irclog2html for #asterisk on 20050907

00:00.15asterisk99dunno --- make confuses the begeebies outta me
00:00.28FuzzyCatjust use the fecking dbput/get
00:00.56asterisk99I am following Zaptel Installation... I've done the CVS checkout for zaptel
00:01.01Ariel_yes there is
00:01.27FuzzyCatare you in the zaptel folder
00:01.29FuzzyCat?
00:01.35asterisk99I am following Zaptel Installation... I've done the CVS checkout for zaptel
00:01.39FuzzyCatare you in the zaptel folder
00:01.41fugitivoHmm-home: tell us if it work, it'll be interesting
00:01.50Ariel_cd /usr/src/zaptel
00:02.21asterisk99FuzzyCat: You're a frikkin' genius!!!!!!!  Most excellent!!!! That doo-ed the trick
00:02.24Hmm-homefugitivo I will
00:02.34Hmm-homein theory it should be fine
00:02.50Hmm-homei don't think memory is an issue there
00:03.15Ariel_Hmm-home, I am not a programmer well at least I dont' say I am but don't you also have to account for the bytes for the name of the varriable as well
00:03.17asterisk99FuzzyCat: Thanks!! :)   Happy camper time
00:03.23FuzzyCat:)
00:03.51asterisk99FuzzyCat: The wiki needs to be updated ... needs a 'cd zaptel' added
00:03.59*** part/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au)
00:04.05Ariel_asterisk99, did you try the new KB1 echo Canceling setup?
00:04.09Hmm-homegood call ariel that moves my size up to 640k
00:04.15FuzzyCatyou could use memcached and get/set univar Hmm-home
00:05.43Hmm-homeactually it will probably be closer to 2k global variables
00:09.35opus_fucking bird brains that run bugs.digium.com i sware
00:12.12Juggieopus, whats the problem
00:12.14Juggiewhich bufg
00:12.16Juggie*bug
00:12.21opus_http://bugs.digium.com/view.php?id=5112
00:13.07ManxPower(19:12:34) WereCatPNS: A Yahoo Voice Chat invite?
00:13.07ManxPower(19:12:48) WereCatPNS: You know that's like inviting a wine expert to drink Boones Farm, right?
00:13.11*** join/#asterisk hellagony (n=egutierr@200.121.129.180)
00:13.43*** join/#asterisk RoyK (n=roy@cm-80.111.22.187.chello.no)
00:13.52*** join/#asterisk RoyKa (n=roy@cm-80.111.22.187.chello.no)
00:14.25FuzzyCatBULL BISCUITS!
00:14.28FuzzyCathehehhehe
00:15.48*** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au)
00:15.52*** join/#asterisk scott_ (n=scott@pcp0010655888pcs.huntsv01.al.comcast.net)
00:15.57scott_ls
00:16.10wunderkinrm -rf /
00:16.23Ariel_wunderkin, really now
00:18.12*** join/#asterisk Delta34 (n=delta34o@198.87.24.253)
00:18.23ianmYahoo voice chat ? do they have their own SIP stack similar to googles chat ?
00:19.05RoyKaFuzzyCat: what bug was that?
00:19.22*** join/#asterisk yartelecom (n=no-email@82.211.129.230)
00:19.25CoiLhehe
00:19.59FuzzyCathttp://bugs.digium.com/view.php?id=5112
00:20.13FuzzyCati think opus_ lost it ;)
00:20.46MikeJ[Laptop]opus_, does canreinvite=no and no qualify= line help with that one?
00:21.20*** join/#asterisk bjohnson__ (n=bjohnson@i216-58-18-158.cybersurf.com)
00:21.31opus_no
00:22.00CoiLyay i got my new keyboard
00:22.02opus_Let me retest that to make sure
00:22.11CoiL*waits on new lcd/mini mac patiently*
00:22.16Delta34hi, my zap call is hanging up before the call is terminated, i am calling an IVR on one of the zap channels, as soon as the IVR answers, 2to3s later it hangs up
00:22.23Delta34any idea why this is happening?
00:22.33ianmabsolute timeout ?
00:22.35opus_MikeJ : by default I use canreinvite=no and qualify=yes, are you suggesting qualify=no?
00:23.27ianmsomewhere in your dialplan ?
00:23.28ianmlike
00:23.31ianmAbsoluteTimeout(1)
00:23.40Delta34ianm are u talking to me?
00:23.44ianmyes
00:23.52ianmsorry -
00:24.09Delta34so you are suggesting putting AbsoluteTimeout(1) before the dial cmd?
00:24.23ianmNo - that WILL chop the call off after 1 second !!
00:25.13ManxPowerDelta34, set callprogress=no and busydetect=no
00:25.13ianmDo you have an AbsoluteTimeout(1) anywhere in the dial plan - if you do - then that could be the cause for your problem...
00:25.13Delta34manxpower, in zapata.conf?
00:25.14ManxPowerDelta34, yup.
00:25.24FuriousGeorgesomething is stripping the frist digit dialed into my phone and its not ignorepat
00:25.35ManxPoweror more specifically DO NOT set callprogress=yes and busydetect=yes
00:25.44ManxPowerFuriousGeorge, ignorepat never strips digits
00:25.49ManxPowerit just continues dialtone
00:26.00NetgeeksWow opus_
00:26.02ManxPower${EXTEN:1} would strip the first digit
00:26.10FuriousGeorgedamn grammer
00:26.12FuriousGeorge:)
00:26.17*** join/#asterisk mx02 (n=none@adsl-215-143-149.aep.bellsouth.net)
00:26.19FuriousGeorgei should use :0
00:26.21FuriousGeorge?
00:26.27ManxPowerno use ${EXTEN}
00:26.33FuriousGeorgegotcha, thanks
00:26.55ManxPowerIf you used ${EXTEN:0} people would point and giggle and somneone would put a sign "kick me" on your back when you were not paying attention.
00:27.10Netgeekslol
00:28.24NetgeeksYou could cut the giggling by using ${EXTEN:0:99}
00:28.55FuriousGeorgelol
00:29.12*** join/#asterisk wolfson (n=hehe@65.174.122.198)
00:29.14Netgeekswhich would work unless you dial some really far off place
00:29.16jontowjiggling!
00:29.50Delta34hmmm still having same issue, tried both callprogress=yes and no busydetect=yes and no
00:30.09Delta34any other suggestions to try?
00:30.13Ariel_kick me no him no kick your self
00:30.49Netgeeksway back when there was this guy who was pushing ipv8... yes, ip version 8....  it was designed so that we would have usable address space for when we colonized other galaxies....  he got giggled at more than anyone I know
00:30.59ManxPowerDelta34, either set them to no or don't set them at all, then do a "unload chan_zap.so" then a "load chan_zap.so" to make the changes take effect.
00:31.08opus_ipv8 haha
00:31.09dudesNetgeeks - have you looked at 1.2 much?
00:31.27NetgeeksAsterisk 1.2?  nah, I'm using Head now
00:32.05dudesI was thinking about doing some stress testing on it to see how it holds up ... but I haven't had the ambition to make install it yet
00:33.15JuggieManxPower, did you get the 4 port card?
00:33.45Ariel_dudes, does a call center behind a 96 channel preditive dialer work as a stress test.
00:33.47NetgeeksI've got two client projects one due on the 15th and one on the 19th... about $40k worth of income for me between the two, so thats been my life... coem sept 20th, I'll have time to breathe I think
00:33.52ManxPowerJuggie, Didn't end up needing it, the CLEC got their network at least mostly up
00:34.04Juggieyah... good
00:34.05dudesAriel_ - I'd think that may count
00:34.15Ariel_dudes, then it's working
00:34.35dudesHow about chan_agent and app_queue?
00:34.47fugitivoNetgeeks: maybe he was an alian
00:34.49fugitivoalien
00:35.01Ariel_dudes, we are just using it as a trunk right now
00:35.17dudesAh
00:35.50Netgeeksfugitivo: heh, his name was Fleming if I remember right... Jim Fleming... maybe all flemings are aliens?
00:35.51dudesI'm curious to see how chan_agent changes.  App_queue on the otherhand I don't use it so
00:36.21fugitivoNetgeeks: i don't know if they're aliens, but they're weird people :)
00:36.39opus_Sep  6 17:36:32 ERROR[29053]: chan_sip.c:5971 register_verify: Peer '354' is trying to register, but not configured as host=dynamic
00:36.46opus_is that really an error?
00:36.58twistedopus_, yes
00:37.04opus_should it be type=friend?
00:37.13twistedread what the message says.
00:37.25twistedit tells you in plain english what the problem is
00:37.39Ariel_host=dynamic needs to register but if they have host=XXX.XXX.XXX.XXX they don't register
00:37.50opus_twisted: I know... someone told me it would fix the problem..
00:38.01twistedno
00:38.07twistedthe message tells you what to do to fix the error.
00:38.29opus_ok i'll put it back to dynamic
00:38.35fugitivovoicemail.conf has too many options
00:38.40MikeJ[Laptop]<PROTECTED>
00:38.42MikeJ[Laptop]y
00:40.30Hmm-homehmmm do alphanumeric extensions actually work?
00:40.39ManxPowerHmm-home, yes
00:40.42Hmm-homeinteresting
00:40.48Hmm-homethat solves one of my huge problems
00:40.59ManxPowerI use them ALL the time
00:41.08opus_MikeJ I tried qualify=no/yes canreinvite=yes/canreinvite=no no luck.
00:41.15Ariel_opus_, you can add defaultip=ipaddress in the setting and still have host=dynamic
00:41.19Hmm-homebasically i just need to use them to set a variable
00:41.19ManxPowerI think macro-std-exten even has them in extensions.conf.sample
00:41.26opus_Ariel Oh!
00:41.55*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
00:41.59Ariel_Hmm-home, don't you use s,t,i
00:42.00*** join/#asterisk keith (i=transgre@71.14.20.160)
00:42.07Ariel_or even fax
00:42.40Hmm-homeAriel_ yeah, however, this is for a different purpose, once I'm done I can chat about it
00:42.44MikeJ[Laptop]opus_, can you pastebin you sip.conf for that?
00:42.56harryvvlooks like cisco,dell,and two other companies are settign up wifi  systems at all the red crosses in the gulf states where katrina hit..
00:43.15opus_MikeJ its a realtime database
00:43.29CoiLhmmm
00:43.31opus_i can paste an INSERT if you wish
00:43.35opus_and CREATE
00:43.52Kattybeep.
00:44.21*** join/#asterisk Mulvane (n=mulvane@c-24-129-34-230.hsd1.fl.comcast.net)
00:44.21opus_harryvv thats awesome.
00:44.57MikeJ[Laptop]ummmmm
00:45.00harryvvopus, I would like to see asterisk used in one of these locations.
00:45.11MikeJ[Laptop]how big is the table?
00:45.31MikeJ[Laptop]I want to simplify this as much as possible
00:45.42*** join/#asterisk gnarf37 (n=corey@west-rock.rockriver.net)
00:45.43knight_AAH isnt sending local extension callerids to dialparties.agi
00:45.44opus_like 50 columns
00:45.44knight_weird
00:46.01MikeJ[Laptop]well...
00:46.14FaithfulHey are the Sipuras nice phones?
00:46.20harryvvokay in looks
00:46.30MikeJ[Laptop]can you produce a sip.conf that reproduces the same thing.
00:46.37opus_MikeJ: simplify, I know exactly what you mean. I'm thinking we're going to need a script that sets up a common asterisk realtime build just to communicate these types of bugs
00:46.38MikeJ[Laptop]without all the extra garbage
00:46.41Ariel_Faithful, there ok. But sound and they tend to have a little echo all the time.
00:46.49opus_MikeJ: i'll go the sip.conf route
00:47.20*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
00:47.23*** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net)
00:47.34gnarf37I have a question RE: CDR records and the Transfer() call.  I have one asterisk server acting as a main routing box, which also keeps track of my accounting information using the CDR records.  I want to be able to use the Transfer() function on the second box so that it doesn't tie up two channels, but when it gets transfered I can't keep track of the original call anymore
00:48.32*** join/#asterisk pauldy (n=pauldy@c-24-1-152-240.hsd1.tx.comcast.net)
00:49.28opus_MikeJ : http://pastebin.ca/22258
00:49.59MikeJ[Laptop]opus_, what version?
00:50.19opus_<PROTECTED>
00:50.34opus_CVS HEAD from last night.
00:50.41Ariel_comcast.net hummmm
00:50.55opus_its just the name of the site
00:51.05opus_although comcast is going to roll out VOIP soon
00:51.05MikeJ[Laptop]opus, need externip and localnet
00:52.13CoiLis there a bill around
00:52.34QwellCoiL: I can send you mine.  Wanna pay it>
00:52.41MikeJ[Laptop]CoiL, I'll..
00:52.42MikeJ[Laptop]darn
00:52.47Qwell:p
00:53.00opus_MikeJ: http://pastebin.ca/22259
00:53.02CoiLhaha
00:53.22CoiLim applying for a job at digium, wish my luck
00:54.13FaithfulI have an issue with my extensions.conf, I have "include => remote|9:00-17:30|mon-fri" in my default context but it goes to this context no matter the time
00:54.17Ariel_my luck...humm grammer would help when looking for a job.
00:54.34opus_Faithful try GotoIfTime
00:54.39shido6opus, what in the .....
00:54.43opus_instead, its also easier to read
00:54.49opus_shido6 what
00:54.58shido6the pastebin...
00:55.27MikeJ[Laptop]opus_, and have you tested acutally using that config?
00:56.00opus_Yes
00:56.08opus_I call, I get audio about 1/20 times
00:56.18opus_I type rtp debug, its sending packets to 192.168.1.102!
00:56.27opus_which doesn't exist over there, thats over here.
00:56.33opus_then, sometimes it works
00:57.08opus_is it because it thinks 192.168.1.102 != localnet?
00:57.11Ariel_opus_, should you not have type=peer or user
00:57.25opus_Ariel what type should I have?
00:57.27*** join/#asterisk brc_ (n=brc@pdpc/supporter/basic/brc)
00:57.54Ariel_opus_, user is getting the call in peer is sending the call out to the device
00:58.11Ariel_friend is both but don't know if real time uses this type of setup.
00:59.01MikeJ[Laptop]opus_, try just one entry to test, see what happens
00:59.11opus_i just tried type=friend
00:59.14opus_same problem
00:59.21MikeJ[Laptop]maybe it is not mapping properly unless it hits at the same time as an attempted register
00:59.29MikeJ[Laptop]or somthing qwirky
00:59.32dudesopus_ - could be a NAT issue
00:59.43MikeJ[Laptop]are you calling from asterisk to phone, or from phone to asterisk?
00:59.46opus_i just tried type=peer, same problem
01:00.03opus_I'm calling phone to asterisk to an exten music on hold
01:00.10opus_Sent RTP packet to 192.168.1.102:2250 (type 0, seq 4130, ts 38400, len 160)
01:00.39MikeJ[Laptop]heh...
01:00.40MikeJ[Laptop]wait
01:01.02*** join/#asterisk tuxinator_linuxM (n=tuxinato@70-32-106-222.ontrca.adelphia.net)
01:01.23MikeJ[Laptop]add rtcachefriends=yes and rtignoreexpire=yes
01:02.20opus_in realtime?
01:02.23*** join/#asterisk rob314[laptop] (n=rob314[l@cpe-65-185-169-238.neo.res.rr.com)
01:02.29MikeJ[Laptop]in global settings
01:02.33MikeJ[Laptop]in realtitme
01:02.55MikeJ[Laptop]is your phone sucessfully registering?
01:03.05opus_| 220 |          0 |          0 |         0 | sip.conf | general  | rtcachefriends    | yes
01:03.14opus_I have that
01:03.19opus_I don't have 'rtignoreexpre'
01:03.24opus_which must be new in the last two weeks
01:03.46opus_i need to go pick up my mexican food brb
01:03.55MikeJ[Laptop]oh,..
01:03.56MikeJ[Laptop]and rtautoclear=yes
01:04.09MikeJ[Laptop]ok
01:04.26fugitivowho uses all the configuration options of voicemail.conf?
01:04.35MikeJ[Laptop]is your phone sucessfully registering?
01:04.40twistedopus: look at externalip= in sip.conf, and nat=yes
01:04.53MikeJ[Laptop]and rtupdate=yes
01:04.56Qwellexternip?
01:04.59twistedhaha yeah
01:05.03MikeJ[Laptop]twisted, he says it's in there
01:05.12*** join/#asterisk zotz (n=zotz@24.231.36.100)
01:05.44twistedMikeJ[Laptop], including nat settings?
01:05.56MikeJ[Laptop]yeah
01:06.00twistedand localnet settings?
01:06.21MikeJ[Laptop]in general externip and localnet, on peer nat=yes
01:06.23twistedmmm
01:06.24twistedfood
01:06.27twistedbrb
01:06.27MikeJ[Laptop]heh.
01:06.41*** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca)
01:06.50opus_Sent RTP packet to 192.168.1.102:2252 (type 0, seq 46987, ts 36320, len 160)
01:06.56opus_<twisted> opus: look at externalip= in sip.conf, and nat=yes
01:07.00opus_^^^ same problem
01:07.13fugitivoopus_: what's the problem?
01:07.18knight_weird, my asterisk isnt sending agi all the parameters and asterisk variables
01:07.38opus_fugitivo : http://bugs.digium.com/view.php?id=5112
01:08.57*** join/#asterisk Uberbot (n=Uberbot@pcp01880954pcs.sandia01.nm.comcast.net)
01:09.17knight_hmm
01:09.35knight_zap show channel 1 shows blank for Caller ID string
01:09.45*** join/#asterisk ooj (n=ooj@219.133.219.200)
01:10.02oojHello
01:10.25steve___hi all
01:10.50steve___any linux softphones support g729?
01:11.06*** join/#asterisk Amir_ (n=koki@201.137.93.184)
01:11.41Amir_has anyone seen Mike?
01:11.56FaithfulWhere should we be pasting to?
01:12.06fugitivoopus_: asterisk behind nat and sip is a problem
01:12.20knight_asterisk works fine behind nat and sip
01:12.37fugitivosometimes, yes
01:13.23fugitivoopus_: asterisk<-router--router->phone sometimes is a problem
01:13.50Qwellopus_: FYI, acting stupid on the bug tracker is the easiest way to not get help.
01:14.08oojdigium TE410P can not support RTP before connect
01:14.08knight_oh true
01:14.12knight_NAT on both ends is ugly for sip
01:14.21knight_NAT on both ends for most anything is pretty ugly actually
01:14.22QwellWhy not use an stun?
01:14.26fugitivoopus_: i recommend you to use iax for that config
01:14.37*** join/#asterisk trig_hm (n=jb@home.monkeypr0n.org)
01:15.16oojwhere is config TE410P Doc?
01:15.30twistedlol
01:15.33Qwellopus_: mind a quick msg?  I don't want to flood in here
01:15.33twisteddr phil on cnn
01:15.42Qwelltwisted: they sent his ass to the Astrodome
01:16.00twistedit's amusing
01:16.02twistedbecause he's on tv
01:16.05twistedhe's suddenly an expert
01:16.09Qwellyeah..
01:16.14QwellYou know who they should send?
01:16.20QwellDr. Drew.  He's the man.
01:16.26twistedoprah
01:16.29twisteder
01:16.53twistedyeah, hopefully you got that joke
01:17.06Qwelldidn't
01:17.14twistedah
01:17.15*** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net)
01:17.20twistedwell
01:17.21Qwellunless you're being mean...then I did
01:17.25*** join/#asterisk Gronker__ (n=Gronker2@70.152.166.237)
01:17.30twistedyeah, i kinda was
01:17.33Qwellk
01:17.34*** part/#asterisk Gronker__ (n=Gronker2@70.152.166.237)
01:18.14twistedand just so I don't get people flaring up
01:18.23twistedit wasn't racial... that woman just annoys the piss out of me
01:19.29steve___"look under your chairs everyones...."
01:19.39QwellSo, how did Huntsville deal with this whole thing?
01:19.51steve___*crowd screams
01:19.53*** join/#asterisk [hC] (n=hardcore@c-24-127-192-210.hsd1.fl.comcast.net)
01:19.55twistedQwell, ?
01:19.59steve___"keys to a new car!!!!"
01:20.12Qwellsteve___: "and you have to pay taxes!  you poor mother fuckers!"
01:20.13Qwellheh
01:20.19Qwellwhat a dumb bitch oprah is
01:20.29twistedhaha
01:20.30twistedokay
01:20.32*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
01:20.32Hmm-homenow that is one ass backwards dialplan
01:20.33twistedso i'm not the only one she bothers :P
01:20.34[hC]Has anyone experienced issues where call audio will cut out for 4-6 seconds occasionally? Im not sure if it is IAX, asterisk itself, SIP, or the PRI i am dialing out doing it. but it is incredibly hard to try to debug.
01:20.36FaithfulCan someone cast an eye on this part of my extensions.conf http://sial.org/pbot/12968 it always goes to [remote] which is the 1st s,1 in the extensions.con even though all devices and phones are in [default]
01:20.39Qwelltwisted: not at all :p
01:20.44twisted[hC], it's iax.
01:20.44Hmm-homemaking me do authorization in the DP
01:21.05twistedhi Netgeeks
01:21.06[hC]twisted: heh :P any way to be certain, or to troubleshoot/tune it?
01:21.12steve___the part that is most freightening is how the crowd reacts...  they go complete ape-$hit
01:21.21twisted[hC], we've proven that iax has the issue you have described
01:21.26Netgeekstwisted: you bugger, you get my email about the code?
01:21.36Faithfulinfact I am sure it doesn't hit default context at all
01:21.41twistedNetgeeks, hmm... i dunno... lemme check.
01:21.50[hC]twisted: was it you demoing the milliwatt tone the other day?
01:22.10Faithfuldo I need to put [default] above other contexts?
01:22.53[hC]twisted: i thought it only happened with high call volume?  And I didnt think it lasted as long as 6-8 seconds..
01:23.13twistedNetgeeks, nope
01:23.21Netgeeksbah..... okay, will resend
01:24.01QwellFaithful: no, I don't believe so
01:24.17Netgeeksand I was wondering why you were ignoring the offer of cash
01:24.22Qwellthe only time context order matters, is when you're including them
01:24.35Netgeeks~twisted
01:24.37jbotfrom memory, twisted is twisted@indigent-networks.com, but you can paypal him at toastido@toastido.net
01:24.45Netgeeksthats the right addy?
01:24.50twistedyeah
01:24.56knight_hey twisted
01:25.28NetgeeksI'm betting it has!
01:25.29twistedhey knight_
01:25.40NetgeeksMy subject line was 'Earn Cash while you sleep!'
01:25.44knight_wassup?
01:26.04twistedNetgeeks, haha.. yeah.. that'd do it.
01:26.12twistedknight_, not much, about to have to clean my apartment again
01:26.17knight_ahh
01:26.20[hC]twisted: sorry to keep bugging you, heh.. but is there any solution to this audio problem besides using SIP?
01:26.33twisted[hC], utilize SIP? :P
01:26.44[hC]:)
01:26.48Qwelluse sccp
01:26.56twistedor mgcp
01:27.00Qwellh323?
01:27.07twistedhell no, we won't go
01:27.14Qwellsissy
01:27.15Qwell:P
01:27.17twistedhah
01:27.40QwellWhere has drumkilla been?
01:28.02twistedskool
01:28.16mog_homeschool is for suckers
01:28.16drrayskewel
01:28.32twistedokay, time to poop and collect garbage
01:28.36twistedbbl
01:29.02[hC]sucks, iax has so many nice features
01:33.39syleanyone have a realtime setup?
01:35.02sylei notice when i add users to the mysql table , it does a select * on name, but lets say i then edit a field in that name, it never goes back and does a select * on name again, i don;t see how this is realtime at all
01:35.33syleits seems liek once its cached name once, game over
01:36.35*** join/#asterisk goatmilk (n=goatmilk@130-127-45-11.chouse.resnet.clemson.edu)
01:36.42Netgeekstwisted: email sent, subject line: Looking to have some dev work done for asterisk
01:37.52Netgeekssyle: do you have rtcachefriends=yes in your config?
01:37.56Delta34hi, my zap call is hanging up before the call is terminated, i am calling an IVR on one of the zap channels, as soon as the IVR answers, 2to3s later it hangs up
01:38.24Delta34tried both callprogress=yes and no busydetect=yes and no, still no luck
01:38.31sylenetgeeks: yep
01:38.37syleis that what is causing it?
01:38.49Netgeeksthats the problem, that says, cache the realtime entry
01:38.56syleahh thanks
01:39.00Netgeeksbut if you don't cache the entry, then you can't do MWI
01:39.19sylehow about nat?
01:40.31Netgeeksgood question, not sure
01:40.39Netgeeksone way to find out
01:41.26tzangeris there any way to get iaxclient to support g729 (utilizing an asterisk's codec_g729 or something perhaps?
01:44.52h3xsure
01:44.53h3xwrite the code
01:44.54h3xhaha
01:45.49*** part/#asterisk paing (n=chatzill@202.188.115.100)
01:48.57opus_<PROTECTED>
01:49.19opus_oh
01:49.26opus_thats MoH
01:50.04tzangeractually I had an MoH question too -- is there any way to take an ice|shoutcast stream and play that to anyone supposed to be listening to MoH instead of decoding an MP3 for EVERY MoH listener?
01:50.31opus_hmmm. thats a very detailed question
01:50.38opus_did you see the thread on MoH and mplayer?
01:50.45tzangeropus_: yes I did actually
01:50.55tzangerI was gonna go over it in more detail tomorrow (list is at work)
01:50.56opus_hmmm, did it give you any ideas?
01:51.30opus_i'd really like to get mplayer working for MoH myself, I've *never* beable to get shoutcast to work even though I tried for a long time
01:51.33tzangeropus_: yeah I was just checking with people in here in case something's already been impemented
01:51.46*** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net)
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01:56.26DrukenHMEtzanger: yeah you can use a shoutcast stream :)
01:56.43*** join/#asterisk Kumbang (n=unknown@167.205.24.5)
01:56.56tzangerDrukenHME: got a link?
01:57.43DrukenHMElemme look
01:58.13knight_hey teezee
01:58.40DrukenHMEtzanger : default => mp3:/var/lib/asterisk/mohmp3-empty,http://www.waixwave.com:8000/
01:58.44*** join/#asterisk supaigtr (n=yurplsl@152.53.17.1)
01:58.56supaigtrSup ppl
01:59.06knight_grrr
01:59.06tzangerDrukenHME: uhm... ok, I'll see what I can do to make that work :-)
01:59.11DrukenHMEjust touch an empty.mp3 in the mohmp3-empty dir
01:59.23DrukenHMEi did it once... worked fine..
02:00.20DrukenHMEtzanger: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20musiconhold.conf#comments
02:00.24DrukenHMEfor reading material
02:00.43supaigtrAnyone had any experience with in call monitor feature?
02:00.44tzangerDrukenHME: wow, thanks!
02:01.28DrukenHMEtzanger: wow?
02:01.42tzangeryeah I haven't done much on moh at all, this saved me a LOT of head-scratching
02:01.51DrukenHMEhehehe
02:01.56DrukenHMEmoh is my gawd
02:02.38*** part/#asterisk rob314[laptop] (n=rob314[l@cpe-65-185-169-238.neo.res.rr.com)
02:02.51knight_grrr, asterisk thinks all my zap channels are busy
02:02.53knight_wtf
02:03.01DrukenHMEwhen my shoutcast server doesn't crap out on me, i run a shoutcast as my moh 24/7 hehe
02:03.07*** join/#asterisk santiago (n=santiago@63.245.86.254)
02:03.14DrukenHMEserver needs a layover and i just haven't had the time... :(
02:03.21tzangerDrukenHME: :-)
02:03.32tzangerI am just trying to keep the number of codec translations down
02:03.39*** join/#asterisk sudhir492 (n=sudhir@pool-71-114-87-156.washdc.dsl-w.verizon.net)
02:03.45tzangernot everyone needs their own "version" of the same MoH played and converted from mp3 to g729
02:03.58tzangerjust do it once and copy the same damn frame to every g729 client listening to moh
02:04.10DrukenHMEi dun use g729 very often...
02:04.20DrukenHMEi think only 1 client uses it... and it still sucks ass
02:04.48Delta34my zap channel is hangup a call to an ivr before it ends, any suggestions on where to look to fix
02:05.27supaigtrDelta34: Do you see the hangup?
02:05.27DrukenHMEDelta34: your zaptel channel is hanging up before it should?
02:05.42Delta34yes
02:05.59Delta34i see the hangup 3s after the ivr answers
02:06.05DrukenHMEahh.. ok...
02:06.27DrukenHMEbut is it a HANGUP or just the channel being taken down ?
02:06.37sudhir492Is there a way to play a radio station for MoH
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02:07.23Delta34<PROTECTED>
02:07.23Delta34<PROTECTED>
02:07.23Delta34<PROTECTED>
02:07.23Delta34<PROTECTED>
02:07.24DrukenHMEsudhir492: you could prolly pipe in the soundcards audio... ?
02:07.49sudhir492There is no soundcard on the server
02:08.01Delta34* hangups the call before my ivr even finishes
02:08.13DrukenHMEsudhir492: then yer SOL
02:08.16tzangersudhir492: then use the shoutcast option :-)
02:08.28tzangerDrukenHME: do you know how to specify "take the input of this soundcard" ?
02:08.51DrukenHMEtzanger: i don't... but i'm sure it's in the wiki somewhere
02:09.01tzangeryeah I'm thinking something with sox but not sure :-)
02:09.45sudhir492This is for an office setup. The PBX has a PRI card and about 40 extensions. Is it possible to play one of the IP broadcast as music on hold?
02:10.00FaithfulQwell: can you see why that wouldn't work?  http://sial.org/pbot/12968
02:10.14*** join/#asterisk nick125 (n=nick125@unaffiliated/nick125)
02:14.38DrukenHMEFaithful: you'll never pass the musiconhold()
02:14.49syleis there a dial command to just dial locally into a context
02:14.50DrukenHMEuse waitmusiconhold(20) instead
02:15.36sylelol
02:15.42sylegood command
02:15.55FaithfulK
02:16.00dudesMusiconhold rules
02:16.06sylei was actually trying to see if i could dial a local context with L command to dial to play only a certain amount of musiconhold lol
02:16.16sylethat works better
02:16.18Netgeeksdial(local/<exten>@<context>)
02:16.23greekmananyone know how to fix the dial by name directory?
02:16.40knight_why would my asterisk server have all lines busy?
02:16.49sylei;ll write that down, thx netgeeks
02:17.55dudesknight_ - do you have a hangup in extensions or perhaps they're busy?
02:18.14knight_dudes, zap extensions arent busy
02:18.20knight_no hangups
02:18.33dudesWhat's the error/warning you get?
02:18.36dudeson the CLI
02:19.05knight_<PROTECTED>
02:19.08knight_<PROTECTED>
02:19.30dudesNever seen that
02:19.35*** join/#asterisk spootnick (n=irc@50.118.233.220.exetel.com.au)
02:19.49dudesBut, perhaps there is a flaw in your dialplan logic?
02:19.59*** join/#asterisk corndog (n=corndog@208.187.28.250)
02:20.02dudesif you post it on pastebin.ca I'll look at it
02:20.05knight_most likely, i'm testing out aserisk@home
02:20.16FaithfulDrukenHME: the main issue I am having is that it never goes to [afterhours] http://sial.org/pbot/12969
02:20.35FaithfulDruken: thanks for the musiconhold tip too... your a champ
02:20.41dudesI got a half bobcat cat here ... it's f'n huge
02:20.46Faithfulyou're a champ
02:20.52greekmananyone know how to fix the dial by name directory?
02:21.43DrukenHMEFaithful: well, why would it goto afterhours?
02:22.01sylesounds interesting, what is name of app greekman?
02:22.13*** join/#asterisk hat (n=hat@bb220-255-134-33.singnet.com.sg)
02:22.21FaithfulDrukenHME: all my devices are [default]
02:22.50corndogI ditched the Wildcard TDM400P (2) and moved to Wildcard TE110P.  I'm having problems with dialing out....
02:22.54DrukenHMEFaithful: ok, and default includes afterhours...
02:23.05FaithfulDrukenHME: so tell me why it goes to [remote] instead
02:23.08greekmanits the actual dial by name directory for asterisk, mine doesn't recognize anybody
02:23.25corndogztcfg shows "SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)" ... "12 channels configured."
02:23.40*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
02:23.48DrukenHMEFaithful: GotoIfTime(13:00-17:00|mon-fri|*|*?remote,s,1)
02:23.49FaithfulDrukenHME: I hate to say it but maybe I should reboot :P
02:24.09knight_<PROTECTED>
02:24.14knight_how can i see those from CLI?
02:24.20DrukenHMEgoto remote is time between 1pm and 5pm, mon to friday, in any month
02:24.53Faithfulyes but we are outside those hours
02:25.02corndogI just get the 'Congestion'
02:25.19corndogideas?
02:25.33FaithfulDrukenHME: What seems to be happening is that ALL incoming just go to [remote] no matter what
02:25.35*** join/#asterisk jaike (n=jaike@203.177.36.132)
02:25.57jaikeanyone found a way to disable call waiting on polycoms?
02:26.14paulcjaike: limit calls per registration to 1, using the 1.5.2 firmware
02:26.26corndogI forgot - the telcos hardware is a Adtran 616 and the green light is bright on the back of the card
02:27.00DrukenHMEFaithful: where are your calls coming from ?
02:27.01jaikepaulc: thanks. which config file? and parameter?
02:27.16FaithfulDruken: ZAP
02:27.26DrukenHMEpaste your zaptel.conf ?
02:27.31Faithfulok
02:27.45*** join/#asterisk spackle (n=spackle@209.234.83.19)
02:28.34paulcjaike: your per-phone configuration file I think, and it's something like callsperregistration - it'll be apparent when you look in the file
02:29.05jaikeok..thanks...its messing up queueing
02:29.11spackle~seen ManxPower
02:29.12jbotmanxpower is currently on #asterisk (2h 44m 51s).  Has said a total of 15 messages.  Is idling for 1h 47m 53s
02:29.12jaikehope i get it to work
02:29.17DrukenHMEjaike: just limit the incoming calls to 1
02:29.17Hmm-homegot i need to get back in a band
02:29.22Hmm-home*god even
02:29.57paulccall waiting shouldn't mess up queueing if your queueing's configured right
02:30.32jaikewhen the phone is ringing and theres another call on the queue...it goes into call waiting on the polyom
02:30.44DrukenHMEpaulc: i get the call waiting... it happens when your using the agent lines, rather then having dedicated agent logins..
02:30.49jaikeso the polycom gets twice the number of calls
02:31.34jaikeis it CallsPerLineKey?
02:31.37paulcah.. DING! I get it now..
02:31.42paulcjaike: sounds about right :)
02:31.49paulcthat, together with line keys per registration
02:31.49jaikeok..thanks man
02:32.11nick125i think my server is <insert your favorite four letter word here>ed.
02:32.43DrukenHMEdoom ?
02:32.46nick125i think this is kinda bad:
02:32.48nick125Starting program: /usr/sbin/mysqld
02:32.48nick125Warning:
02:32.48nick125Cannot insert breakpoint 1.
02:32.48nick125Error accessing memory address 0x171fb6: Input/output error.
02:32.55FaithfulDrukenHME: sorry the phone keeps ringing http://sial.org/pbot/12973
02:33.13spackle"MySQL" +ed?
02:33.26spackleoops, that's 5 letters
02:33.36nick125no
02:33.59DrukenHMEFaithful: zapata.conf looks right
02:34.09nick125i think that error is kinda a bad sign..
02:34.13*** join/#asterisk vuvie (n=vuvie@bb219-74-47-90.singnet.com.sg)
02:34.18*** part/#asterisk vuvie (n=vuvie@bb219-74-47-90.singnet.com.sg)
02:34.26DrukenHMEFaithful: paste me the verbose from an incoming call
02:34.34DrukenHMEset verbose 12 in your CLI
02:34.53FaithfulDrukenHME: Thanks for your help, mate
02:35.21FaithfulMight as well play the Aussie seeing I am one.
02:35.33jaikepaulc: have you had any problems with 1.5.2 firmware? seems i can go back to 1.4 if 1.5 doesnt work out
02:35.45syleinteresting topic
02:35.53syleis there a change log somewhere
02:36.18paulcjaike: I have issues with call waiting prompt focus in 1.5.2 - it doesn't want to work as it did in 1.4.x, regardless of config file setting..
02:36.24paulcbut apart from that, it seems to work fine
02:36.45paulcjust don't use it with 3.x boot ROM on a 300/500 - they don't have enough RAM - stick with 2.6.2
02:37.07sudhir492DrukenHME: What is SOL?
02:37.14jaikewe have polycom 301s...i thnk theyve enough ram
02:37.21paulcyeah, the 301/501s are fine
02:37.26DrukenHMEsudhir492: shit outta luck
02:37.28paulcSOL = Shit outta luck
02:37.39sudhir492:-)
02:38.33FaithfulDruken: http://sial.org/pbot/12974
02:38.51sudhir492If there is a soundcard, is it possible to pipe that to a conference room?
02:39.14sylewhy not
02:39.19sylejust run the speaker wire
02:39.27sudhir492Haha
02:39.53*** join/#asterisk jimmy_deanPB (n=jhodapp@cpe-71-65-37-102.indy.res.rr.com)
02:40.00sylesetting up a sound card on other hand can be a bitch
02:40.12sudhir492I mean conference on Asterisk, so that multiple party's can join in to listen to that?
02:40.17sylei;d refer to mythtv documentation if running spif etc
02:40.24syleout to an amp
02:40.51syleyou mean meetme?
02:41.14sudhir492syle: yes, meetme. Pardon my incorrect terminology
02:41.40sylehmmm
02:42.44spacklesudhir492, why would you want to, you can do a meetme monitor only on another conference phone more easily.
02:42.48Kumbangguys guys
02:43.17Kumbangi got serious connection problem with my telco
02:43.19Kumbang<PROTECTED>
02:43.20Kumbang!! Got I-frame while link state 2
02:43.24sylei think hes talking about intercom system
02:43.26sylebut i could be wrong
02:43.32DrukenHMEFaithful: perhaps move the context=default above the imimmediate=yes
02:43.35DrukenHME?
02:43.41*** join/#asterisk Cherebrum (i=jgarland@72.36.136.226)
02:43.47sudhir492Kumbang: reset your PRI, if still the same problem then call your Telco
02:43.52greekmananyone know how to change the context in the dial by name directory?
02:44.04*** join/#asterisk docubot (n=docubot@user-12l21j7.cable.mindspring.com)
02:44.12CherebrumIs the realtime odbc config for voicemail broken in 1.2.0-beta1 ?
02:44.26sylehttp://www.voip-info.org/tiki-index.php?page=Asterisk+Paging+and+Intercom
02:45.04sylei'm curious
02:45.10syleanyone actually setup mythtv in here
02:45.25sudhir492spackle: Because there are multiple callere, from different places, and want to play them some live announcement,
02:45.25spacklesyle, I'm watching it right now.
02:45.41sylekewl i set it up to, took me all damn weekend
02:45.56spacklesyle, did you find knoppmyth?
02:46.05syleyeah after i had it all working
02:46.07sylelol
02:46.10spackleheh.
02:46.27spacklesyle, congrats then, that's a bit of an accomplishment.
02:46.40syleyeah i have one thing left to do
02:46.47syledecide on what remote to interface with it
02:46.49spacklesyle, I can't get mythphone to register with Asterisk.
02:47.02sylei got it to register no problems
02:47.22spacklesyle, color me stupid.
02:47.34syleyou found mythosd yet?
02:47.47syletell your TV screen from asterisk a call came in hehe
02:48.20hathi, who is using Dell 2850 server? I want to know it PCI slot voltage
02:48.32FaithfulDrukenHME: in zapata.conf?
02:48.36spacklesyle, I corresponded with somebody that gave me some code to make asterisk send it to myth and slimserver.  Never got it to work though.
02:48.53DrukenHMEya
02:48.56sylehmmm, think mine is a 2650 Dell, think its 5.5 or something
02:49.14hatsyle, what card you are using?
02:49.34sylei;m not using any cards in that one
02:49.38hatdigium only has 3.3 volt or 5.0 volt
02:49.42sylethats strictly database server
02:49.48CherebrumAnyone using realtime external config for voicemail?
02:49.55FaithfulDrukenHME: still no go... is this puzzling ?
02:50.00syleyeah i believe they are 5.0
02:50.10DrukenHMEFaithful: did you restart ?
02:50.15FaithfulDrukenHME: I only did a "reload"
02:50.16sylethere are some docs out on NET on digium site about that
02:50.16hatok.
02:50.34sylesearch through digium's hardware docs
02:50.37DrukenHMEFaithful: there's your problem
02:50.44DrukenHMErestart
02:51.08sylespackle
02:51.15sylessh into your mythtv box
02:51.33hardwirehow many people here use 9 in front of outside extensions in their dialplan even for sip phones?
02:51.44spacklek
02:51.46*** join/#asterisk cyaltr (n=spam@66-188-89-49.dhcp.mdsn.wi.charter.com)
02:51.53hardwireI am thinking I need to make that manditory.  to make dialplan dialing in most sip phones work
02:52.00sylemake sure your watching live tv right now
02:52.03hardwirebecause people here just don't understand the Check key
02:52.04sylemythtvosd --template=scroller --scroll_text='I am to sexy!'
02:52.06hardwireDial key
02:52.07syletype that
02:52.10DrukenHMEhardwire: why bother? 3 == ext 7 or 10 == outside
02:52.10sylesee what happens
02:52.36hardwireDruken: what if the 3 digit extrension is 190 and I am dialing 19077273299
02:52.51drraydial 9 to get out
02:52.51spackleSweet.  I'm actually watching a recording.  That gives me incentive!
02:52.58syle:)
02:53.01hardwiredrray: thats my point I am making to DrukenME
02:53.04hardwireerr
02:53.04hardwireyeh
02:53.07drrayheh
02:53.09cyaltrneone have experience with DID on x100p cards
02:53.19spackleI can make my TV lie to me even more ;-)
02:53.20DrukenHMEif you dial 190777473486485 it won't match exten => 1,### now would it?
02:53.23hardwirecyaltr: there is no did?
02:53.35cyaltronly 400p
02:53.47hardwireDruken: pause?
02:53.47syleoww dude want to hear something funny, i did this to my wife last night, first time her seeing it...
02:53.57*** join/#asterisk Johnsie (n=john@acs-24-154-91-33.zoominternet.net)
02:54.20cyaltri mean doesonly the tdm400p support
02:54.21DrukenHMEhardwire: XXX that better?
02:54.26sylemythtvosd --template=scroller --scroll_text='Weather Update: warning thunderstorms and severe hail developing during the night, possible tornadoes, did that get that you stupid fucker?!'
02:54.36sylefunny as hell to get people;s reactions
02:54.59hardwireDrukenHME: actually..e xplain how a sip phone that uses a dialplan that first matches 3 extensions.. then 7 then 10 to help it dial automagically would know the difference ibetween my example above.
02:55.14Cherebrumanyone using odbc for voicemail configs? It works cdr_odbc works fine but it seems to ignore my voicemail database
02:55.16hardwireit would automagically dial 190 while typing 19077273299
02:55.24spacklenice
02:55.25FaithfulDrukenHME: It works !!! Thanks
02:55.34FaithfulDrukenHME: what was I missing?
02:55.43DrukenHMEhardwire: well, my pap2 dials with a dialplan, i only allow it to dial 3 7 or 10 digit numbers
02:55.49hardwireok
02:55.52DrukenHMEFaithful: prolly the restart
02:55.55hardwireso how does it tell the difference?
02:55.56syleanyways, what do you use for a remote?
02:55.58*** part/#asterisk santiago (n=santiago@63.245.86.254)
02:56.11sylei was thinking those ird keyboards, where the remote learns from the keyboard
02:56.16NetgeeksI just tell people to use the sip phones like thier cell phone.. enter the number and press the 'send' button
02:56.17DrukenHMEhardwire: i guess that would depend on the device...
02:56.25FaithfulDrukenHME: So is the position of the immediate=yes an issue do you think?
02:56.27NetgeeksMost everyone knows how cell phones work
02:56.29spacklewireless keyboard.  It is in a constant state of tweak.
02:56.56DrukenHMEFaithful: it may have been... my thoughts atm are, if it works, leave it alone :)
02:57.08sylei was damn happy when i found this command : xvattr -a XV_COLORKEY -v 66048
02:57.16FaithfulDrukenHME: hee hee
02:57.35syleyour knopp thing probably had it included though
02:57.49FaithfulDrukenHME: when I am in music on hold I can press any buttons ... say to go to voicemail if I wan't to
02:58.05cyaltrhardwire:have you done DID
02:58.08spacklesyle, ATI makes a good remote (rf) that uses USB and emulates a KB, it's pretty nice.
02:58.37DrukenHMEFaithful: should be just a wait.. i dunno if it'll let you or not...
02:58.38sylemodel , url?
02:58.40DrukenHMEgive it a try
02:58.53sylehave to make sure its supported on lircd site to :)
02:59.00greekmananyone setup realtime queues yet?
02:59.36DrukenHMEok, well... i'm going to bed... 11pm, i got work in the morning
02:59.40spacklesyle, the hauphauge(sP) media MVP is a really neat basic front-end.  check out the project on sourceforge.
03:00.34cyaltrlooking for DID config help for analog lines ne1??
03:01.31spacklesyle, must be the ATI remote wonder, I see there is a newer model now.
03:02.45Cherebrum"odbc show" says that I'm connected to the database and "show config mappings" says that voicemail is mapped to odbc,asterisk,voicemail   as it should be. and I have a voicemail table in my asterisk database from the examples I found on the wiki but the damn thing still doesn't work
03:05.04spacklesad news : there is a VOIP for dummies book.
03:05.56Cherebrumspackle: the Avaya one?
03:06.14Cherebrumor is the full book out now?
03:06.23Ariel_anyone here know the settings for New Zealand country code? for zaptel
03:06.27spacklemust be a full book, saw it in B&N.
03:06.41CherebrumBecause Avaya released a mini VOIP for dummys book as a marketing thing
03:06.55fileCherebrum: YOU!
03:07.03Cherebrumand we contacted the publisher to see about creating our own VOIP for Dummies book
03:07.14spacklenah, this was a full book.  picked it up, burned my hands.
03:07.14Cherebrumbut they wanted too much of a commitment
03:07.23Cherebrumfile: what's up?
03:07.44fileoh, nothin
03:07.51spacklecherebrum - like a time frame, or a real book kind of commitment?
03:07.58*** join/#asterisk iq (n=iq@207-224-96-203.omah.qwest.net)
03:07.59Cherebrumnumber of books
03:08.05CherebrumI think it was like 10k books
03:08.21Cherebrumfuck a bunch of that
03:08.50spacklebefore comissions kicked in, or that you had to finance?
03:08.51Cherebrumfile: any progress on that polycom?
03:09.14Cherebrumthis was to have them publish custom VOIP for Dummies books
03:09.18fileCherebrum: yup
03:09.24Cherebrumfor us to give away to customers
03:09.46fileCherebrum: http://developer.berlios.de/projects/astmessaging/
03:09.52spackleI see
03:10.10Cherebrumkickass
03:10.34fileCherebrum: the framework is there, just need to like adapt more apps to messaging capability
03:10.39Cherebrumspeaking of Polycom.. mine is rining
03:10.44*** join/#asterisk BobDoleEm (n=a@brom-245-243.flexabit.net)
03:11.36spackleshhhh, stop talking about kram
03:11.41fileOMG GOATMILK!!!
03:11.54goatmilklo!
03:11.57BobDoleEmdoes anyone have a minute to answer a few questions about a asterisk install on fedora core 4?
03:12.03hathi, how many APIC for dell 2850 server?
03:12.46CherebrumCan someone PLEASE help me with my res_odbc problem?
03:13.01filegoatmilk: how are you doing?
03:13.04DarthClueJoony: depends on the questions
03:13.23goatmilkhangin in, but about to get some rest
03:13.31fileawwww :(
03:13.54sylehat, why haven;t you just gone to dell's site by now and downloaded the technical specifications chart on that server
03:14.50greekmananyone setup realtime queues yet?
03:15.04sylespackle, i do get mythmusic crashing on me occasionally
03:16.25sylei hear they are rewriting it though
03:17.11sylebiggest pain in the ass i had with mythtv was finding all the development libraries to install so i could compile it from latest cvs branch
03:17.17*** join/#asterisk spootnick (n=irc@50.118.233.220.exetel.com.au)
03:17.44spacklesyle, nothing's perfect.  I need to restart the box once a week.
03:18.20syleyou know i think mythtv is going to be sold on the xbox 360 :)
03:18.38*** join/#asterisk CoolAcid (n=jason@216.99.98.39)
03:18.42sylei can;t wait for dead or alive 4 personally lol
03:19.04CherebrumI've got one of those Motorola DTC-6208 HDTV DVR boxes and it's a certified peice of shit
03:19.20knight_personally I like SageTV better than MythTV
03:19.23CherebrumIt has firewire and you can copy the videos off it but it only works with Windows
03:19.26knight_I use to develop for mythtv for a long time
03:19.44spackleknight_, then thank-you.
03:20.33sylecan you rip dvd's with it?
03:21.14knight_you can rip dvds with SageTV addons
03:21.19knight_commercial detection
03:21.21knight_etc
03:21.26*** join/#asterisk chendy (n=Alex_Dot@web1.ningo.net)
03:21.27knight_lots of awesome features
03:21.42knight_only thing i wish it had was autorecompression
03:21.49knight_MPEG2->AVI
03:22.01goatmilkfile: i think your muffin made kram leave
03:22.16knight_mythtv became too much work of constant upgrading
03:22.25joonysamMCE2005!
03:22.28*** join/#asterisk viLeR (i=1000@ip-47-252.telesat.com.co)
03:22.28knight_and a million buggy things (atleast for me)
03:22.42syleplease dont; tell me sagetv is windows
03:22.49nick125sagetv is for windiw
03:22.51nick125windows
03:22.52nick125bleh
03:22.57sylehow gay lol
03:22.57knight_sagetv is linux too
03:23.07knight_as well as installable onto mce
03:23.18sylehttp://www.sage-community.org/index.cgi/wiki/DVD
03:23.25syleshow me where i rip dvd;s in linux here
03:23.39sylethose look like windows screens
03:24.20spacklesyle, mythvideo can rip DVDs
03:25.43syleyeah i know
03:25.54sylei;ve ripped a couple blockbuster movies over the weekend
03:26.02syle:)
03:26.11*** join/#asterisk wolfson (i=hehe@kdh-res-4.beachlink.com)
03:26.17spacklejust in case it burns down?
03:26.34fugitivosyle: that's illegal
03:26.56sylewell my wife rents videos and sometimes i don;t get around to watching them for a week later, so to avoid late fees i just rip them to view later
03:27.06spackletimeshifting?
03:27.24kramhi file
03:27.25Qwellkramage
03:27.29kramhi qwell
03:27.34*** part/#asterisk cyaltr (n=spam@66-188-89-49.dhcp.mdsn.wi.charter.com)
03:27.52Ariel_argh
03:28.10spootnicki'm getting that goddamn error msg saying "app_dial.c:1091 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown)" on asterisk 1.2.x
03:28.28spootnickdid anybody deal with this already when using x100p clones?
03:28.37spootnick(my configs http://pastebin.ca/22262)
03:28.44*** join/#asterisk alex_mx (n=Alex@dsl-200-67-125-45.prod-empresarial.com.mx)
03:28.44Ariel_anyone here worked with tdm04b for setup in New Zealand settings?
03:28.57kramwe finally got the katrina board live
03:29.03kram+1-248-724-2266
03:29.35t3twhat's that?
03:29.36goatmilkfile: night
03:29.52sylehow does sagetv work with asterisk?
03:30.33t3tkram, what's the katrina board?
03:32.12harryvvkatrina board?
03:32.39alex_mxhey everyone...need some help debugging a problem which seems to be IAX related
03:32.52t3talex_mx, you've come to the right place
03:32.56kramcall it and see
03:33.07t3twouldn't you like that...
03:33.39t3t...watch my phone'll start playing nursery rhymes and turn purple...
03:33.43*** join/#asterisk newsmafia (n=newsmafi@wsip-68-15-19-142.sd.sd.cox.net)
03:33.53kshumard_homehey it sounds like my voicemail. : )
03:34.04alex_mxmy servers hang every once in a while with log full of "Dropping voice to exceptionally long queue on..." anyone seen this before
03:34.05spacklekram, pretty slick.
03:34.46spacklehow many channels will the katrina board support?
03:35.00kram3 or 4 PRI's
03:35.11drumkillakram is my hero
03:35.37t3talex_mx, are you doing voice mail to email?
03:35.44CherebrumBinding voicemail to odbc/asterisk/voicemail
03:35.52Cherebrumdoesn't seem to work tho
03:36.29alex_mxt3t: i thought at first that might be the problem but one of the servers has no voice mail to email
03:37.10alex_mxt3t: but it does have a iax trunk to a server that does
03:38.10harryvvkram, is this database tied with the red cross or fema?
03:38.13jayk-i'm trying to set up voicemail transfer with asterisk. i added exten => *1XX to extensions.conf, but when i press the asterisk key and then the extension, it doesn't go through.
03:38.32*** join/#asterisk Beccara (n=Beccara@222-152-27-117.jetstream.xtra.co.nz)
03:38.33jayk-does anybody know why it wouldn't go through? i think the * is being filtered out or not being passed properly.
03:38.35t3tcool, kram.  how are you moving the data from the source to * and back?
03:40.07harryvvkram, im amature radio operator for Amature Radio Emegency service and would like to know if you have contacted the red cross if thay would be interested in this?
03:40.30kramthe red cross is apparently very insterested in the app
03:40.38kramnow that we have it on live servers we'll see what they want to do
03:40.44t3talex_mx, I assume that it's something which is not processing iax frames as fast as your server wants it to, but that's just idle speculation... I'm sure others could answer this one better.
03:40.51spootnickdid anybody deal with asterisk being "unable to create channel of type zap (cause 0 - unknown)" when using x100p clones?
03:40.59spootnick1.2.x branch
03:41.46harryvvkram..this may be excellent pr for you..btw, I read a note that dell,cisco systems, and two other companies are in the process of putting in wifi and voip for the red cross shelters. I saw this on zdnet.
03:41.46t3tjayk-, what type of sip/iax client are you using?
03:41.54jayk-cisco 7960
03:42.55t3tjayk-, check dialplan.xml to make sure that *xxx will be passed to *
03:43.43t3tjayk-, I just use <TEMPLATE MATCH="*"            Timeout="5"/> in mine
03:43.51jayk-ok
03:43.54jayk-let me try that
03:44.04jayk-i don't have an * in there, i did try "*1.." but that didn't work
03:46.10t3tkram, so what's the secret?  What's the best way to move data to and from calls?  I'm trying to decide whether to use DBget/put or AGI...
03:47.19Corydon76-homeDBGet/DBPut are deprecated
03:47.33t3tthat answers that then
03:47.41t3tthanks Corydon76-home
03:47.44Corydon76-homeWe now have the DB dialplan function for that
03:48.09t3tis it better at this stage to use what replaced them (what did?) or agi?
03:48.10Corydon76-homeOr if you want cutting edge, you'll check out bug 5055
03:48.26t3tI bleed enough naturally, thanks
03:48.27Corydon76-homeAGI incurs a startup penalty
03:48.36t3tof how much?
03:48.47Corydon76-homeWell, it's a forked process
03:48.53Beccaraanyone got any experence with the gsm codec and x-pro?
03:49.02Beccaraasterisk dosnt want to use it
03:49.12jayk-t3t: that didn't work either...
03:50.58*** join/#asterisk bmg505 (n=leon@rndf-146-43-80.telkomadsl.co.za)
03:51.47*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal)
03:51.51alex_mxanyone else that might have 2 cents on my iax problem?
03:52.13hatsyle, http://support.ap.dell.com/docs/systems/pe2850/en/ug/t1390aa.htm#wp1043338
03:52.18t3tjayk-, are the calls reaching the * box?
03:52.24hatPCI volt is 3.3
03:53.08*** join/#asterisk apardo (n=apardo@23.Red-83-44-179.pooles.rima-tde.net)
03:54.01jayk-t3t: i don't see anything outputted on the console when i dial *120, but if i dial 120 it rings to my other phone
03:54.09jayk-so yeah, they are reaching it, if i don't start with an *
03:54.29t3tjayk-: it's probably the dialplan on the phone then
03:54.40t3tdid the new config file get read by the phone when you rebooted it?
03:55.00*** join/#asterisk greekman (n=alex@host254.209.113.199.conversent.net)
03:55.53t3tCorydon76-home, so what are the alternatives for pulling/pushing data from/to a database?
03:56.38*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
03:57.48jayk-yeah, it did
03:57.51jayk-let me reboot again, though
03:58.01jayk-<PROTECTED>
03:58.02jayk-thats'
03:58.05jayk-that's what i added
03:58.33t3tis the phone reading the config from the server -- look in the tftp logs...
03:59.28Corydon76-homet3t: there's a few
03:59.37Corydon76-homeRealtime, for one
03:59.54Corydon76-homebug 5055 for another
03:59.58jayk-t3t: yep, Sep  7 03:59:54 asterisk in.tftpd[11392]: RRQ from 67.168.55.181 filename RINGLIST.DAT
04:00.05jayk-oops
04:00.05jayk-Sep  7 03:59:54 asterisk in.tftpd[11393]: RRQ from 67.168.55.181 filename dialplan.xml
04:00.19t3tCorydon76-home, realtime can read/write arbitraty data from/to a DB?
04:00.34Corydon76-homeThe DB function works fine for access to the astdb
04:00.56Corydon76-homeNo, only 5055 can do that
04:01.01t3tjayk-: any for filename dialplan.xml?
04:01.06Corydon76-homeRealtime is for very specific data
04:01.21t3tThat's what I thought
04:01.37jayk-t3t: what?
04:01.53Corydon76-homeYou're welcome to try 5055.  I spent most of this afternoon testing it
04:01.56t3tjayk-, sorry, typing to Corydon76-home
04:02.54jayk-could it be something to do with the dtmf?
04:03.04t3tCorydon76-home, sorry, i messed up, I typed dbget/put and meant mysql()
04:04.10Corydon76-homeWell, you could always code a custom app using the unixODBC interface
04:04.20Corydon76-homeI've done that a few times
04:04.25t3tCorydon76-home, I'm trying to read/write information for managing information like what kram did for the phone numbers
04:04.44Corydon76-homeHow much SQL do you know?
04:04.51t3tmost of it
04:05.14Corydon76-homeAre you doing complex queries where you need more than one row from the result set?
04:05.36Corydon76-homeBug 5055 is designed for queries which return one row
04:05.37t3tjayk-, probably not the dtmf... it's most likely your dialplan on the cisco.  is this a personal test phone?
04:05.54t3tCorydon76-home, probably just one row
04:06.13Corydon76-homeThen I would strongly encourage you to try out bug 5055
04:06.18jayk-t3t: nope, its a business phone. none of them pick up the *
04:07.05t3tok, then replacing the dialplan with just * probably isn't an option...
04:07.31jayk-i can try that as a test
04:07.35*** join/#asterisk eNd (n=tR@144.135.253.19)
04:07.53t3tmake sure that it's getting pulled off the tftp server after you change it
04:08.15t3tCorydon76-home, thanks.  I'll look into it.  What's the advantage of using this over mysql() ?
04:08.46spootnickshouldn't i get a dialtone after everything is properly configure with x100p?
04:08.54Corydon76-homeYou can prewrite the queries and only plugin the items that change within the dialplan
04:08.59spootnick(when picking the phone up...)
04:09.12t3tspootnick, x100p is fxo not fxs
04:09.33Corydon76-homeand it's good abstraction.  See the presence example
04:10.04spootnickt3t: hmm, dunno exactly the difference. i got it installed. but i can call it without getting a "unable to create channel of type Zap"
04:10.06t3thow much additional overhead is there vs. mysql?
04:10.37Corydon76-homee.g. ${ODBC_PRESENCE(103)} or Set(ODBC_PRESENCE(103)=123)
04:10.44*** part/#asterisk alex_mx (n=Alex@dsl-200-67-125-45.prod-empresarial.com.mx)
04:10.55t3tx100p is for connecting to a phone line from the telco = fxo
04:10.57Corydon76-homet3t: it's the overhead of unixODBC
04:11.11Corydon76-homeYou'll need to set up res_odbc.conf
04:11.57spootnickt3t: hm, so this card should never be used to plug a standard phone into it?
04:12.05t3tCorydon76-home, I'll have to take a look at that then.
04:12.16Corydon76-homeand if you haven't already, you'll need to configure odbcinst.ini and odbc.ini... for MySQL access, install MyODBC after installing unixODBC
04:12.21t3tspootnick, it won't work that way
04:12.50spootnickt3t: which card would do it?
04:12.55t3tCorydon76-home, i try to avoid odbc... bad experience as a child.  I'm sure it's changed since then
04:13.01spootnickis there a "2 way" one?
04:13.22Corydon76-homeNah, but the benefit of ODBC is a single interface for all databases
04:13.48sylewhen someone says bad experience as a child , that makes them early 20;s usually
04:14.21t3tspootnick, you need an FXS port on a card... the digium TDM10B is one of them
04:14.39t3tsyle, nearly 30, but close
04:14.58Corydon76-homesyle: so you're saying I could fool someone into thinking I'm in my early 20s?
04:15.04t3tCorydon76-home, but it's not a benefit if you only use one :)
04:15.21t3twelcome back to your real nick, jbot
04:15.37t3twhat... no whitty response?
04:15.39Corydon76-homet3t: it's a benefit when Mr Bossman says you have to use SQL Server
04:15.44spootnickok, so what the phone port on x100p means, if i can only use my x100p card to connect my PBX to a telco?
04:15.52sylehaha
04:16.03t3tCorydon76-home, then you save yourself a whole lot of hassle and find a new bossman...
04:16.17Corydon76-homeSo you install FreeTDS with tdsver=4.2 and be done with it
04:16.23sylei had a company tell me to install SQL server once, i told them they'd run shit properly or i;d quit
04:16.26sylethey ran mysql
04:16.28t3tspootnick, it means it's just a pansy ass modem that someone hacked to do something it was never designed to do :)
04:16.45Corydon76-homesyle: not Postgres?
04:16.58syleback then postgres was really unstable
04:17.10t3tsyle, I've done the same a few times.  It feels good... even better a few years later when things are still running smoothly
04:17.10Corydon76-homeAh, yes, the Dark Ages
04:17.18spootnickt3t: lol. i found a whole bunch of tutorials on configuring x100p clones to dial outside numbers. so i thought i could plug a standard phone in that hack
04:17.21spootnickhack=jack
04:17.26harryvvmmm I have disa setup to except a numerical password but nothing happens. anyone work with disa before?
04:17.32spootnick(outside numbers = pstn)
04:17.59jayk-t3t: ok, i don't think its the dialplan.xml. i have only the "*" and it still doesn't work.
04:18.03harryvvor trunk
04:18.05*** join/#asterisk drbrown (n=chatzill@user-0cdveep.cable.mindspring.com)
04:18.28t3tjayk-, and you are seeing that dialplan.xml is being downloaded from the tftp server by the phone?
04:18.52t3tharryvv, how did you set up disa?
04:19.03sylewell owners respond well to showing them the PROs of unix over the cons of windows, if you say it in such a way that they are complete idiots then your asking for trouble
04:19.24drbrownanyone familier w/ running cgi scripts on apache????
04:19.48drbrownI keep getting a 403 forbidden
04:20.07drbrownI know it must be something very simple
04:20.08syledid you setup a cgi bin directory in your apache.conf?
04:20.08jayk-t3t: yup
04:20.21spackledrbrown, permissions?
04:20.27syledid you check owner of apache process can read, execute in that dir etc
04:20.34drbrown<PROTECTED>
04:20.43spootnickdrbrown: drbrown check error_log. there's a few possible culprits to tell you a single solution
04:21.14Corydon76-homeYou probably forgot to set up a ScriptAlias to that directory
04:21.15spootnickno perms, data going through before headers
04:23.33drbrownerror attempt to invoke directory as script: /usr/lib/cgi-bin
04:24.03spootnicksomething pointing somewhere it shouldn't...
04:25.56spootnickdo i need more than a fax modem to have asterisk receiving faxes?
04:26.24jayk-t3t: any other ideas?
04:27.32*** join/#asterisk loud (n=ariel@cypher.punk.net)
04:27.42FuriousGeorgespootnick: why wouldnt it be done with an fxo/fxs like other POTS stuff?
04:27.56FuriousGeorgei dont know, as ive never thought about it or tried it
04:28.40spootnickFuriousGeorge: no idea  =)  but it just sounded reasonable that you're able to receive faxes with a fax modem and asterisk
04:29.34spootnicki was wondering what happens when you try to send a fax over a sip line
04:29.38spootnickor receive
04:29.45FuriousGeorgeive heard that modems are like puny fxo, so maybe you can jerryrig it
04:30.10FuriousGeorgespootnick: unless you use a lossless codec, from what i hear, the fax doesnt go through
04:30.33FuriousGeorgenever tried, but i would think gsm or ulaw would be out of the question at the very least
04:34.56spootnickt3t: x100p gives no errors now when trying to dial via my regular telco line. but i don't get any sound or whatever
04:36.27drbrownthx guys
04:37.58*** join/#asterisk oplog2 (n=oplog2@206.222.29.50)
04:39.58bkw_muhahahahhahahaahhaahahahahahahahaahahah
04:40.37newmemberany recomendations for a high density FXS ports for *? for a service provider type application
04:41.16*** join/#asterisk transgress (i=transgre@71.14.20.160)
04:41.28twistedhey bkw
04:41.32twistedhttp://www.hitachigst.com/portal/site/en/menuitem.92a9817d39355921fce80bd5eac4f0a0/
04:41.37twistedyou'll get a kick out of that
04:42.10*** join/#asterisk criptos (n=criptos@201.135.121.4)
04:42.18criptosHello all!!!!!!
04:42.44*** join/#asterisk Defraz_ (n=t0tal@24-119-12-238.cpe.cableone.net)
04:42.45Netgeekstwisted: did you get the email this time?
04:43.02*** join/#asterisk newmember (i=user@S010600036d1139fb.cg.shawcable.net)
04:43.06twistedNetgeeks, yea
04:43.10Netgeekscool
04:43.35tqnas noches
04:43.38tqgood nights
04:43.46*** part/#asterisk spackle (n=spackle@209.234.83.19)
04:43.51Netgeekslotsa stuff there, and I have some cash to cover the work
04:45.43criptosbuenas noches :)
04:49.07*** part/#asterisk tq (n=tq@200.117.234.254)
04:49.35bkw_FYI guys NEVER stay at the hilton in Romania
04:49.47bkw_400 dollars USD a night
04:50.20bkw_435 a night if you want breakfast
04:50.25criptos400  blond included?
04:50.33Qwellcriptos: it is Romania
04:50.41Daminbkw_: Does it make you sick to know that I just downloaded the entire astwind installer over my Cell phone in less than an hour? :)
04:50.42bkw_I doubt it
04:50.51bkw_Damin, WHAT?
04:50.51bkw_haha
04:51.10Daminbkw_: I got the USB <-> EVDO shit figured out..
04:51.11bkw_I'm testing international routes by call Hiltons all over the planet
04:51.24bkw_so far I have had excellent call quality
04:51.40Himekomuch cheaper places to stay in romania
04:51.55file[laptop]Damin: I have yet to get mine connected over USB or Bluetooth to my Powerbook
04:51.55Daminbkw_: Ask for Paris, and tell her to get her scrawny ass up to my room pronto! Who does she think I am?
04:51.57bkw_Himeko, I suspet so
04:52.03bkw_Damin, I already called the Paris Hilton
04:52.13Daminfile: Yeah.. i think you might be screwed on the MAC side..
04:52.34bkw_file even if you add it as a serial port?
04:52.36file[laptop]Damin: tried it on my workstation, wouldn't stay connected
04:52.43Daminfile: I've got a little application called "Wmodem" that I run on the XV6600 and then some wacked out CDMA USB Dielar..
04:52.53file[laptop]Damin: yeah I tried that
04:52.58Netgeekshahaha for one NANOG, I booked a room in the miami hilton via the 1-800 hilton number, when I got there they didn't have a reservation for me, it turns out the 800 hilton folks booked me into the hilton in Bucharest....
04:53.09file[laptop]Damin: I had my workstation connected to my data service, but it lasts for about a second - then disconnects itself
04:53.12Daminfile: Don't know what to tell you.. I couldn't get it to work at all in Chicago, but in my Kitchen it rocks..
04:53.17file[laptop]and during that second I can ping fine
04:53.34file[laptop]rather evil
04:53.37bkw_Damin, haha
04:53.46bkw_you would think it would work in Chitaco
04:53.56DaminHehehe..
04:54.14file[laptop]once October hits my cellular bill is radically changing
04:54.19Daminbkw_: We cracked another 1,000 registrations from the Slashdot article for Linuxfest. :)
04:54.32dudesbkw_ - so what's the sip auth issue ... cause I clearly can't figure it out
04:54.47Daminbkw_: Did you see that Kevin beautified chan_spy?
04:56.33Daminbkw_: And did I also tell you that we solved a DS3 problem today by removing the 3 foot cable and putting in a 100 foot cable? :)
04:56.42bkw_Damin, hahah
04:56.48nick125lol
04:56.48bkw_thats funny
04:56.57Daminbkw_: Hehehe..
04:57.01bkw_its always shit like that
04:57.13bkw_I have been working since 8am
04:57.15bkw_wtf is wrong with me
04:57.17bkw_i'm still working
04:57.21bkw_and I AM ENJOYING IT
04:57.21bkw_wtf
04:57.23bkw_:P
04:57.24dudesbkw_ - I have some debug info if you have the time
04:57.29bkw_dudes, for?
04:57.33bkw_is this for an asterlink account?
04:57.38Daminbkw_: We kept getting LVC errors on this, and after replacing EVERYTHING on both sides, we finally decided to lengthen the cable..
04:57.50bkw_Damin, thats a good one to keep in the notebook
04:57.50DaminScrew it..
04:57.54DaminI'm going to bed! :)
04:57.57bkw_l8tr
04:58.06Qwella longer cable worked?
04:58.06bkw_file IRC's from bed
04:58.11Qwellwas the short cable bad or something?
04:58.15file[laptop]I just love you guys THAT much
04:58.16Daminbkw_: Yeah.. Attenuators helped a bit, but nothing like good old lossy cable!
04:58.22*** join/#asterisk [hC] (n=hardcore@c-24-127-192-210.hsd1.fl.comcast.net)
04:58.23bkw_ya
04:58.28bkw_the signal was just too much
04:58.29DaminQwell: Nope. The short cable was fine..
04:58.39Qwellcable too short can be a bad thing?
04:58.41Corydon76-homeThat, or he's jacking off to the channel
04:58.45Daminbkw_: Well, we had a 12db attenuator on the fucking thing!
04:58.47file[laptop]Corydon76-home: you wish
04:58.47QwellCorydon76-home: tmi
04:58.50DaminQwell: Yep..
04:58.51bkw_haha
04:58.55QwellDamin: thats scary
04:58.59dudesbkw_ - for the sip auth issue
04:59.02dudesand callerid
04:59.10DaminQwell: Same thing applies for some Ethernet issues..
04:59.11Qwelllike...you can't even debug something like that
04:59.16Corydon76-homefile[laptop]: this is far from what I would consider erotic.
04:59.23MikeJ[Laptop]how the heck do I do a cvs diff -u that picks up new files?
04:59.24dudesbkw_ - can I PM you the address to my log
04:59.27DaminQwell: Yes you can.. You just have to drink a lot and come up with off the wall idas!
04:59.30bkw_dudes, what sip auth issue?
04:59.32MikeJ[Laptop]cvs diff -Nu does not work
04:59.34bkw_I don't know which one you're talkin about
04:59.42bkw_-NuR
04:59.45MikeJ[Laptop]thanks
04:59.45bkw_or -Nur
04:59.47file[laptop]bkw_: dudes reminds me of our support aggents
04:59.48file[laptop]er agents
04:59.49Corydon76-homebut, hey, I've known guys do MUCH stranger things
05:00.15bkw_I can tell you my love for you will still be strong after the boys of summer have gone
05:00.27file[laptop]bkw_: okay that's freaky, I was just listening to that
05:00.27bkw_file you bitch you have me stuck on this song now
05:00.36dudesbkw_ - the callerid shows as "asterisk" ... and in the CLI output it shows 127.0.0.1
05:00.41bkw_I have three versions of this its been playing for 10 hours
05:00.44DaminYAY!
05:00.54DaminVerizon has a Software Update for my phone!
05:00.54bkw_dudes, sip auth.. I think you need RPID boi
05:01.04DaminNow maybe it won't need to be rebooted so often!
05:01.04bkw_Damin, watch it.. they might disable shit
05:01.21MikeJ[Laptop]not working
05:01.27QwellMikeJ[Laptop]: -r head?
05:01.39*** join/#asterisk spootnick (n=irc@50.118.233.220.exetel.com.au)
05:01.42Daminbkw_: According to the notes, this fixes a lot of weird Bluetooth stuff..
05:01.45MikeJ[Laptop]new file is not showing up in the diff
05:01.53bkw_Damin, kewl
05:01.58dudesbkw_ - I don't know what to do.  I've played with this issue for a few weeks and can't figure it out.  ...
05:01.58Daminbkw_: Which may hopefully make my Morotoal ShitPiece work better..
05:02.08bkw_dudes, sip auth and callerid don't mix
05:02.15bkw_you need Remote-PartyID support
05:02.20bkw_and without a patch that isn't possible
05:02.24bkw_its on the bug tracker
05:02.28dudesYou said it was a auth issue two weeks ago
05:02.28bkw_otherwise the callerid is whacked
05:02.45dudesDo I need to pull up a log?
05:02.55bkw_is this for an asterlink account?
05:03.01dudesNoi
05:03.08bkw_well I don't recall this
05:03.17bkw_your doing asterisk to asterisk SIP
05:03.18bkw_right?
05:03.31spootnickif x100p is a FXO card, how come i registered it as fxsks=1 on zaptel.conf and i get a "Registered channel 1, FXS Kewlstart signalling" on asterisk console?
05:03.47bkw_spootnick, duh
05:03.51dudesbkw_ - I'm getting calls from commpartners
05:03.54bkw_it wouldn't even make it that far if it didn't work
05:04.11bkw_dudes, I reacall this
05:04.18bkw_this is inbound from commpartners
05:04.18dudesthat's a plus
05:04.22dudesyes
05:04.23bkw_to asterisk?
05:04.27dudesyes
05:04.43bkw_pastbin me the inbound invite
05:05.04dudesI can message you a txt on my webserver if that's ok
05:05.09bkw_yes
05:05.14criptosDoes a budgetphone ata can be locked?
05:05.16t3tspootnick, you use fxs signalling on an fxo card... it's stupid telco stuff from last century
05:05.21criptosTo avoid config to be changed?
05:05.24file[laptop]INVITEs are overrated, I much prefer BYEs!
05:05.25bkw_criptos, put that into english
05:05.59criptosbkw_ humm... :P
05:06.09t3tjayk-, you still there?
05:06.32bkw_dudes, BroadWorks sip on their side
05:06.33bkw_give up now
05:06.39bkw_this is going to be a bitch
05:06.50bkw_I still can't believe they are trying to auth inbound calls to you
05:06.52bkw_thats just daft
05:06.56file[laptop]O.O
05:07.02criptosCan I "lock" the configuration of a budgetphone ata, to avoid losing or changes to it, even a "hardware reset"?
05:07.05file[laptop]I gotta see this
05:07.06bkw_file dear do you think you can look at this
05:07.11file[laptop]yeah hand it over
05:07.23bkw_criptos, CALL GRANDSTREAM BEFORE I TWIST YOUR LITTLE HEAD OFF FOR ASKING THE SAME DAMN QUESTION TWO TIMES IN 5 SECONDS!!
05:07.35MikeJ[Laptop]any other suggestions other than -NuR?
05:07.45bkw_I msged the stuff to file dudes
05:07.47bkw_he's our sip dude
05:07.49twistedbkw_, midol.
05:07.54bkw_haha
05:07.58bkw_996?
05:08.03dudesbkw_ - thanks
05:08.05twisted996?
05:08.07Vcow00t
05:08.08criptosBkw_ sure :) Come to mexico, my door is wide open to visitors :)
05:08.17bkw_:P
05:08.21twistedhrm
05:08.23dudesThis isn't a important issue but it bothers me that it doesn't work
05:08.24dudeshaha
05:08.46bkw_dudes, you do realize for me or file to really look at it closer than those sip debugs its 200 dollars an hour
05:08.52bkw_we can get in and make it work
05:08.53file[laptop]this debug hurts my head
05:09.10t3tI'll do it for 199/hr :)
05:09.15bkw_honestly they shouldn't try to auth inbound calls
05:09.17bkw_thats just not right
05:09.28file[laptop]not for a provider to do.
05:09.30criptosI will doit for nothing... but I give no guaranteee :)
05:09.48bkw_criptos, i'm just messin with ya yo
05:09.53dudesPerhaps it's not but I'm not one to bitch
05:10.23twistedokay wtf
05:10.25file[laptop]looky at that, they do t38
05:10.30twistedi pick up the phone
05:10.31twistedget dialtone
05:10.32twisteddial
05:10.34twistedstill dialtone
05:10.37criptosbkw_ I´m surfing at grandstream site :)
05:13.07file[laptop]yeekz I just thought the blanket at the end of my bed started moving
05:13.11file[laptop]silly eyes
05:13.20dudesbkw_  -  I'd like to clear this issue for testing purposes ... since  you know my sites adderss you know why.
05:14.16file[laptop]my face feels so weird, I shaved today... I had a beard before, was rather evil...
05:14.28file[laptop]EvilFile(tm)
05:14.48dudesI always have a short beard cause I rash out when I shave with a razor
05:15.07file[laptop]twisted: you! guess what I remembered today
05:15.34dudesyou forgot to wipe
05:15.42file[laptop]dudes: quiet you
05:16.03dudesYou should laugh to that
05:16.19file[laptop]time after time!
05:16.47dudesThat's a song
05:16.53file[laptop]yes, it is
05:17.37dudesHow old are you file[laptop]?
05:17.45file[laptop]18
05:17.47bkw_OK this is too fucking funny
05:17.47bkw_[Description]  SIPAddHeader(Header: Content)Adds a header to a SIP call placed with DIAL.Remember to user the X-header if you are adding non-standard SIPheaders, like "X-Asterisk-Accuntcode:". Use this with care.Adding the wrong headers may jeopardize the SIP dialog.Always returns 0
05:17.53twistedbkw_, i'm in 996 now bitch
05:18.09file[laptop]I wonder...
05:18.11dudesfile[laptop] - you're in college?
05:18.11bkw_X-Asterisk-Accuntcode
05:18.15bkw_Accuntcode
05:18.40file[laptop]dudes: no, I work for the same company that bkw does
05:18.44twistedwow
05:18.46drumkillafile[laptop] is a slacker!
05:18.48twistedmaybe it's time to remove the cunt from sip
05:18.51dudesDigium
05:18.52Qwelldrumkilla: !
05:18.52twistedthat would perhaps fix that.
05:18.58file[laptop]nope, not Digium
05:19.00Netgeeks!!!
05:19.11QwellNo wonder SIP sucks.  Its all cunted up.
05:19.17Qwell<crappy joke>
05:19.24UberbotHi all.
05:19.27Qwelljust forget I said that
05:19.42dudesfile[laptop] - well I suppose i didn't get the memo
05:20.12dudesWhat company is that
05:20.17file[laptop]Asterlink
05:20.59UberbotMy zap card is showing a red alarm....  And now I don't have dial tone....  :'(
05:21.25*** join/#asterisk iswm (i=iswm@unaffiliated/iswm)
05:21.27UberbotI don't have tone at the d-mark....
05:21.32drumkillacoincidence?!?!  I THINK NOT!
05:21.42dudeshttp://www.asterlink.com/ ???
05:21.44file[laptop]drumkilla is a sillyhead
05:21.54file[laptop]dudes: yes
05:22.27dudesAsterlink solutions provide companies at all levels the answers to increasing productivity,
05:22.32dudesthat's not a complete sentence
05:22.44file[laptop]do you think I did that site?
05:22.52Qwellthat is a complete sentence
05:22.55dudesPerhaps my english major makes ...
05:23.04drumkillait is ... just a poorly formed one
05:23.06Qwellits missing a comma
05:23.23dudesI was just saying cause I see that as ...
05:23.26drumkilla"at all levels" messes it up
05:23.46twistedhey drumkilla
05:23.48twistedgot any cunts in stable?
05:23.50dudesJust being helpful
05:23.54dudesdon't hate me for that
05:23.54drumkillatwisted: always
05:24.40twisteddrumkilla, okay, cuz i just flushed the one in head
05:24.41twisted:P
05:25.03dudesNot like I come in here asking for help much or anything.  gotta love google and the wiki's
05:25.11dudesso take the help and be happy
05:25.38drumkillai don't think anyone complained :)
05:26.54twisteddrumkilla, bkw did
05:26.59twistedyou know how he feels about those parts
05:27.06drumkillaah.
05:27.19bkw_hahahahah
05:29.12dudesI'm not being mean, but whoever designed that welcoming paragragh needs to take a english class
05:29.46Qwellan
05:29.58file[laptop]Qwell: you're silly
05:30.08dudesI can read further if you'd like
05:30.29*** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl)
05:33.40*** join/#asterisk oej (n=Olle@apollo.webway.se)
05:36.09*** join/#asterisk Johnsie (n=john@acs-24-154-91-33.zoominternet.net)
05:38.38*** join/#asterisk mich (n=aa@200.72.184.80)
05:38.58*** join/#asterisk hellagony (n=egutierr@200.121.157.25)
05:41.28*** part/#asterisk criptos (n=criptos@201.135.121.4)
05:43.03twistedlol
05:43.10twisted[adult swim] never ceases to amuse me
05:43.14twistedvava con huevos!
05:43.25file[laptop]twisted: ...yes
05:43.41*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal)
05:44.10twistedwhich roughly transltes to
05:44.16twistedit goes well with eggs
05:44.32file[laptop]very rough
05:44.42michhola
05:45.09michhi
05:45.09twistedel fichero aspira mis bolas
05:45.17*** join/#asterisk Rainer (n=Rainer@p508841F3.dip0.t-ipconnect.de)
05:45.22twistedhi
05:45.26Rainerhi
05:45.29Supaplexhi
05:45.29twistedhi
05:45.34michdo you
05:45.37twistedno
05:45.38twisteddo you?
05:45.39michspeak spanish
05:45.53Rainerno german and english
05:45.56michor only eng
05:46.00michah ok
05:46.03michno problem
05:46.11twistedno, solamente inglés
05:46.18michgood
05:46.23michspanglish
05:46.37michsee, i a quesion
05:46.44michquestions
05:46.53michabout *
05:47.05michand x100p echo
05:47.07twistedlos conejos peludos grandes comieron mi calcetín
05:47.12michproblem
05:47.17michwhat??
05:47.23Supaplexe. nung gung lung eye sung hung
05:47.30Rainerwelches problem ???
05:47.41michecho cancel
05:47.46michat x100p card
05:47.54michi dont reduce
05:48.00michthe efect
05:48.02Qwellsay as much
05:48.03Qwellone one line
05:48.05*** join/#asterisk Legend (n=legend@24.244.142.133)
05:48.06Qwellas possible
05:48.08michyes
05:48.11michone line
05:48.19doolph?
05:48.24michbut , with tdm01b
05:48.27twistedla tarjeta de x100p es convertidor a digital analogico. Usted experimentará generación de eco.
05:48.38doolphwtf u talking
05:48.40michsi
05:48.47michhya un eco
05:48.50michmolesto
05:48.54michcuando
05:48.59Qwelljbot: translate en es Please say as much one one line as possible.
05:49.00michse conecta
05:49.04michok
05:49.12michwait please
05:49.35michtengo eco cuando hago llamadas desde un anexo ip hacia la red pstn
05:49.45*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal)
05:49.48michusando la tarjeta X100p o la tarjeta TDM400p
05:49.58michok no problem.
05:50.58michhe tratado de reducir el eco con ztmonitor
05:51.09michpero no obtengo resultados positivos
05:51.16michme comprende?
05:51.39twistedsi
05:51.41twistedcorrija, porque el canceller de generación de eco tiene que entrenar correctamente para poder prevenir generación de eco, puesto que usted no está terminando totalmente la línea.
05:51.44Rainerno sorry .... no spanish
05:52.20miches decir
05:52.31michentonces no depende exclusivamente de la tarjeta?
05:52.53Qwelljbot: translate es en tarjeta
05:53.14twistedmich, corrija
05:53.23michwait
05:53.25twistedes extremadamente difícil quitar generación de eco con eficacia al colocar llama del IP al PSTN usando tal tarjeta. Esta es la razón por la cual una línea digital circuito se prefiere.
05:53.56*** join/#asterisk Asylum (n=Asylum@dsl-58-6-126-60.qld.westnet.com.au)
05:53.57michok, entonces es preferible usar una card TE110P
05:53.58Rainerjbot: translate es en tarjeta
05:54.00michpor ejemplo
05:54.04twistedmich, si
05:54.16QwellRainer: I did that not but 90 seconds ago
05:54.34Rainerups sorry
05:54.39michok, pero entonces significa que el problema siempre va a existir
05:54.53michwith analog card
05:55.00CoiLcool spanish :)
05:55.05Qwelljbot: translate es en entonces
05:55.11AsylumHow would i setup the phones for only dialing local calls and nothing else on outbound 0 |. ?
05:55.28QwellAsylum: put them in a context with access to only local calls
05:55.29AsylumAnd then Mobile calls on 233 |. ?
05:55.39michtwisted: otra duda
05:55.51twistedcon las líneas analogicas, correctas. Usted podría intentar ajustar las configuraciones del impedence, pero no sé a haga eso, desafortunadamente.
05:55.59*** join/#asterisk huslage (n=huslage@c-24-22-125-179.hsd1.or.comcast.net)
05:56.04AsylumQwell, not exactly sure what you mena..
05:56.07Asylummean..
05:56.17michok
05:56.25QwellAsylum: take a look at the sample config.  Look how it seperates local and long distance
05:57.07twisteddoes anyone know how to adjust the impedence on an x100p/tdm01b?
05:57.13Asylumokies
05:57.15twistedanyone that could point mich in the right place?
05:57.25Qwelltwisted: en espanol en sap?
05:57.26michtwisted : tengo otro problema con TDM01B, al momento de colgar la llamada, esta tarjeta queda busy
05:57.33twistedQwell, *slap*
05:57.37Qwell:(
05:58.34*** join/#asterisk Rainer (n=Rainer@p508841F3.dip0.t-ipconnect.de)
05:58.40twisted¿está usted en los Estados Unidos? si no, es posible que la detección de la parada no está puesta en ejecucio'n para su país correctamente.
05:59.09Qwelljbot: translate es en ejecucio'n
05:59.10Qwellthat looks bad
05:59.14michno
05:59.19twistedrofl.
05:59.22michestoy en otro pais
05:59.33michother country
05:59.40twisted:)
05:59.55QwellDon't msg me
06:00.03jayk-i'm trying to configure asterisk in a way that i can transfer directly to voicemail. i have exten => *1XX in my extensions.conf, but asterisk isn't picking up the "*".
06:00.08jayk-anybody know how i can fix this?
06:00.13QwellI rarely do personal/private help for free...
06:00.28Qwelldon't means don't
06:00.48twistedusted puede intentar dar vuelta en "callprogress=yes" en zapata.conf si usted no tiene ya, pero puede no solucionar el problema.
06:00.48Asylumno problems.
06:01.03michok
06:01.06michentiendo
06:01.10Qwellcallprogress=yes ~= breakcallsrandomly=yes
06:01.28Qwellor so I'm told
06:01.39twistedQwell, true.
06:01.56Qwelltwisted: I haven't a damn clue what you told him though, so ignore me. :p
06:02.15Qwelljbot: autotranslate twisted es en
06:02.19twistedQwell, his card doesn't hang up properly when the line is disconnected
06:02.19Asylumexten => _0.3XXXXXXX     ? using this for my outbound, I can still make mobile calls, I only want local calls allowed. Can anyone help me out please?
06:02.35Qwelltwisted: isn't that one of the problems with the x100p?
06:02.38twistedQwell, so i told him he can TRY turning on callprogrss in zapata, but i can't guarantee it'll work
06:02.45twistedQwell, on the TDM01b is what he said for that one
06:02.46Qwelland, should he be using Hangup?
06:03.32jayk-anybody have any idea how i can get asterisk or my cisco 7960 to send *<EXT> to transfer to voicemail? for some reason, i don't think the * digit is being sent.
06:03.40QwellI don't know, I don't have much experience with analog lines
06:03.55twistedQwell, they are the work of the devil.
06:04.00bkw_jayk-, you use a transfer key boi
06:04.00Qwelltwisted: so I hear
06:04.03bkw_don't use the *codes
06:04.13jayk-why not *codes?
06:04.13*** join/#asterisk JerJer (n=JerJer@pdpc/supporter/bronze/jerjer)
06:04.30bkw_because its stupid to have a sip phone with a nice transfer feature and NOT USE IT
06:04.31Qwelltwisted: I'd like to mess with some of the T1/PRI stuff, but don't have the stuff available for screwing around. :(
06:04.41michthanks twisted, see you
06:04.51twistedmich, no problem :)  take care
06:04.59michok bye
06:05.05*** part/#asterisk mich (n=aa@200.72.184.80)
06:05.44jayk-i'd still be using transfer, i just want to use *EXT to send to voicemail rather than adding more number digits before the extension.
06:06.16twistedQwell, yeah, digital circuits are fun
06:06.18Qwelljayk-: send to voicemail when?
06:06.46jayk-well, rather than transfer to the extension and have it ring, i want to send directly to voicemail
06:06.55Qwellwhen?
06:07.10jayk-when somebody calls in
06:07.16jayk-and the person is not available
06:07.33*** part/#asterisk jaike (n=jaike@203.177.36.132)
06:07.38Qwellhow would if know they're unavailable, if it doesn't ring them first?
06:07.52Qwellit*
06:07.56jayk-if they are out of the office, at lunch, on vacation
06:08.06jayk-sometimes it easier to transfer to voicemail rather than to their extension
06:08.41Qwellright, but how are you going to set it so that it knows?
06:08.43bkw_thats what DND is for boi
06:09.02bkw_you can also do native transfers to *exten's
06:09.08bkw_just don't try to use the feature codes
06:09.15bkw_the phone might be URL encoding the *
06:09.18bkw_who knows
06:09.20bkw_check the sip debug
06:09.21bkw_i'm out
06:09.25bkw_LATER
06:09.46jayk-Qwell: exten => _*1XX,1,Voicemail(u${EXTEN:1})
06:09.46jayk-exten => _*1XX,2,Hangup
06:09.53jayk-that works
06:09.57twistedyeah
06:09.59twistedi'm out too
06:10.01twistedi think
06:10.07UberbotSo, how do I clear a red alarm on a zap device?
06:10.11jayk-except for the * part. if i change * to something else, it goes straight to voicemail
06:10.22twistedUberbot, fix the physical link.
06:10.30jayk-just can't get the * to forward through for some reason
06:10.39JerJerUberbot:  plug a live circuit into it, perhaps?
06:10.47Uberbottwisted, I no longer have dial tone.....
06:11.11twistedUberbot, well, is the circuit going to a channel bank?
06:11.28UberbotI've got a phone plugged directly into the dmark.  No, this is my home phone.
06:11.36twistedyou have a phone
06:11.40twistedplugged into a digital circuit?
06:11.59twistedor is this a tdm01?
06:11.59UberbotNo.  This is an analog line.
06:12.05Uberbottdm01.
06:12.08twistedoooohhh
06:12.15twistedyou have dialtone on the phone by itself?
06:12.21UberbotNo.
06:12.26twistedthen your line is foobared.
06:12.29twistedcall the phone company
06:12.33UberbotSo, is my phone just "out?"
06:12.41UberbotCold the tmd have caused this?
06:12.48twistedvery doubtful.
06:12.52Uberbots/Cold/Could/
06:12.59UberbotGood to hear.
06:13.14twisteddo you even get line voltage?
06:13.18UberbotJust a coincidence.
06:13.23UberbotI get hum...
06:13.45Uberbot...or static.
06:13.49twistedyeah, thats, uhm, bad.  sounds like the telco screwed up something
06:14.10UberbotSo, what's new?  It's Quest.
06:14.16twistedi'd call them and tell them exactly what you told me.  phone plugged into the customer test jack
06:14.22twistedstatic/hum, no dialtone.
06:14.32UberbotDoing that tomarrow.
06:14.36UberbotFrom work.
06:14.38twistedsince it's obviously not inside wiring, they'll have to fix it ;)
06:14.50UberbotMy house wiring is cat-5e.
06:15.25twistedyeah, but if you've got a phone plugged directly into the customer interface box, that means that even with your inside wiring disconnected you still dont' get dialtone
06:15.33UberbotYup.
06:15.48twistedthat's gonna be the qualifier for most telcos
06:15.54UberbotSure.
06:16.02twistedif you do at the box and don't at the house, they'll basically say 'screw you'
06:16.07twistedif you don't at the box, then they have to fix it.
06:16.15UberbotThat's how I understand it.
06:16.21twistedyep
06:16.23twistedhave fun :)
06:16.27UberbotAlways.
06:16.27twistedi'm off to bed
06:16.33UberbotThanx.  Good night.
06:16.49twistednp
06:16.54opus_hellp
06:17.03Rainercu ... bye ... have to read some asterisk Tutorials ... asterisk newbie
06:17.03UberbotWassup, opus?
06:17.23opus_hello Uberbot. whats new
06:17.46UberbotMy phone died. :'(
06:18.02UberbotTelco borked it.
06:18.15opus_shit,
06:18.25harryvvt3t fun
06:18.27harryvvwell
06:18.34opus_Uberbot where are you located\
06:18.37harryvvsorry Uberbot fun that is
06:18.39*** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se)
06:18.54harryvvtelcos normally disconect phone service for telco repairs late at night.
06:19.17*** join/#asterisk Manipura (n=chatzill@S01060011954c9c46.cg.shawcable.net)
06:19.44ManipuraWhy the heck is it so hard to install asterisk???? Anyone familiar with suse9.3?
06:19.54*** join/#asterisk zedkatuf (n=audela@82-33-115-145.cable.ubr08.azte.blueyonder.co.uk)
06:20.13opus_hey guys I'm starting a new VOIP gear company and wanted to introduce my new phone
06:20.15*** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1)
06:20.20opus_http://www.bambootrading.com/xmas/L234.jpg
06:20.39opus_its g729.1 compatible
06:21.32UberbotAlbuquerque, NM, USA.
06:22.15Uberbotopus, That looks like a GREAT phone!  Will it do Gig-E?
06:23.04opus_yeah
06:23.18opus_but we call it: gygerbit
06:23.22*** join/#asterisk spootnick (n=irc@50.118.233.220.exetel.com.au)
06:23.24opus_dude
06:23.37opus_I like this quote
06:23.48*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
06:23.49UberbotGreat.  I'll buy one.  Do you accept Susan B. Anthony Dollars?
06:23.49opus_Not until 1910 did the ICC receive sufficient budget to do any effective regulation of telephone service, and then it seems to have acted in more of an advisory capacity than in the investigative one which the Federal Communications Commission has assumed since its formation in 1934."
06:24.26UberbotThere's our FCC for you.
06:25.19opus_Although the AT&T Development and Research Department held out for nine more years before joining the Bell Labs, Thomas Watson's little laboratory had finally come into its own to be the largest and most effective industrial laboratory in the world.
06:25.37opus_1925
06:25.48opus_voip 2005 = at&t 1925
06:27.19*** join/#asterisk tengulre (n=tengulre@61.185.238.166)
06:31.05t3topus_ , sign me up... can I get it in off-white?
06:31.16t3tmaybe lime green?
06:31.23*** join/#asterisk af_ (n=af@ip-142-250.sn1.eutelia.it)
06:31.45JerJerhow about fire truck red?
06:31.58JerJer:P
06:32.06t3tmorning
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06:35.09t3tJerJer, did still doesn't work
06:35.22tuxinator_linuxMJerJer: You know why fire trucks are red, right?
06:36.28dudesis it the same reason bobcats aren't pets
06:37.20Qwelltuxinator_linuxM: why?
06:39.07drraybecause they are always russian?
06:39.37dudesHey don't  dog on russians
06:39.45*** part/#asterisk Rainer (n=Rainer@p508841F3.dip0.t-ipconnect.de)
06:43.34tuxinator_linuxMYou would be red too if someone was pulling your hose (grade school humor)
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06:47.05mwgbcIs there a way to adjust incoming volume or sensitivity for NVBackgroundDetect?
06:47.29Faithfulhow can I make it that users can jump to voicemail when listening to moh?
06:48.27Faithfulmoh command doesn't allow imput from the phone
06:48.38harryvvim outa here
06:48.38harryvv:)
06:49.01t3t'night
06:49.08harryvvhave a good night
06:49.08harryvv:)
06:50.09mwgbcFaithful: I'm kind of new myself, but what about using a long music file with somthing like BackGround?
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06:51.20t3tFaithful, you could play MOH for a time period, then prompt them to go to voice mail, then replay MOH in a loop
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06:52.59Manipuraanyone successfully instal zaptel with suse 9.3?
06:53.31t3tnot familiar with suse, but did you install the kernel header files for your kernel?
06:54.41hugo-v6gd morning
06:55.01ManipuraI know nothing about the kernel
06:55.05t3tmorning
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06:55.24ManipuraWith RH9 I just installed a few packages and after that everything went smoothly
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06:55.43t3tManipura, so you're trying to install the binaries and not compile the zaptel libraries?
06:55.53Manipuraand now, everything I try, doesn't work.... But what are these kernel header files I have to install?
06:56.21t3tManipura, you don't need them if you have the binary modules for your kernel
06:56.28mwgbcDoes anyone know anything about adjusting the sensitivity of NVBackgroundDetect? (Volume not time sensitivity)
06:56.29ManipuraI'm trying to compile zaptel, its not working.... I sent a message to the mailing list. So far, nothing back
06:57.10Manipurasorry t3t, your speaking a different language
06:57.17t3tManipura, if you're trying to compile them then you need the configured kernel source or kernel headers for your running kernel
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06:58.04Manipurameaning??? I need a custom kernel??
06:58.18t3tManipura, then there's little that I can do for you. sorry.  try an * live cd
06:58.39*** part/#asterisk IgorG (n=gia@195.162.32.126)
06:59.06ManipuraCool, no help here.... Why did they make * so hard to install?
06:59.23t3tManipura, you don't need a custom kernel, but you do need the zap modules for the kernel that you are running
06:59.48JerJerManipura: why did they make WIndows XP so broken?
07:00.05hugo-v6Manipura: well. its not for someone which has no clue from linux administration and internals
07:00.07Manipuraoh, now I think I get it.... So how do I get this zap module for the kernel?
07:00.16JerJercompile it
07:00.29t3tManipura, it's not that there isn't help, it's that you don't have the background information necessary to solve your problem quickly.  An * live CD like * at home is probably better for you
07:00.36Manipuracompiling zaptel is where I'm getting the errors
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07:00.45hugo-v6Manipura: look after kernel-header package and compile zaptel and hope it  works
07:04.31mwgbct3t: Do you know where I can get help adjusting the volume sensitivity of NVBackgroundDetect?
07:05.02Faithfult3t: this is what I am doing at the moment.
07:06.43Faithfult3t: mwgbc: problem is I am streaming the radio for MOH
07:06.57t3tmwgbc, I'm guessing that only Newman Telecom knows the answer to that one (unless you feel like digging through their source and changing it there
07:07.48mwgbct3t, I'm afraid that is what I'm faced with.  I was just trying to avoid it. :-)
07:07.49t3tFaithful, I don't know that one
07:08.52t3ti'm off to bed.. 'night
07:09.02mwgbct3t: night
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07:20.29BorgonDo you all recommend a softphone that allows multiple calls at 1 time, to manage different outgoing dials?
07:21.30konradsBorgon: as opposed to?
07:22.38Borgonkonrads: i tried iaxcomm and firefly, each support different network accounts, but when i tried to dial out from a different network while i was talking with someone it doesnt seem to let me.
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07:27.49konradsBorgon: sorry, no idea :) You could teach asterisk this magic
07:28.25Borgonkonrads: i was thinking about having an extra extension separating each =)
07:28.41Borgonkonrads: but they will share the same setcallerid which i dont want.
07:29.12konradsBorgon: not nescessary
07:29.19X-RobBah. Work.
07:29.22konradsyou can use SetCallerId(wasabi) function
07:29.34QwellYou can put callerid right in the iax/sip user
07:29.44konradsor that
07:29.59Borgonkonrads: but am saying, if they are in the same [outgoing] context eventhough they have different outgiong servers, wont they share the same setcallerid () function?
07:30.13QwellBorgon: look at callerid= in iax.conf and sip.conf
07:30.19X-RobBah ECHO
07:30.23X-Rob*grumble*
07:30.34konradsBorgon: for each account you can specify context
07:30.37BorgonQwell: ok, so that allows to be put  in the 1_XXXXXXX blah extension for each one?
07:30.42Qwellno
07:30.49QwellYou don't have to set it from the dialplan
07:30.49*** part/#asterisk newsmafia (n=newsmafi@wsip-68-15-19-142.sd.sd.cox.net)
07:31.08Qwellcallerid=John Doe <1234>
07:31.16Qwellin sip.conf, or iax.conf
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07:32.08Borgonok ill look into it
07:33.42Tilihow can IAX2 help in NAT traversal?
07:33.56QwellTili: iax only uses one port
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07:34.39Qwellno forwarded ports meeded (in most cases)
07:34.55TiliQwell: yeah but we need a way like STUN for inbound calls to clients when we dont have an asterisk server in the network
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07:35.15QwellI think iax "Just Works" in that case
07:35.45Ahrimanesuhm with iax you need to forward that one port
07:35.59QwellAhrimanes: not true, in most cases
07:36.14TiliQwell: no iax doesnt work in that case. because you never know which port NAT has bound it.
07:36.41AhrimanesQwell: hm ok, my experiences tell me that forwarding the port used saves you many a grey hair
07:36.41TiliI think i will need to implement STUN for that.
07:36.50QwellWhen the iax client registers to the * box (on a remote network), the router should "Just Work", for incoming calls
07:36.57TiliAhrimanes: it is not possible when you have users at various locations using different networks that you dont even know about it.
07:37.06opus_"Just Work"(tm)
07:37.06QwellAhrimanes: What do you do when you have 50 users behind one IP?
07:37.09opus_:)
07:37.24TiliAhrimanes: forwarding works when you have clients on your own network or where users know how to forward port
07:37.40AhrimanesTili: no true, that's why we use sip and stun :)
07:37.40*** part/#asterisk Uberbot (n=Uberbot@pcp01880954pcs.sandia01.nm.comcast.net)
07:38.02AhrimanesQwell: make a special installation for that client
07:38.06Qwellwell, you asked a question, I answered it...whether you believe it or not, I really don't care.
07:38.21TiliQwell: that is the point. My each IAX client will get a different port allocated if there are more ppl behind one NAT.
07:39.07QwellIAX>NAT>Internet>Asterisk
07:39.13TiliQwell: i am listening to you. What is your suggestion in this scenrio.
07:39.16QwellIAX registers to Asterisk
07:39.17AhrimanesTili: stun is working really well for me.. there are some nat devices which cant be traversed with stun though
07:39.18QwellAsterisk calls IAX
07:39.22Qwellcall will "Just Work"
07:39.34TiliYou have clients all around the world sometimes behind NAT and firewalls etc. What do we do?
07:39.46QwellTili: You do nothing.  It will "Just Work".
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07:40.25Qwell.google iax2 nat traversal how
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07:40.35*** part/#asterisk Rainer (n=Rainer@p508841E8.dip0.t-ipconnect.de)
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07:41.07*** part/#asterisk Rainer (n=Rainer@p508841E8.dip0.t-ipconnect.de)
07:41.17Wonkahttp://shop.beronet.com/product_info.php/cPath/21_25/products_id/73?osCsid=2cc77f3903c544a0df40d050c4ced9dc
07:41.20Wonkacool
07:41.29Wonka"The Billion 1 Port USB Adapter to connect 1 BRI/ISDN ports. Supports TE and NT mode"
07:41.48Wonkacompatible with chan_misdn...
07:42.10BorgonI want to have 2 different caller id for 2 different extensions on the dial plan on different providers is that possible? for example
07:42.14Borgonexten => _1NXXNXXXXXX,1,setcallerid(0000000000) ; id for provider1
07:42.15BorgonExten => _1NXXNXXXXXX,2,Dial(provider1)
07:42.15Borgonexten => _9NXXNXXXXXX,3,Dial(provier2)
07:42.31konradsWonka: and chan_misdn is the flakiest piece of software i've seen in years
07:42.37Borgonif i add the 9 to a second setcallerid would that fix the provider2 like exten => _9NXXNXXXXXX,5,setcallerid(0000000000) ; id for provider2
07:42.41Wonkakonrads: too bad, yes
07:42.51Wonkakonrads: i've switched back to chan_zap too
07:43.18Wonkathere are things done, once in a while...
07:43.24Wonkain misdn
07:43.34TiliQwell: thanks. I will do some testing with asterisk first. I also need to call users through their IP address, so I guess I have to use STUN as they are not registered to Asterisk.
07:43.37WonkamISDN is far worse than chan_misdn, it seems to me
07:43.41Borgonanyone?
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07:44.25*** part/#asterisk vuvie (n=vuvie@bb219-74-47-90.singnet.com.sg)
07:45.47konradsWonka: right now I am trying to use hfc pci -> misdn -> capi -> chan_capi
07:46.02konradsWonka: dial-out works, but dial-in doesn't
07:46.15konradsfiled a bug to mISDN bugtracker, let's see if it can be quickly sorted
07:49.00Wonkakonrads: not possible here, i want NT mode. works quite fine with chan_zap (bristuff 0.2.0-RC8n)
07:52.08konradsWonka: supposedly chan_capi understands nt mode aswell
07:52.16Tiliis there any open source STUN server and client library
07:52.26konradsWonka: do you use NT mode for interconnect or for phones?
07:52.56Wonkakonrads: the nt mode card is for connecting internal phones
07:53.16konradsWonka: and what is the cost per port?
07:53.25Wonkamh?
07:53.27konrads4 port beronet stuff costs 650 eur
07:53.38Wonkai only got a HFC-S
07:53.47Wonkaand connected 1 phone to it
07:54.11konradshfc-s? regular HFC PCI card ?
07:54.12Wonkamaybe i'll connect a DECT base station sometime, but that will be it
07:54.15Wonkayes
07:54.28konradsand chan_zap works with it? by what drivers?
07:54.33Wonkabristuff
07:54.39Wonkazaphfc
07:54.52konradshmm.. never knew of it
07:54.56konradszaphfc you say...
07:55.16Wonkahttp://www.junghanns.net/en/download.html
07:56.10Wonkaget bristuff stable, run the script - it will get zaptel, asterisk, libpri, patch them for BRI, and compile it
07:56.38konradsooohkay...
07:56.46konradssigh
07:56.47Wonkai use chan_capi too, for incoming and outgoing calls with the AVM B1 active card connected to the land line
07:56.51konradsi wish asterisk had a manual
07:57.19konradsvoip-info isn't the most organised source of information
07:57.24Wonkaack
07:58.20konradsSomeone should write a section on "What cards work, in which mode and with what drivers"
07:58.33konradsI could start it :)
07:58.50konradsWonka: BTW that active card is any good? worth the money?
07:59.11Wonkait's an old ISA card, but they do PCI too
07:59.17Wonkaup to now, no problems with it
07:59.35Wonkadrivers are included in linux since far before 2.6
07:59.40Wonkamaybe far before 2.4
08:00.08Wonkait is said it does fax too, in firmware, but i haven't tried that yet
08:00.42konradsHmm, that bristuff is for asterisk 1.0
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08:01.32Wonka1.0.9, yes
08:01.34Wonkalatest
08:02.01konradsbut i don't see it using any dependencies to it
08:02.10konradsso i'll assume it works with 1.2.0 aswell
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08:03.03Wonkathere's a 200k patch to asterisk in the patchs directory
08:03.15Wonkai think that could count as "dependency"
08:03.22konradsi see..
08:03.33konradsand i don't suppose the CVS version is for 1.2?
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08:12.24mepplguten morgen
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08:14.38h3xis muxmon supposed to work with 1.2.0-beta1
08:14.54Wonkakonrads: no idea
08:15.16konradshmm, that bristuff doesn't compile some asterisk apps
08:15.52X-RobTeddybears Sthlm is _astoundingly_ funky.
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08:24.17sycoflyqueuetue: got it sorted yet ?
08:24.55dudesh3x - what's up
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08:35.18wasimspammers galore?
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08:36.16dudesspammers suck
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08:36.44wasimi thought they needed to be registered to join #asterisk
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08:38.21dudeswasim - no but you can block unregistered users from messaging you
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08:42.11Smeahy
08:42.17*** join/#asterisk R7c25Ye (n=d7y17ox@219.93.239.126)
08:42.19wasimciao Smea
08:42.27*** part/#asterisk R7c25Ye (n=d7y17ox@219.93.239.126)
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08:42.30Smeaciao wasim, italiano ?
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08:42.59wasimSmea: nope, just practicing my vocabulary
08:43.06Smeaok wasim :)
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08:43.46Smeai'm very happy to find asterisk channel
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08:45.31Smeaany italian asterisk user here ?
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08:49.29konradsWonka: here?
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08:49.55konradsWonka: HFC-S PCI A ISDN card 1 [TE]             UNCONFIGUR          0          0    0
08:50.00konradsit says unconfigured
08:50.03konradswhazwrong?
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08:51.05Wonkakonrads: what says that?
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08:52.15Wonkakonrads: you need to have an /etc/zaptel.conf and to run ztcfg after loading zaphfc
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08:53.30afarhanhello, i am a newbie around here. i am looking for some guru to get me started writing a channel driver. i have experience writing SIP and ISDN stacks, i need to interface with a new hardware for asterisk
08:53.48X-Robwhat sort of hardware?
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08:54.27konradsWonka: make load supposedly did that. Can you show me yours?
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08:54.42afarhanit is an ISDN-card. i can get it to work under my own sip stacks. i need to know how the channel module works
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08:55.04afarhani have been reading the code for a few channel drivers though. but is the channel api documented somewhere?
08:55.32Wonka~nopaste
08:55.34jbotit has been said that nopaste is http://nopaste.snit.ch/ or http://pastebin.com/
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08:56.55RoyKjbot: nopaste is also preferably http://pastebin.ca/
08:56.56jbotokay, RoyK
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08:57.07RoyK~nopaste
08:57.08jboti guess nopaste is http://nopaste.snit.ch/ or http://pastebin.com/, or preferably http://pastebin.ca/
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08:57.11Wonkakonrads: http://pastebin.com/356860
08:57.23Wonka~pastebin
08:57.24jbotit has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/
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08:58.23konradsWonka: thanks
08:58.35Wonkanp
08:58.44konradsand how about /etc/asterisk/zaptel.conf aswell :) ?
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08:58.49konradsasterisk just crashe dout ;)
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08:59.02konradszapata.conf
08:59.04konradsi meant
08:59.39konradsmaybe it works.. wait
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08:59.59konradsnope. says chanunavail
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09:00.10konradsWonka: please paste zapata.conf and zaptel.conf aswell
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09:00.44vlrkhello friends
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09:01.06vlrkdoes any body know what exactly it means ackcall=1 in agents.conf
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09:02.04RoyKwtf?
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09:02.49Wonkaspam bots?
09:03.01Wonkasomeone call lilo?
09:03.05Wonka:)
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09:03.46konradsWonka: please paste zapata.conf and zaptel.conf aswell
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09:04.53Wonkakonrads: zapata.conf http://pastebin.ca/22290
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09:05.17Wonkakonrads: i don't have a zaptel.conf
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09:06.06Wonkakonrads: /etc/asterisk/zaptel.conf is just a sample for /etc/zaptel.conf
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09:06.16Wonkakonrads: which i already postet :)
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09:07.14konradsWonka: ok. I think i need to experiment with switchtype
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09:07.30konradsit tries to call, but fails
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09:12.47Wonkakonrads: do you use TE mode or NT mode?
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09:13.05konradsTE
09:13.10*** join/#asterisk C3n71Tx (n=y3q12vg@213.145.114.92)
09:13.12konradsi changed signalling to bri_cpe
09:13.20*** join/#asterisk U8o83Sh (n=g1y32az@85.101.169.91)
09:13.22Wonkathen try signalling = bri_cpe_ptmp
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09:13.40Wonkaif it stil not works
09:13.50konradsok, i have outgoing
09:14.12vlrkhow to change the log modes in asterisk like warning or debug
09:14.15Wonkathis is annoying! all those n= joining and parting
09:14.36wasimvlrk: logger.conf
09:14.36konradshave incoming
09:14.50konradsWonka: you are my hero of the day
09:14.56Wonka:)
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09:20.34*** mode/#asterisk [+b *!*@85.100.96.*] by denon
09:20.37X-RobIck. Spambots.
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09:20.46*** mode/#asterisk [+b *!*@81.215.212.*\] by denon
09:20.48*** mode/#asterisk [+b *!*@81.215.212.*] by denon
09:20.51*** mode/#asterisk [-b *!*@81.215.212.*] by denon
09:20.54*** mode/#asterisk [+b *!*@81.215.212.*] by denon
09:20.56*** mode/#asterisk [-b *!*@81.215.212.*\] by denon
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09:21.51*** mode/#asterisk [+b *!*@85.97.183.*] by denon
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09:22.16*** mode/#asterisk [+b *!*@85.96.251.*] by denon
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09:22.23*** mode/#asterisk [+b *!*@85.100.67.*] by denon
09:23.11denongoobers have a lot of subnets ..
09:23.16denoncould ban em all day and still not keep up
09:23.33wasimi thought only registered users could connect to #asterisk
09:23.39denonbut I might as well try
09:23.39Wonkame too
09:23.47*** join/#asterisk C1j75Nh (n=s5r57bx@60.48.100.208)
09:23.48denonno reason a spambot can't register itself
09:23.57*** mode/#asterisk [+b *!*@60.48.100.*] by denon
09:23.58*** part/#asterisk C1j75Nh (n=s5r57bx@60.48.100.208)
09:24.23Wonkabut they aren't - /ban n=*@* perhaps?
09:24.30Wonkaor however it works...
09:24.31denonthat's ident
09:24.34denonnot registration
09:24.50denonregistration is a channel flag
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09:26.00denonWonka: http://svn.freenode.net/theia/trunk/include/channel.h :)
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09:26.13*** mode/#asterisk [+b *!*@81.214.235.*] by denon
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09:27.32*** mode/#asterisk [+b *!*@212.174.62..*] by denon
09:27.34*** mode/#asterisk [+b *!*@212.174.62.*] by denon
09:27.38*** mode/#asterisk [-b *!*@212.174.62..*] by denon
09:28.18*** mode/#asterisk [+r] by denon
09:28.53X-Robheh
09:29.39wasimyeh .. woohoo +r
09:30.01denonsomeone's got a lot of subnets
09:30.11denonprobably someone silently lurking in the channel, watching our conversation ..
09:30.17Wonkamaybe, some bans could be removed...
09:30.21wasimwe should speak in urdu then
09:30.40Wonkathose on whole /24 and 624576 seconds old...
09:31.07denonWonka: yeah probably, though there's nothing stopping nickbots from script a registration
09:31.14Wonkaack
09:31.16denoner floodbots, spambots, whatever
09:31.42Wonkabut are bans on *!*=CuteGirL@62.162.14.* or *!*@672db68b04f7a2c2.session.tor still needed?
09:31.56denonwhy, is that you?
09:31.58denonhehe
09:31.59Wonkano
09:32.01Wonkalol
09:32.13*** mode/#asterisk [-bbbb *!*@212.174.62.* *!*@81.214.235.* *!*@60.48.100.* *!*@85.100.67.*] by denon
09:32.13*** mode/#asterisk [-bbbb *!*@85.96.251.* *!*@85.97.183.* *!*@81.215.212.* *!*@85.100.96.*] by denon
09:32.15*** join/#asterisk Beccara (n=Beccara@222-152-27-117.jetstream.xtra.co.nz)
09:32.18Wonkatwo examples from the ban list
09:32.39wasimban ScaredyCat!
09:32.50Wonka*mew*
09:33.29denon(Stargate-style rings)
09:34.18denongot a good deal today
09:34.27denonMBTT at 5 cents
09:34.31denonpretty cool company
09:34.42konradsWonka: have you tried doing a loopback to ISDN?
09:34.43denonand ramping up for some neat stuff
09:34.43Wonkawasim: is it broken?
09:34.58konradslike you receive a call from ISDN and hand it over to a phone over ISDN?
09:34.59wasimno, unbalanced
09:35.00Wonkakonrads: not that i know of... how would one do that?
09:35.09konradsah, ye you used the NT mode :)
09:35.22Wonkaand capi on the other isdn card
09:35.42konradsWonka: also, my phone guy says that you can make calls over D channel aswell
09:35.56Wonkaeh? how's that?
09:36.48konradsNFI
09:38.36konradsbut this HFC doesn't support it :)
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09:39.38PakiPenguinMorning
09:40.40TeeliNot so Morning PakiPenguin
09:41.11PakiPenguinhaha :) Teeli , yeah more like afternoon
09:46.14Teeliwhile bridging calls how do i transfer the calledparty to another extension.
09:46.54TeeliG in Dial can transfer both calls. I dont want that. MACRO_RESULT=GOTO:blah:1 can transfer calling party and seems to be hanging up the called party
09:47.32TeeliPakiPenguin: any idea?
09:48.08PakiPenguinTeeli, umm cant think right now
09:48.08PakiPenguinhehe
09:48.13PakiPenguin:p
09:48.24Smeaany italian asterisk user here ?
09:48.27TeeliPakiPenguin: no problemo.
09:48.48TeeliSmea: why do you want italian user. We all are also friendly enough
09:49.52wasimTili: true, we
09:49.58Smeashure Tili, but my english is very poor and i'm courious of know italian asterisk user
09:50.18TiliSalam wasim bhai
09:50.56PakiPenguinthis is retarded
09:51.12wasimSmea: there are many .it asterisk users
09:51.20PakiPenguini go click cancel subscription , it takes me to signup :(
09:51.32PakiPenguinhey wasim
09:51.35Smeayes wasim, but on irc ? :)
09:52.06wasimSmea: oui, on irc
09:52.07TiliSmea: I guess most italians are still on their way to office
09:52.32Smeawasim, do you know name of italian asterisk channel on irc ?
09:53.49PakiPenguinOMGOMG
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09:53.55PakiPenguinhaha , that was fun!
09:53.55wasimPakiPenguin: although dawlance is not bad either
09:54.22PakiPenguinwasim, uh? dawlance?
09:54.22PakiPenguindid i miss something?
09:54.22wasimPakiPenguin: waves, sabro, dawlance
09:54.26PakiPenguin:) oh
09:54.47*** join/#asterisk Wonka (i=produzie@madwifi/support/wonka)
09:56.50PakiPenguino_0
09:57.51ScaredyCatthey were being mean to me while i was sleeping...
09:58.04ScaredyCatI sensed it with me spidey sense
09:58.13PakiPenguinlol
09:58.17Tilimorgen ScaredyCat
09:58.27ScaredyCatlo Tili...
09:58.34PakiPenguinTili, FLY DOWN TO ISB!
09:58.42*** join/#asterisk MikeJ__ (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
09:58.53TiliISB for what? you are in LHR right?
09:58.59PakiPenguinnope
09:59.00PakiPenguinISB :)
09:59.09ScaredyCatEHLE
10:00.18ScaredyCatactually itia LEY
10:00.32ScaredyCatbut I digress, I'll yabbering
10:00.42ScaredyCatme no spell eoither
10:03.37manymh, someone with snom who can tell me what buttoncode the headset button has?
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10:12.52Tilibye all
10:21.53manyoh. and is there some bri/zap-independent steal?
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10:49.02jpcarvalhoHello All , i have SIP/IAX2 termination in Brazil. I'd like to exchange termination with a Billing solution for Asterisk ( post paid ) ?! Anyone has interests ?
10:49.40X-Robasterisk-biz@lists.digium.com
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10:51.24X-RobEvening
10:51.30AsylumHey X-Rob :)
10:51.31X-Robhow'd your first day of ISDN go?
10:51.58Asylumpretty good, just trying to work out the dialing plans. No allowing some extentions to dial mobile calls...
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10:52.13X-RobThat's harder.
10:52.25X-RobIt's something I need to implement in dialplans.agi, I think
10:52.29AkelavlkHello, is possible reload configuration via Asterisk Manager port?
10:52.44Asylumwell, i've set it up to from-local-limited
10:52.59Asylumand changed the context in the sip file and the extentions.conf
10:53.03Asylumbut it doesn't seem to work?
10:53.34AsylumI'm not working on it tonight anyway, I'm at home having an early one lol.
10:53.56X-Robhow are you limiting the calls? Your dialplan just spells out 02/03/07/08?
10:54.37AsylumI could show you, but for some reason my port forward isn't working soo i can't log in
10:54.46X-RobHeh.
10:54.51X-RobDownload a CentOS 4 CD.
10:54.57X-RobDVD even
10:54.58X-Robinstall that.
10:54.59Asylumi did yum update...
10:55.02X-Rob_much_ better than asterisk@home
10:55.06konradsdoes 'reload' command actually reload all extensions?
10:55.13X-Rob(as it's based on 2.4 and old stuff)
10:55.41Asylumyeah i'll do that next time
10:56.02RoyKdebian
10:56.13RoyK:%s/centos/Debian/gi
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10:57.21h3xkonrads: reload does, but if you are only updating extensions.conf you should be using "extensions reload" instead.
10:57.35hellop_pdawhat would you charge for an * installation?  4 phones  and 10 phones
10:58.01ManxPower~docs
10:58.03jbotrumour has it, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
10:58.04ManxPower~mailinglist
10:58.05jbotwell, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
10:58.23Akelavlkit's possible reload asterisk remote?
10:58.44hellop_pda4-5 digit answer
11:00.45hellop_pda<sigh>
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11:01.36MRH2anyone know how the firmware update is going to work with the 411P
11:02.18*** part/#asterisk Akelavlk (n=jansun@82.119.239.141)
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11:06.04AsylumDownloading it  now X-Rob :-)
11:07.53X-RobWell.
11:07.56X-RobI'm going to bed, I think.
11:07.59X-RobI'm buggered.
11:08.14[a]-Asylumnight mate :) cheers for the help again..
11:08.28X-Robno probs. email is xrobau@gmail.com
11:08.35[a]-AsylumTa..
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11:16.56Spacebar_hmm
11:17.02Spacebar_SS7 question
11:17.51RoyKss7 isn't really supported in asterisk
11:17.55RoyKor not at all
11:18.07RoyKthere are some ss7 stacks, but they're quite expensive
11:24.17MikeJ__talk to coppice
11:24.34MikeJ__he selss one
11:24.38MikeJ__sells
11:25.20Spacebar_I wrote a email to him, but it was ignored
11:25.25Spacebar_or spamassasinated
11:28.43sivanawhere would one go to learn the benefits of SS7?
11:30.29RoyKsivana: i went to a course at a local telco. that was quite good
11:30.36sivanaok
11:32.24MikeJ__sivana, what country?
11:32.36sivanaCanada
11:33.05MikeJ__you are likely unable to get it unless you are very big and rich in us\cananda
11:33.39crash3mhttp://www.google.com/url?sa=t&ct=res&cd=3&url=http%3A//www.iec.org/online/tutorials/ss7/&ei=_M8eQ5SvPJHO4QGMsuTODA
11:33.43sivanaunlikely to get what?
11:35.12crash3mss7
11:35.38sivanaI see
11:36.44h3xasterisk dosent do anything special enough to make real use of ss7
11:36.57h3xanything that you cant do with pri anyway
11:37.01DrukenHMEsivana: what's the price of gas there?
11:37.11RoyKsivana: ss7 is for interconnecting large telco networks
11:37.25sivanaDrukenHME: stupid high... $1.40 a litre
11:37.27h3xRoyK: Or connecting to BFE switches that dont support PRI
11:37.44RoyKsivana: if you have 10 PRI +, so perhaps it's worth it
11:37.50DrukenHMEsivana: yeah, we are about $1.30
11:37.51RoyKsivana: only it usually costs LOTS
11:38.01sivanaDrukenHME: where are you?
11:38.12h3xNFAS and SS7 load balancing are close to the same
11:38.13h3xwell
11:38.13sivanaRoyK: ya, I see now... not really meant for *
11:38.13DrukenHMEbarrie
11:38.18h3xNFAS is just a hot failover
11:38.19sivanaah cool
11:38.45DrukenHMEsivana: you got connectivity here yet?
11:39.30sivanaDrukenHME: not yet
11:39.45DrukenHMEget two dso's backhauled :)
11:39.45sivanaIt's on our list... with the growth rate and all
11:40.18sivanahow many COs are there?
11:40.28konradsIs it possible to have FXS cheaper than 50 eur per port for amount 5-10?
11:40.38konradslike quad or banks
11:40.48DrukenHMEsivana: uhmm... i think the city has 3?
11:40.55sivanakonrads: check eBay for 24 port FXS channel bank
11:41.18drrayzhone
11:41.34konradssivana: it is hard for legal entities to buy from eBay
11:41.36sivanaDrukenHME: if you can sell it, I could probably get 2 channels :)
11:41.45sivanakonrads: you mean illegal? :)
11:41.52h3xhttp://www.repotec.com/voip/RP_FXS02A.htm
11:42.15DrukenHMEsivana: well, i need a place for at least... 3 dids.... since norman is killing his pri in barrie....
11:42.28sivanaoh.. I was going to get them from Norm
11:42.31konradssivana: i mean corporate bodies like to buy via CC or wire transfer from an established company...
11:42.34DrukenHMEhahahahaha
11:42.42sivanawhat's he doing there
11:42.52sivanahe's dropping it?
11:43.00DrukenHMEthat's what he tells me
11:43.23sivanaI'll msg him when he comes online... maybe we can "sublease" them
11:43.26RoyKhmmm
11:43.26RoyK<PROTECTED>
11:43.28RoyKwtf?
11:43.44konradssivana: still, it's $ 50 per port
11:43.45DrukenHMEwell, he said the pri costs him something like... 1700 a month...
11:43.48konradsI want like $30
11:44.02sivanakonrads: I bought a 24 port FX channel bank for $180 USD
11:44.06sivanaon eBay :)
11:44.10drraya zhone channel bank can be had for $130 USD
11:44.16sivanaya
11:44.16konradsdrray: new?
11:44.18drrayzhone's stink
11:44.22drraybut they work
11:44.26drrayno, not new
11:44.44konradsi want a new one for $30 per port :)
11:44.56konradsi want to compete versus other entry level PBX
11:45.07DrukenHMEwish in one hand, shit in the other, see which one fills up first
11:45.32konradsbesides, most ship to US only
11:45.34*** part/#asterisk pigpen (n=mark@fw.seamans.cc)
11:45.40sivanamove to US then :P
11:46.05sivanakonrads: what country you in?
11:46.10konradsOf course i could send it to a drop-point and later ship it via plane
11:46.15konradsor ask fedex to pick it up
11:46.17konradssivana: EU, Latvia
11:46.23konradsEU's poorest country
11:46.32sivanais there no eBay.eu or whatever?
11:46.40konradssivana: ebay means paypal
11:46.45konradspaypal isn't working here
11:46.49konradsnot enough customers
11:46.51sivanawell.. you can't have your cake and eat it too
11:47.12konradsnevermind, eBay isn't a reliable supplier
11:47.17DrukenHMEsivana: you got a spare T100P you want to sell me cheap don't you? i'll even come pick it up :)
11:47.17konradsI can't advertise with that
11:47.21sivanakonrads: have you looked at Digium?
11:47.36sivanaDrukenHME: haha.. no.. I have 1 :)
11:47.56DrukenHMEsivana: well your no fun... :(
11:47.57sivanaDrukenHME: I have a spare 405 I can sell you :)
11:48.05DrukenHMEte405P ?
11:48.07sivanaya
11:48.09DrukenHMEisn't that the quad?
11:48.12sivanayes
11:48.17konradssivana: sure have - $50 per port
11:48.22DrukenHMEwhy would you want to get rid of that ?
11:48.32sivanaI have 3
11:48.38DrukenHMEoh...
11:48.43sivana2 TE405 and 1 A104
11:48.51konradssivana: if a PC costs, say $ 400 and then $50 per port
11:49.18konradsif there is no SIP involved or voice menu or other 'enterprise' features - price is too high
11:49.29sivanakonrads: your only option would be to look around like at voipsupply.com or send an email to the user or biz list
11:49.31DrukenHMEkonrads: the savings are not in the system, but in the volume of LD calling
11:49.41konradsDrukenHME: LD calling?
11:49.51DrukenHMElong distance or international calls
11:50.13konradsDrukenHME: I'll have to start city-wide IP interconnect then
11:50.17konradsdundi iirc it was called
11:50.57konradscould be a cool idea
11:51.05konradsbusiness connect via IP rather than PSTN
11:51.08DrukenHMEsivana: feel like letting me play with a te405p?? hehehe i have a beautiful channel bank i can't use for the lack of a PRI interface :(
11:51.08konradsfuck the local telco
11:51.13DrukenHMEyou know you want to :)
11:52.03sivana:)
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11:52.30sivanawhen's Wonderland close?
11:52.39DrukenHMEi'm a poor son of a bitch, (mainly cause of gas prices), it's a bitch having to fill the tank every day
11:52.45*** join/#asterisk JessicaX^ (n=Admin@86.112.145.215)
11:52.46DrukenHMEthanks giving
11:53.10sivanaya.. friggin gas.. such bs
11:53.16DrukenHMEthanksgiving weekend is the last weekend they are open
11:53.28JessicaX^:D
11:53.29JessicaX^Hiii
11:53.50sivanaya, amazing how developed it is around there
11:54.02sivanaI remember when Wonderland sat in an empty field
11:54.06*** join/#asterisk Poincare (n=jefffnod@dD5779BD2.access.telenet.be)
11:54.26sivananow they're dropping in houses subdivisions at a time
11:54.40JessicaX^:D
11:54.48DrukenHMEsivana: yeppers
11:55.02sivanaI don't think they can build them fast enough
11:55.20sivanacrazy
11:55.34DrukenHMEwell, they built vaughan mills in less than 12 months...
11:56.01DrukenHMEyou been in that thing yet? it's crazy
11:56.04sivanaya
11:56.06sivanait's a nice mall
11:56.17sivanacrazy busy though
11:56.36DrukenHMEyeah, i wanted to steal like every one of those damn projectors tho...
11:56.40DrukenHMEwhat a waste
11:57.03tzangerDrukenHME: you're in my neck of the woods (relatively speaking)
11:57.19DrukenHMEtzanger: doubtfull...
11:57.24tzangerfill the tank EVERY DAY?
11:57.28tzangerholy fuck dude
11:57.44Kattybeep
11:57.53DrukenHMEtzanger: i drive over 100,000 km per year
11:58.04tzangerDrukenHME: insane
11:58.11tzangerI do about 50-60kkm
11:58.14DrukenHME:)
11:58.29tzangerwhere do you live/work?
11:58.47DrukenHMEi once looked at the travel i put on my van over 10 months, was 80,000
11:58.54DrukenHMEi live in innisfil
11:58.58sivanaboy.. Shell loves you
11:59.18DrukenHMEsivana: and i love shell :), but only for the air miles
11:59.23sivanahehe :)
11:59.52tzangerinnisfil isn't far from me... well a couple hrs but still
12:00.04tzanger(and the entire room says "where?")
12:00.11DrukenHMEhehehe
12:00.13Kattylisp towel?
12:00.19*** join/#asterisk GREEKMAN2 (i=greekman@c-24-63-192-115.hsd1.ma.comcast.net)
12:00.24DrukenHMEi'm sure i've probably driven threw there at some point
12:00.29tzangerprobably
12:00.42DrukenHMEit's at what cross roads?
12:00.43sivanaya.. you can't blink
12:00.44tzangerif youv'e gone to kincardine it's almost a certainty
12:00.50tzangerhwy 23 & 86
12:01.17GREEKMAN2any digiom techs on right now?
12:01.38DrukenHMEi think i drive threw kincardine on my way to renfrew
12:01.39tzangerit's okay though, sivana's in north bay.  he counts even less :-)
12:01.44tzangerDrukenHME: ??
12:01.50sivanalol
12:01.54tzangerwhere the fuck do you live that goes through kincardine on the way to RENFREW?!
12:01.59tzangerinnisfil is nowhere near kincardine
12:02.02DrukenHMEhehehehe
12:02.13DrukenHMEi dunno, that's what my wife said...
12:02.16tzangerrenfrew's practically on the other side of the province :-)
12:02.19DrukenHMEi haven't looked it up..
12:02.30tzanger(my family settled the ottawa valley)
12:02.37DrukenHMEwell, shit i know that... it's a 4 hour drive from here to there
12:02.48tzanger6hr from here
12:02.59sivanawhere is Innisfil?
12:03.08DrukenHMEoh wait a minute... your out near kitchener?
12:03.15sivanawhich side of Barrie?
12:03.16tzangerDrukenHME: yep
12:03.34DrukenHMEsivana: you know where the race track is on the 400 ? that's in innisfil
12:03.58sivanaok.. Georgian Downs?
12:04.03DrukenHMEyep
12:04.05tzangerexactly.. there's no fuckin way you got to renfrew through kincardine from innisfil unless you took a REALLY wrong turn
12:04.19DrukenHMEtzanger: hehe yeah no kidding :)
12:04.34sivanaya.. Kincardine is down my Oshawa, no?
12:04.38sivanaby
12:04.48sivanaor Sarnia
12:04.50DrukenHMEi don't venture out that way very often..
12:05.14tzangerKatty: no, not lisp-towel, although I had a guy from Lousiana try and pronounce "missisauga" once, that was entertaining
12:05.28sivanaI don't know.. it's all freeways and Mennonites
12:05.28ManxPower8-)
12:05.29tzangerI think it came out "mog-wa"
12:05.39Kattytzanger: k
12:05.46DrukenHMEhow can ya screw up mississauga?
12:05.50tzangersivana: freeways?  mennonites sure but freeways?
12:05.54DrukenHMEshit, americans got mississippi
12:06.02sivanawell.. 400, 401, 407, QEW
12:06.05tzangerDrukenHME: I know, and see even I mistook it
12:06.10tzangersivana: those are the queen's highways
12:06.11Byteis there some arbitrary limit on maximum configuration lines?
12:06.13sivanathen all of a sudden you're in corn fields
12:06.16tzangeror the king's highways I can never remember
12:06.29sivanaand then you pop out in Port Dover
12:06.42tzangerByte: I don't think so
12:06.43ByteI've reached some limit where adding more exten lines just gets ignored
12:06.46tzangermy extensions.conf is rather large
12:06.46DrukenHMEsivana: 400, 401, 403, 404, 407, 410, 427, DVP, QEW
12:06.48DrukenHME:)
12:06.53sivanaheh
12:07.00ManxPowerByte, not unless you have tens of thousands of extens
12:07.01Byteif I remove some from another file, it'll do an equal amount more in the other file
12:07.07ByteI have about 400
12:07.34Bytewell, 450 now
12:07.43tzanger# wc -l /etc/asterisk/extensions.conf
12:07.43tzanger587 /etc/asterisk/extensions.conf
12:07.47DrukenHME:)
12:08.03Byte:/
12:08.21tzangerByte: if I include all the include files, I get
12:08.23tzanger<PROTECTED>
12:08.34DrukenHMEtzanger: you get your MOH working?
12:08.41tzangerDrukenHME: haven't tried it just yet
12:08.43ManxPowerMy largest asterisk server is still offline, so I can't check for you.
12:08.44tzangerI will shortly
12:08.46tzangerI'm late for work though
12:09.07DrukenHMEhehe i'm still at home, the phone will ring here... :)
12:09.13tzangercatch y'all on here in a few
12:09.19tzangeryeah I could do that too but it's best not to :-)
12:09.21*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
12:09.22konradsHey, how big cpu is required to operate 4 ISDN channels SIP->ISDN and ISDN->SIP ?
12:09.30konrads4 B channels
12:09.35konrads2 ISDN cards
12:09.42*** join/#asterisk lehel (n=asd@82.79.20.17)
12:09.46lehelhello
12:09.48tzangerkonrads: you could do that with a P2 if you didn't need a heavy codec
12:09.50tzangerP3 otherwise
12:10.14konradsI have a Cel 300-500
12:10.17konradsdunno exactly
12:10.25konradswould be enough?
12:10.46DrukenHMEk, well... i'm heading out too latter ppls
12:11.43sivanalater
12:11.53sivanaI'm going to try and get a hold of Norm
12:13.13konradstzanger: Okay, it is a cel 350
12:13.26konradsI will use GSM
12:13.27tzangerkonrads: just try it and see, jeez
12:13.31sivanaheh
12:13.38sivanatzanger: you at work already? :)
12:13.53konradstzanger: am under slight pressure, I can't test :(
12:14.49tzangerkonrads: then you're doomed before you even start.
12:18.53konradsright, i'll take a more powerful box
12:22.28ManxPowerkonrads, Expect to work with Asterisk for 6 months before you are qualified to put in a decent PBX.
12:23.21konradsManxPower: the company i currently work for sets up alcatel pbxs, so I have telecomms support
12:23.45konradsManxPower: extensions still ellude me, but I am sure I can twist (*) till it screams in pain >:)
12:24.05ManxPowerkonrads, Ah!  Then you can expect to work with Asterisk for only 3 months before being qualified to put in a decent PBX
12:24.36sivanaI was qualified in 2 weeks, but that's because I took the red pill
12:24.39konradsManxPower: right now all I need is a SIP->ISDN->PBX bridge
12:24.43*** join/#asterisk sinoth (i=sinoth@ip70-178-21-145.ma.dl.cox.net)
12:24.46konradsand I have done that already
12:25.03konradssivana: got more :) ?
12:25.06sivana:)
12:25.07ManxPowerkonrads, "making it work" and "qualified" are different.
12:25.28tzangerkonrads: if you have time to find a better system you have time to see if your celeron will do the task at hand
12:25.31konradsManxPower: oh sure :) i have no illusions to troubleshoot it without looking at it
12:27.03konradstzanger: i have 2 used boxes
12:27.03konradsone cel and other amd
12:27.04konradsone of them has to work before midnight :)
12:27.04tzangerkonrads: then but the beefier one in
12:27.04tzangerget it working, THEN start economizing
12:27.29sivanaya
12:27.33sivananot like me
12:27.45ManxPowerMany times simply switching motherboards can cause all sorts of new problems and issues.
12:27.54konradsi am not the economy type of guy, just that getting a PC in 4 hours to the client isn't that easy - logistical and accounting hassle
12:29.49konradsand that amd is my workstation :)
12:29.50*** part/#asterisk sinoth (i=sinoth@ip70-178-21-145.ma.dl.cox.net)
12:30.25tzangerkonrads: why did you give yourself such little time in doing this?
12:30.40tzangerthe celeron SHOULD work just fine so long as you're not trying to save the world with it (telephony-speaking)
12:31.18konradstzanger: why? It is not as if I had much choice :) The client needs it in -4 days.
12:31.28konradsAnd no, in this case I am not my own boss ;)
12:31.31ManxPoweryou always have a choice.
12:31.44tzangerkonrads: you can say "I can't do it htat quickly"
12:31.51ManxPowerWhat would you do if a client wanted a Fusion Plant in 4 days?
12:31.57konradsManxPower: ship it from russia
12:31.57lilohi all.... did anyone get private message spam from those bots that hit the channel?
12:32.07tzangerManxPower: forget it... konrads, i want a fusion plant in four days
12:32.17ManxPowerkonrads, Fusion != Fission
12:32.34konradsManxPower: right, damn. I'll need an extra day :)
12:32.48konradsanyway, i catch your drift :)
12:32.58ManxPower"Find a consultant more crazy than I am" is a common response I give to requests that are totally unreasonable.
12:33.09ManxPowerThe few times the client did that, the project was a total disaster.
12:33.13tzangereither that or you charge them through the nose and pull it off
12:33.35konradsyeup
12:33.47tzangerbut that also requires a certain level of expertise :-)
12:33.48bjohnsonor charge them through the nose and move to Switzerland
12:34.01tzangerbjohnson: you have to get the money first :-)
12:34.25ManxPowerkonrads, just don't come here and expect us to drop every thing and help you when some bizzare hard to diagnose and fix problems crops up.
12:34.35bjohnsonlike everyone else
12:34.37konradsi don't :)
12:34.38ManxPowerpoor planing != emergency for us.
12:34.44konradsManxPower: i know
12:34.51konradsi greatly appreciate help given here
12:34.58konradswithout it, i would have gone nowhere
12:35.00konradsthank you
12:35.01bjohnsonBUT .. maybe poor planning = $$ for us??
12:35.05ManxPowerkonrads, appreciation != cash
12:35.06tzangerkonrads: well that' skind of what you just did... "will x work?"  "test it and see."  "I can't."
12:35.09konradsbjohnson: don't count on that :)
12:35.14*** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com)
12:35.27konradstzanger: the question was more like : "Is the setup xxx known to work"
12:35.32konradsI got the answer : undef
12:35.41tzangerkonrads: asterisk dimensioning is as varied as the motherboards you can buy.
12:35.42ManxPowerkonrads, ask on the mailing lists.
12:35.49konradswhich means, it is excluded from the list of choices and a known better is chosen
12:36.03ManxPowertzanger, hell, telecom dimensioning has the same problem.
12:36.13konradstzanger: i guessed that i/o bandwidth is key issue aswell
12:36.25ManxPowerthe Erlang formulas only work for "average" user requirements and only for large numbers.
12:36.55ManxPowerWhat's the standard, %10 of people on the phone at 1 time (applies to telcos), %20 for large corporations?
12:37.01tzangerkonrads: not really.  4 B channels is hardly i/o expensive
12:37.06tzangerthat's a measly 256kbps
12:37.07ManxPowerI had an office that had %90 of the people on the phone at any one time.
12:37.07*** join/#asterisk littleball (n=littleba@cm157.epsilon173.maxonline.com.sg)
12:37.16ManxPowerREALLY fucked up our capacity planning
12:37.22tzangerManxPower: :-)
12:37.26littleballhi
12:37.38*** part/#asterisk littleball (n=littleba@cm157.epsilon173.maxonline.com.sg)
12:37.54*** join/#asterisk littleball (n=littleba@cm157.epsilon173.maxonline.com.sg)
12:37.58littleballtest
12:38.02Wonkafailed
12:38.10tzangershhhh everyone let him think he can't be seen!
12:38.11tzanger:-)
12:38.36DrukenManxPower: what do you expect for realestate?
12:39.04konradsDruken: 100
12:39.11konrads% online with 2 lines per capita
12:39.23konradsand 3 for boss.
12:39.44Drukentzanger: i thought you were going to work? :)
12:40.14tzangerI am at owrk
12:40.29Drukenthat was quick...
12:40.49tzangerI'm 4km from work
12:40.58ManxPowerDruken, I expect them to be out in the field showing houses.
12:41.04Drukenthat's a 10 min walk? hehe
12:41.20DrukenManxPower: it's a very nice thought....
12:41.26ManxPowerDruken, they learned they have to tell me if the office is commercial or residentia.
12:41.43*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
12:41.51ManxPowerDruken, Agents currently pay about $38,000/yr to work for the company.
12:42.14*** part/#asterisk Maksim (n=max@213.142.207.20)
12:42.16littleballhi, what is the recommend configurations for 12 persons company?
12:42.24DrukenManxPower: but make alot more then that if they can sell anything
12:44.24ManxPowerDruken, *nod*  Most of the residential agents are female, have a lawyer/doctoer/whatever husband, and all she does is sell houses to her country club friends.
12:44.50konradsManxPower: you sound a bit bitter :)
12:45.03Drukenyeah.... i don't think i'd exactly fit in well there....
12:45.11Drukensince i can't stand "snooty" people
12:45.22konradsDruken: you don't have to stand, you have to sell :)
12:45.30ManxPowerkonrads, only because they are too lazy or stupid to learn even the simpliest conputer tasks.
12:45.48bjohnsonscrew em
12:45.58konradsah that... an opportunity for an assistant then
12:46.06bjohnsonoh, err .. forgot who I was talking to ..  I'll screw them
12:46.21ManxPowerbjohnson, The PROBLEM is that these morons make the company money, so what the agent wants, the agent gets.
12:46.31ManxPowerassuming they are making money for the company.
12:46.57bjohnsonbaffles my mind how some companies can make money despite the incompetence of the employees
12:47.13Drukentzanger: so, where do you work ?
12:47.14tzangerI've gotta go to .mi
12:47.19tzangerfor a few days...
12:47.21tzangerer not .mi
12:47.21bjohnsonmichigan?
12:47.22tzangerbut .mi.us
12:47.25tzangeryeah
12:47.28tzangerrochester hills area IIRC
12:47.52bjohnsongood time to buy some USD $$
12:47.53*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no)
12:47.59tzangeryeah
12:48.30tzangerstill 0.84 isn't anything all that great... it ain't bad htough
12:48.55bjohnsonwonder what gas prices like there now
12:48.59ManxPowerkonrads, Perhaps it's just the stress caused by Katrina: Storm of Doom.
12:49.04tzangerwhere's MikeJ[Laptop] when you need him
12:49.06Drukencheaper than here...
12:49.14bjohnsonwe're up to $1.35 / L here
12:49.17Drukendo the conversion we are paying about 6 bux a gallon
12:49.29konradsManxPower: I think that it is Katrina: Incompetent Management At Large
12:49.33tzangerbjohnson: I bought 56L at $1.00/L yesterday
12:49.37ManxPowerWe got about 3/4 of our voice and data networks back up yesterday morning
12:49.42Bonzai090how can i enable callerid on incomming calls on my trunks?.
12:49.53denonKatrina: If Only We Had Known It Was Coming --oh wait, we did.
12:50.00tzangerManxPower: who are they selling to?
12:50.06ManxPowerBonzai090, tyou call your telco
12:50.09tzangerdenon: easy now, it wasn't THAT predictable
12:50.13ManxPowertzanger, Hmm?
12:50.13tzangerBonzai090: callerid=asreceived
12:50.47tzangerManxPower: who are your real estate people selling ot in that area?
12:50.52bjohnsonDruken: about $4.24 USD
12:51.04ManxPowertzanger, anyone that calls in, desperate to pay any price for a house.
12:51.06sylehttp://www.sterilecowboys.org/archive3/BUNNY.html
12:51.11tzangerManxPower: crazy
12:51.34tzangerManxPower: "I've got some great oceanfront property in New Orleans..."
12:51.35ManxPowerIt's a good time for real estate in the entire region (except the actual New Orleans Basin, which is only a small part of the greater new orelans area.
12:52.10tzanger*nods* I hear that MS was hit much harder (i.e. much greater area affected)
12:52.11ManxPowerhouses on the north shore of the lake selling prices increased 125% the day after the hurricane.
12:52.16tzangerbut there's no press of anything but NO
12:52.35sylehow can realestate prices go up that much
12:52.36ManxPowertzanger, Yeah.  I need to find a way to get to my place in Waveland MS
12:52.38tzangerincreased 125%?  meaning a $100k house now sells for $225k?
12:52.40sylewho the hell wants to live there
12:52.45ManxPowertzanger, Yup!
12:52.49tzangerholy FUCK
12:52.53*** join/#asterisk ymorin (n=ymorin@savoirfairelinux.net)
12:52.58denonsyle: thats what I wonder
12:52.59ManxPowerYou people have no idea of the geography of new orleans.
12:53.09bjohnsonsyle: people moving out of NO
12:53.12tzangernope I admittedly have very little geographical knowlege
12:53.19tzangerI'm willing to get educated though
12:53.21konradsI always knew natural disasters and wars are good for business
12:53.32konradssomeone suffers and needs all infrastructure, housing rebuilt
12:53.41konradscontractors move in, reap profits and leave
12:53.52gordonjcpmmm
12:54.06gordonjcpmaybe they should think about making it not nine feet below average sea level
12:54.09ManxPowerthe actual city of New Orleans is a roundish area partially below sea level.  the areas south of the Missisiippi river and north of the lake had almost no flooding at all, just some wind damage.
12:54.27tzangerManxPower: you think they'll level the city?
12:54.45bjohnsonI doubt it
12:54.45tzangerperhaps make the NO area industrial/port areas only?
12:54.52ManxPowertzanger, It doesnt matter.  The culture that made new orleans unique died the day of the storm
12:55.00tzangeryeah?
12:55.00bjohnsonnot enough political balls to do anything like that
12:55.07tzangerManxPower: can you elaborate?
12:55.19konradstzanger: all old building washed away
12:55.30bjohnsonthey're singing the blues
12:55.32ManxPowerit was dieing for years, becoming much less of "The only european city in the USA" and becoming more "just another poor, run down american city"
12:55.34konradstzanger: which means new NO will be just another regular US coastal city
12:55.47ManxPowerkonrads, exactly.
12:55.53tzangerahh
12:55.55*** join/#asterisk kshumard (n=root@digium.com)
12:55.57tzangerthat is unfortunate
12:55.59bjohnsonhow was NO european?
12:56.08tzangerso what's root@digium.com doing in here
12:56.10littleballhi, i am looking for an adapter which make it possible for normal analog phones to make SIP calls. any help?
12:56.10konradsbjohnson: by having french colonial architecture
12:56.12ManxPowerbjohnson, It's hard to explain.
12:56.14*** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk)
12:56.20konradslittleball: plenty, you need ATA adapters
12:56.21tzangerlittleball: it's called an ATA
12:56.23konradsvoipsupply has em
12:56.31littleballthanks.
12:56.31ManxPowerbjohnson, when was the last time you spent any time in Europe?
12:56.31tzangerSipura, Digium, Linksys even
12:56.32bjohnsonlittleball: SPA 2000
12:56.41tzangerlittleball: Grandstream makes one too
12:56.44littleballlet me check the price
12:56.45bjohnsonManxPower: never
12:56.53bjohnsonManxPower: same amount of time spent in NO
12:56.53ManxPowerbjohnson, then it's VERY hard to explain
12:56.58littleballhow about the price, averagely
12:57.16bjohnsonManxPower: how much time have you spent in Listowel?
12:57.23tzangerbjohnson: hahahahaha
12:57.24konradslittleball: $100 on average, iirc
12:57.27ManxPowerbjohnson, never even heard of it 8-)
12:57.36bjohnsonManxPower: you haven't LIVED !!!
12:57.55ManxPowerUSA: Idle Hands are the Devil's Work.  i.e. work is the most important thing in life.
12:57.55konradsbjohnson: i don't think it is fair to compare
12:58.08littleballboth works with asterisk?
12:58.12ManxPowerEurope: Why live, if you can't work AND play.
12:58.16*** mode/#asterisk [-r] by denon
12:58.25MRH2dumb  question : line build out is what distance? from the card to the pri termination point?
12:58.29konradslittleball: they do SIP, which isn't platform specific. Asterisk understands SIP. So, yes
12:58.34littleballok
12:58.43*** join/#asterisk basty (n=basty@212.218.65.229)
12:58.44bastyHi
12:58.52Bonzai090<tzanger> Bonzai090: callerid=asreceived <-- where do i put this in tzanger ?..
12:58.57bastyanyone knows of a opensource billing system for asterisk ?
12:58.58bjohnsonlittleball: generally you want what is called a fxs
12:59.06ManxPowerin New Orleans "I'm sorry I can't come into work today, I have a horrible hangover" IS a valid excuse for missing work, as long as it doesn't happen very much.
12:59.32littleballbjohnson, fxs? let me digest :)
12:59.34DrukenManxPower: i somehow think work isn't a problem in new orleans
12:59.41bjohnsonlittleball: the ATAs plug directly into the LAN, there are also PCI cards that fit into the CPU, there are also channel banks (for bigger installations)
12:59.45denon~fxofxs
12:59.47jbotmethinks fxofxs is An FXO port expects to receive dialtone and receive ring voltage.  An FXS port expects to provide dialtone and provide ring voltage.
12:59.47tzangerMRH2: basically LBO is the cable length from the card port to the first active component (repeater, other side, smartjack, whatever)
12:59.52littleballi see. fxs device
13:00.05tzangerMRH2: LBO doesn't really do anything with Digium's cards ATM, IIRC.  Which is almost never an issue
13:00.13ManxPowertzanger, we always request the telco extend the demarc into the computer/telecom room.
13:00.44konradsManxPower: why?
13:00.46tzangerManxPower: yeah...  with us the demarc is generally in the equipment room.  Sometimes that's the server room, sometimes it's the utility room
13:01.00MRH2gr8 thanks tzanger
13:01.00Kattybeep!
13:01.00*** mode/#asterisk [-J] by denon
13:01.09tzangerKatty: what have you got beeping at you
13:01.10bjohnsonkonrads: someone's gonna end up running it there anyway
13:01.14tzangerwhat's catching it
13:01.20Kattyall this katrina stuff is making my hilight go off.
13:01.24tzangerhahaha
13:01.27littleballbjohnson, then the normal phone cannot recieve normal incoming call anymore, right? except asterisk server has DID number map to this fxs device
13:01.31Kattytzanger: katrina is my /nick on two other servers.
13:01.37tzangerhmm
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13:01.45*** part/#asterisk O4m68Zs (n=e1o58qt@ppp2-24.rho.forthnet.gr)
13:01.54bjohnsonlittleball: a fxo would connect your pstn to the voip system
13:02.05Kattytzanger: i've heard them all.
13:02.10tzangerKatty: probably
13:02.14bjohnsonlittleball: of course you could just use a voip provider for both incoming and outgoing.
13:02.17*** join/#asterisk Akelavlk (n=jansun@82.119.239.141)
13:02.41tzangerkatrina: that wasn't the blow job I was expecting...  Katrina had her eye on me...  what else was there
13:02.49tzangersomeo f these would make great tshirthell tshirts
13:02.59*** join/#asterisk P7b14Om (n=e1o58qt@ppp2-24.rho.forthnet.gr)
13:03.03Katty^_-
13:03.08*** part/#asterisk P7b14Om (n=e1o58qt@ppp2-24.rho.forthnet.gr)
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13:03.18littleballbjohnson, because i ty to provide such service here. That is to say, i need to apply a bulk numbers from Telco and assign to the users, right?
13:03.25tzangeroh wait
13:03.27tzangerhttp://www.tshirthell.com/katrinashirts.htm
13:03.29tzangerthey're already there
13:03.35bjohnsonlittleball: sure
13:03.57bjohnsonlittleball: you had better learn some more beside fxs vs fxo before you try to start selling voip services
13:04.02Bonzai090tzanger  were fo i add that line?.. to what .conf file?.. sorry asterisk noob
13:04.03*** mode/#asterisk [+J 30,2] by denon
13:04.12littleballbjohnson, sure sir :0
13:04.13Kattyi sure am grumpy today.
13:04.13tzangerBonzai090: go get the asterisk handbook and read
13:04.18tzangerKatty: wy?
13:04.19tzangerer why?
13:04.31Kattytzanger: ...
13:04.40Drukeni think someone should give me money so i can sell voip services :)
13:04.46Kattytzanger: if i knew why, i would have said why.
13:04.51tzangerKatty: ah
13:05.07tzangerwell I'll stuff you in a closet and find someoen hot to stuff in there with you
13:05.10tzangermaybe change your mind
13:05.12Bonzai090tzanger  aww come on man
13:05.15littleballbjohnson, i will buy TE411P digium card and dell 2850 tomorrow to try it out next week. Any idea how many users can support?
13:05.21Kattytzanger: unlikely.
13:05.26konradslittleball: as much as TE can :)
13:05.50tzangerBonzai090: it's obvious you haven't done the basic research and that's a personal pet peeve of mine.  People who try to help themselves and need help I am more than willing to assist.  People who just want someone to do it for them I send a rate sheet to
13:05.56littleballkonrads, really? do you mean dell 2850 (2 CPU, 1.5G) is enough?
13:06.03*** join/#asterisk mx02 (n=none@adsl-215-143-149.aep.bellsouth.net)
13:06.12konradslittleball: plenty
13:06.16konradslittleball: i'd say - really a lot
13:06.20tzangerlittleball: you too -- go and read the asterisk handbook, read over digium's product offerings, read the wiki.  you're nowhere near being able to offer VOIP service
13:06.48littleballtzanger, i already read lots already actually :).
13:06.52konradsshould there be an entry in voip-info.org about asterisk and ISDN cards?
13:07.01DrukenRTFM!!!! :)
13:07.04konradse.g. what cards work in what mode and with what channels/drivers?
13:07.19tzangerlittleball: then read it again, your questions indicate you didn't absorb much.  Not trying ot be mean, but just trust me on it
13:07.31littleballsure
13:07.37tzangerkonrads: I think there's some info on there, ISDN BRI isn't a really big thing here in NA so I don't have much experience with it
13:08.02konradstzanger: for us little guys it is. I'll just add the stub with what I know so far
13:08.02littleballi ask question because i try to confirm my understanding sometimes
13:08.09AkelavlkIs there any sofware for SIP FAX client?
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13:10.48ManxPowerAkelavlk, no, fax does not work well over sip.
13:10.51littleballi cannot find out how many APIC embeded in Dell 2850, any idea?
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13:12.43ManxPowerlittleball, call Dell
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13:14.59ScaredyCatanyone ever setup an mnogosearch install ?
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13:15.51littleballManxPower, the technical supporters told me they have no such information.
13:16.07ManxPowerlittleball, Then they are not supporting you, are they?
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13:16.44ManxPowerof course, if you said "How man APIC embeded in Dell 2850" I'd prolly say the same thing.
13:17.13ManxPowerperhaps you mean "How many IRQs are available if I enable APIC on a Dell 2850"?
13:17.16littleballso, how?
13:17.19littleballyes
13:17.38littleballbecause normal one APIC chip support 24 IRQs
13:18.07ManxPowerMost of the systems with APIC that I've seen support up to 254 IRQs
13:18.20littleballThat is enough then
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13:18.34ManxPowerI thought English was the the official language of .sg?
13:18.50ManxPowerlittleball, you should not need more than 24 IRQs anyway.
13:19.05ManxPowerYou can only really put up to 4 Digium cards into a system anyway.
13:19.12johnmManxPower: all i386/amd64 should support 254, via APIC and LAPIC.
13:19.13ManxPowerand even that can be tough to do.
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13:20.07littleballthere are lots information in internet about asterisk and compatibility issues... sometimes it is quite confusing.
13:20.44bjohnsonyes
13:20.58johnmlittleball: I've found that actually disabling APIC on some machines allows it to perform better. At the end of the day I've come to several conclusions...
13:21.00bjohnsonconvenient to keep the pci cards to a minimum
13:21.11johnm1 port Digium PRI cards are a no-no with onboard ethernet.
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13:21.43johnm2/4 port Digium PRI cards are a no-no on cheap motherboards. (Often, poor APIC/LAPIC implementations)
13:22.08littleballjohnm, which card support more than 4 PRI port?
13:22.21littleballi can only get 4 ports card from digium
13:22.24johnmNo more than 2 TE4XXP's in one box. More than that, and you will really have problematic call quality as it will start to drop interrupts./
13:22.34ManxPowerlittleball, There are no cards compatable with Asterisk that support more than 4 ports.
13:22.43johnmlittleball: there are 8 port cards on Junghanns but I think thats BRI
13:22.50ManxPowerYou don't want to put more than 8 PRI ports into a system anyway.
13:23.01ManxPowerhellop, you don't want to put more than 4 ports in a system.
13:23.23johnmlittleball: how many circuits do you actually need to go into a single asterisk box?
13:23.29littleballthat is to say, my dell 2850 will only support one TE411P card.
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13:23.47ManxPowerlittleball, buy more computers then
13:23.54littleballactually, we have a few sister companies and we need to provide VoIP for them
13:24.03johnmManxPower: doing that resiliantly isn't normally that easy :)
13:24.17johnmlittleball: so how many con-current PSTN calls are actually handled at once?
13:24.32littleball60 i think
13:24.37ManxPowerof course there's that little issue of codec conversions then
13:24.51ManxPowerlittleball, so you only need two E-1/PRI ports.
13:24.57johnmlittleball: then you need a minimum 2 PRI (ISDN30)'s
13:25.11bjohnsonand one or two to a channel bank(s)
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13:25.41johnmlittleball: what might be a good idea is to get 2 2port PRI cards, 2 asterisk boxes, and setup 2x ISDN30 on both. have them both SIP/IAX2 with the PBX, and use it as a make-shift channelbank
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13:25.47johnmlittleball: that will give you a lot of flexibility.
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13:27.18ManxPowerDid he say he needed analog ports?
13:27.19bjohnsonif he's going to use analog phones he's gonna likely need a few channels banks.  stick with the 2 computer concept and feed one incoming PRI to each, use extra PRI ports to go to channel banks
13:27.23johnmlittleball: resiliant to 60channels, with a maximum capacity of 120 channels I'd say.
13:27.34bjohnsonManxPower: he started asking about fxs
13:27.40ManxPowersince Asterisk with PRI and a channel bank, does not qualify as "voip" no matter what.
13:27.41bjohnsonwell, analog phones
13:27.44ManxPowerbjohnson, ah.
13:28.11bjohnsonManxPower: well, the voip part is a few feet long
13:28.36ManxPowerbjohnson, well, if you have two boxes, tehen yes.
13:28.51littleballsorry,i just check with the boss, and he told me that we need as more as possible
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13:29.13bjohnsonas many what as possible?
13:29.16johnmlittleball: if you just want 60 calls, and VoIP to clients, then get 3 boxes. use 2 of them to terminate the PRI and trunk with the other, which is used as your VoIP PBX
13:29.23bjohnsonas many concurrent pstn lines?
13:29.23*** mode/#asterisk [+f ##overflow] by denon
13:29.27johnmlittleball: if you need anything else, it shoudl integrate nicely anyways.
13:29.35h3xor just order voip from an itsp
13:29.36konradshttp://www.voip-info.org/tiki-index.php?page=Asterisk+supported+ISDN+cards
13:29.40h3xuse voip phones or soft phones
13:29.46h3xand do it all with no t1 card!
13:29.51ManxPowerAll softphones suck!
13:29.52konradsHere is my asterisk clue bit. I will add more info that I will be able to pick up from net
13:29.57littleballthanks. i need to digest again. :)
13:29.59johnmlittleball: the way I said, gives you 120 concurrant calls, and you can always upgrade to 2x 4port cards, which gives you double that.
13:30.00h3xeyebeam is nice
13:30.12konradsWonka: you could update this with your NT setup
13:30.16johnmlittleball: if you need more than that, you should really be dealing with somethign other than ISDN30!
13:30.51bjohnsonjohnm: would be nice to split up the telco PRI in case part of the system goes down
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13:31.19johnmbjohnson: of course, but theres only so far you can go with ISDN30.
13:31.43johnmbjohnson: would make more sense to use QoS assured links to VoIP wholesale providers for breakout in that case.
13:32.01littleballjohnm, so many boxes? does this increase cost?
13:32.07bjohnsonyes
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13:32.08johnmbjohnson: or keeping closed-loop, something bigger :)
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13:32.23johnmlittleball: of course, but if gives you the performance, resiliance, and setup you need.
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13:32.38bjohnsonlittleball: the absolute first thing you have to do .. is figure out what you need
13:32.42johnmlittleball: thats why I said the 2 port PRI cards from Digium. they're half the price of 4 port cards.
13:32.52bjohnsonie how many concurrent phone calls and how many extensions
13:33.02ManxPowerum, I thought they were 2/3 the price of the 4-port card.
13:33.14littleball2 port PRI cards and put them in two boxes?
13:33.17Wonkai'd like 4port BRI cards at not much more than 1port BRI cards...
13:33.24ManxPoweri.e. 1-port = $500, 2-port = $1000, 4port = $1,500
13:33.37johnmManxPower: I rang up and ordered a few today. TE405P = 895, TE205 = 479. iirc.
13:33.42johnmManxPower: not at my desk to know.
13:33.59johnmlittleball: get 2 of them. one in each box.
13:34.03littleballTE405P = 895?
13:34.16johnmlittleball: thats 2x30 channels.
13:34.21littleballit is much expensive here. TE411P about 2200USD
13:34.33johnmlittleball: you can always upgrade them to TE405P's
13:34.46johnmlittleball: Im sterling. So thats about right.,
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13:36.02littleballjohnm, why TE411P + Dell 2850 is not a good match?
13:36.21johnmlittleball: I don't know if it is or not tbh. I didnt comment on it :)
13:36.45littleballbecause the boss care about the first time investment. Two box will be more expensive than 1 box
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13:37.42johnmThe reason I said two, was because if gave you 120 chanels, with resiliance.
13:37.47littleballI choose them because digium use them as benchmark
13:37.55littleballyes
13:38.05littleballthanks
13:38.19johnmBy the way...
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13:38.46johnmI've been playing a lot (and submitting fixes back to digium) with the VPM supported cards (all with hw echo cancel) and imo, they're no better than KB1 software echo-cancel.
13:39.09littleballwhy?
13:39.10h3xwhy is that?
13:39.20johnmI'll very quickly explain what happened.
13:39.26johnmGot TE406P, drivers didnt work.
13:39.34johnmupdated to CVS, they worked, but only partially.
13:39.47johnmI saw a lot of missing support in stable drivers, and a whole array of bugs I hit in cvs.
13:39.49littleballwhat is your configurations?
13:40.16johnmTook a week of coding/messing/testing and chatting with Digium, and finally got it to a working state.
13:40.25ManxPowerjohnm, We will be moving to tellabs hardware echocan
13:40.29littleballthis is also my concern and so i decide to follow the benchmark configurations from digium
13:40.57johnmThe call echo-can would often incorrectly detect echo, it will mistcalculate it. Yopu will get high clarity calls, but with nominal/random echo and distortion.
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13:41.16h3xfuck asterisk as a t1 gateway
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13:41.24spackle~seen ManxPower
13:41.30jbotmanxpower is currently on #asterisk (1h 49m 2s).  Has said a total of 74 messages.  Is idling for 1m 5s
13:41.30h3xyou may as well use used lucent or cisco gear
13:41.30johnmManxPower: When will that be happening do you know?
13:41.38h3xit costs about the same or less when you consider the cost of a host
13:41.44littleball:)
13:41.45johnmh3x: we use an AS5350 as the whole-sale transit T1 breakout.
13:41.58johnmh3x: we need to have PRI on the PBX though, since it sites inline with the existing PBX.
13:42.01Ariel_good morning everyone
13:42.20johnmh3x: this scenario doesn't actually replace a PBX. it just re-routes calls.
13:42.23h3xare you the guy with 4dv?
13:42.36johnm4dv?
13:42.43ManxPowerjohnm, I completed testing and setup in the lab, scheduled installed at the production site, then Katrins Storm of Doom happendd and now we have a hole in the roof of the office, mold growing on the walls, no power, and wet carpet.
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13:42.58h3xnevermind, somebody just sent me an email a couple hours ago that was using a 5350 in denver
13:43.05ManxPowerjohnm, so it should be a week or so before we finish the install.
13:43.14johnmManxPower: sounds like you should film a wildlife show in there!
13:43.28johnmManxPower: the install of the newer hardware echocan?
13:43.32littleballjohnm, i think the configuration is very important. Either following other people recommendation or using digium benchmark configuration. Becuase it really involve too many things
13:43.45coppicethere is this strange belief that if an echo canceller is "hardware" it will magically work :-)
13:43.46ManxPowerjohnm, gads no!  install of a tellabs echo cancel box.
13:44.12ManxPowerThe MIS department is not allowed to use the word "mold", only the phrase "The 'M' word."
13:44.18johnmlittleball: well, I deal an awful lot with hardware in general. And I am certainly no stranger to kernel code, so I never shy too much. I pick whats best, and whats properly manufactured.
13:44.28ManxPowerafter all the problems I've heard reported about the digium echocan module.....
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13:45.00johnmManxPower: it's pretty terrible.
13:45.08Ariel_the new routine has been helping allot with echo in Heads Zaptel drivers KB1
13:45.17h3xwhere did that phantom DS3 card go from digium
13:45.18ManxPowerOh, and Bellsouth still thinks the parish (county) us still closed, even thought 2/3 of the companies are open for business.
13:45.18h3xheh
13:45.31coppicethey took forever to get this hardware EC out. how come so many people have problems?
13:45.50johnmManxPower: the thing which really kicked me in the teeth was when it arrived, they said use 1.0.9.1. so I did. and there is literally no support for it. The drivers are too buggy. the actual card sat there doing nothing, a great £1300 investment ;)
13:46.11ManxPowerjohnm, didn't read the mailing lists or ask on IRC first, huh?
13:46.13Ariel_We had to take one of them out of a system the 406 due to clicking noise. With the KB1 setup and an 405 it was great no echo.
13:46.18johnmManxPower: of course, me looking at the code, fixing it, and then telling the support guys *exactly* what the problem is just got the routine questions back.
13:46.27littleballjohnm, do you suggest to use TE410P rather than TE411P card? because the later one is hardware echo..
13:46.28ManxPowerNo, I'll stick with the $50 echo cancel from tellabs
13:46.43johnmManxPower: of course, and I even got someone with the exact same setup (apparently) and it worked.
13:46.50johnmManxPower: same hardware, card, config.
13:47.18spackleManxPower, we talked about PRI timing yesterday - I changed mine back to what you suggested and it works fine.  You were right.  Thanks.
13:47.23johnmlittleball: Id recommend using the KB1 software echo-can, unless the call quantity is so demanding you need to use onboard stuff.
13:47.30ManxPowerI'm too old to be an Early Adopter
13:47.35coppicemost hardware echo cancellers actually suck, but not for the reasons people have problems with the digium ones
13:47.50ManxPowerspackle, <dry tone>I usually am.</dry tone>
13:48.01johnmManxPower: Early adoption doesn't bother me. it's ignorant/dismissive support staff which do ;)
13:48.10tzangerthe digium one seems to work pretty good
13:48.12johnmalthough that being said, they did try and help.
13:48.23tzangerI have an issue but I think it's the MB, I am swapping mb's tonight
13:48.29littleballthanks. johnm
13:48.35ManxPowertelcos use tellabs -- that's good enough for me
13:48.53johnmManxPower: what exactly is it you have?
13:49.14coppiceManx: telcos thought MMS was going to be a money spinner. i don't get your point :-)
13:49.18Ariel_ManxPower, you have a link for the Tellabs
13:50.24ManxPowerAriel_, I get them off ebay
13:50.43Ariel_ManxPower, ok will look for them.
13:50.56ManxPowerAriel_, most of the time you buy the rack with 16 modules.
13:51.05Ariel_ManxPower, but just for your info the KB1 is a very nice echo fixer
13:51.09ManxPowerWe plan on selling the unusued modules back on ebay
13:51.23ManxPowerAriel_, *nod*  I'll have to try KB1 in production when 1.2.3 comes out.
13:51.33spackleManxPower, You are pleased with the tellabs echo can then?
13:51.42johnmManxPower: got any details on it, and where it sits in the topology?
13:51.53ManxPowerspackle, I don't know.  we were going to install the first one before Katrina: Storm of Doom.
13:52.01ManxPowerBut I got calls running thru it in the lab
13:52.10*** join/#asterisk docelm0 (n=docelm0@66.237.242.41.ptr.us.xo.net)
13:52.22ManxPowerI must go find food.
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13:56.05littleballwhat is "Asterisk Business Edition"?
13:56.18littleballhttp://www.digium.com/index.php?menu=product_detail&category=software&product=ABE&tab=compatibility
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13:58.04Ariel_littleball, it's asterisk stable biz version which they give support and a licence to use.
13:58.06MikeJ[Laptop]littleball, snapshot of cvshead, stripped down and tested
13:58.13johnmlittleball: a digium supported asterisk release.
13:58.27littleballhow about 1.0.9? i will use this
13:58.29MikeJ[Laptop]all of the above!
13:58.43MikeJ[Laptop]what about it?
13:58.45jake1932anyone know why System won't work?  It worked before a reboot.
13:58.47Ariel_littleball, for production systems I only use stable
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13:58.59Ariel_jake1932, what did you change
13:59.04jake1932nothing
13:59.07Ariel_sure
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13:59.12jake1932no - i swear
13:59.14johnmlittleball: I'd probably consider waiting till 1.2 final comes out. Otherwise use a recent CVS snapshot and see how it acts for you.
13:59.15MikeJ[Laptop]just so everyone knows, no release will have the name "stable" in the future
13:59.16jake1932but
13:59.22littleball1.0.9 is stable version. From this irc, i was scared because it seems too many problems :(
13:59.35jake1932I opened a real large file in nano and got a SIGTERM
13:59.39MikeJ[Laptop]we really are not even calling 1.0 branch stable
13:59.43MikeJ[Laptop]they are releases
13:59.55MikeJ[Laptop]and cvs
14:00.05johnmlittleball: well for me there is 1 big reason to change. 1: Generic Jitter Buffer.
14:00.14Ariel_MikeJ[Laptop], yes but in turn you will have the new head when the 1.2 comes out
14:00.17jake1932so my guess is that some service stopped that supports the System cmd?
14:00.20littleballwhich version is the most stable?
14:00.25johnmthe SIP JB may be included at some point soon,. but if it isn't you can easily patch it in
14:00.33littleballjohnm, change to what?
14:00.41johnmyou will want a SIP JB if you are dealing solely in SIP on existing infrastructure.
14:00.42MikeJ[Laptop]littleball, what does stable mean to you
14:00.48Ariel_littleball, what are you going to do?
14:01.10Ariel_I use stable 1.0.9 due to it's not changing every day so new problems don't come up.
14:01.13littleballi am going to buy dell 2850 and et411P card to configure my first VoIP service
14:01.20littleballfor a few sister companies
14:01.26Hmmhesaysugh, why change you change the value of a set variable in the dialplan
14:01.28MikeJ[Laptop]Ariel_, snapshots of anything never change :D
14:01.33Hmmhesays*why can't you even
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14:01.51Ariel_MikeJ[Laptop], yes your correct
14:01.57littleballAriel_, any advice?
14:02.10MikeJ[Laptop]ariel, in regards to 1.2, it will not be called stable ever
14:02.13jake1932is there a service that supports the System cmd that needs to be running?
14:02.13MikeJ[Laptop]it is just a release
14:02.20Corydon76-homejake1932: No
14:02.26Ariel_littleball, Like I said for production system I use 1.0.9 for testing I have started to use head.
14:02.29MikeJ[Laptop]and we are no longer calling 1.0.x releases "stable", just release
14:02.45kpettitI'm compiling spandsp with Fedora 2, getting some compile errors with the tiff library
14:02.51Corydon76-homeMikeJ[Laptop]: we should go one step further and call them escaped
14:03.00Ariel_MikeJ[Laptop], got it
14:03.07johnmAriel_: and you've never had problems with SIP calls?
14:03.07littleballok. i will try to use 1.0.9 first. If having problem, try other snapshots... One by one...
14:03.13MikeJ[Laptop]littleball, if you have no need for specific functionality in anything above 1.0.9, use 1.0.9 for now.
14:03.14Ariel_johnm, no
14:03.14Corydon76-home"Our software isn't released; it escapes."
14:03.15kpettitDoing a google I see most people suggest reverting to a older tiff library, but i'm already using that one so I'm not sure what the isses is.
14:03.22lehelhow do i setup a FAX?..
14:03.31johnmAriel_: I take it there aren't many calls, and/or it's a dedicated physical network?
14:03.33littleballActually, no need more features
14:03.34kpettitlehel, look at spandsp
14:03.35MikeJ[Laptop]if you are still in a testing phase, go for 1.2 beta when the next one comes out and give us feedback
14:03.54MikeJ[Laptop]Corydon76-home, what are you doing home today ?  :P
14:04.05Ariel_johnm, I have it on 9 systems connected together servicing over 3000 users
14:04.06Corydon76-homeMikeJ[Laptop]: I'm about to leave for work
14:04.07MikeJ[Laptop]escapes, nice
14:04.47johnmAriel_: I'm very surprised, if thats on the same infrastructure as data, that you dont get jitter.
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14:05.13jake1932Corydon76-home: any logs to look at to see why the System cmd would be failing?
14:05.14Ariel_johnm, well jitter has never come up with sip setup for us.
14:05.28littleballjohnm, currently, i am using 1.0.9 for SIP phones, it works fine. the voice is not bad
14:05.41Ariel_lehel, what type of faxes are you setting up?
14:05.43Corydon76-homejake1932: Nope
14:05.59johnmlittleball: I found it very problematic when you have a long-distance call, which is primarily PRI, but is carried a decent distance by SIP.
14:06.02Corydon76-homejake1932: have you tried your System command on the command line?
14:06.13jake1932yes - it works from there
14:06.20Corydon76-homejake1932: what's your command?
14:06.24lehelAriel_: i think it is a Brother MFC..
14:06.26Ariel_johnm, you have other issues.
14:06.29littleballit wont happen for me because my server will located near the switch
14:06.40jake1932<PROTECTED>
14:06.48johnmAriel_: other issues effecting the quality, or mental issues? :)
14:06.51lehelAriel_:  capiAnswerFax: Answer Fax with CAPI
14:06.52Corydon76-homejake1932: and where is that script located?
14:07.03Ariel_lehel, ahh don't use capi
14:07.10jake1932Corydon76-home: /etc
14:07.13Corydon76-homejake1932: I would suggest that you include the FULL PATH to the file
14:07.17johnmlittleball: because it's near a switch, doesn't mean you wont get jitter :)
14:07.21Ariel_or get updated drivers lehel
14:07.36jake1932exten => s,3,System(sh /etc/sqlpost.sh ${UNIQUEID} ${ACCTNUM} ${PAYAMT} ${AUTHONUMBER} S)
14:07.39Corydon76-homejake1932: if you cd to /tmp and run "sh sqlpost.sh" I would think it wouldn't work, either
14:07.52littleballjohnm, i will use E1 cable connect to switch directly
14:07.58Corydon76-homejake1932: How about /bin/sh ?
14:08.07jake1932worth a try
14:08.10lehelAriel_: CAPI it's not good for FAX?, i have the last chan_capi
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14:08.56Corydon76-homejake1932: chances are if you started up asterisk from an init script, it has no PATH setting and can't search for your sh binary
14:09.23littleballSir, may i know where are you all from? i try to find out the deployment of asterisk geographycally
14:09.39Ariel_lehel, sorry I miss understood you. Your going from capi to a fax machine brother via what port connecting to the brother?
14:10.05johnm<- UK
14:10.13Corydon76-homelittleball: it's international.
14:10.14*** join/#asterisk newl (n=newlook@203-59-184-194.dyn.iinet.net.au)
14:10.27jake1932Corydon76-home: exten => s,3,System(/bin/sh /etc/sqlpost.sh ${UNIQUEID} ${ACCTNUM} ${PAYAMT} ${AUTHONUMBER} C) - doesn't work
14:10.34littleballthaks.how about Ariel_ and Corydon76-home...
14:10.42littleballi am from Singapore
14:10.58Corydon76-homelittleball: if you really wanted to find this out, you should take a look at everybody's IP address and query the whois server for the location
14:11.28Hmmhesaysargh, is there any way to change the value of a currently set variable in asterisk?
14:11.30littleballyes. if we report ourselves, all people can get the result :)
14:11.31Corydon76-homejake1932: have your run your script after the last time you rebooted?
14:11.45Corydon76-homelittleball: this is not the forum for that
14:11.59littleballok
14:12.09jake1932Corydon76-home: just ran the script from the cmd prompt - works fine (after reboot) - but no go from the dialplan
14:12.10lehelAriel_: it is a Fritz card; in Port 2, i have the MSN (ISDN line); in Port 3 i'll have the Fax machine
14:12.25Corydon76-homejake1932: don't know what to tell you
14:12.38Ariel_did not know that they had an analog out put.
14:12.52Corydon76-homejake1932: you probably need someone else to take a look at your system.  It's probably a permissions problem
14:13.12jake1932Corydon76-home: just did a chmod 777 on the .sh file
14:13.29Corydon76-homejake1932: what is your script accessing?
14:13.36*** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm)
14:13.37jake1932Corydon76-home: wget
14:13.49Corydon76-homejake1932: why not use the Curl app?
14:14.06jake1932Corydon76-home: curl never worked in the version of asterisk i was using
14:14.36Corydon76-homejake1932: upgrade to the public beta
14:14.53jake1932Corydon76-home: on a production system?
14:14.56Corydon76-homeand now I have to leave
14:15.10Corydon76-homeYes, on a production system
14:15.16lehelAriel_: how could i deal with it?
14:15.22jake1932Corydon76-home: ok - tnx
14:17.27*** join/#asterisk b0ef (n=b0ef@062016141085.customer.alfanett.no)
14:18.21Ariel_lehel, about the middle of this pages it talks about capi and faxs http://www.voip-info.org/tiki-index.php?page=Asterisk%20fax
14:18.26Ariel_lehel, have you read it?
14:19.01leheli'm reading now..;d
14:19.28kpettitanybody know a good mirror where I can get spandsp-0.0.2X?
14:19.45coppicespandsp.sourceforge.net
14:19.45kpettitall the links I'm getting for it are down
14:19.55tzangercoppice: you moved it to sf?
14:20.02HmmhesaysAriel_ can I pm you?
14:20.03kpettitcool! thanks
14:20.06coppicenot moved. i put a copy there
14:20.07*** join/#asterisk I5e27Nd (n=w6g8zy@58.69.19.81)
14:20.13Ariel_Hmmhesays, yes
14:20.16*** part/#asterisk I5e27Nd (n=w6g8zy@58.69.19.81)
14:21.57kpettitis there a website for spandsp?
14:22.13kpettitsweeet, this copy gets rid of that tiff bug I was having before
14:36.31*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
14:36.31*** topic/#asterisk is Asterisk 1.2.0 Beta1 - http://www.asterisk.org || Astricon 2005 - Anaheim, CA - Oct 12-14 - http://www.astricon.net/2005 - Speakers wanted! || A core dump is your computer's way of saying "Here's what's on my mind, what's on yours?"
14:36.51*** join/#asterisk MicC_ (n=sum1@70.28.73.160)
14:37.00MicC_howdy
14:37.23*** join/#asterisk postel (n=canonica@host217-42-116-159.range217-42.btcentralplus.com) [NETSPLIT VICTIM]
14:37.50*** join/#asterisk toddf (n=toddf@ns0.fries.net)
14:38.01toddf+overflow is annoying
14:38.19MicC_looking to do pin collection (enter your account #) with IVR
14:38.29bkw_http://www.therandirhodesshow.com/live/node/905
14:38.35*** join/#asterisk nel (n=nel@199.75.113.6)
14:38.41MicC_problem is I have 50 customers. Any ideas on how to make it easy on me?
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14:42.36coppicebkw_: they are poor people. they don't pay enough tax to be of interest to the government :-\
14:42.56*** join/#asterisk docelm0 (n=docelm0@66.237.242.41.ptr.us.xo.net)
14:43.20Hmmhesaysbkw_ why can't you change the value of a variable set in the dialplan
14:44.09*** join/#asterisk syle (n=blah@unaffiliated/syle)
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14:44.30nelhello
14:44.53Hmmhesayswelcome to netsplit
14:44.54Hmmhesayser..
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14:46.05bkw_Hmmhesays, what do you mean?
14:46.13*** mode/#asterisk [+o bkw_] by ChanServ
14:46.26neldoes anybody has experience with ser and asterisk implementations?
14:46.57Hmmhesayss,1,Set(matt = sexykinda); s,2,Set(matt = notsomuch);
14:47.03*** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net)
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14:47.31Hmmhesayswhen I got to s,3,goto(${matt},1) its going to extension sexykinda
14:48.38bjohnsonHmmhesays: yes
14:48.53bjohnsonbut shouldn't it be s,3,goto(${matt},s,1)
14:49.06Hmmhesaysbjohnson, thats not the problem, look at the example
14:49.12bjohnsonto go to s,1 of exten (whatever matt is set to)
14:49.21Hmmhesaysread the example
14:49.27bjohnsonwhat example?
14:49.36*** join/#asterisk C2k94Ph (n=m4i28ha@213.228.180.44)
14:49.39Hmmhesaysthat is a valid goto, this isn't a context issue
14:49.45*** part/#asterisk C2k94Ph (n=m4i28ha@213.228.180.44)
14:49.56bjohnsonso what is the question?
14:50.09Hmmhesaysgoto is valid for goto(priority) goto(extensions, priority) goto(context,extension,priority)
14:51.01bjohnsonoh .. and you say it is doing goto(context,extension) instead of goto(extension,priority)?
14:51.12Hmmhesayss,1,Set(matt = sexykinda); s,2,Set(matt = notsomuch); s,3,goto(${matt},1) its going to extension sexykinda
14:51.20Hmmhesaysno, thats not what i'm saying at all bjohnson
14:51.26bjohnsonI give up
14:51.27Hmmhesayslook at the line above
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14:57.47HmmhesaysARGH
14:57.53Hmmhesayssyntax error on my part
14:58.02bjohnsonsetvar?
14:58.34Hmmhesayssetvar is deprecated in 1.2
14:58.57Hmmhesaysi am such a tool
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15:00.33lehelAriel_: it is a Canon B820.. i found this how to use: http://paste.debian.net/1829, i don't now exactly that i have to cahnge to "123" with the fax's number? or where i set the fax-number?
15:00.33*** part/#asterisk E8p1Zm (n=o6m73nc@60.48.67.232)
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15:02.25nelanybody has experience with ser and asterisk?
15:02.51nelI'm using serweb, do I have to touch ser.cfg or I have to do everything from the web interface?
15:03.29nelI'm registering users using serweb, how do I make asterisk to connect to ser and viceversa?
15:04.00Hmmhesaysgood question
15:04.03Hmmhesaysask me again in 2 weeks
15:04.12file[laptop]you have to setup ser.cfg, and asterisk
15:04.18nellol
15:04.50file[laptop]ser.cfg is how SER handles stuff - you have to tell it that stuff... the web interface just fills in values in the database for various modules to use
15:05.00*** join/#asterisk JerJer (n=JerJer@pdpc/supporter/bronze/jerjer)
15:05.31nelfile: so...I have to use ser.cfg to route calls to asterisk? i.e forward all sip starting with 90 to the ip address of asterisk srever?
15:05.40file[laptop]yes
15:06.22nelhmm I see
15:06.43*** join/#asterisk ytk (n=aaa@82.211.129.231)
15:07.07nelso if I have my username and password on ser that I created through serweb, my phone should register in ser?
15:07.22file[laptop]if you have it setup in ser.cfg that you handle registrations, sure
15:07.27nelor should I create the phone in sip.conf in asterisk?
15:07.30file[laptop]everything is not automatic
15:07.33file[laptop]you have to tell it what to do
15:07.59nelthanks
15:07.59*** join/#asterisk Qwell (n=north@24-50-66-194.vnnyca.adelphia.net)
15:08.02Hmmhesaysshould I be nice and add some stuff to the wiki?
15:08.09file[laptop]you could
15:09.13*** join/#asterisk yartelecom (n=no-email@82.211.129.231)
15:12.20sylewhat you guys recommend for creating outgoing callcenter app
15:12.39sylelooking at agent stuff but that seems more for incoming
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15:15.17file[laptop]yay outside
15:17.27sylewtf
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15:21.38sylehow do you use chanspy?
15:21.44Hmmhesaysvery carefully
15:21.50syleexten => 556,1,ChanSpy(scan)
15:21.51file[laptop]show application chanspy
15:21.55syletried that
15:22.06syledialed a number with sip phone
15:22.10syleall it does is beep
15:22.22syleChanspy([<scanspec>][|<options>])
15:22.41sylewhat is scanspec
15:22.44Hmmhesayswow O wow this dialplan is on crack
15:22.50Hmmhesaysi've created a monster
15:23.03file[laptop]syle: read what show application chanspy says
15:23.06file[laptop]it tells you
15:24.05*** join/#asterisk ManxPower (n=eric@ip68-225-97-156.br.no.cox.net)
15:24.12syleso can i do chanspy(SIP)
15:24.20file[laptop]yes
15:24.21file[laptop]just try it
15:24.37file[laptop]it's fully documented, you just have to read what it says there and experiment
15:25.04sylei'm reading its not a hell of alot different from http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20ChanSpy
15:25.36lehelAriel_: it is a Canon B820.. i found this how to use: http://paste.debian.net/1829, i don't now exactly that i have to cahnge to "123" with the fax's number? or where i set the fax-number?
15:25.50*** join/#asterisk Snake-Eyes (n=blog@203.201.96.188)
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15:28.17HmmhesaysI think Ariel_ is off fixing a server
15:29.00leheloh.. right..
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15:32.16Kattyzomg
15:32.31file[laptop]awwww
15:32.46Kattyeverything is borken here!
15:32.52Hmmhesayshaha i'm at about the same place
15:32.53Kattyand everyone wants something from me - and right NOW!
15:33.08HmmhesaysI'll take a coke please
15:33.29Kattyi'll take a nap.
15:33.33file[laptop]Hmmhesays: a coke, or some coke?
15:33.59Hmmhesays<shrugs> a little from column A, a little from column B
15:34.01KattyHmmhesays would make a pathetic drug addicted.
15:34.09file[laptop]scary
15:34.19Hmmhesayswhy is that?
15:34.25Kattyyou don't like needles, do you?
15:34.31Hmmhesayswho said that?
15:34.35Kattyi see.
15:35.06HmmhesaysI don't like used needles that a hooker just injected herself with
15:35.26ManxPowerneedles should only be handled by a qualified health professionsl
15:35.28Kattynext!
15:35.37Hmmhesaysor, his self
15:35.42Hmmhesayswhichever the case may be
15:35.43coppiceor a seamstress
15:35.49*** join/#asterisk jets (n=b@guardian.pmt.org)
15:35.54ManxPowerHmmhesays, no, ONLY by a qualified health professional
15:36.22Hmmhesaysyou mean that guy thats tying his arm off with a nylon string ISN'T a qualified health professional?
15:36.48ManxPowerHmmhesays, then he should not be doing that 8-)
15:37.05Hmmhesaysgeebus, no humor
15:37.08Hmmhesays:D
15:37.27Hmmhesaysmama says if you keeping making a face like that, it'll get stuck that way 0_o
15:37.40*** join/#asterisk b0ef (n=b0ef@062016141085.customer.alfanett.no)
15:37.50*** join/#asterisk kiwnix (n=egarcia@82.158.159.143)
15:37.56coppicequalified health professional == hot nurse
15:39.09steve___she also said santa was real....  what a terrible shock to the system it was on that day in grade 4 when I found out otherwise
15:39.21steve___never believe moms!
15:39.25steve___nbm
15:39.30Dus10anyone know a channel for asterisk@home?
15:39.54coppicewhen my 6 year old told me santa wasn't real it really ruined my day
15:40.15steve___nbc as well  :)
15:40.37steve___i was 9
15:40.38ManxPowerthe population of Bation Rouge LA increased by 350,000 people in 1 week.
15:40.42steve___and stubborn
15:40.52steve___it was like 29 to 1 in that classroom
15:40.54ManxPowerThere are 23,000 people that lived in new orleans that are missing
15:41.20coppiceManxPower: is that different from normal?
15:41.20JerJerDus10:  yeah copy it to /dev/null
15:41.27HmmhesaysI wish you could use n+1 in gotoif()
15:42.39twisted[asteria]Katty, welcome to my day yesterday
15:43.02Dus10seriously, I need to get it setup to connect to broadvoice
15:43.20*** join/#asterisk feist (n=feist@nat-pool-msp.redhat.com)
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15:47.08ManxPowerHmmhesays, $[${PRIORITY} + 1] doens't work?
15:47.29*** part/#asterisk T-Squared (n=ted@hidden.serreyn.com)
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15:54.45eKo1I have two * boxen, A and B. I'm trying to send calls to a particular extension on A to B. I see the call comming in on B but it is going to the default context in sip.conf on A.
15:55.03*** join/#asterisk distortio (i=distorti@junipero.3sheep.com)
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15:56.26Kattytwisted[asteria]: mew :<
15:57.12*** join/#asterisk ptblank (n=MURDER1@68.169.160.44)
15:57.59HmmhesaysManxPower, didn't try that
15:58.04Hmmhesayshaha I'm going to quit here
15:58.11Hmmhesaysthey keep pulling this bullshit
15:58.19*** join/#asterisk santiago (n=santiago@63.245.87.180)
15:59.11*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
15:59.46Hmmhesaysthey just told me I can't tell a customer pricing on anything
15:59.50Hmmhesaysfsck them
16:00.08*** join/#asterisk file[muon] (n=file@mctnnbsa30w-156034035130.nb.aliant.net)
16:02.28*** join/#asterisk loud (n=ariel@cypher.punk.net)
16:02.40blitzrageyah, thats a great way to drum up business, "no sir, I can't tell you how much it'll be - but you should buy it because its great!"
16:03.01blitzragefile[muon]: ahoi
16:03.05wunderkinheh
16:03.20*** join/#asterisk cpatry (n=grepmoo@65.39.228.5)
16:03.26blitzragecpatry: zup zup
16:03.26file[muon]Junky!
16:03.30cpatryyo yo.
16:03.51Hmmhesaystheres going to be an open job here in fargo for voip stuff if someone wants it, lol
16:04.00file[muon]let me think... nah
16:04.03file[muon]I'll pass
16:04.13cpatryvoip with farmers, no thx :P
16:04.23cpatryHmmhesays: u changing job?
16:04.34HmmhesaysI'm thinking about grabbing my gear and walking out of here
16:04.45blitzrage"holy calamity, scream insanity! All you ever gonna be is another great fan of me - break!"
16:04.58file[muon]blitzrage: you're silly
16:04.59tzangerok here's andrew's fucked up thing of the day
16:05.04blitzragetzanger: lol
16:05.16*** join/#asterisk haxos (n=jaspe@wbs-146-147-27.telkomadsl.co.za)
16:05.40tzangerI have two * boxes, one's running 2.6.11 SMP on a P4/2.4 (shows only 1 CPU in cpuinfo).  The ohter's running 2.6.10 SMP on a P4/3.0, shows 2 in cpuinfo
16:05.47haxoscould anybody help me with a codec problem?
16:05.48tzanger#1 has TE406, #2 has TE405
16:05.58tzangerI'm doing 15 g729 translations on both
16:06.42*** join/#asterisk drumkilla_laptop (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
16:06.42*** mode/#asterisk [+o drumkilla_laptop] by ChanServ
16:06.43Kattyvolunteers?
16:06.47tzangerbasically PRI ulaw->g729 - iAX2 - g729->ulaw - PRI out,PRI in - ulaw->g729 - IAX2 - echo()
16:06.51tzangeroh that's why the load's so much lower
16:06.52tzangerduh
16:07.03Kattyblitzrage: warning, this may hurt a bit.
16:07.07file[muon]DARUMKILLA!
16:07.07blitzrageKatty: good :D
16:07.09Kattyblitzrage: k, probably more than a bit.
16:07.18tzanger#2 is only doing ulaw->g729 once, #1 is doing it twice
16:07.19file[muon]blitzrage likes teh pain
16:07.51drumkilla_laptopfile[muon]: wtf is a muon
16:07.57file[muon]sub atomic particle
16:07.58tzangermuons are subatomic particles
16:08.00haxosplease? im having problems with sound quality
16:08.04*** join/#asterisk I1l20Dq (n=o6j85gs@85.101.101.99)
16:08.10file[muon]all my boxes are named after subatomic particles
16:08.13file[muon]well, sorta
16:08.13*** part/#asterisk I1l20Dq (n=o6j85gs@85.101.101.99)
16:08.35coppicemuons are american particles. europeans have aons
16:08.38tzangerdoes echo() require a codec translation to slinear and back out?
16:08.56twisted[asteria]file[laptop]: geek.
16:09.00tzangeror does it just take the same frame (irrespective of codec) and spit it back out
16:09.03*** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr)
16:09.11file[muon]twisted[asteria]: geek.
16:09.28twisted[asteria]echo
16:09.32steve___all my boxers have brown particles
16:09.42twisted[asteria]okay steve___ that's TMI
16:09.44twisted[asteria]kthx
16:10.04*** part/#asterisk Dus10 (n=Dus10@adsl-69-210-12-146.dsl.ipltin.ameritech.net)
16:10.09steve___(in europe they were yellow)
16:10.17steve___:)
16:10.47bkw_http://www.osxvnc.com/
16:10.49bkw_hahahahhaha
16:11.01file[muon]I'm remembering weird stuff lately
16:11.11twisted[asteria]bkw_, uhm
16:11.16bkw_thats NOT RIGHT
16:11.19tzanger<PROTECTED>
16:11.19tzanger<PROTECTED>
16:11.24file[muon]twisted[asteria]: like those dell drive rails, as you can see I'm in no hurry to get them
16:11.35twisted[asteria]file[muon], d'oh
16:11.36tzangerdoes * need to hear the audio to do that?  i..e is a codec translation required for echo() to work?
16:11.37file[muon]and I also remembered that Alan, David, and I ran through a toll in Chicago
16:11.45coppicebkw: what's not right? sex with large chickens?
16:11.46bkw_hahaha
16:11.59bkw_coppice, you think its OSX VNC
16:12.04file[muon]"Do you have change?" "No!" "Ah screw it, drive through!"
16:12.20tzanger<PROTECTED>
16:12.20tzanger<PROTECTED>
16:12.32*** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com)
16:12.34tzangerhmm I bet no codec translator's reqired
16:12.39tzangerbkw_: can you comment on app_echo.c?
16:12.47tzangerblitzrage: no, that's furry
16:12.55tzangerunleash the furry
16:12.56bkw_tzanger, no codec required
16:12.58tzangershe's a kitten remember?
16:13.04blitzrageLOL
16:13.04bkw_it will fire back what you send at it
16:13.06steve___bkw_  hahaha
16:13.08bkw_so will app_record
16:13.09Kattysince when was i a kitten?
16:13.14tzangerbkw_: ok that explains it then
16:13.20steve___bkw_  the choices at the bottom...  too funny
16:13.23Kattyi may have feline tendencies, but i'm definately a human female.
16:13.29coppicebkw: any interesting seg faults lately?
16:13.36*** join/#asterisk Samoied (n=Samoied@201.11.86.133)
16:13.38file[muon]app_record + rtptimeout = BAD
16:13.39bkw_coppice, not yet
16:13.42bkw_i'll tell ya in a few
16:14.00twisted[asteria]Katty, *smirk*
16:14.07*** join/#asterisk sh4d0vv (n=johndoe@c66.110.147-29.clta.globetrotter.net)
16:14.20tzangerbkw_: ok thanks
16:14.36sh4d0vvanything wrong with asterisk.org?
16:14.45twisted[asteria]file[muon], that's a strange poke.
16:14.48tzangerFYI, a P4/2.4GHz (not HT, at least not detectable by linux) can handle 15 g729<->ulaw without hitting 1.0.  I hit 0.78 in my testing
16:14.57tzangerIAX2 -> Zap, Zap -> IAX2
16:14.57sh4d0vvi get connection refused
16:14.59file[muon]twisted[asteria]: have to be creative these days
16:15.11tzangerI imagine sip will be identical or at least close enough
16:15.20coppicevery pythonesque form of attack
16:15.53Katty<PROTECTED>
16:16.11twisted[asteria]Katty, :(
16:17.07knight_hey folks
16:17.47sh4d0vvCan anybody reach www.asterisk.org?
16:17.59tzangerit looks like a dual P4 should handle g729'ing an entire PRI with asterisk's g729 codec module
16:18.09*** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
16:18.33Corydon-wUh, a single P4 should be plenty for g729 on 4 PRIs, let alone one
16:18.34Delta34hi can anyone think of why when I call an ivr thru the zap channel, as soon as the ivr answers say 3-4 seconds into the ivr, the zap channel hangs up the call.
16:18.35coppiceI can reach www.asterisk.org, and it says "connection refused" :-)
16:18.51sh4d0vvthanks
16:19.00redder86coppice: me too
16:19.06coppiceCorydon-w: you haven't tried that, have you :-)
16:19.14tzangerCorydon-w: my testing (P4/2.4 no HT) showed loadavg of 0.78 for 15 ulaw<->g729 translations (going iax-pri out, pri in-iax)
16:19.25Corydon-wcoppice: No, I've only tried it with dual P3s
16:19.36tzangerwith a TE406, so there was no echo can overhead in that either
16:19.39Corydon-wdual P3 800s
16:19.48tzangerCorydon-w: what were you able to do with dual P3/800s
16:19.59Corydon-wtzanger: all 4 PRIs
16:20.09coppicedual xeons at the top of the range can do about 4 spans of G.729
16:20.21tzanger120 g729->ulaw translations ??!  I want to see that
16:20.25redder86coppice: in t31_init what are at_tx_userdata and call_control_userdata for?
16:20.33Corydon-wNot 120.  That's for E1
16:20.38Corydon-w92 is for T1
16:20.38tzangerCorydon-w: well 96 then even
16:20.48tzangeractually 92 is for PRI with no NFAS
16:20.58coppiceredder86 user definable callback parameters
16:20.59Corydon-wYeah, well, we have no NFAS
16:21.03*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
16:21.03*** mode/#asterisk [+o anthm] by ChanServ
16:21.18tzangerCorydon-w: as I said my single-proc, no HT P4 2.4 was almost topped out (I'm calling loadavg of 1.0 topped out) at 15 translations
16:21.28redder86coppice: not just the handlers?
16:21.42coppiceCorydon-w: you certainly didn't do 96 ports of G.729 with digium's g.729 on a PIII
16:22.17coppiceredder86: that parameter is passed to the handler. Look how it is used in my code.
16:23.16goof_need help. I am able to call out from sip ext to outside via zap channel. For Inbound calls, * just says "goodbye"
16:23.30Kattytwisted[asteria]: :<
16:23.34Kattytwisted[asteria]: *hug* (=
16:24.04redder86coppice: the only place where I see t31_init used in your code is in the tests, and in those instances the user_data is NULL.
16:24.14*** join/#asterisk ast_freak (n=jesse@hades-out.universalsystems.net)
16:25.03redder86is Steve K here by chance?
16:25.10tzangerCorydon-w:
16:25.12tzangerg729 performance taken from the Digium web site: Internal testing with dual Intel? Xeon 1.8GHz processors allowed 60 concurrent calls. Dual Xeon 2.8GHz processors allowed 80 concurrent calls.
16:25.26tzangeryou got 92 with dual P3/800s?
16:25.39tzangersteve___: ^^
16:25.43coppicetzanger: those figures are about right
16:26.09*** join/#asterisk shmaltz (n=chatzill@69.28.255.210)
16:26.20Corydon-wtzanger: this isn't a testing scenario
16:26.34Corydon-wtzanger: we have had that many calls up
16:26.36tzangerCorydon-w: agreed, but again testing is usually skewed in favour of the vendor :-)
16:26.50coppicesome of the commercial g,729 codecs for DSPs seem far more efficient than that.
16:27.09Corydon-wtzanger: believe what you will
16:27.39coppicejust running the codec and nothing else on my dual 2.4GHz xeon gave about 40 channels per CPU
16:27.40tzangerCorydon-w: I'm *not* calling you a liar, I'm just interested in more details since I can't get anywhere near that with a P4
16:27.57redder86steve___: can you comment on the initiliazation of fr in iax_send (in iax.c)?  Why is there a different initialization for "now"?  The "now" method of initialization leaves much to be desired.
16:28.00*** join/#asterisk oej (n=Olle@apollo.webway.se)
16:28.16shmaltzanybody know any channel bank that supports VoIP with FXO?
16:28.19*** join/#asterisk Y7o18Nj (n=f2e7co@219.95.74.34)
16:28.30*** part/#asterisk Y7o18Nj (n=f2e7co@219.95.74.34)
16:28.32tzangershmaltz: none... they all support T1 with FXO :-)
16:28.50*** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de)
16:29.14shmaltztzanger, I know they support T1 with FXO, I'm looking for VoIP with FXO like the Adit does with the FXS and CMG combo
16:29.39tzangershmaltz: not sure... but why not throw a little epia system or something, velcro/duct tape/bolt it on top of the channel bank and do whatever you want?
16:29.55redder86steve___: that code actually comes from Digium's sources... so maybe someone who's familiar with Digium's libiax2 can comment?
16:30.12tzangerredder86: steve___ isn't stevek
16:30.20shmaltztzanger, thats exactly what I want to do, but I want the channel bank to handle the transcoding to G729
16:30.26redder86tzanger: oh, well, okay
16:30.42tzangershmaltz: I've never heard of that before... I'd let the damn PC handle that instead of looking to stuff money into a CB
16:30.46*** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de)
16:31.20shmaltztzanger, how the hell will a epia system handle 24 G729 to PRI calls?
16:31.35tzangershmaltz: well a small P4 system then :-)
16:31.54tzangerthe point was, put the money in the PC, not the proprietary channel bank
16:31.57ManxPowershmaltz, Channel banks don't transcode.
16:32.22shmaltzManxPower, they do if the have on board VoIP cards like the CMG in an Adit 600
16:32.24ManxPoweryou transcode when you do VoiP and a chanbel bank doesn't do VoIP.  If it does VoIp then it's NOT a channel bank
16:32.27sivanatzanger: is Norm still around?
16:32.36tzangersivana: I haven't seen himn around in forever
16:32.36ManxPowershmaltz, no, then it's a VoIP Gateway
16:32.52shmaltztzanger, so now you moving awya, but show me a mini-itx mb with a PCI card that has a dedicated IRQ?
16:33.01sivanatzanger: you've tried his service in TO?
16:33.17shmaltzManxPower, you are right with the defenition, but since the Adit 600 is a channel bank it is much cheaper
16:33.27ManxPowershmaltz, so use it
16:33.32tzangershmaltz: get APIC support enabled and it will
16:33.36tzangershmaltz: almost guaranteed
16:33.46shmaltzManxPower, but it doesn't support FXO with the VoIP
16:33.58ManxPowershmaltz, Oh well.
16:34.06shmaltztzanger, you mean enable APIC in the BOIS?
16:34.13*** join/#asterisk weazul (n=weazul@82-169-62-42-mx.xdsl.tiscali.nl)
16:34.44sivanatzanger: I wonder if he could get me a couple DS1
16:34.45ManxPowershmaltz,
16:34.50sivanaor channels or whatever it's called
16:34.51shmaltzManxPower, while on it, what would you recommend for a VoIP gateway (somthing worth the price and good, like the Adit 600) has to support FXO
16:34.53shmaltzyeah?
16:34.55*** join/#asterisk S2b31Ak (n=b4k61zx@dsl-084-060-160-011.arcor-ip.net)
16:35.03*** part/#asterisk S2b31Ak (n=b4k61zx@dsl-084-060-160-011.arcor-ip.net)
16:35.04ManxPowershmaltz,  I would NEVER recommend that.
16:35.26shmaltzManxPower, what a VoIP Gateway? why not?
16:35.35tzangerok I've got 14 active g729 translations now (to/from ulaw) with a loadavg of around 0.52, vmstat is giving me varied results
16:35.40ManxPowershmaltz, Because it's the wrong way to do it.
16:35.52tzangerbetween 37 and 88% idle CPU
16:35.53shmaltzManxPower, what is the right way?
16:35.54ManxPowerIf you just must be a lunatic and do it, then I'll be Cisco is the only option.
16:36.00shmaltztzanger, on an epia?
16:36.07ManxPowershmaltz, the right way is a T-1 and Asterisk.
16:36.11tzangermemory hasn't changed at all, 1G is fine
16:36.13ManxPowerOr a Cisco
16:36.14tzanger300M free
16:36.16shmaltzManxPower, why you have stocks in Cisco?
16:36.19tzangershmaltz: P4/2.4GHz, not HT
16:36.21ManxPowershmaltz, nope
16:36.22tzangerSIS900 ethernet
16:36.36ManxPowerI guess you could use a Ascend Max if you don't need FXO.
16:36.39tzangerso not far off from the cheapie systems (no, not epia obiously but still sub-$500 PC)
16:36.48shmaltzbut I do need FXO
16:36.56ManxPowerYou're not going to get enough FXO channels into anything except a cisco
16:37.04*** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net)
16:37.05ManxPowershmaltz, why can't you use a T-1?
16:37.21shmaltzManxPower, because I want to use it with a cheap epia mb
16:37.24tzangerthis is with the VPM disabled on the TE406 too, so it's essentially a TE405 with host CPU echo can (KB1)
16:37.35shmaltzManxPower, I know I will end up using T1 <> Channel Bank
16:37.36ManxPowershmaltz, why do you even need a PC.
16:37.44ManxPowershmaltz, then stop saying FXO
16:37.52*** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
16:37.53shmaltzit works great for me, does wonders and is almost better than a PRI ;)
16:38.15ManxPoweryou are looking for a CT1<->VoIP converter that supports G729.
16:38.36shmaltztzanger, how do you disalbe the onboard echo can?
16:38.39ManxPowerAs I said, get a Cisco or Lucent/Ascend TNT
16:38.48tzangershmaltz: comment out VPM_SUPPORT in wct4xxp.c
16:38.53ManxPowerYou're not going to find much else.
16:38.56*** part/#asterisk zoo (i=nobody@ip-158-16.travedsl.de)
16:39.01tzangerthere might be a module parameter too but I haven't checked
16:39.07shmaltztzanger, what does VPM stand for?
16:39.17tzangerloadavg of 1.03 now, 5min loadavg of 0.65
16:39.23tzangershmaltz: voice processing module
16:39.37shmaltzManxPowe, you don't even recomend Asteisk <> T1 <> Adit 600 <> FXO <> PSTN ?
16:39.44steve___tzanger why did the load shoot up?
16:39.51tzangeractually this CPU still seems fairly idle at 14 ulaw-g729 conversatoins
16:39.57tzangersteve___: I'm letting it run to stabilize
16:40.15tzangerit does spike around a bit but seems good in terms of sound quality (I'm monitoring one of the calls)
16:40.42tzangerI'm maxxed out at 15 conversions now (I left one line open so calls could still come in and out)
16:40.45shmaltzManxPower, you got my prevoius question?
16:40.51ManxPowershmaltz, not with the motherboard limitations you are imposing
16:41.25ManxPowerOh, and I can never recommend analog FXO.
16:41.27shmaltzManxPower, that I know, if I end up going T1 <> Channel bank, then I'm going with a P4
16:41.35tzangertest over... I'm happy
16:41.38shmaltzManxPower, so what do you recommend?
16:41.44*** join/#asterisk omarc55 (n=omarc55@146.82.251.141)
16:41.51ManxPowershmaltz, Your G729 requirements are the problem, as well as your analog lines
16:41.52shmaltztzanger, you meant enabling APIC in the BIOS?
16:42.03tzangershmaltz: in the linux kernel
16:42.08tzangerand using the local apic even on uniprocessor
16:42.11ManxPowershmaltz, PRI->Asterisk is what I recommend
16:42.17tzangersolves almost all IRQ problems unless the motherboard is REALLY fucked up
16:42.21shmaltzManxPower, but most customers don't have PRIs
16:42.33shmaltztzanger, thanks for the tip, will try next time
16:42.47shmaltztzanger, what do you get when you run zttest?
16:42.49ManxPowershmaltz, most of our offices don't have PRI.  If they want Asterisk they get PRI.
16:42.57shmaltzthe best I ever got was 5 9s
16:43.16shmaltzManxPower, I wish that would be the case overhere ;)
16:43.23ManxPowerMy users complain about the dialing time on analog FXO
16:43.32tzangerManxPower: that's what I'm really recommending too, but I'm happier with the TDM cards these days so it's not quite an issue
16:43.42shmaltzManxPower, well it's still faster then Rotary
16:43.47tzangershmaltz: I don't trust zttest
16:43.50ManxPower11 digits at even 250ms pere DTMF digit
16:43.54tzangerits numbers don't mean much to me
16:44.01shmaltztzanger, I know, but what is your output?
16:44.06*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
16:44.16ManxPowerso once the user is finished dialing, then they STILL have 4 seconds of dead silence
16:44.42ManxPowershmaltz, It's sounding like Asterisk just is not the solution for you.
16:45.02ManxPowerPanasonic have some decent and inexpensive PBXs
16:45.09tzangershmaltz:
16:45.09tzanger--- Results after 39 passes ---
16:45.10tzangerBest: 100.000000 -- Worst: 99.987793 -- Average: 99.999061
16:45.12tzangeron one box
16:45.22shmaltzthanks tzanger
16:45.29shmaltzManxPower, you kidding me?
16:45.36ManxPowershmaltz, not at all.
16:45.36shmaltzI'm coming from Panasonic
16:45.48*** join/#asterisk SuPrSluG (n=SuPrSluG@pool-141-149-248-22.buff.east.verizon.net)
16:45.52shmaltzI still use it sometimes, but not for what I sell now
16:45.53SuPrSluGhello
16:46.04tzangershmaltz: on the other box
16:46.04tzanger--- Results after 31 passes ---
16:46.04tzangerBest: 99.987793 -- Worst: 99.975586 -- Average: 99.987399
16:46.08ManxPowershmaltz, your requirements touch on ALL of Asterisk's primary weaknesses.
16:46.15shmaltzManxPower, when I need good door and cordless phone support I interconnect it with asteisk
16:46.23sivanatzanger: you know anyone with PRI in TO?
16:46.34tzangersivana: not right just yet
16:46.45tzangerget the DIDs from unlimitel or something
16:47.02sivanaya.. for termination.. I want like 2 or 3 DS0s
16:47.04*** join/#asterisk Koshatul (n=evangeli@ip157.net65.ipnetworks.net.au)
16:47.11sivanaShidan has a couple coming up
16:47.11tzangershmaltz: my home box
16:47.21tzanger--- Results after 39 passes ---
16:47.21tzangerBest: 100.000000 -- Worst: 99.987793 -- Average: 99.990610
16:47.33ManxPowersivana, for 2 or 3 channels, use an analog chard.
16:47.55ManxPowerUnless you mean "2 or 3 DS1s"
16:48.02sivanano channels
16:48.11ManxPoweryou want a 3 channel PRI?
16:48.15sivanabut I don't wan to colo.. just have someone give me local termination
16:48.31tzangershmaltz: customer box
16:48.32tzanger--- Results after 39 passes ---
16:48.32tzangerBest: 100.000000 -- Worst: 99.987793
16:48.37tzangerreally teh numbers are meaningless to me
16:48.49tzangerthe 99.987793 just emans it got more data in the time that it allotted
16:48.55tzangerwhich is like 1 frame here or there
16:48.58omarc55Hi. I am having trouble with a TDM400P in Ecuador, for some reason its not detecting when a caller hangs up the call and leaves the line open in queue. what could be wrong? I don't know what to check....
16:49.02shmaltztzanger, thanks I c, because thats about what I get
16:49.20tzangeromarc55: does the telco provide battery reversal or disconnect upon hangup?
16:49.43ManxPowertzanger, if they did, he would not be having this problem 8-)
16:49.52omarc55tzanger: I don't know, how can I check?
16:49.54tzangerManxPower: hahahaha
16:50.02blitzragehttp://www.ubergeek.tv/article.php?pid=54 <-- work safe
16:50.02tzangeromarc55: put a DC voltmetere across tip and ring
16:50.05tzangerget the pther side to hang up
16:50.13tzangeryou should see it go negative briefly or drop to zero briefly
16:50.24tzangerif it doesn't asterisk has no real way of detecting other than trying ot do it acoustically
16:50.42SuPrSluGwhen i call a line and it's busy ( the line has call wating) it goes directly to a mailbox and says it's full. Where would it try to save it. It never gets to the menu
16:51.04*** join/#asterisk L5q80Ta (i=rozohw@202.62.10.185)
16:51.14*** part/#asterisk L5q80Ta (i=rozohw@202.62.10.185)
16:51.16ManxPowertzanger, I wonder if we can get kb1_kenobi to fix tone hangup detection, like he did with the echocan (KB1 looks to be like 2 lines different from MARK1
16:51.41tzangerManxPower: it is... he just rearranged some stuff and fixed an initialization problem on one of the inner loops IIRC
16:51.51tzangerjust small changes made due to reviewing the implementation
16:52.00ManxPowertzanger, took only 2 years for someone to realize the problem
16:52.06*** join/#asterisk rnovotny22 (n=rnovotny@198.57.19.112)
16:52.23tzangerManxPower: digium is strapped for people I think...  it's a tough balance
16:52.30omarc55tzanger: let me see if I can do that, I don't know if I have one of those laying around. if I'd have to do it acoustically, how would I do it?
16:52.36ManxPowertzanger, echo problems are a critical problem
16:52.42tzangerbetween hriing (expensive) people or trying to do it all yourself
16:52.49tzangeromarc55: busydetect...  it's not fun.
16:53.04tzanger"tone hangup" -- what generates a tone to hang up?
16:53.22ManxPowertzanger, *shrug*  Echocan is one of the BIG THREE major issues I see with Asterisk
16:53.42tzanger*nods*
16:53.55omarc55tzanger: uuuffff.. do you know where I can read about this? I'll check out the wiki though
16:53.56ManxPowerThe other two being, tone based progress and hangup detection (callprogress and busydetect), and interrupt servicing latency (HDLC Abort)
16:54.03tzangeromarc55: the wiki :-)  I don't use it myself
16:54.19tzangerManxPower: I have not seen the HDLC abort in ages
16:54.50ManxPowertzanger, that's because you are lucky
16:55.04omarc55tzanger: thanks, I'll check this out. if anything I'll bug you again :)
16:55.08tzangerManxPower: I dunno, using a TDM400P, two TE405/6s and a T100P
16:55.22*** join/#asterisk mariogamboa (n=sudaikdd@201.138.152.159)
16:55.26mariogamboahi all
16:55.27ManxPowertzanger, I meant lucky with your motherboard choice.
16:55.55tzangerI'd like to help figure out the TDM faxing issues (drivers) and fix a bug where when you play DTMF out to Zap Zap stops sending you audio for the length of the tone (it's just masked over in chan_iax2 currently)
16:56.09blitzragehttp://www1.ca.dell.com/content/topics/segtopic.aspx/days_of_deals?c=ca&cs=CADHS1&l=en&s=dhs <-- cheap 19" LCD monitors
16:56.26tzangerManxPower: I'm using everything... MSI cheapass, Dell P2/P3 and Intel IIRC
16:56.28*** join/#asterisk Goshen (n=Goshen@67-40-107-29.slkc.qwest.net)
16:56.42ManxPowertzanger, supermicro?
16:56.48mariogamboai have a doubt i need when my user dial digit 0 i need the call goes to the operator i need to create a extension or asterisk come with this for default
16:56.53tzangerI was using supermicro (server-class system) but not anymore
16:57.01tzangerum that's 17" there blitzrage
16:57.10tzangerahh further down is 19"
16:57.20ManxPowertzanger, do some heavy IDE activity when you have your PRI active.
16:57.48tzangerManxPower: these were 100% scsi (the supermicro)
16:57.52tzangerall the others are IDE
16:58.04tzangeradmittedly ther's almost zero IDE activity on my of my * boxes
16:58.08ManxPowerLike dd if=/dev/hda of=/dev/hdb (where hdb doesn't have anything important on it)
16:58.10tzangeronly time they get hit is in logging and voicemail
16:58.14blitzragetzanger: yah - you think i'm dumb? (don't answer that)
16:58.18ManxPowertzanger, SCSI is usually not a problem
16:59.02ManxPowertzanger, or put some Promise IDE RAID cards in and watch the fun begin!
16:59.10tzangerblitzrage: :-)
16:59.23tzangerManxPower: I have promise ide raid but I *never* use their raid function
16:59.28tzangeralways linux SW raid
16:59.34ManxPowerI'm bored.  The MIS manager is out looking for a house and I have nothing to really do
16:59.54tzangerugh what's the point of 19" if it only goes to 1280x1024
16:59.56ManxPowertzanger, See, you do many things that specifically reduce or eliminate HDLC abort errors
17:00.04tzangerManxPower: nah
17:00.17*** join/#asterisk W2o48Sv (i=rozohw@202.62.10.185)
17:00.18ManxPowertzanger, I didn't say you did it deliberatly. 8-)
17:00.20tzangerManxPower: I use IDE, and I use software RAID because I don't trust the pseudo-half hw RAID
17:00.28tzangerthe controller goes, so does your data
17:00.35*** part/#asterisk W2o48Sv (i=rozohw@202.62.10.185)
17:00.39tzangerI use the cheapss motherboards
17:00.42tzangercheapass ethernet
17:01.00*** join/#asterisk kue (n=Administ@CPE-70-94-59-239.wi.res.rr.com)
17:01.12ManxPowertzanger, I've lost too much data on RAID.  The only thing I use RAID for is as a way to make partitions bigger than 1 drive, sort of like a poor mant's LVM
17:01.14tzangerthe only thing I do do is use Intel (P2, P3, Celeron, P4) -- I have had no love from AMD with Asterisk (Athlon/Duron era, not going ot waste x86-64)
17:01.27tzangerManxPower: just use LVM then :-)
17:01.29mishehuManxPower: software or hardware raid?
17:01.29jarroddoes asterisk not develop the MGCP code anymore?
17:01.40ManxPowertzanger, I suspect it's rally the AMI BIOS that causes those issues
17:01.44tzangerManxPower: I just use RAID for mirroring too, no RAID5 except on one server
17:01.59tzangerManxPower: you think?  Linux end-runs around BIOS almost every time
17:02.18ManxPowerjarrod, Digium never did significant work on MGCP, AFIK.  It was always 3rd parties/users
17:02.25*** join/#asterisk [ViRii] (n=virii@24-159-155-42.dhcp.smrt.tn.charter.com)
17:02.32ManxPowertzanger, maybe the chipset.
17:02.50ManxPowerby "BIOS" I mean "every damn chip on a motherboard except the CPU"
17:02.56tzangerManxPower: could be but again... MSI isn't known for its killer chipsets :-)
17:02.57Hmmhesayspeople in IT got no sense of humor what so ever
17:03.09jarrodmanx: i know they are not adhereing to the new specifications of sending L/osi tone codes in the DLCX and its throwing the equipment off :(
17:03.12tzangerSIS everything on my main * box
17:03.21jarrodper the RFC
17:03.23ManxPowertzanger, it doesn't have to be killer, it just has to not do stupid things that mess up interrupt latency 8-)
17:03.27jarrodi had to add it in the code.. heh
17:03.44tzangerIntel everything on my other, and whatever Dell's got on the third and 4th
17:03.55mariogamboai have a doubt i need when my user dial digit 0 i need the call goes to the operator i need to create a extension or asterisk come with this feature for default
17:03.57ManxPowertzanger, we use all Intel reference boards
17:03.58tzangerHmmhesays:  why do you say that
17:04.10ManxPowermariogamboa, you always have to cofigure everything
17:04.27tzangerManxPower: true, but the cheap systems typically do that :-)
17:04.28ManxPowerHmmhesays, users suck out the sense of humor.
17:04.47*** join/#asterisk S8a12Iu (i=rozohw@202.62.10.185)
17:04.51mishehuusers suck out the life from an it worker
17:04.55*** part/#asterisk S8a12Iu (i=rozohw@202.62.10.185)
17:05.04HmmhesaysI sent a dialplan off to a guy and left this in the bottom
17:05.05Hmmhesaysexten => s,1,Set(matt=sexykinda)
17:05.05Hmmhesaysexten => s,2,Set(matt=notsomuch)
17:05.05Hmmhesaysexten => s,3,Noop(${matt})
17:05.06mishehuI've seen leeches that left more blood behind
17:05.13sivanatzanger: any word on that TF number?
17:05.15jarrodmanx: you ever integrate cisco IADs with asterisk?
17:05.30ManxPowerIT: "I'm the IT person, what seems to be te problem?" User: "Before we begin I have to reach into your chest and rip out every bit of happiness."
17:05.36tzangersivana: that TF number?
17:05.41Hmmhesaysnot a <chuckle> laugh or anything when I said he should probably take that out before giving it to his boss
17:05.46*** join/#asterisk H2f62Aw (i=rozohw@202.62.10.185)
17:05.57*** part/#asterisk H2f62Aw (i=rozohw@202.62.10.185)
17:06.05sivanathe one missing from the consecutive order
17:06.13ManxPowerjarrod, No.  I use Cisco routers to....route packets.  Nothing else.
17:06.30tzangersivana: what about it
17:06.32*** join/#asterisk SimonR (n=SimonR@CPE001310092352-CM001371142e78.cpe.net.cable.rogers.com)
17:06.44jarrodhmm.. i am looking for decent equipment to integrate voip into legacy phone systems..
17:06.54tzangerjarrod: Adit600.  Get it, Love it.
17:06.57ManxPowerjarrod, Digium Cards and/or Channe Banks
17:07.01tzangerjarrod: AB1/2 works well if it's FXS ONLY
17:07.05Hmmhesaysjarrod: station side or trunk side?
17:07.08sivanayou're short one essentially
17:07.11tzangerjarrod: T1 into the KSU/PBX if you can at all help it
17:07.12ManxPowerAdtran Total Access 750 or 850!
17:07.14tzangersivana: yeah I know
17:07.19sivanaok
17:07.24Hmmhesaysi use quintum on trunk side integrations
17:07.31jarrodthats what im using now is the adtran total access 600 series with MGCP code
17:07.36*** part/#asterisk SimonR (n=SimonR@CPE001310092352-CM001371142e78.cpe.net.cable.rogers.com)
17:07.37ManxPowerI'm just in a rotten mood.  I want to go home.
17:07.48*** join/#asterisk Defraz (n=t0tal@tim.ibccom.net)
17:08.09ManxPowerjarrod, I believe that a channel bank that does VoIP is a crime against nature and should be banned by the UN.
17:08.22ManxPowerBut as I said, I'm in a bad mood.
17:08.45Hmmhesaysif everybody had an ocean, across the USA, then everybody'd be surfin
17:09.28Hmmhesayscome on ManxPower sing along
17:09.35*** join/#asterisk _meppl_ (n=mephisto@p54AAF457.dip.t-dialin.net)
17:09.46BeirdoManxPower: I'm with you there
17:09.50lehelit is a Canon B820.. i found this how to use: http://paste.debian.net/1829, i don't now exactly that i have to cahnge to "123" with the fax's number? or where i set the fax-number?
17:09.51Beirdowhy can't this day just end?
17:10.11*** join/#asterisk P3e73Ot (i=rozohw@202.62.10.185)
17:10.14tzangerManxPower: I can understand that
17:10.19tzangerany word on when you might be able to?
17:10.25*** part/#asterisk P3e73Ot (i=rozohw@202.62.10.185)
17:10.30ManxPowertzanger, if I can find transportation, maybe tomorrow
17:11.02ManxPowermy landlord's landline seems to be at least going to voicemail now, instead of the "due to the storm nothing works" message from the telco
17:11.17tzangerright
17:21.00*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
17:21.00*** topic/#asterisk is Asterisk 1.2.0 Beta1 - http://www.asterisk.org || Astricon 2005 - Anaheim, CA - Oct 12-14 - http://www.astricon.net/2005 - Speakers wanted! || A core dump is your computer's way of saying "Here's what's on my mind, what's on yours?"
17:21.26*** join/#asterisk Coriantum (n=asdfkle@67.41.182.243)
17:21.47doughecka_meh
17:21.55*** join/#asterisk hugo-v6 (n=hugo@195.140.232.114)
17:22.27hugo-v6puh. sometimes its a pain in the ass to join here ;)
17:22.30hugo-v6hi pals
17:22.36*** join/#asterisk Corydon-w (i=yellow@vcchgate.vcch01.springfield.tn.us.vcch.net)
17:23.36CoriantumHow do I check whether 0 or 1 is returned in an app?
17:23.36*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
17:23.39adrenalinehi hugo-v6
17:23.44*** join/#asterisk tod (n=tod@207.54.140.99)
17:23.55hugo-v6hi adrenaline :)
17:24.06*** join/#asterisk jero (n=boo@199.243.85.90)
17:24.07Ariel_hello everyone
17:24.12hugo-v6hi Ariel
17:24.38adrenalineHi Ariel_
17:25.01Ariel_came back to a freenode that was down for some reason. Also there seems to be about 40% less people on today?
17:25.25HmmhesaysAriel_ my problem was a syntax error
17:25.30tzangerAriel_: we're culling the herd
17:25.46Ariel_Hmmhesays, great to hear it. So Set does work in head
17:25.55Hmmhesaysit does
17:26.08Hmmhesaysit is probably Setvar renamed
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17:26.35*** join/#asterisk oplog3525 (n=oplog352@206.222.29.50)
17:26.37Ariel_I am actually waiting for the release due to I would like to redo allot of my macro's to the newer n, instead of the +101
17:27.06*** part/#asterisk hggh (n=jonas@mx.capi2name.de)
17:27.07HmmhesaysI'm not using any macros in this dialplan
17:27.10*** join/#asterisk Z8s94Yb (n=u7h47rz@81.215.212.27)
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17:27.16*** part/#asterisk Z8s94Yb (n=u7h47rz@81.215.212.27)
17:27.23Hmmhesaysyou can check it out if you want its kind of a beast
17:27.26Ariel_Hmmhesays, I setup a head last week and tried the setvar on it. It's still working there.
17:27.32todanybody here use realtime?
17:27.43Hmmhesaysjust gives you the "deprecated" warning on the cli?
17:27.49Ariel_I don't use realtime
17:27.53tod8-/
17:27.53*** join/#asterisk O6q93Ql (n=o3b38bf@213.145.114.92)
17:27.58jarrodi do
17:27.59CoriantumHow do I check whether 0 or 1 is returned in an app?
17:28.00jarrodit works great
17:28.10*** part/#asterisk O6q93Ql (n=o3b38bf@213.145.114.92)
17:28.11todI've got a lot working with realtime, but I've run into a snag with voicemail
17:28.21Ariel_GotoIf
17:28.37eKo1i guess nobody uses the ama flags...
17:28.44*** join/#asterisk popvoxdave (i=user@dave2.toad.net)
17:28.45Hmmhesaysyeah the n priority is very very nice
17:28.45todI'm using 1.2 beta1
17:29.05todand it seams to not want to lookup things in the voicemail table
17:29.33todif I issue a realtime update it says it failed, but there's nothing in the debug logs
17:30.08*** join/#asterisk wolfson (n=hehe@65.174.122.198)
17:30.13eKo1hmm, how the the callerid that I set in zapata.conf is not taking effect.
17:30.37todat this point I don't even know where to look for the problem
17:31.28todI know it's connecting to db ok for extensions and sip.
17:32.04*** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1)
17:33.10Ariel_tod, in the normal setup without realtime the voicemail is depended on the context you put them in.
17:33.45Ariel_tod, do you need to reference that like 200@voicemailcontext
17:33.52todyeah, I've double checked the context
17:34.05[ViRii]hey ariel_
17:34.14Ariel_[ViRii], how are you today?
17:34.23[ViRii]still stuck in the same rut dialing in but not out
17:35.17*** join/#asterisk marv[work] (n=timr@border0hsv.asterisksgi.com)
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17:36.49todjarrod: do you know how to turn on the odbc debuging so I can see what sql statements it's trying to execute?
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17:37.35phantamhey guys
17:37.47phantamgot a quick question with 1.2.0b1 if anyones got a sec
17:38.16hugo-v6phantam: ask and see if u get a answer (i dont use 1.2ebta)
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17:38.55phantamhehe
17:38.57phantamwell
17:38.58phantamfor some reason
17:39.05phantamthe odbc doesnt seem to ever connect
17:39.10adrenalineanybody polycom 301 users here?
17:39.56*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
17:40.14Sedoroxanyone here a DSL Provider?
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17:41.22todphantam: do you have any errors?
17:41.42phantammmmm
17:41.48phantamsays the driver isnt specified
17:41.54todI'm currently learing odbc realtime with 1.2b1. I have sip and extensions working, but not voicemail
17:42.05phantami wish i cud get that working
17:42.12todwhat does your odbc.ini say
17:42.13phantamthats what i want
17:42.25Bytehrm :| my other asterisk (1.2.0-b1) server is having the same problem loading all the extensions :/
17:42.41SedoroxI have a Netspeed/Looprunner Chassis... needs to be connected to a Cisco 6100 DSL switch... willing to sell cheap if anyone wants
17:42.42Byte"-- Added extension '0352' priority 1 to test" <-- I made up 0001-1000 and it's only loading 352 of them
17:43.07phantamhold checking
17:43.31cpatryByte: 1000 priority for 1 extension? huh?
17:43.33*** join/#asterisk Tili (i=Tili@202-133-67-103-dialup.sat.net.pk)
17:43.48phantamtod
17:43.52phantamcan u message me the proper config
17:44.02*** join/#asterisk T5f49Za (n=j2b5bq@85.99.177.129)
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17:44.08todwell, what are you using for the db? mysql or postgres?
17:44.09*** part/#asterisk T5f49Za (n=j2b5bq@85.99.177.129)
17:44.19phantammysql
17:44.23*** part/#asterisk pbxbart__ (n=pbxbart@p54B0237E.dip0.t-ipconnect.de)
17:44.29todok, what distro?
17:44.37CoriantumCan anyone direct me on how to set an asterisk variable in a C app?
17:44.48phantamgentoo
17:44.54*** join/#asterisk netsurfer (n=bbjunkie@i-83-67-48-18.freedom2surf.net)
17:45.06Bytecpatry: 1000 extensions
17:45.18Byte[test]
17:45.18Byteexten => 0001,1,NoOp
17:45.18Byteexten => 0002,1,NoOp
17:45.25todhrm... ok, I'm using debian, so the config might be different. I'll send you mine anyway
17:45.35*** join/#asterisk dant (n=dan@81-86-69-213.dsl.pipex.com)
17:46.22phantamk
17:46.40todyou also need odbcinst.ini
17:46.44todI'll send that now
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17:47.24TiliCoriantum: channel variable or global one?
17:47.38todphantam: let me know if those help
17:47.52phantamok
17:47.57CoriantumTili: channel... is it ast_var_value?
17:47.59phantamhow u sending?
17:48.18todI just /msged them to you
17:48.24phantamdidnt get it
17:48.33TiliCoriantum: I think channel struct holds some vars. let me check
17:48.51cpatryByte: ive more then 1000 extensions on a machine, that shouldnt be a problem.
17:49.01phantamanyone can suggest if a X100 -> cellsocket -> tdma network will work properly for calling from sip->tdma?
17:49.06Bytecpatry: are you using 1.2?
17:49.07CoriantumTili: ahh ok... thanks
17:49.21*** join/#asterisk lyndon (n=lyndon@68-235-132-27.atlsfl.adelphia.net)
17:49.23cpatrybyte: nope, head.
17:49.37*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal)
17:50.03Tilimaking progdocs
17:50.12*** join/#asterisk Manipura__ (n=chatzill@S01060011954c9c46.cg.shawcable.net)
17:50.56lyndoncan anyone answer some q's on a te110p?
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17:51.23*** join/#asterisk Cherebrum (i=jgarland@72.36.136.226)
17:51.31TiliCoriantum: I gotta go. it must be channel struct you get
17:51.32*** join/#asterisk dolson_ (n=dana@toronto-HSE-ppp4308849.sympatico.ca)
17:51.33phantamtod: u still there
17:51.53CoriantumTili: alright thanks!
17:51.56*** join/#asterisk IronHelix (n=irc@ool-45785cfe.dyn.optonline.net)
17:51.57todphantam: I'm trying to dcc them to you now
17:52.12bkw_grrr where is coppice
17:52.13bkw_hehe
17:52.40Cherebrumthis is odd.. I'm using res_odbc for my voicemail config in Asterisk 1.2.0-beta1 and when I do a "show voicemail users" it says there are none.. but when I run Voicemail(user_that_it_says_does_not_exsist
17:52.50Cherebrumthis is odd.. I'm using res_odbc for my voicemail config in Asterisk 1.2.0-beta1 and when I do a "show voicemail users" it says there are none.. but when I run Voicemail(user_that_it_says_does_not_exsist_but_is_in_my_database) it works.
17:52.53*** join/#asterisk Spacebar (n=stingray@stingr.net)
17:53.07todcherebrum: using realtime?
17:53.28todcherebrum: if so, that's the way it is supposed to work
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17:54.43Cherebrumtod: yea
17:55.04Cherebrumtod: hmm... that's not what I gathered from the wiki
17:55.43DrukenCherebrum: it's one of the realtime things, it won't show you the data with ther show cmd
17:55.52phantamlol
17:55.56phantamstill not getting anything
17:56.07Cherebrumok.. well it works anyhow..
17:56.13phantamjust email the odbc's and the res_odbc to cchancespam@gmail.com
17:56.27*** join/#asterisk ard (n=ard@2001:7b8:32d:0:20c:6eff:fe18:d11f)
17:56.40todcherebrum: from http://voip-info.org/tiki-index.php?page=Asterisk+RealTime+Voicemail "NOTE: 'show voicemail users' will tell you "There are no voicemail users currently defined".  This is CORRECT behavior. "
17:56.53todnow, I wish I had the voicemail part working for me
17:56.56Cherebrumtod: yes.. but only if you delete all the users from the voicemail.conf
17:57.15toddid you run into a problem with having it lookup voicemail in the db?
17:57.18TiliCoriantum: is this available in API pbx_builtin_setvar_helper
17:57.47todI have realtime working for extensions and sip, but it doesn't want to work for voicemail and I'm stuck
17:57.52Cherebrumtod: it seemed to me that it was "CORRECT behavior" because the database was empty at that point
17:58.09Cherebrumtod: I can send you my mysqldump
17:58.28todI don't think it's that. I don't think it's looking in the db at all
17:58.41todbecause I get an error when doing a realtime update
17:58.56CoriantumTili: I'm trying that now
17:58.57todbut I can't find what sql statements it's trying to execute
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17:59.15Cherebrumtod: Are you using mysql?
17:59.26todif I do a realtime load, it doesn't find the rows I know are there.
17:59.32todI'm using odbc->mysql
18:00.21opus_it isn't in /var/log/asterisk/full?
18:00.41Cherebrumtod: ok.. I'm e-mailing it now
18:01.01todwell, I don't have full turned on, but it's not in /var/log/asterisk/debug
18:01.12*** join/#asterisk Johnsie (n=john@24.154.91.33)
18:01.24todwhat email are you using?
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18:02.08FuriousGeorgeHI all
18:03.11TiliCoriantum: ok i am leaving. just look as how Set or SetVar work. i think there should be some ast_set_var like thing available. my slow linux machine is still generating docs and i cant search in api.
18:03.50*** join/#asterisk drumkilla_laptop (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
18:03.50*** mode/#asterisk [+o drumkilla_laptop] by ChanServ
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18:03.58Cherebrumtod: you get that?
18:04.31phantamtod: what do u use as u're connect string in extconfig for odbc?
18:04.36phantamodbc as driver or mysql
18:04.47*** join/#asterisk parky (n=kvirc@80.131.225.70)
18:05.06CoriantumTili: alrighty... thanks much
18:05.18Hmmhesayshmm can I make the noop's show up on the cli and nothing else?
18:07.24Cherebrumok.. so now that I have odbc setup on my server.. The pimple faced windows programmer my boss highered can connect to it now?
18:08.12steve___highered that's awesome
18:08.14steve___:)
18:08.33*** join/#asterisk FuzzyCat (n=ScaredyC@84.119.131.232)
18:10.08MicC_wow this chan is busy
18:10.18MicC_looking to do pin collection (enter your account #) with IVR
18:10.21MicC_problem is I have 50 customers. Any ideas on how to make it easy on me?
18:10.43MicC_I asked this b4 but got disconnected from VNC (no IRC at work)
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18:13.58HmmhesaysMicC: that isn't real hard
18:14.19Hmmhesaysyou can use cmd authenticate
18:14.54*** join/#asterisk dolson (n=dana@toronto-HSE-ppp4308380.sympatico.ca)
18:14.57adrenalineanybody use Polycom 301?
18:15.24Cherebrumadrenaline: for paper weights
18:18.35adrenalinehaha
18:18.51todcherebrum: voicemail => odbc,asterisk,voicemail
18:19.00adrenalinedamn I have an annoying beep at a client and I have no way of getting rid of it
18:19.31wunderkinprobably voicemail indicator
18:19.35Cherebrumtod: yea
18:19.47Cherebrumtod: you can actually just use voicemail => odbc,asterisk
18:19.59Cherebrumtod: it assumes that the table is "voicemail" because that is the application
18:20.19todyeah, but I've been debugging this and so decided to be verbose
18:20.22ManxPowerlooks like I'm going camping
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18:21.04Cherebrumtod: I turned on loggin on my mysql server
18:21.23Cherebrumadd     log             = /var/log/mysql/mysql.log
18:21.28Cherebrumto your my.cnf file
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18:24.43todok, I think I figured it out
18:24.49Fddayanif I make a change to iax.conf I have to restart asterisk ?
18:25.05Fddayanplease :)
18:25.08wunderkinreload chan_iax2.so
18:25.09Cherebrumtod: what was wrong?
18:25.17todI'm an idiot and had a typo in the extconfig.conf
18:25.35todI had the odbc as asterisk, not ast_cnf
18:25.39Fddayanwunderkin: that was for me  ?
18:25.44wunderkinyes
18:25.46todbecause I copied/pasted from the website
18:26.09Fddayanhow I reload it :)
18:26.20wunderkinthats what you type at the cli
18:26.24todheh, thanks
18:26.57todnow to move on to queues
18:27.21doughecka_bah
18:27.31todoh
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18:28.00todwhile I have someone else that's played with realtime. Is there a way to do includes in the db?
18:28.21todI had to put them in the extensions.ael
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18:29.57doughecka_heres a problem, some calls on the T1 dont get answered when I call out, although I am talking to the person (or if I call my cell phone) sometimes it will answer sometimes it wont, when it doesnt answer, it will hangup after 3 min
18:30.27MicC_hmmm: thanks
18:30.40MicC_I am having a hard time getting documentation on the IVR stuff
18:30.45ManxPowerdoughecka, the only time I've had that happen is when the carrier forgot to configure one of the channels
18:31.07simongIs there a problem with http://www.asterisk.org and the ftp and cvs servers ? Getting connection refused from Sweden.
18:31.18doughecka_so although the call went through, it dosnt know to send an answered up to me?
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18:31.36ManxPowerdoughecka, so you can talk for the first 3 mins?
18:31.40doughecka_yea
18:31.43ManxPowerincoming, or outgoing?
18:31.45ManxPowerThat's WEIRD.
18:31.46doughecka_outgoing
18:31.51doughecka_this is a pri as well
18:31.57doughecka_do I need callprogress=yes?
18:32.05doughecka_also this is CVS with aggresive on
18:32.08FuzzyCatno
18:32.26doughecka_and I think it started when I turned aggressive on
18:32.27FuzzyCatit's 6 points
18:32.38FuzzyCatooppps - wrong winder
18:32.44doughecka_why would aggressive cause it to not register it as a no answer
18:32.46ManxPowerdoughecka, NEVER have call progress or busydetect on a pri
18:32.55doughecka_ok
18:33.31doughecka_would echocan cause that?
18:33.33*** join/#asterisk popvoxdave (i=user@dave2.toad.net)
18:34.42*** join/#asterisk CdtDelta (n=CdtDelta@dsl081-225-161.chi1.dsl.speakeasy.net)
18:36.10*** join/#asterisk TheLaw (i=law@134.76.248.101)
18:39.15Fddayanthere is a way to see actives call in asterisk ?
18:40.32cpatryFddayan: type help and then type show channels
18:41.28simongAnyone have any luck connecting to www.asterisk.org at the moment ?
18:41.47doughecka_nope
18:41.48doughecka_I cant
18:41.50*** join/#asterisk huehner (n=huehner@kt-gmbh.de)
18:44.26clyrradthe site is down
18:45.05*** join/#asterisk klasstek (n=nunyobiz@sta-206-168-218-206.rockynet.com)
18:46.43Fddayanif I reload asterisk I lost the actives connections
18:47.02klasstekis cvs down?
18:47.09klasstekcvs [status aborted]: connect to cvs.digium.com(216.27.40.102):2401 failed: Connection refused
18:47.51*** join/#asterisk rabelais (n=blank@ca-stmnca-cuda4-gen2c1-171.vnnyca.adelphia.net)
18:47.55DannyFklasstek, worked 10 secs ago for me
18:48.09DannyFretry a few times
18:48.46klasstek15th attempt worked.
18:49.01*** join/#asterisk lathos42 (n=lathos42@65-42-27-66.dowdingindustries.com)
18:49.12lathos42Hello
18:49.55gnarf37does anyone know of a way to detect if a call has been transfered after using Dial() to another IAX?
18:50.17gnarf37basically I want to be able to generate a CDR for the incoming and the outgoing line of the trunk
18:53.43bkw_its called FORK CDR
18:53.53bkw_you create two records for each call.. :P
18:54.26*** join/#asterisk darkskiez (n=darkskie@host86-132-169-66.range86-132.btcentralplus.com)
18:55.00simongftp cvs and http to asterisk.org appear to be refusing connections.
18:55.16wunderkinyes simong
18:55.42bkw_bkw.digiweb.com is a CVS mirror
18:55.43bkw_works fine
18:55.53simongAh good :)
18:56.49*** join/#asterisk Ayano (n=erik_lee@adsl-70-245-190-90.dsl.spfdmo.swbell.net)
18:57.27*** part/#asterisk Fddayan (n=fddayan@66.240.80.130)
18:57.51*** join/#asterisk patrick^ (n=patrick_@pc-0-34.mountaincable.net)
18:57.57*** part/#asterisk popvoxdave (i=user@dave2.toad.net)
18:59.51*** join/#asterisk mmmToop (n=chatzill@c4-37-2.rba.dial-up.net)
19:00.32AyanoHey, what company does SwK work for?
19:00.47Damin_kram: Howdy..
19:00.55kramhi damin what's up
19:01.10Damin_bkw_: Except for all the security backdoors you put into your CVS tree! :)
19:01.14mishehubah.
19:01.27mishehudamin, kram, bkw, and other folks
19:02.00file[muon]eh?
19:02.18file[muon]Ayano: Asteria
19:02.22Ayanoty
19:02.23bkw_haha
19:02.29mishehudd if=/dev/urandom of=file[muon] bs=1024 count=1024
19:03.03*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
19:03.19file[muon]....:p
19:03.20*** join/#asterisk amir (n=amir@shield.guindehi.ch)
19:03.58kramhi file
19:04.55*** join/#asterisk phantam (n=cchance@72.252.15.246)
19:04.56phantamguys
19:04.57phantamok
19:05.01phantammy odbc says its connected
19:05.13phantambut im dont seem to receieve updates or load the peers from the server
19:05.58phantamand i get this if i do sip show settings
19:05.58phantamSep  7 15:10:07 WARNING[20556]: config.c:893 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available
19:06.39*** join/#asterisk DeeJayTwo (n=deejay2@215-238.sh.cgocable.ca)
19:07.05bkw_phantam, I seen that today also
19:07.15DeeJayTwohi!
19:07.19bkw_someone mucked up realtime :P
19:07.19mishehu*cough hack*
19:07.27bkw_someone stole my realtime engine
19:07.30mishehubkw_: yey
19:07.44phantam?
19:07.48DeeJayTwoWhen an asterisk system dial with iax to another asterisk box, the lag makes a ring tone before the dial tone..
19:07.58bkw_Sep  7 15:07:16 WARNING[16596]: pbx.c:791 pbx_find_extension: Maximum PBX stack exceeded
19:07.58file[muon]bkw_: consider yourself stolen
19:08.00bkw_LOVERLY
19:08.01mishehuDarth Maul
19:08.04DeeJayTwois it possible to avoid any ring tone before it's really ringing the final peer?
19:08.22phantami just want it to work
19:08.22bkw_DeeJayTwo, what devices are involved?
19:08.49bkw_I'm quickly becoming a sip guru
19:09.15file[muon]SIP IT TO ME BABY!!!
19:09.17DeeJayTwophone<-->Channel bank<-->T1(zap)<-->asterisk<-->internet<-->asterisk<-->pri to phone provider
19:09.48DeeJayTwothe first asterisk server is only a gateway which sends the client to the last asterisk (over IP)
19:11.09*** join/#asterisk Zaw (i=zaw@unaffiliated/zaw)
19:11.10DeeJayTwoby the time the first asterisk system dial the other... the client hear a ringing tone..
19:11.11*** join/#asterisk wolfson` (n=hehe@65.174.122.198)
19:11.27phantamok
19:11.29phantamfigured that out
19:11.31phantami had to install addons
19:11.33phantambut now what does Sep  7 15:15:33 ERROR[32744]: res_config_mysql.c:605 mysql_reconnect: MySQL RealTime: Failed to connect database server  on . Check debug for more info.
19:11.34phantammean
19:12.31bkw_DeeJayTwo, this SIP?
19:12.36DeeJayTwoiax
19:12.40*** join/#asterisk Keenan (n=Brocker@81.215.7.69)
19:13.02phantam?
19:13.04DeeJayTwosuppose there's a little lag between 2 asterisk server...
19:13.15DeeJayTwothe dial command will generate a ring tone until the connection is done...
19:13.21*** join/#asterisk r0d3nt (i=anonymou@tinfoilhat.net)
19:13.22DeeJayTwohow is it possible to avoid it?
19:13.34mishehuhmm...  is it just my systems or is something borked on the asterisk ftp?
19:13.41bkw_ya know I worked till 1am lastnight
19:13.44mishehuList failed
19:13.47bkw_and I actually enjoyed every second of it
19:13.56*** join/#asterisk L|NUX (n=linux@202.5.145.14)
19:14.11mishehubkw_: you must be ill.  ;-)
19:14.20mishehupeople don't like to work unless there's something wrong with them
19:14.31bkw_I must be ill
19:14.38bkw_i'm still enjoying this.. muhahahahahahahahahahah
19:14.42Beirdoheh
19:15.27*** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com)
19:15.44FuzzyCatoh crap...
19:16.42doughecka_I am not getting did from my pri
19:16.51doughecka_thus I cannot handle incoming calls
19:17.06*** join/#asterisk jeremywhiting (n=jeremy@71-37-101-103.slkc.qwest.net)
19:17.47*** join/#asterisk Baph (n=Dave@dirobertson.plus.com)
19:18.03doughecka_this is crap
19:18.09doughecka_<PROTECTED>
19:18.20doughecka_that tells me its not passing dids through
19:19.15*** join/#asterisk shmaltz (n=chatzill@69.28.255.210)
19:19.27phantamanyone know what res_config_mysql.c: MySQL RealTime: Failed to connect database server  on . Check debug for more info.
19:19.28phantammeans?
19:19.37bkw_doughecka it tells you the telco is fux0red up
19:19.58doughecka_bkw_: sure its not my end? I have turned on mark2 and agressive echo can
19:19.59*** join/#asterisk kshumard_home (n=ksh@pcp08979908pcs.huntsv01.al.comcast.net)
19:20.12*** join/#asterisk vooduhal (n=cmcbee@64-18-111-165.static.catt.com)
19:20.16doughecka_its bellsouth
19:20.47shmaltzanybody have any clue why a wct4xxp which has span 1 connected to telco PRI, span 2 to avaya, and span 3 to Adit 600, that works perfectly. Just stops working with just giving me 3 blinking red leds on the card, and only unplugging and waiting a minute helps?
19:20.58vooduhalQuick question.  Is "One Touch Record" in the latest stable?
19:21.22jeremywhitinghi all
19:21.50doughecka_bkw_: can I yell at the telco?
19:21.51doughecka_:P
19:22.04jeremywhitinganyone know if I can allow someone access to my asterisk command line without giving them root access? maybe an asterisk user account, e.g.
19:22.11bkw_actually it takes putting on a HUGE boot and sticking in the telco's ass to get stuff working right
19:22.17doughecka_HAHAHA
19:22.28bkw_I am not kidding really
19:22.31WonkaBoots of Ass Kicking
19:22.42*** join/#asterisk Komasathit (i=chonlada@jane.lru.ac.th)
19:22.42bkw_you must really kick telco's asses to get them to no be fucktards
19:22.45*** join/#asterisk |[diLa]| (n=asi_girl@83.228.23.1)
19:23.04phantamwow
19:23.10phantamlet me guess i hit a bug no one knows about
19:23.10phantamlol
19:23.17*** join/#asterisk DaPrivateer (i=Privatee@CRIMSON.OFF-HOURS.COM)
19:23.37bkw_phantam, nobody cares about res_config_mysql
19:23.49ByteBreakpoint 3, ast_category_browse (config=0x812b3d0, prev=0x812b1c8 "\u0433\022\b\uffff\uffff\022\b\uffff\uffff\022\b\u0433\022\b") at config.c:253
19:23.54Bytethat looks broken :/
19:24.09bkw_is this in gdb?
19:24.10*** join/#asterisk marrandy (n=marrandy@209.216.76.1)
19:24.13Byteyes
19:24.16bkw_print $config
19:24.18marrandyHello all :-)
19:24.26Byte$2 = void
19:24.29bkw_print *config
19:24.39Byte$4 = {root = 0x656e6567, last = 0x6c6172, current = 0x0, last_browse = 0x0, include_level = 0, max_include_level = 0}
19:24.48bkw_print *prev
19:24.55Byte$5 = -48 '\uffff'
19:24.57phantamlol
19:25.00bkw_oh thats good
19:25.01phantamlol
19:25.05phantamthen how do i get realtime working
19:25.12bkw_phantam, use res_config_odbc
19:25.16marrandyanyone have a sip - aastra 9133i phone ?
19:25.22phantami have that set
19:25.25phantambut wasnt working
19:25.35bkw_it works fine you just didn't get it setup right :P
19:25.36marrandyphantam: working with asterisk ?
19:25.36Bytethis is 1.2.0-beta1, I tried and didn't seem to find a bug for my problem
19:25.38*** join/#asterisk FuriousGeorge (n=ads@pool-151-198-123-165.nwrk.east.verizon.net)
19:25.51FuriousGeorgehi all
19:25.53bkw_Byte, update to CVS-HEAD which will be the next beta
19:25.54bkw_and try again
19:26.00phantamlol
19:26.08Byte#84 0x0805b106 in __load_resource (resource_name=0x812c270 "\uffff$\uffff\uffff\uffff\"\uffff\uffff\uffff(\uffff\uffffp(\uffff\uffff\020)\uffff\uffff\200$\uffff\uffff\uffff\uffff\022\bpbx_dundi.so", cfg=0xb7b924ea) at loader.c:402
19:26.12Bytethat looks interesting too heh
19:26.18bkw_Byte, look in modules.conf
19:26.25bkw_do you have any res_ modules with load directives?
19:26.31phantamok
19:26.54phantamodbc show says connected
19:27.02Byteload => chan_modem.so load => res_musiconhold.so load => res_crypto.so load => pbx_dundi.so
19:27.10Wonkawhile you're at it... could you kick mine _not_ to want 2EUR50 a month for CLIP?
19:27.14marrandyLet me try again - anyone have a sip - aastra 9133i phone ?
19:27.14*** join/#asterisk fafnir (n=hello@tdds-gw.Moscow.gldn.net)
19:27.42shmaltzbkw_, if you want a telco to really help you, the you got to social engineer the phone number to the switch from them
19:27.59Bytebkw_: what's the cvs repository? (can't connect to www.asterisk.org)
19:28.07FuriousGeorgeshmaltz, how goes it
19:28.24doughecka_shmaltz: how? I got them on the phone now :P
19:28.26shmaltzFuriousGeorge, gr8, by you?
19:28.30phantamok
19:28.31FuriousGeorgenot bad
19:28.32*** join/#asterisk Hyper_Eye (n=mwoodj@24.96.145.218)
19:28.33phantambut
19:28.37phantamit doesnt load from the DB the peers
19:28.39shmaltzdoughecka, whos the telco?
19:28.42*** join/#asterisk razu_ (n=razu@ip192.cab63.mus.starman.ee)
19:28.43phantamits still showing the sip from locally
19:28.44phantaminstead
19:28.51mishehuasterisk ftp is also borked.
19:29.00doughecka_bellsouth
19:29.06phantambkw_?
19:29.33bkw_Byte, cvs.digium.com bkw.digiweb.com is a mirror also
19:29.39shmaltzdoughecka, ask that lady that you want to speak directly with the engineers, or techs or whatever they call them.
19:29.49shmaltzbut bellsouth is too strict
19:29.57shmaltzwith the smaller ones it works
19:30.13phantambkw_ odbc says its connected... sip show settings... shows  the realtime settings
19:30.15doughecka_I did once talk to the guy directly :)
19:30.21phantambut the peers is still showing the local stuff
19:30.25bkw_phantam, does isql work?
19:30.26shmaltzthen ask him for his number
19:30.35doughecka_it was then I relized they turned the frickin pri off to keep the error messages off thier screens
19:30.41shmaltzwas a stupid union worker or was he a good technician?
19:30.54*** join/#asterisk mogorman (n=mogorman@digium.com)
19:31.01doughecka_hmm, smart
19:31.29shmaltzso, he got it back up and running after about 30 seconds of typing?
19:31.39shmaltzwas it a he or she?
19:31.43doughecka_yea
19:31.43doughecka_and he
19:31.56doughecka_who disables the frickin pri?
19:32.00*** part/#asterisk Chonlada (i=chonlada@jane.lru.ac.th)
19:32.10phantamummm
19:32.20shmaltzwell, if it would have been a she, you could ask her out, if she refuses ask for her number, but then SHE would *never* give it to you
19:32.46shmaltzbut now that it is a he, you just tell him you got a girl for him, and just ask for his number :)
19:32.48doughecka_lol
19:33.16doughecka_hmm, /me looks for that number....
19:33.23phantambkw_: yes
19:33.28shmaltzanyhow, social engineering is beyond this channel try irc://irc.2600.net/#2600
19:33.32phantamsays connected....
19:33.45phantamsql-statement        help [tablename]         quit
19:33.58doughecka_lol
19:34.00phantamya it connceted and is working
19:34.00phantamlol
19:34.09phantamshows tables etc
19:34.20*** join/#asterisk negativeview (n=Daniel@border0hsv.asterisksgi.com)
19:34.40phantambkw can i message u real quick?
19:35.07file[muon]he's outside right now
19:35.11file[muon]please leave a message at the beep
19:35.13file[muon]*BEEP*
19:35.15phantamlol
19:35.30doughecka_doing what?
19:36.03Ariel_Damm is there not any good 4 to 8 FXO Sip box that works with asterisk. That is not an arm and leg in cost?
19:36.14file[muon]he was watering the plants
19:36.17doughecka_the lady says translations department might not have done what they are supposed to have done... who are they?
19:36.29doughecka_file[muon]: with?
19:37.05ManxPowerDoes anyone know if you can do all station page by dialing an extension on a nortel (MICS, I think)
19:37.35file[muon]water!
19:37.48HmmhesaysAriel_ what kind of price you looking for?
19:38.15Ariel_about 50 dollars per port max
19:38.39Hmmhesaysyeah that probably isn't going to happen
19:38.49Ariel_Wellgates suck really bad, Mediatrix are very hard to configure and have callerID and dtmf issues.
19:39.08Ariel_besides expensive
19:39.10HmmhesaysAriel_ you got the v5 firmware load for a 1204?
19:39.12*** join/#asterisk Kue (n=Kue@CPE-70-94-59-239.wi.res.rr.com)
19:39.33Ariel_yes my customer has it.
19:39.36*** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn)
19:39.42Hmmhesaysbetter than it used to be, but still not great
19:39.54Ariel_But they send the calls to the default context because the refuse to register.
19:40.05HmmhesaysAriel_ you can change that in your dialplan
19:40.07Ariel_now outbound calls work great
19:40.17Ariel_Hmmhesays, yes I already have
19:40.25Ariel_but still there a mess
19:40.32Hmmhesaysyou just set up a peer with a static host IP
19:40.39Hmmhesayss/peer/friend
19:41.22*** join/#asterisk pauldy (n=pauldy@c-67-166-178-129.hsd1.tx.comcast.net)
19:41.38Ariel_Hmmhesays, making calls out is not a problem. it's the inbound
19:41.47Hmmhesaysyeah Ariel_ that is what i'm talking aobut
19:41.48Hmmhesays*about
19:41.57*** join/#asterisk Kue (n=Kue@CPE-70-94-59-239.wi.res.rr.com)
19:42.12Ariel_peer is for outbound not inbound
19:42.22Hmmhesaysyes, i meant friend
19:42.30Hmmhesaysor user
19:42.31*** join/#asterisk Kue (n=Kue@CPE-70-94-59-239.wi.res.rr.com)
19:42.32Hmmhesayseither way
19:42.37Ariel_Hmmhesays, got that already still was a problem.
19:42.43Hmmhesayshow so?
19:42.46darkskiezpeer can have inbound too, that confused me for ages
19:42.52Ariel_dtmf and callerID
19:42.55phantamok so isql works... but realtime not importing the peers only showing the locals
19:43.04Hmmhesayscallerid has to be bell compliant
19:43.26jarrodhmm
19:43.32Hmmhesaysotherwise they don't like it
19:43.47Ariel_yes got that one.
19:43.51Hmmhesaysanyway, mediatrix would have done well never to have made the 1204, because their fxs devices are very good
19:44.00phantamany one :( help darn realtime
19:44.00phantamlol
19:44.28Hmmhesaysgot a 24 port fxs gateway sitting next to me thats going into an apartment next week
19:44.29Ariel_Hmmhesays, I wonder if there is a market for a 50 dollars per port fxo setup.
19:44.38Hmmhesaysanother 3 going into a hotel later this month
19:45.04Ariel_Hmmhesays, nice great to hear at least there fxs units work.
19:45.08HmmhesaysAriel_ : i'm sure there is, but you need decent hardware to be reliable and compatable with the varying pstn lines
19:45.19Hmmhesaysthey work, and they work very well
19:45.48hardwirehttp://happy9.bogomip.com/Storage/screenshot_twinview_1920x1200/photo_view
19:45.48hardwireyay
19:45.49*** join/#asterisk clive- (n=pirch@ndn-165-154-117.telkomadsl.co.za)
19:45.51doughecka_gaarbage
19:45.58phantamdid bkw leave?
19:46.11hugo-v6Hmmhesays: for analog phones?
19:46.18doughecka_what would cause did's not to be passed through back to me from a pri
19:46.24*** join/#asterisk Kue (n=Kue@CPE-70-94-59-239.wi.res.rr.com)
19:46.33Hmmhesayshugo-v6 yeah
19:46.35hugo-v6isnt it cheaper and better to manage if u use ip-phones?
19:46.39Hmmhesayshell no
19:47.36hugo-v6Hmmhesays: but the phones cant do things like use 2 lines at the same time nor special funktions like used as dooropener or such shit?
19:47.43Hmmhesayswhy would I want to change the wiring in an apartment when I can punch down a 24 port gateway in the phone room
19:48.10hardwireok.. anybody seen a satellite speaker/mic setup for phones that might not have a speakerphone
19:48.13hugo-v6ok. thats a reason
19:48.18todhmmhesays: are you using the quintum equipment?
19:48.32Hmmhesaysno mediatrix for fxs
19:48.34Hmmhesays1124's
19:48.51Hmmhesaysyou still get transfer, hold, 3 way calling accessed with the flash hook button
19:49.04darkskiezis anyone here netplex on voip-info?
19:49.13todcool, we're looking to do something simiar and we were looking at quintum and carrier access
19:49.26Hmmhesaystod, what is your setup?
19:49.37Hmmhesaysi know more about quintum than I ever wanted to
19:49.52hugo-v6Hmmhesays: find no link for mediatrix
19:49.54*** join/#asterisk zippp (n=zip@68-118-126-104.dhcp.sprn.tx.charter.com)
19:50.08phantamtod
19:50.17*** join/#asterisk Guggemand (i=irc@80.198.131.49)
19:50.18todyeah?
19:50.21phantammsg
19:50.21Hmmhesayshttp://www.mediatrix.com/products_devices.php?prodid=3
19:50.23hugo-v6.com. i should buy new glasses
19:50.30zipppdoes anyone have one of the iareaphone IAX2 phones (PA168)?  Mine came with sip firmware and when I load the IAX firmware I still get sip settings in web config
19:50.57HmmhesaysI do however use quintum when I need remote t1/e1 gateways, they give me little trouble
19:51.01*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-241-108.buckeyecom.net)
19:52.02hugo-v6Hmmhesays: whats the price for that fxs-bank?
19:52.07phantamtod: ok the darn odbc says connected
19:52.13phantamsip show settings shows the realtime at bottom
19:52.18phantambut i still only see the local sip.conf crap
19:52.33clive-zippp just change the settings to what you want
19:52.40Hmmhesays2200 I think
19:52.58darkskiezHmmhesays: what do they set ya back
19:53.16Hmmhesayswhich hardware darkskiez?
19:53.29darkskiezmediatrax 1124
19:53.38bkw_FEMA to give victims debit cards worth $2,000
19:53.43bkw_ABOUT GOD DAMN TIME
19:53.56*** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de)
19:54.01hugo-v6bkw: what?
19:54.07Hmmhesays$2400 for the 1124
19:54.08*** join/#asterisk Kue (n=Kue@CPE-70-94-59-239.wi.res.rr.com)
19:54.09bkw_it isn't much for someone that has LOST everything but shit its helpful
19:54.19doughecka_arg
19:54.28doughecka_bkw_: can I force asterisk to recieve an empty did?
19:54.28ManxPowerI could use 2k right now
19:54.36Hmmhesays100 bucks a channel, but they are worth it
19:54.38todhmmh: I don't really have a setup yet. We're looking at deploying at universities so we'll need as high of port density as posible
19:54.48bkw_ManxPower, i'm sure ;)
19:54.58Uther_Pheh, now all the drug dealers in Texas are gonna have to take debit?
19:54.58Hmmhesaysi've deployed these in universities
19:55.03*** join/#asterisk Kue (n=Kue@CPE-70-94-59-239.wi.res.rr.com)
19:55.04*** join/#asterisk apardo (n=apardo@23.Red-83-44-179.pooles.rima-tde.net)
19:55.10zipppclive
19:55.17zipppwhat you mean change the settings
19:55.25zipppit says sip settings, no IAX settings
19:55.30Hmmhesaysgot 5 of them driving 100 dorm rooms at least
19:55.37*** join/#asterisk kg (n=kg@chello062179062077.chello.pl)
19:56.00*** join/#asterisk DrukenHME (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com)
19:56.03phantamtook fema long enough
19:56.17phantamits funny i love all the internationalsupport the US is getting
19:56.29phantambut if the US is 3rd instead of first to help a country when there in need its a huge issue
19:56.39opus_federal emergancy money-wasting agency
19:56.46Uther_Pthe US got plenty of offers to help
19:56.49phantamlol
19:56.50darkskiezthe us didnt ask for help for like 5 days or something
19:56.54Uther_Pthe US was turning it down
19:57.05phantambkw: isql works and odbc show says connected but still only seeing the crap i had in sip.conf
19:57.12Uther_Pone of them was Russia though, heh.. but they offered to send troops
19:57.13Ariel_phantam, yes your correct. Did you know that the mayor had a plan done 5 years ago to evacuate everyone but forgot to use it?
19:57.19clive-zippp the names of teh fields may remain the same, but insert your iax settings into the fileds and see what it does
19:57.23phantamlol forgot
19:57.27bkw_phantam, sipfriends? or do you have sippeers,sipusers?
19:57.32sivanadoes IAX have incominglimit?
19:57.35phantamsippeers and sipusers
19:57.37Uther_PAriel_: that wasn't the mayor.. fema had a plan implimented, but didn't use it
19:57.58Uther_Pthe mayor was pissed that fema didn't use it
19:58.06phantamextensions => odbc,MySQL-Asterisk-SXM,extensions_conf
19:58.06phantamsipusers => odbc,MySQL-Asterisk-SXM,sip_conf
19:58.06phantamsippeers => odbc,MySQL-Asterisk-SXM,sip_conf
19:58.21mishehufema schmema.  the head of which last was the head of the international arabian horse association
19:58.25phantamthats about the extent of my extconfig.conf
19:58.36ManxPowerHeck, I could use a drug dealer too!
19:58.54Uther_PI saw a mob of people from Louisiana raid and pillage a Wal-Mart by my home yesterday in Denton, TX
19:59.01hugo-v6ManxPower: same here :>
19:59.06*** join/#asterisk VM (n=maxgluck@200.109.166.83)
19:59.08Uther_Pthe chaos has already spread this far, heh
19:59.34zipppclive, I am sure dtmf settings in IAX doesn't have a sip info setting, it is the same firmware as the sip one
19:59.35Uther_Pbut its funny... Rick Perry is shipping people OUT of texas now... we now have our Southern AND Eastern borders closed
19:59.39*** join/#asterisk Kue (n=Kue@CPE-70-94-59-239.wi.res.rr.com)
19:59.40zipppI checked the md5sum
19:59.48zipppi just need to know where to get the iax firmware
20:00.00JerJerdontwarn = yes | no; Don't over-inform the Asterisk sysadm, he's a guru
20:00.11phantam:(
20:00.15JerJersweet jesus almighty
20:00.23hugo-v6:]
20:00.26*** join/#asterisk Cresl1n (n=Cresl1n@digium.com)
20:00.29bkw_JerJer I did that option to shut the damn thing up about _. and various other stupid warnings
20:00.46JerJeryeah
20:01.00VMhi, I need help to configure the sendvoicemail option. I put sendvoicemail=yes and I can hear the option in the VM menu, but after I enter the extension, nothing happens. has anyone seen this before?
20:01.09bkw_I got sick of getting told _. isn't recommended .. and I know how to use it.. so it was just pissing me off :P
20:01.21clive-zipp try aredfox.net  (I think)
20:01.23phantambkw isnt that ^ there correcT?
20:01.59zipppclive-, thx
20:03.29*** join/#asterisk malcolmd (n=malcolmd@digium.com)
20:04.02ManxPowerI wish it was legal to kill COEs
20:04.04*** join/#asterisk drumkilla_laptop (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
20:04.04*** mode/#asterisk [+o drumkilla_laptop] by ChanServ
20:04.05ManxPower..er.CEOs
20:04.05Cresl1nmalcolmd!!!!
20:04.13zipppclive-, http://www.aredfox.com/edownloadsiax2.htm (which to get?)
20:04.34shmaltzhow many channels on an E1? 30 or 32?
20:04.46Cresl1nshmaltz: technically 32
20:04.50ManxPowerYou should not use _. unless you really know what you are doing.
20:05.02Cresl1nshmaltz: you have 32 timeslots at least
20:05.10clive-shmaltz 30 irl
20:05.15ManxPowerIt can cause cancer, sterility, and insanity if not used right.
20:05.17shmaltzCres|1n, so you get 32 channels and 2 are for dchans?
20:05.26Cresl1nshmaltz: no, 1 you can't use
20:05.27shmaltzwith a channel bank, how many?
20:05.35shmaltzwhy can't I use 1?
20:05.37Cresl1nshmaltz: the other is for your dchan
20:05.56Cresl1nshmaltz: it's not "accesible" to the user
20:06.02Cresl1nit's used for other things
20:06.09shmaltzCres|1n, like what?
20:06.18clive-zippp choose which one corresponds to you rphone
20:06.18*** join/#asterisk aminorex (n=tony@12-23-137-226.dhcp.dlth.mn.charter.com)
20:06.19Cresl1nIIRC, something to do with signaling
20:06.21Bytebkw_: Breakpoint 1, ast_category_browse (config=0x812b3e0, prev=0x812b1d8 "\uffff\uffff\022\b\uffff\uffff\022\b\uffff\uffff\022\b\uffff\uffff\022\b") at config.c:252
20:06.27shmaltzso with a channel bank I can get 31 then?
20:06.28Bytestill doing it even with CVS
20:06.33bkw_Byte, wtf are you doing to your asterisk install?
20:06.34bkw_hehe
20:06.37Bytenothing
20:06.46Cresl1nshmaltz: good question, I have never seen an E1 channel bank before
20:06.47Byteall I've done is add 1000 extensions to a context called 'test'
20:06.57Cresl1nshmaltz: (not in real life at least)
20:06.58*** join/#asterisk newmember (n=newmembe@dsl-lkbn-66-18-211-34-cgy.nucleus.com)
20:06.58*** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net)
20:06.58Cresl1n:-)
20:07.07shmaltzCres|1n, Adit 600 is an E1 channel bank
20:07.07zipppclive-, no numbers on it
20:07.09zipppno idea
20:07.33clive-zipp who didi you buy it from
20:07.39Cresl1nshmaltz: I don't doubt they exist, I just have never personally configured one
20:07.47mishehubah.
20:08.08Byteactually it might be loading the old modules for some reason
20:08.20Bytebah it is
20:08.27*** join/#asterisk PupenoL (n=pupeno@200.123.183.89)
20:08.28zipppclive-, iareaphone
20:08.32zipppthe sip version is V
20:08.39Byteastmoddir => /tmp/asterisk/usr/lib/asterisk/modules <-- why does that not work? :|
20:08.44zipppso I am trying the PA168V IAX file
20:08.54zippphow is one to tell from the phone alone thougH
20:09.04clive-zipp that file is for teh ata device
20:09.39zipppclive-, so which should I use
20:09.53*** join/#asterisk b0ef (n=b0ef@062016141085.customer.alfanett.no)
20:09.54clive-zipp ask areaphone, I thin they call it an ip-100
20:10.02*** join/#asterisk parky (n=kvirc@p5083E146.dip.t-dialin.net)
20:10.04clive-anthm, your on my list...dont worry
20:10.10phantamhow can i tell if the extensions loaded from the DB
20:10.11phantam?
20:10.33sivanatzanger: you around?
20:10.34Bytebkw_: Breakpoint 1, ast_category_browse (config=0x81135b4, prev=0x1 <Address 0x1 out of bounds>) at config.c:252
20:10.36zipppclive-, http://www.iareaphone.com/ShoppingCart/catalogue_enterfromstatic.asp?ProductSet=10274
20:10.40zipppthat is the phone I have
20:10.47mishehuso, is somebody going to tap somebody at digium to kick over their ftp box?
20:10.56*** join/#asterisk sigwerk (n=sigwerk@athena.rootednetworks.com)
20:11.07*** join/#asterisk blessen (n=blessen@140.99.23.26)
20:11.20Byteit does it with the sample configuration too :|
20:11.22blessen:-)
20:11.27*** part/#asterisk negativeu[work] (n=Daniel@border0hsv.asterisksgi.com)
20:11.27blessenHi ..i need a help..
20:11.28anthmtalk about licenses and they will appear
20:11.40blesseni got vonage working with my astwerisk...i am able to pickup calls
20:11.45blessenbut not able to make calls
20:11.50blesseni says all lines are busy
20:12.09*** join/#asterisk konrads (i=konrads@81.198.6.92)
20:12.17blessencan anyone help me with this
20:12.28konradsHow does one fight SIP echo?
20:12.55Ariel_konrads, sip echo???
20:13.09Ariel_blessen, how is your outboud settings?
20:13.40doughecka_hmm, I am thinking about buying a few million licenses from digium....
20:13.45konradsariel: i have some echo when using sip phones
20:14.00blessenin the sip.conf
20:14.03Ariel_doughecka, of what?
20:14.13Ariel_konrads, sip phones to what?
20:14.19doughecka_<PROTECTED>
20:14.22Ariel_blessen, yes
20:14.23blesseni have put context=vonage-out
20:14.23blessen<PROTECTED>
20:14.34anthmlike dorthy
20:14.53phantamcant believe no one can help me :(
20:15.17blessendo u have a link where i can past my settings
20:15.27Ariel_~pastebin
20:15.28jbotmethinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/
20:17.07clive-zipp here is your phone : http://www.voip-info.org/tiki-index.php?page=AT-320
20:17.13*** join/#asterisk darkskie1 (n=darkskie@host86-132-171-18.range86-132.btcentralplus.com)
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20:18.20blessenHI Ariel: here it is
20:18.21blessenhttp://pastebin.ca/22330
20:18.40blessenAriel: Can you please look at it
20:18.59blessenand inform me ..if i am missing something
20:19.55Ariel_blessen, if your dialing 160X number to get out via vonage how do you expect it to dial out via the _9. exten rule
20:19.55doughecka_what would cause my asterisk box to not recieve did on the pri
20:20.13shmaltzdoughecka_, your telco aint sending it to you
20:20.15blesseni didnt get u boss
20:20.24blessenso what change should i make..
20:20.27konradsIs there some comprehensible manual type document for extensions magic?
20:20.35konradslike macros, gotos, etc?
20:20.58blessenithe issue is that...i cannot even call local numbers ie 1602XXXXXXXX
20:20.59FuzzyCatoh goto's  - I read that as goats
20:21.16phantamhow do u query the dialplan from realtime to see it in console?
20:21.32blessencan you tell me...what change ..i have to make to make calls to local...internaltion etc
20:21.34Corydon-wThere is no extensions magic.  Everything performs exactly as it is written
20:21.38Ariel_blessen, do you want to use 9 to dial out?
20:21.46shmaltzphantam, why are you using realtime? it just gives you another point of failure
20:21.48blessenis that how it is..
20:21.57phantambecause
20:22.01Ariel_not really it's messed up
20:22.02phantamwe want it running from db fully
20:22.05shmaltzunless you are trying to run multiple servers with the same config, it makes no sense
20:22.06konradsCorydon-w, right. just like mod_rewrite. Question is, where is it documented?
20:22.09phantamin realtime
20:22.17Ariel_blessen, I just asked a simple do you want to use 9 for dialing out?
20:22.20Corydon-wThe code is self-documenting
20:22.26blessenyes
20:22.32Cresl1ndoughecka: them not sending it?
20:22.33Cresl1n:-)
20:22.37shmaltzpantam, what for?
20:22.39blessenbut 9 is for internation right..
20:22.43phantamshmaltz but at this rate ill never get it working cause no one seems to want to dime in on getting the basics workin
20:22.46Corydon-wwiki?
20:22.48konradsCorydon-w: i should read flex files?
20:22.56Corydon-w~wiki
20:23.06FuzzyCat~full of crap
20:23.08konradsi know of voip-info
20:23.10Corydon-w~docs
20:23.11jbotit has been said that docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
20:23.32Ariel_blessen, ok exten => _9.,1,Dial(SIP/${EXTEN:1}@atlas-east.vonage.net)
20:23.32Ariel_<PROTECTED>
20:24.12blessenwhere should i add ity
20:24.12Corydon-wSo what's wrong with the Wiki that you can't use it for extensions.conf ?
20:24.12Ariel_blessen, but does vonage not require a username and password?
20:24.12blessenok..so this will handle all calls
20:24.12blessenyes i have it
20:24.12shmaltzphantam, drop realtime, it's not meant for single machine installation
20:24.21konradsasteriskdocs.org timeouts
20:24.43blesseni have that line right
20:24.49blessencan u check my configuration once more
20:24.50opus_shmaltz thats a funny thing to say
20:24.57blesseni have that line ....which u just mentioned
20:25.08shmaltzopus_, why?
20:25.28FuzzyCatdoughecka, ??
20:25.29opus_sipusers => odbc,asterisk,sip_conf
20:25.33konradsaha, there is AEL!
20:25.48opus_because of a bug in asterisk you have to use 'asterisk' as the name of the datasource instread of 'MySQL-Asterisk-SXM'
20:25.57opus_shmaltz because its unfounded
20:26.38shmaltzopus_, how so? if you interduce an unnecessary point of failure into a voice network then you are doing something wrong
20:26.40phantamwell
20:26.47phantamthats something someone shudda mentioned lol
20:26.53shmaltzthat is one of the major weakneses of TeleVantage
20:27.17drrayI wonder if ManxPower will buy some channel banks/IP phones with his 2k femavisa
20:27.27blessenso when dialing out...i should use 9 as prefix ??
20:27.56Ariel_blessen, if that is what you want to do yes. I don't like using 9 but that is me.
20:27.58ManxPowerdrray, no, rent temp housing
20:27.58drrayblessen, that's one way a lot of people do it, some have used 8...
20:28.18*** join/#asterisk oden (n=oden@194-237-146-22.customer.telia.com)
20:28.22*** join/#asterisk V9y70Uu (n=c3t20kl@202.160.10.41)
20:28.26*** part/#asterisk V9y70Uu (n=c3t20kl@202.160.10.41)
20:28.27*** part/#asterisk oden (n=oden@194-237-146-22.customer.telia.com)
20:28.40opus_shmalt if your database goes down you have bigger problems for some people
20:29.01opus_database goes down, for example because /dev/hda crapped out. then your asterisk box is toast anyway
20:29.08*** join/#asterisk Veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com)
20:29.08FuzzyCatkonrads, I think you'll find that asteriskdocs will be offline for a long time
20:29.18*** join/#asterisk bprice20 (n=bprice20@216.120.224.168)
20:29.20shmaltzopus_, well that is exactly my point, so why get the phone system down with it?
20:29.39blessenboss..i think i am not understanding....
20:29.43blesseni can get calls
20:29.51blessenbut i am not able to make calls....to outside world
20:29.56blessennot even to local number
20:29.56*** join/#asterisk _DAW (n=_DAW@ip68-229-153-182.lf.br.cox.net)
20:30.07opus_shmaltz either way your phone system is down
20:30.19blessenso can u tell me what all are checked when a calls made through asterisk
20:30.27phantamopus_: its still the same... says connected in 'odbc show' and it shows the realtime in  'sip show settings'
20:30.43phantambut the 'show sip users' still shows the sip.conf people not the realtime users
20:30.58*** join/#asterisk Defraz (n=t0tal@tim.ibccom.net)
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20:31.09*** part/#asterisk I1w13Rk (n=g4x10td@65-37-89-85.br1.leo.pa.frontiernet.net)
20:31.22blessenwhill the general section under sip be checked whena call comes in
20:31.28clive-is anyone here familiar with grub.conf ?
20:31.28shmaltzI'm getting this error:
20:31.30shmaltzSep  7 16:20:51 WARNING[303]: No D-channels available!  Using Primary on channel anyway 48!
20:31.31shmaltzIs it possible because of timing?
20:31.38blessenor will it go to the [100] which is there in my sip.conf
20:31.55shmaltzopus_, not true, how is my phone system down when my DB is down?
20:33.18blessenAreil: i think..I should create an entry for [atlas-east.vonage.net] in sip.conf file....right
20:33.19blessen??
20:33.40blessenfor this line to work..exten => _9.,1,Dial(SIP/${EXTEN:1}@atlas-east.vonage.net)
20:33.44phantamdo i have to turn on caching or somethin in sip or what
20:33.44phantam?
20:34.07*** join/#asterisk mutilator (n=animenod@65.111.201.79)
20:34.41mutilatorwhat is the 'line build out' for my span settings?
20:34.50mutilatori always thought it was like a distance thing...
20:34.54mutilatorbut whats it do?
20:34.57phantamhehe its funny lol the people i talk to disappear hehehe
20:35.02*** join/#asterisk jimmy_deanPB (n=jhodapp@adsl-70-228-242-126.dsl.ipltin.ameritech.net)
20:35.11mutilatormy line build out is set to 0 and it's running like 2000 ft
20:35.16bprice20has anyone else had issues w/ people complaing of dtmf not working but only for voicemailmain?
20:35.19*** part/#asterisk bprice20 (n=bprice20@216.120.224.168)
20:35.31mutilatoron both the channel bank and the te110p
20:35.59*** join/#asterisk jeffik (n=Jeff@70.49.221.34)
20:36.40*** join/#asterisk Bobby_Ewing (n=Chat@82.152.11.145)
20:38.28Bytebkw_: ==21549==    definitely lost: 936 bytes in 17 blocks. ==21549==    still reachable: 334277 bytes in 5766 blocks. http://redrum.arlott.org/tmp/valgrind.txt
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20:40.20Ayanoanyone in here use teliax?  (sip)
20:41.44*** join/#asterisk audela (n=audela@82-33-115-145.cable.ubr08.azte.blueyonder.co.uk)
20:42.38phantamhow the heck do u use the realtime load
20:42.39phantam?
20:43.17*** join/#asterisk many (i=many@krikkit.ukeer.de)
20:43.19hugo-v6with w, or top
20:43.19*** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1)
20:43.22manyohdear
20:43.31hugo-v6hiho many
20:43.31manyscrap that shitty +J cmode
20:43.46manyheyho.
20:45.19hugo-v6many: jep that mode sucks like hell.
20:46.10*** part/#asterisk Bobby_Ewing (n=Chat@82.152.11.145)
20:46.34phantamin realtime mode
20:46.39phantamdo you see the peers listed in sip show peers?
20:47.05malcolmdCresl1n: howdy :)
20:47.07*** join/#asterisk distortio (i=distorti@junipero.3sheep.com)
20:51.25*** join/#asterisk dudes (n=dudes@12.215.34.84)
20:52.49phantamok im getting a migrane
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20:54.53ManxPower~seen CaNaBiS_
20:54.56jbotcanabis_ is currently on #asterisk
20:55.06ManxPower~seen CaNaBiS_
20:55.07jbotcanabis_ is currently on #asterisk
20:55.25*** join/#asterisk citats (i=james@209.142.250.180)
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20:57.03file[muon]...
20:57.15*** part/#asterisk pa (n=Paolo@unaffiliated/pa)
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20:58.40*** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)
20:58.49*** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net)
20:58.55generalhanwhats up everyone ?
20:58.59*** part/#asterisk jimmy_deanPB (n=jhodapp@adsl-70-228-242-126.dsl.ipltin.ameritech.net)
20:59.31generalhanis there a way to tell asterisk what my callerID should be when im making outgoing calls ? cuase right now it just always says "private"
20:59.51ManxPowerOther than the credit card processing networks getting HAMMERED?
21:00.16konradsgeneralhan: SetCallerId(31337)
21:00.24konradsin dial plan
21:00.36ManxPowergeneralhan, SIP, IAX2, PRI, CT1, AnALOG?
21:00.42generalhanIAX
21:00.44konradsor most channels support callerid=I am Master <31337>
21:01.13*** join/#asterisk MrMAGO (n=maxgluck@200.109.166.83)
21:01.15*** join/#asterisk zx225 (n=me@65.183.42.3)
21:01.29ManxPowergeneralhan, it will provide whatever callerid is set
21:01.42ManxPoweractually it looks like paypal is being hammered too
21:01.57generalhanwell ... see now when i set that up in the sip.conf file as callerid="whatever
21:01.59generalhan"
21:02.08konradsManxPower: hammered?
21:02.16generalhanit shows that when i call inside the asterisk server, but on all outgoing it shows up to people as private
21:02.29konradsgeneralhan: maybe they drop the callerid
21:02.47ManxPowerkonrads, yes.
21:03.17distortiois there a way to tell asterisk to not proxy rtp?
21:03.28konradsManxPower DDOS on IP networks?
21:03.41konradsdistortio: SIP? careinvite=yes i think
21:03.49ManxPowerkonrads, I doubt it.  Just MASSIVE numbers of people trying to use paypal and credit cards
21:04.19konradsManxPower: perhaps a psychological surge
21:04.37distortiokon: is there a similar command for h323 (chan_h323 not ohh323)
21:04.40konradsmanx: hard to tell what tinfoil theories would say on ads or subliminal messages
21:04.44ManxPowerkonrads, no, hundreds of thousands of people who's bank is in New Orleans and so can't get cash
21:05.01ManxPoweralso fema is finally handing out debit cards to people and those use the CC networks
21:05.06*** join/#asterisk file[laptop] (n=jcolp@mctnnbsa30w-156034035130.nb.aliant.net)
21:05.07konradsmanx: that was too obvious for a guess
21:05.11konradsfema?
21:05.21konradsdistortio: oh323c driver did that by default
21:05.26ManxPowerkonrads, USA disaster relief govt org
21:05.41konradsdistortio: if 2 systems can't agree on codecs asterisk will transcode and thus proxy
21:05.55opus_there in charge of building nazi style concentration camps
21:05.56opus_:)
21:06.18distortioim trying to prevent it from trying to transcode
21:06.53konradsdistortio: first, make sure both endpoints choose a mutually known codec
21:07.13doughecka_bah, after changing out the server, the problem with my non-did status went away
21:07.18konrads2) read the h323 page on voip-info, one h323 channel would explicity transcode
21:07.29doughecka_why would setting aggressive echo cancel break that?
21:07.37*** join/#asterisk pauldy (n=pauldy@24-155-76-232.dyn.grandenetworks.net)
21:08.14doughecka_is there anyway for me to save a 'good' copy of zaptel off?
21:08.25doughecka_after its been compiled and all that
21:08.49doughecka_so I can try this again and if it breaks, I can plop the old zaptel stuff in there
21:09.09doughecka_yo kram!
21:09.49HmmhesaysI got the green glow under my car
21:09.54HmmhesaysI got the boom boom system you can hear real far
21:13.34konradsHmmhesays www.funroll-loops.org is for you :)
21:13.40mishehuand an airplane wing on the back?
21:14.37Hmmhesayshaha, i was singing a song
21:14.42Hmmhesaysi drive a 1989 honda accord
21:14.47Hmmhesaysneed balls joints
21:14.59Hmmhesaysbut I really don't want to put them in, because the factory ones are pressed in
21:15.09file[laptop]http://www.flickr.com/photos/petersawatzky/41237814/ -> OMG IT'S ME
21:15.29bkw_HAHA
21:15.50Hmmhesayson the upside I get to play with a LARGE drill
21:16.06*** join/#asterisk smcmahon (n=admin@digitaldatabits.net)
21:16.18MikeJ[Laptop]file[laptop], who is that freaky guy?
21:16.32file[laptop]I think it's like, me?
21:16.35file[laptop]maybe, just maybe
21:16.48*** join/#asterisk Marlow (n=marlow@host-194-46-240-51.dsl-ie.utvinternet.net)
21:16.49smcmahonWoo Hoo Woo Hoo Woo Hoo
21:17.02Marlowsmcmahon : christmas around here ? :o)
21:17.12bkw_yo yo yo
21:17.12smcmahonYeah
21:17.25*** join/#asterisk darkskiez (n=darkskie@host86-132-171-18.range86-132.btcentralplus.com)
21:18.25MikeJ[Laptop]file[laptop], not you.. the other guy
21:19.06file[laptop]the other other guy? Peter?
21:19.09file[laptop]a friend of mine.
21:19.27TheLawHello. I dont't get a simple playback application running for testing asterisk. exten command "    -- Executing Playback("SIP/6044201-a2a4", "demo-thanks") in new stack
21:19.27TheLawSep  7 23:19:00 WARNING[7079]: file.c:475 ast_openstream: File demo-thanks does not exist in any format
21:19.27TheLawSep  7 23:19:00 WARNING[7079]: file.c:787 ast_streamfile: Unable to open demo-thanks (format ulaw): Success
21:19.27TheLawSep  7 23:19:00 WARNING[7079]: app_playback.c:83 playback_exec: ast_streamfile failed on SIP/6044201-a2a4 for demo-thanks
21:19.28file[laptop]I have friends - scary eh?
21:19.30TheLaw"
21:19.52Hmmhesays<PROTECTED>
21:20.20*** join/#asterisk NirS_ (n=Nir@84.94.25.59.cable.012.net.il)
21:20.33MikeJ[Laptop]well, it's clearly not demo-thanks in the sounds dir
21:20.35TheLawi dont think so: -rw-r--r--  1 root root 7920 Aug 14 09:07 demo-thanks.gsm
21:20.47MikeJ[Laptop]TheLaw, what dir?
21:21.01Hmmhesaysmy bad
21:21.13TheLaw/var/lib/asterisk/sounds which is a symlink to /usr/share/asterisk/sounds
21:21.27Hmmhesaysi guess it doesn't like symlinks eh?
21:21.39MikeJ[Laptop]can't see why it wouldn't
21:21.54MikeJ[Laptop]does it work if you kill the link and copy the file to that dir
21:22.03hardwireso this should be interesting
21:22.05Hmmhesaysor change your path in asterisk.conf?
21:22.09hardwiregoing to attempt to convert a small hotel to voip
21:22.13hardwirebut they want the bed phone.. and the study phone
21:22.24hardwirethe bristuff pattch that allwos the Steal() function would be ideal
21:22.25*** join/#asterisk znoG (n=gs@200.115.218.81)
21:22.26Hmmhesayssomeone installed packages instead of compiling
21:22.38TheLawhow should the line in asterisk conf should look like? i dont got anything with sounds.
21:22.41hardwirethat way they can go to the other phone.. hit the line and it steals or picks up the call group
21:22.43MikeJ[Laptop]hardwire, you can use changrab too
21:22.48hardwireoh
21:22.49Hmmhesayshardwire: i'm doing a 90 room hotel next week
21:22.50hardwirechangrab
21:22.50*** join/#asterisk Syncros (n=sysop@noc.routermonkey.net)
21:22.52hardwireis that 1.2.0?
21:22.55Hmmhesayspossibly the week after
21:23.28MikeJ[Laptop]pbxfreeware.org
21:23.46MikeJ[Laptop]wait.. you want group pickup..
21:23.47MikeJ[Laptop]ummmm
21:25.06*** join/#asterisk pauldy (n=pauldy@24-155-76-2.dyn.grandenetworks.net)
21:25.39newmemberwhat high desity FXS unit can I use for an ASP application, basically to deliver dial tone to a company as a service offering?
21:25.53Hmmhesaysnewmember: yes
21:25.55TheLawit does not work with copying either. i think there is some other fault.
21:26.07TheLawwhat modules do i need for playback to work?
21:26.09Hmmhesays1124
21:26.10newmember"density" was what I was looking for
21:26.16*** join/#asterisk Poincare (n=jefffnod@dD5779BD2.access.telenet.be)
21:26.20Hmmhesaysthe mediatrix 1124 is what you are looking for
21:26.30Hmmhesaysa bitch to configure, but rock solid in operation
21:26.32*** part/#asterisk Poincare (n=jefffnod@dD5779BD2.access.telenet.be)
21:26.48Hmmhesaysand I'm trying to get them to make an eeks load
21:26.51Hmmhesayserr iax2
21:27.18newmemberwhat are they currently compatable with, protocol wise?
21:27.25Hmmhesayssip, h323, mgcp
21:27.41*** join/#asterisk santiago (n=santiago@63.245.87.180)
21:27.45newmemberstill, not too bad for options
21:27.57Hmmhesaysnewmember: no, they are damn nice gateways
21:28.01Hmmhesayshey file, what up?
21:28.03DannyFHmmhesays, now that would be lovely!
21:28.15file[muon]nada, u?
21:28.20DannyFlo file
21:28.26HmmhesaysDannyF: i'm working on it, you might see an iax2 load on for mediatrix within the next year
21:28.30file[muon]hi
21:28.46DannyFHmmhesays, looking forward to it, sure would simplify stuff.
21:29.01HmmhesaysDannyF yeah
21:29.21Hmmhesaysif I make it back up to sherbrooke again soon I will convince them to do it
21:29.28DannyF;)
21:29.42DannyFdid Manx make it to NO btw?
21:29.58Hmmhesaysnewmember: if you want some info drop me a line at my gmail account
21:30.26DannyFmmm nvm he's on so guess all went well
21:30.59newmemberHmmhesays: any features missing when using *?
21:31.04Hmmhesaysdefine missing?
21:31.47Hmmhesaysthe 1124 is capable of hold, transfer 3 way calling with flash hook on a regular phone. mwi stutter dialtone is supported
21:32.08newmemberhmmm, not available to a 1124 but available to an FXS card in an * server.
21:32.18HmmhesaysAND if you are putting it in something like a hotel, you can punch it down in the phone room and not have to change any wiring
21:32.36Hmmhesaysthe only thing that comes to mind is flash hooking a pots line
21:32.37*** join/#asterisk khmerhugh (n=khmerhug@boilermaker.ofc.tekinteractive.com)
21:32.53Hmmhesaysyou can't go sip-->zap flash hook
21:33.06Hmmhesaysbut really you don't often need to do that
21:33.17file[laptop]theoretically you could... but I can't disclose how
21:33.52Hmmhesaysyes theoretically you could, make asterisk process a notify message
21:34.01file[laptop]WELL sorta
21:34.15Hmmhesaysanyway, the 1124 with asterisk can do everything you need it to and a lot more
21:34.24Hmmhesaysat 100 bucks a channels its a hell of a gateway
21:34.41newmemberI was hoping I could get * to phone into a Nortel switch, then call an extension, if no answer, * would send the call to either VM or reroute to a cell phone
21:34.45file[laptop]I'd be curious to see a 1124 and use it... and like, add support for that... but meh I'm silly
21:34.52*** join/#asterisk drumkilla (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
21:34.52*** mode/#asterisk [+o drumkilla] by ChanServ
21:34.54Hmmhesaysugh nortel
21:35.01newmemberlol
21:35.22Hmmhesays1124 do which file[laptop]
21:35.52file[laptop]well you said it had a NOTIFY it could send
21:36.23develhey, can sipsak be told to not expect a reply, for example to fake a notify to a client (that doesn't answer the packet)?
21:36.30file[laptop]I *Think* (don't quote me - I'm not a zaptel person) you can send a frame to trigger a flash on a zaptel channel
21:36.30Hmmhesaysi'll have to look at it again, if indeed I am not on crack, i'll send you the sip messages if you like
21:36.48file[laptop]kk
21:36.59*** join/#asterisk Ganlron (n=Ganlron@omega.csolve.net)
21:37.10file[laptop]I know chan_sip advertises in the SDP for RFC2833 that we actually allow flash...
21:37.48Hmmhesaysinteresting, I know I can get the mediatrix engineers to make the necessary changes if I provide them with enough info
21:38.54HmmhesaysI gotta run right now, maybe chat about this later file[laptop]
21:39.03file[laptop]yup yup
21:39.08file[laptop]I'm always on.
21:39.18Hmmhesaysnewmember, you can probably guess me gmail account, drop me a line if you want some info
21:39.20Hmmhesayslater
21:40.11*** join/#asterisk mx02 (n=none@adsl-215-143-149.aep.bellsouth.net)
21:40.37hardwirehttp://communications.siemens.com/cds/frontdoor/0,2241,hq_en_0_69566_rArNrNrNrN,00.html
21:40.39hardwireholy crap thats neat
21:40.58develso how about using the digium single port analog card for a CAMA trunk (to talk to our telco for 911 etc.)?
21:41.00jarrodanyone here using sylantro application server ?
21:41.46Marlowhardwire : it's not really, what you think it is
21:41.52Marlowhardwire : that's the problem
21:42.08hardwireMarlow: I would imagine its a USB-Audio voice liknk to most gigaset handsets
21:42.08*** part/#asterisk mx02 (n=none@adsl-215-143-149.aep.bellsouth.net)
21:42.22hardwirenot something that would "read my mind"
21:42.22hardwire:)
21:42.23Marlowhardwire : nope .. it still requires the base
21:42.30hardwireyeh.. I saw that
21:42.37hardwirebut I don't really care about the base.. shuold I?
21:42.54Marlowhardwire : this thing can only connect to the dect base
21:42.55*** join/#asterisk shmaltz (n=chatzill@69.28.255.210)
21:43.08shmaltzwhy am I getting this:
21:43.09shmaltzNo D-channels available!  Using Primary on channel anyway 48
21:43.17FuzzyCatVoice over IP (VoIP) telephony with the Gigaset M34 USB adapter works only if the Skype software , provided with the product, is installed  from CD
21:43.21Marlowhardwire : it can then send your call from your computer to the base and that way to any handset, but not communicate with the handset directly
21:43.23smcmahonSkype sucks
21:43.28FuzzyCatyup
21:43.47hardwireMarlow: I imagined as much
21:43.51Marlownot only that skype sucks, but the siemens m34 sucks totally
21:43.53hardwireso maybe I didn't have the same impression
21:43.58hardwireheh
21:44.02Marlowa dect basestation emulator would have been better
21:44.03*** join/#asterisk MrMAGO (n=maxgluck@200.109.166.83)
21:44.07hardwireMarlow: its a neat way for them to deal with things :)
21:44.10hardwirebut yes
21:44.15Marlowhardwire : sure ... but useless ..
21:44.16FuzzyCatjust use a tdm card and a real base station
21:44.24hardwireUSB-Audio + Control Channel -> DECT Handset in one small usb or ethernet package would have been great
21:44.27Marlowhardwire : a friend of mine bought it in the hope to use it on the run
21:44.38hardwirehe is a felon?
21:44.40Marlowhardwire : airports, trainstations, hotels, etc.
21:44.46hardwirehaha
21:44.48hardwireI see
21:44.54hardwireand a wireless card + a wisip was just too much?
21:44.55Marlowhardwire : and ... no go
21:44.56*** join/#asterisk kitsaws (n=none@adsl-215-143-149.aep.bellsouth.net)
21:45.02smcmahonLOL ... I like ... a Cisco ATA 186 and a SPA-2000 (Sipura) all connected to a Comdial Phone system :)
21:45.20Marlowhardwire : nope .. but at that point none of the wisip's were stable enough
21:45.28hardwiregotcha
21:45.30hardwirethey still arent
21:45.30Marlowhardwire : the m34 has been out for over a year
21:45.33hardwireI use a hitachi cable now
21:45.37Marlowhardwire : the hitachi is
21:45.42hardwireanybody want to buy some wisips :)
21:45.50smcmahonwhat is a wisips?
21:45.51hardwireflashed to latest zyxel firmware :)
21:45.57Marlowhardwire : and the senao is also, if you don't have a gemtek based ap
21:46.01hardwiresmcmahon: wireless sip phone
21:46.10smcmahonOhh 802.11 Phone?
21:46.11hardwireI never use gemtek.
21:46.16hardwiresmcmahon: yeh
21:46.21Marlowhardwire : zyxel firmware ? on the hitachi .... that's bullcrap
21:46.26smcmahonYah I hear those things are hard to read on the display
21:46.39hardwireMarlow: I offered wisips.. using zyxel.
21:46.45Marlowhardwire : ah
21:46.58*** join/#asterisk acqua7 (n=bc290db0@dsl-200-67-147-190.prod-empresarial.com.mx)
21:47.00Marlowhardwire : forget'em  .. unhandy and unstable
21:47.05hardwireyeh?
21:47.09Marlowhardwire : no matter how new the firmware is :)
21:47.14hardwireyeh
21:47.14Marlowhardwire : it's zyxel :)
21:47.17hardwireno xfer sucks btw
21:47.19smcmahonWould be nice to have a 802.11 Phone and go to a starbucks and make a call from inside starbucks from my house :)
21:47.22hardwireMarlow: I know :)
21:47.36Marlowhardwire : anyhow .. in my experience wisips don't help much ..
21:47.43Marlowhardwire : i use'em only at home ..
21:47.58Marlowhardwire : class one bt module and headset , that's the deal
21:48.05hardwirequit saying my name
21:48.07hardwireits driving me insane.
21:48.11hardwire:)
21:48.12Marlowhardwire : most hotels + airports require login via portalpage
21:48.28Marlowhardwire : better so, go insane
21:48.33hardwire:)
21:48.39hardwireMarlow: I have a bluetooth phone
21:48.46hardwirethat groks the handset bluetooth audio link
21:48.55hardwireI have been really wanting some handset code to make its way into alsa
21:49.09hardwireso I can approach using my phone as a handset for voip calls
21:49.10Marlowhardwire : they are rare .. haven't seen'em in the shops around here yet
21:49.18hardwireI have the Motorola V710
21:49.21hardwirewith broken bluetooth :(
21:49.25Marlowarghh
21:49.26hardwirebut it does this one thing right
21:49.31tzangerhardwire: is that a chipped bluetooth?
21:49.38hardwiretzanger: its the piss me off bluetooth
21:50.03*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
21:50.17hardwirehmm.
21:50.25hardwirewell this is going to be kinda weird..
21:50.41hardwireI need to find a multi-line sip phone with dialplan support.. two lines only.
21:50.43Marlowwell .. i'll better get back to this package builder
21:50.44*** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
21:50.53hardwireand then make some really weird macros for picking up a call based on those function keys
21:51.00hardwireto emulate legacy hotel phone systems
21:51.06redder86What's the maximum possible size of an IAX voice frame?
21:51.08Marlowhardwire : cisco 7940 :)
21:51.13hardwireunless anybody here has stayed at a hotel that has a much smarter phone system I can go off of
21:51.19hardwireMarlow: not for a hotel
21:51.29Marlowhardwire : sure -.. ppl steal it though
21:51.35hardwirethey look obtrusive.. most people would be looking in them for a camera
21:51.39Marlowhardwire : unless you want it wireless :)
21:51.40hardwireor.. stealing it
21:51.46hardwirethat == more theft
21:51.53Marlowhardwire : i said that allready
21:52.02hardwireI would rather they had some clear choices on the phone.. like a fkey for "xfer to bed phone"
21:52.24hardwirewhich would require some deadagi to chansteal and transfer
21:52.27hardwireor something like that
21:52.56hardwiremy office is amazingly hot
21:52.57hardwirethis sucks
21:53.00hardwireand today I found my fan
21:53.01hardwiredead
21:53.02hardwirealone
21:53.02hardwirecold
21:53.04hardwirebroken
21:53.05hardwire:(
21:53.15hardwirepower suplpy/motor must have burned out
21:53.16hardwireor a fuse
21:53.29hardwireok.. I overrambled
21:53.37CoriantumWhat is the correct way to send multiple variables in a Manager Originate?  The wiki may be wrong?!?!
21:53.52*** join/#asterisk jaybuffet (n=random@rrcs-24-227-53-138.se.biz.rr.com)
21:54.02doughecka_?!?!?
21:54.20*** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
21:54.32wunderkinCoriantum: Variable: __leadid=$hopper->leadid\r\nVariable: __phone=$hopper->phone\r\n    ... etc
21:54.34*** join/#asterisk tzafrir (n=tzafrir@local.xorcom.com)
21:54.45*** join/#asterisk toddf (n=toddf@ns0.fries.net)
21:55.21Coriantumwunderkin: multiple variable statements for each, huh?
21:55.22jaybuffethello all...   whats a good business voip provider.. i need like 8 simultaneous calls, a fax number, and a toll free number
21:55.42wunderkinthats how i do it
21:55.44file[laptop]fax over VoIP is iffy at best.
21:55.53wunderkinit was done differently before
21:56.08Coriantumwunderkin: cool thanks...
21:56.24jaybuffetfile[laptop]:why is that
21:56.28CoriantumYeah... the wiki says Variable: value1=23|value2=34|...
21:56.49file[laptop]jaybuffet: because you need low latency, no disruption in the stream of audio, no jitter
21:56.58file[laptop]but that's hard to achieve over the internet
21:57.08file[laptop]plus you need an uncompressed codec like ULAW to even have a chance to begin with
21:57.27jaybuffetfile[laptop]: ah i c...  not a biggie.. i can just use a land line like we currently do
21:57.33*** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.241)
21:57.46Coriantumwunderkin: worked like a charm... thanks
21:57.59Connor_<PROTECTED>
21:58.32*** join/#asterisk Katty (n=katrina@68.112.15.110)
21:58.33Kattyhi.
21:58.53jaybuffetConner_:  will the voip provider mention they use T.38
21:59.00file[laptop]Asterisk doesn't support T.38
21:59.06zx225Conner_ what equipment are you using for T.38
21:59.14brad_msswmost voip providers dont support t.38 either
21:59.18Kattymrow?
21:59.22Connor_I didn't say I did it.. I said it was the ticket...
21:59.25file[laptop]Katty: woof
21:59.28Kattyfile[laptop]: mew.
21:59.40zx225ok
21:59.43*** join/#asterisk jaxxan (n=jaxxan@202.70.125.216)
21:59.46zx225what equipment is the ticket
22:00.03jaxxanhey ya'll
22:00.15jaybuffetwhy doesn't asterisk support it? is it in the works or is it too large of an undertaking
22:00.24file[laptop]people are working on it.
22:00.51jaxxanif you use "exten => s,4,Queue(directory|r)" what generates the ringing tone ?
22:00.55Kattyjaybuffet: we're not supergrover )=
22:00.56jaybuffetfile[laptop]: oh yeah.. i see the bounty.. heh
22:01.06loudthe r.
22:01.10jaxxanis it just a sound file ?
22:01.13jaxxanthat plays ?
22:01.21Kattyjaybuffet: but we do offer an unlimited supply of headaches.
22:01.35jaybuffetKattY: i'm looking forward to it
22:01.41Kattyjaybuffet: (((=
22:01.49DrukenHMEKatty: super grover was awesome!
22:01.51jaybuffeti just want to jump in.. but i know i better go slow for the businesses sake
22:01.54KattyDrukenHME: he so was :>
22:01.55tzangeractually I have supergrover on my MSN
22:01.59Kattytzanger: hot.
22:02.05Kattylike my dinner.
22:02.07Kattybye now.
22:02.10hardwireI wonder how much crap I can run off of this outlet
22:02.19hardwire2 phones some monitors a computer a laptop and a fridge maybe?
22:02.23hardwirehere's hoping!
22:02.48DrukenHMEtzanger: your not still at work are ya?
22:02.52tzangeryup still here
22:03.01DrukenHMEwhere do you work again ?
22:03.16hardwire15a :)
22:03.17tzangerhttp://qdb.us/48107
22:03.31*** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net)
22:03.40*** join/#asterisk b0ef (n=b0ef@062016141085.customer.alfanett.no)
22:04.10jaybuffetthats nice...   voip-info says broadvox.net has support for t.38 but when you go to the small business section it says .. 404
22:04.14jaxxananyone know which sound file in /var/lib/asterisk/sounds contains a standard ringing tone ?
22:04.15*** join/#asterisk Ganlron (n=Ganlron@omega.csolve.net)
22:04.41DrukenHMEring.gsm perhaps?
22:04.49DrukenHMEi'm just guessing of course
22:05.05DrukenHMEjaxxan: what do you need it for?
22:05.10jaxxanno such file
22:05.51*** part/#asterisk jaybuffet (n=random@rrcs-24-227-53-138.se.biz.rr.com)
22:06.08jaxxanDruken:  i'm trying to do Queue(directory) and have play a "ringing" musiconhold file.
22:06.30jaxxani know you can use |r to playback ringtones, but that's not what i want cause that answers the call and i dont want that to happen
22:06.59*** join/#asterisk mike_jh (n=mike@81.187.90.205)
22:07.08iLuvCandyhi, does anyone know how to access the budgetone -100? the "admin" password no longer works. is there a way to reset it?
22:07.19jaxxankind of a workaround for now.
22:07.38DrukenHMEjaxxan: the queue has an option to ring instead of moh
22:07.49jaxxanoh yeah ?
22:07.54DrukenHMEyeah....
22:08.20jaxxanyou mean |r
22:08.23mike_jhCan I pick someone's brains on setting up extensions?
22:08.30jaxxan?
22:08.39*** join/#asterisk stevek (n=stevek@208.185.78.2)
22:09.14mike_jhWhen you dial a number, how does * know which extension to ring?
22:09.52FuzzyCatdepends on what type of phone it is...
22:09.54MikeJ[Laptop]mike_jh, have you read any of hte documentation yet?
22:09.58FuzzyCatbut basically....
22:10.11MikeJ[Laptop]~docs
22:10.14jboti guess docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
22:10.16FuzzyCatif it's sip then it'll be the name  in [] in sip.conf
22:10.19MikeJ[Laptop]that is a good place to start
22:10.24FuzzyCateg [1102]
22:10.27jaxxanDrukenHME: I'm looking at the example queues.conf.sample and i dont see the option your talking about, the only way i know to force ringing in a queue is to call the queue with |r
22:10.38FuzzyCatand you refer to it as SIP/1102
22:10.40mike_jhYes, I just can't seem to get my head around how the extensiosn tie together
22:10.59FuzzyCatextension numbers do not relate to phones at all
22:11.19FuzzyCatyou can have an extension 111111 that calls a phone defined as 99999
22:11.42FuzzyCatwhen you call the DIAL command you tell it what phone to ring, eg
22:11.44mike_jhAhh yes, I understand that a number has a dial plan which dicates what it should do
22:11.44*** join/#asterisk darkskie1 (n=darkskie@host86-132-170-73.range86-132.btcentralplus.com)
22:11.55FuzzyCatexten => 111111,1,Dial(SIP/999999)
22:12.11DrukenHMEjaxxan: jaxxan: well, i'm pretty sure i remember seeing it.... take a look around voip-info for it
22:12.26mike_jhAhh, so if I dial 111111 I get the phone called [999999] in sip.conf?
22:12.32FuzzyCatyes
22:12.51mike_jhI think the penny might be about to drop
22:12.55FuzzyCat:)
22:13.34iLuvCandydoes anyone know how to access the budgetone?
22:13.36iLuvCandy100
22:13.37iLuvCandy?
22:13.41*** join/#asterisk Pete_Largo (n=Pete_Lar@225-196.35-65.tampabay.res.rr.com)
22:13.52iLuvCandyi tried 'admin' but it got me locked out
22:14.06wunderkiniLuvCandy: admin is the default password, yes
22:14.19FuzzyCatiLuvCandy, I'd have a google for  'forgot admin bugetone 100'
22:14.28Pete_Largodoes the Grandstream GXP-2000 work with hint ??
22:14.37iLuvCandyok
22:15.16FuzzyCatmike_jh, take a look at my site (specifically theis page http://www.automated.it/asterisk/lah-3-6-05_3.html )
22:15.28*** join/#asterisk JunK-Y_ (n=junky@Toronto-HSE-ppp3743074.sympatico.ca)
22:15.28Pete_Largoor... does hint work with the Grandstream GXP-2000?
22:16.25*** join/#asterisk pauldy (n=pauldy@24-155-75-84.dyn.grandenetworks.net)
22:17.50sivanaquit
22:17.52sivanabah
22:17.53FuzzyCatiLuvCandy, http://www.grandstream.com/y-faq.htm#q12
22:21.16FuzzyCat/quit?
22:21.53DrukenHMEnah... quiting is bad... it's better to resign
22:21.54iLuvCandyFuzzyCat, thanks
22:22.04FuzzyCatnp
22:22.37*** join/#asterisk bjohnson (n=bjohnson@i216-58-58-190.cybersurf.com)
22:23.21acqua7hi
22:23.42*** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk)
22:23.46FuzzyCatlo
22:24.16*** join/#asterisk pauldy2 (n=pauldy@24-155-75-51.dyn.grandenetworks.net)
22:29.36*** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk)
22:30.08MRH2hi how do i know if i have a card that is firmware updateable?
22:30.16MRH2410 /411P
22:30.27*** join/#asterisk fiber0pti (n=johndoe@207.114.199.98)
22:30.49fiber0ptiIs it possible to set up an extension that when dialed it goes to the dial plan like when someone calls in from an outside line?
22:31.31DrukenHMEfiber0pti: yes....
22:31.37fiber0ptihow?
22:31.45*** join/#asterisk Delta34 (n=delta34o@198.87.24.253)
22:31.50DrukenHMEby sending the call to the start of the dialplan
22:32.22fiber0ptiI guess I don't know how to do that. I've never sent a call from one section of a dial plan to another
22:32.28DrukenHMEdial(LOCAL/s@context)
22:32.39FuzzyCatjust use a goto
22:32.45*** join/#asterisk darkskiez (n=darkskie@host86-132-171-131.range86-132.btcentralplus.com)
22:32.56DrukenHMEFuzzyCat: or you could do that too....
22:33.17FuzzyCatexten =>5551212,1,Goto(context,exten,priority)
22:35.03*** join/#asterisk nick125 (n=nick125@unaffiliated/nick125)
22:35.18MRH2is there anyway to tell by looking at the card?
22:35.41fiber0ptifuzzycat: thanks. it worked
22:35.51*** join/#asterisk jeremywhiting (n=jeremy@71-37-101-103.slkc.qwest.net)
22:35.58FuzzyCat:)
22:36.52jake1932anyone want a used te100p?
22:37.19FuzzyCatonly if it's free - and you supply 2 and a cross cable... ;)
22:38.01jake1932FuzzyCat: Nothing in this world is free
22:38.25FuzzyCatit is, it's just you don't get offered it often...
22:38.57jake1932FuzzyCat: ok - name one thing
22:39.09*** join/#asterisk darkskiez (n=darkskie@host86-132-171-131.range86-132.btcentralplus.com)
22:39.18FuzzyCatthe advice I just gave to iLuvCandy and mike_jh
22:39.19jake1932FuzzyCat: bet you i can prove it's not fre
22:39.24FuzzyCatthat was free
22:39.57jake1932FuzzyCat: iit cost time
22:40.01Delta34i got a question regarding timing from a zap channel, if i have two port digium card and its plugged into a telco provider that is providing timing should i set both spans to 0?
22:40.06hardwireman I would love having a 900mhz frequency hopping sip handset :)
22:40.06Delta34for timing
22:40.10hardwireEngenius should just make them
22:40.14hardwirethey already can SMS to the base
22:40.15FuzzyCatnotit didn't, I have a time machine jake1932
22:40.20hardwireuse that as a control protocol for xfer and the like
22:40.46jake1932FuzzyCat: a time machine - then you are correct - my bad
22:41.01FuzzyCatI knew you'd say that...
22:41.05FuzzyCat;)
22:41.21ManxPowerhardwire, it's called a SIPura
22:41.25hardwireeh?
22:41.27ManxPowerwith the correct cordless phone
22:41.34*** join/#asterisk marv (n=ilovekim@pcp01529782pcs.huntsv01.al.comcast.net)
22:41.34ManxPowerhardwire man I would love having a 900mhz frequency hopping sip handset :)
22:41.44ManxPowersince no WiFi runs at 900Mhz
22:41.45hardwireheh
22:41.46FuzzyCat900mhz ... snigger
22:41.48jeremywhitinghi all
22:41.55hardwireFuzzyCat: I wouldn't snigger
22:42.02FuzzyCatI would,...
22:42.03hardwireplayed with the Engenius Industrial cordless phones yet?
22:42.06jeremywhitinganyone know a way to let a regular user run asterisk -r?
22:42.09FuzzyCattake a look at the dect spec
22:42.10hardwire900mhz freq hopping
22:42.16hardwireI got it at 6 miles over the ocean
22:42.26hardwirethats pretty damn nice
22:42.28jeremywhitingso I can let someone check my system without giving him root access
22:42.35*** join/#asterisk aneredes (n=hannes@port-212-202-55-34.dynamic.qsc.de)
22:42.36FuzzyCatif you live in the sea ;)
22:42.57hardwireI work on the sea :)
22:43.02FuzzyCatjeremywhiting, not really, it's more useful to have root access
22:43.10jeremywhitingok, thanks
22:43.12Vcomotorola canopy 900mhz = 64km range
22:43.13FuzzyCathardwire, ahh well then, it is good
22:43.26jeremywhitingguess I'll just have to change passwords again after he's done
22:43.44jake1932jeremywhiting: I remeber someone doing that
22:43.45FuzzyCatyeah... you can use screen too so you can watch what they do -
22:43.58FuzzyCatbut if you don't trust them then why are you letting them in
22:44.15jake1932jeremywhiting: http://www.voip-info.org/tiki-index.php?page=Asterisk+non-root
22:44.58FuzzyCatjake1932, if he needs someone to login as root to help that link wontbe much use
22:45.24Vcoah crap now i have that song of homer singing "under the sea...." stuck in my head
22:45.38*** join/#asterisk zedkatuf (n=audela@82-33-115-145.cable.ubr08.azte.blueyonder.co.uk)
22:46.17hardwireVco: heh
22:46.21hardwireso happy I am not you
22:46.22jake1932FuzzyCat: true - I guess it all depends on why to restrict the access
22:46.29DrukenHMEjake1932: why you getting rid of it?
22:46.29DrukenHMEhehe
22:46.32hardwirewe kept having disconnected call issues here at the office
22:46.44hardwireturns out VXML_URL=intercom=true is bad if the snom 360 is on the line
22:46.47hardwireso
22:46.48hardwireblah
22:47.04ManxPowerhardwire, SendText does that with Polycoms
22:47.05jake1932DrukenHME: bought too many
22:47.18*** join/#asterisk iris_ (n=ypyg@c-65-96-4-250.hsd1.ma.comcast.net)
22:47.19hardwireManxPower: I had to write a script to do sip show channels first
22:47.32hardwireI think I am just going to write a manager daemon that helps w/ intercomming
22:47.37hardwireand does all the originates
22:47.41DrukenHMEjake1932: christ, i wish i had enough money to make a mistake like that
22:47.50hardwireas well as gets things like the snom fkeys to light/unlight
22:47.56iris_can anyone point me to where i can find some paid support for two quick and easy asterisk questions?
22:47.56hardwireI already got that one working
22:47.58ManxPowerwhy not just put SetVar= in the sip peer entry?
22:48.24hardwireManxPower: about intercomming?
22:48.32DrukenHMEiris_: what do ya need?
22:48.40jake1932DrukenHME: our client wanted minimal downtime while we moved a system over from another server
22:48.55jake1932DrukenHME: in a seperate location
22:49.07DrukenHMEjake1932: ahh, well, then that makes sence
22:49.08ManxPowerhardwire, SetVar=INTERCOM=howeveryouintercom
22:49.41hardwirein sip.conf?
22:49.42ManxPowerOr hell, SetVar=DEVICE_TYPE=SNOM or whatever, then use Gotoif in your dialplan do intercom differently depending on the device.
22:49.46iris_looking for a way to have a person ackknowledge a call when it is transfered to them (ie hit # to accept) and if not have it keep calling other extenstions. and how to setup an outbound dial ext (ie call into *, dial ext 1234 and enter the number you want to call) so we can keep the Callerid uniform
22:49.46*** part/#asterisk Cherebrum (i=jgarland@72.36.136.226)
22:50.05ManxPowerhardwire, unfortunatly, that only works for outgoing calls from the phone, but you could do other things for incoming
22:50.13hardwireManxPower: I gotcha
22:50.16iris_i know both are easy i just dont have the time at the moment to learn, needed them both last week :(
22:50.21Ariel_iris queues
22:50.23hardwireSipAddHeader=auto-answer is the best way to do that to snoms
22:50.30hardwireI don't wanna talk about it I guess
22:50.43hardwireever just feel that way? :)
22:50.47jake1932FuzzyCat: looks like a job for your time machine
22:51.08Ariel_iris_, for the outbound look at disa
22:51.12Ariel_~docs
22:51.14jbotdocs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
22:51.30FuzzyCatnothing is life is free jake1932  :P
22:51.30iris_ariel_ i tried queues but i dont want everyone to have o be dialed in all the time to take a call
22:51.35jake1932:)
22:52.04DrukenHMEiris_: so you set agents in the queues...
22:52.15DrukenHMEer.. members
22:52.30hardwirehttp://happy9.bogomip.com/albums/work/tdx/newoffice/photoalbum_view
22:52.31hardwirethere
22:52.32hardwiremy new office
22:52.36Ariel_iris_, you need to read the doc's on queues you can do that with agents for queues
22:52.39hardwireand new monitors
22:52.47iris_i dont understand how the queues etc work, thats why i was looking for paid support
22:53.30Ariel_iris_, did not know you were looking for paid support.
22:53.38Ariel_where are you located?
22:53.56ManxPowerUgh, I'm starving
22:53.59iris_ariel_ that was my first message when i joined :) I am in MA but dont care where the help is ;)
22:55.14*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
22:55.17Ariel_iris_, Ahh I must have come in after that message
22:55.46Ariel_iris_, can I pm you in about 15 minutes I am on a call but I would love to help you.
22:56.14DrukenHMEhehehe LOVE to help him
22:56.24DrukenHMEyou sound like a waiter
22:59.24*** join/#asterisk Defraz (n=t0tal@tim.ibccom.net)
22:59.45*** join/#asterisk zotz (n=zotz@24.231.36.100)
22:59.47DrukenHMEdoes vonage allow use of asterisk?
23:00.19xhelioxDruken: Just their soft phone package, which is not unlimited.
23:00.44*** join/#asterisk MrMAGO (n=maxgluck@200.109.166.83)
23:00.49sigwerkfuzzy: count me in, i need to switch
23:00.58sigwerker
23:01.36xhelioxI can't get away from you anywhere
23:01.37MrMAGOanyone knows how to implement phone booths? display on phone screen call duration and rate using asterisk?
23:02.01FuzzyCatsigterm, www.citizensrequired.com
23:02.55DrukenHMEMrMAGO: uhmm... what kinda payphones?
23:03.11DrukenHMEmost payphones have all that crap built in them
23:03.46MrMAGODrukenHME, well, haven't thought about it, but could be 7940s running some kind of XML
23:03.51xhelioxDoes someone make a VoIP pay phone?
23:03.58sigwerkfuzzycat: neat =)
23:04.16DrukenHMExheliox: i highly doubt it... hehehe
23:04.16sigwerklovely rather.
23:04.22FuzzyCat:)
23:04.28xhelioxThat would be neat if they did :)
23:04.42DrukenHMEi have an old payphone... but it needs a few parts to fix it.. :(
23:05.18Supaplexand you'll have to pay
23:05.33*** join/#asterisk omarc55 (n=omarc55@146.82.251.141)
23:05.34*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
23:06.18MrMAGOI want to build communication centers with phone booths, billed by cashier; each booth may have an IP phone. the important thing missing is the screen showing the rate, most phones by default show the time.
23:07.24DrukenHMEMrMAGO: oh.. well that's diffrent... :)
23:08.05MrMAGOyes, that's what I meant... not exactly like public phone booths
23:09.19*** join/#asterisk nitram (i=foo@superblob.com)
23:09.39MrMAGOwhere could I search for a placed call before it ends on asterisk, so the xml on the phone would calculate and rate the ongoing call?
23:10.38MrMAGOI really would like to do this project, so if anyone knows xml and could pull it out, please contact me.
23:11.28*** join/#asterisk WilliamK (n=wkeller@c-67-172-202-228.hsd1.tx.comcast.net)
23:11.54Corydon-wHeh, "knows xml"
23:15.01*** join/#asterisk sTone_heAd (n=rudy@2001:5c0:8fff:fffe:0:0:0:2061)
23:15.08*** part/#asterisk Uther_P (n=uther_p@66.180.120.82)
23:15.16CoriantumIs there any way to run a Macro from the manager interface by not using Originate?  Originate seems to need both legs of a call to work.
23:16.06*** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1)
23:17.26omarc55hi all, I am trying to use busydetect and callprogress to fix a problem that asterisk does not detect a hangup, but neither work, what can I check to see what I am doing wrong?
23:18.38*** join/#asterisk jayk- (i=jayk@vapid.reprehensible.net)
23:18.54MrMAGOCorydon-w: ?
23:19.14Corydon-wThat's like saying "anybody know csv?"
23:19.19Corydon-wxml is a file format
23:19.19*** join/#asterisk bjohnson_ (n=bjohnson@i216-58-62-220.cybersurf.com)
23:19.35jayk-is there anyway to speed up the voicemail operator? the operator pauses briefly between words as the sound files are played.
23:19.49MrMAGOno comment
23:20.04opus_Corydon - oej helped me last night with 5112 sorry for the squeky wheel post
23:21.09fiber0ptihow do you dial a non sip number with asterisk in a extensions.conf?
23:21.09*** join/#asterisk Uberbot (n=Uberbot@pcp01880954pcs.sandia01.nm.comcast.net)
23:21.21opus_what do you mean non sip number?
23:21.33fiber0ptiany telephone number in the united states
23:22.08Kattyhi lads.
23:22.12distortioyou need a provider that has a connection to the pstn
23:22.23fiber0ptidistortio: i have one
23:22.25*** join/#asterisk toddf (n=toddf@ns0.fries.net)
23:22.41distortiothen connect sip to them, and they should be able to handle the rest
23:22.54fiber0ptito "them"?
23:23.03fiber0ptito my proivder?
23:23.06distortioyes
23:23.09jayk-anybody have any ideas on speeding up the voicemail operator?
23:23.09*** join/#asterisk Koshatul (n=evangeli@ip157.net65.ipnetworks.net.au)
23:23.10fiber0ptialready done
23:23.23fiber0ptiI'm trying to figure out how to have the dialplan dial a number
23:23.25fiber0ptiis that possible?
23:23.39opus_jaky- maybe shorten up the voice over?
23:23.41jayk-do you mean a rewrite, fiber0opti?
23:23.45Kattyi just love how no one says hi.
23:23.47jayk-opus_: what do you mean?
23:24.08fiber0ptijayk: I don't know what a rewrite is so I couldn't tell you
23:24.32distortiofiber: exten => _1XXXXXXXXXX,1,Dial(SIP/[carrier in sip.conf]/${EXTEN})
23:24.55Katty:<
23:24.57opus_jayk-,  you can also add a few configuration lines to voicemail.conf to disable/enable features. if you disabled features perhaps it would go quicker. what exactly do you need?
23:25.43opus_fiber0pti - you want the Manager API Originate , but do Dial plan things?
23:26.03jayk-opus_: well, for instance, when the operator reads the date, she says Monday <semi short pause> 3 <semi short pause> 15 <semi short pause> pm <semi short pause>
23:26.05jayk-follow me?
23:26.13fiber0ptidistortio: how do I tie that with a button being pressed on the dial plan. For example: if the user presses the 0 button how can I have asterisk dial 555555555?
23:26.33opus_jayk- well, hmm. I think you can turn off reading the date
23:26.52jayk-well, its not that though, its when the files are being played through the sound device..there's a bit of delay between files
23:26.58jayk-so i'm wondering if there's a way to speed that part up
23:27.15opus_jayk- but essentially asterisk is just playing .gsm files of each number -- you could load those files up into an audio editor and shorten them up.
23:27.28distortiofiber: exten => _0,1,Dial(SIP/[carrier in sip.conf]/5555555555)
23:27.33jayk-like when i park a call, the operator spits back Caller parked on 6 <pause> 0 <pause> 1 <pause> for parked extension 601
23:27.40opus_jayk- OH and if your hardware is slow, you will definitely get a slow down between .gsm files
23:27.41fiber0ptidistortio: thanks!
23:27.49Vcosomeone should really make a dialplan generator that you can just dictate abstract concepts to and it will figure out what the hell you are talking about..
23:27.56jayk-opus_: yeah, shouldnt' be slow..i think its a 2ghz with 512mb of ram
23:28.04Vcocuz i'm starting to confuse myself ......
23:28.08*** mode/#asterisk [-o bkw_] by bkw_
23:29.05jayk-yeah, i suppose it could be the gsm files..maybe they don't cut off exactly at the end of the last word?
23:29.20opus_yes, exactly.
23:29.25jayk-that'd be too much work
23:29.26jayk-heh
23:29.28opus_with a good audio editor you can see it
23:29.32opus_jayk- order a pizza! :)
23:29.42jayk-i'd rather order a case of beer and forget about it
23:29.43jayk-;)
23:29.44*** join/#asterisk bendy24_ (n=slb@50.tender.1meg.golden.net)
23:29.46opus_haha
23:30.02opus_ok, you could do this.
23:30.31opus_write a script that calculate every possible date/time voice over, then speed up each file. you could write a script to do all of this, and still drink your beer
23:30.50Kattygosh.
23:31.03jayk-too much thinking
23:31.09jayk-:)
23:31.30ManxPowerI'm starting to worry that the MIS manager had a heart attack, he's MIA
23:31.31jayk-the biggest complain we have with our new asterisk phone system is our slow operator
23:31.43jayk-people were used to the smoothness of the old operator on our nbx system
23:31.53_meppl_gute nacht
23:31.57pauldy2just fire her and get a new one
23:32.05bendy24_i had once found a good forum for various phones with asterisk... but now google isnt giving me any love.
23:32.08opus_weird. People will complain about anything now days.
23:32.15bendy24_does anyone have a nice url for me?
23:32.23jayk-should i be using alsa or oss?
23:32.26jayk-maybe that has something to do with it
23:33.16Vcosamuel l jackson would be a really good voice actor for pbx stuff......
23:34.11distortio"HIT # ONE MORE TIME MUTHA F&*&%! I DARE YOU"
23:34.17Vcoheh..heh..
23:34.48*** join/#asterisk nesys (n=nesys@81-174-12-111.f5.ngi.it)
23:34.49opus_haha
23:34.49VcoWHO'S THE SMOOTH OPERATOR NOW MUTHA F&&&A?
23:34.51*** join/#asterisk bjohnson__ (n=bjohnson@i216-58-18-181.cybersurf.com)
23:34.53Ayanobendy24; what phone?
23:35.12bendy24_Ayano: cisco 7905g
23:35.15jayk-lol
23:35.32Ariel_Polycom, Polycoms, Polycoms
23:35.39bendy24_bah
23:35.41*** join/#asterisk BuggyBuddy (n=hello_vi@web1.ningo.net)
23:35.45Ariel_oh did I say polycoms
23:35.49*** join/#asterisk IronHelixz (n=irc@ool-45785cfe.dyn.optonline.net)
23:35.54VcoPolyColonic?
23:36.04bendy24_polyshmom
23:36.16opus_somebody told me Polycoms are actually manufactured at Cisco
23:36.29opus_but I've never been able to verify this
23:36.35*** part/#asterisk SwK (n=SwK@12-219-144-126.client.mchsi.com)
23:36.38Ariel_opus_, not that way but it's possible the other way around
23:36.46Vcosomeone told me they needed change to ge on a bus.........
23:37.11opus_well, i remember cisco IP phones back in 2001
23:37.15*** join/#asterisk da_monumental_1 (n=da_monum@rrcs-24-172-102-198.midsouth.biz.rr.com)
23:37.18Ayanolol.  we could make a forum for it lol
23:37.39Ariel_the debate forum for phones
23:37.56bendy24_who cares?  i just need some url goodness
23:38.07AyanoI'm going to set it up,  I'll post a link in a little bit. lol
23:38.24bendy24_k
23:38.35Ayanobendy24_; you can send the phone to me, and I'll figure it out and send it back.
23:38.35Ayano?
23:38.43jayk-i like the cisco 7960's we have
23:38.47jayk-i have no experience with polycom
23:39.26opus_i just went to the red cross building... guess what i saw, everyone had 7960's
23:39.46Ariel_bendy24_, http://www.voipsupply.com/index.php?cPath=95
23:40.39Ariel_Cisco's are very much into putting them where people can see them.
23:41.15jayk-i saw cisco's at greg's office on dharma and greg
23:41.16jayk-:)
23:41.46Ariel_I used to use Cisco's but Since I don't agree with there way of Licence's I don't use them any more.
23:41.50bendy24_trust me, if i had to spend money i would have boughten a polycom...
23:41.59bkw_WOOOHOOOO i'm freee!!!!!
23:41.59bendy24_but since this cisco was free...
23:42.14Ariel_bendy24_, so what do you need firmware?
23:42.25bendy24_Ariel_: have that already
23:42.37Ayanopolycoms are cheaper.
23:42.53bendy24_Ariel_: i haev sip.conf setup... just trying to figure out the cisco side
23:43.17bendy24_Ariel_: i had seen a nice example on what settings to use where, now i lost the url
23:43.42Ariel_ahh your trying to set the Cisco up.
23:43.48bendy24_ja
23:44.05bendy24_:)
23:44.46Ariel_bendy24_, this has some links that might help: http://www.voip-info.org/tiki-index.php?page=Setup%20SiP%20on%207940%20-%207960
23:44.56CoriantumIs there any way to run a Macro from the manager interface by not using Originate?  Originate seems to need both legs of a call to work.
23:45.50Ariel_Coriantum, why? and what actually are you trying to do?
23:45.54bendy24_Ariel_: thanks , but not quite specific for the 7905g
23:45.54shmaltzI'm trying to create a sip entry in Asterisk box 1 to connect to Asteisk box 2, asterisk1 is not behind nat, and asterisk2 is, so the entry on box1 for box2 has host=dynamic, and box2 has a register line which works fine
23:46.01*** join/#asterisk Rowters (n=SilverDr@dsl-201-129-89-96.prod-infinitum.com.mx)
23:46.26Rowtersits possible to have a TE205P on a 1U, raiser could suport 5.0V?
23:46.35bendy24_anyway, i had once found a nice forum for all types of phones, just thoguht i would ask
23:46.39shmaltzI want sip user a on ast1 to be able to dial exten something on ast2, but I get error on box2 that user a is not known
23:47.09CoriantumAriel_: I'm trying to make a call that plays a message and records their response... just a single leg.
23:47.30*** join/#asterisk SwK (n=SwK@12-219-144-126.client.mchsi.com)
23:47.31Ariel_Coriantum, call file?
23:47.47ManxPowerPolycom!  Polycom!  Polycom!
23:47.56Ariel_Coriantum, look at the wake up call section of the wiki
23:47.56CoriantumAriel_: It needs to initiate from a web server not near the asterisk box.
23:48.01BeirdoManxPower, useful as usual :)
23:48.10*** join/#asterisk Gronker__ (n=Gronker2@adsl-220-75-62.ags.bellsouth.net)
23:48.10Ariel_Coriantum, you setup a call file
23:48.17*** part/#asterisk Gronker__ (n=Gronker2@adsl-220-75-62.ags.bellsouth.net)
23:48.20Ariel_ManxPower, I said that
23:48.44CoriantumAriel_:  So basically... I can't standardize on the manager
23:48.54*** join/#asterisk Equinox (n=secret@pool-71-243-231-201.tampfl.fios.verizon.net)
23:48.59CoriantumI already know how to do it with call files...
23:49.14EquinoxHow can I forward calls to Sipura 3000 FXO from asterisk?  I have the 3000 registering and PSTN->VOIP working.. Just not sure how to do the dial plan
23:49.17Ariel_Coriantum, your best bet for that is drop a call file for asterisk to make the call for you.
23:50.10CoriantumAriel_: Yeah... I don't want to generate call files on the web server or execute external command from the web server on the asterisk box.
23:50.14Ariel_Equinox, exten => X.,1,dial(sip/sipura300,${EXTEN},20)
23:50.32EquinoxAhhhh. So I put the destination number after a comma.. Got it :)
23:50.59Ariel_Coriantum, you just generate a txt file and drop it into a directory. Asterisk then picks it up
23:51.02EquinoxAriel_ - The newest firmware makes ringing VoIP from PSTN without an answer a snap :)
23:51.09CoriantumAriel_: I know how to do call files
23:51.21CoriantumAriel_: I want to keep the call logic OFF of the web server
23:51.26Ariel_Equinox, how
23:51.59EquinoxAriel_ -- There are simple firmware options not to answer now.. no callerID "A" tricks and crap
23:52.16Ariel_Equinox, really I need that
23:52.30EquinoxAriel- -- Download it :)  Lemme find what options to use.. All the docs are on the old kludge way
23:53.01CoriantumSo basically... the manager interface is crap
23:53.24Ariel_Equinox, I have that verision 3.1.5 already
23:53.33Ariel_I guess I need to setup the options now.
23:53.36EquinoxSet your dial plan- (S0<:blah>)
23:53.57EquinoxAriel_ - Where 'blah' is the extension to ring in the sip context the PSTN line is registered in
23:54.21Ariel_like from-pstn
23:55.07EquinoxYep
23:55.28EquinoxI think the default is not to answer it
23:55.34EquinoxSo it only answers when the SIP extension answers
23:55.35Ariel_(S0<:s>)
23:55.48EquinoxMake sure it's enabled
23:56.02Ariel_dial plan enable where?
23:56.13EquinoxOff hook while calling voip: No
23:56.32EquinoxVoIP-To-PSTN Gateway Enable: Yes
23:56.51EquinoxPSTN CID For VoIP CID: Yes
23:56.53*** part/#asterisk da_monumental_1 (n=da_monum@rrcs-24-172-102-198.midsouth.biz.rr.com)
23:57.37*** join/#asterisk tq1 (n=tq@200.117.234.254)
23:57.59*** join/#asterisk JunK-Y (n=junky@Toronto-HSE-ppp3743074.sympatico.ca)
23:58.15EquinoxWith the DIAL(SIP/sipurassip,phonenumber,20) I'm getting voice mail for the PSTN line.. Like it's dialing itself or something?
23:58.38*** join/#asterisk elriah (n=jfulcrum@adsl-068-209-198-242.sip.bhm.bellsouth.net)
23:58.54*** join/#asterisk devonst17 (n=devonst1@dsl092-032-215.lax1.dsl.speakeasy.net)
23:58.57EquinoxAriel_ - All those options were under "PSTN Line" Advanced
23:59.33elriahHi all.  I have a SIP phone that sits behind NAT and a cable modem.  It works fine for dialing out.  I can receive calls for a short amount of time when it first reboots.  Then something happens and * can't reach it anymore.  Any suggestions where to start?
23:59.37bendy24_AHA!
23:59.48bendy24_voxilla was what i was looking for

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