IRC log for #asterisk on 20080303

00:00.21kn0xyes...
00:00.26*** join/#asterisk PepOSX (n=angeldav@190.72.147.233)
00:01.25*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
00:01.43*** join/#asterisk zippytech (n=chatzill@dns2.zippytech.com)
00:02.28zippytechany one seen this TRUNK Dial failed due to CHANUNAVAIL
00:03.42JTLemensTS: no, it converts SIP to FXS
00:04.05JTwhy would you need another channel bank?
00:05.12zippytechfailing through to other trunks
00:05.39*** join/#asterisk ninazu (n=who@cpe-67-10-211-160.satx.res.rr.com)
00:06.26jameswf-home~freeswitch
00:06.26jbotfreeswitch is probably an open source soft switch that is *not* a fork of asterisk http://www.freeswitch.org/interview2.htm
00:06.32*** join/#asterisk nighty^ (n=nighty@210.188.173.246)
00:09.36zippytech<PROTECTED>
00:09.37zippytech<PROTECTED>
00:09.39zippytech<PROTECTED>
00:10.04zippytechwhy would it not show an channels
00:10.12zippytechthere should be 4
00:10.33ManxPowermaybe your /etc/asterisk/zapata.conf is not correct?
00:11.49jameswf-homeimagine that
00:15.50robmac67zippytech: what card do you have installed ?
00:16.25angryusergn all ;)
00:17.39*** join/#asterisk craigk (n=craigk@58.174.150.119)
00:22.21Greek-Boyis there a channel state that is the opposite of CHANUNAVAIL? I want to use a GotoIF statement provided there is one available channel in a trunk.
00:24.10*** join/#asterisk CrazyTux[m] (n=CrazyTux@76-204-200-226.lightspeed.hstntx.sbcglobal.net)
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00:28.43*** join/#asterisk duri (n=mduregon@c-76-105-157-51.hsd1.or.comcast.net)
00:29.11duriwhat kind of load can a WRT54GS handle ?
00:29.34*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-0470144e074db085)
00:38.21ManxPowerGreek-Boy: No.
00:38.33ManxPowerBut you can use the app ChanIsAvail
00:39.02ManxPowerGreek-Boy: you sure do like to make things complicated
00:40.42*** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com)
00:48.05ManxPowerThey're not killers, just misunderstood.
00:48.11ManxPowerI blame it on the schools.
00:48.28jameswf-homeI blame the parents
00:48.31*** join/#asterisk egecko (n=sam@cpe-76-176-238-225.san.res.rr.com)
00:49.21JerJeri blame canada
00:50.00JerJeri can hear them up there at night sharpening their ice skates, just waiting for the perfect time to attack
00:50.19ManxPowerI wish they would hurry up.  We could use some decent beer.
00:51.54Greek-BoyManxPower once an extensions reaches hangup() does it ignore priorities after that?
00:53.01Greek-Boyi need to do something after a call is hung up
00:53.25lunaphytewhat is smdi?
00:54.07JerJeri think smdi is a serial connection to legacy PBXs
00:54.15lunaphyteoh, ok.
00:54.53JerJerStation Message Detail Interface
00:55.58lunaphyteah, thanks.
00:56.27lunaphyteoh, is it maybe Simplified Message Desk Interface?  http://en.wikipedia.org/wiki/SMDI
00:56.28*** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com)
00:56.38lunaphyteor is that something else with the same acronym?
01:02.00cleoneany one here have iaxcomm?
01:02.58Greek-Boywhen an extension reaches hangup() does it ignore priorities after that?
01:03.19adeelwhy does the polycom digit mapping feature suck so hard?
01:06.17*** join/#asterisk RoyK (n=roy@ti211110a081-7661.bb.online.no)
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01:18.15*** part/#asterisk beek (n=klinebl@65.211.106.243)
01:19.25Greek-Boysomeone please take a look at http://pastebin.com/d2f881e
01:19.44Greek-Boyi am trying to setup a macro in my dial plan to handle emergency calls
01:19.57Greek-Boybut I seem to be going off track big time
01:24.34Greek-Boyforget that url. check http://pastebin.com/d45525822
01:26.25rajivwhats the equivalent of 'core show locks' in 1.2.x ?
01:30.00ManxPowerrajiv: there isn't.
01:30.34ManxPowerthat feature (invented by rusellb, I think) is new in 1.6.x, it may have been backported to 1.4.
01:31.12rajivany ideas what to do about: WARNING[14800]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0x81aae58', 9 retri
01:33.49Greek-BoyManxPower can u check http://pastebin.com/d45525822 for me?
01:33.50Greek-Boypls
01:33.58ManxPowerno
01:34.02Greek-Boylol
01:34.03Greek-Boyk
01:34.04Greek-Boy:)
01:35.34Greek-BoyManxPower: Can you tell me if hangup() waits until the call is over before going to the next priority?
01:36.19lunaphytewhy might asterisk not load a module after upgrading from 1.2 to 1.4?  http://rafb.net/p/qaTgnw89.html
01:38.26ManxPower1.2 binary modules do not work with 1.4
01:39.26ManxPowerOne of the critera FOR an increase in the tenth's digit is breaking binary compatability.
01:40.56lunaphyteoh, ok.
01:44.27lunaphytei have an older 12sp+ that i was using with chan_sccp, which was (is?) a third party module.  might i be able to use it with asterisk without the need for the additional module anymore?
01:47.40*** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6)
01:47.42ManxPowermost modules have newer releases that support 1.4
01:47.56*** join/#asterisk tengulre (n=tengulre@125.71.208.16)
01:49.15lunaphyteyeah, i was just looking at the site for this module.  it seems to be a bit neglected.
01:51.35*** join/#asterisk akira2014 (n=chatzill@172.Red-88-8-198.dynamicIP.rima-tde.net)
01:51.44akira2014hello
01:51.57jameswf-homeoh snap
01:52.50*** join/#asterisk MoreAllLess (n=jackjust@cpe-76-169-131-140.socal.res.rr.com)
01:52.52akira2014jameswf-home: ?
01:53.06*** join/#asterisk BeeBuu (n=beebuu@219.135.42.4)
01:54.04jameswf-home~random
01:54.20akira2014:)
01:54.49akira2014jameswf-home: can you help me ( another time ) with zaptel
01:54.51akira2014?
01:54.58akira2014or some one else, plz
01:55.46BeeBuucan i dial someone and invite he/she to meet?
01:56.53akira2014as tzafrir and others tell me i've compiled zaptel, libpri & Asterisk from sources....
01:59.04akira2014but i continue without being able to pick up calls
01:59.08akira2014any ideas ?
02:01.28jameswf-homenow killer jelly fish
02:03.58*** join/#asterisk seanbright-home (n=seanbrig@c-69-251-175-43.hsd1.md.comcast.net)
02:04.03anonymouz666Asterisk native wav format is 16bit or 8bit?
02:05.10*** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net)
02:05.11jameswf-homeusualy 8 is the magic number
02:06.26drmessano3 is the magic number, per Schoolhouse Rock
02:06.44jameswf-home1 is the lonliest number
02:06.56_charly_42 is the answer
02:07.16drmessanoHmm.. if only I knew the question
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02:26.32*** join/#asterisk jmacz (n=jmacz@190.25.38.26)
02:33.25jmaczHi everyone, I'm having some problems with an * box which is terminating some incoming calls over a PRI because of the Telco sends a Destination out of order (27) HANGUP cause.
02:33.38jmaczThis does not happens with the old PBX nor the ISDN Tester. Any ideas what it might be?
02:35.06*** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id)
02:37.22*** join/#asterisk Daviey (n=dave@ubuntu/member/daviey)
02:41.35jameswf-homejmacz: http://www.cisco.com/warp/public/121/cause_codes.html
02:42.13*** join/#asterisk [hC] (n=hardcore@S0106001346a4b813.vc.shawcable.net)
02:42.29Greek-BoyI think I finally figured out a solution for emergency calls
02:42.54Greek-BoyI have this in my dial plan [macro-dialemergency]
02:42.55Greek-Boyexten => s,n,GotoIf($["${CHANNEL}" = "${ARG1}"] && $["${DIALSTATUS}}" = "CHANUNAVAIL"] && $["${EMERGENCY_STATE}" = "0"]$?softhangup:setem)
02:42.55Greek-Boyexten => s,n(softhangup),SoftHangup(${ARG1}|a)
02:42.55Greek-Boyexten => s,n,Wait(3)
02:42.55Greek-Boyexten => s,n(setem),Set(GLOBAL(EMERGENCY_STATE)=1)
02:42.56Greek-Boyexten => s,n,Monitor(wav,${EMERGENCY_REC_CALLFILENAME},m)
02:42.58Greek-Boyexten => s,n(dial),Dial(${ARG2})
02:43.00Greek-Boyexten => s,n,Hangup()
02:43.02Greek-Boyexten => s,n,Set(GLOBAL(EMERGENCY_STATE)=0)
02:43.04Greek-Boyoops
02:43.06Greek-Boyi was going to paste a pastepin url
02:43.14Greek-Boyhttp://pastebin.com/d407638e2
02:43.19jameswf-homegah
02:43.24Greek-Boysorry :(
02:43.45Greek-Boymy ctrl+c key shortcut didn't help me on that one
02:44.31jmaczjameswf-home, had already checked that link. We tested the ISDN line with the Telco and it seems OK (passed the physical and data link tests). Besides, the Telco only sends that hangup cause when the call comes from other service provider.
02:45.39jmaczjameswf-home, say I got my PRI from provider A, if the call comes from provider B or C (specially long distance calls), the call is dropped but only with Asterisk (as I said, the call is not dropped with the old PBX nor the ISDN tester).
02:46.18jmaczany ideas? (the thing is driving me nuts)
02:49.46lmadsenGreek-Boy: not that: 1) your context doesn't start with a priority 1, 2) you have a type on the first line ${DIALSTATUS}}  <-- 2 ending curly braces, 3) same line, you have a $? at the end.... you only need ?, 4) you should get into the habit of using commas instead of pipes to separate as pipes went away in 1.6
02:49.55lmadsens/not that/note that
02:51.16*** join/#asterisk joobie (n=joobie@joobie.org)
02:51.18joobiehey guys
02:51.51joobieafter a basic phone that is good quality for SIP.. ive been looking at the "Polycom SoundPoint IP 320 Phone" phones. Anyone got any recommendations, or is this a good phone to be looking at?
02:51.59joobiethere's also the 330 phone.. not sure which is better
02:52.05joobieprice is $50 more for the 330
02:52.15BeeBuuhow can i dial someone's phone invite he/she to meet?
02:53.07lmadsenBeeBuu: stay away from the he/she's.... bad news
02:53.27BeeBuulmadsen: what's wrong?
02:53.36lmadsenI don't understand your question
02:53.57lmadsenand I'm going to bed... g'night!
02:54.25BeeBuufunny guy...
02:55.17BeeBuu~book
02:55.17jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
02:56.53joobieguys - just about to buy a polycom phone
02:57.01*** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com)
02:57.12joobiewondering if i want to use the internet to dial out to the sip provider.. all i need is the phone yea and a box that has ethernet?
02:57.18joobieone for the wan and one for the LAN
02:57.18joobie?
02:58.58Greek-Boylmadsen thanks for your help
02:59.05jameswf-homejust 1 nic
02:59.09Greek-Boylmadsen either than those problems will it work?
03:00.17joobiethanks james
03:00.21joobiealso is that the best basic phone
03:00.23joobiethe polycom?
03:00.30joobiei just want something that is clear.. and simple.. and works well
03:00.46*** join/#asterisk Daviey (n=dave@ubuntu/member/daviey)
03:01.32jameswf-homepolycom is most expensive
03:01.42jameswf-homebasic I would look at aastra
03:02.37joobieahh k
03:02.40joobiealso james
03:02.50joobiehope you dont mind me bouncing these Q's off you:P
03:02.59joobiei have about 20 phones i want to hook up for cheap outbound calls
03:03.19jameswf-homeanalog?
03:03.25joobiedo i need asterisk at all? i mean.. i can see some providers allow you to hook striaght up to them, without asterisk.. just so long as u have a sip phone
03:03.28joobiedigital
03:03.29*** join/#asterisk Kumbang (n=kumbang@125.163.83.153)
03:04.11jameswf-homedepends on how much control you want
03:04.18*** join/#asterisk SteveTotaro (n=root@pool-71-179-207-15.bltmmd.east.verizon.net)
03:04.41joobiewhat are the control mechanisms asterisk can give you ontop james?
03:05.07joobieso you're saying you can do voip phone -> asterisk -> provider .. all using SIP as the transport protcol.. in a way using asterisk as a sip gateway?
03:05.34jameswf-homeasterisk gives you full controll say you only need 10 lines for 20 phones etc...
03:05.57*** join/#asterisk chendy (n=chendy@58.61.40.239)
03:06.00SteveTotarohey, with a little luck and some help with Tzafrir, I got a Junghanns quad port BRI card and a quad port Sangoma FXO board to live happily together on a US BRI no less
03:06.11joobiebut then with SIP providers.. if 10 lines are in use we'll get billed the same as if 20 were yea?
03:06.26jameswf-homeright
03:06.30joobiei see
03:06.51joobiebut i guess from asterisk we can restrict cant we?
03:06.59SteveTotaroi would suggest you have a dedicated point to point if you are going to rely on that for business
03:07.01jameswf-homeyes
03:07.03joobielike can we say "u can only dial local numbers" for particular phones
03:07.23SteveTotarothat is all done in the dialplan
03:07.29joobieSteveTotaro, how come? I thought having asterisk would be a better solution?
03:07.39joobieliek more flexible..
03:07.56SteveTotaroasterisk is a great solution, i was talking about using all VoIP
03:08.11joobieahhh
03:08.26JTSteveTotaro: do document US BRI workage online
03:08.30SteveTotaroeither get a point to point to your voip provider or get voip from your ISP
03:08.34JTSteveTotaro: lots of people here were interested in that
03:08.46JTa number of months back
03:09.02SteveTotaroi would love to consult, gas is too damn expensive ;)
03:09.03joobieSteveTotaro, point-to-point as in from the phone to the sip provider? or the phone, to asterisk, to the sip provider?
03:09.29SteveTotarogoing up to $4/gallon this spring!!!!
03:09.29JTSteveTotaro: don't look at me, where I am we get ETSI BRI :)
03:09.39joobiewow
03:09.41joobiethat is steep
03:09.44JTSteveTotaro: fuel in the US is cheap as hell
03:09.49JTthat's cheap
03:09.58jameswf-homethats only $1 a quart
03:10.02SteveTotaroi know and it is going to just rise
03:10.13JT$1 a litre or so
03:10.25JTfuel is around AUD$1.40/L here
03:10.31SteveTotaroin Liberia they sold gas in old plastic water bottles, it was colored red
03:10.36JTwhich is probably USD$1.20
03:10.36drmessanoEd Begley Jr could care less
03:10.40SteveTotarono working gas stations
03:10.59joobieSteveTotaro, point-to-point as in from the phone to the sip provider? or the phone, to asterisk, to the sip provider?
03:11.04SteveTotarohey DRM, how is hanging
03:11.07drmessanoHe's got a car powered by douchebagedness, and he gets 1000 miles per episode of St Elsewhere he did
03:11.13drmessanoWhaddup Steve
03:11.44SteveTotaroSt Elsewhere, that is a name that defined an era
03:11.53SteveTotarohillstreet blues
03:12.26SteveTotaroanyways, joobie you don't want public internet if you are serious about your business communications
03:12.38drmessanoEd Begley Jr doesn't use Asterisk because he can't get SIP to run on Solar power
03:12.43drmessanochan_douche is not finished
03:12.53joobieSteveTotaro, i am serious.. but i want to keep costs down
03:13.03joobieSteveTotaro, I was thinking about ADSL2+
03:13.03SteveTotarobut the potato battery is money
03:13.12drmessanoHA
03:13.17drmessanoPotato powered PBX
03:13.22SteveTotarohow many lines are you looking at?
03:13.23J4k3drmessano: eh, my gs bt 101's only draw about 1.5 watts (5V) while talking :P
03:13.27jameswf-home~chan_erexic
03:13.31SteveTotarowhat kind of usage? budget?
03:13.37jameswf-home~chan_rexic
03:13.43jameswf-homedoh
03:13.46J4k3~chan_suki
03:13.49SteveTotarotoll free?
03:13.52*** join/#asterisk jmacz (n=jmacz@190.25.38.26)
03:13.57joobieSteveTotaro, cheap as possible.. without going too nasty with performance.. it's a call center.. so heaps of outbound calls.. wnat to keep costs down
03:14.01joobiethat's why i was thinking of ADSL2+
03:14.05joobieand the number of lines is 20
03:14.19SteveTotarothen just get an LD pri
03:14.28joobiehow many lines is that?
03:14.37joobiethe thing is.. if i go PRI / BRI.. i pay higher call costs
03:14.38SteveTotaro23 voice 1 data
03:14.40joobiethat's what i was looking at sip
03:15.15*** join/#asterisk Wi_Fi (n=OUT@cpe-76-169-21-21.socal.res.rr.com)
03:15.19SteveTotarowhat is the cost if your call center is down for one hour?
03:15.29SteveTotarohow about one day?
03:15.37*** part/#asterisk Wi_Fi (n=OUT@cpe-76-169-21-21.socal.res.rr.com)
03:15.54DavieySupplement outbound calls with SIP, not rely upon
03:15.55SteveTotarothe one I implemented lost $27k per hour of downtime
03:16.04joobiewow
03:16.06joobiethat's a lot
03:16.09SteveTotaroDaviey has a good point
03:16.19joobiewhy is sip so bad?
03:16.31SteveTotaropublic internet is so bad
03:16.43J4k3joobie: theres a lot of doom/gloom going on, just get multiple itsps to make calls through
03:16.47J4k3so if the routing goes bad, switch
03:16.53joobieitsps?
03:16.59SteveTotaro~itsp
03:16.59jbot[~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs
03:17.02Davieys/switch/auto switch/
03:17.06joobieahhh
03:17.07SteveTotaroit isn't doom and goom
03:17.13joobieso if you get two
03:17.16joobieu just swtich to the other
03:17.17joobieif it's bad
03:17.19J4k3yep
03:17.22SteveTotaroi pay $150/mo for my LD T1 PRI
03:17.22J4k3get two internet connections
03:17.23joobieand maybe i can get two internet links
03:17.26joobietwo different proiders
03:17.26joobieya
03:17.32J4k3yeah
03:17.45joobieok this brings me to my next question
03:17.51joobiehow much bandwidth do we actually need
03:17.51SteveTotaromight as well have two different call centers
03:17.52joobieper call
03:18.05J4k3SteveTotaro: twice the management staff?  isyounuts?!
03:18.13SteveTotaroin case there is an earthquake or terrorist attack
03:18.15J4k3a second internet connection might cost you $250/month
03:18.32joobieDaviey, i got ya
03:18.38J4k3joobie: depends on how bad of call quality your ears can handle.
03:18.52joobiei dont want bad quality
03:18.57joobielike.. in terms of optimal perforamnce
03:19.00joobiehow much does each call need?
03:19.02SteveTotarono it depends on what your clients will tolerate
03:19.04J4k3~bandwidth
03:19.05jbotbandwidth is, like, This is a measure, in some amount of bits per second, of theamount of data that can be sent over a particular cable, interface, orbus.
03:19.08J4k3~g729
03:19.09jbotg729 is, like, an ITU-standard voice codec which operates at 8kbps and offers quality very similar to GSM. G.729 is patent-encumbered; those wishing to use it with Asterisk must buy a license from Digium.
03:19.11J4k3hrm
03:19.13SteveTotaroyou obviously have little call center experience
03:19.17jameswf-home~wizard
03:19.18jbothmm... wizard is enchancement to howto's
03:19.38J4k3http://www.inphonex.com/support/voip-codecs.php
03:19.55J4k3calculate that, add some padding
03:19.57J4k3etc.
03:20.04Davieyjoobie: Are you sure this project isn't a bit too big for you to handle?
03:20.15J4k320 lines is hard?
03:20.16J4k3wtfbbq
03:20.28joobieSteveTotaro, i do
03:20.31joobiewhy?
03:20.41J4k3joobie: you'd be plenty safe if you could get at least 1.5mbit upload
03:21.04joobieJ4k3, that should be doable.. how much per individual call tho
03:21.05joobieapprox
03:21.06J4k3I think adsl2+ is asynchonous, so be sure to get close to the CO
03:21.08joobiefor a good connection
03:21.12J4k332k + overhead + safety
03:21.16J4k3for g729
03:21.29J4k3figure 40-45k, assuming no other internet activity and/or very good qos
03:21.35J4k3on *Both* ends of the link
03:21.45joobieaahh k
03:21.51joobieKB/s
03:21.53joobieor Kbit?
03:21.55SteveTotarook, get an LD PRI for $150/mo
03:22.04SteveTotaropay a penny a minute
03:22.07joobieSteveTotaro, they are much more expensive here
03:22.13SteveTotarowhere is here?
03:22.15J4k3same here
03:22.17joobieAU
03:22.27J4k3I'd have to backhaul to a city to some clec to do that
03:22.29SteveTotarohow much?
03:22.34joobiei got a 3 line ISDN and it costs 75$/month
03:22.45SteveTotaroisdn what bri?
03:22.47J4k3The original 'Full Rate' GSM speech codec is named RPE-LTP (Regular Pulse Excitation Long-Term Prediction). This codec uses the information from previous samples (this information does not change very quickly) in order to predict the current sample. The speech signal is divided into blocks of 20 ms. These blocks are then passed to the speech codec, which has a rate of 13 kbps, in order to obtain blocks of 260 bits.
03:22.49joobiei think so
03:22.51joobieit's 3 line
03:22.52joobieafaik
03:22.54joobiefor another client
03:22.55J4k3^^ for the person that was complaining about lack of gsm-fr support
03:22.58joobieplus calls are not cheap on it
03:23.14SteveTotarosucks to be you
03:23.28JTjoobie: no 3 line ISDN
03:23.34SteveTotaromaybe voip is your "best" option if cost is the main concern
03:23.35J4k33 lines activated on a PRI
03:23.46J4k3you don't have to light up all 23/30 B's
03:23.59jameswf-homeI got a pri installed just to vote for american idol.... j/k
03:24.00JTJ4k3: don't know of a single telco here who will light up less than 10
03:24.02J4k330 or however many b's there are on an e1 pri
03:24.21J4k3JT: that usually depends on how much bsing you do with your telco salespeople
03:24.31J4k3JT: with the right charm, you can get fiber pulled to your garage in timbuktu
03:24.35SteveTotaroi know one that will do it but they hand it off as analog through a channel bank
03:25.04J4k3SteveTotaro: unplug channel bank, plug in pci card.
03:25.05SteveTotarothere are 31 channels in an e1
03:25.12SteveTotaro1 d 30 b
03:25.23JTsort of right
03:25.28JTthere's 32 timeslots
03:25.32JT30 B channels
03:25.34JT1 D channel
03:25.45jameswf-hometechnicaly 32 but one of the channels is al magical and invisible...
03:25.49SteveTotarosounds like what i just said
03:25.49JT1 timeslot reserved for multiframe synch and network alarms
03:25.57J4k3what a waste
03:25.59JTnot really, there are 32 timeslots
03:26.08SteveTotaroi never mentioned timeslots
03:26.12JTTS0 is multiframe synch and network alarms
03:26.17JTTS16 is the D channel
03:26.21JTbut that's all it it
03:26.23JTit is
03:26.24JTtimeslots
03:26.34JTanyway, only 30 from and end user viewpoint
03:26.41JTs/and/an/
03:26.54J4k3~jt
03:26.54jbotTemplate to compose LaTeX jewel case CD inserts. URL: http://www-stud.enst.fr/~michon/realisations.html
03:26.59Davieyshall we start getting itno US, Europe and Japan's variance?
03:27.04Davieyinto*
03:27.11JTand there are no RBS E1s afaik
03:27.15JTat least not in australia
03:27.20SteveTotaronah, africa and us is enough for me
03:27.22jameswf-home~e1
03:27.22jboti heard e1 is the basic digital telephony circuit used everywhere except the US, Japan, Taiwan and Hong Kong. T1, or slight variants of it, are used in those places. E1 runs at 2.048Mbps. It can be an unstructured channel for data. It can be channelised, to provide 31 time slots of voice or data, each of 64kbps. Sync and alarm information occupies the remainder of ...
03:27.50SteveTotarohrm 31 timeslots, jbot is all knowing
03:27.52jameswf-homethat sucked
03:28.19SteveTotaroSync and alarm information occupies the remainder of ...
03:28.22SteveTotaroguess not
03:28.23jameswf-home~book is all knowing
03:28.24jbot...but book is already something else...
03:28.31jameswf-homebah
03:28.58SteveTotarowell my ccie training taught me differently
03:29.24SteveTotaroit is the cisco way or the highway
03:29.27SteveTotaro;)
03:29.31DavieyCisco :(
03:29.33SteveTotaro~cisco
03:29.33jbotcisco makes the routers that move the pr0n across the internet at a very high rate of exchange. a company full of patent protected punks!
03:29.34JTi trust telecomms engineering training more ;)
03:29.43jameswf-home~porn
03:29.43jbotPorn remains one of the largest problems with Open Source Software. Often causing development delays, flooded links and, in extreme cases, disabling programmers ability to type.
03:29.58Daviey:O
03:30.31jameswf-home~random
03:30.36SteveTotaroDaviey, back to your point about supplementing TDM with an ITSP
03:31.38SteveTotarowhat mechanisms are there to determine when to shut off the voip
03:31.38SteveTotaro?
03:31.38Davieywhen the trunk fails :(
03:31.38JTjoobie: are you in the sticks?
03:31.38SteveTotaroother than that, i am thinking along the lines of poor audio quality
03:31.49Davieyhmm, can't think of anything in *
03:32.24SteveTotarothat would be a MAJOR plus
03:32.25Davieyi suppose you could measure packet loss, use * RT and modify it outside of *
03:33.12DavieyFirstly, what problems make a call quality bad:
03:33.19Daviey* long ping times
03:33.24Daviey* packet loss
03:33.28Daviey* crappy ITSP
03:33.29SteveTotarono, latency is not an issue
03:33.37Daviey* echo
03:33.49SteveTotaroit is but does not hurt voice quality
03:34.09Davieyhmm, UDP :)
03:34.18SteveTotaroyes udp
03:34.25Davieypacket loss DOES hurt udp
03:34.38SteveTotaroi was speaking of latency
03:34.48SteveTotarobut yes, packet loss hurts udp
03:35.39Davieysorry.
03:35.53SteveTotaroi would say anything that could make a customer not want to stay on the phone or prohibit business would qualify
03:36.10Davieyhmm, crappy on hold music then
03:36.12DavieyIVR's
03:36.22SteveTotarowell that is expected these days
03:36.29Davieyoffshore call centres
03:36.33SteveTotarojust keep pressing 0
03:36.40Davieyheh
03:36.58SteveTotaroi setup five offshore call centers
03:37.52SteveTotarothis is one i like to show off but it was not very big http://translate.google.com/translate?hl=en&sl=fr&u=http://www.osiris.sn/article1636.html&sa=X&oi=translate&resnum=3&ct=result&prev=/search%3Fq%3Dbeth%2Bpayne%2Bsenegal%2Bcomputer%2Bfrontiers%26hl%3Den%26client%3Dfirefox-a%26rls%3Dorg.mozilla:en-US:official%26hs%3DgKJ
03:38.09anonymouz666SteveTotaro: using hardphone/softphone or ATA?
03:38.27SteveTotarothis was a funny situation
03:38.55SteveTotaroto close the deal i had to use a system that had a name, so I used a 3com PBX
03:39.18SteveTotaroand a "authentication" server with a quad port E1 card
03:40.50SteveTotaroCSC and the State Dept were not buy "OpenSource Solution" so wording was very important
03:41.34anonymouz666that happens all the time
03:42.07SteveTotaroi was a one man show with 13 bosses, first time in africa
03:42.22SteveTotaroE1 cost $3k/mo
03:42.54SteveTotaroThen when Sonatel (the monopoly) found out what it was for, they wanted more money
03:43.12*** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net)
03:46.52*** join/#asterisk supjigator (n=shanebur@152.53.16.10)
03:47.12*** part/#asterisk beek (n=klinebl@static-71-240-222-16.alt.east.verizon.net)
03:50.05joobieback
03:50.07joobiesorry phone
03:50.15joobieJT, sorta.. about 30kms from the city
03:50.24SteveTotaroi hate phones
03:50.42JTjoobie: which city?
03:50.49jameswf-homemost eople who work with phones for a living hate phones
03:50.58jameswf-homepeople even
03:51.51SteveTotaroi RARELY answer
03:52.11joobieJT, melb
03:52.14SteveTotarogood about returning calls though
03:52.17joobieare you from AU too JT?
03:52.20JTjoobie: sydney
03:52.24joobiecool :)
03:52.28joobieg'day mate
03:52.29joobie:P
03:52.37JTjoobie: you should be able to get decent PRI pricing
03:52.37SteveTotarobut email is such a better way to communicate
03:52.39JThi :)
03:52.52joobieJT with who?.. is PRI just 3 lines?
03:52.57joobiebecause i need 20 lines outbound...
03:53.10JTjoobie: 10-30
03:53.37jameswf-homepri is 23
03:53.42jameswf-homemax
03:53.46JTjoobie: not in australia
03:53.47SteveTotaroe1 pri is 30
03:54.02JTjameswf-home: even
03:54.03joobieJT.. but line rental on that is like 25$ a line ya?
03:54.10jameswf-homeyou can get a partial pri
03:54.11JTjoobie: nope
03:54.16joobiehow much JT?
03:54.22JTjameswf-home: like i said, 10-30 was correct for australia
03:54.26joobiealso call costs are standard too.. they wont compare to voip call costs....
03:54.46JTjoobie: don't be so sure
03:54.50SteveTotaro(10:52:06 PM) jameswf-home: pri is 23
03:55.00SteveTotaro(10:52:11 PM) jameswf-home: max
03:55.06JTjoobie: quality and realiability is much better
03:55.10jameswf-homeis there an echo?
03:55.15JTjoobie: and you can get good call rates
03:55.17joobieJT, what are the costs for like a 20 line pri?
03:55.24JTjameswf-home: 23 is for a T1 PRI
03:55.34JTjameswf-home: we do not use T1 PRI in Australia
03:55.38SteveTotaroyou just said pri is 23 max
03:56.15SteveTotarothey use a much better e1
03:56.23SteveTotaroi wish we did as well
03:56.32SteveTotarobut BRI has to go
03:56.40jameswf-homema bell is the future they dont change
03:56.52jameswf-homewe are lucky we have AC power
03:57.05SteveTotaroi prefer DC power
03:57.13jameswf-homeU.S. is often bass ackwards
03:57.24JTjoobie: umm, $250/mo maybe
03:57.30SteveTotaroGO METRIC!!!
03:57.31JTmaybe $300
03:57.33JTdepends on telco
03:57.35joobieJT with who
03:57.40JTjoobie: Primus
03:57.43joobiehmm
03:57.46joobieinteresting
03:57.50JTOptus will be clost to $400
03:57.51SteveTotarominute costs?
03:57.52JTclose
03:58.01JTSteveTotaro: to call what?
03:58.02SteveTotarohe is outbound
03:58.20SteveTotaronot sure, joobie, who you telemarketing to?
03:58.28JTSteveTotaro: the costs for calls would vary depending on where you call
03:58.33joobieSteveTotaro, local and international
03:58.57SteveTotaroso get an itsp for your international
03:59.15joobiewell
03:59.17joobiethat was my next q
03:59.21joobieif i go the route of sip
03:59.28joobieare there any decent AU based ones?
03:59.31SteveTotaroobviously rates vary but i got a flat rate of a penny a minute anywhere in the US
03:59.40JTlocal calls are 10c+gst untimed with primus
04:00.25SteveTotaroi am unfamiliar with the terms "10c+gst untimed"
04:00.35joobieOngoing/monthly costs
04:00.36joobieISDN 10 lines (PRI) = $ 345 (based on Telstra
04:00.38joobiethat is from whirlpool
04:00.39*** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal)
04:00.58joobieAAPT's ABC product may be suitable. It can provide ISDN10/20/30 and internet access upto 2M over the same tail greatly reducing costs (no install and $20/month/line).You dont need to stick with ISDN10/20/30 either you can choose any number of lines and only pay for what you need (ISDN14 for example)
04:01.07joobieanother whirlpool post
04:01.14joobieso that would be
04:01.17JTjoobie: only if someone is moronic enough to choose telstra, or unlucky enough to have them as only choice
04:01.24joobie400$ / month just for rental
04:01.26JTjoobie: forget about whirlpool
04:01.32JTjoobie: and quote it yourself
04:01.42JTprimus is the cheapest if you can get it
04:01.46JTexcept maybe verizon
04:01.53JTverizon has steep install charges
04:02.01JTand very hard to talk to sales
04:02.12SteveTotaroverizon in Australia, cool
04:02.23J4k3verizon, haha
04:02.25J4k3sucks2bu
04:02.29JTyeah they bought MCI Worldcom and UUnet iirc
04:02.41SteveTotaroat least you have options unlike some countries
04:03.03J4k3yeah
04:03.04JTSteveTotaro: gst is 10%
04:03.09JTgoods and services tax
04:03.32joobieJT.. what about SIP providers, as an alternative
04:03.38JTit's payable at invoice time but is input tax creditable
04:03.42JTjoobie: they mostly suck
04:03.42joobieknow any good ones.. i mean ill look down the PRI path.. but want to compare against native sip
04:03.47jameswf-homediahria or frozen vomit.... sometimes options arent all they are cracked up to be
04:03.49JTbut perhaps Isphone, iVox
04:03.50joobiemostly? there is hope?
04:04.52joobieJT, are they decen tin melb too tho? or just sydn
04:05.10JTjoobie: they're big enough for it not to matter in such major metro areas
04:05.31JTsymbio networks might also be an option
04:05.51SteveTotaroi think frozen vomit is a much better choice
04:06.06SteveTotaroever see "joe dirt"
04:06.31SteveTotaroat least you don't need a VSAT
04:06.53JTjoobie: but on the "forget about it" list: engin, mynetfone, faktortel, koala telecom
04:07.15JTengin is not that bad, they're just not that good
04:07.22SteveTotaroJT, i think you should qualify the forget about it list
04:07.39joobiehehe
04:07.39JTmynetfone will only do 1 phone call per account - stupid
04:07.45joobieJT.. out of all those - which do u think is the best
04:07.45JTkoala telecom - clowns
04:07.52joobieout of theones u suggested
04:07.58JTfaktortel - clowns whose sydney dids were down for over 2 weeks
04:08.12JTengin - random dropouts and sometimes there's echo
04:08.12drmessanoDid someone say Clowns?
04:08.13SteveTotarofaktortel just sounds dumb
04:08.18J4k3ass clowns
04:08.24drmessano~happyclownpbx
04:08.30jbot[HappyClownPBX] is currently in closed beta, approaching 12GB in size, uses Asterisk for it's core, and it pwns
04:08.30JTother than that, they're ok, unlimited concurrent calls
04:08.30drmessano:(
04:08.54SteveTotarofor real about happyclownpbx?
04:08.55JTjoobie: btw, my favourite ITSP for personal use: Pennytel
04:09.01SteveTotaronever heard of it
04:09.04JTi dunno if i'd use them for business
04:09.15JTcheapest rates, quality is usually good
04:09.18*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
04:09.21JTsometimes there are hiccups though
04:09.56jameswf-homeI hear its based on web 3.0
04:10.20joobieahh
04:10.24joobieJt.. but for business use
04:10.28SteveTotarowhat is that age old saying
04:10.29joobiewhcih one do u think is best?
04:10.46SteveTotaroprice, quality, support, pick two (thats not it though)
04:11.13SteveTotaroactually maybe it is
04:12.03J4k3well, its also vs volume
04:12.05*** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal)
04:12.10J4k3you can get great prices if you're pushing 100k minutes/month, etc.
04:12.15jameswf-homeI thought it was you cant polish a turd but if you can convince enough people its a diamond you will get a million downloads
04:12.31J4k3ie - decent sized 24x7 international call center
04:13.02drmessano<jameswf-home> I thought it was you cant polish a turd but if you can convince enough people its a diamond you will get a million downloads <---- Sorry, but that's trademarked by trixbox
04:13.29SteveTotaroyou can print ink on paper and convince people it is worth something (the dollar)
04:13.32drmessanoI'm gonna need you to disease and decyst immediately
04:14.31J4k3SteveTotaro: I can hand you a piece of plastic, you swipe it, and *hand it back to me*
04:14.36J4k3for lots more than a dollar :)
04:14.47SteveTotaroor less
04:15.06SteveTotaroheck just give me the numbers on #ccnumbers
04:15.34jameswf-home#ccnumbers has 0 users
04:15.51SteveTotaroyou dont know what irc server
04:15.56SteveTotarovery few do
04:16.04J4k3thanks to channel logging and google, it'll have 17 secret service agents in it tomorrow afternoon
04:16.34SteveTotarodo you think they care?
04:16.42J4k3yeah
04:16.46J4k3they don't like bullshit-paper competition
04:16.52SteveTotaroit's part of the interest rate and fees paid by the merchants
04:17.39SteveTotarochargebacks just cost merchants
04:18.12J4k3which increases costs which increases prices
04:18.37SteveTotaroand all banking wants a paperless economy
04:18.52J4k3haha
04:18.58SteveTotarothat's fine but secret service doesn't give a damn
04:19.04J4k3so instead of bullshit dollars that people occasionally pay tax on
04:19.10SteveTotarothat is why the US dropped the $1k bill
04:19.16J4k3they'll barter, tax free*
04:19.25J4k3(* - not legally, but who can track?)
04:19.27SteveTotarothere is a huge barter movement
04:19.50SteveTotaroi barter as much as possible
04:20.01SteveTotaroadmin a PBX for colo space
04:20.02jameswf-hometoo bad cant barter gas
04:20.32J4k3I need an electric car
04:20.39J4k3power here is considerably cheaper than gas for now.
04:20.45SteveTotaroif you own a farm you can get tax free gas, much cheaper, but "illegal" to use in regular street vehicles
04:21.36J4k3usually its marked with dye
04:21.47J4k3some high output UV LEDs in your fuel tank can fix that quickly
04:22.33ManxPowerI wonder if you could get the fuel to generate the power to run the electric car...
04:23.04SteveTotaroBGE (Baltimore Gas and Electric) just deregulated, prices have jumped 50% immediately
04:23.16J4k3some 3v LEDs beats the hell out of the in-tank electric fuel pump running 14-15v
04:23.19JTi found a manufacturer who make luxeon style high output UV LEDs
04:23.24J4k3yay for electromechanical devices immersed in fuel
04:23.34JThell, they make almost any wavelength LED in a high output multi watt version
04:23.38JTfrom infrared to UV
04:23.47SteveTotaroi breaking the tail light but not the filament and putting that in the gas tank
04:24.29SteveTotaroturn on your headlights or hit the breaks
04:24.49J4k3not enough oxygen to do anything spectacular
04:24.59jameswf-homeJ4k3: dream killer
04:25.18SteveTotarowith the right additive...
04:25.33SteveTotaroi don't give all my special ops secrets away
04:26.06*** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net)
04:27.38SteveTotarowho need o2 when you have N2o
04:27.54jameswf-home~~
04:27.55jbotEvery moment in which I'm called upon is torture.
04:27.58jameswf-home~~~~~
04:27.59jbotgrrrr
04:30.08TJNII~~~
04:30.09jbotI HATE YOU, tjnii!!!
04:30.23Nivex~die
04:30.24jbotACTION takes two shots to the head and crumples to the ground, lifeless.
04:30.41Daviey~people_are_bored
04:31.42jameswf-home~taco
04:31.43jbotTACO TACO TACO!
04:31.53SteveTotaro~verizon
04:31.53jbotVerizon is utter garbage. Do yourself a favor and stay away from that company.
04:32.01SteveTotaro~qwest
04:32.02jboti guess qwest is a company with secksie backbones but lame peering (www.qwest.net). or a company that randomly scrambles routes and pisses off network engineers worldwide
04:32.06jameswf-home~vonage
04:32.07jbotfrom memory, vonage is a bunch of monkeys
04:32.36joobiehey JT
04:32.37jameswf-home~canada
04:32.37jbotWe're really sorry we beat you at hockey in the olympics, it's just that we're much much much much MUCH better than you.
04:32.39SteveTotaro~jameswf
04:32.40jbotjameswf loves unsolicited technical support
04:32.46joobiei just spoke to iprimus
04:33.13jameswf-home~fonality
04:33.13jbotFonality is hiring, "no Asterisk knowledge needed" or, "I just installed asterisk now what?", and will be the first guys against the wall when the revolution comes!
04:33.17SteveTotaro~jameswf-home
04:33.28joobie0m contract - $2,000install.. 12m contract - $1,000 install.. 24m contract - $0 install ............ $250/month for ISDN20 .. all costs exclude GST
04:33.44SteveTotaro$2k install!!!
04:33.50joobiein terms of call costs he's going to send that through on the email
04:34.01joobieSteveTotaro, that's AUD..
04:34.15J4k3disposable aussiedollars.
04:34.27jameswf-home~jameswf-home
04:34.27jbotwhen -home is added it means he is on his own time dont call his boss
04:34.28SteveTotaroso what does that equate to U$D, like $4k?
04:35.02Daviey~SteveTotaro
04:35.03jbotmethinks stevetotaro is an IRC nub
04:35.20jameswf-home~unixdog
04:35.21jbot<unixdog> Everyone use BSD gah linux sux progress is overrated use my project gah
04:35.21SteveTotarosee, all knowing
04:35.35joobieSteveTotaro, it's around $1,850 USD
04:36.04DavieySteveTotaro: $2K install is bargain!
04:36.08SteveTotarothat is a tidy bit
04:36.21SteveTotaromy install was waived
04:36.40J4k3unixdog speaks the truth, linux *does* indeed suck
04:36.51DavieyOne of my clients paid >$15K
04:36.51SteveTotaroit is an LD circuit though so I pay for inbound and outbound
04:37.19jameswf-home~linux
04:37.37jbotlinux is, like, the cure for cancer, AIDS and slavery to corporations
04:37.37SteveTotaropenny a minute all tollfree
04:38.00Daviey~ubuntu
04:38.06SteveTotaro~fedora
04:38.06jboti guess fedora is stevetotaro is <action> tells stevetotaro that becoming RHCP certified, is The One True Way to Supreme Linux Knowledge
04:38.10J4k3~nug
04:38.10jboti guess nug is a verb for when your girlfriend graciously feeds you chicken nuggets while you're driving. a lot of fun. any girl who does this for you is a keeper.
04:38.20J4k3hahaha
04:38.22J4k3wor dup
04:38.42Davieyfedora :(
04:38.49SteveTotaro~fedora
04:38.49jbotrumour has it, fedora is stevetotaro is <action> tells stevetotaro that becoming RHCP certified, is The One True Way to Supreme Linux Knowledge
04:38.53Davieyfedora :(
04:38.55J4k3~sexualchocolate
04:38.55jbotI told you that boy could sing
04:39.00SteveTotarowho put that in there?
04:39.49SteveTotarodaviey, do you see your name or is it mine?
04:39.55SteveTotaro~fedora
04:39.56jbotfrom memory, fedora is stevetotaro is <action> tells stevetotaro that becoming RHCP certified, is The One True Way to Supreme Linux Knowledge
04:40.19TJNII~inflatable dildos
04:40.19jbotOh yea, stretch that hole!
04:40.28Daviey< jbot> rumour has it, fedora is Daviey  is <action> tells Daviey that becoming RHCP certified, is The One True Way to Supreme Linux Knowledge
04:40.30SteveTotarogross
04:40.39jameswf-homelmao
04:40.46SteveTotarofrom memory, fedora is stevetotaro is <action> tells stevetotaro that becoming RHCP certified, is The One True Way to Supreme Linux Knowledge
04:40.51DavieySteveTotaro is irc nub :)
04:41.01SteveTotarotrue enough
04:41.07J4k3~yourmom
04:41.07jboti guess yourmom is a man
04:41.09JTSteveTotaro: install is waived if you get a 24m contract ;)
04:41.11SteveTotaroQwell put that in
04:41.19SteveTotaro~qwell
04:41.19jboti guess qwell is 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.
04:41.33DavieyJT: gah, you need at least 5 years to get it waived in the UK
04:41.53JTjoobie: i was right on the money for monthly cost there it seems pretty standard
04:41.59Davieywell for a high costing one
04:42.00SteveTotaro1 year waived, no monthly option
04:42.01JTDaviey: wow, that's an incredibly long contract
04:42.03JTfor pri...
04:42.59SteveTotarotook qwest about 50 calendar days to get it installed and provisioned correctly
04:43.14DavieyI had a customer that wanted fibre net access and PRI to 3 sites from the telco.. cost them nerly $200K
04:43.19SteveTotarog'night all
04:43.30DavieySteveTotaro: whats the time there?
04:43.37SteveTotarogoing to do some app_rpt stuff tomorrow, should be cool stuff
04:43.47jameswf-homeIt took qwest 5 techs and 2 weeks to get me a second pots line... only got fixed cause I threatened to go up the pole my self
04:43.51SteveTotaro11:43 says my atomic clock
04:44.15DavieyHmm, i should probably go to bed..
04:44.21DavieyMon Mar  3 04:44:20 GMT 2008
04:44.23Davieynn
04:44.44SteveTotaromostly site survey stuff tomorrow but need to be creative
04:45.12J4k3I hate marketing
04:45.15SteveTotaroand send out some bills
04:45.31jameswf-homeits only 21:45 here
04:45.31JTSteveTotaro: app_rpt is neat
04:45.35SteveTotaroyou should outsource your marketing
04:45.46J4k3SteveTotaro: no money to do that
04:45.53SteveTotaroyes, i have three high end repeaters to play with
04:46.30jameswf-homehave to spend $$$ to make $$$
04:46.34SteveTotarosometimes the money you don't think you have can be made by freeing up time spent on things you hate and doing things you like
04:47.25SteveTotaroespecially if you are not a marketing guy, could be wasted time
04:49.25joobieJT ya u were
04:50.10JTyes
04:52.01joobiecall rates tho
04:52.06joobiethat might be a different story
04:52.15JTjoobie: they all depend on spend
04:52.19JTthe more traffic you have
04:52.22JTthe cheaper they are
04:52.37joobiehey guys
04:52.41joobieahh
04:52.43joobiewell
04:52.50joobiewhat about analogue vs digital handsets
04:53.03JTthat's a no brainer... SIP :)
04:53.12joobiewhy?:P
04:53.15joobiethe cost is 150$ / handset
04:53.18joobieif i go analogue
04:53.21joobiei can get them much cheaper
04:53.29joobieand then put an analoge card in asterix box
04:53.40JTbecause analogue handsets have almost no features
04:53.45JTand analogue ports are not free
04:53.51joobiehow much are they
04:54.03joobielike i can get an analogue for for say 50$
04:54.07joobieso i save 100$ per handset
04:54.19SteveTotaroget a tenor AX 24port sip to FXS
04:54.27JTbut... they're crap
04:54.32joobieheh
04:54.33SteveTotarocallcenter phones get beat on
04:54.59SteveTotarojust go with the standard plantronics analog headsets
04:55.16joobiei was thinken polycom 330
04:55.18joobieerr 320
04:55.21SteveTotaroand a tenor ax (at least that is what I have had great luck with)
04:55.23*** part/#asterisk duri (n=mduregon@c-76-105-157-51.hsd1.or.comcast.net)
04:55.41SteveTotaroi keep the expensive stuff in the NOC
04:55.46joobiehttp://voip-warehouse.com.au/polycom-soundpoint-ip-320-phone-p-4516.html
04:55.48joobiecheck out that price
04:55.53joobie159$
04:56.02SteveTotaroand put thin clients and crappy analog headsets on the floor
04:56.46JTanalogue phones lack call handling ability
04:56.56joobiefark
04:56.58joobiethey are steep tho
04:57.08SteveTotarofor an outbound call  center??
04:57.12joobieSteveTotaro,
04:57.16SteveTotarowhat functionality do you need?
04:57.16joobie2900$ for that
04:57.21joobiemay as well get the polycom handsets
04:57.22joobieit'll be cheaper
04:57.23joobieand digital
04:57.35joobiejust route it straight to a standard etehrnet nic
04:57.40SteveTotaroit will be cheaper until they are abused and broken
04:58.03JTnot all callcentres are in the bronx ;)
04:58.09joobieahahah
04:58.13SteveTotaroi have never seen a call center agent give a sh*t about the equipment
05:01.50joobiebtw
05:02.01joobiewat about asterisk at home vs the full product
05:02.14joobiecan i get awawy with the gui of @home?
05:02.21joobiefor 20 line setup
05:02.24JTasterisk at home hasn't been updated for years
05:02.30joobieahh
05:02.54joobiei hear the full asterisk is a PITA to setup
05:03.07JT~book
05:03.07jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
05:03.42joobienice.. thanks jt
05:04.17joobieu kno i think google books has\ most ofthe oreilly books for free
05:04.24joobieits odd
05:04.33*** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl)
05:05.05*** part/#asterisk MoreAllLess (n=jackjust@cpe-76-169-131-140.socal.res.rr.com)
05:06.28joobiesee that is just weird
05:06.28joobieoreilly charge to view that book
05:06.28joobienormally
05:06.28joobieonline
05:06.28joobiebut u can get it free
05:06.30joobieduno wats up with that.. they changed their model or sumthen
05:08.24MatBoywow, the asterisk.org site is slow most of the time
05:08.32MatBoyactually the digium site
05:09.27JTjoobie: the asterisk books from o'reilly have always been freely downloadable
05:10.04MatBoyJT, I think no-one ever would have bought it when it was a payed version
05:10.18JT"payed"?
05:10.21joobieu sure JT?
05:10.21jameswf-homehmmmm http://travel.state.gov/passport/ppt_card/ppt_card_3926.html
05:10.23MatBoypaid
05:10.24MatBoy:)
05:10.25joobiei thought they charged
05:10.26JTjoobie: yes
05:10.28joobiethey used to have demo books
05:10.30joobielike a few chapters
05:10.33joobiethe rest was paid
05:10.38JTjoobie: for this book.
05:10.53joobieahh certain books are free?
05:10.59joobienot all?
05:11.01MatBoyjooby asteriskdocs.org
05:11.06JTi know this one is available for free
05:11.29MatBoythere you can download it
05:12.07joobieya got it bro
05:12.07joobiethanks
05:12.29MatBoyok, cu you in the bronx :P
05:12.57joobiewerd my bruda from anutha mutha
05:13.01MatBoyIt's Bell day today !!
05:13.11joobiebell day?
05:13.13MatBoyyep
05:13.23MatBoyhttp://en.wikipedia.org/wiki/Alexander_Graham_Bell
05:13.35MatBoywould be nic eto add into the topic for fun :)
05:13.38jameswf-homema bell day
05:13.49MatBoyjameswf-home, hey !
05:13.56MatBoyAsterisk made me sleep better and wake up better
05:13.57MatBoy:)
05:14.09jameswf-homeAsterisk can cure cancer
05:14.15jameswf-home~asterisk
05:14.15jbotsomebody said asterisk was the best free PBX in the world, or #asterisk on irc.freenode.net, or http://www.asterisk.org
05:14.25jameswf-homethats not it
05:14.30MatBoyow
05:14.56MatBoyno really, I have a great feeling about it
05:15.03MatBoythis is the OSS solution that I needed
05:16.31MatBoyjameswf-home, but what did you actually mean with it ?
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05:20.22MatBoyI'm actually figuring out what is the best to have national numbers with the country area that are in the system direcly called to sipfriends and what is not known in the system is routed to the outsideworld
05:23.41MatBoyI think a default prefix
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05:42.48AJayMNIf Im using g729.. the phone will use 1 license.. but if my provider makes me connect to them with g729 with that call is that another g729 count? or is that considered a pass-thru and only registers as 1?
05:42.51jameswf-home~sleep
05:42.51jboti guess sleep is overrated, and a poor substitute for caffeine.
05:43.16MatBoyjameswf-home, hehe, I almost didn't slept for 2 days
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05:43.53newmemberCan I run two instances of a SIP phone on the same computer?
05:43.54jameswf-homesome Jack-Ares on the plane got me sick....
05:44.12MatBoyJack-Ares ?
05:44.19jameswf-homearse
05:44.25MatBoyow hehe
05:44.50MatBoynewmember, yes, but it won't woirk... maybe you can bind some tools to seperate IP's on your PC ?
05:44.57MatBoydunno if those clients exist
05:45.12Der-Timhi there
05:45.21MatBoyat least, I thought it was not possible
05:45.22Der-Timgood morning from germany
05:45.27newmemberMatBoy: interesting ID
05:45.35newmemberMatBoy: interesting idea
05:46.31MatBoynewmember, yeah, but I didn't see any clients yet that can be bound to an IP, but I never needed it.. it's actually very nice for testing, so let me know if you want when you found something. I now do it using my VM and Host or a laptop next to me
05:50.20AJayMNany of you guys using g729?
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05:54.14drmessanoAJayMN
05:54.39drmessanoIf your device supports G729 and your ITSP is using G729, that is passthru
05:54.59*** join/#asterisk MoreAllLess (n=jackjust@cpe-76-169-131-140.socal.res.rr.com)
05:56.42AJayMNso it would only hit the count if they were in a conference room on the local system?
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06:01.48drmessanoIf there is transcoding, its using a license
06:01.52drmessanoor licenses
06:04.22drmessanoYou're welcome
06:04.26drmessanoJerk
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06:08.56BeeBuudrmessano: are you still there?
06:10.04drmessanoyes
06:10.33BeeBuuhow can i call someone and invite to meet?
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06:12.06drmessanoYou can call them and transfer to meet
06:12.17drmessanoor have them call you and transfer them
06:12.28drmessanoYou still trying to do this multi server conference?
06:12.36BeeBuuwhich command can transfer to meet?
06:12.48BeeBuudrmessano: yes,i am.
06:12.58drmessanoHave you googled like I advised you?
06:13.22BeeBuuyes, just get ChannelRedirect()
06:13.29BeeBuubut that's beyong me
06:13.36drmessanoNope
06:13.37drmessanohttp://www.trixbox.org/forums/trixbox-forums/help/multi-site-conference
06:13.41drmessanoRead that
06:13.57BeeBuuthanks,drmessano,you are so nice..
06:14.19BeeBuulet me check it...
06:15.12drmessanoThat's not a perfect solution, but it's a damn good try at it.. I'm sure someone can polish it up a lot more and make it bulletproof
06:23.22craigkdoes anybody know if it is 'faster' or 'better' to use the asterisk built in berkley db, or AGI and an external database ? I guess I am looking for an opinion on speed v reliability v features comparing berkley db to AGI and mysql/sqlite
06:27.04*** join/#asterisk Abu-Abudrahman (n=chatzill@84.36.147.94)
06:29.29Nuggetbdb wins in speed and in reliability.
06:29.57Nuggetmysql is a big lose on the complexity and reliability fronts, and will *always* be a more poorly tested configuration.
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06:39.09Abu-Abudrahmanam trying to install asterisk the last versions using svn but when i run the ./configure i got that error "configure: error: C preprocessor "/lib/cpp" fails sanity check" OS is Solaris 10
06:39.22Abu-Abudrahmanany suggestions ?
06:39.34Nuggetwhy are you using the development code from svn?
06:39.55Abu-Abudrahmani was following instruction on digium fourms
06:40.14Abu-Abudrahmann the author wrote the howto usin svn
06:40.40Nuggetyou're sure you have a c++ compiler installed on the machine
06:40.40Nugget?
06:40.52Abu-Abudrahmanyes am sure
06:42.00Abu-Abudrahmanbash-3.00# pkg-get -i gcc3g++
06:42.02Abu-AbudrahmanWARNING: gpg not found
06:42.03Abu-AbudrahmanNo worries... you already have version 3.4.5 of gcc3g++
06:42.23BeeBuui want dial out and auto play something to callee?
06:50.18BeeBuu~
06:50.23BeeBuu~~
06:50.23jbotEvery moment in which I'm called upon is torture.
06:50.47drmessano~~~~~~~~
06:50.47jbotYou know, this got old a long time ago.
06:50.51drmessano~~~~~~~~~
06:50.51jbotI'm ignoring you now.
06:50.55drmessano~~~~~~~~~~
06:51.00drmessanobah
06:51.04drmessano~~~~~~~~~~~
06:51.04jbotNo, really, STOP!
06:51.08drmessano~~~~~~~~~~~~
06:51.08jbot~~~~~~~~~~~ are YOU READY?????????? ~~~~~~~~~~~~~~~~~~
06:51.12drmessano~~~~~~~~~~~~~
06:51.16drmessano~~~~~~~~~~~~~~
06:51.19drmessanoHmm
06:51.26drmessano~~~~~~~~~~~~~
06:51.28drmessano~~~~~~~~~~~~
06:51.28jbot~~~~~~~~~~~ are YOU READY?????????? ~~~~~~~~~~~~~~~~~~
06:51.33drmessano~~~~~~~~~~~~~~~~
06:51.48drmessanoAt some point it's supposed to shoot me or something
06:51.55drmessano~~~~~~~~~~~~~
06:52.05drmessano~~~~~~~~~~~~~~
06:52.07*** part/#asterisk hads (n=hads@reef80.anchor.net.au)
06:52.13drmessano~~~~~~~~~~~~~~~~~~~~
06:52.47drmessanodicks?
06:56.08*** join/#asterisk ArashHemmat (n=ArashHem@91.184.88.227)
06:57.55J4k3~no
06:57.55jbotYES
06:58.06J4k3~norway
06:58.06jbotrumour has it, norway is a country in Scandinavia. A beer costs EUR 1,80 there. The capital of Norway is Oslo. And yes norway isn't a member of the EU, or a great skiing nation, or actually it is a great skiing nation, or make sure you know where your towel is before you go
06:58.15J4k3~kenya
06:58.15jbothmm... kenya is Where can you find Lions?  Only http://mastaile.mine.nu/kenya1.mov !, or http://www.weebls-stuff.com/toons/kenya/
06:58.26J4k3jbot: forget norway!
06:58.26jboti didn't have anything called 'norway!' to forget, J4k3
06:58.46drmessano~dubai
06:58.55J4k3dubai is for smoking
06:59.28drmessano~dubai
06:59.28jbotHalliburton!
06:59.32J4k3haha
06:59.34drmessanobetter
06:59.35J4k3~bush
06:59.35jbotsomebody said bush was chick plumbing or the current president and potential dictator, or the guy that made stupidity fashionable.
06:59.46J4k3~cheney
06:59.49*** join/#asterisk dominic1 (n=dob@213.221.82.242)
07:00.06J4k3jbot: cheney is <reply> dick
07:00.06jbotJ4k3: please, watch your language.
07:00.13drmessanoha
07:00.15J4k3wtf
07:00.27J4k3dick is a name, not a 'bad word'
07:00.28J4k3wtfbbq.
07:01.12drmessano~cheney
07:01.12jbotI bet you thought I was going to say "diiiiiick"
07:01.21drmessanohandled
07:02.16J4k3haha
07:02.53J4k3I need to stick this evdo card in my laptop and hack an antenna onto it.
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07:06.29drmessanoAll I have is tin foil, and I am using that for my hat
07:16.24*** join/#asterisk lnx (n=lnx@183-83-66.ip.adsl.hu)
07:16.27lnxhi all
07:24.01*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)
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07:36.49BeeBuuhow can i run multi command with asterisk -rx?
07:37.05drmessanowhat do you mean?
07:37.26BeeBuuSeparate with ;?
07:38.05BeeBuuasterisk -rx "command 1;command 2..."
07:38.12BeeBuucan i do that?
07:38.23drmessanoNo idea
07:38.31drmessanoWhat are you trying to do?
07:38.44BeeBuuauto run some command..
07:38.57BeeBuuwith console
07:39.03drmessanoduh really
07:39.11drmessanoI never would have guessed
07:40.10drmessanoJust seems like a silly way to go about whatever it is you're trying to do
07:40.43drmessanoBut without knowing more than "trying to do stuff", i'll just leave it as "uh, dunno"
07:41.38*** join/#asterisk agallo (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it)
07:46.09tzafrirBeeBuu, no. this doesn't work
07:46.15lnxhmh i have try to write out a return code of function send_text via AGI perl script. Text has been arrived. http://pastebin.com/d33413141
07:46.29lnxbut out file is empty
07:46.34BeeBuutzafrir:how can i run multi command?
07:46.57tzafrirBeeBuu, only decent way I know: send commands directly to the unix-domain socket
07:47.00lnxBeeBuu: what kind of command wish to run?
07:47.23BeeBuudial to someone and play some sounds.
07:47.43BeeBuulnx: any thougths?
07:48.24lnxi have medi it via call file cause my server has not sound card :)
07:48.36lnx/medi/made
07:48.48tzafrirthe base of the script I have:
07:48.58tzafrirwhile read line; do
07:49.09tzafrirecho -n "$line"
07:49.19BeeBuutzafrir: what's that?
07:49.20tzafrirsleep 0.001 # separate between commands.
07:49.20tzafrirdone
07:49.28tzafrirand that piped to:
07:49.37lnxBeeBuu: shell script :)
07:49.45tzafrirsocat - /var/run/asterisk/asterisk.ctl
07:50.02tzafrirthis is only for sending commands
07:50.31BeeBuuwould you A-Z?
07:50.46BeeBuuwould you teach me with A to Z?
07:51.05drmessanoBeeBuu, you may want to take a look at the book
07:51.14drmessanoIt can answer a lot of these things for you
07:51.22BeeBuuwhere's the book?
07:51.26BeeBuu~book?
07:51.27jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
07:51.27drmessano~book
07:51.27jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
07:51.47J4k3~sike
07:51.47jbot[Sike!] The preceeding statement was intented to be humourous, and may have involved false statements used in a humorous manner, or an exaggeration of a real life situation with humourous intent, but in no way should be taken as pure fact or not considered questionable at best.
07:52.03*** join/#asterisk PepOSX (n=angeldav@190.72.147.233)
07:52.30lnxte-asterisk-book.com is a nice one too
07:53.03BeeBuulnx: where iss it?
07:53.40lnxon the Moon BeeBuu
07:53.48BeeBuu:-D
07:53.56tzafrirI believe recent nc also supports unix-domain sockets, so you may also use it. socat is packaged in e.g. debian
07:54.21J4k3its in the air, dawg
07:54.57tzafrir~j4k3
07:54.57jboti heard j4k3 is a dream killa
07:55.01drmessanoJ4k3: Did you ever play the game Castles?
07:55.08BeeBuutzafrir: i conected to 5038 now
07:55.27tzafrirBeeBuu, that's another option. Requires setting up a user in advance
07:55.38drmessanoIn order to even start it, you needed to answer a question from the book...
07:55.48drmessanoThat would be a good arrangement for #asterisk lol
07:55.50lnxBeeBuu: u must read some docs @ www.voip-info.org
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07:56.27BeeBuulnx: i do,but there are so many...
07:56.32lnxlol
07:56.35drmessanolnx: Isn't it better just to ask every single minute question in here?
07:57.00lnxIsn't :)
07:57.15drmessanoSurely you jest :)
07:58.19lnxBTW drmessano can u tell me please why return code of send_text does not present @ logfile http://pastebin.com/d33413141    :))
07:59.38drmessanoNo, but I bet tzafrir can
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08:01.17Asterisk-nobhi
08:01.27*** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net)
08:04.29Asterisk-nobI've a question about the pound key mechanism, like press 1 followed by the pound key, so for this we simply create exten => 1#,1,dial(sip/ext) or it has some otherway to do?
08:06.30BeeBuutrafrir: i can
08:06.43BeeBuunot send command to AMI
08:06.56tzafrirlnx, have you checked that the open was successful?
08:07.19BeeBuutzafrir: nc 127.0.0.1 5038
08:07.46BeeBuuand send command,but doesn't work...
08:07.47tzafrirBeeBuu, sure. That's to the manager interface.
08:08.16tzafrirBut the protocol there is slightly more complex. For starters you need a manager username and password
08:08.19*** part/#asterisk MoreAllLess (n=jackjust@cpe-76-169-131-140.socal.res.rr.com)
08:08.19tzafrirand then:
08:08.44tzafrirsomething of the sort of: (/me drafts)
08:08.50BeeBuumust login?
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08:10.46jeanmiii_iHi
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08:12.17lnxtzafrir: yes
08:13.00lnxtzafrir: outfile has been summoned :)
08:13.24jeanmiii_iOne a call has been initialized, is it possible to change the media info (IP and port for rtp) using a reinvie or update ?
08:14.56*** join/#asterisk jivco (n=jivco@85.187.217.6)
08:15.10jeanmiii_iI am trying to find out if it is possible to do a man-in-the-middle attack by updating the media of two phones by asking them to no longer send/receive rtp to/from eachother but to go through another IP/port that would then forward to the phones
08:15.14lnxtzafrir: i have try to give a variable to per script from extensions.conf , exten => 10,3,AGI(testscript.pl,${DIALSTATUS})
08:15.28jeanmiii_iI am not sure if I am making myself very clear ....
08:15.56lnxtzafrir: then, no result
08:17.01lnxi have no more idea at all to get variables
08:22.09obnauticuswhat is caller id callback?
08:22.13obnauticusI don't understand the term
08:30.55tzafrirBeeBuu, http://pastebin.ca/925785
08:30.58JTobnauticus: when you call asterisk, it detects your callerid as being one to use callback on, and disconnects the call, then it calls you back
08:31.03tzafrirIf you feel like taking this further
08:31.14tzafriryou can replace the sed line with dos2unix
08:31.45tzafrirSorry, busy elsewhere now
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09:02.17lnxtzafrir: hmmh it seems the script makes the file but doesn't write into,
09:02.40tzafrirtry a different print, then?
09:02.56tzafrirprint L "whatever\n";
09:03.05tzafrirperhaps you need the \n?
09:03.12lnxtzafrir: if i move open() down; the file does not become
09:03.29lnxonly if open is in the 1st line
09:03.41lnxk testing :
09:03.42lnx)
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09:05.04lnxtzafrir: nothing, but   -- AGI Script testscript.pl completed, returning 0
09:05.21lnxit drives me crazy :S
09:05.44BeeBuutzafrir:thanks
09:07.04tzafrirlnx, it returns 0 because this is what you returned
09:07.50BeeBuutzaafrir: sleep 0.1 is enough?
09:08.24lnxtzafrir: it seems print to file by-pass
09:09.33*** join/#asterisk CaRb0n^ (n=playa@203.81.237.252)
09:09.48tzafrirBeeBuu, I have no idea
09:09.56tzafrirPure guess work
09:10.14BeeBuuwhat's sleep for? wait for result?
09:10.16tzafrirTry watching it in action (hint: tee)
09:13.24*** join/#asterisk af_ (n=getsmart@88-149-241-244.dynamic.ngi.it)
09:14.06[hC]huh
09:14.08[hC]outcall is pretty nice
09:14.11[hC]for an outlook integrated plugin
09:14.21lnxtzafrir: $AGI->send_text('Hello World'); is succesful, what do you think why print does not work?
09:14.39lnx[hC]: nice :)
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09:21.34akira2014hello
09:21.37tzafrirlnx, sorry, busy
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09:24.37jeanmiii_ia phone is calling an extension on my aserisk PBX, I have written a script that sends an INVITE to the phone (using the callID of the estasblished call, so which should be considered as a reinvite)
09:24.51jeanmiii_ithe phone is considering the INVITE as a new call
09:25.10jeanmiii_iinstead of a reinvite (even though I am using the same from/to/call-id)
09:25.21*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
09:25.37jeanmiii_ido you any clue about what I might be doing wrong
09:25.54agallothere is a way to stop FORTINET to drop multiple outgoing INVITEs sent by asterisk outside? it believe its some kind of spoofing attempt; i mean how to delay invites each others? seems no settings about this in sip.conf
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09:33.23CaRb0n^<CaRb0n^> even if i dial 7777
09:33.23CaRb0n^<CaRb0n^> it plays the ivr but does not accept any didgits 9keys0 after that
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09:49.07lnxwhat is the reason of that asterisk variable has no value? my $status = $AGI->get_variable('DIALSTATUS'); ?  Local SIP dialed via call file, dial is successful...
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09:59.25Der-Timhi there
10:00.14Der-Timi'm using a dialstring like this Dial(IAX2/username:password@host/${EXTEN}) but i get an error on the remote site: Registration Refused
10:00.51tzafrirDer-Tim, that is a dial, not a registration
10:00.55tzafrirLook elsewhere
10:01.20tzafrirSpecifically, at the 'register =>' strings in iax.conf
10:02.38Der-Timtzafrir: i don't use the register string, as an author of a german asterisk book wrote, that this is not needed...
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10:08.29BeeBuucan i run any AGI program with CLI?
10:09.57akira2014tzafrir did you remember me ?
10:10.20akira2014i still in troubles with zaptel, after rebuilding all from sources
10:10.40tzafrirakira2014, what troubles?
10:10.56tzafrirrandom disconnections?
10:11.12akira2014no, i cant pick up calls from pstn
10:11.30tzafririt causes a hangup?
10:11.31akira2014wen a call caomes nothing happens
10:11.39akira2014no
10:11.47*** join/#asterisk hijacked (n=argh@cerebus.clandestineresearch.com)
10:12.34akira2014asterisk sees the zap channel, but nothing happens wen a call comes from landline
10:12.50tzafrirhttp://lists.digium.com/pipermail/asterisk-users/2008-March/206851.html
10:14.45tzafrirmaybe downgrade to 1.4.7.1 (but in this case, be sure to rebuild asterisk)
10:15.13tzafrirhmm... "nothing happens" is something different from the problem described there
10:15.44akira2014the problem related in this link is different
10:16.22lnxtzafrir: do you know why my ${DIALSTATUS} is empty?
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10:16.58akira2014my problem is exactly nothing appens, no messages from zaptel in log, no asterisk console messages.... nothing no where :(
10:17.37tzafrirlnx, sorry, wasn't following it
10:17.59tzafrirAll I can suggest is to use extra tracing
10:18.12akira2014strace ?
10:18.29tzafrirakira2014, any debug messages when you enable that debug?
10:18.39akira2014no
10:18.49akira2014verbose is 100 and debug is 100
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10:21.22lnxtzafrir:  i have put verbose after answer   exten => 10,2,Answer()   exten => 10,3,AGI(verbose "MUUUU ${DIALSTATUS}",1)  and the output is "MUUUU "  -only
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10:23.08Der-Timwhat can i do if i see such an error message: Auto-congesting call due to slow response ?
10:23.38Der-Timi'm facing a problem with an static ip asterisk host without nat and a dynamic ip host behind nat...
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10:26.21jeanmiii_iI have put a .call file in /var/spool/asterisk/outgoing to make an outgoing call but the file remains in the directory without any call being triggered (and no error message whatsoever)
10:26.50jeanmiii_ido I have to enable something somewhere in order to have the .call files triggering a call ?
10:27.17sergee~seen danpwi
10:27.30jbotsergee: i haven't seen 'danpwi'
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10:31.13budol~asterisk
10:31.13jbotmethinks asterisk is the best free PBX in the world, or #asterisk on irc.freenode.net, or http://www.asterisk.org
10:31.39budol~zaptel
10:31.39jbot[zaptel] zapata telephony interface. A low level interface designed to abstract hardware access to a variety of devices for BRI, PRI or analogue access.
10:31.59budol~bri
10:31.59jbotBRI == Basic Rate Interface, usually consisting of (1) 64kbps bearer (B) channel, and (1) 16kbps signalling (D) channel
10:32.24budol!pri
10:32.29budol~pri
10:32.29jbotpri is Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, R1T1,R2T1,R4T1, etc.
10:37.15*** join/#asterisk xezz (n=asdasd@trust-it.gr)
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10:38.00JTwhat the hell
10:38.05JTjbot: forget bri
10:38.05jbotJT: i forgot bri
10:38.10xezzhello, i've installed mod_ssl but http access still exists with https, any idea on how to disable http ?
10:39.13JTjbot: BRI is Basic Rate Interface - a form of ISDN that consists of 2 * 64kbit/s Bearer (B) channels and 1 * 16kbit/s signalling channel (D)
10:39.14jbotJT: okay
10:39.48g0mb0sergee?
10:39.56g0mb0http://bugs.digium.com/view.php?id=11993
10:44.17Der-Timmmh, got a new problem by now... :-(
10:44.35Der-TimCall rejected. no authority found...
10:44.51Der-Timbut there is an context "from-internal" on the remote system
10:53.27beasty_is it possible to let sip user make an outgoing call by a IAX channel
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10:59.39*** topic/#asterisk is Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk 1.6.0-beta4 (2008/02/21), Asterisk 1.4.18 (2008/02/07), *-Addons 1.4.6, 1.6.0-beta2 (2008/02/21), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox -=- FYI: no sharks in this channel
11:00.38sergeeg0mb0: that was you, right?
11:00.48g0mb0yes
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11:09.38sergeeg0mb0: oops, i meant danpwi from http://bugs.digium.com/view.php?id=9299, 11993 is second in my list after 9299
11:09.57g0mb0ok
11:10.01g0mb0gotta go, thanks
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11:11.24nixguycould anyone recomend a nice guy where  users can change forwarding of phone numbers listen to messges etc?
11:11.28nixguygui
11:11.30nixguynot guy :)
11:15.14tzafrirdestar?
11:15.20tzafrirhmm... not listen
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11:23.36xezz<PROTECTED>
11:24.10angryuserhello everybody, i have strange traffic amounts, i have installed pfsense with traffic shaper, and when i call it show's 88kb/s!? why so much?
11:26.17jblackangryuser: perhaps 40Kb a sec in, 40Kb a sec out.
11:26.43jblackThough offhand I can't think of a 40Kb codec.
11:26.48angryuseri got queue in and out, in 88 out 92
11:26.58jblackThere's 20Kb codecs, 56Kb codecs....
11:27.16beasty_anyone ever got this ?
11:27.18beasty_WARNING[21381]: app_dial.c:1628 dial_exec_full: Had to drop call because I couldn't make SIP/jdecoste-081d4aa0 compatible with IAX2/329909001611-5
11:28.04angryuser<jblack> maximum that i am able to use is G711 64kbit/s that is 8 ko
11:28.17angryuser<jblack> i dont have any special codec installed
11:28.37angryuserkinde strange
11:28.39jblackI didn't say those things.
11:29.38jblackDidn't you say you're using 88kb a sec, as in 88 kilobit (~ 10 kilobytes)
11:30.13angryuser<jblack> no it was 88 ko
11:30.20jblackI didn't say that either.
11:30.23angryuserlike 10 times more
11:30.52jblackangryuser: "<jblack> no it was 88 ko" is quoting me as having said "no it was 88 ko"
11:30.57JTangryuser: 64kbit/s dows not include sip and rtp overhead
11:30.59JTwhat is ko?
11:31.27angryuser<JT> ah sorryn damn french it's kb ;)
11:31.33JTkB or kb?
11:32.13angryuserwhatever it's like 512kbit/sec for one call (all traffic)
11:32.50jblackThat doesn't sound like one call to me. Perhaps you have unrelated traffic in that count.
11:33.15JTkb = kilobits
11:33.19JTkB = kilobytes
11:33.37angryusernope i am sure of that, i got like near 0 before call, and 88kB during
11:33.58angryusermaybe pfsense i lying ?
11:36.49angryuseri will play with codec's let's see
11:38.16angryusercodec's permitted alaw ulaw libc gsm, not a traffic consuming
11:38.23JTi think it's really kbit/s
11:38.51angryuser49.30Kb/s -----status of queue
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11:39.07angryuser<JT> kbit's you think ?
11:39.38JTkb/s is kilbits per second
11:39.50JTkilobits
11:40.09angryuseri will ask someone to load a big file
11:43.22angryuser<JT> yes it was kbit/s! stupid me ;)
11:45.02atopIf I recompile Asterisk with dont_optimize and malloc_debug flags set, can the resulting code be ran for a few days without problem, or will it run like a pig?
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11:49.41beasty_anyone knows if i can let my SIP/ users call out using a IAX line ?
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11:57.57akira2014when a call comes from a zap channel and i make a sip/XXX why my phone rings forever even if the other side has hangup? thk's
12:03.29tzafrirakira2014, where are you at?
12:03.47tzafrirdoes your provider give any sort of disconnect supervision?
12:04.01akira2014i'm in spain
12:04.06tzafririf not, use busydetect :-(
12:04.08akira2014my provider is telefonica
12:04.33tzafrirhmm... I'm not sure if it uses polarity
12:05.04akira2014i will search in google about polarity on my profider
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12:07.49akira2014tzafrir, yes they use, i've forget to add it on zapata
12:09.03tzafrir<PROTECTED>
12:09.52akira2014ok, now it works
12:10.09akira2014thk's tzafrir
12:13.41agalloi've 16 account registered onto the same sip server, is there a way to give a little delay between each outgoing "REGISTER" attempts ?
12:14.13jblacknot that I know of.
12:14.42mostyagallo, do you have a problem with the registrations?
12:15.03agallomosty, yes but i think its related to some Fortinet Fortigate filtering out traffic
12:15.14agalloits seems does not like to spam many register at the same time
12:15.41mostywell that's easy to test, just disabled the filter temporarily
12:15.52*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net)
12:16.37agallomosty, well its not a problem of me, its the DID provider that sucks! lol :)
12:17.34mostycontact the DID provider's tech support
12:19.00agallomosty, they say its my router fault; i tested with other DID providers, other routers, other xDSL line and its not my fault; i'm getting 1 OK at the 1st REGISTER and no reply for the other 15. Their fortinet is filtering out stuff or there is some routing/spoofing problem with their packets
12:20.58mostystill sounds like you need to talk to them
12:29.25jblackHere's what you can do.
12:29.36jblackchange the qualify time for them so that they're all different.
12:30.12jblackYou'll still spam yourself out for the first round on all of them, but the later reregisters will catch back up on the 2nd round.
12:32.03agallojblack, qualify handle the OPTIONS message not the register, afaik
12:32.17jblackPersonally, I think the problem is not that they have abuse protection, but that the abuse protection is a little too stringent.
12:32.23jblackYou got something better to try?
12:33.07agallojblack, yes since they say its router fault i'm trying with 2 different routers and 2 different xDSL lines... also trying if i've some strange stuff in /proc/sys/dev/ ... but i never had problems with other providers....
12:33.27jblackmy thinking is it's possible that with qualify, when the timeout comes up, it'll notice that registration hasn't happen, and it'll reregister.
12:34.03jblackagallo: So your something better is to go with a different provider.. That's reasonable.
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12:37.56agallojblack, unluckly i cannot switch it since it takes months to switch the telephone numbers too :-(
12:38.21jblackThat doesn't sound like a better idea to me.
12:38.29agalloi'm very upset, this time its not asterisk fault, i cannot fix it myself :-P
12:39.04jblackI'm not upset.
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12:43.45coppice"its out of my hands" sounds so much better than "I have to spend hours fixing stuff"
12:44.31jblackDidn't he say months, not hours?
12:45.00coppicethat's even worse, isn't it?
12:45.57coppiceI just hate work
12:45.59agalloits 2 month minimum to switch telephone number between DID providers :)
12:46.08agallocoppice, true, lets abolish it
12:46.17jblackIt's as if you weren't being sarcastic.
12:46.40Der-Timre
12:46.51coppiceI'm not. I honest and sincere
12:47.02jblackagallo: So, try the qualify thing. Perhaps you'll be able to get all 15 registered over a period of 15-20 minutes. Then, you can just worry about not restarting *
12:47.16Der-Timsorry for asking, but what does "no authority" mean? a remote asterisk is rejecting a call and i don't know why
12:47.32jblackThat might buy you the time you need to port the numbers to a handful of providers.
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12:48.45jblackder-tim: That sounds like a bad user/pass. I'd check the sip/iax regististrations to make sure that the servers are registering to each other with the right accounts and passes.
12:49.11jblackheh. Bill Richardson has a beard.
12:49.12Der-Timjblack: there's nothing like registration... just username / secret in the iax config
12:49.34agallojblack, tryed but there is no OPTIONS going out since they're not registered (indeed i never get a packet back from the server)
12:49.47jblackder-tim: Try registering (you can register with iax). See if that casts some light on the problem.
12:50.43Der-Timjblack: well, i tried it with the user credentials in the dial string...
12:50.57jblackheh.
12:51.21Der-Timjblack: if i'm using registration, am i in need of a [usernam] type=user section?
12:52.13jblackNot to do the registration itself. But if you're lacking an authentication stanza for a machine that's saying "you're not authenticated"... Well, to be polite.. shouldn't that be telling you something?
12:54.48jblackPresuming that machine B is trying to call machine A, and machine A is telling B that "youre not authenticated".. well, that comes down to two possibilities.
12:55.21jblackEither 1) Machine A doesn't have a guest account. or 2) Machine B isn't using a account that A recognizes
12:56.14jblack2) Can be broken up into a half dozen possibilities... such as "Not trying to authenticate at all".. "Bad username." "Bad password".. "wrong hostname setup"... etc etc
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13:00.00Der-Timjblack: nothing of those possibilities... the username / password is the same on both machines... for sure... all i get are two different messages, which don't make sense atm
13:00.13coppiceanyone know what happened to vovida.org?
13:00.31Der-Timthe first is "no authority"
13:01.11Der-Timand the other is "no registration for peer 'username'"
13:01.35Der-Timwell, i think, i should take a look later... ;-)
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13:21.55flujanhi all, I need a tool to convert wav files to g729.
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13:24.57beasty_is it hard to setup a musiconhold with mp3 ?
13:27.52PepOSXbeasty_, nop
13:27.57PepOSXuse playback
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13:31.11jblackHeh. Do trumpets go off when you get home?
13:31.19stansmithand the lights flash
13:32.04jblack"Children, daddy is home! Meet him at the front door and bow to him!"
13:32.28stansmithmaybe some day... i still live at home with mommy
13:32.54jblackOh, so those flashing lights are the alarm system....
13:33.20jblackDidn't your parents teach you "gotocollege.getajob.moveout" ?
13:33.32stansmithdone with college, at my job right now, ......
13:33.47jblack2/3 of the way there.
13:34.09stansmithi think my mom is scared to let her lil baby go, since im the youngest and what not
13:34.23stansmithchicks totally dig it by the way
13:34.24stansmithanyways..
13:34.36jblackNah. Mom is waiting for you to gtfo so that she can go on an ocean cruise. Trust me.
13:34.47stansmithhaha
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13:35.09jblackThe flashing lights should have clued you into that. ;)
13:35.10[TK]D-Fenderbeasty_: Install "asterisk-addons" for MP3 suppotr
13:35.35jblack[TK]D-Fender: Hey, you want an invite to hulu? The site is a place to watch TV shows.
13:35.48[TK]D-Fenderjblack: Thanks for the offer, but no need.
13:35.59jblackokey-dokey
13:36.47PepOSXLOL @ okey-dokey
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13:37.54[TK]D-FenderPepOSX: And Playback has nothing to do with MoH
13:38.07beasty_[TK]D-Fender: now i get this
13:38.09beasty_WARNING[30303]: res_musiconhold.c:947 local_ast_moh_start: No class: default
13:38.32[TK]D-Fenderbeasty_: Go set up your "musiconhold.conf"
13:38.34PepOSXMoH?
13:38.37beasty_did that
13:39.08[TK]D-Fenderbeasty_: Well it seems to clearly think your class [default] does not exist
13:39.55*** join/#asterisk shido6 (n=shido6@204.126.120.132)
13:40.41*** part/#asterisk suahmed (n=Administ@69.88.13.17)
13:41.31*** join/#asterisk real0ne (i=cleo@41.251.86.36)
13:41.35beasty_[TK]D-Fender: http://rafb.net/p/nrdcv071.html
13:41.51real0neany one here use iaxcomm?
13:43.40[TK]D-Fenderbeasty_: "module reload res_musiconhold.so"
13:45.43beasty_mm
13:48.19ManxPowerI hate mornings
13:50.55stansmithsomeone has a case of the mon-daze!
13:52.17ManxPowerI hate auto mechanics too.
13:52.51*** join/#asterisk hi365 (n=hi365@77.125.78.34)
13:53.25HavokmonMy last mechanics experience " You need a new computer"
13:53.46HavokmonMonths later, afte rI put in a new computer, and Check Engine light still comes on I discover....
13:53.55agallojblack, *couch* after my chief phoned their chief now "magically" works :)
13:53.58HavokmonThe o2 sensor they installed was unplugged ://
13:54.09stansmithlol my co-worker is snoring so loud
13:59.01jblackagallo: Yay for the old boy's network
13:59.23ManxPowerI got a new clutch, I needed a new clutch anyway, but it did not fix the problem I was having ("bucking" when going some speeds)
13:59.25*** join/#asterisk mattman99 (n=chatzill@ppp121-44-207-170.lns3.mel4.internode.on.net)
14:02.00beasty_how do i convert a .mp3 to .wav for asterisk ?
14:02.34stansmithbeasty_: google "convert mp3 to wav"
14:02.36ManxPowerbeasty_: however you would do it if you didn't have Asterisk.
14:02.45mattman99linux or winows?
14:02.51beasty_linux
14:03.09*** join/#asterisk hi365 (n=hi365@213.151.52.239)
14:03.17*** join/#asterisk tcseke (n=chatzill@217.20.134.239)
14:03.53[TK]D-Fenderbeasty_: just use asterisk-addons
14:04.11*** join/#asterisk hi365 (n=hi365@213.151.52.239)
14:04.12mattman99mpg123 will do it
14:04.38[TK]D-Fenderbeasty_: and I see your error in your MoH config.  You put "mode=file", this should be "mode=files".
14:05.49ManxPowerI thought it was mode=muffins
14:06.44beasty_idd
14:07.00beasty_[TK]D-Fender: it's my boss his 'ubuntu' apt-get install
14:07.10*** join/#asterisk hi365 (n=hi365@213.151.52.239)
14:07.18beasty_and feisty doesn't have a asterisk-addon package since it's br0ken
14:07.32ManxPowerbeasty_: you don't want to install from packages anyway
14:07.34ManxPowernot for Asterisk
14:07.44ManxPowernot for Zaptel
14:08.00[TK]D-Fenderbeasty_: Then I guess this just reinforces that you should have simply compiled from source like the rest of us
14:08.06jblackbeasty_: Theres good packages in hardy heron, which should be out soon. Excluding zaptel drivers, that is.
14:08.40beasty_ManxPower: well i had it compiled
14:08.41jblackI use packages, and understand the reasons why you may want to use them too.
14:08.58beasty_my boss removed it ... and installed from packages
14:09.02stansmithzaptel is easy to compile, the only dependency is glibc
14:09.45jblackzaptel is also dependant upon the kernel headers, for which there have recently been some impedance mismatches.
14:09.57stansmithtrue
14:10.37ManxPowerbeasty_: Asterisk is one of the VERY few applications that I compile from source
14:10.53mattman99i agree manx
14:10.58stansmithyoung chris sounds just like jay-z its disgusting
14:11.00beasty_true
14:11.05beasty_can you tell that to my boss
14:11.07ManxPowerThe others are SpamAssassin, ClamAV, and a commercial web interface to IMAP
14:11.15beasty_idd
14:11.19ManxPowerNotice all of those change fast
14:11.49jblackI'm on your boss'es side. I don't think production work should depend upon the recently buildable crack.
14:12.46*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
14:12.49ManxPowerjblack: Oh, I don't CARE if you use package or build from source.  What I care about is people coming here expecting us to help fix something that is broken or changed in their package .vs. Asterisk
14:13.36*** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
14:13.37*** mode/#asterisk [+o lmadsen] by ChanServ
14:13.50ManxPower"Asterisk isn't reading the files in /etc/asterisk.  I'm using package X from distro Y."  "Where is Asterisk compiled to look for it's config files?"  "I have no idea, I didn't build it."
14:13.52jblackmanxpower: sure, I can see how that's inconvienant.
14:14.13ManxPowerjblack: it also takes away support resources from people that need them.
14:14.45stansmithi mean, the book outlines the dependencies, so there really isnt an excuse for not being able to compile it from source
14:15.12ManxPower*grumble* I supposed I should take the mechanic that worked on my truck on a "test drive".  Need to make sure I have the buckets and cement first.
14:15.20jblackHeh. Production systems are where one can derive support income. Supporting CVS from 45 minutes ago doesn't feed your kids.
14:15.36ManxPowerjblack: I would never run CVS.
14:15.47ManxPowerI want a stable, working system, that is the same across all servers.
14:15.55*** join/#asterisk fskrotzki_ (n=fskrotzk@host198.textwise.com)
14:16.09ManxPowerCVS is French for "have more time than money", you know.
14:18.17jblackwhatever works for you
14:18.33stansmithsvn > cvs ?
14:18.35*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
14:19.40*** join/#asterisk Katty (n=Katty@hera.copi-rite.com)
14:19.45Kattyo hai! i upgraded your pbx.
14:19.50jblackFor some value of "greater than"
14:20.00Kattyx is greater than or equal to 4.
14:20.10Kattyand y=5
14:20.13Kattyplot it.
14:20.45Kattythat'd be like half a parabola.
14:22.54[TK]D-FenderKatty: Not without an exponent, otherwise you'll only get a straight line....
14:23.03stansmithLOL
14:23.06stansmith0wn3d
14:23.08[TK]D-FenderKatty: (Mew)
14:23.13Kattyyou are SO right.
14:23.33Kattyway to be on your toes this morning, mister mathematical ballet performer.
14:23.49jblackNo he's not.
14:24.00Kattyoh sure he is.
14:24.11[TK]D-FenderIts like they say, there's only three kinds of people out there ; those that can do math, and those that can't.
14:24.11jblackHe's on the balls of his feet, because there's more ways to give a curve than an exponent.
14:24.58Kattyanymore talk like that on a monday morning, and i'll make sure your exponentially removed.
14:25.13Kattyhave pity on my poor sleepy brain!
14:25.18jblackYour contribution here is logarithmic, at best. :)
14:25.50tzangerhaha no cable yet Katty?
14:25.58Kattyjblack: i resent that.
14:26.00jblackoh, that came out mean. I'm sorry. It was meant to be merely funny.
14:26.31stansmithquestion - is there some sort of switch statement that can be used in extensions.conf or must one resort to a lot of GotoIf()'s?
14:26.32Kattyjblack: i'm, by far, more than just some little f(x) = c log x!!
14:26.43Kattytzanger: no :/
14:26.59Kattyjblack: and you're wrong. i'm a female. i'm in no way a constant :P
14:27.14tzangeramen, sister!
14:27.43jblackSo, you're a high order polynomial? Up down, up down, everywhere a tight curve?
14:27.52jblackuh. gah. sorry again
14:28.04Kattyno references this early in the morning.
14:28.10jblackI can't stay out of trouble this morning
14:28.33tzangernothing wrong with tight curves
14:28.39jblackI didn't mean to imply innuendo. I'm still on my first pot of coffee. I swear
14:28.40stansmithO!!
14:28.40tzangereven girls like 'em
14:28.49Kattyi think i'm going to have to cite the story of the little equation that could
14:29.02Kattythe little equation that ate EVERYTHING
14:29.17Kattyand that little equation COULD EAT YOU!
14:29.21jblackI was referring to the old, sexist stereotypical rollercoaster of female emotions
14:29.26jblackYes Ma'am
14:29.33tzangerthere is just way too much innuendo in this channel this morning.  I love it
14:29.40jblackI didn't mean it@!
14:29.43tzangerthere isn't anything old or sexist about that
14:29.44Kattyjblack: yeah, but there are reasons for that.
14:29.52*** join/#asterisk BrokenNoze (n=root@host81-149-254-218.in-addr.btopenworld.com)
14:29.52tzangerit's a given fact
14:29.52Kattyjblack: and don't tell me males don't have mood swings
14:29.52jblackYes ma'am
14:29.55Kattyjblack: cause i know better :P
14:30.01tzangeroh we do
14:30.03tzangerfor sure
14:30.05Kattyjblack: i live with a very moody man right now
14:30.17Kattypoor guy has to do a tower climb today :<
14:30.19jblackSure... they go from quiet... to angry.. to quiet... to brooding.. back to quiet.
14:30.21*** join/#asterisk docelm0 (n=vircuser@h74.196.180.12.cable.rstb.cablerocket.net)
14:30.21BrokenNozeHi All, anyone know if there's a way to bridge two active channels?
14:30.29tzangerjblack: haha
14:30.39Kattyjblack: cave syndrome
14:30.44jblackUgh.
14:31.03[TK]D-FenderI am NOT a passive-agressive powder-keg just waiting to explode, and I'll *KILL* anyone who even looks at me funny at the suggestion of it!
14:31.33ManxPowerBrokenNoze: almost every call in asterisk is bridged between two channels.  Perhaps a less generic question with a bit of background would be more productive?
14:31.44Kattymorning Manx (=
14:31.45jblackHere, it's just me and a 14 year old girl. So, don't you go talking to me about female emotions. I'm ON THE FRONT LINES!
14:31.46yangI am wondering about video phone that use h.263 & h.264 does this have a working support in asterisk ?
14:31.57Kattyjblack: i am /so/ incredibly sorry.
14:32.08BrokenNozeManxPower: OK, then attended tranfer via the Manager API...
14:32.13Kattyjblack: you have my most sincere condolences.
14:32.14jblackNot as sorry as I am. :)
14:32.25ManxPowerBrokenNoze: now you at least are asking the right questions
14:32.39BrokenNozejust that question never seems to get me anywhere so thought i'd change it a little ;-)
14:32.41Kattyjblack: i can't even imagine how you can put up with a 14 year old..
14:32.50tzangerjblack: mine start young.  I've got a 4yo and an 8yo lil bundle of emotion, and a 33 year old leader in that territory
14:33.10[TK]D-Fenderjblack: Just because you see the fall-out with a front-row seat doesn't mean you necessarily have a clue whats really going on :)
14:33.16tzanger[TK]D-Fender: hahahahaha
14:33.26BrokenNozeManxPower:  if you can answer that question with a positive I'll owe you bigtime!
14:33.30jblackI feel like a elephant that can't turn around fast enough to deal with a pesky hornet.
14:33.36[TK]D-Fenderjblack: So grab some popcand enjoy the show!
14:33.52Kattyjblack: yeah, and if you tried...
14:34.01Kattyjblack: it'd be hell on earth, with a few exponents.
14:34.07[TK]D-Fenders/popcand/popcorn and/
14:34.23Kattyi remember being 14, just barely.
14:34.29*** join/#asterisk kraypius (i=user@c-67-175-202-53.hsd1.il.comcast.net)
14:34.50jblackI've found things work best if I just trudge through life, and ignore the mild stings. She doesn't mean them anyways.
14:35.02Kattyjblack: no, no she doesn't...
14:35.07ManxPowersend her off to boarding school.  problem solved
14:35.11kraypius<PROTECTED>
14:35.16KattyManxPower: horrors!
14:35.20KattyManxPower: you may never have children.
14:35.25ManxPowerAnd try to help others avoid the mistake of having children!
14:35.26jblackEvery once in awhile, I'll knock down her metaphorical hornet's nest to remind her that while I lumber, I can affect things she likes.
14:35.42HavokmonI have a 14 year old girl..   Brothers in arms :)
14:35.50ManxPowerKatty: My children have four paws.
14:35.55KattyManxPower: cheers.
14:35.58BrokenNozekraypius: X-lite
14:36.03jblackYou sexual deviant you.
14:36.11Kattywhat did we say about references!
14:36.21x86:P
14:36.26stansmith?
14:36.28ManxPowerjblack: you have no idea.
14:36.33jblackewwwww.
14:36.40*** join/#asterisk docelmo (n=vircuser@h74.196.180.12.cable.rstb.cablerocket.net)
14:36.50ManxPowerAnyway, I'm off to torture an auto mechanic.  Wish me luck.
14:37.08BrokenNozeOK, new question. How do i bridge two active calls in asterisk?
14:37.16jblackWe may not hear from him for days if he's on the losing end of that battle. ;)
14:37.28jblackbrokennoze: There's a bridge app
14:37.39[TK]D-FenderBrokenNoze: There is a redirect AMI call you can use.
14:37.47jblackExcept it's called something else
14:37.57*** join/#asterisk putnopvut (n=putnopvu@216.207.245.1)
14:38.00BrokenNozeIt doesn't bridge two ACTIVE channels though does it?
14:38.41BrokenNozei use redirect to redirect one active channel to another point in the dial plan, not to another actual open channel
14:38.42[TK]D-FenderBrokenNoze: Yes
14:38.54jblackPerhaps chanspy
14:39.02x86BrokenNoze: chanspy will do it
14:39.12x86or you could park/pickup
14:39.34BrokenNozeI'm basically directing a call to a hold extension
14:39.52jblackOh, Try the transfer application
14:40.10BrokenNozetransfer is blind though isn't it?
14:41.19anonymouz666putnopvut: do you care if I ask you directly an app_queue user question?
14:41.21jblackwell, there's the TRANSFERSTATUS variable....
14:41.23*** part/#asterisk PepOSX (n=angeldav@190.72.147.233)
14:41.34jblackWith typical j n+101 behaviour
14:41.38BrokenNozetrying to simulate an attended transfer by holding one channel, connecting another to the remote party, dropping the first party and connecting the held channel to the dialled endpoint
14:41.49Kattyanonymouz666: morning (=
14:41.53anonymouz666hello kaldemar
14:41.55anonymouz666ops
14:41.56nixguyis there a variable for the incoming number?
14:41.57anonymouz666Katty!
14:41.58*** join/#asterisk joshaidan (n=joshaida@S0106001c1023e838.tb.shawcable.net)
14:42.07jblackBrokenNoze: Hmm. Did you check to see if your phone already supports it with flash?
14:42.17nixguy$EXTEN is outgoing, so what is it for incoming?
14:42.25jblackflash - new number - "Hello there. have a call for ya" - hang up
14:42.27putnopvutanonymouz666: What's your question?
14:42.34stansmithomg omg omg
14:42.39BrokenNozejblack: no good, i need to do it via Manager as we support multiple SIP hardphones
14:43.06anonymouz666putnopvut: Once the caller is waiting in a queue, is it possible to reinject the PRIO?
14:43.18anonymouz666before I know it is possible.
14:43.23[TK]D-FenderBrokenNoze: Use "Redirect" and point to a dialplan context that will prompt the call like you described
14:43.27anonymouz666just using QUEUE_PRIO
14:43.37putnopvutanonymouz666: what do you mean by "reinject?"
14:43.41*** join/#asterisk lirakis (i=lirakis@pr0tected.us)
14:43.42anonymouz666change the prio.
14:44.01anonymouz666it is a important customer, he should be the first
14:44.19anonymouz666but he is already waiting...
14:44.25BrokenNozenixguy: CALLERID(num) or CALLERID(name)
14:44.29b11d.
14:44.37putnopvutanonymouz666: Well you can set the QUEUE_PRIO variable before he joins the queue, but once he's waiting you can't change his priority.
14:44.43stansmithis there a more "elegant" way to capture user key press for up to 3 digits delimited by "#" then this? --->  http://www.pastebin.ca/926030
14:44.54*** join/#asterisk af_ (n=getsmart@88-149-241-244.dynamic.ngi.it)
14:45.19[TK]D-Fenderstansmith: "core show application read"
14:45.34putnopvutanonymouz666: there are hackish ways you can go about moving a person up in the queue...
14:45.35[TK]D-Fenderstansmith: Yes, this should have been 1 line of dialplan.
14:45.40BrokenNozeFender: that works for the first two parts, but when i actually have the two active calls in progress...
14:45.41nixguyBrokenNoze: k thnx
14:45.43stansmith@#$@!
14:46.01Kattynixguy: are you australian?
14:46.18putnopvutanonymouz666: but no way of just manipulating the QUEUE_PRIO variable to move a caller already in the queue up.
14:46.26[TK]D-FenderBrokenNoze: What are you going to  do with your destination's CURRENT call?  You're saying "Person B, here's a call NOW, deal with it"
14:46.28anonymouz666putnopvut: ChannelRedirect on the Zap channel and then make hit the QUEUE_PRIO before join the queue
14:46.44putnopvutanonymouz666: yes something like that.
14:46.54anonymouz666putnopvut: you think that could be a interesting feature?
14:47.21putnopvutanonymouz666: I don't know really. It seems like something you should take care of prior to joining the queue to me.
14:47.29*** join/#asterisk docelm0 (n=vircuser@h74.196.180.12.cable.rstb.cablerocket.net)
14:47.32Kattydocelm0: mew.
14:48.05anonymouz666alright, thanks for your time.
14:48.11BrokenNozeFender: no Person A is calling person B to tell him that person C is holding. when person A hangs up, person B and person C need to be connected
14:48.12putnopvutanonymouz666: no problem.
14:48.43[TK]D-Fenderstansmith: and your PB showed I could punch in any multiple of 3-digits I wanted.
14:48.49x86BrokenNoze: that sounds like an attended transfer to me
14:48.51BrokenNozeif person A can't get hold of person B because he's already on a call, he'll just grab person C back and apologies for not being able to redirect
14:48.55*** join/#asterisk twisted (n=root@pdpc/supporter/active/twisted)
14:48.56*** mode/#asterisk [+o twisted] by ChanServ
14:48.59Kattyhi twisted!
14:49.03BrokenNozex86: that is EXACTLY what it is
14:49.09x86BrokenNoze: yeah that's easy :)
14:49.18x86BrokenNoze: you using analog channels?
14:49.43BrokenNozeno
14:49.52BrokenNozeSIP
14:50.30BrokenNozex86: lay it on me! I've been after it for months!!
14:50.50stansmith[TK]D-Fender: thats the idea, a caller can have an account with a suffix up to 3 digits
14:50.55*** join/#asterisk PepOSX (n=angeldav@201.243.76.220)
14:51.16[TK]D-Fenderstansmith: But your context allows me to enter 12 if I wanted.
14:51.29stansmithyea thats fine
14:51.34[TK]D-Fenderstansmith: Yuo only needed 1 single line of dialplan with "Read" to do this properly.
14:51.58*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
14:51.58*** mode/#asterisk [+o russellb] by ChanServ
14:52.43Kattyhi russel!
14:52.54stansmith[TK]D-Fender: actually, its somewhat tricky with Swift(), but i believe this is my work around in case anyone is interested ---> http://www.pastebin.ca/926040
14:53.16stansmithi havent been able to determine whether ${SWIFT_DTMF} is persistant or only holds the last digit
14:53.33stansmith*hopefully* it only  holds the last digit, ill have to check the source
14:54.22BrokenNozex86: so what do i do?
14:54.26*** join/#asterisk RoyK_ (n=roy@fw.fortel.no)
14:54.37Kattyroy (=
14:55.18[TK]D-Fenderstansmith: It should only take 1 line fo dialplan for your read.. thats a lot of filler for nothing...
14:56.05stansmith[TK]D-Fender: yea but if the caller presses something while Swift() is executing, swift will try and go to that extension (via swift.conf --> "goto_exten=yes"), so i need to catch that key press
14:56.31*** join/#asterisk JenniferAkemi (n=akemi@206-248-164-21.dsl.teksavvy.com)
14:56.31stansmithif your talking about the noop(), i put that in cause i didnt want to renumber everything, n priority ftl
14:56.36lnxi have put some exten => 10,7,NoOp(exten_Dialstatus ${DIALSTATUS}) in my dialplan and ${DIALSTATUS} is always ampty, but the call is succesful. Can u tell me why? please
14:56.54JenniferAkemigood morning everyone
14:57.01[TK]D-Fenderstansmith: Ah, it aborts on DTMF?  Ok, you will need an IVR approach then
14:57.26russellbKatty: greetings :)
14:57.30[TK]D-Fenderlnx: pastebin your entire context and a sample call's CLI output at verbose 10
14:57.32[TK]D-Fender~pb
14:57.33jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
14:57.33stansmith[TK]D-Fender: yea, swift will go to the extension if it exists, but i think this is a work around, i need to test it
14:57.35[TK]D-Fender^^^^^^^^^^^
14:57.41*** join/#asterisk ccvp (n=ccvp@66.0.46.210)
14:57.54[TK]D-Fenderstansmith: You should read "_X", not "_XXX", and collect each digit.
14:58.08[TK]D-Fenderstansmith: And keep a length count so they don't enter too mcuh.
14:58.46*** join/#asterisk LjL (n=ljl@ubuntu/member/ljl)
15:00.37Kattyhi ccvp (=
15:01.07ccvphello Angela
15:02.11*** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com)
15:03.49lnx[TK]D-Fender: http://pastebin.com/m7cf4d2ab
15:05.04*** join/#asterisk CVirus (n=GoD@196.205.192.185)
15:05.06[TK]D-Fenderlnx: You never call "Dial" anywhere is there, of course ${DIALSTATUS} is going to be empty.
15:05.32*** join/#asterisk wmaulik (n=wmaulik@158.59.192.218)
15:06.43lnx[TK]D-Fender: asterisk does not use Dial() via outgoing/  ?
15:06.50*** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584628.dsl.bell.ca)
15:07.47stansmiththat is tits, i think my solution works
15:08.22*** join/#asterisk bkw_ (n=brian@adsl-64-149-54-40.dsl.tul2ok.sbcglobal.net)
15:08.23stansmithi think im gonna submit that to the asterisk cookbook wiki
15:08.26*** join/#asterisk acron17 (n=joe@p5089F6C0.dip.t-dialin.net)
15:08.34acron17hi there
15:08.42stansmith~hi acron17
15:08.43jbotMany greetings, acron17, most strange traveller, to this IRCdom of plenty.
15:09.14acron17i know that asterisk can bi configured to use info-messages for dtmf
15:09.26acron17that works great with me voip-app
15:09.31stansmithacron17: IVR?
15:10.16Kattyto play warcraft or not play warcraft...
15:10.20acron17just how the dtmf tones are transported
15:10.26cmantitosilly Katty. the answer is ALWAYS play.
15:10.28cmantito:P
15:10.33stansmithcs > wow
15:10.37cmantitonahh
15:10.42ccvpacron17
15:10.44ccvps/bi/be
15:10.46ccvpimo :)
15:10.47stansmithlol
15:10.48Kattyhrmm
15:10.55*** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com)
15:10.55Kattyto play the hunter, the mage, or the pally.. hrmm
15:11.00stansmithacron17: are you asking a question or stating a fact?
15:11.01acron17is there way a voip-app may tell asterisk which kind of DMTF mode to use?
15:11.01ccvpcan u get level 80's
15:11.03ccvpin wow yet?
15:11.08cmantitonot yet
15:11.08Kattynot yet
15:11.08cmantitosoon
15:11.18ccvpi wonder how hard it'll be
15:11.26ccvpbig experience curve, ie: 2-3months RL time, or PL in few days heh
15:11.27cmantitoKatty: what realm/region/faction are you, out of curiosity?
15:11.39Kattycmantito: horde, llane
15:11.44ccvpi never got into WoW
15:11.48ccvpim waiting for blizzards next free game
15:11.49ccvpDiablo 3
15:11.49cmantitoKatty: For the horde!
15:11.55Kattycmantito: (=
15:12.09ccvpDiablo 2 > WoW
15:12.11Kattyi'm not that serious...
15:12.20Kattybut i have a 70 mage and a 70 hunter, and my pally is uh... 13
15:12.24stansmithLOL
15:12.26cmantitohehe
15:12.31cmantitoI have a lvl 50 shammy
15:12.34ccvpi played wow once like 1.5 years ago
15:12.35drfreezeHello
15:12.39ccvpand had some class that had a blue ghost
15:12.42ccvpas my pet, it was a tank
15:12.44[TK]D-Fenderlnx: that var only gets set when App_dial is called
15:12.54cmantitoI was serious for a little while, but I've found something else that holds my fantasy interests
15:12.58cmantitomore dorky than wow
15:13.02drfreezeI've had some problems this morning with the not being able to hear incoming calls
15:13.05Kattya warlock gets a blue 'ghost' tank
15:13.16[TK]D-Fenderlnx: and the fact you are IN the dialplan at all means the call is in progress and has ALREADY been answered
15:13.25drfreezeOne just occurred where I could hear the incoming call, then they went silent
15:13.40cmantitoI've taken up LARPing :3
15:13.56acron17is there way a voip-app may tell asterisk which kind of DMTF mode to use?
15:14.03drfreezeOn a system that is working normally, what could be a potential problem that is causing this?
15:14.04lnx[TK]D-Fender: i understand thankl you
15:14.08joshaidanIs there an * postgresql app that lets your run queries in extension.conf similar to MySQL()
15:14.20stansmithjoshaidan: yes
15:14.27stansmithres_postgresql i think
15:14.28[TK]D-Fenderjoshaidan: use ODBC
15:14.32stansmith^
15:15.21stansmithO'REILLY has a book called "cookbooks in a nutshell"?
15:15.21acron17no answers or no hints?
15:15.42*** join/#asterisk pithen (n=pithen@mail.graphlogic.com)
15:15.45tzangerstansmith: heh
15:16.24angryuserofftopic someone with good knowledge of openvpn here? what is this ? write UDPv4: Operation not permitted (code=1) (openvpn log) unable establish connection
15:16.44Kattysounds firewall related
15:16.51stansmithsounds permission related
15:16.58Kattycould be that too
15:17.09pithenHey all.. I have an * system with a Digium (4xFXO) card.. for the most part everything works fine, but occassionally the system will not answer calls from PSTN, and when dialing an outgoing call I just get loud static.. Reboot the system and all is fine again. Any thoughts? Card going bad?
15:17.13stansmithyoull get a message similiar to that when you try to do something root as a non-root user
15:17.18*** join/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net)
15:17.30Kattypithen: i've had plenty of those problems
15:17.32stansmithpithen: how do u dial out with 4 fxo?
15:17.39Kattypithen: i think it's just the cards really
15:17.46JenniferAkemilarping seems like it would be better exercise than wow at least
15:17.52pithenstansmith, did i swtich my terminology..fxs?
15:17.57ThatKidKelWhen calling Dial(SIP/xxxx) is it not possible to get the SIP failure reason?  Other than CONGESTION or BUSY?  Ie.  404
15:18.01stansmithpithen: i get confused too haha
15:18.08acron17russellb: you said in asterisk-dev, that there is a possibility to do that,,,,
15:18.12Kattypithen: we had to reboot our server from random hiccups once a month or so
15:18.20pithenKatty, this is almost every day nowe
15:18.29Kattyhrmm. that is a bit excessive.
15:18.31lnx[TK]D-Fender: how can i call an agi script without a callfile please
15:18.38Kattydo you have another card you can throw in there to test?
15:18.48drfreezeAnyone know why this doesn't reboot a polycom phone: sip notify polycom-reboot ext
15:18.52stansmithlnx: exten => s,n,AGI([script name])
15:18.59[TK]D-Fenderlnx: What do you need your script to do anyways?
15:19.21pithenKatty, yeah the boss is ready to toss out the whole system.. I don't have anything else on hand, but was hoping there was some way to test the card before I go out and blow another $x00
15:19.27[TK]D-Fenderdrfreeze: because you need to set your provisioning configs to accept the notice.
15:19.29stansmithlnx: place the script in /var/lib/asterisk/agi-bin/ (default location)
15:19.47Kattypithen: i can sympathize with that. for awhile, the boss was ready to toss ours too
15:19.51stansmithchmod +x !
15:20.09stansmithchmod +x digiumcard should fix it
15:20.10lnx[TK]D-Fender: call a number and analise ${DIALSTATUS} automatically
15:20.14Kattypithen: i'm sure there's a way to enable debug for the card, but i couldn't tell you what to really look for.
15:20.19lnxstansmith: ty
15:20.33pithenKatty, ill look into that, thank you
15:20.42drfreeze[TK]D-Fender: ahh. How do I do that?
15:20.46[TK]D-Fenderlnx: Think about what kind of channel you are placing your call against.
15:20.56[TK]D-Fenderdrfreeze: Go check your admin guide.
15:21.16stansmithlmadsen: is it still possible to create an account for asteriskcookbook.com ?
15:21.31lnxstansmith: exten => s,n,AGI([script name]) when invokes this line?
15:21.33lmadsenstansmith: yes, you have to follow the instructions on the site though
15:21.41lmadsenstansmith: which basically means you have to email the guy who will set it up
15:21.44lnx[TK]D-Fender: maybe i must create one
15:22.03stansmithah...create account wasnt creating account
15:22.18stansmithlnx: i dont understand your question
15:22.26[TK]D-Fenderlnx: Just look at the channel you are originating and think about what kind of channel would allow you to deal with it not being answered....
15:23.00pithenAnybody have experience with Rhino? Quality hardware? Im hoping to get a new system altogether with a 24 channel fxs bank and a fxo card
15:23.18Kattypithen: we had a rhino server once.
15:23.25lnx[TK]D-Fender: i have to call *94 example
15:23.30[TK]D-Fenderpithen: Not worth it.
15:23.33pithenKatty, i notice thats past tense :)
15:23.39*** join/#asterisk Fusoya (i=quality@togi.homeunix.org)
15:23.44Kattypithen: tho, we were more curious about the Pretty Software
15:23.49[TK]D-Fenderpithen: Just get a SIP gateway
15:23.55Kattypithen: eh, i guess it worked okay with rhino equipment
15:24.04Kattypithen: but, for the price, ...
15:24.08Kattypithen: not really worth it
15:24.11[TK]D-Fenderlnx: What is this talk about dialing "*94" have to do with your call-file>
15:24.13[TK]D-Fender?
15:24.26coppicearen't rhinos an endangered species? :-\
15:24.36Kattycoppice: i don't think so.
15:24.38pithenmy current setup involves about a dozen Sipura units..its hell to maintain
15:24.46[TK]D-Fenderpithen: as CB's go, Rhino is decent, but its not a solution I advise unless you need it or ahve the extra T1 ports to spare.  Even then....
15:24.58coppice[checks with the WWF] wrong!
15:25.08Kattynot the first time ;)
15:25.08[TK]D-Fenderpithen: use a bigger SIP gateway like AudioCodes MP-124 or Mediatrix 1124
15:25.24pithen[TK]D-Fender, will check those out
15:26.09coppicechoose a big name like mediatrix. they aren't any less buggy, but everyone has worked arounf their bugs so they are compatible :-)
15:26.11lnx[TK]D-Fender: it was a testing :) now i know that i must use Dial()
15:27.21FusoyaIs there a way to fix the "ring requested on channel already in use" zaptel bug that wouldn't require me to restart the server?
15:27.56ccvpin topic, what exactly is switchvox
15:28.04ccvpan expensive turkey provider for businsesses?
15:28.05ccvpseems pricey
15:28.07Fusoya(ast1.2.13 zap1.2.11)
15:28.09ccvpturn key
15:28.35stansmithbruce@oreilly.com no worky?
15:29.19lnx[TK]D-Fender: my goal is : make timing calls and analise ${DIALSTATUS} of that calls. I hope it is acchieve in enxtensions.conf
15:30.11lnx[TK]D-Fender: and *94 is auto answering :)
15:32.49lnx[TK]D-Fender: it would be a testing mechanism to examine calls are going well
15:33.12HavokmonRegarding configs, is an upgrade from * 1.2 to 1.4 pretty much seamless?  I've seen some differences in some configs - like parking....
15:33.32FusoyaNobody knows a way to un-hang a zap channel without a full restart in Asterisk 1.2?
15:34.32[TK]D-FenderHavokmon: Go read upgrade.txt and all the dozens of articles about whats been changed
15:35.04[TK]D-FenderFusoya: "soft hangup Zap/1-1", etc
15:35.08ccvpd-fender
15:35.21ccvpis [TK] a guild/clan tag for you in an online game? :)
15:35.23Fusoya[TK]D-Fender: Ooooh, let me try that one.
15:35.26ccvpTha Killaz? :)
15:35.41[TK]D-Fenderccvp: Many years ago, yes.  I keep it for sentimental reasons.
15:35.45ccvpheh
15:35.53ccvpdid you play qw/dm?
15:35.56*** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net)
15:35.58[TK]D-Fenderccvp: and no, not "Tha Killaz"
15:36.06stansmith~caturday
15:37.06ccvpbleh wtf, its 75 degrees yesterday and today, and a front is rolling through around 2am
15:37.12ccvpand gonna drainop 50 degrees in 1 hour, massive tornados yet ag
15:37.14ccvpagain
15:37.21stansmithccvp: location?
15:37.24ccvpnorth, AL
15:37.27ccvpjust below TN
15:37.41stansmithits a nice 50 something here in columbus..hope that front doesnt reach here
15:37.47ccvplook at TX now
15:37.49ccvpon weather.com
15:37.52ccvpthick purple line of storms
15:37.54ccvppurple > red
15:38.26ccvplook at that front in TX
15:38.28ccvpits gigantic
15:38.52ccvphttp://www.weather.com/maps/news/severewinterforecast/floater1_large_animated.html?from=hp_main_maps
15:39.34stansmithat least its not gonna snow
15:39.49*** join/#asterisk ajohnson (n=ajohnson@63.147.46.186)
15:39.51ccvpthat area in TX where there is a blizzard now
15:39.55ccvphad 83f 2 days ago
15:40.06ccvpglobal warming ftw :)
15:40.13stansmithyou can say that again!
15:40.23*** join/#asterisk dominic1 (n=dob@213.221.82.242)
15:40.26*** part/#asterisk dominic1 (n=dob@213.221.82.242)
15:42.56Fusoya[TK]D-Fender: Doesn't seem to be able to hang up this call
15:43.37pithenSo heres a dummy question.. these sip gateways you've pointed me to all have RJ21x connectors.. is that something I can just go to HomeDepot and pick up a junction block or something?
15:45.07x86RJ21X? that's an Amphenol 25-pair connector
15:45.40x86no, Home Depot does not sell them
15:46.20pithenmeh..im going to have to rip apart the whole network closet for this project ;)
15:46.24x86need to go to some wiring specialty place... here we go to Graybar, which is a wholesaler
15:46.40Kattyif you're doing Background(menu) and you want to wait 15 seconds before repeating it...
15:46.46*** join/#asterisk SirThomas (n=tomc@mail.kendeco.com)
15:46.50x86you can probably hit up some local wiring vendor for RJ21X cables
15:46.53Kattydo you use Wait(15)? For some reason, when i use wait, it won't accept my input
15:47.14[TK]D-Fenderpithen: I haven't seen a mass gateway with 24x RJ11 before.
15:47.17x86WaitExten is what you want Katty
15:47.19De_MonKatty use WaitExten
15:47.21Kattythanks.
15:47.37[TK]D-Fenderpithen: Closest I've seen is the SPA-8000 w/ 8 FXS
15:47.57[TK]D-Fenderpithen: When when you think of it is better than 4x 2-port gateways
15:47.57FusoyaLOL, I love how the bug for this (6147) just got closed for no reason
15:48.17x86Fusoya: happens quite frequently it seems ;)
15:48.29[TK]D-FenderKatty: "Set(TIMEOUT(response)=15"
15:49.37*** join/#asterisk ming_zym (n=ming_zym@123.103.29.198)
15:49.55pithen[TK]D-Fender, thats what was appealing about the rhino bank, it had rj11's.. i suppose if im going to do this though i autta do it right
15:50.11[TK]D-Fenderpithen: Which Rhino has RJ11's?
15:50.15[TK]D-Fenderpithen: News to me...
15:50.38coppicei think their analogue cards do
15:50.39tzafrir[TK]D-Fender, huh?
15:51.11De_MonFusoya http://bugs.digium.com/view.php?id=6147?
15:51.21De_Monno that can't be what you're talking about
15:51.34tzafrirwe provide RJ11-s . or an optional bundled interface. It's much simpler to test the unit when you have an RJ11 output for each port
15:52.01FusoyaDe_Mon: Yeah it is.
15:52.19De_MonFusoya the last comment was in 2006, and says exactly why it was closed
15:53.02FusoyaDe_Mon: "I don't believe this bug is going anywhere" isn't a reason in my book to close a ticket. :)
15:53.06FusoyaBut that's me
15:53.49FusoyaI can't find any more recent ones on this issue, but I see reports elsewhere of people having it very recently, with recent versions
15:54.11De_Monread the whole comment and if you still feel that way, we'll just have to agree to disagree
15:54.41FusoyaWe'll have to do that then, because I've read it.
15:54.44De_Monbunching a bunch of problems into 1 ticket is an excelent reason to close that ticket and make them open separate issues.
15:55.00FusoyaThere was no reason to believe the issues were seperate.
15:55.03FusoyaSeparate, rather.
15:55.22FusoyaAll produce the same symptom... how do you track a bug if not by symptom when the cause is unknown?
15:55.42pithenhmm..maybe im mistaken "Analog Output: 24 Loop start lines via 25 pair amphenol connector " i thought i saw a picture with 24 ports on it
15:56.01FusoyaIs it better for every user to open a ticket that says "hey I don't know why but I'm getting the same error as the people who posted the last 18 tickets"
15:58.12[TK]D-Fenderpithen: So you'll need a break-out box
16:00.17*** join/#asterisk BCS-Satori (n=somewher@75.148.21.113)
16:01.04BCS-SatoriI noticed that my app followme was not loaded in asterisk but the module was there, so i did load app_followme.so which worked from the console.  Will this stay there forever or when I reboot will it disappear or should i just add it under modules?
16:01.24stansmithBCS-Satori: pb modules.conf
16:01.32stansmither..check modules.conf
16:02.08BCS-Satoristansmith: its not listed there before or after the load command
16:02.14JenniferAkemiBCS-Satori: you must have autoload=yes or have it loaded explicity
16:02.21stansmith^
16:02.50BCS-Satoriautoload is set to yes, so i guess i need to laod it explicity then
16:03.04JenniferAkemii think you should really only need one or the other
16:03.32JenniferAkemiyou could always try rebooting and see if it's loaded :)
16:03.53JenniferAkemii'm right now playing with modules.conf with autoload=no
16:04.47*** join/#asterisk XnOSX (i=d491af1a@gateway/web/ajax/mibbit.com/x-d01891759a901cbf)
16:04.48*** join/#asterisk kraypius (i=user@c-67-175-202-53.hsd1.il.comcast.net)
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16:06.04kraypius<PROTECTED>
16:06.22[TK]D-Fenderkraypius: ~freepbx
16:06.26[TK]D-Fender~freepbx
16:06.26jbot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
16:06.42stansmithlol
16:07.32JenniferAkemido i need adsi for callerid on call waiting?
16:07.46stansmithhow would a rookie programmer implicity set TIMEOUT()'s for each context in extensions.conf?
16:07.51stansmithhypothetically speaking that is
16:08.22[TK]D-FenderJenniferAkemi: No.
16:08.33JenniferAkemicool thanks [TK]D-Fender
16:08.36[TK]D-Fenderstansmith: There is no "implicit".  You have to set them
16:08.58*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)
16:08.58*** mode/#asterisk [+o anthm] by ChanServ
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16:10.35Kattyhi anthm (=
16:10.47anthmhi =D
16:11.15*** join/#asterisk VaNNi (n=VaNNi@216.70.165.200)
16:11.33stansmithwhich of these 2 context is recommended in production?   http://www.pastebin.ca/926142
16:12.02stansmithpretty much having While() vs. "t" extension
16:13.18pithenOkay.. final question to all before I get out of your hair: as far as FXO cards go, I need a 4 port card; Digium the best in terms of reliability/price, or should I consider other manufacturers (which?)?
16:13.45stansmith!dialogic
16:13.54stansmith(w/ asterisk that is)
16:14.12[TK]D-Fenderstansmith: 1st, without "waitexten", but rather "autofallthrough=no", and also with an actual patter that will match.  Next it'd be nice if your timeout counter worked.
16:14.38[TK]D-Fenderpithen: I use Sangoma A200d's exclusively for that
16:14.48*** join/#asterisk sudhir492 (n=sudhir@adsl-18-47-35.mco.bellsouth.net)
16:14.53x86wow... all of my branches made a combined total of 47,392 calls LAST WEEK ALONE ;)
16:14.58stansmithi have REPEAT defined in [globals]
16:15.15sudhir492Is there an FXO/FXS card for USB?
16:15.24x86[TK]D-Fender: A20002D-x :)
16:15.28[TK]D-Fenderstansmith: Wasn't talking about the global which I took for granted you'd at least done properly...
16:15.50coppicesudhir492: dunno. ask tzafrir :-)
16:15.52[TK]D-Fenderstansmith: You have ANOTHER error.  Stare at it till you see it, or your eyes bleed.
16:15.58x86sudhir492: Xorcom makes USB channel banks, talk to tzafrir
16:16.10tzafrirsudhir492, hi
16:16.58sudhir492hi tzafrir
16:18.04*** join/#asterisk docelmo (n=vircuser@h74.196.180.12.cable.rstb.cablerocket.net)
16:19.13*** join/#asterisk XnOSX (i=d491af1a@gateway/web/ajax/mibbit.com/x-de5349e0ec31b467)
16:19.46x86tzafrir: what kind of density does Xorcom offer with those things?
16:19.56x86tzafrir: can yall do 48 ports in 1U yet?
16:20.23*** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net)
16:20.32ThatKidKelCDR Question--Is it possible to have CDRs written to the database AND in cases of DB failure to the Master.csv?
16:20.43stansmithThatKidKel: yes
16:20.48*** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net)
16:20.53ThatKidKelcan you point me to documentation?
16:20.58tzafrirx86, 32 ports . But you can connect many of them.
16:21.04BCS-SatoriIs there a way to detect a hangup and execute the next line of code in a exten series?  For example, i am setting up a monitor that will send the message they record to email, but if the user hangs up the phone it will never email. the only way it currently emails is when they press #.  Is there a way to decect the hangup and move to next step?
16:21.06stansmithThatKidKel: its really up to you, i do it via an AGI script
16:21.23ThatKidKelah...
16:21.55pithen[TK]D-Fender, excellent..thank you so much
16:22.13stansmithThatKidKel: i think it gets written to Master.csv regardless
16:22.35ThatKidKelok.
16:22.58ThatKidKeli've noticed something werid in my Master.csv..  If its a NO ANSWER, i get two records..
16:23.08ThatKidKelOne with the destination of "s" and the other with the real destination
16:23.35stansmiths = destination to go here when there is nothing explicitly defined
16:24.11stansmithi use "s" as an entry point to the context
16:24.14JenniferAkemimaybe the phone is calling back
16:25.38stansmith[TK]D-Fender: were you trying to tell me to do "exten => _1,n,..." rather than "exten => 1,n,..." ?
16:25.52stansmithwhen you said pattern matching
16:26.07[TK]D-Fenderstansmith: No.
16:26.24[TK]D-Fenderstansmith: You had nothing DIALABLE in there.
16:26.30*** join/#asterisk sergey_masushko (n=sergey@66.243.68.219)
16:27.07stansmith[TK]D-Fender: Swift() jumps to the extension...so when i press 1, it will go the the "1" extension and execute that
16:27.18JenniferAkemithere isn't even a exten => 1,n.... in your thing
16:27.23[TK]D-Fenderstansmith: http://www.pastebin.ca/926142 <- Where do you see anything you can actually DIAL in there?  You have no dial patterns.
16:27.37stansmithim not trying to dial though..its an IVR
16:27.41stansmithstrictly an IVR
16:27.47[TK]D-Fenderstansmith: You are NOT getting it..
16:27.53stansmithi know im not :-(
16:28.11[TK]D-Fenderstansmith: You can't dail anything in there.  Everything comes up "i" and sends you in circles.
16:28.53stansmithoooooooooooo
16:29.10stansmithhaha, the "..." meant there are extensions in there that i didnt PB cause i thought they were irrelevant
16:29.12[TK]D-Fenderdial*
16:29.20stansmiththey were just taking up space
16:29.46*** join/#asterisk CunningPike (n=arodgers@204.239.12.183)
16:29.54[TK]D-Fenderstansmith: You show me something that laks some important pieces and I will assume you didn't think to actually make it DO anything.
16:30.38JenniferAkemi[TK]D-Fender: do you speak french too?
16:30.46[TK]D-FenderJenniferAkemi: Naturally.
16:30.56JenniferAkemi[TK]D-Fender: nice :)
16:31.11[TK]D-Fenders/laks/lacks
16:31.15stansmithmy question was more directed towards "should i use a while loop or the "t" extension for the main extension in that context"
16:31.15*** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com)
16:31.20stansmithsorry for the mix up [TK]D-Fender
16:31.26JenniferAkemiman i wish i didn't do install samples
16:31.36JenniferAkemii wonder if thats why all these conf files are here.
16:31.39[TK]D-Fenderstansmith: You ask which is better for production and I'll check if the ENTIRE thing works or not :)
16:31.42stansmithJenniferAkemi: it is
16:31.59JenniferAkemii wish it would just make them blah.conf.sample
16:32.10stansmithhaha, i was like "well he prob doesnt care to see "exten => 1,1,Goto(login,s,1)"
16:32.31*** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com)
16:32.38[TK]D-Fenderstansmith: Yes I want to see if you've done anything else silly...
16:33.23[TK]D-Fenderstansmith: and what I was pointing out earlier is that your 2 context samples showed a bug between them.
16:33.34[TK]D-Fenderstansmith: exten   => s,5,Set(COUNT=$[${COUNT} + 1]) <- good.
16:33.44[TK]D-Fenderstansmith: exten   => t,1,Set(COUNT=${COUNT}+1) <- bad.
16:33.55[TK]D-Fenderstansmith: Please pay attention to your use of * evaluations.
16:34.14[TK]D-Fenderstansmith: Which is why I told you your rety loop was busted
16:34.17[TK]D-Fenderretry*
16:34.42stansmithahhhh - yea, the second example was my first iteration developing the IVR, i didnt really realize how $[] was supposed to be used
16:35.03stansmithby second iteration, i knew better *cough* read the f-ing manual *cough*
16:35.13[TK]D-Fenderstansmith: 57th times the charm....
16:35.37stansmithdid i mention im fresh out of college ?
16:35.44stansmithgood thing is, im highly trainable
16:35.52stansmithlike a young puppy
16:35.55*** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com)
16:36.25ZPerteedoes anyone have any experience with integrating asterisk with older pbx?
16:36.42JenniferAkemiwhat kind of experience
16:36.50stansmithwhat kind of integration?
16:37.00JenniferAkemii'm connecting asterisk to my older stuff
16:37.12JenniferAkemibut, it's just like being a pri provider
16:37.24JenniferAkemiit being the older switch
16:37.58stansmithDe_Mon: thanks, i just saw it
16:38.04ZPerteeI have an avaya partner acs system and I want to put asterisk as an analog extension on it.  however, asterisk won't answer my call
16:38.06stansmithur message that is
16:38.13JenniferAkemiwhat is a "skinny channel" in asterisk?
16:38.17*** part/#asterisk sergey_masushko (n=sergey@66.243.68.219)
16:38.31drmessano--> )(
16:38.32ZPerteeI will also sit asterisk in front of avaya as well
16:38.50[TK]D-FenderJenniferAkemi: SCCP protocol (Cisco Phones)
16:39.01JenniferAkemi[TK]D-Fender: thank you
16:39.04ZPerteewhat signalling would I use on my Digium fxo card?
16:39.16stansmithfxs_ks is recommended
16:39.18drmessanoSCCP, one letter off from CCCP.. no coincidence
16:39.28stansmithZPertee: wait dont listen to me
16:40.09ZPerteeok
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16:43.13stansmithZPertee: maybe i am right
16:43.19stansmithare you referring to zapata.conf?
16:43.26[TK]D-FenderZPertee: Typcally fxs_ks
16:43.34stansmithZPertee: yea i was right
16:45.31tzafrirdrmessano, but also one letter from scp
16:45.34ZPerteeok.  so if asterisk doesn't answer then either the dial plan is wrong or the fxo card?  any idea how I configure my fxo cards in asterisk now?  supposed to be automatic but I can't figure out how to see if the card is even working or not
16:46.36stansmithZPertee: in zapata.conf, u need to tell it where to enter the dialplan at
16:46.46tzafrirZPertee, asterisk doesn't answer?
16:46.51stansmithfor me, it is "context=init" because i have a context [init]
16:47.15tzafrirwhy does it sound familiar
16:47.18tzafrir?
16:47.30stansmitho lol
16:47.43stansmithZPertee: in your context, is the first thing Answer()?
16:47.55ZPerteeyes and I have it pointed there
16:47.58stansmitho..
16:48.08ZPerteecan I test the card from the asterisk console?
16:48.15ZPerteeor get any info?
16:48.28stansmithZPertee: are you running zaptel service?
16:48.41stansmithi.e - is ur driver loaded?
16:48.56tzafrircat /proc/zaptel/*
16:49.01stansmith(i like `lsmod | grep zap`)
16:49.11ZPerteethanx...newb
16:49.30ZPerteeyep its running
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16:49.52ZPerteei'll play with it some more, prob something stupid...usually is
16:50.16stansmithZPertee: if you edited zapata.conf or extensions.conf you should restart asterisk
16:50.35ZPerteeok
16:50.36stansmithdialplan reload will reload extensions.conf while asterisk is running, but i dont know the command to reload zapata.conf
16:50.43stansmith"dialplan reload"
16:51.33*** join/#asterisk supjigator (n=shanebur@152.53.16.10)
16:51.39ZPerteewhats difference between "reload" and "dialplan reload"?  anything
16:51.50stansmithreload reloads everything perhaps?
16:52.03stansmithi know there is a command to reload everything, that may be it, i have never used it
16:52.34ZPerteeseems to take longer... so its possible
16:52.42Qwellhelp reload
16:53.21ZPertee<PROTECTED>
16:53.21ZPertee<PROTECTED>
16:53.27tzafrirstansmith, for most things: 'reload' or 'modules reload chan_zap.so'
16:53.46stansmithah..i tried "modules reload zapata" and it error'd
16:55.11ZPerteewould it hurt anything if I hava an ata connected to * with extension 1.  and then I dial ext.1 and and have a telephone line plugged from ata to * fxo?
16:55.27ZPerteedial from softphone
16:55.40ZPerteetrying to test system
16:57.07stansmithi test via softphone
16:57.12stansmithmake sure u edit sip.conf accordingly
17:00.25stansmithdoes anyone here use asterisk to port knock their gateway/firewall? i.e - is that idea original or old news?
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17:02.23HavokmonYou mean allow incoming connections (RDP/ssh/etc) from the same IP a user registered their phone from?  Interesting thought...
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17:06.03ZPerteeIf my fxo card on my * box is configured correctly should I at least be able to see an incoming call in the asterisk console?
17:06.15stansmithi mean to have asterisk on same box as your gateway, then call it when you are not home, dial extensions which are really just the ports you want to knock on, and if its the right sequence then run a simple bash script to alter iptables
17:06.36stansmithim waiting on my w2 so i can buy a card at home and try it
17:06.47Qwellstansmith: you haven't gotten it yet?
17:06.54stansmithi mean, the actual cash
17:06.57Qwelloh
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17:07.21stansmithZPertee: yes
17:07.41ZPerteestansmith if your firewall can be controlled by the command line couldn't use just use the System() application
17:07.45stansmithyou might see something like "starting simple switch on ... "
17:08.08stansmithyea, more then 1 way to skin a cat..typical with *nix
17:08.24ZPerteeyou gotta love it though
17:10.47*** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net)
17:12.02grandpapadotHi all.  In this outbound dialing extension: exten => _91NXXNXXXXXX,1,dial(Zap/g1/${EXTEN}) If g1 has no free lines, how do I get congesion playing to the callers?  I tried just adding a priority right below this one with congestion but it's not working.
17:12.38*** join/#asterisk ManxPower (n=manxpowe@209.16.72.139)
17:13.14ManxPowerWhat's that online website/store that has to do with mozilla that sells VoIP stuff.  I just can't remember the name of it.
17:14.12ManxPowervoxilla was the one I was thinking, just found it
17:14.14stansmithvoipzilla?
17:14.24stansmiththats actually a good name...
17:17.41grandpapadotIf a zaptel group has no free channels shouldn't Dial() exit and the call flow to the next priority?
17:19.28[TK]D-Fendergrandpapadot: it does, so you've done something wrong...
17:19.35[TK]D-Fendergrandpapadot: pastebin is your friend.
17:19.42*** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net)
17:20.38coppiceexcept when it remembers your embarassing mistakes
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17:37.38generalhancan someone tell me what this 'fullcontact' line is all about in the sip DB
17:37.53grandpapadot[TK]D-Fender: Looks like n+101 is where the calls were going ...
17:37.59generalhanaccording to how the table is setup it cannot be NULL, but i cant figure out what i need there
17:38.27[TK]D-Fendergrandpapadot: You should not be doing priority jumping since 1.2
17:38.44x86priority jumping is such a hack
17:39.08x86grandpapadot: create a macro that checks dialstatus and makes decisions based on that
17:39.23Wayhighgot a refund from x100p.com.
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17:45.39tzafrirWayhigh, interesting
17:45.46tzafrirThey actually respond
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17:48.27Wayhightza: yeah but it took me a lot.. they're going through some HR problems apparently
17:48.45Kattymmm, lunch.
17:48.48Kattyi haz it!
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17:48.58stansmithlolkat?
17:49.08Kattystansmith: precisely.
17:49.13stansmithha
17:49.30Kattystansmith: this is why asterisk syntax sometimes does not parse.
17:49.33Kattystansmith: it does not speak kat.
17:49.59stansmithhaha wow
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17:51.23greekguy8888anyone in here?
17:51.23[TK]D-Fender*b00m*
17:51.23stansmithomg netsplit!
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17:51.25Wayhighi've got $43 to spend on voip gear.. what can I get?
17:51.25russellbthat was hot
17:51.25Wayhigh:P
17:51.41mvanbaak;)
17:51.46mvanbaakhowz you ?
17:51.52russellbquite good.
17:51.55mvanbaakgood
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17:52.00Katty[TK]D-Fender: "no boom today, boom tomorrow. there's always a boom tomorrow.... What? somebody's gotta have some damn perspective around here. Boom! Sooner or later. BOOM!"
17:52.03Katty[TK]D-Fender: name the quote.
17:52.15mvanbaakcan * do video conferencing ?
17:52.34stansmithi collect straws
17:52.42Kattycpm: you sir, must be russian.
17:52.47greekguy8888lawl
17:52.57*** join/#asterisk Mavvie (n=edwin@ppp121-44-57-203.lns10.syd7.internode.on.net)
17:53.04[TK]D-FenderKatty: B5 FTW
17:53.14Katty[TK]D-Fender: horay! my reference didn't go unnoticed!
17:53.21mvanbaakI have to setup this 'proof-of-concept' system with video conferencing
17:53.27mvanbaakI want to use * where possible
17:53.41stansmith~B5
17:53.46stansmith!starwars
17:53.46Kattyjbot: babylon 5?
17:53.52greekguy8888does anyone know if there is a workaround for transferring calls deliverd by queue via the sip transfer button on a phone? (currently only releases agent channel if u use asterisk native transfer)
17:53.53*** join/#asterisk SteveTotaro (n=root@pool-71-179-157-126.bltmmd.east.verizon.net)
17:53.53Kattyjbot: B5?
17:54.29Kattysomeone teach jbot that B5 and Babylon 5 = http://en.wikipedia.org/wiki/Babylon_5
17:55.22Wayhighjbot: B5 is also at http://en.wikipedia.org/wiki/Babylon_5
17:55.23jbotWayhigh: okay
17:55.41Wayhighjbot: Babylon 5 is also at http://en.wikipedia.org/wiki/Babylon_5
17:55.41jbotWayhigh: okay
17:55.43sudhir492Does someone here have a recommendation for an ATA other than Linksys?
17:55.47Wayhigh~wayhigh
17:55.48jbotAsterisk mouse WAZ in his 1U, eatinz his thermo ribbons.. HE R MOUSEKILLA
17:56.00greekguy8888the granstream ata's are only other option i think
17:56.29stansmithwow this chick is snoring so loud!
17:56.31tzafrirsipura :-)
17:56.32stansmithshould i wake her up?
17:56.45greekguy8888give her an elbow lol
17:56.48stansmithlol
17:56.50Wayhighsudhir: depends what you're looking for.. I'm having good luck with my x100p.com S100-FX
17:57.09Qwell~cheap
17:57.09jbotsomebody said cheap was a bad idea.  If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE.
17:57.11Wayhighstansmith: depends on if he's a gnaw your arm off kinda gal
17:57.37stansmithhaha...i kinda need both my arms today
17:57.47greekguy8888nice
17:58.37tzafrirWell, IAX ATAs are probably better targetted for traveling and such
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18:03.12stansmithlol, ecoterrorism?
18:04.39stansmithLOL are you even allowed to do that in IRC?
18:04.43x86oh noes now i'm teh ecoterrorismnist!
18:04.48stansmithLOL
18:04.58Qwellstansmith: I don't think there's a specific rule against it..
18:05.02x86stansmith: "allowed" and "IRC" should not be used in the same sentance ;)
18:05.11stansmithhaha
18:05.38generalhancan anyone tell me what this warning is all about ? app_voicemail.c:2262 inboxcount: Failed to obtain database object for 'asterisk'!  i just loaded a new 1.4 install with realtime and this is just constantly scrolling
18:07.18greekguy8888looks like your realtime vm is not connecting
18:07.39greekguy8888<trying this again>   does anyone know if there is a workaround for transferring calls deliverd by queue via the sip transfer button on a phone? (currently only releases agent channel if u use asterisk native transfer)
18:07.44*** join/#asterisk ice_croft (n=nolan@85.172.5.106)
18:07.48generalhangreekguy8888: any ideas as to why that might happen ?
18:07.57*** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk)
18:08.33greekguy8888u did not setup your realtime correctly, could be a number of thinngs... the fact it can't find an object says either its not connecting to the db, OR... you don't have a vm box setup for what is being accessed
18:09.11greekguy8888probably the first thing if its consantyl scrolling
18:09.34generalhani see. well lets see if i can figure out why it wouldnt be connecting
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18:14.24x86generalhan: execute this from CLI: "realtime mysql status"
18:14.27*** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net)
18:14.37x86replace mysql with whatever database driver you're using
18:14.54x86generalhan: for CDR, do this "cdr mysql status"
18:15.17generalhanx86: it says connected
18:15.29x86then your database is not setup properly
18:15.31generalhanbut you know what ... im not getting that warning anymore, and i think i know why
18:15.33x86did you load the schema?
18:15.41x86;)
18:15.53stansmithschemas ftw
18:16.07generalhani had copied over my configs to the new server ... so there was a sip.conf entry for phones that arent in the DB, so i think it was trying to load the VM info for those phones that wasnt setup
18:16.20generalhanso when i removed the sip.conf references the warning went away
18:17.00generalhanbut ... should i still be able to see the phones that are setup in the DB by using sip show peers in the CLI ?
18:17.07generalhancause it produces nothing
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18:19.22dexpdxAnyone run across a scenario where cdr_odbc connect's just fine bust doesn't store records in that datbase
18:19.44dexpdxwith cdr-csv working fine, isql connecting & inserting just fine
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18:20.25hmmhesaysodbc killing the connection after so many hours?
18:20.32dexpdxnope
18:20.47dexpdxstair off a fresh restart no inserts
18:20.55dexpdxs/stair/strait/
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18:24.18JayTee52What is the best or preferred voice recoginition add on for Asterisk for creating a Dial by name voice menu?
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18:25.44dexpdxHere is my CLI output: shows no errors: http://pastebin.ca/926420
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18:41.29generalhancan anyone tell me what ${DATETIME} was replaced by in 1.4 ??
18:41.50stansmithgeneralhan: http://www.voip-info.org/wiki-Asterisk+variables
18:41.57generalhanty
18:44.00ccvpwhat would be a good piece of software to deploy to a website, that would be community driven, where people could whistle blow about a particular topic, and submit images, and videos as evidence of something going on, that would have a web 2.0 feel to it?
18:44.43stansmithwhat is a wiki?
18:44.56ccvpfor $200 please
18:45.00ccvpThanks Alex Trebek
18:45.18Qwellccvp: youtube
18:45.31ccvpwell something i can integrate w/ adsense
18:45.32ccvpheh
18:45.38ccvppersonal gain, not no profit
18:45.51stansmithadsense can integrate with anything html, no?
18:45.52Qwellmonetary gain from a whistle blower site?
18:46.00Qwellyeeeeeeaaahhhhh....  good luck with that
18:46.02ccvpthe more people that come , the more CPM
18:46.15ccvpadsense is a nifty tool qwell , when used w/ adsense
18:46.24ccvppayments based on ad impression, and more payment when their actually clicked
18:46.30cpmthe more cpm what?
18:46.46ccvpimpression counts
18:46.52dexpdxanyone here familiar with cdr_odbc?
18:47.01dexpdxi.e. troubleshooting it
18:47.06ccvpgoogle adsense also has a calculation that pays pays based on how many people see an add, but its smaller
18:47.11ccvpcompared to if someone actually clicks an ad
18:47.13stansmithccvp: impressions get paid out by every thousand visitors though, no?
18:47.17ccvpyep
18:47.30ccvpwhen i tested a site of mine with TOR
18:47.37ccvpso my ip isnt linked to me, and clicked an add twice
18:47.42ccvpI got $2.80 cents in 4 minutes
18:47.47stansmiththats illegal
18:47.49ccvpbut I wont abuse, risk getting my adsense account terminated
18:48.09ccvp:)
18:48.19stansmith;)
18:48.37stansmith1.40 a click? i thought it was pennies a click
18:48.43ccvpsome ads pay more
18:48.50ccvpim not an expert on adsense
18:49.04ccvpmy MIS 595 class has 2 weeks on adsense next semester
18:49.07stansmithwhoa whoa whoa
18:49.30ccvpim surprised that hardly
18:49.33ccvpanyone knows about adsense
18:49.37ccvphow it works etc,,,,,
18:49.48ccvpi know us tech geeks despise ads, but theres a market , billions of people use the internet
18:49.51stansmithccvp: here check me out, im about to make u a lot of money
18:50.28stansmithyou set up an asterisk box, and then anytime someone calls you via voip, you use their IP to fetch the URL of a page that has an ad on it
18:50.31stansmithwammy
18:50.37stansmithlike stealing candy from a baby
18:50.47ccvpyou cant target google ads
18:50.51ccvpits hidden in javascript, heh
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18:51.38stansmithi was thinking more along the lines of the impressions
18:51.47ccvpfractional
18:51.55stansmithslow progress > none
18:52.00ccvpbetter bet would be to malware 10's of thousands of people
18:52.04ccvpand have drones do the click fraud heh
18:52.05stansmithlol
18:52.42stansmithwhat domain were u gonna put the adsense on?
18:52.55ccvpcannot say heh
18:52.57stansmithshorter, more basic domains = higher return
18:53.05stansmithvoipzilla.com?
18:53.07ccvpyep, but alot of them are already taken
18:53.08*** join/#asterisk vap0rtranz (n=jpittman@75.110.17.157)
18:53.19ccvpback in the day in like
18:53.22ccvp1994,1995, some guy made like
18:53.28ccvp280 million before patent squatting was law
18:53.35ccvpreserved 500+ website domains ofm ajor corporations
18:53.41ccvpbefore they had web presences
18:54.18stansmithha..i was interning at a company that was hired to redo sex.com..we had all the traffic info for that domain..made a million dollars every month or so just redirecting international traffic
18:54.32stansmithof course, the name itself goes for a couple million
18:55.06stansmithat least
18:56.38*** join/#asterisk ice_croft (n=nolan@85.172.5.106)
18:58.11vap0rtranzwhy does my last peer in users.conf match against all incoming calls? so dialing the next-to-last peer's DID fails with 'rejected ... extension not found.'
18:58.46stansmith~externalivr
18:59.17grandpapadotAnyone here use Packet 8?  What phones are they selling under their brand?  We have a customer that wants to convert and I was just doing some preliminary research on the phones.
19:04.58*** part/#asterisk jlb (n=jlb@75.148.162.90)
19:05.04dexpdxgrandpapadot: probably what ever they can get their hands on for the smallest overhead
19:05.15*** join/#asterisk JunK-Y (n=junky@modemcable153.55-201-24.mc.videotron.ca)
19:07.19*** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl)
19:10.35Tommy3zzz
19:11.51Tommy3any recommendations on where to go for local phone number over voip for my asterisk box. shopping on the web wore me out with poor results.
19:11.54J4k3packet8 stuff is ghettofied
19:12.09dexpdx<PROTECTED>
19:12.10dexpdxerr!
19:12.13J4k3they're about onpar with vonage....  amateur hour.
19:12.19*** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net)
19:12.23J4k3err, overpriced amateur hour
19:12.46Tommy3using skype now, but not sure I can use it with asterisk...
19:13.04*** part/#asterisk ice_croft (n=nolan@85.172.5.106)
19:13.32lmadsenyou can't
19:14.35J4k3Tommy3: there are lots of different offers out there, the issue is getting something that works for tou
19:14.38J4k3err you
19:14.55Kattymmmmmapple!
19:15.05J4k3gapple
19:15.10Kattyalso good
19:15.16Kattyand very juicy!
19:15.28Kattybit spensive tho :<
19:15.56BrianR___At least with number portability you can switch if things don't work out.
19:16.04Kattyi gave up soda tho
19:16.13BrianR___I've used both gafachi and vitelity for US48 termination...
19:16.15lmadsenI try not to drink pop anymore
19:16.17Kattymiss it :/
19:16.25Kattylmadsen: yeah. lotta stuff in it
19:16.27lmadsenmy problem is I get addicted to it :)
19:16.31alrsI was late this morning so I was unable to stop at the grocer and restock my Earl Grey cache
19:16.31Kattylmadsen: especially the diet version.
19:16.34lmadsenKatty: gives you a bit of a beer belly too :)
19:16.37BrianR___My feeling is that the gafachi calls are slightly higher quality. They also do t.38 fax
19:16.42Kattylmadsen: i imagine
19:16.49Kattylmadsen: it made me SO incredibly hungry
19:16.52Kattylmadsen: i could eat an elephant
19:16.55alrsBrianR___: I've had really good luck with Gafachi, you?
19:16.56lmadsenand gut rot
19:17.03Kattylmadsen: and that's Not Good when you can't go get a snack :<
19:17.04stansmithelephants are at the top of the food chain
19:17.08lmadsenindeed
19:17.17Katty)_=
19:17.22Kattyhates you
19:17.28lmadsenI made pasta and a veggie sauce for lunch, and that was only 2 hours ago, and I'm hungry again
19:17.38lmadsenyou hate me? that is sad...
19:17.40Kattywhat kind of pasta?
19:17.44lmadsenfuscilli
19:17.50J4k3Earl Grey?!?!  floor sweepings!
19:17.51Kattyis that made from flour or egg?
19:17.58lmadsenhrmmm.... good question
19:17.59*** join/#asterisk e` (n=e@38.102.196.202)
19:17.59stansmithlol
19:18.03Tommy3is there a good service that I should look at. I can only find those rediculous monthly rates so far. would like something around $35/year with unlimited calling and a local phone number. would rather not mess with per minute stuff (hate suprizes). Ok, got gapple, gafachi, vitelity. enough for me to go do research. thanks guys! and Katty too!
19:18.07Kattywhat kind of veggies did you eat with it?
19:18.10lmadsennow you have me curious... /me goes to the kitchen to check
19:18.12alrsJ4k3 I've got a bergamot problem
19:18.27Kattylmadsen: i'm going to wager it wasn't whole wheat pasta.
19:18.32Kattylmadsen: and you didn't eat enough veggies
19:18.40Kattylmadsen: and thus your being eaten from the inside OUT
19:18.40J4k3Tommy3: vitelity offers various plans, depending on rate center
19:18.56lmadsenKatty: onion, garlic, orange pepper, tomato, and some hot peppers with butter, chili power, lemon and black pepper, and red chili flakes
19:19.01J4k3ie - where I live I can get a $7.95/month unlimited incoming line that offers multiple call paths (I've pushed 3 calls over it, so far)
19:19.08J4k3err 3 simultanious calls
19:19.08Kattylmadsen: that sounds lovely.
19:19.15Kattylmadsen: but how much did you use per serving total of veggies? 1c?
19:19.17*** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com)
19:19.21lmadsenKatty: ya... I have whole wheat pasta in the cupboard, but wanted to try the fuscilli
19:19.28J4k3but, you go 25 miles east of that rate center, where my girlfriend lives, and the best they offer is $1.50/mo + $1.49/minute
19:19.37Kattylmadsen: it sounds exotic!
19:19.45lmadsenKatty: oh.. I usage a whole tomato, a whole orange pepper, whole hot pepper, whole small onion, etc....
19:19.51ZPerteewhat is the cheapest way to get a SIP DID?  don't care what the number is as long as it is in the USA
19:19.54BrianR___alrs: No major complaints.
19:19.56Kattylmadsen: ahh..
19:19.59lmadsenKatty: so I got like... a good chunk of veggies
19:20.03Kattymad.
19:20.09BrianR___alrs: COmpared to vitelity, I think whatever Gafachi is using for echo can does a better job.
19:20.15Kattylmadsen: stop running up and down stairs!
19:20.17lmadsenI like to do half veggies, half pasta
19:20.24Kattylmadsen: that's about what i do
19:20.27lmadsenKatty: isn't that like... good for you? :)
19:20.32[TK]D-FenderZPertee: www.ipkall.com <- free
19:20.35lmadsenplus I don't have any stairs in my condo
19:20.49BrianR___alrs: Also, there's some problems with REINVITE on Vitelity and they don't seem interested in helping me. When I transfer a call on Vitelity, I get a trombone shaped audio path.
19:20.51Kattylmadsen: 6oz grains, 5oz protein, 3 C Dairy, 1.5 C Fruit, 2.5-3 C veggies is what i eat a day
19:21.09BrianR___alrs: But when I transfer on Gafachi, my asterisk winds up out of the audio path.
19:21.12lmadsenKatty: ya... I need a better meal plan. I do pretty good, but I don't have a set plan
19:21.12J4k3gafachi seems kinda outrageously overpriced til you hit 10k/month for termination
19:21.20Kattylmadsen: 1C rice = 2 oz grains
19:21.30Kattylmadsen: go to mypyramid.gov
19:21.30BrianR___J4k3: I agree that their termination prices are high..
19:21.33Kattylmadsen: they have a handy generator
19:21.36lmadsenKatty: oh nice!
19:21.38lmadsenI'll check it out for sure
19:21.39Kattylmadsen: pretty accurate too
19:21.46lmadsenI've been meaning to put together a good diet plan
19:21.48lnxwhen i use SIP/0001*001 channel why my endpoint is ringing? :) And when exten => s,1,Dial(SIP/ipphone) executed? Where can i find base documentation about channels etc... ?
19:21.48BrianR___Katty: I though that site was for social security...
19:21.56lmadsenKatty: well... not diet -- healthy eating plan
19:22.01Kattylmadsen: indeed.
19:22.09Kattylmadsen: we don't use the four letter D word in my house
19:22.21ZPertee[TK]D-Fender: thanks
19:22.41BrianR___I currently use gafachi for my fax stuff, vitelity for my home voip (because they have e911)
19:22.44Kattylmadsen: i blogged it too, if you wanan see what i did: http://angela.sleekgeek.org/category/health/mypyramidgov/
19:22.49J4k3diet is overrated, everyone needs more sexercise.
19:22.53lnxi must learn SIP basics
19:23.04stansmith~sexercise
19:23.16[TK]D-Fenderlnx: You need to learn * basics... go read.. THE BOOK
19:23.16stansmithjbot like 0/4 today
19:23.17jbotACTION smooches 0/4 today on the lips
19:23.18[TK]D-Fender~book
19:23.19jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
19:23.22lmadseni use 0% milk, eat cereal like multi-grain cheerios and Just Right, eggs and veggies, whole grain (most of the time) pasta and veggies, long grain and wild rice, etc...
19:23.36lmadsenthen I tend to snack on trail mix and such
19:23.45stansmithtrail mix is the whip
19:23.48Kattylmadsen: beginning http://angela.sleekgeek.org/category/health/mypyramidgov/page/9/
19:23.51stansmithgood, and good for u!
19:23.55J4k3trail mix = simple carbos deluxe.
19:23.56[TK]D-Fenderlmadsen: Nuke the grain overload....
19:24.01J4k3straight to your ass.
19:24.08[TK]D-Fenderlmadsen: and indeed trail-mix = EVIL
19:24.13lmadsenya but it's tasty
19:24.17stansmithgood, and good for u!
19:24.18lmadsenI'm not trying to be a stick here :)
19:24.19Kattydepends on the trailmix
19:24.28Kattydried peas have a nice crunch
19:24.29lmadsenstuff I have is mostly cranberries
19:24.41Kattyi tend to snack on fruit
19:24.50Kattya ginormous apple takes a good 10 minutes to eat
19:24.51lnxhttp://www.asteriskdocs.org/  how nice
19:24.59[TK]D-FenderKatty: Pretty much all of them are a caloric overload and the serving size accounts for too much of your dietary intake.
19:25.10lnx~book
19:25.11jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
19:25.14Katty[TK]D-Fender: baroo?
19:25.23Katty[TK]D-Fender: apple is not a caloric overload
19:25.28[TK]D-FenderKatty: talking about trail-mix here..
19:25.32Katty[TK]D-Fender: oh, right.
19:25.38Kattyyes. trailmix can be bad.
19:25.41*** join/#asterisk BCS-Satori (n=somewher@75.148.21.113)
19:25.43Kattybut! can also be properly accomodated for
19:26.03Kattyyou could eat a little less rice and pasta throughout the day to accomodate for some trailmix snackings
19:26.14Kattyor... mayhaps, make it a weekly treat. that won't screw anything up.
19:26.23BCS-SatoriI am looking for an 8 port FXO gateway to use with asterisk.  Besides the grandstream is there another that is better or is the grandstream the way to go
19:26.39J4k3heres a question:  why hasn't somebody come up with an asterisk gui that isn't a trainwreck (freepbx)
19:26.48J4k3seems like there'd be millions of dollars for that.
19:26.49lmadsenI don't think anyone here is ever going to say grandstream is the way to go
19:26.57lmadsenJ4k3: because it's hard
19:27.06J4k3grandstream is the way to go.... if you've got $60 to spend and need two handsets
19:27.06[TK]D-FenderBCS-Satori: Sangoma A200d.
19:27.15KattyBCS-Satori: Sangoma++
19:27.34J4k3otherwise I'd avoid buying grandstream's stuff
19:27.38J4k3gs's ATAs made me cry
19:27.41BCS-SatoriI personally never used grandstream i heard they were cheap
19:27.43J4k3they got returned
19:28.04[TK]D-Fender~gs
19:28.05jbotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
19:28.05*** part/#asterisk pkunkra (n=chris@cpe-74-73-10-32.nyc.res.rr.com)
19:28.22SteveTotarogs ATAs are decemt
19:28.24J4k3oh lord, somebody woke up ol duhfender.
19:28.25SteveTotarodecent
19:28.29J4k3;)
19:28.48J4k3SteveTotaro: they're ok.  I didn't wanna get stuck having to support them :)
19:29.06SteveTotaroi sold them when i had a webstore
19:29.13BCS-Satori[TK]D-Fender: the Sangoma A200d seems very nice, is there anything thats an external product sorta like SPA400
19:29.25SteveTotaromany bricked ATAs during firmware
19:29.30[TK]D-FenderBCS-Satori: Only decent ones are quite pricy.
19:29.40[TK]D-FenderBCS-Satori: AudioCodes MP series
19:29.57CanWoodGreat... we just bought 4 Grandstream phones and set up and Asterisk box yesterday with the intentions of rolling out a 12 user site
19:30.02SteveTotarobut in my use, the ATAs have been good for things like analog polycom conference phones
19:30.05[TK]D-FenderBCS-Satori: Thats the thing... FXS is quite aggressively priced, FXO not so.
19:30.10*** join/#asterisk cesar_CR (n=cesar@200.91.75.45)
19:30.11J4k3CanWood: which model phones?
19:30.18CanWoodGXP2000
19:30.22CanWoodyep
19:30.25SteveTotarothey seem ok
19:30.34J4k3gxp2000's don't seem too bad
19:30.35CanWoodand two TDM808B cards
19:30.48BCS-Satori[TK]D-Fender: Thanks for the information
19:30.50[TK]D-FenderCanWood: 16 FXO?
19:30.51SteveTotaroi just took on a support role for a company that has gxp2000
19:30.57CanWood2x8
19:31.00SteveTotarothey like them
19:31.04*** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net)
19:31.09CanWoodgood to hear
19:31.11J4k3I think I'm going to take the plunge and get a nxtvox nxa800p and a couple modules
19:31.16[TK]D-FenderCanWood: Fractional PRI is not an viable option where you are?
19:31.18J4k3I need an FXO and an FXS would be fun to play with
19:31.39*** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net)
19:31.42SteveTotaronxtvox is the one in china right?
19:31.47CanWoodOur current system is analog in and the transtion will be smoother if we change one piece only
19:31.47J4k3yeah
19:31.49SteveTotarothey mail from chine
19:31.52SteveTotarochina
19:31.54J4k3so does half of ebay
19:31.58J4k3so its ok.
19:32.10Kattyi would like to hire an assassin.
19:32.20Kattydo we have any ninjas?
19:32.24SteveTotaroi took the plunge from a different place
19:32.36Katty:<
19:33.24stansmithecoterrorism?
19:34.03SteveTotarowww.getvoicecards.com
19:34.20SteveTotaroebay worked out a few bucks cheaper
19:34.37Kattystansmith: that sounds like some sort of prehistoric dino.
19:34.52*** join/#asterisk Servergod (n=servergo@70.97.159.120)
19:35.03Kattystansmith: ecoterroroserious.
19:35.06ServergodHi all!
19:35.17Kattysurious.
19:35.25Kattysomething like that.
19:35.36x86I'm using MySQL for CDR, and I'm wondering what (if anything) uses the 'userfield' field, or if that's something I can put whate ever arbitrary value I want in?
19:35.39budolhi Katty
19:35.43Servergodphones are displaying caller id number twice and no name. Any ideas?
19:35.53Kattybudol: allo.
19:35.55SteveTotaro<PROTECTED>
19:35.55SteveTotaro<PROTECTED>
19:36.05Servergodie 702-xxx-xxx as number and 702xxxxxx as name
19:36.06budolKatty : how are you
19:36.11*** join/#asterisk phillipk (n=phillipk@fw.datafax.net)
19:36.14Kattybudol: digesting apple.
19:36.23Kattybudol: that's all i'm doing today.
19:36.42x86any ideas?
19:36.44Kattybudol: and giving stansmith a lot of crap.
19:36.52Kattybudol: no relation to apple digestion.
19:36.53stansmith:-(
19:36.58stansmithew ha
19:37.10SteveTotarostansmith sounds like a fedora guy
19:37.15SteveTotaro~fedora
19:37.16jbotrumour has it, fedora is stevetotaro is <action> tells stevetotaro that becoming RHCP certified, is The One True Way to Supreme Linux Knowledge
19:37.21stansmithwrong!
19:37.24Kattywhenever someone says fedora, i think about eudora
19:37.25stansmithArch
19:37.27Kattythat old email client
19:37.40Kattyi think it was eudora anyway ^_-
19:37.46J4k3when people say 'fedora' I think of a big hat.
19:37.47Kattydebian++
19:38.07Kattywhat was mandrake an offshoot of?
19:38.07SteveTotaroi think of a gangster with a tommy gun
19:38.07BCS-SatoriSteveTotaro: any experience with the OpenVox?
19:38.08stansmithLPCI > RHCP?
19:38.11stansmithdebian i think
19:38.23x86stansmith: LPIC*
19:38.32SteveTotaronope, but the card is at my office, installing it tomorrow
19:38.33J4k3well, linux is more like a whole family-sized bag of cheap wet noodles.
19:38.36x86stansmith: Linux Professional Institute Certification
19:38.38stansmithoops, thanks x86
19:38.41Kattyubuntu is kinda nice
19:38.46x86stansmith: there are three of them, I have two ;)
19:38.50Kattyall i remember about mandrake was all the Easy Wizards that Didn't Work
19:39.05J4k3manrape
19:39.07stansmithi met one of the authors that published the oreilly book at linuxfest2007...great book
19:39.09x86stansmith: two LPIC certs, CompTIA Linux+ cert, and a Brainbench Linux cert
19:39.33stansmithi want to get the level 1 cert, i think i can do it with studying very little...just havent gotten around to it
19:39.34SteveTotarois linux + like A +?
19:39.36x86I avoid distro-specific certs like the plague
19:39.37Kattywhere is Nugget when you need him! guess i'll fill in for him.
19:39.38*** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net)
19:39.41Katty<Nugget> Linux is poo.
19:39.44x86SteveTotaro: kinda
19:39.44SteveTotaroor network +?
19:39.54x86SteveTotaro: it's linux+ :)
19:40.02x86SteveTotaro: similar test setup
19:40.10Kattyi did A+ for awhile
19:40.16J4k3linux kernel is poo, the 23434534535 craptastic wheel-reinvention distros that run on top of it all suck furiously in their own little way
19:40.21budolKatty : lol
19:40.21Kattyi decided it was redonkulus and i had better things to do :/
19:40.22SteveTotaroVxWorks rocks!!
19:40.32Kattylike chase the ferrets around the house
19:40.32stansmithso pretty much we all agree Arch is the best distro? great!
19:40.43J4k3Microsoft, I'm sure, loves linux, since it just clogs up a few hundred thousand clued people into this constant wheel reinvention routine
19:41.08J4k3without linux, microsoft would probably actually some day get competition
19:41.22Kattyi must agree completely.
19:41.27SteveTotaronah
19:41.28J4k3but for now, everyones gonna waste time on garbage like... the uber-bloated linux kernel and the total waste of time xwindows is.
19:41.40stansmithJ4k3 hates freedom...must be a terrorist..get him!
19:41.56J4k3stansmith: eh?  BSD license is a lot more free than the GPL
19:42.00J4k3so don't do into that 'freedom' bullshit
19:42.01stansmith:-(
19:42.14Servergodcan anyone help with caller id display?
19:42.18SteveTotaro64 degrees out, crap, i gotta fire up my Kawasaki!!!
19:42.20Kattyhow fitting that freedom and bullshit be in the same sentance.
19:42.32J4k3I can put a product on the market running a bsd 4.4-lite based OS and not releases sources and *gasp* not get my pants sued off.
19:42.38KattySteveTotaro: eep.
19:42.42tzafrirServergod, everyone is busy with more important flamefests
19:42.48*** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net)
19:42.51J4k3gpl-violations = why I'd NEVER EVER release a linux-running commercial product.
19:42.57tzafrirServergod, or waiting for you to provide details
19:43.02stansmithlol
19:43.15tzafrirJ4k3, ever heard of BSD violations?
19:43.28Servergodphones are displaying caller id number twice and no name
19:43.28Tommy3is cisco 7940 a good phone for sip conversion? work with asterisk???
19:43.31Servergodie 702-xxx-xxx as number and 702xxxxxx as name
19:43.33J4k3tzafrir: I'm sure somebody out there has violated the BSD license, but it'd be quite a bit harder.
19:43.38J4k3ie - removing the copyright line
19:43.48tzafrirJ4k3, AT&T lost a trial for BSD license violations
19:43.50Servergodi can see the name comming in properly from the provider
19:43.56J4k3tzafrir: hence why I said bsd 4.4-lite
19:43.59SteveTotaroi have heard of S&M violations
19:44.00*** join/#asterisk bkw___ (n=brian@adsl-70-234-182-211.dsl.tul2ok.sbcglobal.net)
19:44.04Kattybkw___: YOU
19:44.13stansmithJ4k3: associating gpl with gnu/linux kind of a fallacy, no?
19:44.14[TK]D-FenderTommy3: Where are you located?
19:44.16Kattybkw___: never say hi anymore :<
19:44.28tzafrirMost of the current GPL violations are very similar in nature: the distributors never bother to do the basic thing
19:44.36*** part/#asterisk gerhard7 (n=gerhard@82-169-26-19.ip.telfort.nl)
19:44.36Kattywhat kind of license thing does gentoo have?
19:44.38Tommy3huntsville
19:44.44SteveTotaroservergod, go to www.voip-info.org
19:44.55SteveTotaroand search for set caller id
19:44.56J4k3yes, because complying with the gpl leads to jerkoff kids creating warranty problems - see the wrt54g and why Linksys has done everything possible to avoid linux now.
19:45.36[TK]D-FenderTommy3: Then forget Cisco and go Polycom.
19:45.37SteveTotarothe wrt54g running linux sold more of those units because of those jerkoff kids
19:45.43*** join/#asterisk ManxPower (n=manxpowe@209.16.72.139)
19:45.43J4k3I could, for example, base a product off an x86-running system running FreeBSD, carefully avoid GPL code, and never have to release a damned bit of source.
19:46.01J4k3SteveTotaro: a few... but warranty hassles cost a lot more than the profit they got off the unit.
19:46.26tzafrirJ4k3, and not use Asterisk, gcc, bash
19:46.31J4k3so, in reality, those nerdy kids that returned hardware got in cisco's pocket, deep.
19:46.35SteveTotarowarranty is void if you load unofficial firmware
19:46.54*** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net)
19:46.56J4k3tzafrir: gcc is only required for compiling...  compilers are for developers :)
19:46.58stansmithwarranty is voided if u open the case...check the sticker underneath where the blue meets the black
19:47.06*** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net)
19:47.22Tommy3Why? problems with cisco phones (cheap on ebay) all I know about them....
19:47.24J4k3SteveTotaro: no, its not, at least in the USA.
19:47.34SteveTotaroyes it is
19:47.52J4k3SteveTotaro: theres a federal act saying you can load any oil you want into your car, and any firmware you want onto your linkydink... its just a matter of if said oil or said firmware *breaks* the warrantied product is when the warranty is over.
19:48.03tzafrirJ4k3, your loss. You avoid the pool of useful programs
19:48.08[TK]D-FenderTommy3: Cisco's SIP implementation sucks and maintenance is a PITA
19:48.20J4k3and its up to the manufacturer to prove your oil or your firmware broke the unit.
19:48.28*** join/#asterisk pat2man (n=pat2man@ip67-90-247-203.z247-90-67.customer.algx.net)
19:48.34J4k3in cisco's case that'd just add more costs, so they had to eat it
19:48.43J4k3while they poked R&D to move to make the unit less linux-friendly.
19:48.44SteveTotaroAccording to Linksys, flashing firmware from sources other than Linksys does void the WRT54G's warranty.
19:49.07J4k3SteveTotaro: yeah, and your dealership would like to lead you to believe changing your own oil will do the same
19:49.14J4k3when in reality its quite the opposite.
19:49.16stansmithbut so what? if u flash the firmware, why should linksys be responsible?
19:49.24J4k3should they not be responsible?
19:49.26SteveTotarothey aren't
19:49.27stansmithno they shouldnt
19:49.29J4k3if the hardware breaks, its their problem
19:49.31stansmithyea
19:49.33J4k3if you break the unit with software, its your problem
19:49.43stansmithif i drive my car off a cliff its not hyundai's fault
19:49.51SteveTotaroif you break your hardware, it is your problem
19:49.59stansmithgreat, so we all came to a conclusion
19:49.59J4k3yep, and its not hyundai's fault if you buy pennzoil thats actually maple syrup
19:50.20stansmithbut that begs the question..was it ever penzoil to begin with?
19:50.21J4k3but, alas, if you pour out their quaker state and add in mobil1 synthetic, you're not violating the warranty.
19:50.27Tommy3Fender: OK, convinced... I was thinking about downloading the firmware and setting up a server to convert it, but you are right, it does take quite a bit of work according to what I have read.
19:50.43SteveTotaroj4k3, your argument is /dev/null
19:50.47stansmithLOL
19:50.50J4k3SteveTotaro: no, its not.
19:50.55stansmithSteveTotaro: +2 xp
19:51.10J4k3SteveTotaro: and its gone to court before...  you've got desktop pc manufacturers not wanting to back hardware failures on PCs that people loaded linux on... its gone to court and the manufacturer lost.
19:51.32SteveTotarototally different situation
19:51.33*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
19:51.45[TK]D-FenderTommy3: What kind of prices were you finding?
19:51.53J4k3not a different situation at al
19:51.55J4k3er all
19:52.00SteveTotaroif i put maple syrup in my car does it void the warranty?
19:52.00J4k3a PC and your router are the same idea
19:52.05J4k3its a processor, some ram, some storage...
19:52.16tzafrirCan that  WRT54G run Linux anyway? isn't ther a separate L model for that?
19:52.26lmadsenyes there is
19:52.26stansmithtzafrir: the older models can
19:52.27J4k3yes, but only on parts directly affected by the maple syrup
19:52.31lmadsenor pre v.5
19:52.38SteveTotaroi have several running busybox
19:52.39J4k3they couldn't say, not cover the rear door hinge cuz you put maple syrup in the engine.
19:52.42SteveTotaroeven asterisk
19:52.43stansmithafter that, linksys started giving the newer models less vram
19:52.43Tommy3Fender: $60
19:52.46SteveTotaroopenvpn
19:53.05J4k3tzafrir: they'll all run linux, its just a matter of having any ram/flash left over :)
19:53.21lmadsenI have a WRT54GL running DD-WRT and have openvpn connected so all my phones and such can access the VPN
19:53.28Tommy3Fender: (probably heavily used or stolen :)
19:54.03[TK]D-FenderTommy3: Well..... hate to say it, but yeah, $60 is extremely ahrd to pass up and even I would probably go for it if I'm not concerned with warranty, etc...
19:54.16J4k3linksys's big mistake on their product was crappy CFE, so when you did mis-flash the unit fell on its ass
19:54.30J4k3toshiba, asus, buffalo... they didn't have that problem cuz their cfe wasn't shit
19:54.32SteveTotarowrong again j4k3, some run VxWorks
19:54.35[TK]D-FenderTommy3: But if we're talking new for business use, then I'd just say Polycom.
19:54.40SteveTotarohttp://www.linuxdevices.com/news/NS4729641740.html
19:54.50J4k3SteveTotaro: yes, and vxworks can be removed and linux put in their place
19:55.14*** join/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net)
19:55.23J4k3SteveTotaro: don't even need to jtag the unit... 100% network based bootloader replacement.
19:55.37Tommy3Fender: Understand the business use.... I just wanted something to experiment with (cisco). I have NEC voip phones an service at my office. have not researched them for sip yet.
19:55.48J4k3dd-wrt has instructions for the changeover, I think openwrt does too
19:55.51ThatKidKeli just noticed that my cdr-csv folder is filling up with a bunch of different files, each named after a particular accounting code
19:55.57J4k3but you're stuck with the limitations of 8mb ram and 2mb flash
19:55.59ThatKidKelhow can i stop that, and only log to Master.csv
19:56.11SteveTotaroyes but you said they all ran linux
19:56.20J4k3dd-wrt does an amazing lot in 2/8, openwrt is a bit less tight.
19:56.25SteveTotarocan you ever admit you are wrong?
19:56.29x86tzafrir: you can run linux on the WRT54G, the only difference with the WRT54GL model is more flash
19:56.29J4k3SteveTotaro: they all *can* run linux.
19:56.39J4k3and no, I never said they all ran linux out of the box... who cares if they do?
19:56.45*** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
19:56.53stansmithu guys made the author leave
19:57.05x86SteveTotaro: yeah, it's true... all WRT54G's can run Linux
19:57.06SteveTotaro(02:51:25 PM) J4k3: tzafrir: they'll all run linux, its just a matter of having any ram/flash left over :)
19:57.19x86SteveTotaro: missed that part :)
19:57.21J4k3SteveTotaro: you took the line out of context.
19:57.23SteveTotaroi know they can
19:57.28stelioskJ4K3 : Actually its not the CFE that gets warped, but the area where the variables are stored
19:57.28SteveTotaroi guess i did
19:57.43*** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br)
19:57.45J4k3steliosk: yeah, and most manufacturers had the ability to clear/ignore that enough to tftpboot again
19:57.51SteveTotarobut if you read the article, they have a linux version for linux people with more ram
19:58.09SteveTotaroso your argument is/was /dev/null
19:58.10J4k3if you can tftpboot, you can make a distro that runs 'mtd erase nvram' and get that whole problem fixed
19:58.21x86SteveTotaro: more flash, iirc
19:58.21J4k3now, asus's broke when you cleared nvram, but otherwise it was fairly bulletproof.
19:58.28J4k3err no, toshiba's die when you clear nvram
19:58.30J4k3asus's recover
19:58.40J4k3linksys's recovers, but usually if the nvram is corrupt it won't ever tftpboot.
19:58.49J4k3and tftpboot is disabled by default on the linksys, which is ghey
19:58.55J4k3saves about 2 seconds of boot time, though.
19:59.02SteveTotarothat is why you enable it first
19:59.12J4k3real hardware has it enabled already
19:59.21J4k3hence the difference between linksys and the rest of the wrt-alikes.
19:59.37J4k3(linksys sells the worst of the worst for the highest price, sans buffalo)
19:59.43SteveTotaroreal hackers enable stuff that can help them from bricking stuff before the hack
20:00.02J4k3of course, but some did it anyways since it *can help legitimate end users too*
20:00.17J4k3real hackers with lots of time on their hands
20:00.25J4k38mb over jtag = slow operation
20:00.29J4k3or even 2mb
20:00.44stelioskwell not the one you get for 2$ woth of parts
20:00.59J4k3its still slower than just upgrading the sumbitch via network
20:02.11stansmithSteveTotaro: why did you think i was a fedora guy?
20:02.27SteveTotaro~fedora
20:02.28jboti guess fedora is stevetotaro is <action> tells stevetotaro that becoming RHCP certified, is The One True Way to Supreme Linux Knowledge
20:02.33SteveTotarocan't you read?
20:02.57SteveTotaroi actually am a fedora guy
20:03.08SteveTotarofc9 beta
20:03.22SteveTotaroalthough i guess it is all beta
20:03.39stansmitharch is rolling release
20:03.41dexpdxis cdr_odbc supposed to connect and disconnect with every record
20:04.38SteveTotaroi don't much care what distro so long as i don't wind up in dependency hell
20:05.09*** join/#asterisk lesouvage (n=lesouvag@62.140.137.125)
20:05.28SteveTotaroallright, if I don't get some motorcycle time in today, I will regret it, later everyone
20:05.38stansmith~bye SteveTotaro
20:05.39jbotno
20:05.42stansmithwtf
20:08.47*** join/#asterisk Greek-Boy (n=email@41.221.58.4)
20:11.04*** join/#asterisk Idle_ (n=brian@S010600a024969312.ed.shawcable.net)
20:11.23Idle_is there a way to create distinctive rings on asterisk with my wildcard?
20:13.30stansmithIdle: what kind of rings?
20:15.28*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
20:17.12*** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler)
20:17.12*** mode/#asterisk [+o angler] by ChanServ
20:17.28b11dok.. I have this problem. I have a PRI going into a Sangoma A104d card..  when I enable the hardware EC, FAXing stops working, but echo is gone. When I disable hardware EC, FAXing works, but I get echo on voice calls.  Any advice?
20:19.04ManxPowerb11d: The EC should AUTOMATICALLY disable EC when it detects a fax tone because EC causes faxes to not work.  This is the way every single EC on the planet should work this way.
20:19.31b11dit doesnt though..
20:19.38b11dI have "faxdetect = both" on the PRI channels
20:20.46b11di have "echocancel = yes" "echocancelwhenbridged = no" and "echotraining = no"
20:20.48b11don those channels as well
20:21.19b11dshould i pastebin my zapata?
20:21.59*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
20:24.49tzafrirb11d, the echo cancelling code in zaptel has an independent fax tone detection that should disable the EC
20:24.56*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)
20:25.12tzafrirRegardless of the asterisk chan_zap faxdetect
20:25.58kraypius<PROTECTED>
20:26.10kraypiusand i get a busy signal
20:26.11tzafrirb11d, but if the channel is dedicated for faxing, just disable EC
20:26.46ManxPowerkraypius: I'm sorry, but we do not support Amp, FreePBX, Asterisk-GUI, or other GUIs here.
20:27.02b11dwell it passes a fax down the next available channel
20:27.09b11dso I never know what one will be fax and what one will be voice
20:27.10b11dits random
20:27.58ManxPowerb11d: you need to contact Sangoma to find out why their EC is not being autpmatically disabled
20:28.27b11dzap show channel shows it being disabled though..
20:28.32b11dyet it doesnt
20:28.38b11dso maybe it actually isnt being shut off
20:29.25b11dhttp://www.pastebin.ca/926725
20:29.37b11dthats a sample of the zapata.conf -- make sure im not on crack and missing something stupid please :)
20:29.50tzafrirb11d, what is the complete line you see there? How many taps?
20:30.06b11d128 taps or something similar..
20:30.08b11done sec
20:30.44b11dEcho Cancellation: 128 Taps unless TDM Bridged, currently OFF
20:30.51tzafrirSo there is currently no active EC, because there is no call. But the channel is configured to use an EC
20:31.09b11done shows "ON" right now
20:31.45b11dso I should be enabling hardware EC, but setting 'echocancel = no' in zapata?
20:32.56b11dall I want is EC for voice and no EC for fax ;)
20:33.20dexpdxanyone know the best way to trace a cdr_odbc error
20:33.34[TK]D-Fenderb11d: "echocancel=yes" and it should deactivate automatically
20:33.53b11dwell.. I have echocancel = yes now..
20:34.00b11dand when I enable hardware EC, faxing stops!
20:34.06vap0rtranzwhy are bloody inbound cids being matched by the last peer entry?!
20:34.27ManxPowerb11d: you ALREADY have your answer.
20:34.35vap0rtranzeh?
20:34.42b11di guess im misreading..  im stressed out today :)
20:34.56dexpdxvap0rtranz: dialplan fallthrough?
20:35.11vap0rtranzdexpdx: really?  i'll disable and test ...
20:35.37dexpdxvap0rtranz: IAX peers?
20:35.41b11dI need to contact Sangoma then you figure?
20:35.53vap0rtranzdexpdx: all sip; sorry didn't clarify that
20:36.52dexpdxvap0rtranz: exactly what is going wrong
20:37.26vap0rtranzseems to be a dialplan thing.  the global context is incoming, but the only inbound cid that doesn't get rejected is whatever trunk (peer) is listed last in users.conf
20:37.32*** join/#asterisk MmixX (i=mmixx@202.124.138.69)
20:38.46dexpdxwhat is the source of the inbound calls?
20:39.28vap0rtranzdexpdx: you mean has is registered upstream correctly??
20:40.14dexpdxvap0rtranz: let me get this right: you have a bunch of sip peers, correct?
20:41.19*** join/#asterisk zobia (n=laurashr@222.212.67.156)
20:41.26zobiaHello everyone
20:41.45*** join/#asterisk [hC] (n=hardcore@vpn.voxter.com)
20:42.02vap0rtranzdexpdx: a dozen or so in users.conf.  i can always get which one is last to answer the incoming did correctly.  all of them will do this if they're last in the line.  i'm matching the incoming cid in the dialplan b/c we have to do different things to different incoming calls.  it really seems like a dialplan problem but i can get each line to work if they're last in the users.conf ... :(
20:42.06zobiai want to change the voicemail's filename. anyone knows how can i change this. or make a copy when generate the voicemail file
20:42.40zobiai have read the app_voicemail.c but don't know where to change. and i want to use the callerid as the filename.
20:42.49dexpdxare the incoming calls are coming over what kind of trunk: sip, iax, zap?
20:43.04dexpdxis that trunk's default context different?
20:43.15vap0rtranzdexpdx: all sip, that's what i meant before
20:43.28dexpdxwhat is the type of error that the other sip peers give when issueing a dial?
20:43.31vap0rtranzdexpdx: yes they are!
20:44.06vap0rtranzdexpdx: i was just about to cram the trunks into one context ... *crosses fingers*
20:44.42dexpdxwhat is the failure message when you try and send a call to one of the SIP peers
20:44.51docelmoAnyone in here in the UK?
20:45.03vap0rtranzdexpdx: no sending.  this is all incoming
20:45.42*** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net)
20:46.04dexpdxvap0rtranz: when you have one call coming in from one sip peers and you want to route it to another you generally issue a Dial(SIP/${INSERT_SIP_EXTEN_HERE}/${EXTEN})
20:47.04vap0rtranzdexpdx: i'm confused.  this is for inbound processing by asterisk from a upstream registrar.  dialing out works.
20:47.13mvanbaakdoes anyone know if * can do videoconferencing ?
20:47.38dexpdxvap0rtranz: paste your extensions.conf to pastebin and give the link
20:49.37*** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose)
20:50.20*** join/#asterisk RoyK (n=roy@ip-122-26-149-91.dialup.ice.no)
20:51.27vap0rtranzdexpdx: we've got alot of internal numbers in there.  not sure how the boss would feel.  anyways, it really sounds like your troubleshooting the trunk dialing outbound which is not the problem here.  it's incoming call processing.  i have _[DID] matching each incoming # in extensions.conf under an [incoming] context which is the default; asterisk should just match that and continue processing (to a menu)
20:52.05*** join/#asterisk annielou (n=anne@c-76-119-139-36.hsd1.ma.comcast.net)
20:52.44*** part/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net)
20:52.54stansmithi love it when a plan comes together!
20:53.32*** join/#asterisk egecko (n=sam@cpe-76-176-238-225.san.res.rr.com)
20:53.50dexpdxvap0rtranz: I don't know how you end indent on a sending a call from one peer to another with out having a Dial cmd
20:53.59draygonis there a good doc on installing asterisks as a non-root user on CentOS?
20:54.29dexpdxdraygon: I don't think you will have any problems as long as you don't bind to any ports lower than 1024
20:54.42dexpdxor whatever the system port cutt off
20:54.49stansmithdraygon: the book goes through the installation using centos
20:54.50stansmith~book
20:54.51jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
20:55.36vap0rtranzdexpdx: ah, i think i see where you're going.  i'm not at the provider end right now.  i'm at the far end where the cid of the did (or whatever) has already been sent downstream.  i need asterisk to correctly process the cid that has already been sent
20:58.01dexpdxvap0rtranz: your not making sense
20:58.17vap0rtranzdexpdx: i know nothing
20:58.26dexpdxvap0rtranz: then go read
20:58.30dexpdxvoip-info.org
20:58.35vap0rtranzok, so where does asterisk put an incoming call?
20:58.43dexpdxwhere ever your tell it
20:58.54vap0rtranzthe context in sip.conf is default, right?
20:59.03dexpdxextensions.conf is where you configure your dialplan contexts
20:59.33dexpdxvap0rtranz: should be 'default' but I always use a custom context for inbound traffic per "peer"
20:59.56dexpdxexten => 5552221212,1,Dial(SIP/1212/${EXTEN});
21:00.02vap0rtranzdexpdx: good, that's how this was setup; for each peer, there was its own context
21:00.09vap0rtranzdexpdx: precisely
21:00.11dexpdxwould be an example of how to route calls coming in on 5552221212 to exten 1212
21:00.29vap0rtranzdexpdx: only diff is i'm Goto (something else)
21:00.54dexpdxso you issue a Goto to jump to another context?
21:01.11vap0rtranzdexpdx: the problem is asterisk doesn't get as far as 5552221212, even with _5552221212; it's always "extension rejected"
21:01.29*** join/#asterisk iamthelostboy (n=nathan@125-236-212-46.adsl.xtra.co.nz)
21:01.45dexpdxwell maybe you are not matching the inbound number correctly
21:02.35vap0rtranzdexpdx: true, what i had thought.  but the error message is explicit.  the number i dailed for inbound is incorrectly seen as coming from the peer that's listed last in users.conf *ACK*
21:02.44dexpdxtry doing a Verbose('DEBUG: number is matching') instead of a Dial or what cmd and if you don't see the msg in the cli then you are doing something wrong
21:03.24*** join/#asterisk NirS (n=NirS@87.68.17.141.cable.012.net.il)
21:03.24dexpdxso the inbound number is falling into the the wrong context?
21:03.52vap0rtranzdexpdx: weird thing is asterisk knows what extension i'm trying to send the inbound call to.  so it goes: "Call from '[last-peer-in-users.conf]' to extension '[correct-inbound-cid]' rejected because extension not found
21:04.20NirSg'day all
21:04.25NirSactually, g'night all
21:04.26NirS:)
21:04.27vap0rtranzdexpdx: [last-peer-in-users.conf] should be the same as [correct-inbound-cid]?
21:07.05*** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org)
21:07.18dexpdxusers.conf not where you specify dialplan contexts
21:07.39*** join/#asterisk enjay5150 (n=chatzill@ip70-190-63-195.ph.ph.cox.net)
21:08.22*** join/#asterisk axisys (n=axisys@155.70.141.45)
21:08.32stansmithvap0rtranz: i love it
21:08.34vap0rtranzdexpdx: right, i've only specified the context for the trunks there.  i'm assuming that context is for inbound & outbound dialing? b/c it's type = peer
21:08.47vap0rtranzstansmith: eh?
21:08.56*** join/#asterisk lirakis (i=lirakis@66.252.24.133)
21:09.07*** join/#asterisk jlb (n=jlb@75.148.162.90)
21:09.13lirakisanyone know a good open source sip stack / library
21:09.23dexpdxvap0rtranz: rtfm
21:09.31stansmithlol
21:09.43*** join/#asterisk Tebi (n=tebi@gw.aller.fi)
21:09.44*** join/#asterisk newmember (n=chatzill@static-66-11-81-65.ptr.terago.ca)
21:09.50*** join/#asterisk shmaltz (n=chatzill@mail2.dmaven.com)
21:10.05vap0rtranzwhatever.  it works by setting the context in users.conf to default and the default context in sip.conf to default
21:10.08vap0rtranzthanks anyways
21:10.12shmaltzwho overher is using switchvox?
21:10.17shmaltz~switchvox
21:10.28newmemberDoes a Cisco 7960 phone need a TFTP server or can I just set the SIP proxy manually on the phone?
21:10.37shmaltz~time
21:10.38jboti heard time is 1 dimensional, or everlasting, an illusion, or 2008.03.03 21:10:38 GMT
21:10.45shmaltz~swithcvox
21:11.00shmaltz~switchvox
21:11.07lirakisshmaltz: its not there!
21:11.14jlbThe voip-info wiki claims that you can only store static config OR realtime config in a database, but not both. Anyone know if this still true in 1.6 (or even 1.4) ?
21:11.15shmaltzok,
21:11.36shmaltzjlb, why use realtime?
21:12.06lirakisshmaltz: because changes are dynamic ?
21:12.18shmaltzlirakis, what type of changes?
21:12.25lirakisshmaltz: .. or rather, the uptake is dynamic
21:12.28*** join/#asterisk DaleG (n=dale@hlfxns0149w-142177089010.ns.aliant.net)
21:12.38shmaltzlirakis, what type of changes?
21:13.25jlbshmaltz: because users/etc. are in the database
21:13.40shmaltzjlb, why not change the text files and issue a reload?
21:14.18jlbshmaltz: because asterisk isn't the only consumer of the data
21:14.27lirakisshmaltz: peers users queues, etc... read the webpage
21:14.52shmaltzjlb, so why not do both? duplicate it to a database as well and have a deamon update the config files, I know Verizon does it that way
21:15.00stansmithredundant
21:15.17jlbbecause that's a terrible architecture, isn't sensitive to frequent updating, etc.
21:15.35shmaltzjlb, really? how so?
21:15.47shmaltzjlb, if Verizon can do it so can you
21:16.31stansmithjlb: couldnt you use DB as master record, and use a lil perl or some other script magic to query results into text file?
21:16.41stansmithcron it up to run whenever
21:16.41*** join/#asterisk Dovid (n=Dovid@bzq-79-180-58-141.red.bezeqint.net)
21:16.42stansmithwammy
21:16.45stansmithall  updates go to DB and we are all happy
21:16.47jlbwell if I have to pick a file to update and reload, I would pick the one that changes infrequently (sip.conf) and not the realtime user data
21:16.54DaleGhey, I created a small prog that's a web-server to provision IAXy's (for Linux/OS X, others).  Point your browser at it, fill in the fields, click provision, and it does it's thing.  Looking for people to test it out...
21:16.54Dovidhi. i am running tests with a new carrier: which one of these is for Xeon ?
21:16.55Dovidhttp://asterisk.hosting.lv/#bin
21:17.46jlbI can definitely do something that will work... I was mostly wondering if the odd mutually-exclusive behavior of static and realtime config had gotten changed/fixed recently
21:18.24shmaltzanyone here using switchvox
21:18.26shmaltz?
21:18.37lirakisshmaltz: dude... come on
21:18.40shmaltz#switchvox is as dead as they come
21:18.48shmaltzlirakis, comeon what?
21:19.15stansmith@#$@~!
21:19.53shmaltzanyone here using switchvox?
21:20.36lirakisshmaltz: are you a robot?
21:20.41stansmithlol
21:21.22lirakisshmaltz: check for a robotic brain too
21:21.28stansmithshmaltz: i wasnt cursing at u, i was cursing at the fact that ${CHANNEL(channeltype)} only displays "Zap" and not the exact line
21:22.09[TK]D-Fenderstansmith: Its only supposed to give you the TYPE, just like its name implies
21:22.33shmaltzstansmith, ${CHANNEL} should have that
21:22.40shmaltzI guess I'm wrong for 1.4
21:23.02shmaltzanyone here using switchvox?
21:23.27stansmithi typed "core show [tab twice]" and didnt see variables.. i was hoping there was something inside asterisk that would allow me to see all variables available, is this feasible?
21:23.50[TK]D-Fendershmaltz: Ask a few more times, its not like you're wasting your breath.
21:23.58lirakisshmaltz: i use swithvox
21:24.12lirakisshmaltz: .. oh.. wait.. nevermind
21:24.18[TK]D-Fenderstansmith: "see variables
21:24.20[TK]D-Fender"?
21:24.21shmaltzlirakis, are you?
21:24.25lirakisshmaltz: you almost had me fooled last time you asked
21:24.41stansmithno such command
21:25.00[TK]D-Fenderstansmith: that was a QUESTION.  What do you mean "see variables"?
21:25.09JenniferAkemiif i type core show version and it says 1.4.18 does that mean i'm not running 1.4, but am running the trunk version or something?
21:25.14tzafrirstansmith, core show function <tab><tab>?
21:25.24stansmithyea
21:25.26[TK]D-FenderJenniferAkemi: 1.4 is a series.
21:25.27stansmithlike in the bash shell
21:25.37stansmith[TK]D-Fender: yes
21:25.42stansmithlike how u can see all the functions
21:25.51[TK]D-Fenderstansmith: "core show functions"
21:25.56stansmithno i know that
21:26.17stansmithi was hoping/wondering there was a way to see variables? rather than looking at http://www.voip-info.org/wiki-Asterisk+variables
21:26.22stansmithwhich is half-deprecated
21:26.36stansmithas in.. core show variables?
21:26.44mvanbaakI think there's a txt file in the source that list variables
21:26.44tzafrirstansmith, in 1.4 you can see globals
21:26.50stansmithah yes
21:27.01tzafririn 1.6 you can also see channel-specific variables
21:27.08tzafrirsee and set
21:27.33newmemberCan I run a Cisco 7960 without a TFTP server and just manually add a proxy?
21:27.44[TK]D-Fenderstansmith: On moment you are asking about variables, the next you're asking about functions.  Make up your mind.
21:27.53JenniferAkemi[TK]D-Fender: the reason i asked is cuz i got the g729a codec, and it says that it was compiled against an older version of asterisk and may cause instability, so i was wondering if i needd to download the one from the unsupported directory (the trunk one)
21:28.18JenniferAkemi[TK]D-Fender: I think i might have done an svn asterisk at some point trying to get something else to work so i'm not sure if i'm running something newer than 1.4 or not
21:28.19stansmith[TK]D-Fender: i never asked about functions...i was asking if there is a way to view available variables..similiar to how you view functions ( "core show functions" )
21:28.33vap0rtranzwhat's an * looping function?  something like do .. until
21:28.36stansmithJenniferAkemi: if you are running 1.4.18, u are running 1.4
21:28.41[TK]D-Fenderstansmith: "core show channel [channel]"
21:28.54lirakisnewmember: http://phoenix.labri.fr/documentation/sip/Documentation/Material/Clients/Hardphone/Cisco/Admin/AdminGuide-us.pdf
21:28.59tzanger[TK]D-Fender: I get an error "[channel] not found"  :-p
21:29.17[TK]D-Fendertzanger: Smart-ass :p
21:29.20JenniferAkemiok. how come i get the message about g729a being compile against an older version of asterisk?
21:29.22newmemberlirakis: ty I will take a look
21:29.27*** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com)
21:29.31lirakisnewmember: next time ask google first
21:29.58newmemberlirakis: you are assuming I haven't
21:30.00JenniferAkemivap0rtranz: probably while.
21:30.11lirakisnewmember: yeah.. b/c that took me like 3 seconds to get on google
21:30.19JenniferAkemivap0rtranz: there was an app_while or something i noticed in modules.conf
21:31.03stansmith[TK]D-Fender: rest assured..the info you pass onto me, i pass onto others when possible, we all appreciate it :)
21:32.16MatBoysomeone good experience with the TE-410P cards ?
21:32.18vap0rtranzJenniferAkemi: oh wow.  http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+While?    that is very cool.
21:33.36stansmithMatBoy: ask a more specific question
21:33.48MatBoystansmith, stability
21:33.59*** join/#asterisk ice_croft (n=nolan@213.132.86.246)
21:34.06JTpretty specific...
21:34.06vap0rtranzJenniferAkemi: the book had some silly GoTo statements.  it was like a blast from the past ... BASIC that is :P
21:34.44MatBoystansmith, I can buy 2 for $500,-
21:35.21JenniferAkemivap0rtranz: glad to help
21:35.33MatBoynew ones actually
21:35.44JenniferAkemiHow do i know if my g729a is still registered
21:35.53stansmithlegally?
21:36.15JenniferAkemiyeah
21:40.25*** join/#asterisk Nasra (n=maxshipp@190.166.70.107)
21:40.57*** part/#asterisk lirakis (i=lirakis@66.252.24.133)
21:41.00*** join/#asterisk remmo (n=junk@203.32.47.250)
21:42.41NasraSteveTatero
21:42.49*** join/#asterisk seanbright (i=seanbrig@65.207.74.18)
21:43.11MatBoyyep, legally
21:43.41J4k3since when did you have to register your g729a installation?
21:43.41NasraIs that you?
21:43.56J4k3afaik all you needed was enough licenses to cover specific usage
21:44.16J4k3it'd also help greatly if digium's g729 versions actually worked.
21:44.28*** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net)
21:45.05NasraI am in the process of installing a small server...
21:45.19Nasrawith linux
21:45.29Nasradebian
21:45.37DaleGAny people out there have any large installations of IAXy's?
21:45.46*** join/#asterisk SparFux (n=raoul@e182030185.adsl.alicedsl.de)
21:46.55*** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk)
21:47.04*** join/#asterisk stansmith (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net)
21:48.11adeelyou know, if your end point & sip provider support g.729, you don't need a license for 'pass-through' mode
21:48.19*** join/#asterisk angom (n=angom@200.79.141.128.dsl.dyn.telnor.net)
21:48.26stansmiththe answer to the question i had a little bit ago was ${CDR(channel)} ... ${CDR(channel)} displays whatever line is being used..in my case it is = to "Zap/4-1"
21:48.30JenniferAkemiwell i bought g729a licenses
21:48.35stansmithjust in case anyone else has trouble with that later on ^^^
21:48.39JenniferAkemibut now my g729 function thing is gone from the CLI
21:48.41SparFuxLicenses?
21:49.08adeelJenniferAkemi, do you still have the codec loaded?
21:49.12*** part/#asterisk stansmith (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net)
21:49.15JenniferAkemihow do i tell adeel
21:50.52adeelJenniferAkemi, show modules should print out the loaded modules
21:51.14adeelJenniferAkemi, should start with codec_ and have g729 somewhere in it
21:51.38JenniferAkemiyeah i have codec_g729a.so
21:52.49*** join/#asterisk Servergod (n=servergo@70.97.159.120)
21:53.19Servergodanyone ever have a cisco186 only ring twice then give a 480?
21:53.43enjay5150JenniferAkemi: have you restarted since you registered/installed?
21:53.49adeelJenniferAkemi, hmmm...well then you should have access to your g729
21:54.38*** join/#asterisk bkw__ (n=brian@adsl-76-196-203-125.dsl.tul2ok.sbcglobal.net)
21:54.39SparFuxLicense? I have no one , but /usr/lib/asterisk/modules/format_g729.so
21:54.41JenniferAkemiyes i restarted many times since i registered
21:55.04JenniferAkemibut since then i've been doing stuff in modules.conf
21:55.21adeelJenniferAkemi, can you try doing a 'module reload codec_g729a.so'
21:55.44JenniferAkemicodec_g720a.so does not support reload
21:56.09JenniferAkemisorry typo, codec_g729a.so does not support reload
21:56.10adeelJenniferAkemi, so do a 'module unload codec_g729a.so' followed by 'module load codec_g729a.so'
21:56.12SparFuxfirst unload then load
21:57.17JenniferAkemii get unregistered translator lintog720 lines, does that mean it's not registered?
21:57.43adeelJenniferAkemi, if you get that while unloading, then yes, it has removed it
21:57.44Dovidhi. i am running tests with a new carrier: which one of these is for Xeon ?
21:57.45Dovidhttp://asterisk.hosting.lv/#bin
21:57.57JenniferAkemiwhen i load it
21:57.59JenniferAkemioh wait
21:58.04JenniferAkemii think we're good
21:58.15JenniferAkemiit says it found license, and how many there are
21:58.40JenniferAkemiexcept i still don't have the g729 cli command
21:58.53adeelJenniferAkemi, type in 'help' and see if anything shows up
21:59.44JenniferAkemiyes, i tried that, but no it isn't listed
22:00.01JenniferAkeminm
22:00.15JenniferAkemii was confused. it was show g729 that i type, not g729 show
22:00.22JenniferAkemithanks for your help adeel
22:00.26adeelJenniferAkemi,  no problems
22:00.47Servergodanyone ever have a cisco186 only ring twice then give a 480?
22:00.55*** part/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net)
22:01.01*** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net)
22:01.09enjay5150servergod, you have a timeout?
22:02.44Servergodhttp://pastebin.com/m79f86ac0
22:02.57Servergodnot that i can see
22:03.11Servergodit just sems to do this with the cisco ata186/8
22:03.41Servergodlemme do one w sip debug
22:03.55dexpdxOk, I'm at my witts end with this cdr_odbc thing
22:04.48dexpdx[Mar  3 14:03:39] ERROR[14748]: cdr_odbc.c:358 odbc_load_module: cdr_odbc: Unable to connect to datasource: cdr
22:04.56dexpdxisql f'ing works!
22:04.58dexpdxwtf
22:06.57JTdexpdx: why are you telling us?
22:07.04dexpdxI need some help
22:07.18JTdexpdx: how can anyone help you with so little info?
22:07.19adeeldexpdx, can you connect to your odbc from the console, outside of asterisk?
22:07.31dexpdxadeel: yum isql works perfect inserts/selects etc
22:07.40dexpdxs/yum/yup
22:07.52dexpdxJT: that's all the info I get
22:07.57JTrubbish
22:07.59JTyou have configs
22:08.03dexpdxwith debug and verbose set higher than 10
22:08.03JTpastebin them
22:08.07dexpdxJT: sure
22:08.08dexpdxnp
22:08.09dexpdxone sec
22:08.51adeeldexpdx, okay, so you can do a direct isql, have you tried testing your odbc config?
22:09.26dexpdxadeel: which odbc config - asterisks
22:10.32vap0rtranzanyway to get the output of reloads to a text file w/out enabling mega-verbose debugging?  asterisk doesn't pipe the info to stdout from -rx :(
22:10.55lanningscreen
22:11.20vap0rtranzlanning: heh. besides the screen
22:11.38dexpdxJT: http://pastebin.ca/926909
22:11.45lanningnot "the screen"
22:11.49lanning"screen"
22:12.26lanninghttp://www.gnu.org/software/screen/
22:12.43dexpdxor 2>&1
22:12.48dexpdxie redirect STDERR to stdout
22:13.41vap0rtranzdexpdx: doesn't play nice with sudo :)
22:13.51vap0rtranzlanning: awesome, already installed too
22:13.56dexpdxasterisk -rx 'module reload' 2>&1
22:14.02dexpdxvap0rtranz: the hell it doesn't
22:14.39dexpdx[jason@pstn-gw1 ~]$ sudo /usr/sbin/asterisk -rx 'exit' 2>&1 >foo
22:14.40dexpdx[jason@pstn-gw1 ~]$ cat foo
22:14.40dexpdxNo such command 'exit' (type 'help' for help)
22:14.46dexpdxproof right there
22:15.16vap0rtranzdexpdx: screen is good.  had forgotten about it.
22:15.21lanningswap them
22:15.42lanningsudo /usr/sbin/asterisk -rx 'exit' > foo 2>&1
22:16.02lanningredirect STDOUT, then copy the new STDOUT to STDERR.
22:16.03ManxPowerlanning: too bad that won't work
22:16.23*** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net)
22:16.39ManxPowerexit is NOT valid in an -rx
22:16.49x86stop now is ;)
22:16.50dexpdxManxPower: I know
22:16.52SparFuxlanning: first of all, you need quote with sudo, to have user id still with stuff after >
22:16.53ManxPower"stop now" would be what you want.
22:16.53dexpdx;)
22:16.54vap0rtranzManxPower: thank you.
22:16.57dexpdxNo such command
22:17.08dexpdxManxPower: I didn't want to stop my instance of asterisk ;)
22:17.13*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
22:17.22x86dexpdx: just run asterisk from inittab ;)
22:17.42ManxPowerdexpdx: what exactly DO you want to do?
22:17.43lanningthe error (no such command) should goto the file (foo), unless asterisk is writing to /dev/tty.
22:18.00x86inittab+asterisk==lovinit
22:18.07dexpdxManxPower: I was showing vap0rtranz that you can redirect the output of -x to a file with sudo
22:18.15ManxPowerAh.
22:18.21ManxPowerpoor thing didn't even know that?
22:18.23SparFuxfrom inittab?
22:18.40Kattyo hai!
22:18.47dexpdxx86: you go ahead and drop what ever calls you want on any of your instances just to should the output of a command
22:18.58dexpdxs/should/show
22:19.52x86dexpdx: inittab doesn't drop calls
22:19.53vap0rtranzManxPower: i did.  that hasn't worked.  but screen looks to be good
22:20.06x86dexpdx: inittab just assures that if asterisk dies, it's immediately restarted
22:20.28x86dexpdx: if you want to run commands, use AMI
22:20.49dexpdxx86: so if I issued a 'stop now' in the CLI because asterisk gets restarted under inittab it wouldn't drop any calls
22:20.53dexpdx?
22:20.56dexpdx*cough*
22:21.24dexpdxI don't use inittab anyways I just use safe_asterisk
22:21.25x86dexpdx: those were two different things
22:21.39x86dexpdx: of course you use inittab... unless you use upstart
22:22.07dexpdxx86: of course I use inittab but I don't start asterisk directly from it
22:22.08dexpdxerr
22:22.16x86you should, is what I'm saying ;)
22:22.20dexpdxwhy
22:22.24lanningx86, the issue was not "restarting asterisk" it was "demo a command output redirection, without any restart"
22:22.26dexpdxdon't need to
22:22.31x86so if it dies (which happens from time to time), it will automagically get restarted
22:22.41x86lanning: gotcha
22:22.44dexpdxx86: just like safe_asterisk restarts it
22:22.56x86dexpdx: except inittab is better
22:22.59vap0rtranzdexpdx: did you actually try redirection of reload?  it doesn't even work as root
22:23.13dexpdxexcept not
22:23.19x86dexpdx: sure it does
22:23.23dexpdxvap0rtranz: yes, I have done it
22:23.24vap0rtranzexit works
22:23.24x86s/does/is/
22:23.30*** join/#asterisk mosty (n=mostyn@ppp191-34.static.internode.on.net)
22:23.36dexpdxx86: why?
22:23.39x86dexpdx: use AMI to capture the output of a command
22:23.49x86dexpdx: because what happens if safe_asterisk dies?
22:24.11x86dexpdx: have you ever seen init die on a box?
22:24.17lanningyes
22:24.17dexpdxx86: yes
22:24.19dexpdx;)
22:24.40dexpdxx86: I don't make a habbit of putting my auto-restarting processes in inittab
22:25.14x86dexpdx: scrap safe_asterisk for asterisk+inittab
22:25.32dexpdxx86: how often do shell scripts simple die on your systems?
22:25.32x86dexpdx: i guess if you're just playing around, it doesn't really matter
22:26.01lanningx86, you are barking up the "personal preference" tree...
22:26.03dexpdxI also prefer not to have to edit inittab and -HUP everytime I want asterisk to "stay off"
22:26.04x86dexpdx: it's plausible if oom_killer decides to chomp it :)
22:26.11lanninguse "vi" not "emacs"
22:26.19x86lanning: nope... oom_killer wont nuke init... period
22:26.19vap0rtranzlanning: *applauds*
22:26.31x86lanning: no, i have reasoning
22:26.38x86not just personal preference
22:26.55lanningeither way works.
22:26.59x86sure
22:27.03x86one works better though ;)
22:27.06lanningenough said
22:27.18vap0rtranzx86: oh bother
22:27.18x86*nods*
22:28.10dexpdxyou do not run trivial restartable processes from inittab and you do not run programs that kill non trivial programs
22:28.19dexpdxprograms/services/processes
22:28.51*** part/#asterisk ice_croft (n=nolan@213.132.86.246)
22:28.51dexpdxi.e if asterisk is important enough to run from inittab your don't run a program that will kill it
22:28.57*** join/#asterisk nirz (n=nir@89-138-78-170.bb.netvision.net.il)
22:29.02x86dexpdx: i never said to
22:29.02dexpdxthat will - that could
22:29.18dexpdxthen why are you concerned that oom_killer will kill it?
22:29.19x86dexpdx: from a high-availability standpoint, inittab makes sense
22:29.29x86dexpdx: oom_killer is part of the linux kernel
22:29.41dexpdxx86: it's a matter of taste - I don't like running things from inittab
22:30.06x86dexpdx: then dont :)
22:30.20dexpdxI also don't run other services on my asterisk machines ;)
22:30.30x86me either
22:30.32vap0rtranzdexpdx: not even ntp!
22:30.34vap0rtranz*gasp*
22:30.38dexpdxclient
22:30.39dexpdx;)
22:30.52x86well, I run apache just to do CDR reporting stuffs
22:31.02dexpdxmy pbx's sync to an "operator" host that has ntp/dns etc on it
22:31.26dexpdxx86: that's part of the reason why I'm trying to get cdr_odbc working
22:31.27vap0rtranzdexpdx: an my ip phones sync to the asterisk box, per the docs of tiered ntp synching :)
22:31.52x86dexpdx: what DB?
22:31.59dexpdxx86: oracle
22:32.06x86nice :)
22:32.15J4k3oracle?  and you're on irc trying to get support?!?! :D
22:32.21x86hahaha
22:32.23x86no kidding
22:32.23dexpdxJ4k3: the problem is not with oracle
22:32.31J4k3you can pay it, shut up.
22:32.33x86you'd think he'd be using CM ;)
22:32.36J4k3err, afford it.
22:33.06J4k3x86: he should be using a consultant :P
22:33.13x86*nod*
22:33.14J4k3I mean, if you can afford a money bucket like any oracle product
22:33.22J4k3you can afford a money bucket like a consultant.
22:33.24x86or Cisco Call Manager
22:33.32dexpdxJ4k3: ok enough already - I'm perfectly comfortable w/ oracle it's asterisk that I have less xp with
22:33.46J4k3buying ccm is just the art of buying a lot of support time.
22:33.53x86*nod*
22:34.02J4k3of course, that goes for all cisco products
22:34.02x86dexpdx: so what's the problem?
22:34.05dexpdxI've never used CCM - I used Avaya in the past
22:34.08*** join/#asterisk eth01 (i=foobar@gentoo/user/eth01)
22:34.16J4k3cisco: we don't make it better, we just give you a lot of white-shirt-wearing support.
22:34.32dexpdxx86: http://pastebin.ca/926909 take a look
22:34.34dexpdxno connect
22:34.43x86J4k3: and sometimes they give you propaganda t-shirts as well, free of charge!
22:35.03x86dexpdx: do you see any connection attempt at all?
22:35.06dexpdxit really shouldn't matter that I'm using oracle as long as I'm going through the unixODBC layer
22:35.07[TK]D-Fenderx86, No, thats a "bundled  expense"
22:35.26dexpdxx86: from the oracle side or asterisk side?
22:35.33J4k3x86: oddly my shirts quit coming when I quit paying for smartnet
22:35.42J4k3'free' my ass :D
22:36.26J4k3(any of you have seen kung fu hustle?)
22:36.38J4k3cisco my ass, smartnet my ass, free my ass, support my ass...
22:36.47dexpdxpeople wouldn't need cisco or avaya if someone would make hardware for asterisk that supports hot-swapping hardware
22:36.51dexpdxi.e. new pri ports
22:37.21J4k3dexpdx: well, I seriously doubt any of the non-* related pbx's out there run on top of linux
22:37.37J4k3they may use linux for certain parts of the system, but not the primary kernel.  linux is a huge limiting factor IMHO.
22:38.01dexpdxJ4k3: the how-swapping in the OS is not the problem...
22:38.18J4k3dexpdx: but I bet the kernel goes *hiccup* when it occurs :)
22:38.25dexpdxin fact all someone would have todo is build a "firewire" chassis of some sort
22:39.02x86dexpdx: ATCA? :)
22:39.11dexpdxJ4k3: I'm not arguing that there are problems with it... I'm just saying that kind of stuff needs to happen before softswitches can really compete
22:39.17J4k3but, IMHO, asterisk's primary market is a lot smaller than that
22:39.24x86dexpdx: there are ATCA chassis that are more than capable of running linux and asterisk
22:39.38J4k3* sells well in places that would normally pay out the ass to their telco for centrix services.
22:39.48x86J4k3: centrex?
22:39.58dexpdxx86: does asterisk support the addition of new hardware with out reload?
22:40.05J4k3x86: err, wasn't centrix bell's name for switch-based-pbx?
22:40.32x86J4k3: i'm using Asterisk in an organization with 7 sites doing outbound callcenter stuff... about 50,000 calls a week going out 7 Asterisk boxes
22:40.57x86J4k3: Centrex is basically a POTS line with crazy features on it
22:41.04J4k3around here the telco peddles switch-based pbx services for about $100/month/line + rediculously high standard PSTN LD rates.
22:41.37*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
22:41.39x86heh
22:45.35*** part/#asterisk eth01 (i=foobar@gentoo/user/eth01)
22:48.03dexpdxres_odbc seems to connect just fine
22:48.17dexpdx*splat*
22:49.10dexpdxoh, fucking weird
22:49.24dexpdxif I load up res_odbc and it connects cdr_odbc connect just fine
22:49.31dexpdxno res_odbc and cdr_odbc won't connect
22:49.34dexpdxstrange
23:00.39vap0rtranzcan group be used to call any sip channel in it? i mean, what does channelavail checking if not some macro that plays with statuses returned by dial() ...
23:02.18mostyvap0rtranz, huh?
23:04.17vap0rtranzmosty: if there's a dozen channels that can dial out, but some might be in use, what's doing the logic for which channel to pick?  the wonderful world of *now is missing the channelvariables.txt files and my answer is probably in there
23:05.04mostyvap0rtranz, what channel specification are you dialing with?
23:05.05vap0rtranzsip
23:05.11vap0rtranzbut will add zap later
23:06.10*** join/#asterisk PepOSX (n=angeldav@201.243.76.220)
23:06.22*** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com)
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23:16.29mostyso Dial(SIP/100&SIP/101) for example?
23:17.30vap0rtranzmosty: that looks messy for 12 lines ... but is that the only way?
23:17.42*** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk)
23:18.35vap0rtranzi had started writing a While() loop checking the returning channelstatus for each trunk configured for outbound ... just wondered if there was something else
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23:20.08[TK]D-Fendervap0rtranz, Can't see how you would loop those.....
23:20.29*** part/#asterisk LjL (n=ljl@ubuntu/member/ljl)
23:22.09vap0rtranz[TK]D-Fender: yea i got stuck on the returned statuses, that why i was hoping the channelvariables.txt would say something.  looks like i'll just create the amount of trunks on the fly (write a dial string with the amount of trunks that any customer has bought)
23:22.15SteveTotarofffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffff\gk
23:22.34vap0rtranzSteveTotaro: you collaspe?
23:22.52kraypiuslawl
23:23.08SteveTotarodog was laying on my wireless keyboard on the floor
23:23.41SteveTotaronot sure how he got that combination of characters though
23:24.29*** join/#asterisk xcompass (n=compass@sr-78.srsv01.resnet.ubc.ca)
23:27.08*** join/#asterisk Jason99 (n=jason@jason.unitz.ca)
23:27.29*** join/#asterisk Washy (i=Washy@gateway/tor/x-276b0656a12bea20)
23:28.20WashyQuestion: Is it possible to get PTSN access without buying service?
23:29.02Jason99Hello all... have a question about Asterisk 1.4..  Last week we upgraded to 1.4 from 1.2 and we've been hearing random DTMF tones on the line but no one is pressing keys.. has anyone else heard of this?
23:29.45*** join/#asterisk angryuser (i=nononon@df01t2-213-44-161-86.d4.club-internet.fr)
23:30.03*** join/#asterisk ManxPower (n=manxpowe@ip72-204-168-61.no.no.cox.net)
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23:30.25vap0rtranzJason99: yes.  it's awful. who's your carrier?
23:30.51[TK]D-Fendervap0rtranz, No.  You should not be doing a "multi-dial".  That will be quite bad.  Just dial them 1 at a time, back to back.  if the 1st gets answered then it will not proceed to the 2nd.
23:31.09*** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com)
23:31.27Jason99vap0rtranz: We're using an AudioCodes Mediant 2000 which has Rogers PRI connected to it
23:31.27tzafrirWashy, sure
23:31.45Washyhow?  do you need home phone service?
23:31.51vap0rtranz[TK]D-Fender: so i should not use Dial(SIP/trunk_1& .. &SIP/trunk_n)? i should just list the trunks in a dialplan
23:32.05Washydo you need any sort of paid service?
23:32.06mostyvap0rtranz, you can use call queues with the ringall policy, or you could use variables, Dial(${SALES}) where you have previously set SALES=SIP/100&SIP/101 etc
23:32.15[TK]D-Fendervap0rtranz, No, you should dial them sequentially
23:32.18Washyhow do you get a normal phone number?
23:32.36vap0rtranz[TK]D-Fender: mosty and you are saying conflicting things ...
23:32.50vap0rtranzmosty: that's what i was thinking ... but evidently bad idea
23:32.55tzafrirWashy, all you need is a cousin at some service provider
23:32.55ManxPowerYou don't usually want to do paralell dial instead of a queue
23:33.11ManxPowerit will REALLY piss people off when their phone keeps ringing but someone else catches the call
23:33.23mostyvap0rtranz, no it's not a bad idea, it's just a different behaviour- it depends what you want this to do exactly
23:33.25[TK]D-Fendervap0rtranz, You seem to be asking about going through a number of outbound providers to get to the PSTN, is that correct?
23:33.46Washytzafrir: shut up
23:34.41vap0rtranz[TK]D-Fender: maybe i'm just overthinking.  but surely * has some way of knowing you've registered several providers and it picks one to dial out with ...
23:34.56[TK]D-Fendervap0rtranz, No, it doesn't, and I'll take your answer as a "yes"
23:35.12[TK]D-Fendervap0rtranz, * dials exactly, and only what you explicitly tell it to in your dialplan.
23:35.26ManxPowervap0rtranz: No, you Dial() if it fails it will exit and set a dialplan variable DIALSTATUS and HANGUPCAUSE
23:35.39[TK]D-Fendervap0rtranz, and indeed you do NOT want to be doing "Dial(SIP/sadggh&sip/SJHGSDJS....) and combining them there.
23:35.56vap0rtranzmosty: ah, well we have some customers who want separate lines for separate things, so i can't just put all the registered sip providers in one group and let them dial out.
23:36.02[TK]D-Fendervap0rtranz, just dial them sequentially like I have described.
23:36.14[TK]D-Fendervap0rtranz, No, you can't.
23:36.18vap0rtranz[TK]D-Fender: ok
23:36.53Washyanyone?
23:36.56[TK]D-Fendervap0rtranz, so set up your dialplan to go through your resources in the order you wish to prioritize based on what was dialed.
23:36.59mostyvap0rtranz, what kind of things do you want to separate the calls based on?
23:36.59*** join/#asterisk joobie (n=joobie@joobie.org)
23:37.33[TK]D-FenderWashy, somebody is paying for it, and sure there are "free" services, but ehy usually have a different kind of "cost"
23:38.00Washyexamples?
23:38.07vap0rtranzmosty: in one case, one extensions gets its own "priority" sip line, in & outbound.  in another, the lines are used for different customers so has to be separated out (for ease of call accounting it seems)
23:38.39[TK]D-FenderWashy, ipkall.com offers free inbound service
23:38.54Washywhat's the cathc?
23:38.58Washycatch?
23:39.05outtoluncmissed the convo, but least cost routing can be 'tiered' for more important clients <G>
23:39.24mostyvap0rtranz, then each of your sip phones should start in a different context, based on that reasoning. from there you can dial via different services
23:39.57vap0rtranzmosty: and the sip providers in separate contexts
23:40.32vap0rtranzi'm going to go KISS and just enumerate through the lines :) under the correct contexts of course
23:40.33[TK]D-FenderWashy, For this on, not perceivable "catch".  GrandCentral is one that can be used with some trickery for free outbound via Gizmo.
23:40.40mostyeg in [group1] you can Dial(SIP/provider1...etc) followed by Dial(SIP/falloverprovider...etc)
23:40.59joobieguys
23:41.11joobieis there a good / cheap device that can convert SIP to analogue?
23:41.22joobiei want to support around 10-20 phones with the device
23:41.37joobieso i can run analogue phones around the office, rather than voip
23:41.44[TK]D-Fenderjoobie, good != cheap.
23:41.47joobiehehe
23:41.50ManxPowerWell, you can get the cost down to something like $30/port if you are willing to have 5 - 10 devices, power supplies, etc
23:41.58ManxPowerSipura 2100 or similar
23:42.00joobiealso are there any down sides to this solution?
23:42.08[TK]D-Fenderjoobie, but your best value would probably be 3 x Linksys SPA-8000 (8 FXS each).
23:42.12outtoluncjust get a t1 card and a channel bank
23:42.16ManxPowerjoobie: It's analog and you have to manage 1 device for every 2 lines
23:42.28ManxPoweror whatever [TK]D-Fender recommends
23:42.28[TK]D-Fenderouttolunc, far from "cheap"
23:42.32[TK]D-FenderManxPower, lol
23:42.44joobiewhy one device for 2 lines?.. it's still an RJ12 ya?
23:42.47outtoluncnot all channel banks are expensive
23:42.51joobieone-to-one?
23:43.00outtoluncand a sip device * 20 isnt' cheap either
23:43.00vap0rtranzmosty: thanks for the help
23:43.06[TK]D-FenderManxPower, the SPA-8000 does 8 ports at the same cost ratio as 2-port ATA's so I suggest it these days... and its a lot less to configure...
23:43.07vap0rtranzand [TK]D-Fender
23:43.43joobiehmm
23:43.45joobiewhat about the quality tho
23:43.54joobiedo u lose anything in quality / features going this path
23:43.59joobieas opposed to true digital handsets?
23:43.59adeelis it true that * doesn't work well on a machine with more than 1 NIC?
23:44.28ManxPowerjoobie: you lose almost every single feature a digital set has
23:44.39outtoluncanyways <G>
23:44.59Jason99has anyone heard of random dtmf tones on asterisk 1.4 even if no digits were pressed?  Using rfc2833... Has anyone found a solution?
23:45.15ManxPowerI do agree with outtolunc, if you are going be stupid and short sighted enough to go with analog at least use a channel bank
23:45.23[hC]Can I add an extra field to a CDR that gets passed forward through the other * boxes a call might go through? I have a CPE asterisk box that sets an account code right now for if they used a 'long distance code' to dial out, but that accountcode is trampled when the call comes to me, to send to the pstn
23:45.24outtoluncnods
23:45.34mostyJason99, i have heard of people with that issue, have you checked bugs.digium.org ?
23:45.36WashyIs IPkall doing something with your incoming #?
23:45.44Washysomething naughty
23:45.45[TK]D-Fenderouttolunc, new math for you : http://www.voipsupply.com/product_info.php?products_id=2912
23:45.48adeelJason99, can you elaborate your setup?
23:46.06*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
23:46.17drmessanoNaughty?
23:46.17[TK]D-Fenderouttolunc, $250 = 8 ports * 3 = $750 for 24 ports, no need to mess with hardware in your server or hit-or-miss ebay-ing
23:46.31adeelJason99, i had that problem when i was using * 1.2 machines to connect to * 1.4 and NAT/reinvite's
23:46.34joobiewhat are some of the major features ManxPower
23:46.38mosty[hC], i forget if there's a way to pass channel variables over IAX besides mangling the dial string
23:46.41[TK]D-Fenderouttolunc, And relieves IRQ load etc off your server.
23:46.44joobiethis will be for a call center.. so it's just outbound calls.. nothing fancy
23:46.50Jason99ATA --> Asterisk 1.4 --> Asterisk 1.4 --> AudioCodes Mediant 2000  --TDM--> Rogers PRI
23:46.58joobiei mean, is there quality of audio loss or something?
23:47.05ManxPowerjoobie: go to polycom, cisco, or aastra's site to learn about their IP phones.  Polycom calls them "SoundPoint"
23:47.12WashyI dunno, like listening to your calls or something
23:47.16outtoluncfirst is it even out, and the cost is $300 for 8 ports so $300 * 3 = $900 check your math
23:47.24drmessanoWashy: No, why?
23:47.37joobieManxPower, I know.. ive been looking at the 330
23:47.38[TK]D-FenderJason99, Set your mediant to rfc2833 encoding towards * and you should not get DTMF issues.  Avoid in-band at all costs...
23:47.39outtoluncwhy is trying to help so flippin hard in here nowdays?
23:47.41vap0rtranzadeel: Jason99 hits this problem on the head.  i don't think it's necessarily 1.2 -> 1.4;  it seems to just pop-up sometimes even with all 1.4
23:47.50*** join/#asterisk craigk (n=craigk@58.174.150.119)
23:47.57ManxPowerjoobie: we did ONE analog Asterisk install.  NEVER again.  Users could not remember how to do call waiting or transfer calls.
23:48.07[TK]D-Fenderouttolunc, Page says $250.
23:48.08joobieManxPower, but im just curious.. if i went this analogue way with a sip->analogue device.. would i lose any voice quality.. noticable that is.. vs jsut straight digital to the end-user?
23:48.09adeelvap0rtranz, well random dtmf should imply an incorrect setup
23:48.13Washywhy do they provide free service then?
23:48.18ManxPowerand of course they blamed their problems with memory on the phone system
23:48.28Jason99[TK]D-Fender: i am using rfc2833 on the Audiocodes towards *
23:48.34vap0rtranzJason99: are the beeps typically towards the beginning of the conversation?
23:48.37joobieManxPower, that's not an issue here as it's only outbound calls
23:48.42joobiebut good to know:)
23:48.45[TK]D-FenderJason99, sure you set * to it as well to avoid double detection?
23:48.51drmessanoWashy: http://lists.digium.com/pipermail/asterisk-biz/2005-July/007081.html
23:48.51ManxPowerjoobie: for the most part, assuming you do not need to compress the voice (like if it went over the internet) there is not a significant audio quality loss
23:48.57joobieManxPower, what about the voice quality.. ? both mic and speaker? are they noticable degraded if you go analoge?
23:48.58vap0rtranz[TK]D-Fender: i bet he is rfc2833 all throughout the network
23:49.17Jason99vap0rtranz: it's anywhere in the call.. people report that it happens when people are speaking loudyly/laughing...
23:49.23adeelJason99, you might want to setup rfc2833compensate
23:49.28WashyBut asterisk doesn't enable you to interface with PTSN for free, right?
23:49.30joobieI do need to compress the voice
23:49.32*** join/#asterisk SteveTotaro (n=root@pool-71-179-144-229.bltmmd.east.verizon.net)
23:49.38joobiegiven this, will there be a difference?
23:49.40ManxPowerjoobie: Dude, I can't tell you what the quality of the speaker/mic of the phones you buy.
23:49.44adeelJason99, hmmm....i wonder if the loud speech is the problem
23:49.45mostyWashy, no of course not
23:49.46drmessanoWashy: Why not?
23:49.48joobieahh k
23:50.02vap0rtranzadeel: isn't that relaxdtmf?
23:50.09mostyWashy, no pstn phone company will give you pstn access for free
23:50.26SteveTotaroyou can splice into a 200 pair
23:50.34joobieim just curious man.. i mean, if i go the digital -> analogue .. I know there's a feature loss, but that's not a consideration here because it's only a single outbound call at a time that's required from these phones. I'm wondering if there's any other noticable differences with a setup like this.. any pros / cons
23:50.38Jason99adeel: will rfc2833compensate break anything else?  Is that something I should set as default in sip.conf ?
23:50.48Washywhat's a CLEC?
23:50.58ManxPowerWashy: a non-bell phone company
23:50.58[TK]D-Fenderjoobie, Using ATA's for phones is jsut fine if you have the extra phones laying around.
23:50.58SteveTotarogoogle know
23:50.59adeelvap0rtranz, they're 2 separate settings...relaxdtmf is one thing, while rfc2833compensate is supposed to compensate for pre-1.4 DTM transmission
23:51.03SteveTotaro~clec
23:51.03jboti guess clec is Created by the Telecommunications Act of 1996, a CLEC is a service provider that is in direct competition with an incumbent service provider. CLEC is often used as a general term for any competitor, but the term actually has legal implications. To become a CLEC, a service provider must be granted "CLEC status" by a state's Public Utilities ...
23:51.16ManxPoweryou don't ever want to use relaxdtmf
23:51.21[TK]D-Fenderjoobie, I linked to a product would would quite suitably fit your needs.
23:51.23ManxPowerif you need to use it, then you have OTHER issues.
23:51.27vap0rtranzadeel: i meant people's voices causes beeps, which seems to be one thing Jason99 notices
23:51.33SteveTotarosometimes you want to use relaxdtmf
23:51.38SteveTotaronever say never
23:51.40ManxPowerSteveTotaro: no you don't.
23:51.44[hC]anyone know if cdruserfield is carried forward in an iax call?
23:51.48joobie[TK]D-Fender, so you think buy like 2-3 of those devices and wack them all together?
23:51.50SteveTotarospeak for yourself buddy
23:51.52ManxPowerIf you have to use relaxdtmf then you have a gains issue with your telco
23:52.07*** join/#asterisk xcompass (n=compass@sr-78.srsv01.resnet.ubc.ca)
23:52.08SteveTotarowhatever works
23:52.10ManxPowerfix the audio level problems and the need for relaxdtmf goes away
23:52.11adeelvap0rtranz, i guess the question is what kinds of phones Jason99's clients are using
23:52.28SteveTotaroif relax dtmf gets the job done
23:52.29drmessanoSteveTotaro: Rewire your central office, lazy ass
23:52.37drmessano:)
23:52.44SteveTotarorather than fighting with your carrier for months about gains
23:52.54Jason99adeel: just to add to the mix.. the beeps are always on the far end.. my clients don't hear them, its the other end
23:52.58[TK]D-Fenderjoobie, Yup.  If I was budget conscious and wanted analog, that's what I would do.
23:52.59SteveTotaroand them having no idea what the hell you are talking about
23:53.14joobie[TK]D-Fender, what if it wasnt budget conscious and wanted analog?:P
23:53.29adeelJason99, so that means you're transmitting them...
23:53.38adeelJason99, what type of phones are you using?
23:53.48[TK]D-Fenderjoobie, then I'd spend like double the money on an AudioCodes MP-124 24-port redundant gateway :)
23:53.57Jason99adeel: not sure, it happens to several different clients, they all use different phones and different ATAs
23:54.02joobieeheh thanks TK
23:54.02*** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320)
23:54.08[TK]D-Fenderjoobie, thats what I'd use in large installs.
23:54.10Jason99adeel: if you think that could be a cause, i will find out
23:54.16SteveTotaroquintum tenor ax rulez
23:54.32Yourname``So, in the CLI of a 1.4 dualcpu duocore box, remote unix connection keeps happening and disconnected. No manager enabled.. what's going on??
23:54.45SteveTotaroor just go with an adtran or adit 600
23:54.52SteveTotarofor your analog
23:54.54[TK]D-Fenderjoobie, for SMB use I more than happy to stack a few SPA-8000's together.  Once you hit 5 units or so it DOES start to get a little ridiculous though.
23:55.06joobiehehe yea i can imagine
23:55.14joobiethanks heaps TK
23:55.17joobiealso the uplink to those
23:55.18[TK]D-Fenderjoobie, The solution should be scaled to the need.
23:55.24joobiedo u just consolidate via a switch / router
23:55.27joobieand just push out the net?
23:55.38joobieor do u go to an asterisk box.... and then out the net
23:55.43joobieyea - i hear ya
23:55.45[TK]D-Fenderjoobie, I only use elephant guns on ant problems when I want a free light & sound show as well :)
23:55.45SteveTotarothe solution should not be scaled to the need but the future need
23:55.46adeelJason99, you're not using canreinvite anywhere on the path are you?
23:55.56joobielol
23:56.02Washyy is stanaphone not accepting new customers?
23:56.07[TK]D-Fenderjoobie, I don't usually use or advocate using an ITSP for corporate PSTN
23:56.12joobieit's a good approach TK.. nto overkill.
23:56.26SteveTotaromaybe stan is going broke
23:56.28SteveTotaro?
23:56.33drmessanostanaphone is dead
23:57.03Jason99adeel: from the ATA through the 2 Asterisk servers, canreinvite=no, on the AudioCodes canreinvite=yes
23:57.07joobie[TK]D-Fender, i hear ya.. the cost problem is a factor here.. it's turned out to be pricy for the PSTN install so the client wants to go voip over itsp
23:57.25[TK]D-Fenderjoobie, lines are nasty in your area?
23:57.25joobie[TK]D-Fender, if that is a given though.. do u see the need to go through asterisk or just route straight out to the ITSP from the linksys device
23:57.32joobie[TK]D-Fender, they are expensive
23:57.48joobieit's around 1k install on a 24-m contract.. and about 250$/month ongoing.. minus calls
23:57.59joobie-24m +12m
23:58.10[TK]D-Fenderjoobie, you'll probably want * there so as to handle the calls.... you don't want literally independant jacks do you?  like 100% separate lines from each other that can't dial between themselves, etc...
23:58.10SteveTotarojust drop a tenor AX on your lan
23:58.14[TK]D-Fenderjoobie, Do you
23:58.17[TK]D-Fender?
23:58.21SteveTotaroand have it register to your itsp
23:58.30adeelJason99, so that means your using old analog telephones? i wonder if your ATA has any dtmf settings
23:58.38[hC]argh. how annoying. So there's no way to carry forward custom CDR fields between IAX/SIP connections?
23:59.01mosty[hC], there are indirect methods
23:59.05joobieTK, there's no requirement for phones to be able to dial each other.. they can be 100% independant
23:59.07ManxPower[hC]: SIP and IAX are not the same thing.
23:59.14ManxPowerThere is IAXVARS or something like that
23:59.22ManxPowerSIP has custom headers you can set/get
23:59.25joobie.. of course the feature wouldn't hurt, but it's not a requirement. It's outbound calls only... like a call center
23:59.32Jason99adeel: yes, the ATA are set to rfc2833.. using Mediatrix, Linksys SPA and Dlink DVG
23:59.35vap0rtranzJason99: do other IVR's work? as in, any buttons pressed are always recongized correctly
23:59.44Jason99vap0rtranz: yes
23:59.47[hC]ManxPower: Hm. So I guess the way to do it is to check for the presence of those on the otherside and write to cdr then.

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