irclog2html for #asterisk on 20061006

00:00.24dovidraidenz: try using the help function in CLI. i am not playing with 1.4 till all issues are fixed and wiki is updated. dont have time to play to run in to issues
00:00.47*** part/#asterisk diclophis-work (n=jbardin@65.203.37.58)
00:01.38raidenzI did and I saw the help on the function but still doesn't help.
00:02.10*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
00:02.10*** mode/#asterisk [+o mog] by ChanServ
00:02.23dovidwur kind of phone r u using ?
00:02.55dovidwut*
00:08.15teknoprepeverything i am readying say do this
00:08.16teknoprepparkedhints=yes
00:08.20teknoprepwhere do i put that ?
00:08.22teknoprepon the extension ?
00:09.57*** join/#asterisk tuxd00d (n=tuxinato@128.187.161.255)
00:13.40dovidagain what phone r u uisng ?
00:15.24C6Vetteguess its a secret.
00:16.21*** join/#asterisk docelmo (n=vircuser@m015f36d0.tmodns.net)
00:17.18C6Vettewhen doing a "show queue whatever"... How can i tell hold long its been since the received a call. It shows a  (last was 30 secs ago) but
00:17.24C6Vette<PROTECTED>
00:17.37C6Vettewhen they are not on a call
00:18.01teknoprepdovid for this extension its a gxp-2000
00:19.19dovidah. didnt play much with those. sorry. they arent worth the time of day.
00:19.51dovidc6Vette: dont know much with queues. try help command or wiki
00:20.13diablopicocan anyone help me with an insmod command for the te212p ?
00:20.55teknoprepto patch asterisk.. do i have to fully recompile it ?
00:21.41diablopicowhen you make , it should only remake any files that change
00:22.12diablopicoand of course those dependant on the changed files
00:27.01*** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com)
00:28.49*** part/#asterisk cbm11211 (n=Administ@66.28.182.170)
00:34.26*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
00:34.36apturalong day.
00:34.52*** join/#asterisk cekc (n=cekc@66-17-9-220.biz.bkfd.arrival.net)
00:35.13kronicis there a setting required for the agent function to return values?
00:35.42kronicI have an agent logged in, attempting to return values from the ${AGENT} function yields nothing :S
00:35.57apturaDo most pbxs use the same numbering sceem be it accessing vm or call forwarding or is that a customer choice? Example one pbx may use *98 to access its voice mail vs another pbx uses *67
00:39.16justinu|laptophttp://www.nanpa.com/number_resource_info/vsc_definitions.html
00:41.15justinu|laptopare there any SIP phones that support DND override?
00:41.43dovidby dnd over ride u mean ?
00:42.02justinu|laptopi mean if someone sets their phone in DND, I should be able to make it ring anyways
00:42.14apturaThanks
00:42.16justinu|laptopwith a proper passcode or some other authentication
00:42.37apturafor example a familly emergency
00:42.54justinu|laptopfor example: i'm the pbx admin, and i don't care whether you want to be disturbed or not
00:42.59justinu|laptops/pbx admin/boss
00:43.52apturaUnless your in a very important meeting or involved in a presentation.
00:43.54Hymiewell, >I can tell you the uniden UIP/200 won't be doing that any time soon :Þ
00:44.24justinu|laptopthis is yet another reason why the phone should just be a dumb device, ala SCCP
00:44.34justinu|laptopnot some pseudo intelligent network element, like SIP
00:44.59Hymiejust implement asterisk do not disturb, via dialplan logic and database variables
00:45.11Hymiethen that can be over ridden as you choose
00:45.17justinu|laptopthat doesn't prevent someone from placing their SIP phone in DND
00:45.20justinu|laptopwhich you can't override
00:45.33Hymieit does, if you turn that option off on the phone.. what phone is it?
00:45.44HymieI can disable that in the uniden UIP200, and that's a crappy phone ;)
00:46.10justinu|laptopwell, basically I like the idea of DND... i just think there should be a way to override it if needed
00:46.24HymieI just told you how
00:46.30dovidjustinu|laptop: i believe not because asterisk is told to not let anything get to it
00:47.00justinu|laptophaving to turn off the DND button on the phone isn't my idea of a solution
00:47.02dovidunless u create a seperate extension that they have to authenticate and from ur dial plan u change thier dnd status but then they will get all other calls.
00:47.36*** part/#asterisk r_evolution (i=r_evolut@208.251.203.246)
00:48.05Hymiejustinu: SIP phones won't ring when the DND button is on.. Asterisk can't do anything about that. You said you wanted a dumb phone, but you don't want to disable the DND button on the phone.... not quite getting you here
00:48.08justinu|laptopthe problem is that the DND status is something the phone can set without having to check with the switch to see if its allowed to set DND status
00:48.18dovidor create two extensions that go to one sip account and have them dial an exten to put them at unavail and when the gen exten is called it checks a variable id they wana be botherd
00:48.35justinu|laptopwhat I meant was that I want a phone which works with a protocol like SCCP
00:48.38Hymiedovid: that won't work
00:48.51*** join/#asterisk Malawar (n=Malawar@adsl-75-21-232-9.dsl.sgnwmi.sbcglobal.net)
00:48.54justinu|laptopwhere when ppl press buttons on the phone, it sends messages to the softswitch, and the softswitch decides how the phone behaves
00:48.55Hymiedovid: when the phone has DND on, it won't ring.. period
00:48.58dovidHymie:  y not ?
00:49.17justinu|laptopyeah, that's a defect in the design of the phone, IMO
00:49.19dovidHyme: for his/her case i wouldnt use dnd
00:49.29dovidI would have this
00:49.36Hymiejustinu: hey, I'm not saying other ideas aren't good, but I'm not interested in that right now, >I have to deal with what I have
00:49.52dovidi would use a gotoif to see if they want to be botherd
00:49.54justinu|laptopi understand where you're coming from
00:49.56dovidfor thier reg exten.
00:50.12justinu|laptopit's not something a SIP phone couldn't do
00:50.14dovidand have the user dial en exten to set the variable
00:50.15MalawarI need.. a business friendly VOIP provider
00:50.16justinu|laptopallow DND override
00:50.26justinu|laptopif we had an opensource SIP phone, we could easily implement it
00:50.31dovidMalwar: teliac isnt the cheapest but good
00:50.48dovidteliax*
00:51.06MalawarIt doesn't have to be the cheapest, just reliable, and not some company that's going to turn around and charge extra/cut me off for using too much call volume (coughbroadvoicecough)
00:51.27justinu|laptopdovid: i saw it, but you're kinda missing the point
00:51.44dovidok. explain
00:51.47justinu|laptopthe problem is that with most sip phones, the user can press DND, and the phone will never ring again
00:51.51justinu|laptopunless they disable DND
00:52.09justinu|laptopwell, then these friggen morons who press DND by accident, and never notice the display flashing "DO NOT DISTURB" call you up
00:52.13justinu|laptopand bitch to you their phone never rings
00:52.26justinu|laptopand that /I/ obviously broke something
00:53.04dovidlol
00:53.06dovidi got that too
00:53.11justinu|laptopyou see my point now?
00:53.12dovidso just disable dnd on the phone
00:53.19dovidand make em dial the exten
00:53.52justinu|laptopagain, i don't see that as a solution
00:54.08justinu|laptopbecause then I have to train them what exten to dial, and the damn phone has a DND button... why should I disable it?
00:55.14dovidand u cant disable it on the phone its self ?
00:55.29dovidand if its an idiot issue then ur SOL people goto learn
00:56.03justinu|laptopi wonder if SIP 3PCC might allow it
00:56.14Malawarhmm
00:56.22Malawarwhat I really need, is an incoming-only line
00:59.43apturadovid, reliability should be the most important feature.
01:00.26apturaa customer or client should not have to give there pbx a second thought when making that call.
01:01.26dovidaptura: what point are u talkin about ? it is reliable to an extent. i trust it
01:02.25apturaI am talking about voip carriers
01:02.46dovidah ok
01:02.56*** join/#asterisk cekc (n=cekc@adsl-71-131-128-183.dsl.sntc01.pacbell.net)
01:03.11kronicwhat's the best method for obtaining an agents name?
01:03.12dovidapturs: thats y i have failover routes. if route A is out then use Route B
01:03.22kronicsince the agent func is non-existent
01:03.30dovidvoip just isnt as relaible as pots and i dont think it ever will in the near future
01:03.35apturaI would rather charge my customers a little more because my reputation is counting on the carriers quality.
01:03.38justinu|laptopthis is interesting: http://www.ietf.org/rfc/rfc3880.txt
01:03.41dovidsame here
01:03.41justinu|laptopwonder if anything implements that
01:04.28dovidthose are too long for me to read. can u sum it upo ?
01:04.29dovidup*
01:04.30justinu|laptopheh, someone posted on asterisk-biz about it
01:04.31apturabut
01:04.36justinu|laptopbut no one replied
01:04.56apturavoip is a good substitute should a local telco go down...like in new orelans.
01:05.07Malawaryarr, all these other providers seem to want to stuff their hardware down my throat as well as a service plan :(
01:05.43apturaWhich providers?
01:05.43Malawari'm just looking through them atm
01:05.45Malawarseveral so far
01:05.49dovidi use myphonecompany for inbound and voipjet and teliax for out
01:06.27Malawarall I need is inbound, but I need to be able to handle multiple calls concurrently
01:06.49apturadovid, how many Minutes
01:06.50aptura<PROTECTED>
01:08.13apturalooking at this call forward bussy query dont know what it means. ; query
01:08.14apturaexten => *45*,1,Playback(call-fwd-on-busy)
01:08.14apturaexten => *45*,2,Set(temp=${DB(CFB/${CALLERIDNUM})})  ; Get CFB key
01:08.39apturaIs it quering if its set or not set?
01:09.28dovidyes
01:09.39dovidapturs: it sets the status to the vairable temp
01:10.13dovidaptura: myphonecompany gives unlimited inbound. however if u hog multiple channels all the time they may get pissy
01:10.29apturaof course
01:10.30aptura:)
01:10.43dovidits $5.00 for inbound. I once was able to get 9 concurent channels. (it was the most cell phones we had in the office)
01:11.04apturalike a netmeeting?
01:11.11apturawhat were you trying to do
01:11.20dovidi did one client that needed 4 concurent so they said to just sign up for 2 lines so it costs $10.00 a month and they can use 4 concurent with no sweat
01:11.33dovidi was just testing to see how many channels they would give me
01:11.38*** join/#asterisk dir (n=dir@58.69.13.29)
01:11.38aptura5 dollars unlimited right?
01:11.46dovidyes. just inbound
01:11.55apturaThats pretty good.
01:11.56MalawarI don't see that plan on their site :/
01:12.03dovidthan again if u hog it they wont be happy like the avg. provider
01:12.09dovidfor some reason its not on the sire
01:12.10apturaWhats the reliability of the voice like
01:12.13dovidsite*
01:12.21dovidits called a mydeviceplan
01:13.28dovidu have to email cc@myphonecompany.com
01:13.28justinu|laptop~seen [TK]D-Fender
01:13.36jbot[tk]d-fender is currently on #asterisk (4h 23m 45s). Has said a total of 50 messages. Is idling for 3h 2m 49s, last said: 'ok, BBIAB'.
01:13.36apturaokay what is the quality of the call like
01:13.36dovidi have been with them for a year and a half and they were down once for 20 min
01:13.36dovidotherwise no issues with them
01:13.36dovidthey are a brnach of exchange telecom
01:13.36apturaand Quality of the call? as in clear or garbled?
01:13.45apturaor jitter?
01:13.45dovidreal clear
01:13.52apturaWhere is there noc at that does the termination?
01:14.00dovidi believe in NYC
01:14.10apturavoipjet is there also
01:14.16apturaAnd you are?
01:14.26apturawhere are you that is
01:14.40dovidjust do a who is on thier IP's and u can see where they are
01:14.46dovidwhere am i ? in NJ and Israel
01:14.49apturaI know
01:15.01dovidu can ask them
01:15.09dovidi deal with some one there by the name of sam litt
01:15.12apturawhen you make the calls you get the incoming calls you are in nj right?
01:15.33dovidno. it hits my dedicated server on the west coast
01:15.56dovidwhich uses teliax to call my cell phone being that I am in Israel now
01:16.15*** join/#asterisk tuxd00d (n=tuxinato@128.187.191.179)
01:16.18dovidthier outbound rates arent the best for US48
01:16.27apturaI am just trying to picture the distance from where you are to myphonecompany noc is.
01:16.39apturahow far the rtp traffic has to flow.
01:16.40dovidill to a tracert
01:16.41dovidone sec
01:17.33apturanot really interested in ping statistics
01:18.15apturaPeople on the west coast..well some of them company about call quality comming from vonage which is east coast. I suspect with multiple router hops and traffic that jitter would be a issue.
01:19.25dovidhere u go
01:19.26dovidhttp://pastebin.ca/192567
01:19.38dovidhas never been for me
01:20.20dovidi would suggest u try a few providers and do the "can u hear me now" test
01:20.29dovidgoto for a smoke
01:20.30dovidbrb
01:21.13apturaI have worked with a number over two years
01:23.24dovidback
01:23.29dovida number of ?
01:23.48Malawaranyone know of any other inbound-only services? Just shot off an email to myphonecompany
01:23.49Malawar:P
01:23.57apturavoip providers
01:24.24dovidMalwar: teliax is good but they charge per minute
01:24.25quid246not many inbound-only providers.
01:24.30apturasixtel has improved and is working for me.
01:24.53apturaohh inbound only well sixtel does both.
01:25.08dovidu dont have to use em for both ways
01:25.11Malawarthe provider doesn't have to be inbound only, just looking for an inbound-only plan :P
01:25.14dovid<PROTECTED>
01:25.33dovidMPC is 2 way just set up asterisk to only use em for inbound
01:25.38quid246Axvoice has unlimited in, AFAIK... $10 for 2 DIDs.
01:26.00quid246They even have 212's last time I looked.
01:26.21dovidwow. i like 212
01:26.26dovidaxvoice.com ?
01:26.29quid246yep
01:27.03*** join/#asterisk Yarrick40k (n=jmoser@static-71-98-31-11.mdsnwi.dsl-w.verizon.net)
01:27.14quid246My 212 actually is phonetic... took me an hour or two seraching through their list to find it htough
01:27.27*** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net)
01:27.32dovidall i see on axvoice is 18.99 a month. none cheaper
01:27.36quid246BYOD
01:27.39quid246Bring Your Own Device
01:28.08quid246Pay As You go... onle $9 now
01:28.11doviddidnt see byod. looking again
01:28.17dovidokies
01:28.39dovidhow many channels do they give u ?
01:28.40MalawarI'm also trying to avoid places like BroadVoice that have scary Terms of Service
01:28.45dovidlol
01:28.50dovidi would never use em again
01:28.56DonXAnyone use ngt?
01:28.58dovidthey can keep ur number if u port in
01:29.09dovidcall quality sux with them
01:29.10quid246haha nice
01:29.11dovidngt ?
01:29.32DonXyah, www.ngt.com
01:29.32Malawarat first glance Broadvoice was very nice looking.. but their ToS states they can charge $100/day if you're using the service in a way they arbitrarily deem unfit :/
01:29.44DonXthey have onvoip.net
01:30.09quid246Malawar: Adn don't forget the clause for "in house legal fees".
01:30.14Malawarquid246, exactly
01:30.34Malawarscary :/
01:30.38Jason99Malawar: What area are you look for numbers in?
01:31.46Malawarnot really picky, but something within the state of NY would be nice :P
01:31.56dovidu can also try voxbone.com
01:32.25DonXHas anyone ever used Broadworks?
01:32.55*** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net)
01:33.05*** join/#asterisk b4ka (n=jh@71-226-114-200.fibertel.com.ar)
01:33.10Jason99Malawar: just checking, we're in Canada only.
01:33.28b4kahey, im a bit of a problem, trying to make an asterisk talk to a strata dk424
01:33.36b4kait works from time to time
01:33.49b4kabut mostly not, complaining about no dchan
01:33.54b4kaanyone has a clue?
01:34.23b4kaits a t1 crossover cable to a sangoma A102
01:36.17justinu|laptopb4ka: one side needs to use internal clock, otherside needs to be clocked from the loop
01:38.48b4kajustinu|laptop it is
01:39.01b4kathe telco is in master on the sterisk
01:39.09b4kaand i server as master to the pbx
01:39.17b4kaserve*
01:39.55b4kaand its weird, now it is working flawlessly, if i disconnect the cable and plug it back, it all goes to hell
01:42.47apturajustinu|laptop ever seen compatibility between a nortel mcs 5200 media server and asterisk? I cannot use a company as my termination service because there may be timming issues. the mcs is a 2 million dollar class IV switch.
01:43.41justinu|laptopshouldn't be a problem
01:43.56justinu|laptopnot T1 timing at least
01:44.21*** join/#asterisk cekc (n=cekc@adsl-71-131-128-183.dsl.sntc01.pacbell.net)
01:45.04apturawell there engineer did try in the past and the voice exchange between two parties would drop off every 15 min.
01:45.09*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
01:45.13justinu|laptop* connects to the DMS all the time
01:45.24justinu|laptopbig brother to their mcs :)
01:45.56justinu|laptopaptura: talking about pri link?
01:46.24justinu|laptopasterisk periodically restarts the B channels, that might piss off some switches
01:46.28justinu|laptopyou can disable it
01:46.28apturayes
01:46.45apturaWhere would it be disabled?
01:46.52justinu|laptopzapata.conf i think
01:47.06apturaokay
01:47.48apturaIs the restarting not a option or the code needs to be changed?
01:48.02justinu|laptopits an option, iirc
01:48.07apturaI see
01:59.48kronicwhat's the best method for obtaining an agents name (${AGENT} does is non-existent)?
02:13.19*** join/#asterisk freebsd_fan (n=unsure@catagiuri305.giuri.unige.it)
02:13.34*** join/#asterisk nvrs (n=RUR@Quebec-HSE-ppp3613717.sympatico.ca)
02:15.01*** join/#asterisk fnordus (n=dnall@24.85.128.203)
02:18.13*** join/#asterisk Splas (n=jwb@brooklyn.paravolve.net)
02:18.48dovidseen ~me
02:18.58dovidseen ~jbot
02:20.06b4kaoh now, i found the way to make the t1 between the strata and asterisk to work. i have to plug the cable SLOW LOL
02:20.35Qwellb4ka: get a new cable
02:20.45b4katried 3
02:20.58Qwellpebcac?
02:21.15ShadowHntrstupid question
02:21.15Qwell(I haven't decided on what the c's stand for yet)
02:21.21b4kacould it be that the signal is too strong?
02:21.36ShadowHntrif i set up asterisk in a strictly SIP setup with SIP phones, do i need a digium card?
02:21.40b4kaand it makes ground before stablishing the connection
02:21.48QwellShadowHntr: no
02:22.06ShadowHntrcool. cause i read about the digium cards, and i knew *what* they were, just didn't know why i'd need one.
02:22.10ShadowHntrQwell: thanks. :)
02:23.08*** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com)
02:23.53dovidShadowHntr: see: http://www.voip-info.org/wiki-Asterisk+timer+ztdummy
02:24.09ShadowHntryeah i've been reading that wiki
02:24.17ShadowHntrjust been trying to make heads or tails of some of it.
02:24.53*** join/#asterisk japerry (n=falc0n@216.231.51.209)
02:25.12*** join/#asterisk angom_h (n=Angel@www.aysco.com.mx)
02:25.27*** join/#asterisk murf (n=steve_mu@216.166.159.235)
02:31.02dovidShadowHntr: the wiki is the king. and remember google is ur friend
02:31.28ShadowHntryeah i've kinda discovered that it may be easier to avoid the cisco phones. too much hassle to configure for a first-time voip user.
02:31.42ShadowHntralthough i am a highly proficient linux admin and a network guru
02:31.50ShadowHntrjust learning as i go
02:31.55ShadowHntrappreciate the pointers. :)
02:32.16*** join/#asterisk michaelo (n=michaelo@adsl-226-208-109.gsp.bellsouth.net)
02:33.27*** join/#asterisk wulfy814 (n=wulfy814@c-67-165-37-20.hsd1.pa.comcast.net)
02:37.25*** join/#asterisk fnordus (n=dnall@24.85.128.203)
02:38.20*** join/#asterisk aptura (n=none@S010600a0c93f6f7e.vs.shawcable.net)
02:41.51*** join/#asterisk fnordus (n=dnall@24.85.128.203)
02:44.23wulfy814if I have just changed the echo cancellor and recompiled zaptel
02:44.32wulfy814do I need to reboot for the change to take effect?
02:44.49dovidno
02:44.55dovidjust stop and start asterisk
02:45.21dovidwell if u just recompiled then it isnt running so just start asterisk
02:46.06wulfy814let me rephrase...
02:46.21wulfy814I just edited zconfig.h and did a make, make install
02:46.25wulfy814asterisk was running at the time
02:46.46wulfy814I changed to Mark3
02:47.18*** join/#asterisk AJaymn (n=me@24-159-236-181.dhcp.mdsn.wi.charter.com)
02:48.24*** join/#asterisk [Outcast] (n=bill@222-154-75-119.jetstream.xtra.co.nz)
02:48.54heison~seen coppice
02:49.07jbotcoppice <n=chatzill@229.166.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 1d 9h 25m 18s ago, saying: 'I get weird reports from anything after 1.2.9 about NULL frames being thrown at rxfax, which cause a crash when they are freed'.
02:52.07*** join/#asterisk fnordus (n=dnall@24.85.128.203)
02:55.03*** join/#asterisk Joel1978 (n=Joel1978@12-226-85-195.client.mchsi.com)
02:55.56dovidstop asterisk. make and restart
02:57.17b4kaanyone knows how to send a restart to all the B channels on a pri line from asterisk?
02:57.36Yarrick40kanyone here using a Cisco 7940 with SIP firmware?
03:02.36*** join/#asterisk sudhir492 (n=sudhir@leesburg-bsr3-68-65-168-202.chvlva.adelphia.net)
03:03.32apturab4ka whats the issues?
03:09.34sudhir492Will a kind soul please help me configure Polycom 301 phones for paging
03:09.44sudhir492I mean Polycom 501
03:11.11*** join/#asterisk xxoxx (n=xxoxxx@tor/regular/xxoxx)
03:12.25[TK]D-Fendersudhir492 : PM me root and I'll get you set up.
03:13.53aptura[TK]D-Fender I would also like to know
03:15.33*** join/#asterisk linuxmigration (n=jeremy@dsl254-075-124.nyc1.dsl.speakeasy.net)
03:16.32*** join/#asterisk AJaymn (n=me@24-159-236-181.dhcp.mdsn.wi.charter.com)
03:17.23AJaymnAnyone know of a good reference for using the sound files. as in how to have Allison speak #s like "Your Account Balance is" "1 hundred
03:17.30AJaymn" and 24 dollars"
03:17.37*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
03:19.50b4kaaptura: i cant friggin make asterisk to talk to the strata through the pri properly
03:20.04apturastrata?
03:20.11b4katoshiba strat
03:20.13b4kapbx
03:20.19[TK]D-Fender*shudder*  FUGLY phones....
03:20.22b4kasome dude documented it working on the wiki
03:20.33b4kabut i cant fucking make it work
03:20.38apturanever even heard of them
03:20.51b4kathey are very common
03:20.55b4kasmall pbxs
03:21.45b4kathis is so retarded, why the fuck connecting the cable slowly makes it work!
03:26.06*** join/#asterisk xpato (n=pato@pc-33-21-104-200.cm.vtr.net)
03:26.22wmandraYarrick40k: I'm using 7940's and 7960's with sip firmware
03:26.41xpatohas anyone worked with the sipura spa3000?
03:27.12Yarrick40kwmandra:  I think I got it worked out.  Have to do firmware updates in stages apparently
03:28.04wmandrayarrick: just follow the instructions on the wiki..... they'll get you through it
03:28.49Yarrick40kwmandra: But winging it is so much more fun!
03:28.50Yarrick40k;)
03:28.57wmandralol
03:29.26Yarrick40kthere a way to reboot a 7940 without jackin the power cord?
03:29.33QwellYarrick40k: sip or sccp?
03:29.38Yarrick40ksip
03:29.46Qwell* + 6 + settings
03:30.22Yarrick40kthanks
03:31.02[TK]D-FenderQwell++
03:31.09Qwell?
03:31.25[TK]D-FenderQwell : Jbot rating thing....
03:31.30Qwellright
03:31.46[TK]D-Fender~Qwell++
03:32.02[TK]D-Fenderjbot : Qwell
03:32.06jbotqwell is probably a patented liquid formula that contains three plant-based bio-active agents that work together in a perfectly balanced combination. These agents act synergistically to boost your good cholesterol and slash the bad.
03:32.16[TK]D-Fenderlol
03:32.26[TK]D-FenderI forget how to do it now!
03:32.35[TK]D-Fender<- dismal failure
03:33.39linuxmigrationjbot: Qwell++
03:40.26kronicI'm receiving this error with Monitor(): ast_writefile: Unable to open file
03:40.47kronicpermissions are correct, does asterisk create directories if their non-existent
03:53.15*** join/#asterisk bmg505 (n=leon@c1-236-7.rndf.isadsl.co.za)
03:57.35X-Rob_kronic, that's correct. it does not create directories.
04:01.08*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
04:01.28*** part/#asterisk jtexter3 (n=jtexter3@ip68-97-73-114.ok.ok.cox.net)
04:01.28kronicit does, its actually due to the fact it doesn't handle relative filepaths
04:04.13apturaHere we go, Poe overhead paging system.
04:04.16apturahttp://www.cyberdata.net/products/voip/voip-loudspeakeramp.html
04:04.43[TK]D-FenderCute, but "ick
04:05.19*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
04:05.19*** mode/#asterisk [+o mog] by ChanServ
04:05.40apturaick?
04:06.45apturaTk, what about this product you dont like?
04:07.34*** part/#asterisk ShadowHntr (i=sentinel@c-68-52-48-169.hsd1.tn.comcast.net)
04:09.00[TK]D-Fenderaptura : Low power guaranteed, and the device is tied to one tech.  Better off with an FXS based unit + ATA
04:09.23[TK]D-Fenderaptura : much more flexible, not to mention cheaper
04:09.34apturaohh because it is poe not enoug power to run the speakers
04:10.13[TK]D-Fenderaptura : And again we've been talking about phones who we know are capable of paging as it is.
04:10.27[TK]D-Fenderaptura : Not that is won't have enough power, jsut that it'll be weak.
04:10.28*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
04:11.48*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
04:13.59apturaSo what do you know works well
04:14.15*** join/#asterisk Winkie (i=slain@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com)
04:15.09[TK]D-Fenderaptura : the Viking paging units have a good rep
04:15.23apturaYea just saw them.
04:15.45apturaAnd then there is the one way page across all phones such as a emergency.
04:16.38apturaAnyway I am going to call it a night wife is home and dinner is cooking :)
04:16.50apturaThanks for the advice on the vikings.
04:17.00linuxmigrationany distro that is particularly well suited for asterisk or most people just go by personal preference?
04:17.51Yarrick40kcentos/trixbox is the preferred choice
04:18.02Yarrick40kthough I've seen it work with redhat, suse, etc
04:18.09Yarrick40kdepends on if you want to spend time rolling your own
04:18.47*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
04:18.52linuxmigrationsay, for manual install in a production env
04:19.12JTYarrick40k: ????
04:19.21JTYarrick40k: trixbox is NOT the preffered choice here
04:19.46JTthere is no advantage to using centos for asterisk either
04:19.46Yarrick40kJT: sorry, didn't mean to step on toes
04:19.54JTwhatever distro you are most comfortable with
04:20.26Yarrick40kJT: so what /is/ the preferred choice then?
04:20.55JTYarrick40k: standard asterisk, on the distro you are most comfortable/confident in personally
04:21.11JTno point learning a new distro just to do asterisk
04:21.39linuxmigrationhow is http://powerontech.com/freepbx-on-debian.htm ?
04:22.06linuxmigrationanyone use that tutorial?
04:22.16JTlinuxmigration: i'd recommend against freepbx in a production environment
04:22.53[TK]D-Fenderlinuxmigration : Whichever distro you are most comfortable administering
04:23.22Yarrick40kJT: why the recommendation against freepbx in production?
04:23.25linuxmigrationi'm not actually setting up a production server.  that was just an example
04:23.48[TK]D-Fenderlinuxmigration : Though I'd pick from Slackware, Debian, CentOS, RHEL if I were you.  Just more stable sources and regular devel stuff available
04:23.49JTYarrick40k: it's a toy which puts a gui on top of asterisk
04:24.29Yarrick40kJT: Right, but the underlying pieces (ie asterisk) is no different.
04:24.57JTYarrick40k: except that freepbx obsfucates access to it
04:25.03JTand you run versions of freepbx, not asterisk
04:25.07JTso it's a fork
04:25.19[TK]D-FenderYarrick40k : When you have a GUI controlling everything for you, who cares?  You are basically turning a massively flexible PBX into a TOASTER.
04:25.30linuxmigrationheh
04:25.33Yarrick40ktrue, don't get me wrong.
04:25.50[TK]D-FenderYarrick40k You'd be almost better to buy a proprietary PBX.
04:25.50Yarrick40kI'd rather do all the management from the command line anyways /shrug
04:26.22[TK]D-FenderWhich is why some managers still go with far less functional proprietary PBX's.
04:26.32Yarrick40ktrue
04:26.38JTat least there's a company to back it up
04:26.42Yarrick40kright
04:26.50file[TK]D-Fender: ! ! !
04:26.57JTthe proprietary pbxes, that is
04:28.23[TK]D-Fenderfile : I don't want relationship!
04:28.40[TK]D-Fenderfile : I just want ...
04:28.47file! ! !
04:28.54[TK]D-Fenderz0mg
04:29.29file[TK]D-Fender: wazzzzzup
04:33.02*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
04:34.33[TK]D-Fenderfile : Enjoying a little time off.  Contract last night took me from 5pm to 1 AM.
04:34.54*** join/#asterisk Defraz (n=t0tal@fw.centrisys.com)
04:34.55[TK]D-FenderSo trying to relax.  Not sure if I'll have another job lined up for tomorrow night or not.
04:35.10*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)
04:35.34fileeep, evil
04:38.05Juggiefile!
04:38.09filemoo
04:38.35Juggieyou hurt that poor irc virgin today :) (nic)
04:38.51fileha
04:39.15Juggiei just got home from the sens/leafs game
04:39.18Juggiewas a blast, did you watch it?
04:39.25fileno
04:40.03Juggieyou missed out, leafs won 6-0 :)
04:40.47*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
04:42.15*** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
04:49.36*** join/#asterisk bungalow (n=tasat-@c-24-7-62-81.hsd1.ca.comcast.net)
04:50.06bungalowanyone here familiar with building modules for asterisk 1.4?
05:15.03*** join/#asterisk tuxd00d (n=tuxinato@128.187.128.38)
05:15.21*** join/#asterisk linuxmigration_ (n=jeremy@dsl254-075-124.nyc1.dsl.speakeasy.net)
05:19.57*** join/#asterisk stephane_ (n=stephane@gw.sortilege.net)
05:24.07*** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com)
05:27.56*** join/#asterisk dir (n=dir@124.106.223.26)
05:30.16*** join/#asterisk Leoman (n=leo@70.241.166.244)
05:31.09LeomanGood evening.
05:32.36*** part/#asterisk Yarrick40k (n=jmoser@static-71-98-31-11.mdsnwi.dsl-w.verizon.net)
05:35.30LeomanI am looking for a way to allow a person at x place to be able to dial a number on y's local pstn.  Could I do that with asterisk?  I'm thinking that by placing to pbx's, one at x place and the other at y's place I can pick up a phone at x and call someone thru y's phone. x an y connect via the internet. Am I correct in my thinking?
05:37.34*** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar)
05:38.23Leomanjuanjoc estas?
05:38.34juanjocSi
05:38.51Leomanque tal como estas? tengo una pregunta de principiante
05:39.00Leomanes mas bien teoria
05:39.26Leomanhablas ingles? ya lo pregunte en ingles pero nadie respondio....
05:39.33juanjocSi, hablo inglés
05:39.42Leomanmira esto
05:39.45LeomanI am looking for a way to allow a person at x place to be able to dial a number on y's local pstn.  Could I do that with asterisk?  I'm thinking that by placing to pbx's, one at x place and the other at y's place I can pick up a phone at x and call someone thru y's phone. x an y connect via the internet. Am I correct in my thinking?
05:40.07juanjocWhat kind of connection to the PSTN do you have?
05:40.13juanjocA Zaptel card? SIP?
05:40.39Leomannone yet...I am in the planning stages.  I only have a regular phone with dsl connections
05:41.27juanjocThen you need to get a Zaptel card to connect you computer to the PSTN
05:41.37juanjocOr get a SIP connection to the PSTN
05:42.14Leomansay I get a zaptel card. With that in mind, could I do what I am thinking about doing?
05:42.42JTyes
05:43.00Leomansomething like:    phone -> pbx -> internet -> pbx w zaptel card -> pstn?
05:43.31JTsure
05:43.48Leomanhow do I connect the 2 pbx's?
05:44.14Leomanover the internet that is .
05:44.21JTsip or iax protocol
05:44.34juanjocYou connect them via a VoIP protocol, either SIP, IAX or H.323
05:44.50juanjocIAX might be the best option if you're connecting two Asterisk's
05:45.15Leomanwill I need to buy services from someone to do connect the two pbx's via IAX?
05:46.40*** join/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com)
05:46.45juanjocIf you just want to connect your PBX's you just need an Internet connection
05:46.55Leomanawesome.
05:47.40juanjocIf you want to access the telephone network you'd have to buy the service from a telephony provider
05:48.35juanjocThat might come in handy, for example, if you want to have a phone number in another country assigned to your PBX
05:49.26Leomanexactly what I'm looking for.  To be able to pick up the phone at x location and get a dial tone at y location , which by the way is in another country.
05:49.48juanjocYou'll normally get those services via SIP
05:49.58juanjocThere are lots of providers for this
05:50.19Leomanhow much are they per month? set fee or variable fee?
05:50.36juanjocIt depends, but they are generally cheap
05:51.01juanjocThe price per minute is normally cheaper than the one you'd get for a normal PSTN connection
05:51.16juanjocWhat country are you interested in?
05:51.44Leomanwhat if I just to a point to point connection between pbxs with IAX?  Is the quality of the call lower?
05:51.46LeomanPanama
05:52.17JTiax and sip are just voip protocols
05:52.22JTcodecs run over them
05:52.27juanjocNot necessarily, the quality will be maily determined by your Internet connection and the codec you use
05:52.30JTand they are the main determinate of call quality
05:52.47LeomanI don't why the need for the SIP provider.
05:53.02juanjocTo connect to the telephone network
05:53.05Leomansorry...I meant to say, I don't see why the need for the sip provider
05:53.20JTLeoman: you don't if you are connecting to the pstn directly
05:53.30juanjocUnless you want to connect to the PSTN on Panama via a PBX there
05:54.01Leomanyes, that's what I'm looking to do. place another pbx there and connect it to the one here.
05:54.35JTyes it will do it
05:54.41JTasterisk will do a lot of things
05:54.54juanjocThen you only have to connect both instances of Asterisk with an IAX link
05:55.05juanjocOr a SIP link
05:55.06orlockhmm
05:55.10juanjocAny of them will do
05:55.15orlockour provider is giving us a few trial sip trunks
05:55.20orlockmultiple DID's on one account
05:55.21orlock:)
05:55.42juanjocLeoman: I'm going to sleep, it's very late here, bye
05:55.45Leomanexcelent
05:55.56Leomanthanks for your help juanjoc
05:55.59juanjocnp
05:56.32orlockJT: ever heard of "TDM encryption"?
05:56.37LeomanJT, the connection between instances of asterisk is documented in asterisk?
05:57.35JTLeoman: www.voip-info.org
05:57.46JTthere's stuff on some approaches to take there
05:58.06JTorlock: no, but i've thought of it
05:58.14Leomanok, thanks. I'll get on reading there. Thanks for the answers.
05:58.42orlockJT: http://www.accessproviders.com.au/index.cfm?content=104
06:00.48JTdodgy, 500000 is hardly any encryption keys
06:01.12orlockJT: yeah, plus it doesnt actually have any real details
06:02.38JTof course
06:02.50JTit's proprietary! you wouldn't want hackers to know would you
06:03.37orlockJT: heh, "xor" i bet
06:04.16JTmaybe they're talking about spread spectrum spreading codes, they're XORed with the data :PO
06:04.19JT-O
06:04.39JTalthough it says TDM, so i doubt there's spread spectrum also
06:04.47orlockyeah, its 5.4Ghz
06:04.49JTtdm is poo for range and SNR
06:05.31orlockServer not found
06:05.31orlock<PROTECTED>
06:05.34orlock<PROTECTED>
06:05.36orlock<PROTECTED>
06:05.38orlock<PROTECTED>
06:05.40orlockargh
06:05.42orlockfuck
06:05.44orlockgoddman
06:05.46orlocksorry!
06:05.58jmlswho's godman ?
06:06.48JTwho said godman?
06:07.36jmlssorry, orlock asked for goddman. I know, he meant goddamn. it's early. give me a break
06:07.45jmls<PROTECTED>
06:07.48JTheh
06:08.04jmlsbloody kids
06:08.23jmlsgot me up at 6am.
06:08.30jmlsgrumpy daddy.
06:08.38jmlsback to * development ...
06:10.21*** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr)
06:14.31LeomanJT, is this the right card needed ? -> "WildCard X100P card", single FXO Port, PCI interface For Asterisk IP-PBX
06:15.02JTyou cannot buy any real X100Ps anymore
06:15.11JTthe ones around don't work very well
06:15.34JTyou'd be better off getting a TDM400P or an external hardware FXO device like a Sipura
06:15.41Leomanthat one says it's 100% compatible with the wildcard X100P
06:15.59JTbullshit
06:16.03Leomanthat's at least, what they say in the website I found that one.
06:16.03JTdon't believe it
06:16.06JTthey're all fake
06:16.13JTor made with rejected chips
06:16.21JTthey may work
06:16.28JTthey may work for a little bit
06:16.30Leomanit's only 16 bucks.,...
06:16.32JTor may not work at all
06:16.40JTwell if you want to try, go for it
06:16.48JTjust expect to buy something better at some stage
06:17.20Leomanso what does one do to get a real X100P?
06:19.05*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
06:19.13JTfind someone who has one, and buy it... they're no longer produced
06:19.20JTand it's hard to verify if it's real
06:19.28LeomanI see...ouch
06:19.49wmandrai bought mine from digium years ago
06:20.22Leomando the fake ones produce echo problems?
06:22.26wmandrai'd actually consider selling mine.... i'd rather replace one of my unused FXS modules with an FXO module
06:23.06*** join/#asterisk kmilitzer (n=km@office-gw.westend.com)
06:25.48*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
06:26.20*** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com)
06:30.50*** join/#asterisk DarKnesS_WolF (n=wolf@62.114.197.43)
06:31.01*** join/#asterisk batphone (n=bugz@cpe-24-162-13-42.houston.res.rr.com)
06:32.09*** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone)
06:34.49*** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com)
06:42.32Joel1978so then what are they selling at http://www.x100p.com ?
06:42.38*** part/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com)
06:44.13*** join/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com)
06:44.50JTi have no idea, Joel1978
06:45.05Joel1978they say they sell authentic original x100p cards
06:45.36JTprobably also fakes
06:45.45JTdigium were the ones who sold real X100Ps
06:45.53Joel1978for fakes their site is quite impressive
06:45.54Joel1978lol
06:47.09JTyeah
06:47.33JTit's possible they're not, but i have no idea how that would be..
06:48.23Joel1978probably farmed out production to another company as i don't think digium wanted to play on the low end of the hardware spectrum
06:48.57JTerr
06:49.07JTthe Intel chips are no longer in production
06:49.18JTso they'd have to be seconds or copies
06:49.27Joel1978probably better than before =)
06:49.40JTask about the X100P when the regulars are awake and active in here
06:49.47JTsome of which work for digium
06:49.59JTif they farmed it off, it's a secret to everyone
06:51.09wmandrais there a way to have * match 1001 and +1001 to the same extension without creating 2 seperate entries and without using "." ??
06:59.03Bobcat991966My vote is stay away from the X100P. I had one from Digium last year and it sucked.
06:59.30Joel1978i've heard that as well
06:59.55JTit was always just a rebadged winmodem with a heatsink on the chipset IC
07:00.09Bobcat991966I have the TDM400 now and I am not all that impress with it, but at least it works most of the time.
07:00.28Joel1978why not impressed?
07:00.37Bobcat991966Echo
07:00.49Bobcat991966not really bad but its still does echo
07:01.09*** join/#asterisk ShadowHntr (i=sentinel@c-68-52-48-169.hsd1.tn.comcast.net)
07:01.15Joel1978and you've changed settings applicable to resolving that with no improvement?
07:01.24ShadowHntrquestion - how resource hungry is asterisk in general
07:01.30ShadowHntrpure sip
07:01.38Bobcat991966Oh yes, I have spent hundreds of hours trying to tune it
07:01.58Bobcat991966It depends on what codec you use
07:02.22JTdepends if you transcode
07:02.51Bobcat991966I have heard good things about Rino cards,,, they have hardware echo cancelation
07:03.02Bobcat991966Rhino that is
07:03.22JTif it's that imortant an application, you'd be using digital connections, too
07:04.29ShadowHntri'm looking at a <1000MHz server
07:04.53ShadowHntrwell maybe
07:04.54JTShadowHntr: how many connections, what codec?
07:05.04ShadowHntrone line out
07:05.06Bobcat991966How man simultanious cards and what codec are you going to use. will you be using the conf bridge
07:05.10ShadowHntrnot sure on the codec yet
07:05.23aiksa[LV]any idea why outgoing digital call on chan_misdn would fail?
07:05.28JTone line out, how many voip connections?
07:05.39ShadowHntrlet's see, probably one or two at most.
07:05.40aiksa[LV]without even getting to remote equipment?
07:06.13JTShadowHntr: that's hardly any load, you'll be fine
07:06.18Bobcat991966you will be fine with a 1G processor and 512 megs ram
07:06.19ShadowHntrcool
07:06.53ShadowHntri was looking for a VoIP provider that let me use my own equipment. then i found broadvoice. :)
07:07.13Bobcat991966na go with Telasip
07:07.16ShadowHntrcause i could use asterisk and put an IP phone in each room.
07:07.58aiksa[LV]http://pastebin.ca/192776
07:08.12Bobcat991966Broadvoice has a bad rep with their so call unlimited plan
07:08.20aiksa[LV]anyone?
07:08.39ShadowHntrBobcat991966: who?
07:09.01ShadowHntrwho are they
07:09.02ShadowHntr:/
07:09.07Bobcat991966Telasip
07:09.13wmandrai have broadvoice now, and i'm shopping for a new provider
07:09.17JTaiksa[LV]: hmm, you haven't even said anything about what the setup is
07:09.28wmandrabroadvoice has been a huge headache
07:09.50Bobcat991966Good BYOD provider that lets you set you own outbound callerid and never gives me a problem with the anount of calls I make
07:10.13Joel1978ha
07:10.17Joel1978really?
07:10.33Joel1978that's the single thing i've been looking for is setting cid
07:10.38diablopicohey ,, does anyone know how to make Zaptel compile using bigmem ?
07:11.08Bobcat991966They wont let you set name, but they will let you set any CLID number you like
07:11.08ShadowHntri'm not impressed with their website.
07:11.31aiksa[LV]JT: i am trying to do txfax through misdn channel
07:11.34Joel1978number is good enough for me
07:11.51aiksa[LV]JT: with call spool file
07:12.14Bobcat991966Telasip is not much on their website but thay are excelent at customer service
07:12.16*** part/#asterisk [Airwolf] (n=airwolf@attilla.nl)
07:12.39ShadowHntrany others that are asterisk friendly?
07:13.04aiksa[LV]JT: chan_misdn documentation (in source folder) said that now options can supplied at the end of channel string
07:13.16aiksa[LV]where /h means outgoing digital call
07:13.33aiksa[LV]asterisk 1.2.9.1
07:13.40aiksa[LV]lastest chan_misdn
07:26.46JTaiksa[LV]: you haven't even said what card you're using
07:29.54*** join/#asterisk elduffy (n=elduffy@pd9569322.dip0.t-ipconnect.de)
07:32.02ShadowHntrBobcat991966: telasip is BYOD?
07:33.42aiksa[LV]JT, oh - very very sorry, beronet 4 port isdn bri
07:34.04aiksa[LV]ports configured as terminal equipment
07:34.41JToh hrm
07:34.51JTi have no experience with that card of chan_misdn myself
07:34.59JTis that card compatible with bristuff?
07:37.46ShadowHntrhmmm
07:37.49ShadowHntranyone use voicepulse?
07:39.03*** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk)
07:43.28aiksa[LV]JT: not sure, but i remember trying to get it work with bristuff with 2.4 kernel
07:43.56aiksa[LV]yet after migrating to 2.6, we chose mISDN
07:45.08aiksa[LV]asteriskguru.com says "- BN4S0 : HFC based 4 port BRI card, known to work with chan_misdn"
07:53.29*** join/#asterisk skywriter (n=test@mail.splendor.net)
07:54.12diablopicocan anyone help me with a zaptel compile problem ?
07:55.19diablopicohmmmm
07:58.20*** join/#asterisk Cyt (n=danielcy@85.75.176.202)
08:12.24aiksa[LV]what is chan_fax?
08:12.56*** part/#asterisk Joel1978 (n=Joel1978@12-226-85-195.client.mchsi.com)
08:13.27*** join/#asterisk scottmcl (n=scott@host217-40-20-25.in-addr.btopenworld.com)
08:15.01aiksa[LV]and how does it work anyway?
08:15.09scottmclHi i am having a big problem getting a call to transfer from a remote interface, can anyone help?
08:17.16scottmclIs anyone actually on the channel at the moment?
08:17.41aiksa[LV]yup
08:18.01aiksa[LV]what are you trying to achieve
08:18.01aiksa[LV]?
08:18.17scottmclis that is just aiksa and scottmcl.... :o(
08:19.51scottmclaiksa - are you looking for help on asterisk or able to help with techy problems?
08:19.56aiksa[LV]no, some just dont look into channel untill they get notified
08:20.13aiksa[LV]scottmcl: i can solve some of them, and i am still looking for help on some of them
08:20.30aiksa[LV]like chan_misdn with outgoing data call
08:20.38*** join/#asterisk inspired (n=mikael@85.221.7.59)
08:21.37scottmclaiksa: ok cool, let me know when you are free to have a chat about my problem .....i have hunted high and low for a solution but can't find one
08:21.39aiksa[LV]scottmcl: now tell me what are you trying to achieve: call transfer from extenranl line?
08:25.02scottmclaiksa: if someone external calls into a phone from a web interface the user has i am trying to transfer the call else where.
08:25.30*** join/#asterisk skywriter (n=test@mail.splendor.net)
08:25.45scottmclaiksa: i have looked at calling the transfer command on the phone via *2 by injecting DTMF tones but this does not seam to work.
08:27.19aiksa[LV]you have that enabled in features.conf?
08:27.52aiksa[LV]does the transfer work on other phones (internal)?
08:28.17*** join/#asterisk oQPa (i=name@237.Red-83-44-33.dynamicIP.rima-tde.net)
08:28.46scottmclyep
08:29.35aiksa[LV]so only softphones from web is affected?
08:34.02*** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com)
08:35.46scottmcli hae not tried softphones.......the web interface is an application i have written that lets you speeddial people
08:36.35aiksa[LV]oh, its using asterisk manager interface?
08:36.41aiksa[LV]or spool?
08:36.56scottmclyep....asterisk manager interface...
08:37.10aiksa[LV]you are using orginate?
08:37.34scottmcli am using ELMEG 290 and i have even tried calling the commands on the phone but there is no techy instructions for the phone
08:37.55scottmclyep using orginate
08:38.13aiksa[LV]have not worked with originate command, but i would suspect that it has to do with the  'tT' command option for Dial application command
08:38.58scottmclwhen it is a new call....but if you already have a call in place how do you get that call to go intro an attended transfer?
08:39.15scottmclwithout hitting the key on the phone
08:40.22*** join/#asterisk RoyK (n=roy@80.239.107.70)
08:40.28aiksa[LV]oh you want to perform redirect from the same interface on the web
08:40.33aiksa[LV]?
08:40.37scottmclyep
08:40.38RoyKmorning
08:40.45aiksa[LV]RoyK: morning
08:40.46scottmclmorning
08:41.31aiksa[LV]scottmcl: and right now you are just trying to perfomr another caqll with *2 ?
08:42.57scottmclaiksa: there is very little help on how to do it so i have guest that if i called via Asterisk Manager Interface playDTMF *2 on the active call line ....then the number this would start a transfer
08:43.12aiksa[LV]it appears that AMI has a function called Redirect
08:43.13scottmclcurrently it just plays the tones to the caller
08:43.38aiksa[LV]http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect
08:43.55aiksa[LV]I beleive you should be able to achieve functionality you desire with that
08:44.36aiksa[LV]i dont think playing dtmf on the line will work, AMI is not exactly replacement of keypad
08:45.18scottmclaiska: yep...is redirect an attended transfer....the wiki is not clear?
08:45.39aiksa[LV]http://www.voip-info.org/wiki/index.php?page=Asterisk+manager+Example%3A+Transfer
08:47.17aiksa[LV]scottmcl: i believe you should implement that behaviour through scriptting AMI
08:52.24*** join/#asterisk chexum (i=chexum@gateway/tor/x-976b73865dec5f7e)
08:54.44*** join/#asterisk dir (n=dir@124.106.223.26)
08:56.43*** join/#asterisk psk (n=psk@golia.caltanet.it)
09:01.29scottmclaiksa: i will have a look at the redirect command in a bit, thanks for your help
09:01.53aiksa[LV]scottmcl: ur welcome
09:07.08*** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net)
09:08.46*** join/#asterisk spr1te (i=spr1te@194.187.130.229)
09:09.39*** join/#asterisk oej (n=oej@pc-1694.puv.fi)
09:16.38*** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com)
09:22.12*** join/#asterisk ebi (n=ebi@ebi.bitflux.ch)
09:23.53ebiHi all. I'm looking for nice voip hardware and probably someone of you can help me :) I'm looking for a wireless (wlan or dect with station doesn't matter) phone that can easily switch between 2 profiles      [I like the Siemens SL75 WLAN but didn't find out if it's possible to switch profiles]
09:26.27*** join/#asterisk xnon (n=xnon@200.8.86.187)
09:33.55CytHi!!! Please, can I adapt this application: 0000,1,dial(${TRUNK}c/9871234321,20,r) to dial out usign a sip line on my PBX?
09:34.53*** join/#asterisk madafaka (i=hts@mobile.dusan.info)
09:46.04*** join/#asterisk S^P (n=ss@203.81.196.20)
09:46.28S^PHi I have a sip provider account how I connect to it using asterisk?
09:46.36S^Pregister=> ?
09:50.08*** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg)
09:52.31qdkS^P: sip peer or friend.
09:53.27S^Pfriend
09:54.04S^Pmy ATA was connected to it.
09:56.11*** join/#asterisk lintechnokrats (n=chikki@61.17.68.129)
09:57.48L|NUXS^P : use this register => user:pass@host:port/extension
09:57.48L|NUX:)
09:57.59*** part/#asterisk ebi (n=ebi@ebi.bitflux.ch)
09:58.45S^PI did and reloaded conf , but no listing in sip show registry :(
09:59.14L|NUXhumm
09:59.25L|NUXsip reload
09:59.28S^Pya
09:59.34L|NUXwhich version you are using ?
09:59.41L|NUXand which provider ?
10:00.04S^PAsterisk 1.2.12.1
10:00.08S^Pprovider RNK
10:00.12scottmclaiski: are you online still?
10:00.13L|NUXhummm
10:00.35L|NUXwell pb your sip.conf
10:01.50*** join/#asterisk dir (n=dir@124.106.223.26)
10:07.02*** join/#asterisk zotz (n=zotz@24.244.163.225)
10:09.56scottmclDoes any one know how to do an Attended transfer via the Asterisk Managment API?
10:14.40scottmcl....stund you all into silence :o)
10:20.57*** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se)
10:20.58jeremy_ghi
10:22.36*** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com)
10:25.41*** join/#asterisk leopardus1 (n=leopardu@195.158.71.173)
10:29.40*** join/#asterisk vpanayotov (n=vdp@83.228.51.12)
10:31.34*** join/#asterisk oej (n=oej@pc-1694.puv.fi)
10:32.26*** part/#asterisk vpanayotov (n=vdp@83.228.51.12)
10:35.02*** join/#asterisk hohum (n=dcorbe@bb-87-80-166-23.ukonline.co.uk)
10:35.17lintechnokratshi all
10:35.21lilalinuxwhen I have 2 extensions where the first is a prefix of the second, will the seconds be executed ever?
10:36.55x86give an example
10:37.23*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)
10:37.31lilalinuxexten => _X.,1,Dial(CAPI/ISDN1/...)
10:37.45lilalinuxexten => 1000, 1, ...
10:37.59S^Plilalinux: no
10:38.00x86those are two different extension patterns
10:38.18lilalinuxwhich will be executed?
10:38.46lilalinuxwhen 1000 is dialed
10:38.56*** join/#asterisk chexum (i=chexum@gateway/tor/x-aeec775f9d4813c4)
10:39.08S^Pas far as I knw _X.
10:40.02lilalinuxjust tried it: the second is executed
10:40.37S^P:(
10:42.28lilalinuxbut the question is why.
10:44.04S^PI experienced the opposite of it while asterisk book says it mathes the best match that what you just expereince.
10:45.29*** join/#asterisk tparcina (n=tparcina@36-25.dsl.iskon.hr)
10:46.15lilalinuxcan we exclude that both are executed at the same time?
10:46.18tparcinahi asterisk guru's
10:46.35lilalinuxhi tparcina
10:46.57tparcinawhich one of you knows where unatuhenthicated incoming IAX2 call goes?
10:47.22tparcinaif you don't have anything defined in general section?
10:47.50tparcinai mean, where it goes => in which context
10:48.35tparcinawho answers this question in a minute gets a plus ;))  (ps I know the answer)
10:49.37S^Pcheck ur guest setting
10:50.07S^Pby default it goes in default context of extension.conf
10:50.38S^Pnumber depend on what they dialed or s, (I guess)
10:50.46tparcinayes, and I don't have default context so unatuhenthicated call goes to context that is defined for last user in iax.conf :))
10:51.02tparcinasoeurity threat? it sure is!
10:51.09S^Phaha
10:52.22tparcinafor example, if i have no default context and i define context that gives posibility to make outgoing calls, someone could make outgoing calls on my account
10:52.57S^Phmmmmm
10:55.20tparcinaquestion: in iax.conf i hav edefined users with Ip address. because of that i can't put username/password. and when that user is calling me, he is treated as unauthenticated user. why?
10:55.36tparcinahow to make calls from that user as aunthenticated?
10:56.10tparcinaproblem is that in iax.conf i can't define context in general section as I can in sip.conf
11:02.23scottmclDoes any one know how to do an Attended transfer via the Asterisk Managment API?
11:08.46*** join/#asterisk xnon (n=xnon@200.8.86.187)
11:08.55*** join/#asterisk zorball (n=adf@dsl51B686F0.pool.t-online.hu)
11:08.56zorballhello
11:10.12zorballI have an passive isdn card /dev/ttyI0 i4l mode
11:10.16zorballIt is work
11:10.29zorballIf somebody call it I can handle
11:10.49zorballBut If i want to forward the call to another isdn card /dev/ttyI3
11:10.53*** join/#asterisk xnon (n=xnon@200.8.86.187)
11:10.58zorballit is not ringing
11:11.03zorballit say called
11:11.10zorballbut doesnt happen anything
11:11.33zorballThe /dev/ttyI3 is connected to an isdn telephone
11:11.43zorballCan it work ?
11:13.09florzzorball: You mean an isdn telephone is connected to the card that can be "reached" through /dev/ttyI3 as if it were an NT1?
11:14.44zorballThere is a pbx and there are some telehone on the isdn pbx and I want a gateway between the isdn pbx and the telehone line
11:14.48zorballwith two isdn card
11:15.58zorballflorz: ... yes
11:16.37florzzorball: Well, that won't work, you need a card that supports NT mode and a driver that does, too.
11:17.20S^PI register to a sip provider using register=> .... now how i send call on it?
11:17.57zorballflorz: Thanks
11:18.12zorballflorz: And if I put my /dev/ttyI0 as an extension?
11:18.35zorballI can only forward one line to another extension in same time?
11:18.46zorballflorz: And if I put my /dev/ttyI03 as an extension?
11:18.54florzzorball: hmm?
11:19.05*** join/#asterisk Lloydie-t (n=lloydie-@thomasclan.plus.com)
11:19.29zorballso /dev/ttyI3 I put the isdn pbx as an extension
11:19.48zorballand /dev/ttyI0 put into NT
11:20.23florzzorball: I4L doesn't support NT mode
11:23.04zorballok thx
11:23.28zorballbut if i put to /dev/ttyI3 to the isdn pbx as a another telephone
11:23.57zorballand if the call comeing i forward it to another phone which is on the isdn pbx too
11:24.21Lloydie-tCan I get some help with a q931.c problem
11:26.08*** join/#asterisk szundi (n=szundi@152.66.243.163)
11:26.43szundihi everyone, can somebody help me with a sip nat problem?
11:26.58razutzafrir , tzafrir_home : are you here ?
11:27.20tzafrir_homeonly one of us
11:27.50*** join/#asterisk KriS83 (n=kmarcrof@212.202.141.92)
11:28.32KriS83Hi
11:29.02*** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)
11:29.08*** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)
11:29.29szundii have a snom 300 ip phone, it is on a nat 172.* ip, but it has a virtual public ip address over an encrypted channel. so the server receives the sip session init from 10.* ip address of the phone, but the phone puts it's ip 172* in the sip headers. asterisk replies well to 10.* via sip, but the voicemal voip sound packets go to 172.*
11:30.26szundican somebody tell me the proper sip.conf nat=yes|no|... options or anything? it's 1.4.0 beta checked ot 2 weeks ago from trunk
11:30.46scottmcldefine('CMS_VERSION',         'v2.26' );
11:30.54scottmcloops wrong window
11:30.56scottmcl:)
11:31.05szundii'v tried google but nothing found about that, it seems my config is correct (but we see not)
11:31.06scottmclDoes any one know how to do an Attended transfer via the Asterisk Managment API?
11:32.25*** join/#asterisk zeppelin_ (n=zeppelin@201.41.0.159)
11:32.53szundihallo, can someone help me?
11:33.01szundiring ring :)
11:35.28zeppelin_hello all !!! :)
11:35.34leopardus1is trixbox using asteriskrealtime?
11:35.57RoyKasteriskonceuponatime
11:35.58zeppelin_leopardus1,  no !
11:38.00*** part/#asterisk Lloydie-t (n=lloydie-@thomasclan.plus.com)
11:43.05zorballFRITZ! CARD PCI can run in NT mode?
11:44.21zorballWhat kind of card know NT mode which cheep?
11:47.00florzzorball: HFC PCI-A based cards
11:51.41HarryRouch
11:51.42*** join/#asterisk denon (i=denon@synapse.subneural.net) [NETSPLIT VICTIM]
11:51.42*** join/#asterisk zeppelin_ (n=zeppelin@201.41.0.159) [NETSPLIT VICTIM]
11:51.42*** join/#asterisk zorball (n=adf@dsl51B686F0.pool.t-online.hu) [NETSPLIT VICTIM]
11:51.43*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) [NETSPLIT VICTIM]
11:51.43*** join/#asterisk lintechnokrats (n=chikki@61.17.68.129) [NETSPLIT VICTIM]
11:51.43*** join/#asterisk psk (n=psk@golia.caltanet.it) [NETSPLIT VICTIM]
11:51.43*** join/#asterisk tuxd00d (n=tuxinato@128.187.128.38)
11:51.43*** join/#asterisk bmg505 (n=leon@c1-236-7.rndf.isadsl.co.za) [NETSPLIT VICTIM]
11:51.43*** join/#asterisk xxoxx (n=xxoxxx@tor/regular/xxoxx) [NETSPLIT VICTIM]
11:51.43*** join/#asterisk sudhir492 (n=sudhir@leesburg-bsr3-68-65-168-202.chvlva.adelphia.net) [NETSPLIT VICTIM]
11:51.43*** join/#asterisk [Outcast] (n=bill@222-154-75-119.jetstream.xtra.co.nz) [NETSPLIT VICTIM]
11:51.43*** join/#asterisk b4ka (n=jh@71-226-114-200.fibertel.com.ar) [NETSPLIT VICTIM]
11:51.43*** join/#asterisk Renacor (n=kvirc@66.238.64.20) [NETSPLIT VICTIM]
11:51.44*** join/#asterisk Thus0 (n=Thus0@169.111.102-84.rev.gaoland.net) [NETSPLIT VICTIM]
11:51.44*** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) [NETSPLIT VICTIM]
11:51.44*** join/#asterisk diablopico (n=diablopi@ip68-101-147-222.sd.sd.cox.net) [NETSPLIT VICTIM]
11:51.44*** join/#asterisk Holos (n=asdf@204.101.26.106) [NETSPLIT VICTIM]
11:51.44*** join/#asterisk quid246 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) [NETSPLIT VICTIM]
11:51.44*** join/#asterisk DonX (i=don@the.lostserver.net) [NETSPLIT VICTIM]
11:51.44*** join/#asterisk shodan (n=shodan@ip206.99-113-216.pppoe4.joliette.intermonde.net) [NETSPLIT VICTIM]
11:51.44*** join/#asterisk tessier_ (n=treed@adsl-75-5-99-178.dsl.sndg02.sbcglobal.net) [NETSPLIT VICTIM]
11:51.44*** join/#asterisk hardwire-afk (n=hardwire@89-208-58-66.gci.net) [NETSPLIT VICTIM]
11:51.44*** join/#asterisk trelane_ (i=trelane@pdpc/supporter/sustaining/trelane)
11:51.45*** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) [NETSPLIT VICTIM]
11:51.45*** join/#asterisk Death_INC (n=sam@pdpc/supporter/sustaining/sbingner) [NETSPLIT VICTIM]
11:51.45*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) [NETSPLIT VICTIM]
11:51.45*** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) [NETSPLIT VICTIM]
11:51.45*** join/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) [NETSPLIT VICTIM]
11:51.45*** join/#asterisk nvzn (n=nvzn@dormir.dreaming.org) [NETSPLIT VICTIM]
11:51.45*** join/#asterisk lilalinux (i=e-trolle@langweiligneutral.deswahnsinns.de) [NETSPLIT VICTIM]
11:51.45*** join/#asterisk Dibbler_ (n=Dibbler@dsl-217-155-254-174.zen.co.uk) [NETSPLIT VICTIM]
11:51.45*** join/#asterisk type0 (i=type0@159-76-74-65.gci.net) [NETSPLIT VICTIM]
11:51.46*** join/#asterisk _Vile (i=_Vile@198.175.14.242) [NETSPLIT VICTIM]
11:51.46*** join/#asterisk faberk64 (n=faberk@213.199.15.249) [NETSPLIT VICTIM]
11:51.46*** join/#asterisk SwK (n=Silik0nJ@12-214-191-109.client.mchsi.com) [NETSPLIT VICTIM]
11:51.46*** join/#asterisk ptblank (n=MURDER1@68-169-166-65.lmdaca.adelphia.net) [NETSPLIT VICTIM]
11:51.46*** join/#asterisk wmandra (n=wmandra@c-68-37-251-85.hsd1.nj.comcast.net) [NETSPLIT VICTIM]
11:51.46*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) [NETSPLIT VICTIM]
11:51.46*** join/#asterisk gmfm (i=gmfm@rtr.enterprisemtg.net) [NETSPLIT VICTIM]
11:51.46*** join/#asterisk Eonz (n=Icarus@irc.americatelnet.com.pe) [NETSPLIT VICTIM]
11:51.46*** join/#asterisk MrChimpy (n=MrChimpy@212.158.8.162) [NETSPLIT VICTIM]
11:51.46*** join/#asterisk pif (n=ldm@zenon.apartia.fr) [NETSPLIT VICTIM]
11:51.47*** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) [NETSPLIT VICTIM]
11:51.47*** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) [NETSPLIT VICTIM]
11:51.47*** join/#asterisk Mavvie (n=edwin@ppp43-109.lns2.syd6.internode.on.net) [NETSPLIT VICTIM]
11:51.47*** join/#asterisk euphobot (n=don@adsl-64-170-69-114.dsl.lsan03.pacbell.net) [NETSPLIT VICTIM]
11:51.47*** join/#asterisk ltd (n=z@202-161-25-75.dyn.iinet.net.au) [NETSPLIT VICTIM]
11:51.47*** join/#asterisk jnc (n=jnc@208.100.19.13) [NETSPLIT VICTIM]
11:51.47*** join/#asterisk scud_ (n=scud@gate.hhsys.org) [NETSPLIT VICTIM]
11:51.47*** mode/#asterisk [+o denon] by irc.freenode.net
11:51.47*** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) [NETSPLIT VICTIM]
11:51.47*** join/#asterisk Idle (n=brian@S010600a024969312.ed.shawcable.net) [NETSPLIT VICTIM]
11:51.47*** join/#asterisk zparta (i=zparta@146.net90.skekraft.net) [NETSPLIT VICTIM]
11:51.48*** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de) [NETSPLIT VICTIM]
11:51.48*** join/#asterisk sivana[home] (n=richard@sivana-155-134.vianet.ca) [NETSPLIT VICTIM]
11:51.48*** join/#asterisk groogs (n=greg@d38-54-164.commercial1.cgocable.net) [NETSPLIT VICTIM]
11:51.48*** join/#asterisk I-MOD (i=opticron@c-71-207-209-230.hsd1.al.comcast.net) [NETSPLIT VICTIM]
11:51.48*** join/#asterisk AndyCap (n=aoy@pdpc/supporter/sustaining/AndyCap) [NETSPLIT VICTIM]
11:51.48*** join/#asterisk luke-jr_ (n=luke-jr@user-0c93tin.cable.mindspring.com)
11:51.48*** join/#asterisk jontow (i=jontow@hijacked.us) [NETSPLIT VICTIM]
11:51.48*** join/#asterisk tamp4x (n=syntheti@vonmail.vonworldwide.com) [NETSPLIT VICTIM]
11:51.48*** join/#asterisk brookshire (i=mbrooks@hijacked.us) [NETSPLIT VICTIM]
11:51.49*** join/#asterisk sivana (n=sivana@mixdown.ca) [NETSPLIT VICTIM]
11:51.49*** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) [NETSPLIT VICTIM]
11:51.49*** join/#asterisk mishehu (i=mishehu@207.229.186.131) [NETSPLIT VICTIM]
11:51.49*** join/#asterisk breno (n=breno@barovia.aureal.com.pe) [NETSPLIT VICTIM]
11:51.49KriS83Short and simple question.. how would I be able to do this: When a call comes in on a certain line, I'd like to see this in my snom: "sales: Extension it's coming from"
11:51.51x86weeeeeee
11:51.51HarryRmaybe I shouldn'tve pressed that big red button :)
11:51.51x86lol
11:52.21*** join/#asterisk Budairc (n=chatzill@200.215.57.174)
11:57.13S^Phi i register to a sip provide using register=> ... how i can dial thought it?
11:58.46benjkonly your provider can tell you
12:01.17S^Pwhen i connect ATA i can dial easily :(
12:04.27madafakawhere does asterisk keep sound files it uses for SayNumber?
12:04.28*** join/#asterisk soylentgreen (n=fgast@193.238.89.34)
12:04.32madafakai want to record them in my language
12:05.07*** join/#asterisk jrprado (i=jrprado@200.146.21.252.adsl.gvt.net.br)
12:05.46*** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com)
12:06.41jrpradohi, hi
12:07.08jrpradosomebody already configured telextreme in asterisk ?
12:08.09*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)
12:10.09creativxmadafaka: /var/lib/asterisk/sounds
12:10.19*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)
12:10.25madafakacreativx, tnx man...just found /var/lib/asterisk/sounds/digits :)
12:10.46madafakajust to find out how to record voice into gsm/ulaw format :)
12:11.04benjkyou can record them in other formats, then use sox to convert
12:12.04*** join/#asterisk Mugatu_ (n=mugatu@unaffiliated/Mugatu)
12:12.36creativxmadafaka: record()
12:12.43creativxalthough using that for digits might not be the best way
12:12.49creativxi'd rather use a prof sound editor
12:12.56creativxso that you dont get odd gaps and shit
12:13.12madafakasure :)
12:15.55*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
12:19.12jrpradosomebody already configured telextreme in asterisk ?
12:20.47*** join/#asterisk fourcheeze (n=rich@office.callmaster.co.uk)
12:21.02fourcheezedoes the Polycom IP501 have buddies?
12:21.13fourcheezebuddies as in those people whose presence you can see
12:22.57fourcheezeI've found a Contact Directory but it doesn't seem to show status
12:23.01fourcheezeam I missing something
12:25.35[TK]D-Fenderfourcheeze: Yes it supports buddy watch
12:25.54[TK]D-Fenderfourcheeze: You need to enable presence support in sip.cfg first
12:26.17fourcheeze[TK]D-Fender: I'm not provisioning them from the network - can I do this in the gui?
12:26.23[TK]D-Fenderfourcheeze: If you don't have a "Buddies" soft-key on Idle then you haven't done this
12:26.32[TK]D-Fenderfourcheeze: I doubt it.
12:26.36fourcheezedamn
12:26.48fourcheezeI hate all that tftp crap
12:27.21[TK]D-Fenderfourcheeze: The web GUI should never be used and I hope they remove it outright ASAP to make room for needed features.
12:27.34[TK]D-Fenderfourcheeze: It supports much more than just TFTP.....
12:30.31tzangerMugatu?
12:30.32tzangerheh
12:30.45tzanger"My entire panty line is made in sweatshops along the Malaysian border!"
12:31.32tzanger[TK]D-Fender: did you see that commit yesterday?  They fixed (worked around) presence for 2.0.1 polycom fw
12:32.01jrpradosomebody already configured telextreme in asterisk ?
12:32.02[TK]D-Fendertzanger: Polycom release?
12:32.14*** join/#asterisk zotz (n=zotz@24.244.163.225)
12:32.15[TK]D-Fenderjrprado: Who are they?
12:32.23tzanger[TK]D-Fender: no, * commit worked around it
12:32.51[TK]D-Fendertzanger: Whats the reason for the failure exactly, and how much is * to blame for the disparity?
12:33.08leopardus1how much does a small asterisk system sell? ;|
12:34.13jrpradoa company VoIP
12:34.32[TK]D-Fenderleopardus1: * is free, and the "system" is just a computer.  Buy whatever you want.  Cards if needed vary based on those needs.
12:34.41[TK]D-Fenderjrprado: Link please...
12:34.51jrpradohe work with user, pass and authenticationID
12:35.29[TK]D-Fenderjrprado: Sounds like jsut about every provider out there.  I'm sure you can set them up nearly identical to some other popular provider.
12:35.41tzanger[TK]D-Fender: the patch says "polycom phones only handle xpidf+xml, even if they say they can handle pidf+xml as well"
12:36.00tzangerit's a VERY small patch
12:36.11tzangerbasically if the useragent is Polycom, force XPIDF_XML
12:36.13[TK]D-Fendertzanger: 5 chars or less? ;)
12:36.26tzangerotherwise allow them to select PIDF_XML (RFC3863)
12:36.34tzanger1 extra line
12:36.45tzangerremove 3 lines, add 4
12:37.04leopardus1[TK]D-Fender  : what do you mean - cards if needed vary based on those needs?
12:37.17*** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it)
12:37.52[TK]D-Fendertzanger: Cool... unfortunately I'll have to wait for 1.4's release before taking advantage of it....
12:38.45tzanger?  1.4 isn't released?
12:38.52tzangerthey fixed it in 1.2 as well
12:38.59*** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com)
12:40.28[TK]D-Fendertzanger: Well I wonder if we'll see another 1.2 release before 1.4 goes "gold"
12:40.43*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
12:41.21[TK]D-Fenderleopardus1: Well you can run * without any special hardware if you just want to use it for VoIP calls.  If you want to plug other kind of lines & phones in you will need extra equipment.  The cost of which varies depending on that need.
12:43.00leopardus1[TK]D-Fender : What I really mean, is this system really selling?
12:43.27mutdoes anyone know anyone i can talk to about a lucent stinger config problem? getting desprate now =\
12:43.36leopardus1[TK]D-Fender : sorry I'm actually talking about trixbox!
12:48.00[TK]D-Fenderleopardus1: Trixbox is just a crappy packaged up Linux ddistro CD.  Costs NOTHING.  Its all just free software.
12:48.01sudhir492D-Fender: I am still struggling with paging on Polycom phones :-(
12:48.18[TK]D-Fendersudhir492: I msg'd you last night offering to log in to help you fix it.....
12:48.32[TK]D-Fendersudhir492: You must have waled away from your computer right after asking it.
12:48.43sudhir492Thanks for that. I must have stepped out
12:49.06sudhir492I remember, I got a call and I had to step out in emergency
12:49.11[TK]D-Fendersudhir492: PM me connect details and I'll get you fixed up.
12:49.30sudhir492thanks. How do I PM
12:50.25*** join/#asterisk Jan_Chu (n=jan@213.150.51.91)
12:51.09Jan_Chuhi, i have a "problem" with asterisk, i have a fully functional IVR, but when i use it, it has a LONG wait time before it forwards me to the chosen destination
12:51.25Jan_Chuanything i can do about this... we are talking 3 sec of silence
12:53.25[TK]D-FenderJan_Chu: Pastebin your IVR dialplan and CLI output for it
12:53.43[TK]D-FenderJan_Chu: www.pastebin.ca
12:53.59fourcheeze[TK]D-Fender: what's my quickest route getting the polycom to read a sip.cfg from somewhere?
12:54.17*** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr)
12:55.13Jan_Chu<- somewhat of a noob, [TK]D-Fender where do i find these config files ?
12:55.20creativx/etc/asterisk
12:55.29creativx:P
12:55.37creativxno wait.. polycom. my bad
12:55.38creativx:)
12:55.51Jan_Chuhehe ya, but in witch files :D
12:56.01*** join/#asterisk joelsolanki (i=joelsola@202.160.161.94)
12:56.17benjkwitch files, haha, I like that
12:56.29Jan_Chuhehe =)
12:56.35Jan_Chui think FreeBPX is more of my place.
12:56.44Jan_Chuthis is more develobment i think.....
12:56.44[TK]D-Fenderfourcheeze: FTP
12:56.46joelsolankiHello all. I had visited call center client. i got the exact scenario now.
12:57.12[TK]D-FenderJan_Chu: If you are using FreePBX, you indeed should not be asking that in here.
12:57.25benjkJan_Chu, what config files are you looking for>
12:57.27benjk?
12:57.43Jan_Chusorry wront place, going to the noob channel now =)
12:57.44joelsolankiThey are using Parsec call center software/hardware in that they have 2 E1 cards and both E1 cards are connected to cisco 3640 2 E1 cards.
12:57.45*** part/#asterisk Jan_Chu (n=jan@213.150.51.91)
12:57.49*** join/#asterisk flackes (n=flack@host-80-194-73-131.static.telewest.net)
12:58.00joelsolankinow i want to replace cisco 3640 with asterisk + 2 E1 cards.
12:58.34joelsolankiwhat i see is that they Parsec software/hardware is using g711 and cisco 3640 is doing transcoding to g729
12:58.40*** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr)
12:59.02joelsolankinow i m planning to keep asterisk with P4 / 1GB ram and 2 E1 card in that.
12:59.17fourcheeze[TK]D-Fender: does the 301 support buddys too?
12:59.23*** join/#asterisk ambriento (n=cyndi@201-27-80-238.dsl.telesp.net.br)
12:59.25[TK]D-Fenderfourcheeze: All of them
12:59.26joelsolankiso will P4 with 1 GB ram will transcode 60 channels on that hardware ?
12:59.49flackesthink that is to low spec tbw
12:59.50flackestbh
13:00.08joelsolanki?
13:00.11joelsolankimeans ?
13:00.13flackesdoes anyone here use the GXP - 2000 and have problems with it not responding?
13:00.18[TK]D-Fender"To Be Honest"
13:00.20flackesdont think that spec is high enough
13:00.39joelsolankiany one can recommend me ?
13:00.46joelsolankiRoYK: u there ?
13:00.52[TK]D-Fenderflackes: Everything Grandstream is flakey junk and should be avoided.
13:01.01flackeswell i have 100 phones and im using 2 x 3.0gb xeons
13:01.05flackeslol
13:01.16RoyKjoelsolanki: sure. whatup_
13:01.24flackeswhat would you recomend.. the phone has a few bugs but they seem to work ok
13:01.45joelsolankiRoyK: i visited the callcenter customer
13:01.46flackesAsterisk is very processor intensive
13:01.54tzangerflackes: depends on what you're doing
13:02.06RoyKjoelsolanki: so good for you. what has that got to do with me?
13:02.50joelsolankiNothing I have question regarding transcoding from g711 to g729
13:03.04joelsolankilet me give u complete scenario so u can understand ok ?
13:03.23benjkGrandstream Barbietones are alright for putting into your grandparents house or the kids room or as a kitchen phone, stuff like that
13:03.41HolosCan I fork a Playback or Background to start to play and immediatly go into the next step? I have am using a Background() followed by a Read() and If I enter digits before the Background() is up, it tires to locate that extension in the context. I'd like it to be at the Read() right away.
13:03.49[TK]D-Fenderflackes: Polycom or Aasta are the first shoice for any kind of business, and Linksys only for places that its hard to get the other 2 at a reasonable price in.
13:05.02joelsolankiRoyK: Parsec software/hardware has 2 E1 cards in it and both are connected to cisco 3640 2 E1 cards. and parsec software/hardware is using g711 and i guess cisco is transcoding to g729. now i m planning to keep 1 asterisk server with config of P4/2.4 Ghz / 1 gb ram and 2 E1 cards in it. so question is that will the hardware will transcode 60 channels from g711 to g729 ?
13:05.05[TK]D-FenderHolos: :  Read(variable[|filename][|maxdigits][|option][|attempts][|timeout])
13:05.11joelsolankiOr i need higher end server ?
13:05.25RoyKjoelsolanki: quite possibly. try it
13:05.27[TK]D-FenderHolos: I do believe if you pass Read the file to play it will play up to the point of the first digit.
13:05.40Holos[TK]D-Fender: Of course... thats why the Filename is in there.. Duh.. I just looked at that yesterday...
13:06.08joelsolankido u recommend any hardware ?
13:06.49*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
13:07.52fourcheeze[TK]D-Fender: ok, I've set me up an ftp site - do I need a firmware on there or can I just upload the config?
13:09.06pigpenIncoming call: Got SIP response 500 "Internal Server Error" back from 10.3.1.7
13:09.16pigpen^^^I am getting this from my polycom 601...ideas?
13:09.30pigpenSIP 2.0.1
13:09.34Holospigpen: All the time? or just intermittantly?
13:09.45pigpenintermittantly...
13:10.10pigpenI should also note that my buddy watch is no longer working on 2.0.1
13:10.17[TK]D-Fenderfourcheeze: You don't need firmware there, just the proper config files.
13:10.20pigpenIt was working fine on 1.6.7
13:10.23Holospigpen: Ya, I get those as well, I haven't worried much about it.
13:10.32pigpenyeah...just annoying.
13:10.52[TK]D-Fenderpigpen: There is a bug in the way Polycom claims to support presence which has JUST been fixed.
13:11.07pigpenah....next ver of the sip software I am sure...
13:11.10pigpenthanks for the info...
13:11.16[TK]D-Fendertzanger>[TK]D-Fender: the patch says "polycom phones only handle xpidf+xml, even if they say they can handle pidf+xml as well"
13:11.21Holospigpen: I have seen these a lot more when the call is forwarded. But I have been getting them on all firmwares.
13:11.26[TK]D-Fender<PROTECTED>
13:12.21pigpenI see this mostly when one of my buddy watch exten's recieves a call...
13:12.30pigpenthanks.
13:12.43*** join/#asterisk bpiper (n=bpiper@70.159.49.40)
13:16.16*** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org)
13:16.43[TK]D-Fender~pb
13:16.54jbotfrom memory, pb is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste
13:16.55Hymiehey guys.. does anyone know why pickup() won't work when an extension is being run via a dial(something&something&something) instead of a plain dial(something)?
13:17.06*** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com)
13:17.34*** join/#asterisk jbalcomb (n=jbalcomb@216.28.180.158)
13:17.59[TK]D-Fendertzanger: I can't see reference to the patch on Mantis....
13:19.08pigpen[TK]D-Fender, thanks...but I am trying to get things "cleaned up" with the way that the Siemens switch sees messages, etc...  After this, then we can get down to the isssue...
13:19.10*** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr)
13:24.34*** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com)
13:27.03*** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net)
13:27.48tzanger[TK]D-Fender: it's not on mantis
13:28.13tzangerin branch 1.2 it's r44432
13:28.13*** join/#asterisk trelane (n=trelane@unaffiliated/trelane)
13:28.29tzangerin 1.4, r44433, and trunk r44434
13:31.05*** join/#asterisk KriS83 (n=kmarcrof@212.202.141.92)
13:31.08[TK]D-Fendertzanger: Sigh.  Ok, guess I'll just have to sit back and wait for a release now....
13:31.26[TK]D-Fendertzanger: Any word on the "500 internal error" flooding?
13:32.01KriS83Hi, is there anyway I can prefix incoming calls to be shown like this on a SIP Phone: "sales: ${CALLERID}"
13:32.37tzanger[TK]D-Fender: I think that may fix that if the polycoms are saying they support PIDF
13:32.40[TK]D-FenderKriS83: Yes, change the callerid.
13:32.43tzangernot sure, I'll have to update and see
13:33.11[TK]D-FenderKriS83: Set(CALLERID(name)=sales: ${CALLERID(name)})
13:33.26KriS83Thanks
13:33.27Hymiehey guys.. does anyone know why pickup() won't work when an extension is being run via a dial(something&something&something) instead of a plain dial(something)?
13:36.35[TK]D-FenderHymie: Pastebin your related dialplan and CLI output
13:38.47Hymie[TK]D-Fender: give me 5, I'm trying to pickup(extension&extension) to match the dial(extension&extension).  FYI, pickup(extenstion) works if there is no & in the dial statement
13:39.49[TK]D-FenderHymie:   Pickup(extension[@context]): This application can pickup any ringing channel
13:39.51[TK]D-Fenderthat is calling the specified extension. If no context is specified, the current
13:39.52[TK]D-Fendercontext will be used.
13:40.11[TK]D-FenderHymie: Nowhere in there doe is imply that you can do this to multiple channels.  Why do you suddenly think it will?
13:40.32Hymie[TK]D-Fender: I didn't say I wanted it to pickup multiple channels
13:40.32*** part/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)
13:40.48Hymie[TK]D-Fender: it won't pickup any extension in a dial(blah&blah) statement
13:40.58Hymieso I tried pickup with the same as the dial line, just as a test
13:41.03HymieI dont' know what the code says
13:41.13Hymiehold on, it may be the context
13:41.19Hymiemight be my bonehead
13:44.23[TK]D-FenderHymie: I was suspecting that possibility but was waiting for the incriminating pastebin ;)
13:44.43Hymiehmm
13:44.47Hymieoddly, that didn't work
13:44.51Hymiealthough it seemed like the answer
13:45.06Hymiewups
13:45.08Hymiehold
13:47.39Hymie#$@$@#)($
13:48.07Hymiepastebin.ca is slow, wtf
13:48.24Hymie!pb
13:48.26Hymie~pb
13:48.31jbotwell, pb is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste
13:49.31*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
13:50.40Hymie[TK]D-Fender: log output -> http://channels.debian.net/paste/3959
13:52.59Hymie[TK]D-Fender: http://channels.debian.net/paste/3960 <--- the dialplan parts
13:54.10Hymie<PROTECTED>
13:54.10Hymie<PROTECTED>
13:55.51*** join/#asterisk skywriter (n=test@mail.splendor.net)
13:56.09skywriteris asterisk supported on fedora core 5
13:56.11fourcheeze[TK]D-Fender: ok, I'm officially a polycom convert
13:59.48intralanmanwow, WTF just happened there
13:59.49skywriterwhat happened?
13:59.49Hymieit's just a netsplit, heh
13:59.49skywriteris asterisk supported on fedora core 5?
13:59.49fourcheezewhat exactly is a netsplit?
13:59.49*** join/#asterisk flackes (n=flack@host-80-194-73-131.static.telewest.net)
13:59.49Hymieskywriter: not sure what you mean there... it will compile on it, I'm sure
13:59.49Hymieskywriter: and there are likely rpms for it
13:59.49flackesAny one have a clue how to Reboot an asterisk phone from Asterisk
13:59.49Hymiefourcheeze: one server has lost connection with the node it was connected to
13:59.49skywriteri hade the following message when i tried to compile zaptel
13:59.49Hymieflackes: what type of phone?  SIP?  some phones support reboot with SIP, but I don't know how to issue the command
13:59.49skywriterYou do not appear to have the sources for the 2.6.15-1.2054_FC5smp kernel instal led.
13:59.50flackesyea they are SIP
13:59.50Hymieskywriter: install the kernel headers, for the kernel you have installed
13:59.50skywriterany ideas?
13:59.50skywriterhow?
13:59.50Hymieskywriter: there should be something yum can download
13:59.51*** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) [NETSPLIT VICTIM]
13:59.51*** join/#asterisk Mugatu (n=mugatu@unaffiliated/Mugatu) [NETSPLIT VICTIM]
13:59.51*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) [NETSPLIT VICTIM]
13:59.52*** join/#asterisk tparcina (n=tparcina@36-25.dsl.iskon.hr) [NETSPLIT VICTIM]
13:59.52*** join/#asterisk leopardus1 (n=leopardu@195.158.71.173) [NETSPLIT VICTIM]
13:59.52*** join/#asterisk dir (n=dir@124.106.223.26) [NETSPLIT VICTIM]
13:59.52*** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) [NETSPLIT VICTIM]
13:59.52*** join/#asterisk Cyt[away] (n=danielcy@85.75.176.202) [NETSPLIT VICTIM]
13:59.52*** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) [NETSPLIT VICTIM]
13:59.52*** join/#asterisk bungalow (n=tasat-@c-24-7-62-81.hsd1.ca.comcast.net) [NETSPLIT VICTIM]
13:59.52*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) [NETSPLIT VICTIM]
13:59.52*** join/#asterisk ruskie (n=ruskie@sourcemage/mage/ruskie) [NETSPLIT VICTIM]
13:59.53*** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) [NETSPLIT VICTIM]
13:59.53*** join/#asterisk stugster (n=Stuart@80-192-33-73.stb.ubr09.edin.blueyonder.co.uk) [NETSPLIT VICTIM]
13:59.53*** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) [NETSPLIT VICTIM]
13:59.53*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) [NETSPLIT VICTIM]
13:59.53*** join/#asterisk Falle (n=falle@falle.se) [NETSPLIT VICTIM]
13:59.53*** join/#asterisk Pj_ (n=pj@fernande.happycoders.org) [NETSPLIT VICTIM]
13:59.53*** join/#asterisk Damin (n=damin@nucleus.nacs.net) [NETSPLIT VICTIM]
13:59.53*** join/#asterisk mover (n=dlu@83.125.8.7) [NETSPLIT VICTIM]
13:59.54*** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr) [NETSPLIT VICTIM]
13:59.54*** join/#asterisk aydiosmio (i=aydiosmi@judecca.aculei.net) [NETSPLIT VICTIM]
13:59.54*** join/#asterisk RaYmAn-Bx (i=rayman@kbhn-vbrg-sr0-vl212-213-185-15-16.perspektivbredband.net) [NETSPLIT VICTIM]
13:59.54*** join/#asterisk klem_ (n=klem@klem.estpak.ee) [NETSPLIT VICTIM]
13:59.54*** join/#asterisk Makenshi (n=chaz@2001:630:1c0:2001:20c:29ff:fe4d:1bd5) [NETSPLIT VICTIM]
13:59.54*** join/#asterisk ]D4v|zZ[ (n=david@unaffiliated/visincito) [NETSPLIT VICTIM]
13:59.54*** join/#asterisk MstlyHrmls (n=mh@66.195.193.151) [NETSPLIT VICTIM]
13:59.54*** join/#asterisk GerjanT (i=gerjan@frontgate.watchthe.net) [NETSPLIT VICTIM]
13:59.54*** join/#asterisk sulan (n=ksjoberg@1-1-4-23a.lio.sth.bostream.se) [NETSPLIT VICTIM]
13:59.55*** join/#asterisk Lyfe (n=lyfe@69.8.146.78) [NETSPLIT VICTIM]
13:59.55*** join/#asterisk CtRiX (n=CtRiX@aretha.navynet.it) [NETSPLIT VICTIM]
13:59.55*** join/#asterisk Money5ack (i=moneysac@wer.will.spontanficken.de) [NETSPLIT VICTIM]
13:59.55*** join/#asterisk KDan (i=nobody@sleek.sleektech.nl) [NETSPLIT VICTIM]
13:59.57*** join/#asterisk TexasJay (n=me@ns1.accu-com.com)
13:59.57TexasJaymorning :)
13:59.57skywriterany command should i type
13:59.57Hymieflackes: I'd search on voip-info for sip reset.. look at the uniden UIP200 page for a ghost of a hint, talking about it
13:59.58skywriteri m a beginner
13:59.58*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
14:01.15Hymieskywriter: I don't know anything about fedora, someone here might
14:01.15*** join/#asterisk denon (i=denon@synapse.subneural.net) [NETSPLIT VICTIM]
14:01.15*** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) [NETSPLIT VICTIM]
14:01.15*** join/#asterisk bpiper (n=bpiper@70.159.49.40) [NETSPLIT VICTIM]
14:01.15*** join/#asterisk jrprado (i=jrprado@200.146.21.252.adsl.gvt.net.br) [NETSPLIT VICTIM]
14:01.15*** join/#asterisk kore (i=kore@mindwipe.org) [NETSPLIT VICTIM]
14:01.15*** join/#asterisk smurf (n=smurf@debian/developer/smurf) [NETSPLIT VICTIM]
14:01.15*** join/#asterisk Rhizome (n=Rhizome@host-81-191-151-89.bluecom.no) [NETSPLIT VICTIM]
14:01.15*** join/#asterisk inspired (n=mikael@85.221.7.59) [NETSPLIT VICTIM]
14:01.15*** join/#asterisk zeppelin_ (n=zeppelin@201.41.0.159) [NETSPLIT VICTIM]
14:01.16*** join/#asterisk lintechnokrats (n=chikki@61.17.68.129) [NETSPLIT VICTIM]
14:01.16*** join/#asterisk psk (n=psk@golia.caltanet.it) [NETSPLIT VICTIM]
14:01.16*** join/#asterisk tuxd00d (n=tuxinato@128.187.128.38) [NETSPLIT VICTIM]
14:01.16*** join/#asterisk bmg505 (n=leon@c1-236-7.rndf.isadsl.co.za) [NETSPLIT VICTIM]
14:01.16*** join/#asterisk sudhir492 (n=sudhir@leesburg-bsr3-68-65-168-202.chvlva.adelphia.net) [NETSPLIT VICTIM]
14:01.16*** join/#asterisk [Outcast] (n=bill@222-154-75-119.jetstream.xtra.co.nz) [NETSPLIT VICTIM]
14:01.16*** join/#asterisk b4ka (n=jh@71-226-114-200.fibertel.com.ar) [NETSPLIT VICTIM]
14:01.16*** join/#asterisk Renacor (n=kvirc@66.238.64.20) [NETSPLIT VICTIM]
14:01.16*** join/#asterisk Thus0 (n=Thus0@169.111.102-84.rev.gaoland.net) [NETSPLIT VICTIM]
14:01.16*** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) [NETSPLIT VICTIM]
14:01.17*** join/#asterisk diablopico (n=diablopi@ip68-101-147-222.sd.sd.cox.net) [NETSPLIT VICTIM]
14:01.17*** join/#asterisk Holos (n=asdf@204.101.26.106) [NETSPLIT VICTIM]
14:01.17*** join/#asterisk quid246 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) [NETSPLIT VICTIM]
14:01.17*** join/#asterisk DonX (i=don@the.lostserver.net) [NETSPLIT VICTIM]
14:01.17*** join/#asterisk shodan (n=shodan@ip206.99-113-216.pppoe4.joliette.intermonde.net) [NETSPLIT VICTIM]
14:01.17*** join/#asterisk tessier_ (n=treed@adsl-75-5-99-178.dsl.sndg02.sbcglobal.net) [NETSPLIT VICTIM]
14:01.17*** join/#asterisk hardwire-afk (n=hardwire@89-208-58-66.gci.net) [NETSPLIT VICTIM]
14:01.17*** join/#asterisk trelane_ (i=trelane@pdpc/supporter/sustaining/trelane)
14:01.17*** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) [NETSPLIT VICTIM]
14:01.18*** join/#asterisk Death_INC (n=sam@pdpc/supporter/sustaining/sbingner) [NETSPLIT VICTIM]
14:01.18*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) [NETSPLIT VICTIM]
14:01.18*** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) [NETSPLIT VICTIM]
14:01.18*** join/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) [NETSPLIT VICTIM]
14:01.18*** join/#asterisk nvzn (n=nvzn@dormir.dreaming.org) [NETSPLIT VICTIM]
14:01.18*** join/#asterisk lilalinux (i=e-trolle@langweiligneutral.deswahnsinns.de) [NETSPLIT VICTIM]
14:01.18*** join/#asterisk Dibbler_ (n=Dibbler@dsl-217-155-254-174.zen.co.uk) [NETSPLIT VICTIM]
14:01.18*** join/#asterisk type0 (i=type0@159-76-74-65.gci.net) [NETSPLIT VICTIM]
14:01.18*** join/#asterisk _Vile (i=_Vile@198.175.14.242) [NETSPLIT VICTIM]
14:01.18*** join/#asterisk faberk64 (n=faberk@213.199.15.249) [NETSPLIT VICTIM]
14:01.19*** join/#asterisk SwK (n=Silik0nJ@12-214-191-109.client.mchsi.com) [NETSPLIT VICTIM]
14:01.19*** join/#asterisk ptblank (n=MURDER1@68-169-166-65.lmdaca.adelphia.net) [NETSPLIT VICTIM]
14:01.19*** join/#asterisk wmandra (n=wmandra@c-68-37-251-85.hsd1.nj.comcast.net) [NETSPLIT VICTIM]
14:01.19*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) [NETSPLIT VICTIM]
14:01.19*** join/#asterisk gmfm (i=gmfm@rtr.enterprisemtg.net) [NETSPLIT VICTIM]
14:01.19*** join/#asterisk Eonz (n=Icarus@irc.americatelnet.com.pe) [NETSPLIT VICTIM]
14:01.19*** join/#asterisk MrChimpy (n=MrChimpy@212.158.8.162) [NETSPLIT VICTIM]
14:01.19*** join/#asterisk pif (n=ldm@zenon.apartia.fr) [NETSPLIT VICTIM]
14:01.19*** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) [NETSPLIT VICTIM]
14:01.19*** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) [NETSPLIT VICTIM]
14:01.19*** mode/#asterisk [+o denon] by irc.freenode.net
14:01.20*** join/#asterisk Mavvie (n=edwin@ppp43-109.lns2.syd6.internode.on.net) [NETSPLIT VICTIM]
14:01.20*** join/#asterisk euphobot (n=don@adsl-64-170-69-114.dsl.lsan03.pacbell.net) [NETSPLIT VICTIM]
14:01.20*** join/#asterisk ltd (n=z@202-161-25-75.dyn.iinet.net.au) [NETSPLIT VICTIM]
14:01.20*** join/#asterisk jnc (n=jnc@208.100.19.13) [NETSPLIT VICTIM]
14:01.20*** join/#asterisk scud_ (n=scud@gate.hhsys.org) [NETSPLIT VICTIM]
14:01.20*** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) [NETSPLIT VICTIM]
14:01.20*** join/#asterisk Idle (n=brian@S010600a024969312.ed.shawcable.net) [NETSPLIT VICTIM]
14:01.20*** join/#asterisk zparta (i=zparta@146.net90.skekraft.net) [NETSPLIT VICTIM]
14:01.20*** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de) [NETSPLIT VICTIM]
14:01.21*** join/#asterisk sivana[home] (n=richard@sivana-155-134.vianet.ca) [NETSPLIT VICTIM]
14:01.21*** join/#asterisk groogs (n=greg@d38-54-164.commercial1.cgocable.net) [NETSPLIT VICTIM]
14:01.21*** join/#asterisk I-MOD (i=opticron@c-71-207-209-230.hsd1.al.comcast.net) [NETSPLIT VICTIM]
14:01.21*** join/#asterisk AndyCap (n=aoy@pdpc/supporter/sustaining/AndyCap) [NETSPLIT VICTIM]
14:01.21*** join/#asterisk luke-jr_ (n=luke-jr@user-0c93tin.cable.mindspring.com)
14:01.21*** join/#asterisk jontow (i=jontow@hijacked.us) [NETSPLIT VICTIM]
14:01.21*** join/#asterisk tamp4x (n=syntheti@vonmail.vonworldwide.com) [NETSPLIT VICTIM]
14:01.21*** join/#asterisk brookshire (i=mbrooks@hijacked.us) [NETSPLIT VICTIM]
14:01.21*** join/#asterisk sivana (n=sivana@mixdown.ca) [NETSPLIT VICTIM]
14:01.22*** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) [NETSPLIT VICTIM]
14:01.22*** join/#asterisk mishehu (i=mishehu@207.229.186.131) [NETSPLIT VICTIM]
14:01.22*** join/#asterisk breno (n=breno@barovia.aureal.com.pe) [NETSPLIT VICTIM]
14:01.24Hymieskywriter: I don't know anything about fedora, someone here might
14:01.24*** mode/#asterisk [+o anthm] by ChanServ
14:01.27*** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) [NETSPLIT VICTIM]
14:01.27flackeswill do :D
14:01.28flackesso its possible then?
14:01.28*** join/#asterisk madafaka (i=hts@mobile.dusan.info) [NETSPLIT VICTIM]
14:01.28*** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) [NETSPLIT VICTIM]
14:01.28*** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) [NETSPLIT VICTIM]
14:01.29*** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) [NETSPLIT VICTIM]
14:01.29*** join/#asterisk freebsd_fan (n=unsure@catagiuri305.giuri.unige.it) [NETSPLIT VICTIM]
14:01.29*** join/#asterisk Bobcat991966 (n=chatzill@cpe-069-132-138-111.carolina.res.rr.com) [NETSPLIT VICTIM]
14:01.29*** join/#asterisk key2 (n=key2@251.9.39-62.rev.gaoland.net) [NETSPLIT VICTIM]
14:01.29*** join/#asterisk bkw__ (n=brian@adsl-70-143-58-55.dsl.tul2ok.sbcglobal.net) [NETSPLIT VICTIM]
14:01.29*** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net) [NETSPLIT VICTIM]
14:01.29*** join/#asterisk nettie (n=nettie@85-18-54-38.ip.fastwebnet.it) [NETSPLIT VICTIM]
14:01.29*** join/#asterisk pigpen (n=mark@fw.seamans.cc) [NETSPLIT VICTIM]
14:01.29*** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) [NETSPLIT VICTIM]
14:01.30*** join/#asterisk suma (n=suma@cm53.omega182.maxonline.com.sg) [NETSPLIT VICTIM]
14:01.30*** join/#asterisk litage (n=nick@203.220.55.70) [NETSPLIT VICTIM]
14:01.30*** join/#asterisk tecnico (n=tecnico@24.96.146.69)
14:01.30*** join/#asterisk dr0ck (i=dr0ck@nat/digium/x-cfeaabf69e97c02a) [NETSPLIT VICTIM]
14:01.30*** join/#asterisk Niklas- (i=niklas@moo.dk) [NETSPLIT VICTIM]
14:01.30*** join/#asterisk hwt (n=hwt@curb.thorkildssen.com) [NETSPLIT VICTIM]
14:01.30*** join/#asterisk angler_ (i=angler@nat/digium/x-cf7f236b46e8a20c) [NETSPLIT VICTIM]
14:01.30*** join/#asterisk svanlund (n=dasv@slave.rixport80.se) [NETSPLIT VICTIM]
14:01.30*** join/#asterisk citats (n=james@mrplow.gnuinternet.com) [NETSPLIT VICTIM]
14:01.31*** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) [NETSPLIT VICTIM]
14:01.31*** join/#asterisk digitalp (n=steve@dns1.nyc.dns-roots.net) [NETSPLIT VICTIM]
14:01.31*** join/#asterisk stoffell (n=stoffell@pot.catsanddogs.com) [NETSPLIT VICTIM]
14:01.31*** join/#asterisk orlock (i=[V7ui+42@202.44.174.4.static.nexnet.net.au) [NETSPLIT VICTIM]
14:02.48*** join/#asterisk KriS83 (n=kmarcrof@212.202.141.92) [NETSPLIT VICTIM]
14:02.48*** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) [NETSPLIT VICTIM]
14:02.48*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) [NETSPLIT VICTIM]
14:02.48*** join/#asterisk S^P (n=ss@203.81.196.20) [NETSPLIT VICTIM]
14:02.49*** join/#asterisk arcanine (n=arcanine@203.82.44.179) [NETSPLIT VICTIM]
14:02.49*** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) [NETSPLIT VICTIM]
14:02.49*** join/#asterisk vader-- (n=johndoe@204.183.88.101) [NETSPLIT VICTIM]
14:02.49*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) [NETSPLIT VICTIM]
14:02.49*** join/#asterisk macTijn (i=martijn@linda.net.insecure.nl) [NETSPLIT VICTIM]
14:02.49*** join/#asterisk Mw3 (i=mw3@national.t-error.hu) [NETSPLIT VICTIM]
14:02.49*** join/#asterisk marcus2 (i=marcus@atlantis.outer.org) [NETSPLIT VICTIM]
14:02.49*** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) [NETSPLIT VICTIM]
14:02.49*** join/#asterisk So3kris (n=jan-will@ids.netland.nl) [NETSPLIT VICTIM]
14:02.49*** join/#asterisk xachen (i=justin@pdpc/supporter/student/xachen) [NETSPLIT VICTIM]
14:02.50*** join/#asterisk yatesy (n=yatesy@unaffiliated/yatesy) [NETSPLIT VICTIM]
14:02.50*** join/#asterisk bXi (i=bluepunk@irssi.co.uk) [NETSPLIT VICTIM]
14:02.50*** join/#asterisk festr__ (n=festr@ns.regnet.cz) [NETSPLIT VICTIM]
14:02.50*** join/#asterisk encode (n=encode@blah.i.hate.w1ndo.ws) [NETSPLIT VICTIM]
14:02.50*** join/#asterisk claviola (n=claviola@debian/developer/claviola)
14:02.50*** join/#asterisk eliXier (i=GTI16V@gti.twice-irc.de) [NETSPLIT VICTIM]
14:02.50*** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM]
14:02.50*** join/#asterisk joe (n=nnnjsaue@66.107.33.195) [NETSPLIT VICTIM]
14:02.50*** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) [NETSPLIT VICTIM]
14:02.51*** join/#asterisk infinity1 (i=foobar@208.184.76.100) [NETSPLIT VICTIM]
14:02.51*** join/#asterisk tzanger (n=tzanger@mixdown.ca) [NETSPLIT VICTIM]
14:02.51*** join/#asterisk kram (n=mark@pdpc/sponsor/digium/kram) [NETSPLIT VICTIM]
14:02.51*** join/#asterisk trelane (n=trelane@unaffiliated/trelane) [NETSPLIT VICTIM]
14:02.51*** join/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com) [NETSPLIT VICTIM]
14:02.51*** join/#asterisk Winkie (i=slain@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) [NETSPLIT VICTIM]
14:02.51*** join/#asterisk japerry (n=falc0n@216.231.51.209) [NETSPLIT VICTIM]
14:02.51*** join/#asterisk nvrs (n=RUR@Quebec-HSE-ppp3613717.sympatico.ca) [NETSPLIT VICTIM]
14:02.51*** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) [NETSPLIT VICTIM]
14:02.51*** join/#asterisk postel (n=jp@wikimedia/Postel) [NETSPLIT VICTIM]
14:02.52*** join/#asterisk SomeJ (n=jasonh@12.19.114.51) [NETSPLIT VICTIM]
14:02.52*** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com) [NETSPLIT VICTIM]
14:02.52*** join/#asterisk rpm (n=russell@S01060002b3d10d24.cg.shawcable.net) [NETSPLIT VICTIM]
14:02.52*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) [NETSPLIT VICTIM]
14:02.52*** join/#asterisk LakeSolon (n=blake@64-83-227-227.dhcp.stcd.mn.charter.com) [NETSPLIT VICTIM]
14:02.52*** join/#asterisk Corydon-w (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) [NETSPLIT VICTIM]
14:02.52*** join/#asterisk evilmnky (i=bit@kyna.dalbaech.net) [NETSPLIT VICTIM]
14:02.52*** join/#asterisk reza_ (i=reza@abort.boom.net) [NETSPLIT VICTIM]
14:02.52*** join/#asterisk znoG (n=gs@162-148-235-201.fibertel.com.ar) [NETSPLIT VICTIM]
14:02.53*** join/#asterisk JT (n=jon@unaffiliated/jt) [NETSPLIT VICTIM]
14:02.53*** join/#asterisk edwar64896 (n=medwards@72.83.233.220.exetel.com.au) [NETSPLIT VICTIM]
14:02.53*** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) [NETSPLIT VICTIM]
14:02.53*** join/#asterisk ajsbsd (n=aaron@216.157.144.227) [NETSPLIT VICTIM]
14:02.53*** join/#asterisk h3x (n=h3xor@64.192.116.17) [NETSPLIT VICTIM]
14:02.53*** mode/#asterisk [+oo kram Corydon-w] by irc.freenode.net
14:02.56skywriterwhen i try to install zaptel on fedora core 5 i got the following message
14:02.56skywriterYou do not appear to have the sources for the 2.6.15-1.2054_FC5smp kernel instal led.
14:02.56skywriterany  ideas
14:02.56TexasJayquick queue question:  what's the correct queue type for making asterisk dial the next agent, remembering who the last successful agent was?
14:02.56Hymieskywriter: I told you what you need to do, now you just need to find out how to install the kernel headers for your kernel.  You need to use YUM, or whatever you use to install packages on fedora...
14:03.51Hymieskywriter: search for the kernel number above, and "headers"
14:03.51Hymieor some such
14:03.51*** join/#asterisk olivier__ (n=olivier@obs92-4-82-239-116-113.fbx.proxad.net)
14:05.33quid246skywrite:  yum install kernel-devel
14:05.33Hymiethere you go! ;)
14:05.33[TK]D-FenderTexasJay: RRMEMORY
14:05.33skywriterhow can i know?
14:05.33Hymiecrazy fedora! ;Þ
14:06.41quid246gee, thanks would be nice
14:08.42Hymie[TK]D-Fender: any ideas?
14:09.30*** part/#asterisk jbalcomb (n=jbalcomb@216.28.180.158)
14:09.37*** join/#asterisk jbalcomb (n=jbalcomb@216.28.180.158)
14:09.39fall0utso, what is #asterisks reccomendations for appliances/front-ends/all-in-one solutions for small officez?
14:11.09[TK]D-FenderHymie: I didn't get your 2nd pastebin due to the netsplit
14:12.14[TK]D-Fenderfall0ut: We don't have one.  GUI's are not supported here.  If you want something cheap and easy, go with Trixbox.  For a bigger office, call www.williamsglobal.com and ask about Fireworx
14:12.30fall0utjust want reccomendations
14:13.47pigpenHi all...Siemens tech support is saying that Asterisk is sending the "channel number" in the "connect message".   Apparently, it doesn't affect functionality, but it make more warning messages during troubleshooting...any way to disable this?
14:14.47[TK]D-Fenderfall0ut: I just gave you 2.
14:17.02fall0utthnx
14:20.05Hymie[TK]D-Fender: http://channels.debian.net/paste/3960 <--- the dialplan parts
14:20.10Hymie[TK]D-Fender: there you go
14:22.03[TK]D-FenderHymie: Pretty obvious problem... there is no Exten 100 in [reception] .....
14:22.38Hymie[TK]D-Fender: hmm, I didn't try 9@reception...
14:22.40Hymieer
14:22.42Hymie0@
14:22.47Hymiehold on
14:23.00*** join/#asterisk gennak0001 (n=Miranda@207.190.248.178)
14:23.02Hymieer
14:23.15[TK]D-FenderHymie: there is no NUMBER in [reception] period.  Please adjust your vision and direction :)
14:23.23Hymieyeah
14:23.53HymieI thought it was checking for the actual extension, I started with SIP/100 in there, then moved it to 100 without thinking about the paragram shift
14:23.57Hymiewill try
14:24.02[TK]D-FenderHymie: I can already see how I'd fix this.... use your imagination.....
14:24.13Hymieyeah, a simple goto
14:24.20[TK]D-FenderHymie: SIP/100 is a DEVICE, not and EXTEN.... theres a clue for you....
14:24.27[TK]D-Fenderan*
14:24.40*** join/#asterisk tdonahue (n=tdonahue@207.138.151.58)
14:24.40Hymieparagram shift
14:25.17pifiu-laptopmorning fender
14:25.26[TK]D-FenderHymie: Excellent :)
14:25.33[TK]D-Fenderpifiu-laptop: y0
14:25.39*** join/#asterisk Qwell (i=qwell@nat/digium/x-b4935957fd8a87e7)
14:25.39*** mode/#asterisk [+o Qwell] by ChanServ
14:26.04pifiu-laptopi got one of those linksys 1 line voip desktop phones, im going to give it a try sometime in the next couple of days
14:26.15pifiu-laptopthey look pretty simple which is good
14:26.16*** join/#asterisk vexorg (n=vexorg@CPE0003478eef7c-CM0016b531e87c.cpe.net.cable.rogers.com)
14:27.06*** join/#asterisk Qwell (i=qwell@unaffiliated/qwell)
14:27.06*** mode/#asterisk [+o Qwell] by ChanServ
14:28.49[TK]D-Fenderpifiu-laptop: EW..... hope you didn't spend personal money on it....
14:30.13*** join/#asterisk Tili (n=tili@202.133.67.19)
14:30.40Tilii get this error in compiling zaptel on debian libtonezone.a(tonezone.lo)(.gnu.linkonce.t.__i686.get_pc_thunk.bx+0x0):/usr/src/zaptel-1.2.6/tonezone.c:41: first defined here
14:33.00[TK]D-FenderHymie: I believe the word you were looking for is "paradigm".  Indeed it was sort of grasping for it in place of my use of "direction".
14:34.13Hymie[TK]D-Fender: no.. no.. my spelling is correct if you are welsh
14:34.56[TK]D-FenderHymie: Watergate 2.0!
14:35.27mutwatergate was about bad spelling?
14:35.28*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
14:35.41Hymiemut: Nixon will tell you so ;)
14:35.46mutheh
14:35.52HymieI WAS FRAMED, IT WAS THE SPELLING!
14:40.59*** part/#asterisk bpiper (n=bpiper@70.159.49.40)
14:41.19*** join/#asterisk juanjoc (n=juanjoc@200.73.189.82)
14:43.31Hymie[TK]D-Fender: thanks dude, mucho appreciated and such
14:43.57[TK]D-FenderHymie: Quite welcome
14:50.02*** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net)
14:50.36pingwinis there a way to "expire" a variable after say 6 hours?
14:50.40pingwinor reset?
14:50.48pingwina method perhaps
14:50.51pingwinsomething that would work in my dialplan
14:52.06*** join/#asterisk pablus (n=nn@test.conama.cl)
14:52.29pablusmorning
14:52.31Hymiepingwin: you could always place a file in /tmp that had the same name as the variable
14:52.31intralanmanpingwin: for the same 6 hours every day? or just 6 hours from when you set it?
14:52.56Hymiepingwin: then put the time when you set that variable, in it, using a CLI command from asterisk, such as date using unix time
14:53.05Hymiepingwin: then, next time you reference the variable
14:53.23Hymiepingwin: you could run another cli command, one that subtracts the current date from the one in the file
14:53.25pingwinhrmmm. that may work
14:53.33Hymiepingwin: and resets the variable if it is > number seconds
14:53.41pingwinwouldn't that be the same as just setting two variables. one with the timestamp it was created?
14:54.01Hymiepingwin: sure, could do that too, I suppose ;)
14:54.08pingwinor can asterisk dialplan even deal with timestamps?
14:54.14Hymiepingwin: I don't know if it can
14:54.20Hymiepingwin: which is why I mentioned the above
14:54.36Hymiepingwin: but, even if it can't, you can likely set a variable to the output of a date command for timestamp
14:54.47*** join/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net)
14:54.49Hymiepingwin: then exclude the additional step I had, of writing it to a file
14:54.53rene1~seen oej
14:55.24jbotoej <n=oej@23.Red-88-7-53.staticIP.rima-tde.net> was last seen on IRC in channel #asterisk, 7d 23h 42m 52s ago, saying: '~seen kpfleming'.
14:55.24Hymieoej was never here, rene1
14:55.24rene1ahh i se
14:55.24rene1thx
14:55.24Hymierene1: you are in an alternate universe, rene1
14:55.28Hymierene1: please give me money, rene1 ;)
14:56.01pingwinHymie: cool, thank you, I'm going to give that a try
14:56.20rene1Hymie: sure, to your nigerian acct?
14:56.24rene1;)
14:56.30Hymiehehe
14:57.09*** join/#asterisk p1p (i=1@mail.comp911.com)
14:57.17*** part/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net)
14:58.19*** join/#asterisk anthonyl (i=anthony@nat/digium/x-76a5d67dc2e45d3a)
15:00.07*** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66)
15:01.15pingwinanyone know of a good tutorial for writing perl scripts for asterisk? or does asterisk only really work via the system call?
15:02.17macTijnpingwin: see AGI
15:02.20HarryRpingwin, you could start at http://www.voip-info.org/wiki-Asterisk+AGI or http://home.cogeco.ca/~camstuff/agi.html
15:02.26macTijnthat :)
15:02.37pingwinsweet, thanks for all your help :)
15:02.44pingwinasterisk fucking rocks guys!
15:03.53*** join/#asterisk brettnem (n=brettnem@72.29.102.158)
15:04.00brettnemhey all
15:04.11brettnemanyone in here using gnudialer?
15:05.01brettnemwow.. quiet in here today
15:05.21TexasJayanyone here an knowledgable about the aastra 480i?
15:06.19brettnemI've had a lot of trouble getting that phone to work
15:06.23brettnemwhere in Texas are you?
15:07.33pigpenbrettnem, careful ..he may be "packin"
15:07.45pigpenYou know Texan's
15:07.58brad_msswit's quiet b/c everyone is taking the day off for GatorGrowl ... obviously, that's a national holiday, right ? I mean most businesses around here are closed or closing early
15:08.48brettnempigpen: where abouts?
15:08.52pigpenmiddle.
15:09.03pigpenSan Antonio
15:09.10brettnemahh, I'm in Dripping Springs
15:09.23pigpenah yes...the spring with a leak.
15:09.37brettnemwhich is why all our wells are drying up
15:09.41pigpenI am working today out of Boerne...
15:10.21brettnemfor those of you not from texas, that's pronounced "BERNIE"
15:10.33brettnemor I think it is.. heh
15:10.51pigpenyep...dam Germans...hehe
15:11.00brettnemgerman texans
15:11.06*** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-154-197-7.red.bezeqint.net)
15:11.12pigpenyep...
15:11.19*** part/#asterisk skywriter (n=test@mail.splendor.net)
15:12.18brettnemok, so no one in here has used gnudialer?
15:12.18pigpenand sorry, I have never touched an aastra....
15:12.18pigpenor gnudialer...
15:12.18pigpenI use macdial...
15:12.18pigpenand no I am not a hippie.
15:12.18brettnemmacdial
15:12.52jmlsleave the poor gnu alone. Pick on something bigger. Like a Kongdialler ..
15:13.16TexasJaybrettnem:  what problems are you having with the 480i?  I quite like it.  Only issue is I wish I could figure out a way to disable the "X missed calls" thing that flashes on the screen.
15:13.40pigpensome people are really hung up on that....
15:13.47brettnempigpen: macdial isn't a call center dialer.. it's a TAPI
15:14.10*** join/#asterisk rados_ (n=rados10@c-68-62-71-76.hsd1.mi.comcast.net)
15:14.14brettnemTexasJay: I've had a heck of a time configuring it to work behind nat with asterisk
15:15.00TexasJaybrettnem:  with Asterisk behind the NAT with you or not?
15:15.13TexasJayI presume not otherwise you wouldn't have mentioned it. :)
15:15.34brettnemasterisk public... phones nat
15:17.20TexasJaycan't say i've been in that position... yet
15:17.26pingwinone other question please... when the p (call screening) flag is raised on a dial, it only records the name of the person if the callerid info is missing. is there anyway, other than something custom, to require the "say your name" option to occur for every phone call coming in?
15:17.36pingwinor for that dial.
15:17.52*** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com)
15:20.18*** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290)
15:22.33*** join/#asterisk lorinc (n=ang@caracas-4634.adsl.interware.hu)
15:26.46*** join/#asterisk droops (n=droops@adsl-074-245-001-031.sip.jan.bellsouth.net)
15:29.25*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)
15:32.36*** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net)
15:33.22*** join/#asterisk ToTo (n=ToTo@host138-138-dynamic.2-87-r.retail.telecomitalia.it)
15:34.11*** part/#asterisk chexum (i=chexum@gateway/tor/x-aeec775f9d4813c4)
15:36.48brad_msswwhat are the recommended pay-as-you-go voip providers these days ?
15:44.20*** join/#asterisk groogs (n=greg@d38-54-164.commercial1.cgocable.net)
15:45.16*** join/#asterisk _ViperNetworks (n=Nitesh@65.48.63.178)
15:46.00*** join/#asterisk aptura (n=s_a_l_e_@S010600a0c93f6f7e.vs.shawcable.net)
15:47.12quid241If you need outgoing only... check out rapidvox.com
15:47.28fall0utGuys at carriers.icall.com are nice
15:47.32fourcheezebrad_mssw: depends a lot where in the world you are
15:47.43*** join/#asterisk nextime (n=nextime@host86-151-dynamic.14-87-r.retail.telecomitalia.it)
15:47.45fall0utthey are using XO for origination, nice front-end to setup stuff
15:47.46brad_msswfourcheeze: hah, true, US ... Eastern
15:48.05*** join/#asterisk anthonyl (i=anthony@nat/digium/x-e7f99bf8fc50337f)
15:48.10fall0utI'm using them for testing out new softswitch
15:48.20brad_msswi've used so many providers over the last couple of years ... most of them are aweful ... just wondering if there is some 'fresh talent'
15:48.43fall0utCheck out icall
15:48.56*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
15:49.09scottmclDoes any one know how to do an Attended transfer via the Asterisk Managment API?
15:49.34apturabrad, it may be the nature of the backbones that are causing the latency and jitter. What do you think
15:49.36fall0uteither unlimited or $0.01/min origination, and different rates for termination
15:50.09fall0utits all pre-paid, too
15:50.47*** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar)
15:51.10*** join/#asterisk Synoptic (n=synoptic@modemcable252.182-203-24.mc.videotron.ca)
15:51.14Synoptichi everyone
15:51.54[TK]D-Fenderscottmcl: How do you imagine doing that?  How does AMI get the audio channel to complete?
15:52.10Synoptici'm having a little problem with a cheap ATA. Might just be some configuration. After hanging up the phone on the ATA, asterisk thinks it is still connected. Could it be the Polarity setup in the ATA ?
15:52.11apturaIf the voip carriers suck then I wonder if anyone who has used asterisk has had any issues with site to site direct conectivity issues.
15:52.13brad_msswaptura: yeah, I always do some basic latency/jitter tests before trying out a carrier
15:52.30*** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-152-194-84.red.bezeqint.net)
15:52.47[TK]D-FenderSynoptic: Doesn't make sense.  Hung up is a 100% guarantee
15:53.20quid241fall0ut:  Some good rates there, but then you look at some of the prices for other Area codes.. 14c a min... ouch.
15:53.55apturabrad, where do you think the problem exist? the voip carrier? the backbone?
15:54.10olivier__Synoptic> your ata is connected to * via SIP ?
15:54.37brad_msswaptura: well, a lot of problems are the voip carriers themselves ... especially got fed up with teliax because they'd just reboot their systems in the middle of the day (or so it seemed), no one would be able to connect for minutes
15:54.46Synopticolivier__: yes, and i'm in North America.
15:55.06apturabrad, thats a very bad idea. Telcos drop channels for maintence at 2 am.
15:55.55brad_msswjunction networks has been generally good ... though they do glitch from time to time
15:56.00apturathats because no one is on. The telco technicans would often listen for the last calls made..unknown to the caller waiting for them to get off the phone before thay change trunks to remove a frame out.
15:56.46HolosCan Asterisk do math like "${EXTEN} minus 10"?
15:56.59apturaI bet the life of a CO tech or egnineer was probebly lonely
15:57.12fall0utquid241: yea
15:57.18fall0utbut who uses a company like this for bulk term
15:57.48apturabrad u have customers using those services?
15:58.14brad_msswaptura: no 'customers', we just use it for our own business ...
15:58.19apturaokay
15:59.17[TK]D-FenderHolos: ${${EXTEN}-10}
15:59.42*** join/#asterisk sb_mx (n=sb_mx@200.78.229.18)
15:59.46[TK]D-FenderHolos: $[${EXTEN}-10] rather
16:03.55*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
16:03.55*** mode/#asterisk [+o mog] by ChanServ
16:04.49sudhir492Hi aptura
16:07.12TexasJayis it possible to dial multiple extensions and have them all be able to answer?
16:07.22*** join/#asterisk krondorl (n=krondorl@acid.auricnet.ca)
16:07.50TexasJaywithout the first answer trumping the other answers, that is.
16:10.03sudhir492msg D-Fender TK, Will you please join #polycom
16:10.14Synopticok
16:10.36Synoptici'm in the astrisk CLI, trying to figure out why asterisk thinks my ATA FXS is still in use..
16:11.38aydiosmioDEBUG
16:11.39rpmSynoptic: wrong signalling
16:13.35Synopticrpm, when it happens, all calls coming from ISP to all extension have no sound. and I cannot place a cal lanywhere inside the pbx system...
16:14.23*** join/#asterisk ikey (i=ikey@220.226.35.125)
16:14.31Synopticdo you still think it is wrong signaling?
16:15.30rpmsounds like rtp is broken then
16:15.41rpmor your codec is incompatible.
16:16.47fall0utfirst off, is it running SIP or MGCP or what?
16:17.04Synopticfall0ut, I'm using an ATA using SIP inside the same network as my *
16:17.14Synopticno nat and such
16:17.19fall0utwhat ATA?
16:18.00SynopticGeneric Cheapo Ebay copy of pap2 style..
16:18.19fall0utso what makes you think it's still in use?
16:18.29*** join/#asterisk adorah (n=admin@87.68.169.166.cable.012.net.il)
16:18.33fall0utsip show channels shows active calls?
16:19.06SynopticChannel Location State Application(Data)
16:19.06Synoptic0 active channels
16:19.06Synoptic1 active call
16:19.17Synopticandthe phone is hung up
16:19.45Synopticand now, I cannot make any call, looks like * is getting confised or something
16:20.12scottmcl[TK]D-Fender : well you can do blind transfers using the redirect command .....some people seem to suggest using playDTMF then sending *2 ..... or what ever you have set the command too
16:23.41*** join/#asterisk bytefoo (n=bytefoo@207-114-255-134.static.twtelecom.net)
16:24.21bytefooin asterisk can i make a voice menu that, when you make a selection will run a script or kick off a cron job?
16:24.46*** join/#asterisk juanjoc (n=juanjoc@200.73.189.82)
16:26.54aydiosmiobytefoo: yes, just set the IVR to send to an extension.conf entry that calls an AGI()
16:27.12aydiosmiothen you can make that AGI a perl script that runs a series of commands
16:27.17[TK]D-Fenderscottmcl: Of course you can do blind transfers.  You're just "throwing" the channel somewhere else without real intervention.
16:27.48bytefoosweet thanks aydiosmio, i don't know what half that meant as i'm still new to all this but I wanted to make sure it was possible ;)
16:28.01[TK]D-Fenderscottmcl: But if you want to do Attended, the AMI has no audio/dtmf interface to allow you to enter where.
16:37.58*** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com)
16:41.33*** join/#asterisk _alex_mx_ (n=alex@200.78.229.18)
16:50.54_alex_mx_hello googling led me to a post of mine of sept 2005 with the same problem that went unresolved.  We solved it by getting rid of all our iax devices and going pure sip.  Now we have 2 servers under heavy load with iax2 trunk between them and here it is again..."channel.c: Dropping voice to exceptionally long queue on IAX2/" server becomes unresponsive and only solution is to restart.  We do voicemail to email but log file doesn't show one being done bef
16:50.55_alex_mx_ore the errors start.  Running latest 1.2 release on both servers.  Any ideas?
16:54.49_alex_mx_file, ? posting on the correct channel this time...:P
16:55.06file?
16:55.06Bobcat991966does naybody have googletalk working with asterisk 1.4 svn?
16:59.28MIXX941Hi all. Trying to get outgoing termination going through IAX and am getting the "Call Rejected by x.x.x.x: No Authority Found" error 50. The service doesn't require registration or a username/password, just to send a special pin before the phone number in extensions.conf. Been searching online and haven't found a situation exactly like mine (meaning most of them required usernames/passwords or registering, and that was the problem). Any ideas on wh
16:59.30MIXX941at to check?
17:00.20apturaBobcat991966 CHECK IN voip-info.org
17:01.00intralanmanMIXX941: what's your dial command look like?
17:01.29intralanmandial(IAX2/peer/PIN${EXTEN}) or something like that?
17:01.34Bobcat991966Ya I hve been there looked at some sample configuration but im not able to get it to work. when I dail a google talk extenshion I get Everyone is busy/congested at this time (1:0/0/1)
17:02.37Bobcat991966Im wondering during the inital install ofasterisk if there was a confiuration option that needed to be set prior to doing that make install
17:04.29MIXX941intralanman: it was "Dial(IAX2/PIN${EXTEN}@provider)" ...but I tried your way and now get a different error
17:05.07intralanmanwell, at least it's a different one
17:05.12intralanmanlol
17:05.43MIXX941yeah
17:05.49MIXX941"I don't know how to authenticate automation to"
17:06.17*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
17:07.22*** join/#asterisk test34 (n=test34@unaffiliated/test34)
17:12.10*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
17:12.10*** mode/#asterisk [+o russellb] by ChanServ
17:13.58*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
17:18.28*** join/#asterisk Bobcat991966 (n=chatzill@cpe-069-132-138-111.carolina.res.rr.com)
17:20.10*** join/#asterisk larryrichardson (n=larry@64.56.99.31)
17:21.21*** join/#asterisk denon (i=denon@synapse.subneural.net)
17:21.21*** mode/#asterisk [+o denon] by ChanServ
17:21.42MIXX941does anyone know what "chan_iax2.c:7205 socket_read: I don't know how to authenticate automation to x.x.x.x" means? online searches turn up nothing for that.
17:22.11*** join/#asterisk alexis101 (n=as@MTRLPQ02-1177996380.sdsl.bell.ca)
17:23.35alexis101hi guys i have a little problem with my queue System ... When i called the queue command with the n option asterisk still do a retry on the agent , can someone tell me why?
17:28.21*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
17:28.29*** join/#asterisk mr_canny (n=root@200.138.113.82)
17:29.54*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
17:30.31*** join/#asterisk C6Vette (n=info@72-166-37-114.dia.static.qwest.net)
17:33.09*** join/#asterisk hfb (n=hfb@pool-71-108-114-33.lsanca.dsl-w.verizon.net)
17:36.17*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
17:36.24*** join/#asterisk diclophis-work (n=jbardin@65.203.37.58)
17:36.39diclophis-workhello all
17:36.57diablopicoanyone know how to give a device  ( te212p ) its own IRQ ?
17:37.07diclophis-workusing SIP, is there any way to have more than 2 channels connected together?
17:38.23diclophis-workdiablopico: you can disable all the other devices?
17:38.50larryrichardsonI have a ? on Queue notification. I need to have a queue announce the agent that is accepting the call before it is transferred.  It looks like there is a Manager "eventwhencalled", but I don;t know how to tap into this. ANy pointers?
17:39.11diablopicoi am running bare bones as it is
17:39.28*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
17:40.08diablopicoi8 have moved the card to each of the PCI slots on the motherboard , and it always shares an interrupt.
17:41.44scottmclDoes any one know how to do an Attended transfer via the Asterisk Managment API?
17:46.08rob0What devices are sharing the IRQ?
17:46.40[TK]D-Fenderdiablopico: Tried setting it in your BIOS?  What MB are you using?
17:46.55diablopicothe vidio card at the momoent
17:47.46[TK]D-FenderOUCH.... very bad
17:47.50diablopicomotherboard = SuperMicro ( X7DVA-E
17:48.09[TK]D-Fenderdiablopico: Oh well... caught by the Digium IRQ curse.....
17:48.36diablopicoThe BIOS wont let me set an irq for the PCI slot
17:48.56diablopicoit all worked until i rebooted
17:49.09japerryomg
17:49.11tzafrir_homediablopico, another avenue of magic: kernel parameters
17:49.14japerryI wanna take a hammer to asterisk
17:49.27tzafrir_homee.g: acpi=no , or pci=noacpi
17:49.44japerryseems like the only tool to fix it --- now RANDOMLY incoming calls are getting forwarded to our main extension and not the phone they're supposed to goto
17:50.09[TK]D-Fendertzafrir_home: I don't think he has the requisite goat to sacrifice for that black an art ;)
17:50.17*** join/#asterisk ManxPower (n=manxpowe@stirprop-s4-0-0-21.ndcr2.datasync.net)
17:50.20japerryso IE: I call 555-6001 and it gets routed to 555-6000. but sometimes it will ring 555-6001, but its random
17:50.23diablopicoyou mean to recompile the kernel with different parameters ?
17:50.36*** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net)
17:50.57[TK]D-Fenderdiablopico: No, he means change your bootloader to call your existing kernel with different options
17:51.05tzafrir_homejaperry, anything interesting in the trace?
17:51.37japerrythats what I'm attempting to do now
17:51.41tzafrir_homedo you have a full log with decent verbose level ?
17:52.00japerrytzafrir_home: well I'm using the asterisk CLI with verbosity 10
17:52.19diablopicothanks all , it looks like i have some homework to do....
17:52.32tzafrir_homejaperry, also make sure (in logger.conf) that this is sent to a log file
17:52.47*** join/#asterisk soylentgreen (n=fgast@nebukadnezar-em0.only640k.org)
17:53.57japerryahh yes yes I've turned on debug into a file
17:54.11japerryI was only getting notice,warning,error in messages
17:54.35tzafrir_homejaperry, verbose is typically the most useful. Debug is often too verbose
17:54.42japerryok
17:56.29jmlsdiablopico: did you update the system before the reboot (kernel etc )
17:56.59japerryso yah the logs are showing anything different
17:57.14japerrythe odd part is that I think something might be wrong with what number is being sent through DTMF?
17:57.45japerrybecause asterisk only gets the last 4 digits, which has been working--however, its now randomly not picking up those numbers
17:58.06japerrybasically it says 'Executing goto --IVRmenu' when it should be saying 'Executing goto ext 601
17:58.17japerrybut I dial our 602 ext outside, and it works fine
17:59.22*** join/#asterisk zotz (n=zotz@24.244.163.225)
18:00.19*** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net)
18:01.59hmmhesaysok my mediatrix 2102 is freaking out sending out registrations constantly
18:04.54hmmhesaysanyone else run into this?
18:07.03hmmhesaysdead in here today
18:08.03jmlssorry, not dead. Just the chiili con carne ..
18:08.12jmls*chilli
18:08.27Bobcat991966does anybody know how to install  iksemel on asterisk 1.4 svn?
18:09.08jmlsBobcat991966: do yourself a favour, get the source, and compile it with no optimizations.
18:09.22Bobcat991966why so?
18:09.37jmlsBobcat991966: see bug #7672
18:10.27jmlsspecifically look at the note (0050803)
18:12.50jmlsfile: ouch
18:13.06jmlsfile: I'm not dead. Honest
18:13.12filea likely story
18:13.29jmlsI feel happy, perfectly happy.
18:13.30hardwirefreak
18:13.43jmlsfrack
18:15.13scottmclDoes any one know how to do an Attended transfer via the Asterisk Managment API?
18:15.41jmlsscottmcl: I'm sure somebody does. When you find out, let me know ! I want that as well.
18:17.00[TK]D-Fenderscottmcl: Was something in my answer not clear to you?  AMI has no audio interface so how are you supposed to talk to the person you want to transfer them to?!?!
18:17.57jmls[TK]D-Fender: I want to be able to get my application to do the work of the agent (hold->more->transfer->dial-speak->transfer)
18:18.36jmls[TK]D-Fender: so I want to: Click On Target->Speak-Click and for the call to be transferred to the target
18:18.37[TK]D-Fenderjmls: You can't SIP phones are "smart" and they do their own thing.  You can't tell the to do anything.
18:19.05jmls[TK]D-Fender: I was hoping app_redirect (when it hits the streets) would do this sort of thing
18:19.06[TK]D-Fenderjmls: So forget about forcing them into a transfer they don't initiate.  UA = king.
18:19.55Bobcat991966jmls: thats an interesting bug, but unless im reading it incorrectly it has nothing to do with adding iksemel.so which is required to make asterisk work with Googletalk.
18:19.56*** join/#asterisk beyond (n=beyond@200.192.160.100)
18:20.14scottmclD-Fender : i know it is not an audio interface but i want to triger the process from somewhere that is not the phone....
18:21.04jmlsBobcat991966: asterisk will make use of the iksemel libraries and calls when using googletalk - so if there is a bug in iksemel, googletalk may find it as well
18:21.46[TK]D-Fenderscottmcl: You can steat the call, but you can't walk them through a transfer.
18:21.48scottmclD-Fender : i.e. you are on a call, then you want to click a button on a screen that holds the call then dial a new number.....like *201162227777
18:21.50moghey jmls i should have my rework done to it tonight / tommorrow morning
18:21.57mogso i will send it to ya
18:22.05jmlsmog: cool!
18:22.24mogi need to rewrite their makefiles and then it should be good for testing
18:22.36scottmclD-Fender : some people say you can do palyDTMF to do it....but not sure if this is the correct way
18:22.39jmlsmog: do you agree with what I'm saying to Bobcat991966, or am I talking male cow dung
18:22.40Bobcat991966is bug #7672 what you ment to say because this bugs title is Cannot simultaneously open >1 new browser window
18:23.22jmlsBobcat991966: 0007672: Asterisk core dump using ast_aji_send
18:23.22mogthat bug is related to ikesemel sucking
18:23.31Bobcat991966ahh ok
18:23.33scottmclD-Fender: you can start new calls....i have speed dial screen but the transfer process has been pissing me off for weeks
18:23.36mogand should be fixed with this rework
18:23.47[TK]D-Fenderscottmcl: There is no correct way to do what you want.  Is against SIP's concepts
18:23.53mogBobcat991966, what happens is their library screws up somewhere inside of gnutls and dies
18:23.56mognext time we call it
18:23.57aydiosmioSIP DENIAL
18:23.58mogwe die
18:24.11mogwe could check every time we deref the iksemel parser
18:24.11jmlsBobcat991966: wtf ? "Cannot simultaneously open >1 new browser window"
18:24.17mogand maybe should
18:24.26mogbut bug is going to be solved this way
18:24.36mogas the iksemel parser should never fail
18:24.41jmlsBobcat991966: are you looking at http://bugs.digium.com/view.php?id=7672 ?
18:24.47*** join/#asterisk denon (i=denon@synapse.subneural.net)
18:24.47*** mode/#asterisk [+o denon] by ChanServ
18:25.07Bobcat991966jmls,thats beter, makes sence now
18:25.11Bobcat991966thanks
18:25.17jmls<PROTECTED>
18:25.36jmlsBobcat991966: what were you looking at ???
18:26.14Bobcat991966https://bugzilla.mozilla.org/show_bug.cgi?id=7672
18:26.26jmlsok .....
18:26.36*** join/#asterisk dahunter3 (n=dahunter@pool-71-110-86-116.lsanca.dsl-w.verizon.net)
18:27.24mogwhy did you go to mozilla's bug page instead of our projects...
18:32.27*** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252)
18:32.33Nuggetwow.  what an idiot.
18:32.36scottmclD-Fender : so injecting DTMF tones will not work?  My other thought was to interface with the phone some how but i have elmeg ip290 and very little instruction
18:35.02jmlsno, it's a valid place to go - if you search for iksemel bugs you land up there
18:35.53apturaIs there any really good free web editors other then nvu?
18:36.04jmlsvi!
18:36.24jmlslike writing the web in assembler :)
18:36.25[TK]D-Fenderscottmcl: Injecting tones?  An attended transfer mens you speak to the person you wnt to transfer them to first.  You still aren't speaking to anyone.  Like I said the most you can expect to be able to do is STEAL the call and throw them somewhere else completely.  No DTMF required.
18:36.55[TK]D-FenderI do all my web programming in Notepad2
18:37.06mogvim is nice for any edditting
18:38.02NuggetI'd expect you to prefer an eddittorr with a sppellcchheckkeerr. ;)
18:38.07filemog: mog.
18:38.13jmlsmog: really. Hope you didn't use it for irc ;)
18:38.16scottmclD-Fender : is there any other way of remotely calling an attended transfer....with out touching the phone....i am sure there must be a way.  The old shit phone system could do it.....
18:38.17apturaohh common
18:39.12apturascott, there is always a way. what old pbx do you use?
18:39.29*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
18:39.33mogNugget, vim7 has spellcheck
18:39.39mogfile: file
18:39.48mogjmls, i use gajim for all my messaging needs
18:40.56hmmhesayssomeone write me a t.38 passthru patch for asterisk 1.2.10
18:41.05[TK]D-Fenderscottmcl: : Theres something very wrong with your idea.  How the hell do you you expect to TALK to the other en first thereby making it an ATTENDED transfer?
18:42.08HolosCan anyone spot the error in this: Voicemail(${MATH(${EXTEN}-10|i))}@all|u)? It works fine if I take the |i or ,i out, but then I get XX.XXXXXX as a result.
18:42.11jmlsNugget: I said play nice !
18:42.23*** join/#asterisk xezz (n=xez@62.103.27.89)
18:42.25xezzhello
18:42.37xezzi 've just updated to asterisk 1.2.10
18:42.40jmlsxezz: welcome to the lions' den
18:42.40xezzlatest i think
18:42.55xezzand asterisk wont start :/
18:42.57jmlsisn't 1.2.12 the latest ?
18:43.03xezzyes
18:43.04hmmhesayst.38 passthru would be great
18:43.13xezzi've just updated to that
18:43.19[TK]D-FenderHolos: Voicemail($[${EXTEN}-10])@all|u)
18:43.23jmls12 <> 10
18:43.32xezzbut asterik isnt'getting up
18:43.40jmlsknow how he feels ;)
18:43.41xezzis there a known bug or something ?
18:43.56jmlspretty hard to say without you telling us *what* the problem is
18:43.57Holos[TK]D-Fender: So I don't need the MATH function?
18:44.13[TK]D-Fenderxezz: And you've told us SO much about the error messages its given you that we must clearly have everything we need to help you!
18:44.19[TK]D-Fender</sarcasm>
18:44.24[TK]D-FenderHolos: Nope.
18:44.26filemy sarcasm detector exploded
18:44.47jmlsmissed the opening <sarcasm>. Must be using vi again ..
18:45.14[TK]D-Fenderjmls: I reopen it regularly and it last for days before I expend enough to reset it ;)
18:45.25jmlsyikes
18:45.35*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
18:45.36xezz[TK]D-Fender : Cannot connect to Asterisk Manager with admin/amp111
18:45.46xezzThis module requires access to the Asterisk Manager. Please ensure Asterisk is running and access to the manager is available.
18:45.48jmlssee: we've scared the mog away
18:45.54[TK]D-Fenderxezz: That isn't an * startup problem.... try again...
18:46.05jmlsbzzzzt#
18:46.06[TK]D-Fender(you KNOW what I smell coming, don't you people?!)
18:46.10[TK]D-Fender<sarcasm>
18:46.15xezzAsterisk could not start!
18:46.15xezzUse 'tail /var/log/asterisk/full' to find out why.
18:46.16jmlstrixbox trixbox trixbox trixbox trixbox trixbox trixbox trixbox trixbox trixbox
18:46.22xezzhaha
18:47.09[TK]D-Fenderxezz : Give us something real to work with......
18:47.18xezzhow ?
18:47.23xezzthats what i know...
18:47.47xezzbefore updating to to 1.2.12
18:47.47[TK]D-Fenderxezz: Start * manually and see what it says.
18:47.50jmlswhat are you using to run asterisk ?
18:47.57xezzamportal start
18:48.09xezz./etc/init.d/asterisk start
18:48.13jmlssimply try asterisk -vvvvvvvvvvvvc
18:48.14[TK]D-Fenderxezz: Start it normally yourself or ask elsewhere....
18:48.17jmlsans see what errors you get
18:48.26jmls*and
18:48.29jmlsbloody vi
18:48.43jmlsno, not "* and bloody vi"
18:48.51jmls"*and" (bloody vi)
18:49.07xezzasterisk -vvvvvvc outputed about 30 pages
18:49.12xezz:/
18:49.13jmlsuh huh
18:49.19jmlswhat was the last line ?
18:49.34xezz<PROTECTED>
18:49.46jmlsthen back to the # prompt ?
18:50.04*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
18:50.06*** mode/#asterisk [+o mog] by ChanServ
18:50.22jmlsthe cat's back!
18:50.39moglol
18:50.40[TK]D-FenderYou know you can't jsut upgrade * on a trixbox install don't you?  it depends on Zaptel having a matching version, adnthat means recompiling EVERYTHING.  And BRI is NOT a normal part of Zaptel.
18:50.49mogjust upgrading my server
18:50.58jmls(for you non uk people - mog == moggy == cat)
18:51.17moghttp://en.wikipedia.org/wiki/Mog has lots of definitions
18:51.22mogbut none refering to me...
18:51.28mogyes
18:51.30moger yet
18:51.39jmlsI like Myelin oligodendrocyte glycoprotein.
18:51.43larryrichardsonOk, anybody with an idea on how to announce to a caller (in a queue) that "Agent #xxx is handling your call" when the transfer occurs?
18:51.44jmlssounds dangerous
18:52.20jmlslarryrichardson: there is an option for this. maybe in a patch. hold.
18:53.32xezz[TK]D-Fender
18:53.35xezzthats the error
18:53.37xezzUnable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)
18:53.56larryrichardsonthanks jmls
18:53.56[TK]D-Fenderxezz: That error isn't WHY * doesn't start, just REPORTS it.
18:54.15xezzk
18:54.27jmlslarryrichardson: http://bugs.digium.com/view.php?id=6910
18:54.35xezzhow can i find out why it isnt starting then ?
18:55.07jmlsxezz: it's a problem with your BRI stuff thingy. Which is not a part of asterisk.
18:55.07[TK]D-Fenderxezz: That error means nothing to us for your problem.  I jsut told you that it appears that you are nailed by module dependency and you can't jsut try and upgrade * out from under Trixbox like that without redoing everything.
18:57.07scottmclD-Fender : on the remote attended transfer all you are doing is simulating pressing the button on the phone from somewhere else...that is not the phone....so you talk to them the same as you normally do....i do not see why this is so complicated.
18:57.48larryrichardsonjmls: Thank you SOOOO MUCH! 3 days of searching completed!
18:58.11jmlslarryrichardson: chalk up one beer.
19:01.54*** join/#asterisk Splas (n=jwb@brooklyn.paravolve.net)
19:04.35*** join/#asterisk intralanman (n=lanman@pool-70-104-160-230.norf.east.verizon.net)
19:04.47larryrichardsonjlms: Does this mean it is in the current release of * ? Kind of ambiguious...
19:05.55*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
19:09.07sudhir492D-Fender: Paging is finally working for me :-)
19:10.33jmlslarryrichardson: No, it's not in yet - the patch has not been accepted into the main source tree just yet.
19:11.53larryrichardsonahhh... OK, i'll keep an eye on it then...
19:13.37[TK]D-Fendersudhir492: What did you do?
19:14.07*** join/#asterisk paulhuynh (n=paul@c-68-82-4-138.hsd1.de.comcast.net)
19:14.17paulhuynhgood afternoon
19:14.24paulhuynhI need help with my asterisk
19:14.48paulhuynhwe use TDM400P on our asterisk and reciever aout of echo
19:14.49*** join/#asterisk tmccrary (n=tmccrary@68-77-164-10.ded.ameritech.net)
19:14.59sudhir492There was a typo - at one place it was alertinfo instead of alertInfo . I told you I must be doing something stupid
19:15.04jmlslarryrichardson: although I think it needs a lot of work from reading it. I also think that it should be handled better. For example, the sound file should be passed as part of the addqueuemember or as part of the agent / member definition and added to the queue member structure. this would make coding a lot easier.
19:15.54paulhuynhany idea on how i can correct my echo issue
19:16.04tmccraryWith presence, Asterisk and Polycom phones, is it possible to have other phones (attendant) see when a phone is off hook? I've gotten it to say when phones are registered or not (online/offline) but I cannot get it to say when the phone is busy
19:16.09paulhuynhalso what si the power connect do for the tdm400p card?
19:17.17Holospaulhuynh: Powers the card (With FXS) and may be needed for more then 4 FXO's
19:17.25paulhuynhoh ok
19:17.45paulhuynhso i have 4 fxo soi do not need power or shoudl i plug it in anyway?
19:17.52HolosSangoma cards require external power when there are more then 4 FXO's installed, so Digium may be the same. PCI bus can't supply a lot of power, so I'd plug it in.
19:18.06paulhuynhOK
19:18.15paulhuynhhow about my echo issue
19:18.23HolosYou're receiving a lot of echo?
19:18.35paulhuynhvery bad echo on our side when call comein via zaptel
19:19.04paulhuynhyes i can hear myself talk on the phon but my client dont hear any echo at all
19:19.05HolosCall Digium support first.. they're there for that.  Other then that, make sure the card is tuned (zt_tune?) and that the Gain levels are correct. (This is a analog card right?)
19:19.20paulhuynhcorrect
19:19.38*** join/#asterisk _alex_mx_ (n=alex@200.78.229.18)
19:19.47paulhuynhdigium support i though it is just for install and not config?
19:20.09HolosEcho dectection and cancelation requires the correct gain levels before it really shines. Find a milliwatt test line, and tune the card.
19:20.11paulhuynhmy cureent gain level is set to 0.0
19:20.27Holospaulhuynh: Where are you from?
19:20.30paulhuynhwhat do you mean?
19:20.44paulhuynhhow do you do a milli watt test on the line?
19:20.58paulhuynhthis line should be very clean
19:21.14paulhuynhit coming off a T1 my telco provide it to it
19:21.23Holoshttp://www.voip-info.org/wiki/view/Asterisk+fxotune
19:21.39paulhuynhme and plug into a tdm400p 6feet away
19:22.35tmccrarySo, it looks like Asterisk does not support PUBLISH, so you cannot monitor line status from other phones
19:22.42diclophis-workwhats this all about: "Got SIP response 484 "Address Incomplete" "
19:23.02diclophis-worki am trying to dial from 1 asterisk machine to another using SIP...
19:23.29diclophis-workthe Dial command on the originating machine looks like this Dial(SIP/11231231234@stg-pbx)
19:23.53diclophis-workand on the recieving machine i have a extension in the dialplan for the context the calls get through .. i think
19:26.00diclophis-workarg.. my extensions were mismatched
19:26.05diclophis-worknow I am getting a 603 message..
19:26.09diclophis-work"declined"
19:26.51diablopicohello,,,, does anyone know if UNICALL is supposed to work with newer versions of * ???
19:29.34Holosdiablopico: http://www.sineapps.com/news.php?rssid=1510 says that 1.12.1 supports it.
19:29.36diclophis-workwhat is unical
19:30.17diablopicothanks Holos
19:31.37*** join/#asterisk aptura (n=s_a_l_e_@S010600a0c93f6f7e.vs.shawcable.net)
19:37.02paulhuynhhelp i get this
19:37.14paulhuynhwhen try to run fxotune
19:37.15paulhuynhTuning module 1
19:37.15paulhuynhCould not fill input buffer
19:37.15paulhuynh..........Failure!
19:37.15paulhuynhTuning module 2
19:37.34syzygyBSDcan asterisk run on a cluster?
19:37.41paulhuynhsome tim eit woudl said module 1 is OK
19:38.02syzygyBSDsay, to offload the compression to other boxes
19:38.22paulhuynhit randomly through out the different error for the module
19:40.51paulhuynhany taker?
19:40.53*** join/#asterisk afrosheen (n=cj@txprotoa2.august.net)
19:42.03apturascarry
19:47.53*** join/#asterisk southtel (n=slester@76.17.115.183)
19:49.16inv_arp[work]in sed how can i match the n'th occurence of a char? ex.. abcd1abcd  want to match the 2nd b
19:50.26tzafrir_homepaulhuynh, are you able to use asterisk with the card?
19:51.23tzafrir_homeinv_Arp, you can use a more complicated pattern:
19:52.04tzafrir_homeinv_Arp, s/b[^b]*\(b\)/something/
19:53.02tzafrir_homeOr do you want the second line that mhas a b?
19:54.53pablushmm
19:57.41pablusa cyber angel
19:57.46pablushmmm
20:02.07*** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66)
20:06.05*** join/#asterisk fiber0pti (n=John@207.114.199.107)
20:07.31afrosheenanyone familiar with sip-t
20:08.13afrosheeni.e. a sip t1?
20:10.21afrosheenarg where are the useful people these days
20:11.18trelanesip t1?
20:11.31trelaneafrosheen, plz 2 hand over what you're smoking
20:11.49trelaneyou hath drunk from the marketing koolaid
20:12.03afrosheensome itsp's are providing bundles of a t1 plus a sip trunk that goes straight into their pop
20:12.07*** join/#asterisk lirakis (n=tbright@h-68-165-94-219.nycmny83.covad.net)
20:12.09lirakishey all
20:12.14afrosheencommpartners is branding it 'sip direct'
20:12.24trelaneafrosheen, ok then call it a T1 wtih a sip trunk, not a SIP T1
20:12.42trelaneI have several such packages deployed wtih a local ISP
20:12.46afrosheensorry, the admin here has been calling it that all day like it's recognizable
20:12.47trelaneI'm using one in the office
20:12.52lirakisim trying to figure out how to send the "remote  party id" in the packet headers
20:12.56trelaneafrosheen, you know what to do then right?
20:13.01lirakishas any one done this, or know how to do it?
20:13.06trelanelirakis, SIPAddHeader()
20:13.08afrosheentrelane: yeah but my clue bat is too large to swing in here
20:13.23lirakisi have set "sendrpid=yes"
20:13.23trelaneafrosheen, then go down to maintainance and get a ball-peen hammer
20:13.27afrosheenhaha
20:13.41lirakistrelane: you will have to forgive my ignorance.. i dont really know where to use that function
20:13.47trelaneafrosheen, after liberal reprogramming with networking tool #4 (the hammer) your admin will be cluefull, or dead
20:13.51lirakistrelane: in the dial plan some where?
20:14.03lirakistrelane: extensions.conf?
20:14.05afrosheenanyway we currently have a normal bonded t1 that everything is sharing including phones and our call quality blows so were going with this bundled deal soon, wondering if there might be any gotchas
20:14.22*** join/#asterisk docelmo (n=vircuser@c-69-138-91-104.hsd1.de.comcast.net)
20:14.22trelanelirakis, before you call Dial() in the dialstring do a SipAddHeader(sip:<ip of sip server>/ :whaver you want in the header>
20:14.45lirakistrelane: hmmm
20:14.46lirakisokay
20:14.48trelane<PROTECTED>
20:14.49afrosheenwe're getting 2 sla's (1 from commpartners and 1 from wiltel) so I think we're safe so far
20:15.03trelaneafrosheen, ooh good so both can blame the other!
20:15.10hmmhesayscan anyone help me out with a crisco as5300?
20:15.30trelanehmmhesays, yeah, reduce the trans fats in your diet, crisco leads to heart attacks!
20:15.36hmmhesaysLOL
20:15.51hmmhesaysI need to figure out if this thing supports g.729 and if it is possible to enable
20:15.53docelmosay anyone know what the license is for FCC carriers?
20:15.58lirakistrelane: hmm.. okay
20:16.15trelanelirakis, that header allows auto answer
20:16.20lirakiswell.. im just getting ready to head out.. i will have to play with it some more on monday.. or possibly on my own pbx this weekend
20:16.33afrosheentrelane: so how are the packages working that your local ISP is doing
20:16.38trelanelirakis, you can do some pretty elite things with phones when you can control what headers you send
20:16.50trelaneafrosheen, on the grounds of self-amusement I plead the fifth
20:16.53trelane(not well)
20:17.00paulhuynhi can use the card with asterisk but echo is so bad on it
20:17.05lirakistrelane: .. so if i wanted to send a Passerted ID i could do that in the header with the same function right?
20:17.09trelaneafrosheen, he's already been a repeat-recipient of the clue-by-four award
20:17.12trelanelirakis, yep
20:17.27lirakistrelane: im having a little trouble because we have trixbox here.. and the dialplans are kinda.. huge and split up
20:17.35lirakistrelane: in know this isnt a trixbox channel..
20:18.08trelanelirakis, HEY LETS TAKE A REASONABLE DIALPLAN... AND PUT IN LOTS OF POINTLESS MACROS... AND THEN... AND THEN... WE'LL CALL IT TRIXBOX!
20:18.08lirakistrelane: .. but im having trouble figuring out where in the darn dial plan to put it... i guess i should watch the console when i make a call to see what it is executing
20:18.15lirakistrelane: yeah
20:18.15trelaneyep
20:18.16afrosheentrelane: but a point-to-point T1 into our sip provider's pop should be great right?
20:18.17lirakisi agree
20:18.25trelaneafrosheen, should be yeah :)
20:18.26lirakistrelane: .. but its what was here when i got here..
20:18.32*** join/#asterisk klasstek (n=nunyobiz@c-67-177-199-232.hsd1.co.comcast.net)
20:18.44trelanelirakis, convince the boss to let you start rewriting the dialplan from scratch
20:18.49lirakistrelane: okay.. i do need to run.. thanks for your help
20:18.52lirakisttyl
20:18.52tmccraryWHERE CAN I BUY THIS TRICKY BOX OF POINTLESS MACROS, I MUST KNOW
20:18.59lirakislol
20:18.59trelanetmccrary, damn right!
20:19.03afrosheenTHEY HAVE THEM AT THE CAPS LOCK STORE
20:19.14apturaI have a questions about my second line on my ip500. I want it to be my bussiness line and show my CID. The first line will be my residential zap line. I dont have access to the xml files so if anyone here cares to aid me in this would be appriciated.
20:19.16trelaneafrosheen, I SHOP THERE FREQUENTLY!!!!!
20:19.48trelaneaptura, contact the vendor?
20:19.50apturaThe second business line will be sip to my sip provider
20:19.53[TK]D-Fenderaptura: You can config it in the web gGUI just fine if you're already sticking to that lame-o road.
20:19.55afrosheenaptura, try hitting the web interface on the phone
20:20.14apturaafrosheen I rarely use that thing because of its problems.
20:20.27afrosheenwell if you won't use the xml files...or the web interface, what else is left
20:20.36[TK]D-Fenderheck you can do it right on the phone if you really care to.
20:20.56afrosheen[TK]D-Fender, I know you didn't just advocate the phone's menus for configuration
20:20.58[TK]D-Fenderafrosheen: Direct ont he phone physically
20:21.22[TK]D-Fenderafrosheen: You ASKED.  I never said I ADVISED it.  Please read what I say carefully :)
20:21.31trelaneheh
20:21.48paulhuynhi need help with some echo issue
20:21.52trelaneafrosheen, you could throw it out and get a snom
20:21.53apturaTK this is a area I have not delt with yet.
20:21.53[TK]D-Fenderafrosheen: Anybody configuring Polycom's outside of provisioning should be slapped.
20:21.54paulhuynhi have tdm400p
20:21.55trelanepaulhuynh, what hardware?
20:22.03trelanecall digium for free support or use fxotune
20:22.05paulhuynhanalog card'
20:22.11paulhuynhok
20:22.18paulhuynhwill do that ritgh now
20:22.40trelanethey're really good at getting the hardware working and since they're on the phone with you it's a quicker response (and anyway, you paid for it!)
20:23.29tmccraryD-fender, have you used Polycom phones before? Have you worked with the 601 expansion module?
20:24.58apturaI purchaced it from atacom
20:25.33[TK]D-Fendertmccrary: I own every model short of the new IP650.
20:25.51[TK]D-Fendertmccrary: I though my rep was universally known around here *sigh*
20:25.55trelaneaptura, atacom or atacomm?
20:26.19trelane[TK]D-Fender, it's good, between you and me there's a polycom expert and a snom expert
20:27.02tmccraryOk, with the 601 expansion module, I have mine setup so it can see when buddies are online or offline (reg'd or unreg'd). However, I cannot seem to get it to display when the line is busy or idle on the expansion module. When I run sip show subscriptions in asterisk, asterisk displays this info.
20:27.15tmccraryCan the expansion module do this with asterisk?
20:27.36*** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com)
20:28.29[TK]D-Fendertmccrary: IP601's come stock with SIP 1.6.3. and presence DISABLED entirely.  You need to enable it in your provisioning files, and then upgrade that phone to 1.6.6 or higher to be able to monitor more than a handful of people.
20:28.40[TK]D-Fendertmccrary: It most certainly can.
20:28.52[TK]D-Fendertmccrary:  So upgrade your firmware, and tweak your sip.cfg
20:28.59apturaatacomm
20:29.00tmccraryI upgraded to 2.05 or whatever (latest)
20:29.35tmccraryWhat function in sip.cfg allows this? I have <feature feature.1.name="presence" feature.1.enabled="1"
20:30.30tmccraryand the 601 *sees* the other phones as buddies, but does not seem to get notified when they are busy. But asterisk does
20:31.36apturaAtacomm has changed there page. In bod free shipping anywhere in the country :)
20:31.42apturabold
20:32.07[TK]D-Fendertmccrary: There is a bug between * and SIP 2.0.1 that was only resolved TODAY. YOu'll need the latest SVN for that combo to work.
20:32.16tmccraryaha, maybe its because reg.1.type is private?
20:32.28[TK]D-Fendertmccrary:  No, your reg is fine
20:32.34*** join/#asterisk bmd (n=bmd@64.50.19.206)
20:32.45tmccrarydamn that sucks
20:33.10C6VetteIs there a way to have a separate queue_log for each queue? I have 3 queues and would like them separated if possible.
20:33.45[TK]D-Fendertmccrary: Not SO bad... fix is out... just need to recompile, though I don't like working off anything but FTP official releases myself....
20:33.54tmccraryyeah, thats what I meant :)
20:33.55[TK]D-Fendertmccrary:  Or downgrade to 1.6.7
20:34.05[TK]D-FenderBBIAB, heading home
20:34.58*** join/#asterisk justinu|laptop (n=Justin@12.44.122.130)
20:35.32*** join/#asterisk sx-wks (n=sxpert@navsys.org)
20:36.45*** join/#asterisk brif8 (n=brif8@ns1.ttienterprises.org)
20:37.29brif8Hi All,  On a linksys PAP2,, anyone know the admin  username and password by default ?  I just bought one and it won't let me in ??
20:38.08trelanebrif8, RTFM?
20:38.22apturaAtacomm does not provide any configuration support
20:38.40apturaGuess its up to me anyway :)
20:38.47trelaneaptura, none of this stuff is THAT hard to configure!
20:38.57brif8trelane: I have repeatedly it only talks about my routers access which isnt a linksys
20:39.23trelaneso you bought a PAP2... and the PAP2 instructions don't contain the login and password infor for the PAP2?
20:39.56*** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net)
20:41.07*** join/#asterisk SexyKen (n=Ken@c-71-202-149-39.hsd1.ca.comcast.net)
20:41.13SexyKenHey guys -- have you heard of Trixbox?
20:41.25TrixVoxnope
20:41.30TrixVox:)
20:41.46trelaneSexyKen, read the topic
20:42.03trelane<PROTECTED>
20:42.22SexyKenI see.
20:42.23SexyKen:-)
20:42.53trelanebrif8, where did you get the PAP2?
20:42.55SexyKenSorry about that :-)
20:42.58trelaneno worries
20:43.21brif8trelane: radioshack  I think it wants me to get a vonage account and I don't want one
20:43.31trelaneyeah it appears locked to vonage
20:43.53trelanewhich will be going out of business any day now
20:44.05trelaneworst ipo ever
20:44.20apturavonage is giving voip a bad name
20:45.19tmccraryI looked into become a vonage reseller a few years ago
20:45.27tmccraryyou had to have 300k customers+
20:45.34*** part/#asterisk bmd (n=bmd@64.50.19.206)
20:45.56pifiu-laptop<[TK]D-Fender> pifiu-laptop: EW..... hope you didn't spend personal money on it....
20:45.58pifiu-laptopno i didnt
20:46.04pifiu-laptopbut why are they that bad? lol
20:46.56tmccraryWhat?
20:48.22brif8trelane: yeah so any ideas ?
20:49.13trelanebrif8, good luck, you'll need it
20:49.50*** part/#asterisk acrg (n=aragon@decoder.geek.sh)
20:52.49*** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2)
20:53.07*** join/#asterisk Norskman (n=jack@jackbuechler.plus.com)
20:53.38Norskmananyone know anything about SIP  versions of the Cisco 7970 V8.04SR1 - I need to know where ot place my files
20:54.17anthonylare you asking about the firmware
20:54.20*** join/#asterisk ikey (i=ikey@220.226.35.125)
20:54.31*** join/#asterisk AvoidingDeadlock (n=ASSERT_K@adsl-69-219-50-18.dsl.chcgil.ameritech.net)
20:55.11*** join/#asterisk ToTo (n=ToTo@host138-138-dynamic.2-87-r.retail.telecomitalia.it)
20:55.20NorskmanTHe SEP000F90CEF9BF.cnf.xml file - my TFTP server says it can;t find it even thouh its in the same directory as the f/w and it loaded that into the phone just fine
20:55.38*** part/#asterisk brif8 (n=brif8@ns1.ttienterprises.org)
20:56.13*** part/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net)
20:57.04anthonylis the file name correct and matching the one the phone is sending via the tftp request?
20:57.16Norskmanyes 100%.
20:58.07Norskmane.g. it can send and the phone receives the XmlDefault.cnf.xml file which is in the directory
20:58.08anthonylthe error is on the server side?
20:58.19anthonylit could be a permissions thing
20:58.31funkmasteris any1 here making use of skypho?
20:58.47NorskmanNot 100% - the file is in the right place. No SEPfile from the TFTP server, no file in the phone recieved =no login to server
20:59.08Norskmanwhere on the permissions, I am using a windows Pc to send the fiules.
21:00.24Norskmana read request is being made of that file to the TFTP server. then the TFTP server says File <Sep....> error 2 in system call Create file cannot find the filespecified
21:00.36Norskmanand now its looping looking for the config file.
21:00.41anthonylwell if the request is not specificying any directory it is just trying to grab it from your tftp-root
21:00.47*** join/#asterisk dahunter3 (n=dahunter@pool-71-110-86-116.lsanca.dsl-w.verizon.net)
21:01.07NorskmanTFTProot being the directory pointed at by the TFTP server?
21:01.59anthonylya
21:02.19NorskmanI thought so..
21:02.53NorskmanSO I am doing it right and its absolutely driving me mad. I have spent weeks trying to get this work and still no luck. Can;t work out where the problem is.
21:03.03NorskmanLoads of weird error messages.
21:03.17*** join/#asterisk hardwire (n=hardwire@89-208-58-66.gci.net)
21:03.19hardwireBOO
21:03.21anthonylhave you used wireshark to double check to make sure things are 100%?
21:03.36Norskmanwhats wireshark please?
21:03.55hmmhesaysso who wants to help me with this as5300
21:04.04anthonylNorskman,  it's a packetsniffer
21:04.09anthonylwireshark.org
21:04.11hmmhesaysi'm having a helluva time with it
21:04.34Norskmanok will look.  I have ethereal loaded - presumably thats ok as well
21:04.39NorskmanSame sort of ting
21:06.48*** part/#asterisk Norskman (n=jack@jackbuechler.plus.com)
21:09.12*** join/#asterisk Optic (n=dfraser@miso.capybara.org)
21:09.20Optici'm having weird segfaults :(
21:09.44OpticConnected to Asterisk SVN-branch-1.2-r44580 currently running on operator (pid = 18701)
21:09.51Opticbacktrace here: http://pastebin.ca/193400
21:09.56*** part/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com)
21:10.11anthonylwireshark is the new-er etherreal
21:10.13Opticany ideas?
21:10.49anthonylOptic, humm
21:11.03anthonylwhat version of asterisk?
21:11.08Opticsee above :)
21:11.13OpticSVN-branch-1.2-r44580
21:11.42Optici just svn update'd and rebuilt today because of the crashing, but i've had a crash on a rebuilt verson too
21:11.48Opticbasically it's today's svn stable
21:12.03Opticit seems to be crashing every couple of hours
21:12.39C6Vetteno bt full
21:12.51Optici can do a bt full for you :)
21:12.51anthonylOptic, has this machine been stable for a while before?
21:12.58Opticyes
21:13.32anthonyli would submi a report to bugs.digium.com with the bt full attached
21:14.36anthonylsubmit*
21:14.37tzafrir_homeOptic, do you get any core file?
21:14.39Opticyes
21:14.55Opticis there a way to turn off gdb's pager or send the bt full to stdout non-interactively?
21:14.58Opticit's long :)
21:17.05C6VetteI just turn on the logger in the ssh client. If your not local that is.
21:17.43Opticah
21:17.53*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
21:17.55Opticcat >gdb.cmds
21:17.59Opticset pagination off
21:18.01Opticbt full
21:18.06Optic^D
21:18.13Opticgdb -x gdb.cmds asterisk corefile
21:18.14Optic:)
21:20.06*** part/#asterisk tmccrary (n=tmccrary@68-77-164-10.ded.ameritech.net)
21:21.21Optichttp://pastebin.ca/193413
21:21.26Opticnow w/bt full
21:22.24Opticyes
21:22.42Opticyes, if i'm getting a backtrace :)
21:22.51Opticoops :)
21:25.03C6Vetteyea,.. I would post a bug with the bt full
21:25.25Opticok
21:25.48Optici mean, it's possible that it's a hardware problem with the box or something
21:25.51Opticbut I feel like it isn't :)
21:26.31C6VetteIf it was working finr before the update i would ASSume it wasnt the hardware.
21:27.06C6VetteSee some memory errors
21:27.13C6Vettelooks like to me.
21:30.07Opticyeah
21:30.15C6VetteIm having memory leak problems on one of mine. at the end of the day I have only a few k available
21:30.17Opticthere, posted :)
21:30.22Optichttp://bugs.digium.com/view.php?id=8109
21:30.30Opticthis is a pretty busy site too, 30 or so users
21:30.34Opticand it's normally pretty stable :(
21:30.51C6VetteIm running aroung 70 users
21:30.55Opticnice :)
21:31.04Opticwhat's your PSTN connection?
21:31.11Opticwe're 10 channels on a PRI here
21:31.13C6VetteVOIP sip
21:31.21C6Vetterunning about 12000 calls an hour
21:31.26Opticwow
21:31.33Opticthat's awesome :)
21:32.10anthonylOptic,  cool
21:32.12C6Vetteits a handfull sometimes
21:32.58apturaman
21:33.25apturaand asterisk is the box handeling these calls?
21:34.03C6Vetteyea. Im wanting to split it into 2 servers one for handing all the sip clients and one to handle the outbound/inbound calls
21:34.23apturanow that i have my ftpserver shutdown the ip500 is not locking up. Time to make the changes
21:34.23Opticok, i'm out
21:34.26Opticthanks for your help ;)_
21:35.25C6Vettenp L8r
21:36.30*** join/#asterisk slobberknocker (n=slobberk@63.149.122.93)
21:37.46slobberknockerhas anyone experienced poor quality music on hold with the files included with asterisk sounds? I have played them on the pc and the sound fine, but when listening to them on hold they have intermittent periods that sound like static.
21:38.28*** join/#asterisk xnon (n=xnon@200.8.86.187)
21:40.59*** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net)
21:41.00TripleFFFF<PROTECTED>
21:41.05TripleFFFFegetting lots of these
21:41.51*** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com)
21:42.24xnonanybody know any easy to install billing system for asterisk?
21:42.27*** join/#asterisk skopii (n=skopii@fuckcingularwireless.com)
21:42.40TripleFFFFFrom: "Unknown" <sip:Unknown@74.13.210.125>;tag=as723c47f2
21:42.41TripleFFFFlol
21:42.44TripleFFFFok !!!
21:42.54TripleFFFFsomeone trying to connect as a guest lol darn sniffers
21:43.24aydiosmioit begins
21:43.39aydiosmioadd tor and you've got a giant newwork of fraud-phones
21:43.44aydiosmionetwork
21:45.01TripleFFFFtor ?
21:45.20TripleFFFFill push all the non auth invites to FBI Lol
21:45.28skopiiHello, I am looking to setup a pbx with sales,and support queues. I have got asterisk installed and am playing around with freepbx. What I dont get is whether or not I need to buy a PRI+Digium Wildcard or if I can use a SIP trunk for both inbound and outbound calls.
21:45.45TripleFFFFtext to speeach ("Im a hacker my ip is X.X.X.X please arrest my dumb ass for being so lame ")
21:46.17slobberknockeranyone on the moh question?
21:46.48C6Vetteslobberknocker, using mpg123?
21:46.54slobberknockernative
21:46.55*** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net)
21:46.59*** join/#asterisk VOIPoTD (n=td@adsl-75-42-28-214.dsl.hstntx.sbcglobal.net)
21:47.06C6Vettegood
21:47.09[TK]D-Fenderskopii : You can do it any way you want.  VoIP only / PRI / Analog / Whatever.
21:47.16fiber0ptiJust developed a new operator panel for asterisk.. looking for feedback. Anyone want to install it?
21:47.28slobberknockerfiber0pti:
21:47.31slobberknockeri am interested
21:47.41C6Vettefiber0pti: how does it connect ?
21:47.56fiber0ptiIt connects via the Asterisk manager
21:48.01skopiican you tell me why in gods name anyone would want to use a PRI if they could use VOIP then?
21:48.21fiber0ptislobberknocker: if you would like to try it out please go to i9technologies.com/isymphony
21:48.25fiber0ptidownload the server and the client
21:48.32fiber0ptiThey both require java.
21:48.32*** join/#asterisk awannabe (n=brad@ip24-251-135-202.ph.ph.cox.net)
21:48.38slobberknockerok, i will play with it tonight.
21:49.23awannabehas anyone used a adtran atlas 550 for testing FXO and PRI cards in *?
21:49.34*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)
21:50.08fiber0ptislobberknocker: I should be on all weekend.. Please feel free to send feedback
21:50.24fiber0ptiWe're going to be changing the UI within a week.. should be much better than it is now
21:50.40*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)
21:50.43slobberknockerok
21:51.43slobberknockerfiber0pti: is this first revision?
21:52.21fiber0ptislobberknocker: it is.
21:53.33slobberknockerno ideas on my moh sound quality then eh?
21:54.30[TK]D-Fenderskopii VoIP is considerably less reliable, you have to pay for both bandwidth and the service you run on it.
21:55.01[TK]D-Fenderskopii :Higher sound quality (max), and many other reasons
21:59.55*** part/#asterisk slobberknocker (n=slobberk@63.149.122.93)
22:01.46*** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net)
22:02.36skopii[TK]D-Fender: is it possible to setup rendunant calls on the inbound sort of active/active load-balanced config?
22:02.44*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
22:03.02skopiithen use the same setup for outbound with a PRI as a failover?
22:04.02wmandrais anyone here using asterisk with LCS??
22:04.43xnonanybody know php-pcntl?
22:05.00xnoni cant find these pack!
22:05.47wmandrai'm working on a CSTA Gateway for * that allows pc to pstn and remote call control with LCS and communicator, and just wanted to see if there was any interest .......
22:05.48xnonanybody know any biling system for asterisk easy install!
22:05.56*** join/#asterisk droops (n=root@adsl-074-245-001-031.sip.jan.bellsouth.net)
22:06.09*** join/#asterisk schirpich (n=kvirc@66.238.64.20)
22:06.32*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
22:08.35qdkwmandra: LCS?
22:09.37wmandralive communications server
22:10.54qdkwmandra: and what's that?
22:11.27wmandraqdk: http://www.microsoft.com/office/livecomm/prodinfo/default.mspx
22:11.54wmandrait's basically an internal IM server, but alos has phone control capabilities
22:12.28wmandrait's actually really nice.... i've been using it here and i love it
22:13.57*** join/#asterisk necudeco (n=chatzill@190.40.219.33)
22:13.58*** join/#asterisk ShadowHntr (i=sentinel@c-68-52-48-169.hsd1.tn.comcast.net)
22:14.02necudecohi
22:14.19necudecoI need help to install asterisk in debian
22:14.19qdkwmandra: sounds expensive... and i still have no idea of what its good for.
22:15.38wmandrathe communicator client is basically an IM client / softphone rolled into one. the nice thing is how it integrates into outlook
22:15.44qdknecudeco: so right about now you expect people to spoonfeed you?
22:16.07qdkwmandra: so you can place a call from outlook?
22:16.41wmandraother users that have me added to their contact list can see my presence status in real time (ie. in a meeting, on the phone)
22:17.28wmandraif a call comes into my * deskphone a popup appears with the callerid and i have the option to answer the call or forward it to another number like my cell
22:18.04qdkwmandra: ok, sounds a bit like a feature i have planed for my solution using *, web and other quite free tools.
22:22.01*** join/#asterisk dahunter3 (n=dahunter@pool-71-110-86-116.lsanca.dsl-w.verizon.net)
22:22.13*** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net)
22:24.06*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
22:24.44*** join/#asterisk BlepsoaF (n=pbaker@nnat-gw.adeptra.com)
22:25.31BlepsoaFHello all, I've had experience with Cisco 79XX phones before, but soon I will be moving to a new office and have the chance to purchase new equip.  Can anyone make any recommendations about good SIP phones for the price to use with asterisk, or has anyone had experience with Polycom phones?
22:26.53apturapolycom are good
22:27.07apturaaastra are also good
22:28.13*** join/#asterisk syzygyBSD_ (n=chatzill@66.226.228.204.cpe.speedyquick.net)
22:28.26afrosheenyeah the polycom ip601's and 501's are nice
22:28.44afrosheenthey're a pain in the ass to learn at the beginning but once you get provisioning down they're easy enough to work with
22:28.51apturajust beware that if you dont do line two right it may lockup. im trying to undoo mine right now.
22:29.12apturamine is locking up now afrosheen
22:29.29afrosheenyour polycom?
22:30.17ShadowHntri'm considering grandstream phones here
22:30.21BlepsoaFhmm
22:30.40ShadowHntrthe polycoms look nice
22:30.43apturayea
22:30.44ShadowHntrbut they're out of my price range
22:30.46apturamy ip500
22:31.05afrosheenShadowHntr, are you doing this for your home?
22:31.15apturai added a second line..going out to my provider. I just deleted it and rebooting the phone. Hopfully this fixed it.
22:31.16ShadowHntrafrosheen: yeah - home and/or home office.
22:31.27ShadowHntri've been researching
22:31.32afrosheenthe ip300 or the newer 430 are probably what you want then
22:31.41ShadowHntrand it seems like the deal is VoicePulse
22:31.51ShadowHntrand maybe move to Voicepulse Connect for Asterisk later on
22:32.17ShadowHntrmy ultimate goal is to put an IP phone in each room
22:32.25ShadowHntrtotal of 3 or 4 phones
22:32.29ShadowHntrusing Asterisk
22:33.20afrosheenwell if you're going to be cheap just buy some iaxy's from digium, plug a standard phone in
22:33.30afrosheenthat way you get cordless phones also
22:33.31ShadowHntrnah i'd like them all to be SIP phones
22:33.44afrosheenwell they'll be iax phones which is even better :)
22:34.25ShadowHntrwell
22:34.33ShadowHntri'm not gonna be ready for this for probably a few months
22:34.37afrosheenor if you want purely sip phones, the polycom 430 is around $149 each, 2 lines and a speakerphone
22:34.41justinu|laptopiaxy's aren't exactly cheap
22:34.43*** join/#asterisk robin_z (n=you@adsl.redpoint.org.uk)
22:35.10afrosheeniaxy's are about $85 each
22:35.23afrosheenI think you should get at least one just for the portable phone aspect
22:35.29justinu|laptopi'd get a real SIP ATA for less
22:35.56justinu|laptopjust for the fact that real ATAs can resolve a DNS name, and the iaxy can't
22:35.59afrosheenwe have an iaxy we give to our road warriors, you never know what kind of firewall they'll end up with at whatever hotel or coffee shop
22:36.39afrosheenand you can trust that an iaxy will gaijin smash anything on it's way to the server
22:38.50apturahave your road warriers come up against any walls and could not get though?
22:39.57apturaThis is really sucking. Man If I had the forsight of this phone locking up a head of time I would not have had to make these changes.
22:40.39afrosheenaptura, not yet, double nat is a joke for iaxy also
22:41.28robin_zjust avoid Sipura SPA2102s when buying ATAs
22:42.04robin_zabout as much fun as slamming your fingers in the desk drawer
22:42.08apturaokay i had to race though these menues on the phone before it locked up..a 10th time.
22:42.24apturaOkay looks like it may have worked.
22:42.32*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
22:43.25*** join/#asterisk kink0 (n=k@161.pool62-37-205.static.uni2.es)
22:43.30*** join/#asterisk iCEBrkr (i=icebrkr@69.9.167.70)
22:43.33iCEBrkr<PROTECTED>
22:43.36iCEBrkrhuh?
22:43.54*** join/#asterisk alkiser (n=support@bdsl.66.14.163.224.gte.net)
22:44.55kink0hello
22:45.57iCEBrkrAnyone else having problems with VoicePulse?
22:47.59*** join/#asterisk [hC] (n=hardcore@66.119.167.162)
22:48.24iCEBrkrhrrm, apparently, it's fixed now
22:48.32iCEBrkrIt just said it's reachable
22:48.42iCEBrkrbut I'm still getting  Got SIP response 481 "Call/Transaction Does Not Exist" back from 2
22:49.54apturanew issues?
22:50.34*** join/#asterisk chief3rd (i=ExUser@cuscon30833.tstt.net.tt)
22:50.46TrixVoxiCEBrkr: Working okay here... looks like they just lowered their rates or something though.
22:51.01iCEBrkrIt seems as if they just fixed it
22:51.21iCEBrkrBut whatever that Call/Transaction does not exist message is new to me.
22:51.37iCEBrkrI can make calls
22:51.44*** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com)
22:51.45iCEBrkroh well
22:52.21TrixVoxnice, most of europe was lowered... canada too
22:56.01*** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net)
22:56.14TripleFFFFgot booted off lol
22:58.12*** join/#asterisk vilito (n=lsackett@68.80.4.115)
23:01.04*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
23:01.04*** mode/#asterisk [+o anthm] by ChanServ
23:02.56*** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6)
23:03.24*** join/#asterisk spr1te (n=spr1te@quarta.synapse.net.ua)
23:07.07fiber0ptiAnyone want to give me feedback on a java operator panel?
23:07.22rpmfiber0pti: java operator panel?
23:07.25TripleFFFFhate hjava
23:07.49fiber0ptirpm: Nod. It runs on java.. both server and client
23:07.58fiber0ptiTripleFFFF: What operator panel do you use?
23:11.03TripleFFFF? vi
23:11.13TripleFFFFvi is the best panel there is..
23:11.18TripleFFFFits like notepad but better
23:11.32TripleFFFFand get the job done with less cpu then any other means
23:13.22*** join/#asterisk AdmoIRC (n=Miranda@user-0c93s2v.cable.mindspring.com)
23:15.01vilitohi all, i have an interesting problem.  i built and install asterisk 1.2.12.1 and everything works fine except that everytime i reboot, the /var/run/asterisk dir is removed
23:15.34vilitoi just did a df and i see that varrun is not on the regular fs, but seems to be like a tmp fs.
23:16.07vilitomy question is, how do i prevent or ensure that /var/run/asterisk is always present before asterisk runs?
23:16.49*** join/#asterisk linuxmigration_ (n=jeremy@dsl254-075-124.nyc1.dsl.speakeasy.net)
23:19.02*** part/#asterisk robin_z (n=you@adsl.redpoint.org.uk)
23:20.27*** join/#asterisk SwK (n=Silik0nJ@12-214-191-109.client.mchsi.com)
23:27.12iCEBrkrisn't /var/run/asterisk the lock file so it knows that it IS running
23:31.35apturaman people can be like sheep not so observant that the tenant was dead for three years in the apt and no one knew. The persons pension continiosly paid the rent.
23:34.14*** join/#asterisk bmg505 (n=leon@c1-236-7.rndf.isadsl.co.za)
23:37.16*** join/#asterisk RoyK (n=roy@gprs-ggsn5-nat.mobil.telenor.no)
23:38.37*** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com)
23:40.04*** join/#asterisk Dr-Linux|work (n=Linux@202.59.73.131)
23:42.51BlepsoaFvilito: I'm not sure why you would wnat to do that, but write a wrapper script in bash
23:50.53*** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue)
23:50.55EyeCueasterisk-1.2.9.1_1          <  needs updating (port has 1.2.12.1)
23:50.56EyeCueooo :D
23:56.51*** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org)
23:57.24*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org)
23:58.38*** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net)
23:58.40BlepsoaFhow is 1.4 anyhow
23:58.54BlepsoaFhopefully it goes production ready by the time I need to move into my new office
23:59.34VOIPoTDAny consultants here or someone with a few hours know AEL2 very well?
23:59.43VOIPoTDIf so, pm me

Generated by irclog2html.pl by Jeff Waugh - find it at freshmeat.net! Modified by Tim Riker to work with blootbot logs, split per channel, etc.