but there's very few things I can say about Comdial without using language I wouldn't use in front of say, my mother.
18:17.30malverian[work]I think the root of my problems is russel's "massive cleanups" on the 19th
18:18.02trelaneI didn't do it ;)
18:18.06trelanegood :)
18:18.47malverian[work]ast_freak, You're probably running asterisk as root.
18:19.29malverian[work]asteriskmonkey, , You're probably running asterisk as root.
18:19.52asteriskmonkeyyes i am :) thanks noticed that in my top program ... how do i change what user it runs as
18:20.03malverian[work]asteriskmonkey, -U <user> -G <group>
18:20.21malverian[work]asteriskmonkey, `asterisk -h` for more information.
18:20.25asteriskmonkeyk
18:21.00sahafeezquestion, i am trying to get asterisk not to try to load mysql support. is that in modules.conf
18:21.07hohumsupport for polycom and cisco hand sets seems good
18:21.14hohumdunno about any other IP handset
18:21.21hohumand dunno about MGCP either
18:21.28hohumall my phones run SIP
18:21.49hohummalverian: what type of phone did you say you were using again?
18:22.12malverian[work]SNOM 320
18:22.15marc324is openssh secure?
18:22.17sahafeezusing, polycom here. works fine. pain to setup the phones on the polycom side
18:22.44malverian[work]I'm using a pretty heavily modified version of app_valetparking to get my orbits working.
18:23.20malverian[work]Is there any kind of regression testing that gets performed on Asterisk builds?
18:23.48shmaltzanybody here know where I can check the status of call forwarding for zap channels that have been forwarded using zaptel and not the dialplan?
18:25.21*** join/#asterisk darkskiez (n=darkskie@host86-133-149-211.range86-133.btcentralplus.com)
18:26.06sahafeezhow do i disable asterisk trying to load mysql realtime on startup?
18:27.27*** join/#asterisk orbi (n=orbiwork@65-86-47-114.client.dsl.net)
18:27.56orbiDoes anyone know where i can find instructions on flashing a Cisco 7912 phone to SIP firmware?  I can't find instructions on google anywhere.
18:28.51*** join/#asterisk groogs (n=greg@d221-73-237.commercial.cgocable.net)
18:29.05groogsp
18:29.34wmandraorbi, try http://voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx
18:30.00*** join/#asterisk arp2 (i=infinite@obscure.metachar.net)
18:31.43orbiwmandra: that's good for The 7940/60s but doesnt seem to work for the 7912
18:32.17wmandrais there even a SIP image available for the 7912
18:32.50orbiyeah
18:33.47wmandratry http://www.cisco.com/en/US/products/hw/phones/ps379/prod_release_note09186a0080443a25.html
18:35.02asteriskmonkeyMy asterisk dosnt seem to want to start with the -U -G args
18:35.44orbinada :(
18:35.48asteriskmonkeydo i have to put it like this asterisk -U asterisk -G asterisk or like this asterisk -U <asterisk> ?
18:36.13*** join/#asterisk MUD (n=MUD@206-248-138-115.dsl.teksavvy.com)
18:36.48malverian[work]-U asterisk -G asterisk
18:36.58asteriskmonkeyok i try again
18:36.59malverian[work]Make sure the user and group exist and have ownership of the directories necessary..
18:37.12malverian[work]asteriskmonkey, Eg /var/lib/asterisk /var/spool/asterisk , etc
18:37.14wmandraorbi: or try http://adisworld.oodi.ca/2005/10/10/using-cisco-ip-phones-with-asterisk/ at the bottom of the page there are instructions for the 7905 which should be similar to the install in the 7912
18:39.09shmaltzanybody here know where I can check the status of call forwarding for zap channels that have been forwarded using zaptel and not the dialplan?
18:41.01asteriskmonkeymalverian : those 2 directories are owned by asterisk th one dir thos /var/spool/asterisk/voicemail/default is owned by root though
18:41.04asteriskmonkeyis that correct?
18:41.40malverian[work]They should all be owned by asterisk
18:41.43malverian[work]Recursively.
18:41.54malverian[work]Also.. /var/run/asterisk (presumably) and /var/log/asterisk need to be owned by asterisk.
18:41.55fugitivoanyone using E1 with MFCR2 ?
18:43.04*** join/#asterisk r0d3nt (i=r0d3nt@66.0.156.250)
18:43.06rikstacan someone please tell me about this error "Oct 23 20:05:31 WARNING[18453]: channel.c:1178 ast_waitfordigit_full: Unexpected control subclass '8'
18:43.09asteriskmonkeymalverian: all the directories are owned by asteisk except thta vm dir, asterisk is running as root atm as i cant get it to start as asterisk
18:43.16asteriskmonkeyanywhere else i should look?
18:44.47malverian[work]chown that directory to asterisk.
18:44.51malverian[work]and then run it as asterisk
18:46.17znoGguys, when a user tries to dial out, i want to ask for a pin and either allow them out or not based on their entered pin. Is it best to do it with a AGI?
18:47.33asteriskmonkeymalvarian: like this chown asterisk:asterisk vm?
18:47.45shmaltzanybody here know where I can check the status of call forwarding for zap channels that have been forwarded using zaptel and not the dialplan
18:47.50fugitivoznoG: no
18:48.10znoGfugitivo: then? how would you do it?
18:48.17fugitivoznoG: Authenticate()
18:48.29znoGusing authenticate?
18:48.30znoGah yes.
18:48.34znoGjust found that on Google :)
18:48.46fugitivo:)
18:49.29shmaltzznoG, how do you plan on controlling the pin?
18:49.33malverian[work]asterisk chown -R
18:49.41malverian[work]UGH...
18:49.47malverian[work]Why is app_queue broken..
18:49.49malverian[work]This is stupid.
18:49.56znoGshmaltz: i'll have to see how Authenticate works first, i thought of just keeping it in a DB, or it just depends on how Authenticate works
18:50.16shmaltzznoG, authencticate is a mess
18:50.29shmaltzyou should either write your own dialplan to handle it
18:50.31fugitivoznoG: well, authenticate is just a simple function, if you want a pin for each user, you'll need to find another way
18:50.41bjohnsonznoG: there is info on the wiki about different ways to do user authentification
18:51.02asteriskmonkeyahah :) for those that dont know how to chown --- chown -R asterisk:asterisk /var/spool/asterisk/voicemail/
18:51.13Augheyjust use a variable for the Authenticate argument.  Load the variable with the user-dependent password
18:51.22shmaltzor you could use the new VMAuthenticate to use voicemail.conf files to reequest the pin
18:51.54bjohnsonyou can do that with the old voicemail, but you have to go into vm first
18:52.10znoGshmaltz: or AGI, right?
18:52.33shmaltzznoG, AGI is only an option as a last resort
18:52.39bjohnsonwhy would you use AGI unles you HAVE to?
18:52.47shmaltzI personaly never use AGI for such apps
18:53.12znoGsee, not everyone in voicemail.conf will have outgoing access..
18:53.15shmaltzI only use AGI for intergration with external DBs for IVR systems
18:53.28znoGi'm looking at http://www.voip-info.org/wiki-Asterisk+user+authentication
18:53.40bjohnsoniirc the vm system will allow the user to change their own pwd
18:53.54shmaltzznoG, so you do it in the dialplan, use the variablies that vmauthenticate returns to figure out to which DISA to drop the user
18:54.11shmaltz~iirc
18:54.12jbotfrom memory, iirc is "if I recall correctly"
18:54.29shmaltzbjohnson, exactly
18:54.42shmaltzthats why I think that authenticate is a mess
18:54.48VxJasonxVAnyone know how to clear up these 'Maximum retries exceeded on call %hash%@ipaddress' warnings?
18:54.50asteriskmonkeyah daminit damnit damit i cant get asterisk to run as root .. is there anywhere else i can find out why not
18:55.07znoGthe idea is that a user dials 12341234 and once dialed, it asks for a pin. If successful, it dials the number, else it says "sorry" and hangs up.
18:55.33stevekVxJasonxV: it means that the other side is dead/unreachable: make the other side respond, and those messages will go away.
18:55.51shmaltzVxJasonxV, yeah, make sure that you have qualify=yes in sip.conf, that way asterisk will know they are not reachable and not try. provided that its a udp timout issue behind nat
18:56.18shmaltzznoG, so this i show you do it:
18:56.18VxJasonxVstevek, and shmaltz I haven't made any calls though...
18:56.35VxJasonxVIt happens any time I start/reload the asterisk process.
18:56.39shmaltzVxJasonxV, then take out the register lines
18:57.04VxJasonxVI only have it registering to gizmo, and that succeeded
18:57.22shmaltzznoG, create the 12341234 exten
18:57.23shmaltzhave asterisk ask for a pin using the vmauthenticate cmd
18:57.31znoGshmaltz: right
18:57.48shmaltzI'm not done yet
18:57.50shmaltzwait
18:58.04znoGshmaltz: 12341234 is any random number (ie. any number starting with 4 longer than 4 digits should be asked for a pin on every attempt to dial)
18:58.24znoGso i guess when you said "create the 12341234 exten" you also meant i can create _4XXXXXX
18:58.27shmaltzznoG, obvoiuslyh
18:59.27shmaltzusing the AUTH_MAILBOX variable route the call to a macro that either denies or allows that user to complete the call
18:59.37VxJasonxVhmmm
18:59.43znoGnice
18:59.46VxJasonxVwell, the register was removed.  I reloaded, and it's still coming up
18:59.48znoGthanks shmaltz
18:59.53shmaltznp
19:00.28mistralanyone know anything about using the sipura 3000 with asterisk ?
19:00.43shmaltzVxJasonxV, then you have a SIP client that is trying to connect to asterisk and asterisk has problems comunicating with it
19:00.43znoGshmaltz: out of curiosity, can the "wait for pin" be customized? in the sense that I want to play a strange tone .gsm (users are used to that with the current system)
19:00.52shmaltzmistral, shoot
19:00.55VxJasonxVAhhhh, it's probably that ip phone
19:01.23mistralits got an fxo port but for the life of me i cannot get it to ring asterisk
19:01.33shmaltzznoG, yes, just use the playback command to play what ever you want, and then use the vmauthenticate with the s option
19:01.58shmaltzmistral, you have to configure a dialplan in the sipura box that it should use
19:02.23znoGshmaltz: ah yes.. the only problem I see is if I want a user to have a password but no mailbox
19:02.24mistralyeah its easyier said than done. Are you fimiluar with it ?
19:02.28shmaltzI always make it into a hotline and configure the s extension in asterisk to do what I want
19:02.40bjohnsonshmaltz: which versions include vmauthenticate?  just head?
19:02.56shmaltzznoG, just specify the mailbox with vmauthenticate
19:03.08shmaltzthat way the user will only be asked for the password
19:03.28shmaltzbjohnson, AFAIK, only HEAD
19:03.55shmaltzmistral, yes I am
19:04.01shmaltzmistral, RTFM
19:04.10mistralyeah i have done 3 times now :D
19:04.17bjohnsonor set up a second exten to catch users without a vm and do a different auth system for them
19:04.30*** join/#asterisk durex (n=ironman@weber.anpa.org.br)
19:04.41bjohnsonread the bottom part of the spa 3000 wiki page
19:04.44shmaltzmistral, how do you think I know, I don't remember it by heart, and I don't have access to a sipura box at the moment
19:04.49bjohnsonuse the forwaring thing
19:04.51durexhi. how to make all connections pass trhough my asterisk?
19:05.06bjohnsondurex: conifrgure them that way
19:05.11bjohnsonand don't use reinvite
19:05.19shmaltzdurex, by wearing slippers
19:05.20durexbjohnson but how?
19:05.32shmaltzRTFM durex
19:05.42shmaltz~docs
19:05.48jboti heard docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
19:05.49bjohnsondurex: you're gonna have to ask a more specific question
19:06.02bjohnsondurex: I don't know if you can't connect at all or you're disconnecting part way through a call
19:06.19*** join/#asterisk acertain (n=acertain@200.119.18.102)
19:06.46bjohnsoneg * can establish a call and then back out of the data flow so the end devices talk to each other directly
19:07.21durexyes, i don't want that end devices talk to each other directly. It will complicate my voip rules.
19:08.02shmaltzdurex, configure your sip settings to have canreinvite=no
19:08.19znoGshmaltz: the annoying thing is the boss just wants users to enter one single pin (no username or mailbox number first)...
19:08.24shmaltzdurex, but I can't see how this will complicate your voip rules
19:08.25*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
19:08.36znoGshmaltz: ie. everyone has their own pin, but they should be able to dial from any phone using one pin, no mailbox number
19:08.44shmaltzznoG, so you supply the usename based on caller id or whatever
19:08.58znoGshmaltz: right, but as i said any user should be able to dial out from any phone using a pin
19:09.10znoGshmaltz: i thought about caller ID, but the user can move around
19:09.11shmaltzznoG, http://www.voip-info.org/wiki-asterisk+cmd+VMAuthenticate
19:09.40shmaltzoh, hold on
19:10.54*** join/#asterisk orbi (n=orbiwork@65-86-47-114.client.dsl.net)
19:11.05shmaltzznoG, for the moment the only thing I can think of: then the mb will have to match the password
19:12.00orbianyone have any ideas why i could pick up the phone, get a dialtone but not be able to dial any extensions (including the demo extension) - if i attempt to dial the dialtone stays w/o registering an input.  Skinny Protocol.   If i hang up i get the on hook notice, waitfordigits <0
19:12.15malverian[work]Hmm.. I think Queue would work better as a channel driver...
19:12.19shmaltzznoG, or the only way to really do it is to use something like authenticate, but then the users wont be able to change their passwords by themselfs
19:12.40malverian[work]Then you can just do Dial(Queue/support) or something..
19:12.48malverian[work]It has so much duplicated code from app_dial it's silly...
19:12.52shmaltzmalverian, nah
19:13.02malverian[work]shmaltz, Why not?
19:13.05VxJasonxVis there some - vs. _ wonkeyness when it comes to context= and context section names?
19:13.07shmaltzthe way it's now is much more flexiable
19:13.29VxJasonxVfor example, subdomains can't have _'s, so you have to translate -'s into _'s?
19:13.34malverian[work]shmaltz, Give me an example of something that couldn't be handled as a channel driver?
19:13.43*** join/#asterisk trentster (n=marktren@rndf-146-5-208.telkomadsl.co.za)
19:13.46shmaltzyou can always use dial(local/whatever) which can point to an extensions doing the queue
19:13.52*** join/#asterisk r0d3nt|m (i=r0d3nt@66.0.156.250)
19:14.31*** join/#asterisk mutilator (n=animenod@65.111.201.79)
19:14.37shmaltzgive me any reason it should be a part of a channel driver?
19:14.48malverian[work]I'm not sure why I brought that up, but I do have a complaint with the queues..
19:14.53malverian[work]Or perhaps it's ignorance..
19:15.06shmaltzshoot
19:15.08malverian[work]What's the best way to have a queue (eg. "support") that has about 10 numbers in it.
19:15.20malverian[work]And then "support-overflow" that will include those 10 numbers and have like 5 more
19:15.24shmaltzyou mean 10 agenst/reps?
19:15.27malverian[work]Without having to just copy and paste the contents.
19:15.54malverian[work]shmaltz, Well, I don't want to overcomplicate it. I just want to use SIP/xxx
19:16.11shmaltzok, but what do you mean by copy and paste?
19:16.24malverian[work]Eg...
19:16.26malverian[work][support]
19:16.31malverian[work]member=>SIP/112
19:16.39malverian[work].... (10 or so more)
19:16.44malverian[work][support-overflow]
19:16.48malverian[work]include => support
19:16.53*** join/#asterisk juice (n=juice@mo-65-40-248-239.dyn.sprint-hsd.net)
19:16.55malverian[work]member=>SIP/192
19:17.00malverian[work]Something like that.. is that possible?
19:20.23shmaltzmalverian, I have no clue it it's possible or not, (the include is for sure not possible), but you could try member=Local/extenpointingtosupportqueue
19:20.40malverian[work]Ahh.. true.
19:20.52trentsterwhats the command to debug asterisk again "asterisk -xxxxxxxxx?
19:21.05shmaltzasterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvvcr
19:21.06malverian[work]vvvv
19:21.08orbii found the answer on how to get a SIP load on a Cisco 7912 and that's just a pain in the ass you wouldn't believe
19:21.09shmaltzc for console
19:21.17shmaltzand r for remote connection
19:21.25orbiso im sticking with Skinny
19:21.31shmaltzorbi, tell me about it
19:22.11trentstershmaltz, thanks
19:22.19sahafeezany issues with head from today.
19:22.44shmaltzsahafeez, tell me
19:23.26orbiok, i pick up the damn phone, it starts simple switch, but if i dial a number it doesnt do anything
19:23.37orbii hang up and it's still waitfordigit <0
19:23.53orbiits like im missing a "Send" button somewhere.
19:24.04shmaltztry the # key after you dial
19:24.18orbiI did
19:24.20orbino dice
19:24.29orbithere are no softkeys or anything
19:25.18Kattyjustrightkeys.
19:25.22malverian[work]Meh...
19:25.27bjohnsonznoG: not much security involved if all it takes is to guess a password
19:25.29malverian[work]I'm just going to fix app_dial, F this :-P
19:25.33bjohnsonnot a username/password pair
19:25.54sahafeezsorry what i was asking any big issues with head from today. i was going to update
19:25.55orbii've been stuck at this point for 2 days now
19:26.04orbi:/
19:26.27orbiI was going to try the chan_sccp module
19:26.34orbibut it crashes when the phones try to register :P
19:26.51*** join/#asterisk kippi (n=kippi@cpc3-hatf3-6-0-cust42.lutn.cable.ntl.com)
19:27.27*** join/#asterisk juice (n=juice@mo-65-40-248-239.dyn.sprint-hsd.net)
19:27.37kippihey
19:28.00sahafeezhow do i disable asterisk from trying to load sql support on startup?
19:28.09sahafeezi have been looking thru the realtime docs but..
19:29.58kippiI have GXP-200 handset but when I logon to my voicemail and enter my password for the mailbox it dosn't want to take it. It seems to take 2 digits and then take the next two. Any ideas what this could be?
19:32.04znoGbjohnson: i agree, that's how it is now though, boss seems to like it this way
19:32.58shmaltzznoG, you can then do hardcoded dialplan without either the authenticate or vmauthenticate to accomplish that
19:33.22shmaltzznoG, beware though that some phones keep these numbers for a redial
19:33.36bjohnsonha
19:33.39bjohnsonthey all would
19:33.54shmaltzbjohnson, not the Polycom or cisco
19:34.15shmaltzonce the stream is esteblished the keys are not rememberd
19:34.16n0rf-sahafeez: check out modules.conf. add noload => [modulename] there
19:34.43shmaltzmost system phones (toshiba, avaya, panasonic) don't remember paswords either for the redial
19:35.08sahafeezn0rf- did that. not there
19:35.22shmaltzsahafeez, of course it's there
19:35.40znoGshmaltz: the phones here at least don't seem to remember keys after the stream is established
19:36.25znoGshmaltz: what do you mean do a hardcoded dialplan? how do I do a simple check to see if the users' "pin" exists in a text file? (this is why I thought using AGI would do the trick, as I just need to find out if a users' pin exists in a file, and since it's not a username/password match, as long as it's in the file is good enough for me)
19:36.51shmaltzznoG, why use a text file?
19:36.54shmaltzuse the DP
19:36.59sahafeezshmaltz: no, its not. not that i see.
19:37.22orbiwhere can i find instructions on making a buttomtemplate file?
19:37.25jalsotdoes anybody use muxmon? for me it doesn't want to do it's job
19:37.25znoGshmaltz: global variables?! or asterisk's internal DB?
19:37.45sahafeezshmaltz: if i have autoload, then i need to spec, which file not to load. i do not know what the file is..
19:37.58sahafeezcalled
19:38.08shmaltzznoG, dialplan, use the dialplan
19:38.26znoGshmaltz: ok, which function exactly to do the pin checking?
19:38.32shmaltzsahafeez, I'm working on it hold on
19:38.41shmaltzznoG, GotoIf
19:38.49znoGshmaltz: if you mean DBGet/DBPut then ok, makes sense.
19:38.56znoG(although they are now deprecated)
19:39.34*** join/#asterisk darkskiez (n=darkskie@host86-133-149-211.range86-133.btcentralplus.com)
19:39.40znoGshmaltz: and where would I store the list of allowed pins?
19:39.44shmaltzznoG, you can use that as well, but you could use hardcoded as well
19:39.54shmaltzsahafeez, its app_realtime.so
19:39.57bjohnsononly if users can't change them
19:40.07VxJasonxVhmmm
19:40.11shmaltzbjohnson,, they can't
19:40.24VxJasonxVthe mysql database meant to hold my CDR (and more?) has no tables
19:40.33sahafeezshmaltz: thanks. brain dead. i was looking for sql type stuff
19:40.42shmaltzVxJasonxV, RTFM its on the wiki
19:40.44VxJasonxVis this an installation step I missed?  or is the program supposed to do that itself if it doesn't find them?
19:40.46VxJasonxVok
19:42.00sahafeezshmaltz: still tries to load the mysql stuff
19:43.07*** join/#asterisk carrar (i=tim@osburn.com)
19:43.09n0rf-sahafeez: there are several applications that can use mysql
19:43.28shmaltzsahafeez, what is the message you getting?
19:44.04sahafeezfigured it out. i just add noload for the name in the error. makes sense now. that you.
19:44.06*** join/#asterisk kiwnix (n=egarcia@82.158.153.207)
19:44.07sahafeezthank you even
19:44.29n0rf-:)
19:44.59carrarAny one in here have a recommended Asterisk server vender who uses Tyan boards?
19:45.02sahafeezI added: noload=app_realtime.so
19:45.02sahafeeznoload=cdr_addon_mysql.so
19:45.27sahafeezi want to setup this up at some point but not untill i get everything working without sql
19:45.31sahafeezthen i will move it.
19:46.02*** join/#asterisk darkskiez (n=darkskie@host86-133-149-211.range86-133.btcentralplus.com)
19:46.03shmaltzgtg
19:46.07shmaltzc ya
19:47.21znoGshmaltz: and where would I store the list of allowed pins?
19:48.04shmaltzznoG, in the dialplan (/etc/asterisk/extensions.conf)
19:48.15shmaltzok, gtg
19:48.17shmaltzc ya guys
19:48.18shmaltzbye
19:49.42znoGi still didn't get what he meant by storing the list of pins IN the extensions.conf file
19:50.56bjohnsonznoG: do a bunch of GotoIf's
19:51.09bjohnsonznoG: one right after the other
19:51.23znoGhaha thats awful
19:51.31znoGbetter go with the DB route i think
19:51.34bjohnsonthe whole concept is awful
19:51.45znoGyea, username/pass would be best
19:51.49znoGusing VMauthenticate
19:51.53bjohnsonyou could do it by including those gotoif's from another file
19:52.25bjohnsonif user's need to be able to change their own passwd via the phone system, voicemail is the only way to currently do it
19:52.55bjohnsonwell, no you couldn't allow that
19:52.58znoGor using AGI :)
19:53.04bjohnsonthe system wouldn't know which pwd to change
19:53.12znoGno, that's right
19:53.29znoGi'm not going to allow them to change their pass, so its ok
19:53.34orbiGuys, any idea why (oh, why) a phone using the skinny protocol wouldnt recognize digits are being dialed?
19:53.55bjohnsonthen a straight text file formated as a context would be easiest to maintain
19:54.24bjohnsonorbi: I don't know skinny, but dmtf sounds like your problem
19:54.47bjohnsonmust be some way to set it on the phones AND in *.  Make sure they are both using the same standard
19:54.59*** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn)
19:56.30znoGi think i'll just have a DB family called PINS, with keys for each pin and a boolean value associated to each one, which shows if they are enabled or not
19:56.53orbibjohnson: hmmm, good thought.
19:57.08znoGbjohnson: this way somebody could add new pins using their phone (an admin, of course)
19:58.17*** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com)
20:00.07carrarWho sells well built servers to use with Asterisk? (hardware)
20:02.25*** join/#asterisk swm_ (n=admin@digitaldatabits.net)
20:03.14*** join/#asterisk jtdintulsa (n=jtdintul@lancer.mbo.net)
20:05.28orbibjohnson:  isn't all DTMF the same DTMF?
20:05.36*** join/#asterisk greekman (n=alex@host254.209.113.199.conversent.net)
20:06.03sahafeezreading the asterisk BOOK. the examples do not match current.
20:06.04trelanecarrar, only one I've really had any dealings with is ATAcomm (www.atacomm.com), he's in here pretty frequently
20:06.06asteriskmonkeycarrar: what do you need
20:06.32bjohnsonorbi: no
20:06.39bjohnsonorbi: not in sip or iax anyway
20:08.39sahafeezin the book there is a file called, enter-ext-of-person. i do not see it or something like it in /var/lib/asterisk/sounds
20:09.18znoGthen create it
20:09.54harryvvcareer you can try Arial. He has sold over 100 asterisk pbx and configured them.
20:10.08carrarok
20:10.26*** join/#asterisk swm_ (n=admin@digitaldatabits.net)
20:10.30harryvvcareer, what is this going to be used for?
20:10.39sahafeezyes, i know that. was looking for canned sounds now. i guess what i am asking is they pulled stuff from the install between the book and HEAD
20:10.44glm2ksahafeez: that sound file is in my /va/rlib/asterisk/sounds dir
20:10.52glm2ker, var/lib
20:10.59carrarwell I just need the hardware big enough to handle two quad cards
20:10.59sahafeezglm2k: what version
20:11.07glm2k1.0.8 HEAD
20:11.21harryvvI see
20:11.29carrarthe software and configuring I will do
20:11.34Corydon-w108 has never been HEAD
20:11.34sahafeezroot@voice-gateway:/var/lib/asterisk/sounds# pwd
20:11.35sahafeezroot@voice-gateway:/var/lib/asterisk/sounds# ls -la enter*
20:11.42harryvvCarrar, how many channels would you be piping though that?
20:11.50glm2kCorydon-w: then HEAD at around that time
20:12.01carrar184?
20:12.01glm2kcirca 2005/08
20:12.14sahafeezah, i am 10/5
20:12.17carrar23*8
20:12.17Corydon-wWhat's the output of 'show version'?
20:12.20harryvvis that sip or sip/zap or what?
20:12.23sahafeezits not there
20:12.32carrarT1's
20:12.38carrarof voice
20:12.42carrarPRI's
20:12.50glm2kCorydon-w: Asterisk  built by jplc@buran.netfone2x.com on a i686 running Linux
20:12.59hohumI wish Digium made DSP boards
20:13.10hohuminstead of leaving it up to the Host CPU
20:13.11tzangerhohum: wy
20:13.19tzangerhohum: that's the entire point of Zaptel
20:13.20hohumI'd be willing to pay extra for the hardware
20:13.26hohumI realize that
20:13.26Corydon-wglm2k: it's been signicantly altered if it's giving you that version string
20:13.29tzangerhohum: go get yourself a dialogic board then :-)
20:13.33tzangerand you WILL pay extra
20:13.34glm2kCorydon-w: aye
20:13.42hohumbut have you ever tried to use an asterisk box as a TDM gateway?
20:13.46hohumit doesn't work too well
20:13.58sahafeezis there an app that will play each .gsm file so i can here each one
20:13.59hohumIMHO
20:14.37hohumand I would run Dialogic if Asterisk supported PRI on any other type of board besides Zaptel
20:14.53glm2kegad, please. no Dialogic
20:14.58tzangerhohum: actually it works pretty damn well for me, and nufone's network is *all* asterisk-to-TDM, and it works very well for them
20:15.24tzangerwhat specific problems do you have?
20:15.31*** join/#asterisk bronc (i=bronc@phalse.2600.COM)
20:15.33tzangeror are you trying to terminate a quadspan of PRIs on a Duron900 or something?
20:15.48hohumno, dual Xeons at the time
20:16.00hohumbut then again it's been a while since I've tried it with Asterisk
20:16.06broncis it just me, or is the iaxclient library not compile any longer?
20:16.07hohumI'm just talking in general
20:16.21tzangerhohum: generally speaking, I disagree with you.  :-)
20:16.25hohumokay
20:16.38hohumand you're entitled to that opinion, I'm not trying to take that away from you
20:16.51hohum:)
20:16.56tzanger:-)  I'm just saying that asterisk as a TDM gateway seems to work pretty damn well for people I work with
20:17.10hohumI don't doubt that
20:17.19tzangerfor large large rollouts I think I'd recommend unlocked MaxTNTs terminating DS3s myself since you could do voice, data and fax on them
20:17.41hohumnever played with thouse
20:17.48carrarthey double as heaters
20:17.50hohumlately I've been tinkering with Brooktrout boards
20:17.54broncahh
20:17.58broncthe Makefile is horked
20:18.05tzangeryes they do spit out a fair amount of heat
20:18.06tzangerand noise
20:18.12broncit never looks to see if you're on FreeBSD
20:18.14broncjust linux
20:18.16hohum8 span PRIs, and 1 port DS3 cards that fit nicely into a Compact PCI chasis
20:18.17tzangerand I consume on average an extra two burgers if I'm lugging the shit around
20:18.21broncwhile the included libs are freebsd aware
20:18.26tzangerbrooktrout boards?
20:18.27broncwhat nice code...
20:18.30hohumyeah
20:18.41hohumhttp://www.brooktrout.com (I think, let me double check)
20:18.53hohumyeah
20:18.55hohumthat's it
20:19.00hohumthey make compact PCI boards
20:19.09hohumthat have a SIP stack
20:19.33*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
20:19.47hohumand some of their products are pretty high density
20:19.56tzangerwhich board?
20:20.01hohumuhm
20:20.35tzangerI don't see anything (quick glance mind you) relating to DS3 or even T1
20:20.44hohumhold
20:20.49tzangerTR1034 is fax and T1
20:21.09hohumhttp://www.brooktrout.com/products/tr2500/
20:21.26tzangerahh there we are
20:21.49tzangerthat's a pretty hefty card
20:22.11tzangerI bet it's got a pretty heft price too
20:22.22hohumI don't know how much we're paying for them
20:22.35hohumthey work well though
20:23.20tzangerI bet
20:23.30*** join/#asterisk swankier (n=swank@S010600080219cb14.vc.shawcable.net)
20:23.32brad_msswwhy is SIP recommended over IAX for faxing?
20:23.50swankierHello channel
20:23.52hohumthey're a hell of a lot less expensive for the port density than alot of things
20:24.02swankierhas there been any status change on zaptel drivers for OpenBSD
20:24.03swankier?
20:24.14trelanebrad_mssw, I don't think iax support has been finished for faxing
20:24.21*** join/#asterisk eKo1 (n=fake@65.169.159.98)
20:24.29hohumand compact PCI cards really aren't *that* costly
20:25.32tzangerhohum: there's always the "compared to what" factor
20:25.43*** join/#asterisk TedC (n=ted@gray.impulse.net)
20:25.45swankieranyone?
20:25.48|dennis|Question: IF i do not do a make asterisk-samples, i will not get any of the config files. What i want to know is which config files are needed  just to get asterisk running with a few sips and a tdm04b card..
20:25.50hohumI don't know, I don't play with media gateways much
20:25.51tzangerswankier: no idea
20:25.56hohumer
20:25.57hohumsorry
20:26.00hohumTDM gateways
20:26.12hohumso I haven't seen much
20:26.12tzanger|dennis|: first answer: stop overoptimizing
20:26.24hohumI've seen really expensive Cisco solutions
20:26.24tzanger|dennis|: second answer: stop overoptimizing
20:26.32tzanger|dennis|: third answer: get it working, THEN wonder
20:26.33hohumin the form of a 250,000 dollar AS5850
20:26.44*** join/#asterisk bmg505 (n=leon@rndf-146-37-221.telkomadsl.co.za)
20:26.57hohumwhich can hold up to 96 PRIs and I think like 12 DS3s
20:27.00tzanger|dennis|: seriously...  a hundred kilobytes of text in /etc is NOT going to kill you.  Get it working, THEN start pissing around getting it small
20:27.38tzangerhohum: yeah that's big..  I generally just fill a half rack with two MaxTNTs, a terminal server and a router/pop box
20:27.41|dennis|tzanger: hmm.....ok...was not really worried about storage...but about what the default config files might leave open in terms of config issues...like connections externally and so on..but yeah...you are right..thanks..
20:27.52swm_XDMCP is a BITCH to setup on lastest Slackware! But the Latest slackware supports all 4 processors in my server! woo hoo
20:28.08bmg505join ##slackware
20:28.13trelaneswm_, umm any linux will support 4 processors
20:28.15tzanger|dennis|: it's all pretty locked down by default
20:28.20bmg505mmmh ja noob alert :)
20:28.30hohumthat's a lucent product?
20:28.33swm_Linux Slackware 8.x would only find two processors
20:28.37tzangerswm_: uh, you mean the latest default kernel.  Slackware supported multiprocessor as soon as the kernel did
20:29.03|dennis|tzanger: thanks...
20:29.35tzangerhohum: well after amalgamations and corporate purchases and all that, yes, I think so
20:30.28swm_Oh I guess that was the situtation.
20:30.41swm_I dont know why they put KDE in the /opt directory now :(
20:30.52swm_I can only get KDM to work and not XDM to work
20:31.59hohumI'm working on implementing a very simple B2BUA
20:32.08hohumwhich supports RADIUS
20:32.23*** part/#asterisk oej (n=Olle@apollo.webway.se)
20:34.29swankierI will purchase an AS5850 fully loaded for anyone who can help me with a wcfxo driver in OpenBSD
20:34.32swankier..... promise
20:34.34swankier:)
20:34.52hohumwhy OpenBSD?
20:35.07swankierbecause it is my preferred platform :)
20:35.54tzangera fully loaded AS5850
20:35.58tzangerthat's a lot of machine
20:36.02broncdamn who wrote iaxclient
20:36.10broncit only compiles on Linux
20:36.15swankieractually, we have a 5850
20:36.18swankierit's just not fully loaded
20:36.25X-RobWhy don't you put a bounty up for it? Will probably be cheaper than $30,000.
20:36.25broncthey claim it compiles on like windows, freebsd, solaris, etc, etc
20:36.29broncand i just tried
20:36.38broncon all of them
20:36.38marc324ne1 runs * on amd processor?
20:36.39bronclol
20:36.57swankierX-Rob: I could hire a developer for most of a year and have it developed as well...
20:37.00bronceven with linux compat mode on freebsd it horks
20:37.03swankierbut I'm a small shop
20:37.05swankierI can't afford that right now :(
20:37.16swankier(for $30,000 that is)
20:37.50hohum30k developer that knows C?
20:37.50hohumI'd like to see that
20:39.13bronchire someone from Ukraine or India
20:39.18broncI do it all the time
20:39.53hohum*shrug*
20:40.03marc324*all the time*
20:40.17broncya
20:40.24broncfor Perl, C/C++ and Java or .net crap
20:40.26X-Robmarc324 - yeps. Wups, he's just hired another one.
20:40.34X-Robuh-oh, there he goes again.
20:40.37X-RobIt's like a nervous twitch.
20:40.53marc324bronc--$/hour ?
20:41.00broncdepends on the job
20:41.07X-Robswankier - well, don't offer $30k worth of hardware.
20:41.11bronci can get a dedicated developer for a month for $2k
20:41.16swankierX-Rob: that was a joke ;)
20:41.20broncdoing web application stuff
20:43.33tzangeryou own't get me for a month for $2k that's for damn sure
20:44.39hohuma week maybe :)
20:44.55tzangerwell maybe ten days.  :-)
20:46.58swankierso the general consensus is... no zaptel drivers on OpenBSD, right?
20:47.28bronci hear the drivers kinda work on *bsd
20:47.32broncbut not very well
20:47.49hohumsome of them work on FreeBSD IIRC
20:47.55hohumbut not all of them
20:47.56bronci saw at astercon that they are making drivers in 1.2 and 2.0 for Freebsd 6
20:47.56hohumYMMV
20:48.56hohumif a driver works in FreeBSD though it isn't guaranteed (and probably won't) work in OpenBSD or NetBSD
20:49.19Igbothomas long as you're not chasing your own tail
20:49.35broncmost likely they have the drivers now working with linux compat
20:49.57broncso you figure you could get it for work on *bsd
20:49.57hohumreally?
20:50.01broncthat's my guess
20:50.01marc324p4 630 vs opteron 142 -?
20:50.12broncsince they said they are working on bsd aware drivers for freebsd 6
20:50.53eKo1marc324: opteron
20:50.57hohumFreeBSD is my platform of preference but I use what ever OS is best suited to the application which I'm trying to run
20:51.06Igbothommarc324; opteron (just) in this case
20:51.12hohumso in the case of Asterisk, I run Linux
20:51.24Igbothomboth are at the bottom end of their ranges
20:52.30hohumwhen I need a good PC-based router (sick, I know) I use OpenBSD
20:53.20marc324p4 630 vs athlon64 3000+?
20:53.43Igbothomcloser again, depending on your needs
20:54.01Igbothomif it is heavy FPU usage, the P4, if not, the Athlon64
20:54.14VxJasonxVGarrr, freaking Grandstream budgetone phone.  Will the asterisk console really not give me a specific reason why my budgetone couldn't register?
20:54.24VxJasonxVI get a generic registration failed, but no reason.
20:54.34*** join/#asterisk [hC] (n=hardcore@8.10.2.42)
20:54.47hohumVx: tethereal/ngrep
20:55.08hohumwatch what the Asterisk box tells the phone
20:56.08VxJasonxVhmmmm
20:56.15VxJasonxVI think I have one of those installed
20:56.51*** join/#asterisk denon (i=denon@synapse.subneural.net)
20:56.51*** mode/#asterisk [+o denon] by ChanServ
20:57.25harryvvmark342 im running the opteron42
20:58.01marc324what board?
20:58.30Rowterfor massive calling .call files would work ok?
20:58.57synthetiqwhat file do u edit to test if u can go over a t1
20:59.03Igbothomas for Opterons, have a look at the Sun Fire X2100, X4100 and X4200 systems - rather nice
20:59.09drumkillaRowter: manager interface is probably a better way to go
20:59.36*** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net)
20:59.53harryvvmarc324 the msi k8t neo master 2 far
21:00.05harryvvbut
21:00.25shido6...
21:00.31harryvvthere are better boards
21:00.40harryvvsome limitations to this one.
21:00.47Igbothomthat's my issue with AMD - the mainboards
21:00.59harryvvIgbothom this one works fine
21:01.04Igbothomwith Intel CPUs it is easy - buy an Intel board, they are rock solid
21:01.26harryvvI would trust this server as a asterisk box.
21:01.37Igbothomit has a decent CPU  :)
21:01.39harryvvIt was designed for 3d animation
21:02.19synthetiqhow do you generate fak phone calls in asterisk
21:02.43synthetiqdigiumw as working on my machine 2 days ago and the guy left whatever making test calls still
21:02.53Igbothomthe Sun boxes look nice - all Opteron-based
21:03.42harryvvIgbothom interesting that sun would go that route
21:04.18Igbothomthey have had some Opteron boxes for a while, but this is the first in-house design based on Opteron - the rest were just re-badged (OEM)
21:05.20*** join/#asterisk gorauskas (n=gorauska@66-224-20-131.atgi.net)
21:06.23bjohnsonsynthetiq: look for .call files
21:06.28bjohnsonor call digium
21:07.42VxJasonxVHey hohum, would you happen to know if a virtual server would prevent you from using either of those tools?
21:08.02harryvvtyan tiger great board.
21:08.14hardwireheh
21:08.14hohumyou have to be logged into root to sniff packets
21:08.15hardwiregrowl.
21:08.21hohums/into/in as/
21:10.45Rowterdrumkilla, why you think so?
21:13.48orbianybody idea why a phone using the skinny protocol wouldnt recognize digits are being dialed?
21:13.59orbithe phone acts like im not even pressing a button
21:15.13*** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net)
21:16.37*** join/#asterisk paryl (n=paryl@209.236.78.59)
21:17.12paryli'm trying to figure out why this isn't working...
21:17.15parylexten => ${ARG1}/${MACRO_EXTEN},1,VoicemailMain(${ARG1})
21:17.15parylexten => ${ARG1}/${MACRO_EXTEN},2,Hangup
21:17.15parylexten => s,1,Dial(SIP/${ARG1},20)
21:17.15parylexten => s,2,Voicemail(su${ARG1})
21:17.15parylexten => s,3,Hangup
21:17.34*** part/#asterisk mkrufky (n=mk@68.160.103.77)
21:17.42paryli'm wanting it to work so that if you dial your own extension you're automatically dumped into voicemail
21:20.07darkskiezparyl: cant you do the voicemail matching in the main context not in the macro?
21:20.52paryldark: well, i guess, but isn't that the whole point of a macro?  to cut down on lines that you have to type?
21:21.22Igbothom:)
21:23.43*** join/#asterisk kippi (n=kippi@cpc3-hatf3-6-0-cust42.lutn.cable.ntl.com)
21:23.47kippihey
21:25.24kippiis there away you can phone a queue
21:25.25kippi?
21:26.06fugitivomake an extension with the queue
21:26.57kippihmm i keep on getting a 499 error when trying to ring any extension
21:29.14fugitivopaste your Dial line
21:30.14*** join/#asterisk iCEBrkr (n=icebrkr@242858hfc41.tampabay.res.rr.com)
21:30.21*** join/#asterisk cripito (n=ncripito@ip67-154-143-190.z143-154-67.customer.algx.net)
21:30.22cripitohi
21:30.50Aughey(sorry if someone already answered this)  Here's a quick question: I'm using a PSTN line, and I'd like incoming calls to first ring the secretary phone for 10 seconds before being answered by the automated system.  But I don't want Asterisk to actually answer the channel until the secretary picks up or it gets answered by the automated system.  If I just Dial(Sip/200) without doing an Answer beforhand, will that work?
21:31.22iCEBrkrAughey: Asterisk has to answer it to do the transfer/call handling
21:31.38sahafeezdo you have to define the parking extention (700) in extensions.conf
21:32.14AugheySo if someone called, and hung up on the first ring, Asterisk would have still answered it and they would get a connection charge
21:32.15cripitoexten => ???, 1, Dial(SIP/200, 10)
21:32.28iCEBrkrAughey: Most likely, yup
21:32.36Augheyhrm
21:32.46sahafeezalso, does the include => parkedcalls have to be in every context you want to use it?
21:32.50cripitoand aughey yes
21:32.54orbianybody idea why a phone using the skinny protocol wouldnt recognize digits are being dialed?
21:33.10AugheyIt would be nice if you could ring an extension, and when it's picked up go to the next extension line and only then do the answer and actually establish the connection between channels
21:33.13Igbothomorbi; dtmf settings?
21:34.00orbiIgbothom: I didn't think cisco phones had multiple DTMF settings?
21:34.04cripitotry the cmd i send u aughey. and put the answer behind that command...
21:34.13cripitoalso u can do someother things
21:34.22Igbothomorbi; dunno, don't have any here
21:34.33iCEBrkrAughey: Asterisk would have to keep the line-voltage pulsing to mimick a ring in order to do that via PSTN
21:34.43orbiIgbothom: is there a place in Asterisk to change the DTMF Settings?
21:35.00Igbothomsip.conf
21:35.15orbiIgbothom: we're using Skinny
21:35.20orbi:~(
21:35.34Igbothomaha - then wherever you configure that
21:35.38orbiok
21:35.40orbitankoo
21:35.52Augheybut the device can detect a ring.  It wouldn't be able to detect a hang-up until it didn't detect a ring within a rings length
21:37.12Augheybecause when it detects a ring, it goes to the s extension.  You answer it with an Answer command.  But you don't have to answer it.
21:38.19Augheyso your first command in the s extension would be a DialButDoNotActuallyConnect command to an extension.  On success or failure of that command the Answer would be done connecting the channels
21:38.40harryvvI thought that asterisk only offers blind calls. obviosly that is not the case.
21:41.38znoGis there a way to keep a counter updated in a Dialplan? ie. i want to ask for a password 3 times, then hang up...
21:42.17denonso, stupid Q, but a 7960 tags everything to * with 802.1p right?
21:43.18*** join/#asterisk Laibsch (n=Laibsch@p54B98F02.dip0.t-ipconnect.de)
21:43.28orbinothing i find in the Skinny.conf indicates where i'd modify DTMF settings
21:43.36orbiany ideas on a reference for Skinny.conf?
21:43.38drumkilladenon: i thought 802.1p was only used in conjunction with 802.1q ...
21:43.58denondrumkilla: not sure ..
21:44.07denonwas just setting up qos on these workgroup switches
21:44.30denontrying to figure out the best way to do it .. 802.1p or dscp
21:44.36drumkillahm, well i guess not
21:44.37denonor by IP port, but that could get ugly
21:44.47LaibschHi.  I understand this question might not be very well liked among you, but what is a good GUI for configuring asterisk on a Debian system to get a system up and running in the shortest time frame?
21:44.48*** join/#asterisk fulgas (n=fulgas@a81-84-116-219.cpe.netcabo.pt)
21:45.08LaibschGUI can of course be web server based.
21:45.29IgbothomLaibsch; AMPortal
21:45.29Augheyvi
21:46.04LaibschIgbothom: Thank you for the hint.  Do you happen to know if that is available for Debian?
21:46.09Igbothomyup
21:46.14denondrumkilla: I usually just prioritize based on MAC, and give it the MACs of the phones .. .but these switches wont do that
21:46.40swm_anyone know of any voip software for the symbian series 60 phones?
21:46.42Laibschhttp://www.sineapps.com/news.php?rssid=1002 -> seems to be.  Cool.
21:46.50swankierhmm... I'm talking to consultants to have them develop and OpenBSD zaptel driver.
21:46.59orbiIgbothom: Is there a good reference for the parameters available in skinny.conf to your knowledge?
21:47.03swankierI'm just gauging interest in it right now
21:47.06swm_anyone know of any voip software for the symbian series 60 phones?
21:47.10denondrumkilla: so you dont think I can rely on 802.1p?
21:47.16Igbothomdunno - I don't use Cisco phones here
21:47.19drumkilladenon: I have no idea what the phone does.
21:47.51swm_Nokia phone, cell phone, but has a OS where you can load programs onto it, also has bluetooth for connectivity....
21:47.55drumkillait supports 802.1q, so i'd think it supported p as well ...
21:48.11drumkillait has been a while since I have looked at that stuff
21:48.11denonany special configs to make it happen?
21:48.16denonor it just magically work?
21:48.37drumkillait's probably an option in the phone, if it has it
21:48.50drumkillasounds like a job for ethereal!
21:49.41justinui know I want a Nokia E61`
21:49.57drumkilladenon: i don't see an option for it in my 7960
21:50.37denonhmm .. these are gig-e switches, maybe I should just assume there wont be an issue
21:50.37denonhehe
21:52.12justinuthe 7960s will use something called CDP (cisco discovery protocol) to figure out which vlan to tag traffic with
21:52.36*** join/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl)
21:52.44harryvvnokia will be comming out with a wimax phone in about two yeas.
21:55.20denonjustinu: yeah, but these arent ciscos switches, so no cdp
21:55.23denonand its all on one vlan
21:55.27denonjust qos prioritizing
21:57.52justinuyeah
21:58.28justinui think you can run cdp off a server, if you want to
21:59.57JunK-Y~astricon2005
21:59.58jbotastricon2005 is probably at http://www.midsouthmarketplace.com/~krice/gallery/view_album.php?set_albumName=album02
22:00.10darkskiezit may be possible to write your own CDP daemon for other switches.
22:00.57*** part/#asterisk darkskiez (n=darkskie@host86-133-149-211.range86-133.btcentralplus.com)
22:01.06*** join/#asterisk darkskiez (n=darkskie@host86-133-149-211.range86-133.btcentralplus.com)
22:01.25*** join/#asterisk glomph (n=black@85.133.18.154)
22:01.55*** join/#asterisk damned (n=vpol@damned.vpol.org.ru)
22:02.36*** join/#asterisk darkskiez (n=darkskie@host86-133-149-211.range86-133.btcentralplus.com)
22:10.20*** part/#asterisk mogorman (n=mogorman@gateway.digium.com)
22:10.44*** part/#asterisk Laibsch (n=Laibsch@p54B98F02.dip0.t-ipconnect.de)
22:18.16asteriskmonkeyis there anywhere i can set what user comedian mail writes the files as?
22:20.18darkskiezpatch app_vm.c
22:20.19darkskiez:)
22:20.43asteriskmonkeydarkskiez: littme more info man :)
22:20.52asteriskmonkeyyou saying there is a patch file i have to compile?
22:21.08darkskiezasteriskmonkey: no, you have to write some code and modify asterisk to do that.
22:21.29darkskiezasteriskmonkey: its adding approx two lines of code i believe for a quickhack(tm)
22:21.42asteriskmonkeycrap, what i dont understand is why after i upraded it writes wavs as root not asterisk anymore
22:21.43FuriousGeorgei noticed today that my fxo sometimes fail to detect a hangup
22:22.02asteriskmonkeytried chowning everything asterisk wont start as user asterisk...
22:22.11darkskiezasteriskmonkey:  probably because you are not running asterisk as user asterisk anymore.
22:22.28darkskiezasteriskmonkey: checked the permissions on the /dev bits and bobs?
22:22.52*** join/#asterisk logicalonline (n=Ken@209.242.52.25)
22:22.52asteriskmonkeydarkskiez: ah no not yet.. will that cause it not to start
22:23.14logicalonlinehello, has anyone gotten the intercom button to work on the aastra 480i?
22:23.27darkskiezasteriskmonkey: probably, what is the error it does starting as not-root
22:23.42asteriskmonkeyit dosnt give me one it simple goes back to cli
22:24.42FuriousGeorgei designed my dialplan so that if one channel was busy it would use another technology to dialout.  obviously i prioritize by whats cheapest.  i noticed today that my fxo werent doing so hot at detecting a hangup, and would just keep going between the call out macros on iax2 and zap/g1 (neither succeding).  im looking at my logs and i notice this:  Ooh, voice format changed to 4     Dunno what to do with control type 15
22:24.52FuriousGeorgewow that was a lot, sorry about that
22:25.12asteriskmonkeydarkskiez: it dosnt give me one.. just goes back to cli is there a command that i can type to get an error msg?
22:25.14*** part/#asterisk puppet (i=puppet@1av10.nu)
22:25.16FuriousGeorgebut what's control type 15?
22:25.50darkskiezasteriskmonkey: su asterisk;  asterisk -cvvvvvvvvdddddddd
22:27.36X-Robdarkskiez - uh. You are aware that verbose and debug only do up to 4, right?
22:27.41X-Robanything beyond that == 4.
22:27.55FuriousGeorgehttp://pastebin.ca/26544 <----  anyway thats my logs.  if you look you see it does go between my macros but isnt calling out like it should when zap is busy
22:27.58darkskiezX-Rob: my keyboard repeat is too high.
22:28.13VxJasonxVhmmm
22:28.27VxJasonxVsomething is causing the asterisk console to be colorless
22:28.31VxJasonxVwhile being run in screen :(
22:28.50X-Robscreen is filtering out the colours
22:28.56X-Robset your TERM to be ANSI before you start screen
22:28.58X-RobNEXT!!!
22:29.01VxJasonxVthanks :)
22:29.50asteriskmonkeydarksiez: seems to work now... i ran it from the user asterisk and runnign as asterisk now.. wierd.. couldnt get it to go under root...
22:29.54VxJasonxVX-Rob, hmmm, ANSI or ansi ?
22:30.03X-Robansi, lower case.
22:30.12VxJasonxVapparently that didn't do it either.
22:30.23VxJasonxVI just did: expert TERM=ansi && asterisk -vvvvddddc
22:30.37X-RobYes
22:30.42[hC]export
22:30.44[hC]not expert
22:30.45VxJasonxVoops
22:30.46X-Robthen TERM=ansi screen -R
22:30.57X-Robnotice how I said 'before you start screen'?
22:30.57VxJasonxVwell, regardless, it didn't error [hC], so your point is moot :)
22:30.59*** join/#asterisk alephcom (n=Miranda@207.34.97.130)
22:31.09VxJasonxVI typed it correctly in the term, just not here :P
22:31.13[hC]ah
22:31.14[hC]:)
22:31.15VxJasonxVindeed I did not.
22:31.46*** join/#asterisk Laibsch (n=Laibsch@p54B98F02.dip0.t-ipconnect.de)
22:31.49darkskiezeditline doesnt seem to work in remote attachment on osx :(
22:31.55*** join/#asterisk Weezey (i=WeezeyD@206.210.109.233)
22:32.12Weezeyanyone have WRT54GP2-NA firmware?
22:32.34LaibschIgbothom: Did you ever install AMPortal on a Debian machine?  The debs seem to be totally borked.
22:32.37VxJasonxVhmmmmmmm
22:33.00VxJasonxVX-Rob, is there anything else a noob may have looked over? :P
22:33.14IgbothomLaibsch; Xorcom Rapid is Debian based, and it was installed on that while I was playing with it
22:33.21*** join/#asterisk swm_ (n=admin@digitaldatabits.net)
22:33.44*** join/#asterisk fifer (n=sirfifer@207.202.227.161)
22:35.16*** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net)
22:35.35LaibschIgbothom: For example they try to overwrite files in /etc/asterisk/.  The current debs from  http://rapid.dotsrc.org/ will not install on a Debian testing system.  Xorcom Rapid sounds nice.
22:36.05Igbothomit did, but the config files are a bit... non-standard
22:36.58LaibschWhat do you mean "it did"?
22:37.14*** part/#asterisk logicalonline (n=Ken@209.242.52.25)
22:37.27LaibschThe server I just tried to set up flat out refuses to install those debs.
22:39.35*** join/#asterisk cp5 (n=samy@adsl-69-110-135-211.dsl.irvnca.pacbell.net)
22:39.40fiferI'm having a logger issue.
22:39.54Weezeyheh, I thought that said, I have a longer issue
22:40.04cp5what signalling types in zaptel/zapata are robbed bit signalling? i can't seem to find any documentation on it
22:40.06Weezeyand I was like, what are you impaling chicks now?
22:40.13fiferI have setup log rotation but after the logs are rotated they are not written to by the logger anymore unless I do a logger reload
22:40.15fiferAny ideas?
22:40.40fiferI'm seeing this on two very diferent * setups, one Debian the other RHEL
22:40.53*** join/#asterisk damned (n=vpol@damned.vpol.org.ru)
22:42.24*** join/#asterisk fugitivo (n=ajf@209.13.241.231)
22:42.57*** join/#asterisk syle (n=blag@unaffiliated/syle)
22:43.53*** join/#asterisk sapo_original (n=Fenix@200.138.76.58)
22:44.02*** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com)
22:45.18LaibschIgbothom: Does Xorcom Rapid use the debs from http://rapid.dotsrc.org/?  I doubt it.
22:45.31Igbothomas do I  :)
22:45.56LaibschCan you please explain what you meant by "it did"?  What is it?
22:46.24justinucp5: e&m, fxo, fxs, etc.
22:46.45sapo_originalhi guys
22:46.47cp5justinu, grazi
22:47.09sapo_originalanybody knows dan, the onwer the project diax?
22:47.30justinucp5: basically anything other than ISDN PRI/SS7
22:48.03*** join/#asterisk lters (n=lters@mrtcdsl-034.mis.net)
22:48.20cp5justinu, awesome, thanks. is e&m ALWAYS robbed bit?
22:48.25justinucorrect
22:48.32cp5ah
22:50.19enderwhat is 'robbed bit' ?
22:51.16justinuit's when you take bits away from the audio path and use them as signalling bits
22:53.00sapo_original???
22:54.36ltersit is a technology for tl signalling
22:56.04*** join/#asterisk fugitivo (n=ajf@209.13.241.231)
22:58.37enderinteresting.  We're using em_w to communicate w/ a Fujitsu PBX
23:00.14|baby|alguien habla español?
23:02.24*** part/#asterisk darkskiez (n=darkskie@host86-133-149-211.range86-133.btcentralplus.com)
23:03.12eKo1|baby|: Si.
23:04.16*** join/#asterisk SplasPood (n=sp@brooklyn.paravolve.net)
23:04.18eKo1robbed bit == inband
23:04.27|baby|eKo1 sabes si existe el patch bristuff para el ultimo asterisk del CVS?
23:05.30znoG|baby|: si, que pasa?
23:06.40|baby|eso, k me falla el atxfer cuando pulso la tecla para transferencia asistida, y marco la extension... si no contesta se cuelga la llamada en proceso y no puedo recuperarla
23:07.03|baby|uso el bristuff-0.2.0-RC8f-CVS
23:07.13eKo1No uso bri
23:07.32|baby|ok :(
23:07.36|baby|y como haces para transferir llamadas?
23:08.02*** join/#asterisk screenname (n=appdev@pcp0011162839pcs.pennington.tn.nash.comcast.net)
23:08.06eKo1pues, los teléfonos tienen un botón que dice 'transfer'
23:08.21|baby|yo uso PAP2 con un telefono normal
23:08.26|baby|la tecla R no funciona tampoco
23:08.28eKo1yo lo oprimo, marco el número, y viola, se tranfiere la llamada.
23:08.41znoGprobaste con la tecla numeral, baby?
23:08.53|baby|znoG #?
23:08.56znoGsep
23:09.04|baby|si, pero yo tengo en features.conf
23:09.10eKo1puede ser que el pap2 no puede hacer transferencias
23:09.12|baby|puesto atxfer => #
23:09.22*** part/#asterisk Laibsch (n=Laibsch@p54B98F02.dip0.t-ipconnect.de)
23:09.28|baby|y cuando la marco, funciona...
23:09.44|baby|pero si la extension a la que llamo no contesta... no puedo recuperar la llamada :( se cuelga!
23:10.33Vcoumm.....so...Never had a working H323 channel before...
23:10.42Vconow it worked right out of the box...
23:10.53justinuneat
23:10.59Vcohow the hell do i restrict access?
23:11.33justinunever worked on h323 before
23:11.44FuriousGeorgeisnt there some way to allow zap channels to initiate a transfer with a #
23:11.56Vcoflash
23:11.59Vcono?
23:12.07cripitowow!! #asterisk in spanish?
23:12.12FuriousGeorgeflash isnt a pound
23:12.25|baby|cripito yeahh! XD
23:12.53justinunosotros tenemos mas influencia con sus hijos que tu tiene
23:12.54FuriousGeorgeay dios mio, el asteric bilinguo
23:12.59Vcoyea..just pound...ext....pound
23:13.00justinupero los queremos
23:13.09cripito:) bueno si la cosa es en espa~Nol ok
23:13.10cripito:P
23:13.11justinucreado y gegalo de los angeles, juana's addiccion!
23:13.14FuriousGeorgetu tienes* justinu
23:13.36justinui'm surprised that was my only mistake
23:13.44viLeRyo no sabia que había tanto hispanoparlante por aca
23:13.52|baby|joer...
23:13.52justinuno habla espanol
23:13.54cripitopues parece q hay una buena cantidad :))
23:13.56|baby|no si ahora todos hablremos español
23:13.57FuriousGeorgey yo tampoco
23:13.58|baby|XDDDD
23:14.00Vcobut seriously, this H323 allows anyone to call through the server
23:14.05Vcowhich i kinda dont' like...
23:14.41cripitowell definitelly the spanish comunity in the asterisk is more big that the ppl think at the first time ;)
23:14.44cripitoany way...
23:15.16FuriousGeorgeVco: de ahora en adelante no se hable el ingles aca.  guerra popular!
23:15.26FuriousGeorgewe can be zapatistas
23:15.49|baby|jaja
23:16.00|baby|nadie usa atxfer para transferir llamadas??? :(
23:16.29FuriousGeorgei use regular transfer on zap and sip channels
23:16.42FuriousGeorgewas just wondering aloud if there was some way to initiate a transfer using # on zap
23:16.49cripitome 2..
23:16.54Vcoyea..
23:17.00Vco# dial ext and # again
23:17.08FuriousGeorge|baby|: por que?
23:17.14cripitobasically the phones have support for that...
23:17.35FuriousGeorgecripito: Vco:  the phones need to support # transfer?
23:17.51FuriousGeorgeas in # flashes the line when its off the hook?
23:17.54Vcofar as i know it was just handled by the fxs
23:18.09FuriousGeorgethat cant be
23:18.09|baby|el telefono lo soporta, porque la llamada se transfiere si el otro contesta...
23:18.09justinucould be
23:18.15|baby|el problema es cuando no contesta nadie, que la llamada se corta
23:18.22Vcolike, it's a zap thing..is it not?
23:18.29|baby|con attended transfer unicamente
23:18.32|baby|*2
23:18.37|baby|;atxfer => *2 ; Attended transfer
23:18.41justinuanyone need or is interested in RTCP support in asterisk?
23:18.48eKo1maybe you should flash them before to make sure they are there
23:19.03Vcoor you mean attended xfer?
23:19.39cripitogeorge incluso in soportar el #
23:19.42FuriousGeorgei think means "not a blind transfer"
23:19.45cripito:)
23:19.52*** join/#asterisk mattHelm (n=root@adsl-68-90-140-81.dsl.fyvlar.swbell.net)
23:20.00Vcoi just use parking
23:20.03cripitocasi todo nuevo sip phone have un key for that
23:20.14Vcoor blindxfer
23:20.29cripitoi mean almost all the new sip phones have support for blind transfer or attended transfer
23:20.45Vcothats for a sip phone tho,,,not a zap
23:21.05cripitoin zap.... basically the features....
23:21.25cripitou can program the features to work with the phones...
23:21.53FuriousGeorge|baby|: o sea:  si tu inicias a un xfer, y tu cuelgas la lina, disconectas a la persona que estas tratando de transferir
23:21.54cripitoor look in the *XX in the chan_zap.c
23:22.23cripito:D attd transfer right?
23:22.23mattHelmAnyone want to entertain a "no d-channels" question?
23:22.38justinuis there any way to get the latest polycom sip image and boot loader? it's not on the freedomphones link
23:23.19justinui don't really care about the sip image, but i'd like the new boot loader so I can do HTTPs provisioning
23:23.25Vcohmm..k..specified h323 to only listen on internal interface
23:24.00Vcowould proabably be a good plan to  have that by default
23:24.22|baby|FuriousGeorge no, si yo inicio una atxfer, y la persona de la extension que llamo no me contesta... no puedo recuperar la llamada de nuevo, se cuelga
23:24.24Vcorather than anonymous-assclown=sure why not
23:27.11FuriousGeorge|baby|: ah.  no es posible pasar cuantos segundos quieres que intenta el atxfer=> 1...${EXTEN},30)
23:27.16FuriousGeorge?
23:28.07|baby|creo k no
23:29.05FuriousGeorgepues la verdad es que no se decirte.  siempre he usado el xfer normal del zap, o implementado por el cliente
23:29.16FuriousGeorgeen sip.
23:30.21cripitobaby basicamente te pasa porque tu ya colgaste...
23:30.28cripitono hay retorno atras
23:30.44cripitoes blind
23:31.31|baby|cripito no, yo no cuelgo el telefono
23:32.22|baby|yo marco *3 (musicohold a la persona que me llama), me da tono... marco la extension y si la extension no contesta... y la llamada se corta y yo no puedo recuperarla
23:32.32|baby|si marco *3 de nuevo, o cualquier cosa... no vuelvo a la llamada
23:32.56|baby|en cambio si la extension me contesta, hablo con la extension y cuelgo .. la llamada se transfiere
23:33.00marc324what motherboard to get for opteron cpus?
23:33.02cripitointeresante
23:33.18cripitopastebin tu features.conf
23:33.57ltersjustinu, I hope u find the firmwares.
23:34.17ltersjustinu, cause I will need them too...
23:34.27cripitoyeap .. keep us posted ;)
23:35.39|baby|[featuremap]
23:35.39|baby|blindxfer => *3 ; Blind transfer
23:35.39|baby|disconnect => * ; Disconnect
23:35.39|baby|automon => *1 ; One Touch Record
23:35.39|baby|atxfer => # ; Attended transfer
23:35.49|baby|tengo las teclas cambiadas # y *3
23:38.19cripito<PROTECTED>
23:38.19|baby|si, en realidad para atxfer estoy marcando #
23:39.10justinuiters: no luck yet
23:39.28mattHelm<PROTECTED>
23:39.35mattHelmWhat is the cause of that usually?
23:39.45justinuyou're not receiving any T1 signal
23:40.04mattHelmTelco's fault?
23:40.10justinupossible
23:40.22justinucan you loop up your niu towards yourself to check your cabling?
23:40.42mattHelmNo D-channels available!  Using Primary on channel anyway 24
23:40.57justinuwithout a T1 carrier, you can't have a D channel, so makes sense
23:41.04mattHelmHmmm. I'm from the old school of external CSU/DSU
23:41.20justinunew circuit?
23:41.35mattHelmNo, has been mostly working for a few months.
23:41.42justinuok, anything that you know of changed?
23:41.44mattHelmA few dropped calls every day.
23:41.48mattHelmNothing changed.
23:41.50*** join/#asterisk paxr0 (n=Walter@200-126-112-99.bk8-dsl.surnet.cl)
23:42.04justinuworth a call to the telco then
23:42.11justinuthey can usually get things going pretty quickly
23:42.47mattHelmThanks, that's what I thought but don't know how to tell what's going on the the digium card.
23:43.01justinuseems like you got the right info from it somehow
23:43.54mattHelm<PROTECTED>
23:52.33*** join/#asterisk n3u7 (n=neutrin0@CPE000d8802a707-CM0011e6c7edb1.cpe.net.cable.rogers.com)
23:52.58*** join/#asterisk mogorman (n=mogorman@gateway.digium.com)
23:53.34*** join/#asterisk wmandra (i=wmandra@pcp04943183pcs.verona01.nj.comcast.net)
23:55.52n3u7yst another night trying to get asterisk installed
23:56.04n3u7www.darkphiber.ca/asterisk
23:56.06drumkillan3u7: make install!!!
23:56.32n3u7drumkilla:I'm working on ztool right now
23:56.37drumkillayou need newt
23:56.39n3u7*zttool
23:56.46encodeX-Rob: are you around?
23:56.50n3u7it boots and quits
23:57.08n3u7which is an improvment over ubunto , knoppix and sarge

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