irclog2html for #asterisk on 20051004

00:00.45Jam1ehmm
00:00.51Jam1enow it wont play nice with sipgate
00:01.34Jam1eah I see
00:01.36Jam1ehmm
00:02.32*** join/#asterisk [hC] (n=hardcore@8.10.2.73)
00:02.42[hC]hey guys... anyone here using *cowers* a sangoma t1 card?
00:04.09FuriousGeorgeim taking off, if anyone has any pointers for properly transfering calls via a remote client, please /msg me
00:04.58drbrownwill zaptel drivers compile with an amd 64 kernel????
00:05.01*** part/#asterisk BrianR___ (i=brianr@c-24-61-206-174.hsd1.ma.comcast.net)
00:08.22Jam1eHmm
00:08.52Jam1eI'm using asterisk and Sipgate, and getting "The service could not be connected" but my softphone is connected to asterisk, and asterisk is connected to sipgate
00:11.50*** join/#asterisk terrapen (n=cjs@fw-01.satx.bikeworld.net)
00:12.01terrapenhowdy all.
00:12.08*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
00:12.19terrapencan anyone suggest a currently non-sucky VoIP DID provider?
00:12.29terrapenI've been through four of them in the last year
00:12.36terrapeni want this next provider to be my last.
00:13.22InfraRedwhere are you based?
00:13.36terrapen<PROTECTED>
00:13.38rayvdall my base are belong to PIGEONS
00:14.10peggerterrapen i am currently using sellvoip.net , have been for just about a month
00:14.23terrapeni need business-class service
00:14.29terrapencan they deliver?
00:14.43terrapendo they have someone there to answer their phones when I call to report a problem?
00:14.50peggerterrapen when you say business class what do you mean by they
00:14.54pegger*that
00:15.37peggerterrapen well i have called once at night and once during the dya and they icked up right away when i called during the day
00:15.38terrapenwhat i just said
00:15.56pegger*needs time to type
00:16.42peggeris that usefull?
00:17.09terrapenwell
00:17.13terrapensort of, yes
00:17.39peggerterrapen are you going to now call them up and ask them questions?
00:17.47*** join/#asterisk justinnnnn (n=justinnn@61.95.68.85)
00:17.49justinnnnnhey ppls
00:17.52justinnnnncan someone help me pls
00:18.06justinnnnnatm im using a e100p with our isdn 10 service... channel 16 is the b channel
00:18.21terrapenpegger, perhaps
00:18.21justinnnnnrecently our service got upgraded to 20 channels.. with 31 being the d channel
00:18.31justinnnnnand when i changed zaptel.conf
00:18.40justinnnnnfrom this
00:18.41justinnnnnbchan=1-10
00:18.42justinnnnndchan=16
00:18.49justinnnnnto this
00:18.54justinnnnnbchan=1-02
00:18.56justinnnnnbchan=1-20 i mean
00:18.59justinnnnndchan=31
00:19.14justinnnnnand reload zaptel modules.. when i started asterisk it crashes when it loads the zaptel modules :(
00:19.29justinnnnnwhen i change it back to 1-10, 16 it works fine
00:19.46*** join/#asterisk pifiu (n=myassisb@208.205.181.170)
00:19.48peggerterrapen well if you do I would be curious to hear about what kinds of questiosn you ask and what kind of answers they are able to give you
00:20.00terrapenok, will let you know
00:20.06*** part/#asterisk terrapen (n=cjs@fw-01.satx.bikeworld.net)
00:20.08*** part/#asterisk MajestiK (n=MajestiK@68.149.67.26)
00:20.14pifiuhey guys what do you guys usually recommend to record while the phone first rings and we wait to answer?
00:20.20pifiulike for example
00:20.23justinnnnn???
00:20.37pifiu"thank you for calling xxxx . please hold while we direct your call
00:20.40pifiuis that fine?
00:21.03fmanoe1/t1 ?
00:21.04[hC]this channel is raelly getting out of hand.
00:21.15[hC]wow. i look educated.
00:21.17[hC]really*
00:21.29justinnnnnfmano is an e1
00:21.49fmanoeurope rulez... ehhehe
00:22.01pifiuin what way HC?
00:22.02peggerwhat kind of ping ms do most people get in the servers they are connected to?
00:22.02justinnnnnits wierd that asterisk just crashes it doesnt like 1-20
00:22.05justinnnnnbut 1-10 is fine
00:22.07pifiuanyone got any advice?
00:22.12justinnnnn??????????
00:22.32pegger[hC] out in hand in hwat way
00:22.45fmanoi've gotted a prblem when skipped channel's. 2 hfc + 1 zaptel
00:23.09fmano1-2+3 4-5+6 7 worked
00:23.27*** join/#asterisk cio (n=na@adsl-072-149-159-016.sip.bhm.bellsouth.net)
00:23.44fmanoif i putted the zaptel in the middle... it gives error
00:23.55justinnnnnhmm maybe i should put 1-21 ?
00:24.14fmanoyes.. or try put the d-chennel in middle
00:24.42fmanoi'm sorry. never worked in PRI
00:24.53[hC]uh.. you cant just pick where you want your dchannel to go
00:25.12fmanoit was just a guess
00:25.31[hC]dchannel on a standard t1 is channel 24.
00:25.36[hC]1-23 are usable voice channels
00:25.48fmanothe e1 is the double
00:25.51|Vulture|there is no dchannel on a T1 I thought
00:25.55fmano* almost *
00:26.03justinnnnnhc.. the d channel changed from 16 to 31..
00:26.11|Vulture|T1=1-24 T1PRI=1-23+1d
00:26.15justinnnnnso i changed it in zaptel... but when i change the channels bit it crashes asterisk
00:26.32cioHow good is the digium hardware as the CPE for a PRI?  (effectively skipping a channel bank)
00:26.42justinnnnnits working well for me
00:26.44justinnnnnexcept this issue
00:26.45*** join/#asterisk hyphenated (n=cgilmour@nzns1.eservglobal.co.nz)
00:27.01justinnnnnlike when asterisk starts it crashes and doesnt load zaptel module
00:27.08justinnnnnthen i change chanel bak to 1-10 she works fine :(
00:27.32[hC]justinnnnn: are you using a t1 pri?
00:27.39justinnnnnpri
00:27.42justinnnnne1
00:27.45fmanojustinnn : country ?
00:27.46[hC]e1.
00:27.46hyphenateddoes anyone have a good wiring guide for E1 cables? one with pretty colours
00:27.49justinnnnnaustralia
00:27.59Qwellhyphenated: google images
00:28.05[hC]justinnnnn: follow the e1 example for zapata.conf and zaptel.conf on voip-info.org
00:28.17hyphenatedI never thought of the image search.. :-)
00:28.25justinnnnnya ill try again
00:28.29justinnnnnmaybe i missed something :(
00:28.32[hC]I may be wrong (i have never done an e1, just t1) but you may have to specify voice channels 1-12, dchan=13, then 14-31
00:29.00fmanosomething like....
00:29.02justinnnnnour provider said just chan 31 will be the d channel
00:29.19[hC]ok then 1-30 for regular voice channels and dchan=31
00:29.26[hC]like i said, never done an e1, just a t1
00:29.31justinnnnnya that also crashed it
00:29.33justinnnnnbut 1-10 doesnt
00:29.36[hC]but there are plenty of instructions on voip-info.org
00:29.37justinnnnnevan if u leave 31 as the d
00:30.00fmanohere in portugal is 30B+1D
00:30.15tzanger30B+D is E1 PRI
00:30.20fmanoyes
00:30.25hyphenatedQwell: still can't find anything useful :-(
00:30.29tzanger32 channels, 1 for timing, one ofr signaling, 30 for bearer channels
00:30.45hyphenated(found ones that aren't RJ45 at both ends..)
00:31.18justinnnnnhmm i think u might be right
00:31.19justinnnnnbchan=1-15,17-31
00:31.19justinnnnndchan=16
00:31.25fmano"here in portugal is 30B+1D" -> info from Portugal Telecom
00:31.30justinnnnnhmmm
00:32.02hyphenateddoes anyone have an E1 cable lying around and can read out the colors? ;-)
00:32.15fmanonot meee
00:32.18justinnnnnisnt it just a straight through cable ?
00:32.20justinnnnni have no clue tho
00:32.30fmanoi think so...but....
00:32.58fmanojustinnnn: does it worked ?
00:33.31peggerhello everybody I am having trouble with my extensions.conf, it is working but when ever I try to enter any extension I get a error.  please let me know what I am missing http://pastebin.com/382195
00:34.10hyphenatedwhee, a search of mailing list had the answer for me. magic
00:34.18fmanojustinnn: go see this
00:34.20fmanohttp://archives.free.net.ph/message/20050819.210241.5b8d2aee.en.html
00:35.40pifiubesides allison smith does anyone know anyone else who does recordings?
00:36.46fmano"ISDN PRI signaling is also used for E1 connections. Such an E1
00:36.46fmanoconnection has 32 channels. For Asterisk, channels 1-15 and 17-31 carry
00:36.46fmanovoice and are called "B" channels, while channel 16 carries signaling
00:36.47fmanoinformation and is called the "D" channel. Channel zero is used for
00:36.47fmanoframing.
00:37.28pegger???
00:39.03peggerany idea why i can not dial extensions    http://pastebin.com/382195
00:39.07fmanopegger: i'm a newbie. but did u checked the "dialout rules" ?
00:39.26*** join/#asterisk jskcr (n=jskcr@unaffiliated/jskcr)
00:40.09jskcrhy all
00:40.42peggerfmano not exactly about the dial out rules but i can dial out with the dial command so that cant be the problem
00:41.03peggerDial() works , i am jut trying to dial extensions inside of asterisk
00:41.38hardwirepifiu: I am tempted to hire a morning show DJ
00:41.42hardwirejust because they are so popular
00:43.04InfraRedhire one to do what
00:44.52peggerwhy do i keep on getting Oct  3 17:22:20 WARNING[20374]: pbx.c:1937 ast_pbx_run: Invalid extension
00:45.00peggerwhat am i dong wrong
00:45.18fmanosorry pegger.... :( i dont now... very, very fresh in this stuff
00:46.04fmanofor me, and in *@home the solution was on the dial out .|.~
00:46.06fmanofor me, and in *@home the solution was on the dial out .|.
00:46.13fmanoi think...
00:46.29fmanoand a dial rule like "XX"
00:46.44fmanobut this was on zap
00:47.08peggerso in order to dial an extension you have to have a dial out rule?
00:47.10fmanoafter that worked like a charm
00:47.21peggerjust to dial a 4 digit extension
00:48.37fmanomy problem was slightly diferent....intercepting an outbound ISDN line, and manipulate her...
00:49.00fmanoto integrate ISDN + Legacy PBX
00:49.11fmano+ VoIp
00:49.19peggerfmano i am guessing you set this up for a business are you a contractor?
00:49.36fmanono.. i'm a TI...
00:49.56fmanoBut sometimes in portugal. we invent a lot...
00:50.08InfraRedTI?
00:50.16InfraRedtexas instruments?
00:50.23fmanono.....
00:50.24bendy24hes a chip?
00:50.27InfraRed:)
00:50.28*** join/#asterisk digime (n=drooth@ip68-111-235-172.sd.sd.cox.net)
00:50.30bendy24a ti chip?
00:50.52bendy24is this the first instance of intelligent silicn?
00:50.53fmanoi'm not made "by" silicium....
00:51.13InfraRedcalcuim ?
00:51.14InfraRed:)
00:51.23bendy24made "from" silicon?
00:51.26fmanoconsider me a computer technician, with lots off skills
00:51.31bendy24ohhhh
00:51.34bendy24IT...
00:51.38InfraRedformat c: ?
00:51.44bendy24heh
00:51.44InfraRedthat's my skill
00:51.45InfraRed:(
00:51.46fmanoformat c: d: e: f: g:
00:51.54InfraRedwww.fdisk.it
00:51.56fmanoRAID included
00:52.17Juxtoff skils are good skills
00:52.22Juxthow about on skills?
00:52.43fmanodont abuse from my english...
00:52.49bendy24im sorry
00:52.51fmanoeheheheh
00:52.59bendy24eh?
00:53.01fmanoit doesn't matter.....
00:53.37Juxtso for under $500 i got myself a dual cpu poweredge 1650 with raid and 2 gb of ram
00:53.41Juxti feel like i robbed someone LOL
00:53.57bendy24um
00:54.04bendy24did you grab one for me?
00:54.04SarahEmmuhh, how juxt?
00:54.11Juxtfleabay!
00:54.21bendy24does it work?
00:54.23Juxti am compiling the kernel on this baby right now
00:54.25bendy24lightning strike?
00:54.45bendy24sheesh
00:55.09fmanowhere are u from bendy ?
00:55.23bendy24canada, eh.
00:55.42wunderkinyou couldn't tell from the accent?
00:55.44fmanoless 8 hours ?
00:55.52bendy245
00:56.10fmanoa friend of mine came from there yesterday
00:56.15*** join/#asterisk outtolunc (n=me@ppp-71-129-1-85.dsl.pltn13.pacbell.net)
00:56.21bendy24cool
00:56.48fmanowellllll.... here are 2:00 AM... it's late
00:57.01bendy24and where are you located?
00:57.08fmanoPortugal
00:57.23fmanojust look across the Atlantic
00:57.44bendy24you mean spain?
00:57.44*** join/#asterisk chidex (i=richard_@82-45-239-141.cable.ubr01.enfi.blueyonder.co.uk)
00:57.56fmanono.. we are not spannish....
00:58.09fmanowe've got our own contry
00:58.19fmanoitīs name is PORTUGAL
00:58.25outtolunchehe
00:58.27bendy24hrph
00:58.40bendy24they told us spain ownz portugal in school
00:59.12fmanoin witch class ?
00:59.19bendy24geography
00:59.37fmanokick the butt of the professor
00:59.41bendy24just kidding :)
00:59.44bendy24gotta reboot
00:59.46fmanook...
00:59.55fmanook.. see u guys....
01:00.02fmanogo to beddddddd
01:00.10fmanokid + wife wainting
01:02.14chidexwondering whether anyone could help me with an asterisk problem?
01:02.32outtoluncwould help if we/they knew the problem
01:02.33SarahEmmdon't ask to ask, just ask :)
01:03.32outtoluncmy friend george has a prob with this umm... <G>
01:03.39chidexattended transfer causes ny asterisk to reboot after all parties have hung up
01:04.48outtoluncreason, being not sure if 'supervised transfer' has ever worked (across all protos)
01:05.43outtoluncmaybe if you stated what you were doing exactly, with what proto's, hard/softphones someone might be able to help
01:05.58chidexit's been advertised as one of its features though
01:07.00outtoluncah, so it must be true (in all aspects known to man)
01:07.29chidextransferring a sip calls between to GS-486 ata's
01:07.44outtoluncnow see that wasn't so hard
01:08.05chidex:)
01:08.17outtoluncthe next question you would be asked by 'someone else' would be what revision of firmware those devices had
01:08.53chidex1.0.6.7
01:09.14outtoluncand are the devices on the same network with the asterisk box?
01:09.16chidexbut this occurs on the softphone as well (eyebeam)
01:09.21chidexyep
01:09.43outtoluncok, so now formulate your question using that info for someone that 'could' help you
01:10.03*** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com)
01:10.12*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)
01:10.15hardwirevoip001*CLI> show uptime
01:10.15hardwireSystem uptime: 2 days, 4 hours, 44 minutes, 26 seconds
01:10.16hardwireyay
01:10.23hardwireyou may not think that its that cool.. but it is
01:10.41hardwirehi lilo
01:11.01hardwireouttolunc: nice
01:11.06hardwireI would like to be there some day
01:11.10hardwirebut more at the box level
01:11.15hardwirenot at the asterisk level
01:11.22hardwirewhich will always be changed by me
01:11.38outtolunci did it because i just hit the main power cord with my leg <G>
01:11.47hardwirehah
01:11.51outtolunchense why i disappear a few minutes ago
01:12.00hardwiredoh
01:12.04outtoluncnods
01:12.04*** join/#asterisk CrazyYoss (n=nobody@c-24-5-170-39.hsd1.ca.comcast.net)
01:12.22hardwirevoip001*CLI> show uptime
01:12.22hardwireSystem uptime: 2 days, 4 hours, 46 minutes, 32 seconds
01:12.27hardwireeven a few minutes later.. quite amazing to me
01:12.30outtolunccongrats
01:12.38hardwireyes
01:12.40hardwiretwo whole days
01:12.43*** join/#asterisk bjohnson (n=bjohnson@i216-58-58-152.cybersurf.com)
01:12.44hardwireone of them not even a work day
01:12.53hardwirelets go for a week
01:12.59outtolunclife has to start somewhere <G>
01:14.07CrazyYossim trying to get asterisk up and running. After compiling when I try and run it I get the following error message "[chan_modem_bestdata.so]/usr/lib/asterisk/modules/chan_modem_bestdata.so: undefined symbol: ast_unregister_modem_driver"
01:14.51chidexhardwire, have you discovered any bugs with your server?
01:14.53hardwireare you using a modem?
01:15.06hardwirechidex: tons!
01:15.08hardwireCrazyYoss: chan_modem isn't loading..
01:15.12hardwirewhich you probably don't even need
01:15.33hardwireun modules.con do a 'noload => ' on all the modem modules
01:15.36chidexany in relation to supervised transfer
01:15.48hardwirechidex: yeh.. its too easy my users can't get it
01:15.54CrazyYosshardwire: roger that, I'll give it a shot
01:15.55hardwirebut thats not really a software bug
01:16.21hardwirechidex: having issues bridging sip calls together?
01:16.23*** part/#asterisk opus__ (n=opus@dahphish.org)
01:16.33hardwireyou may want to check out canreinvite
01:16.38chidexhardwire: no
01:16.39*** join/#asterisk denon (i=denon@synapse.subneural.net)
01:16.39*** mode/#asterisk [+o denon] by ChanServ
01:16.52outtolunchttp://www.voip-info.org/tiki-index.php?page=Asterisk+config+features.conf
01:17.03hardwirechidex: well .. weird I guess .. whats the ussie?
01:17.06hardwireissue
01:18.50outtoluncthe whole point i 'tried to make' was that not only does supervised transfer require alot of config, it is/was also known (back when i actually cared) that GS phones with supervised transfer had a lot of issues and various firmware revs 'claimed' to fix it
01:19.00hardwireso here in alaska we have a fund from the gov't for all sorts of reasons
01:19.04hardwireits like $800 this year
01:19.09hardwirethat they just give everybody in alaska
01:19.12chidexWhen I do a supervised transfer between a GS-486 and a softphone (eyebeam), the sip call bridges fine. But as soon as both parties hangup the asterisk server restarts
01:19.25hardwireits annoying.. everybody advertises ways to blow your entire fund
01:19.40hardwirechidex: thats a little odd
01:19.49hardwirewhat does /var/log/asterisk/messages or debug or full say about that?
01:20.06chidexouttolunc: this happens on the softphones as well
01:20.41outtoluncwell then, i'd move to a proactive approach and start your asterisk as /usr/src/asterisk/asterisk -vvvvvvvvgc
01:20.51outtolunc(the 'g' means dump a core)
01:20.59outtoluncand if you have gdb installed
01:21.00hardwirepoop it out
01:21.27outtoluncyou can then gdb /usr/src/asterisk/asterisk /usr/src/asterisk/path to core file
01:21.27hardwireweird.. I didn't realize supervised was so much an issue
01:21.31hardwireI use snom phones
01:21.39hardwireit seems like asterisk takes care of everything
01:21.50hardwireI don't want this call anymore give it to that phone.. OKAY!
01:22.08outtoluncbut you are only gonna get minimal output inless you use make valgrind (which means you need to install valgrind)
01:22.54chidexjust going through the log now
01:22.57outtoluncso, i fear i'm not able to help you <G>
01:24.51wunderkinhomey g dog
01:28.56chidexwhen I try to start asterisk that way I get the following error: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) as opposes to starting it with safe_asterisk script
01:29.32outtoluncthats because you added an 'r'
01:29.49outtolunc[18:20] <outtolunc> well then, i'd move to a proactive approach and start your asterisk as /usr/src/asterisk/asterisk -vvvvvvvvgc
01:29.55outtoluncsee an 'r' in that
01:30.06file[laptop]I see an r in a few places :P
01:30.16outtolunc-vvvvvvvvgc
01:30.20outtoluncin *that
01:30.31chidexthat's what I did
01:30.45Dr_Ray-vvvvvvvvvrgc
01:30.59file[laptop]pfft
01:31.00outtoluncNO r
01:31.13outtolunche's starting it and staying IN the console
01:31.15file[laptop]it was VON :P and I was jetlagged.
01:31.20outtolunchehe
01:31.21Dr_Raymy bad
01:31.34outtoluncnp
01:31.39file[laptop]back to Enterforaprize
01:31.41*** join/#asterisk Ridgeback (n=jircii@104.243.8.67.cfl.res.rr.com)
01:31.59Ridgebackhey all
01:32.09outtolunchello
01:32.27Ridgebackanything new today?
01:32.28outtoluncplease insert you visa card for 3 more minutes
01:33.04outtoluncanother change to manager events
01:33.09Ridgebackhmmmm
01:33.14outtoluncother than that, not much
01:33.50Ridgebackim building anew kernel of abuddy of mine. throwing in QoS, and runnign wonderhsaper. its really helped out here for my * switch
01:34.00outtoluncinduce, invalids, create errors, mod, mod mod mod, todays edition
01:34.59Ridgebackhave you played with the new AEL?
01:35.05outtoluncme, naw
01:35.08[hC]any of you guys using a sangoma card?
01:35.24Ridgebacknope
01:35.54outtoluncnot me, i got t400p (not the tdm one), and te410pv2
01:36.11*** join/#asterisk criptos (n=criptos@201.138.231.2)
01:36.21Ridgebacki had a zaptel card for fxo work... dropped it for voipjet
01:36.38criptosvoipjet? what is that?
01:36.51outtoluncan iax provider
01:36.55Ridgebackits a good iax voiprovidor
01:37.07outtoluncnotes nufone is also
01:37.10Ridgebacki use them for all my pstn work now
01:37.24Ridgebacki used nuphone for incoming ,VJ for outgoing
01:37.41outtoluncif you did, why is it so hard to spell <G>
01:37.53Ridgebackim crappy typer!  ;)
01:37.56outtolunchehe
01:38.00outtoluncnp
01:38.22outtoluncheck i wasn't sure there wasn't *another provider with that name
01:38.43Ridgebackproabably 1000's of them by now.
01:38.53outtoluncnods
01:39.00Ridgebackafetr asterisk became mainstream micro telcos popped up everywhere
01:39.09outtoluncthey come and go like the proverbial packet
01:39.32Ridgebackno kidding with no "ack" when you want your money back
01:39.33Ridgebacklol
01:39.40outtoluncmyself, i've used nufone.net for like 3 years now, and voipjet for 1
01:39.49Ridgebackyep those two work great
01:39.57Ridgebacknufone for 1800 is awesome
01:40.26outtolunci provisioned 2 866's in like 2 minutes last night for testing
01:40.26*** join/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted)
01:40.26*** mode/#asterisk [+o twisted] by ChanServ
01:40.32Ridgebackdo yo do asterisk for work or just for fun?
01:40.40outtoluncme?
01:40.47Ridgebackyes
01:41.04outtolunchehe, well i've not made *any* money on asterisk
01:41.10Ridgebackhahah
01:41.12outtoluncbut it's not been fun <G>
01:41.14JunK-Ywe have to choose between masturbation or asterisk, we all opt for asterisk :P
01:41.23outtolunci just do mod's
01:41.27Sk3tChon authenticate what is the "enter" key after passworD?
01:41.32Sk3tCh# ? * ?
01:41.44outtolunc#
01:41.47Ridgeback#
01:42.02Ridgebackouttolunc: not fun working with asterisk?
01:42.05twistedpound
01:42.07twistedbang
01:42.08twistedslash
01:42.18twistedbang
01:42.19twistedbackslash
01:42.32outtoluncwell i have a day job, i've been working with asterisk for 3+ years
01:42.33Sk3tChit is an ip telephone
01:42.35Ridgebackouttolunc: i have 5 switches in my private network, about 30 people use it constantly...have had 0 problems. super stable.
01:42.49outtolunci *use asterisk all the time
01:43.01outtolunci make mods where i can/want too
01:43.09outtolunci give give give
01:43.14Ridgebackouttolunc: you write your own applications?
01:43.22outtoluncbut for some reason, noone gives back <G>
01:43.46SarahEmmi give back outtolunc!
01:43.51outtolunchehe
01:43.54outtoluncty
01:44.14twistedoooh
01:44.15QwellBUAHAHAHA
01:44.18twistedyou expect us to PAY you
01:44.20outtoluncyep, 1 cookie deposited
01:44.33outtolunchehe
01:44.37outtolunci never have
01:44.41outtoluncand never will
01:44.41twistedi know
01:44.42twisted;)
01:44.43*** join/#asterisk menger (n=menger@dsl-53.69.240.220.rns01-dryb-mel.dsl.comindico.com.au)
01:44.51file[laptop]uh oh it's Matt
01:44.52Ridgebackouttolunc: I breifly experimented with building a simple app for my own education. I got my app to install, compile, even spit stuff out to the event log. but trying to play with a rtp stream, or simply dialing a number got me all confused.  couldnt find much help out there so i quit :(
01:45.01twistedoh this is great
01:45.07Ridgebackwhats wrong?
01:45.09twistedlast night i disconnected all loud noise making devices
01:45.13twistedand today
01:45.17twistedmy mac decided to take up the slack
01:45.20outtolunci do it because after 28 years of 'doing this shit for a living' it's fun (when it works)
01:45.21Ridgebackoh cra
01:45.22jskcrhya twisted how are ya
01:45.30twistedjskcr, living
01:45.32Dr_Raywhat mac do you have?
01:45.35jskcrya same here
01:45.41twisteddual G4 1.25
01:45.44jskcranyone here using astbill?
01:45.48Sk3tChi pushed the # but i dont grant access
01:46.09outtolunctwisted you didn't disconnect the gf again did you <G>
01:46.17twistedi have a gf?
01:46.24Qwellgf, bf, whatever
01:46.26outtolunc[18:45] <twisted> last night i disconnected all loud noise making devices
01:46.28Qwellits all the same in #asterisk
01:46.34twistedno
01:46.37twistedi do not have a bf
01:46.39outtolunck
01:46.41twistedlet alone a gf
01:46.43Dr_Rayasterisk is a harsh mistress
01:46.45Ridgebacklol
01:46.56twistedalthough i'm working on the latter ;)
01:47.16Ridgebackasterisk is a bitch we all love :)
01:47.22twistedi almost have enough allison prompts to make dial-a-girlfriend
01:47.28outtolunchehe
01:47.45Ridgebackso what are the top ten reasons * is better than a GF?
01:47.46Dr_RayI want to pay allison to record "Who's a dirty boy?"
01:47.46Ridgeback1.
01:47.49outtolunci wonder how much 'unzip your pants' cost <G>
01:48.22twistedwell, i have "unzip" in one phrase, and "your pants" in another
01:48.24Ridgeback1. your * switch doesn't mind you talking to other * switches
01:48.27twistedso I COULD just merge the two
01:48.28outtoluncthis is getting alittle sideways, even for #asterisk <G>
01:48.43mengerlol
01:48.44outtolunchaha
01:48.49outtolunchey menger
01:48.51mengerhi otl
01:48.55twistedomg this is funny
01:48.57Ridgeback2. you can hang up on asterisk to stop listening to the damn thing complain!
01:49.02twistedon friends they're trying to get a sofa upstairs
01:49.05twistedand outside, so are they
01:49.26*** part/#asterisk criptos (n=criptos@201.138.231.2)
01:49.48mengerotl: is fine, i just came on to see who is about
01:50.12QwellWhy don't we just get Allison to read a dictionary?
01:50.25Ridgebackget her to read a porno!
01:59.41*** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com)
02:02.43UberbotWho's Allison?
02:02.50tzangerhttp://bash.org/?111338
02:03.24UberbotHuh?
02:03.40SarahEmmallison:
02:03.41SarahEmmhttp://members.shaw.ca/allisonsmith/html/index.htm
02:03.42hypa7iaUberbot: allison is the "voice of asterisk"
02:04.00UberbotMMmmmm Allison... :-D
02:05.00FuriousGeorgehi all
02:06.05*** join/#asterisk cio (n=na@adsl-072-149-159-016.sip.bhm.bellsouth.net)
02:06.24hypa7iajbot_: Allison is The IVR Voice, http://theivrvoice.com/ and http://thevoice.digium.com/
02:06.38hypa7ia~Allison
02:07.04FuriousGeorgei got this remote sip client on one of my asterisk rigs, and everything works fine in terms of nat and bandwidth, however when the remote sip user transfers a call he uses the transfer button on eyebeam, which appears to cause the client to bridge the call (not *)
02:07.21FuriousGeorgethis is problematic when the call is being bridged for you accross the internet (the client is remote)
02:07.32hypa7iadrumkilla: why won't jbot listen to me
02:07.33hypa7ia:-(
02:07.41drumkillai'm sowwy
02:07.49drumkilla~thwack hypa7ia
02:07.50jbotACTION bludgeons hypa7ia on the forehead with a AS/400
02:08.03*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfl2l.dialup.mindspring.com)
02:08.28drumkilla~Allison
02:08.29jbotsomebody said allison was The IVR Voice, http://theivrvoice.com/ and http://thevoice.digium.com/
02:08.29FuriousGeorgei kknow app_dial does transfer, ive exerimented with T and t in the dialplan by the dial command
02:08.46hypa7iayay!
02:08.52drumkillahypa7ia: it worked when i msg'd it ...
02:08.59FuriousGeorgedoes anyone know wtf im talking about?
02:08.59hypa7iaahh
02:09.01*** join/#asterisk ronaldl79 (n=chatzill@c-24-8-54-203.hsd1.co.comcast.net)
02:09.08ronaldl79Hey gys
02:09.09ronaldl79guys
02:09.10hypa7iadoesn't work if you just tell it in the channel?
02:09.11FuriousGeorgehi
02:09.17ronaldl79Anyone having termination issues with BroadVoice today?
02:09.20drumkillahypa7ia: no clue ... maybe it just doesn't like you
02:09.40ronaldl79I seriously doubt my configuration is at fault...
02:09.49ronaldl79Nothing has changed....
02:10.10ronaldl79In other news, I just loaded Cepstral TTS on my box...sounds great!!!!
02:10.21SarahEmmhehe
02:10.23SarahEmmcepstral is good :)
02:10.28ronaldl79Indeed
02:10.31SarahEmmi just wish they had a linux/xscale version :P
02:10.36ronaldl79I'm not sure which voice I'm going to register....
02:10.36SarahEmmor *anyone* other than festival did :P
02:10.56ronaldl79David sounds alright, I'll need to listen to those demos again.
02:11.06SarahEmmand am using TTS, anyway.
02:11.07ronaldl79I think there's this one English dude who sounds REALLY good...
02:12.18FuriousGeorgeiow, it appears that when i take a call from the pstn on my remote client and transfer it, the call goes fromL  called_party <-> pstn <->  <-> remote client <->  <-> PSTN <-> tranfered party.  i would like asterisk to be in the middle not remote client.  does that make sense?  if so, any way to correct
02:13.02tzangeryour voice is linda when you can't talk?
02:13.25SarahEmmtzanger: cepstral linda.
02:13.39SarahEmmwhen i use TTS, which isn't often due to.. form factor issues
02:13.51*** join/#asterisk dudes (n=dudes@12-215-34-84.client.mchsi.com)
02:13.57tzangerah
02:14.21SarahEmmnobody makes a good TTS system for strongarm/xscale, and typing on a laptop out in public isn't easy.. </whine>
02:14.51tzangerSarahEmm: is TTS heavy floating-point?
02:15.00SarahEmmi have no idea :)
02:15.01SarahEmmit's possible
02:15.10SarahEmmbecause multiple mfgs make it for WinMobile/xscale
02:15.11SarahEmmbut not linux
02:15.22SarahEmmi.e. cepstral...
02:15.39SarahEmmi asked them about a linux/xscale port, they evidently have one but aren't interested in helping people doing one-off projects
02:15.51FuriousGeorgeis it just that remote sip clients shouldnt be transfering calls?
02:15.59InfraRedVI VI VI - the editor of the beast!
02:16.24*** join/#asterisk Goshen (n=Goshen@c-67-172-238-57.hsd1.ut.comcast.net)
02:18.34chidexstrange problem, I called my asterisk box using my mobile phone via one of my sipgate incoming trunks. But I was able to do
02:18.34chidexa transfer from my mobile phone (pressing *1), thus allowing me to dial which ever number I wanted throught the asterisk server. Any suggestions
02:18.34chidexin disabling remote callers access to the call transfer ability?
02:19.21FuriousGeorgeare you using the ,,T switch with dial
02:19.22FuriousGeorge?
02:19.41FuriousGeorgeare your calling the wrong zap channel (thereby sending the call to an fxs instead of an fxo?
02:19.44FuriousGeorge)
02:20.08*** join/#asterisk Moc (n=mochouin@modemcable173.101-70-69.mc.videotron.ca)
02:22.21chidexI'm calling the DID number assigned to me by sipgate which in turn is registered on my box
02:23.53*** join/#asterisk ManxPower (n=eric@adsl-67-65-233-194.dsl.lgvwtx.swbell.net)
02:26.02*** join/#asterisk Strider_ (n=aalmenar@gentoo/user/strider)
02:26.39*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
02:27.23Strider_is there any way to know the size of a variable in extensions.conf ?
02:27.32Strider_like .size() in java ?
02:27.39mog_homelength
02:27.45mog_homeor something like that
02:27.52mog_homeits in README.variables
02:27.55mog_homeor something like that
02:28.01Strider_ive been reading all the docs
02:28.12Strider_but i cant find about it
02:28.22Strider_any pointer on where to look ??
02:28.26mog_homeone sec
02:28.30mog_homeill grab it for you
02:28.30chidex<FuriousGeorge> : sorted it out
02:29.15*** join/#asterisk esandeen (n=sandeen@sandeen.net)
02:29.22esandeenhey all, hope this isn't too OT - tried my first VOIP experiment w/ a PAP2 and sipphone.com - HORRIBLE latency.  This is over DSL... any suggestions on how to isolate the problem?
02:29.27tzangerStrider_: $LEN(${VAR})
02:29.27tzangerI think that's how it's written
02:29.41Strider_tzafrir, thanks, i will try it
02:29.55ManxPoweresandeen, horrible latency or jorrible jitter?
02:29.57Strider_tzanger, thanks !
02:29.59brimstoneStrider_: ${LEN(VAR)}             * String length of VAR (integer)
02:30.08brimstonecp'd from docs/README.variables
02:30.16esandeenManxPower, well I think latency... talk (wait) listen (wait) talk (wait)
02:30.29*** join/#asterisk jskcr (n=jskcr@unaffiliated/jskcr)
02:30.35Strider_brimstone, maybe i have an old
02:30.42Strider_README.variable
02:30.44ManxPoweresandeen, there could be a zillion possible causes.
02:31.02brimstonemine's from CVS HEAD from a week or so agao
02:31.16ManxPowerping the host you are doing voip with.  If ping times are less than 300ms then it's not a network issue
02:31.17esandeenManxPower, figured.  Any idea how to narrow it down?  I think I heard DSL does some sort of packet scattering that makes it much worse
02:31.42ManxPoweresandeen, buffering on DSL modems only kicks in when the link is saturated.
02:32.02ManxPowerpeople run VoIP over DSL all the time with only minor issues, as long as the link is not saturated.
02:32.06mog_homethere it was thanks brim
02:32.14esandeenManxPower,  hm, ok, and mine's not  :(
02:32.17brimstonenp mog_home
02:32.19esandeenso wonder what it is.
02:32.24mog_homewell just like i said
02:32.27mog_homebut no probleem
02:32.33esandeenmaybe one of the 300 settings on the pap2 that I did not set :)
02:32.56ManxPoweresandeen, An ISP CAN set very large buffers on their routers and if their uplink is overloaded things will suck
02:33.00esandeenManxPower, so am I correct to call this a latency problem, if it's perceived as pauses in the conversation...
02:33.14ManxPoweresandeen, Correct, as long as the audio actually sounds fine.
02:33.35esandeenManxPower, ok, thanks.  will keep digging.  My DSL provider offers (outrageously priced) VOIP so in theory they must make it work ok :)
02:33.44ManxPoweresandeen, also try a different provider.  There are many of them out there you can sign up for for a small amount of money
02:34.27ManxPoweresandeen, if your provider offers voip they COULD set things up so anyone elses voip sucks.  only heard of that happening once, however (madison river clec/isp)
02:34.58esandeenManxPower, yep I've heard of such a thing.  can't tell what sort of device they use - not a pap2/spa-2000 sort of thing, looks like
02:36.09ManxPoweresandeen, think of it as a problem to research, in finding the cause and solution you may have to learn massive amounts about networking, voip, routers, QoS, and packet latency.  All good things to know.
02:36.38jskcresandeen:  or find a bad cable
02:37.09esandeen:)
02:37.33esandeenI'll trade filesystem knowledge for voip knowledge :)
02:37.56brimstoneonly if that filesystem knowledge results in a ext2 driver for tiger...
02:38.03pegger<PROTECTED>
02:38.04esandeenbrimstone, sure.
02:38.05pegger???
02:38.29esandeenbrimstone, oh, does it not work in 10.4?  had one for 10.3.  and  a co-worker helped write that one :)
02:38.37brimstonenot in 10.4
02:38.41esandeenah
02:38.44brimstone<10.4 works great
02:38.45mog_home<PROTECTED>
02:38.51esandeenyeah I heard 10.4 changed quite a lot of kernel apis
02:38.52brimstonestill only one thread
02:38.52mog_homebig one
02:38.56esandeenok sorry going OT :)
02:39.06mog_homehalf a thread
02:39.13*** join/#asterisk jarnaud (n=jarnaud@c-67-191-4-38.hsd1.fl.comcast.net)
02:39.16jarnaudhello all
02:39.23brimstonesup jarnaud?
02:39.31jarnaudwhat?
02:40.12brimstonei dunno
02:40.59jarnaudI'd like to do a system(stupid_cmd.sh) in the dialplan and use the result of the command. Is it possible?
02:41.14brimstonejarnaud: AGI
02:41.28brimstoneat that point, you want AGI
02:41.33jarnaudAGI is cool
02:41.43jarnaudBut I would have to recode everything in AGI
02:42.05mog_homeyes you can do that
02:42.08*** join/#asterisk hellagony (n=hellagon@62-43-8-76.user.ono.com)
02:42.15mog_homeyou could have it write a text file
02:42.16brimstonewhy not just make a perl, or sh for that matter, to the stupid_cmd.sh ?
02:42.19mog_homeand use readfile
02:42.24brimstoneor write a call file
02:42.29mog_homeno
02:42.38Dr_Raycall files rule
02:42.39mog_homei mean there is an app to read a text file in asterisk
02:42.45jskcrfestival
02:42.50mog_homeso just have it write results to a call file
02:43.00mog_homewith same uniqueid as channel
02:43.02mog_homeand you will be fine
02:43.17jarnaudmy stupid.sh is a mysql call actually
02:43.24jarnaudthat return a number
02:43.34jarnaudthat I'd like to have in a dialplan variable
02:43.34mog_homeso have it write that number to file
02:43.40mog_homellike this
02:43.58mog_homeexten =>1234,1,system(blah.sh > ${UNIQUEID})
02:43.59mog_homeand then
02:44.17mog_homeexten =>1234,2,readfile(${UNIQUEID})
02:44.22jarnaudyou rock !
02:44.24mog_homeor that isnt syntax for it one sec
02:44.47mog_homeReadFile(varname=file,length)
02:45.07jarnaudthanxalot
02:45.38mog_homewow its been a while sense i wrote that
02:46.05brimstonehaven't been to english lately either huh?
02:46.37file[laptop]eep
02:47.09mog_homelol
02:47.12mog_homei speak no english
02:47.24brimstoneit shows
02:47.26brimstone:P
02:48.22*** join/#asterisk likwid-- (n=likwid@nc-71-0-190-14.dyn.sprint-hsd.net)
02:49.25jarnaudare u sure it's cmd readfile()?
02:49.41jarnaudast1.2?
02:50.45mog_homeits in my ast head
02:50.51mog_homei wrote it months ago
02:51.25jarnaudok I see
02:51.32jarnaudso you're the man!
02:51.54mog_homehmm maybe
02:51.59mog_homenot till i write something big
02:52.20JunK-Yand THE MAN has posted message for ducks? :P
02:53.20mog_homelol
02:53.21mog_homeeep
02:53.50JunK-Yive checked on google earth seems to be a bit far, but who cares, we'll have fun down there.
02:55.37mog_homeowww
02:55.42mog_homewhat was that for
02:55.48*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
02:55.50Corydon76-homeFor the letter
02:56.00*** join/#asterisk pussfeller (n=todd@12.150.129.171)
02:56.14mog_homethe letter?
02:56.21Corydon76-homefrom Digium
02:56.33mog_homefrom digium?
02:56.35Corydon76-homeI can read about 2 of the signatures
02:56.47*** join/#asterisk rking (n=rking@ip68-105-231-56.lu.dl.cox.net)
02:57.06Corydon76-homeMalcolm is about the only one whose signature is legible
02:57.16mog_homesignatures from what?
02:57.34Corydon76-homeMark sent me a thank you note for working on Asterisk
02:58.52mog_homeoh groovy, you do rock
02:59.05drumkillayay Corydon76-home !
02:59.09tzangernice
02:59.13Corydon76-homeHeh
02:59.17JunK-Yya, thats cool.
03:00.20Corydon76-homeNow it's just a matter of trying to get 1.2 out... ;-)
03:00.35denonany of you guys use linux pppd much?
03:00.40QwellYou guys have like...9 days.  You can do it! :D
03:00.40denonas a client, I mean
03:01.08Corydon76-homeAnyway, I've been trying to figure out who all signed it... since the signatures are illegible...
03:01.32mog_homeahh
03:01.36mog_homei think i could have
03:01.42mog_homebut mine would be an m or an o
03:01.47mog_homewith a squigly line
03:02.01*** join/#asterisk Weezey (n=chatzill@206.210.109.226)
03:02.46Corydon76-homeOh, I see one that could start out as "Matt"
03:02.59Corydon76-homebut that could also be Matt F
03:03.25Weezeywhen using call forwarding, how do I make it go out another context?  The only way I can figure right now is to prefix the number,  like _4NXXNXXXXXX, Dial(${EXTEN:1}
03:03.42mog_homeheh
03:04.06jskcrgoto
03:04.50Corydon76-homeThere's one signature that kind of looks like "Zach"
03:04.54jskcrexten => _4NXXNXXXXXX,1,Goto(context,s,1)
03:05.02mog_homelol
03:05.02Corydon76-homeor I suppose it could also be "John"
03:05.09mog_homethere is a john
03:05.35Corydon76-homeand it's also possible that neither of those are correct
03:05.36Weezeyjskcr: right, but is there a way to make calls forwarded from a set go out a different context?
03:06.06Weezeyjskcr: without having to prefix it.
03:07.04jskcrweezey set a var and then check for it
03:07.15jskcrthen use gotoif
03:07.34NiHodial(local/extension@context) for crying outload
03:07.59WeezeyNiHo: it does that automatically.
03:08.10Weezeyit forwards fine.
03:08.22NiHothat uses the local cannel to call the extension in the specified context
03:08.29Weezeyit just goes out as if it's the caller, thus added my caller ID code.
03:08.54NiHothe dont answer it and just dial() it direct
03:08.56QwellCorydon76-home: you'd never be able to make out my signature...
03:09.01WeezeyIf I check to see if it's local vs. SIP that might do the trick though....
03:09.35*** join/#asterisk jselect (n=kvirc@dsl-210-15-200-169.QLD.netspace.net.au)
03:09.40NiHoyou dont have to send it out the context the peer is in, you can dial(FOO/bar) from any context
03:10.11NiHoeven if FOO/bar is in context bar and the dial is in context foo
03:10.24WeezeyNiHo: agreed, but if you press Cfwd on a SPA or Linksys, it sends it to Local/#@your context automatically
03:10.34*** join/#asterisk _Thor (i=Christia@user-vc8fl7l.biz.mindspring.com)
03:19.51*** join/#asterisk dudes (n=dudes@12-215-34-84.client.mchsi.com)
03:26.15_Thorhello
03:28.02Weezeysolution: _NXXNXXXXXX,1,SetAccount(USERNUMBER)   _NXXNXXXXXX,2,GotoIf($[${CHANNEL:0:5} = Local]?4:3)   _NXXNXXXXXX,3,Goto(normaloutgoingdial,${EXTEN},1)   _NXXNXXXXXX,4,Goto(callforwardeddial,${EXTEN},1)
03:28.58WeezeyComedian Mail.  Mailbox?
03:29.18*** join/#asterisk mhnoyes (n=mhnoyes@user-38lc0ba.dialup.mindspring.com)
03:29.34*** join/#asterisk tclineks (n=tclineks@ppp-70-243-238-201.dsl.tpkaks.swbell.net)
03:29.34SarahEmm1234
03:29.36SarahEmm;)
03:29.40QwellPassword?
03:29.49SarahEmm1000
03:30.05QwellYou have three hundred fourty-five, new messages.
03:30.18Weezey1
03:30.24tclinekshello, asterisk newbie here, just installed (recompiling kernel for zaptel atm) was wondering about any suggestions for a slick oss web gui
03:30.57WeezeyOld Skanky Slut gooey, eh?
03:31.23tclineksWeezey: that's the one =)
03:32.50nestAr:o
03:33.18mog_homethere is no very good oss web gui
03:33.28mog_homethe best is Asterisk management portal
03:33.43mog_homebut it is my opinion it doesnt create human readable or editable file
03:33.44mog_homes
03:33.56drumkilla~amp
03:33.58jbotrumour has it, amp is an Audio MPEG Player.  [non-free], or http://amp.coalescentsystems.ca/
03:34.16mog_homethat is just wrong
03:34.20mog_homebut the link is right
03:34.21blitzrageyah
03:34.25SarahEmm~amportal
03:34.27Qwellits an or
03:34.27outtolunc~amp'd
03:34.40drumkilla~mog
03:34.41jbotsomebody said mog was half man, half dog.  A mog is his own best friend.  The most famous Mog is, of course, Barf.
03:34.46Qwellperhaps the order should be swapped
03:34.53drumkillamog_home: hahaha
03:35.01SarahEmm~sarahemm
03:35.03jbot[sarahemm] kitrichernesses.
03:35.04mog_homebut i prefer the mog from ff6
03:35.06SarahEmm*giggles*
03:35.07SarahEmm~kitrich
03:35.09jbotkitrich is, like, a cross between an ostrich and a kitten
03:35.17Qwellmog_home: yeah, that mog is way more famous
03:35.17drumkilla~drumkilla
03:35.18jbotsomebody said drumkilla was Russell Bryant, the Asterisk release branch maintainer <russelb@clemson.edu>
03:35.22drumkillayay
03:35.24*** join/#asterisk Micc (n=dotirc@c-24-19-175-112.hsd1.wa.comcast.net)
03:35.25blitzrage~blitzrage
03:35.26jbotit has been said that blitzrage is a super cool fellow
03:35.29Qwell~qwell
03:35.36outtoluncwaits
03:35.36Qwellscrew you guys
03:35.36mog_homeawww
03:35.39blitzragelol!
03:35.40outtoluncand waits
03:35.44mog_homepoor qwell
03:35.47Qwell:p
03:36.15MiccAre there problems with audio on older versions of asterisk? I've got audio problems when I use nufone for any period of time.
03:36.24drumkilla~qwell
03:36.25jbotqwell is probably one crazy mofo.  he'll bite your head off.  WATCH OUT!
03:36.32Qwelldrumkilla: <3
03:36.39outtolunchehe
03:36.41blitzrage./msg jbot qwell is not quite a quill, but close
03:36.44QwellMicc: I've heard there are problems with iax stable-head
03:36.46*** join/#asterisk opus__ (n=opus@dahphish.org)
03:36.50blitzragedrumkilla: you beat me too it :)
03:36.51opus__How do I turn on QoS on the SIPURA?
03:37.01file[laptop]blitzrage: are you the phantom of the opera?
03:37.04blitzrageopus__: 1) Pick up said device
03:37.06MiccQwell, so even latest version has problems?
03:37.11blitzrageopus__: 2) Open window
03:37.20blitzrageopus__: 3) Toss said device out window
03:37.20QwellMicc: no, I mean, when going from stable to head
03:37.29opus__blitzrage are you serious?
03:37.40blitzrageopus__: 4) Now thats some quality service
03:37.43file[laptop]VoIP it baby, like you know you should!
03:37.55MiccQwell, ohhhhh! Maybe that is what happened to my first installation!
03:38.08opus__does SIPURA even support the ToS bit?? what gives
03:38.15MiccQwell, so if I start a new version on a new machine from cvs head right now it should be good?
03:38.29blitzrageok... bed time... need to be productive tomorrow
03:38.47Qwell~qwell
03:38.49jbotwell, 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.
03:38.51Qwell:D
03:39.23mog_homeheh
03:40.05opus__is there any device that can magically set the ToS bit? Like, it sits between the NIC cards
03:40.20drumkillaopus__: iptables
03:40.23opus__like the second NIC port on the polycom ip-500
03:40.29opus__I meant, hardware device
03:40.32drumkilla... or asterisk can do it ...
03:40.46drumkillawell that'd be the sillyest piece of hardware i had ever seen
03:40.51opus__i can't believe SIPURA doesn't support QoS
03:40.55SarahEmmnini
03:41.24ronaldl79Any BroadVoice users?
03:42.21tclineksmog_home: thanks, AMP was what I has already untarred =)
03:42.24opus__broadvoice sucks ass
03:42.31MiccIs broadvoice still having dtmf problems? Last I heard their provider was having problems with DTMF.
03:42.34mog_homegood luck
03:42.57opus__broadvoice is #1 in bizzaro world
03:43.34tclineksmog_home: i'm not going to use it initially
03:43.47tclineksshould I use mysql off the bat?
03:43.53mog_homeyou will learn more in a few weeks of doing config files
03:43.58mog_homethan ever with amp
03:44.03mog_homedo it with flat files
03:44.08tclineksmog_home: in a few hours i bet
03:44.14mog_homethen learn mysql,postgres or odbc
03:44.25mog_homebut flat files are pretty easy
03:44.46mog_homegood stuff then its easy for you to add support for it
03:45.30tclinekshaha
03:45.36mog_homelol
03:45.43tclineksi'll go without for now
03:49.56tclineksI'm putting this on a server machine that I didn't get alsa/oss configured on -- what amount of the sound processing requires hardware? (i'm guessing it needs to be properly configured)
03:50.35*** join/#asterisk bmg505 (n=leon@rndf-146-33-134.telkomadsl.co.za)
03:52.17mog_homeyeah to use chan alsa or chan oss you need a sound card
03:57.04tclineksmog_home: and it is necessary to use one of those
03:57.12tclineksmog_home: ?
03:57.20mog_homenope
03:57.25mog_homei dont have it on my box
03:57.37mog_homeonly really usefull as an intercom
03:58.40tclineksmog_home: ah, great, thank you
03:58.45file[laptop]I have a craving for pizza
03:58.54mog_homeno problemo
04:00.24*** join/#asterisk esandeen (n=sandeen@sandeen.net)
04:00.35esandeenManxPower, tracked it down to a bad dns setup :)
04:00.53tclineksif I'm missing /dev/zap should I create it ala http://lists.digium.com/pipermail/asterisk-users/2004-December/075521.html?
04:01.09*** join/#asterisk iCEBrkr (i=icebrkr@24.129.130.158)
04:01.24mog_homeread README.udev
04:01.27mog_homein your zaptel source
04:01.46tclineksdoing so solved two WARNING problems =)
04:01.53tclineksmog_home: on it
04:03.27mog_homeheehee
04:05.11file[laptop]eeeeeeeeep
04:05.19Corydon76-homefilefilefilefilefilefilefilefilefile
04:05.23mog_homepoor file
04:05.40twistedoops
04:05.49Corydon76-homeNot poor file, rich file
04:06.18twistedonly after you make him your cabana boy
04:06.27Corydon76-homehahah
04:06.51Corydon76-homeYeah, like I can afford a cabana boy
04:07.11*** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com)
04:07.29JamesDotComman, reading about NANP just makes me want to cry
04:07.37JamesDotComwhy must the US screw everything :|
04:07.47ManxPowerJamesDotCom, Why?
04:07.57ManxPowerJamesDotCom, It's pretty simple
04:07.57JamesDotComjust the complexity involved in it all
04:07.57*** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au)
04:08.01JamesDotComcompared to any other country
04:08.11Corydon76-homeNANP isn't only the US
04:08.20JamesDotComand the inflexibility of it all now that it's been defined
04:08.21ManxPowerHuh?  It's simpled compared to most countries.  i.e. it's NOT variable length
04:08.24Corydon76-homeIt's also Canada, and the Caribbean
04:08.35QwellManxPower: by far the biggest benefit
04:08.36tclineksJamesDotCom: NANP?
04:08.44ManxPowerQwell, Yup
04:08.45JamesDotComnorth american numbering plan
04:08.45Qwell~nanpa
04:08.46Corydon76-homeNorth American Numbering Plan
04:08.47JamesDotComokay
04:08.53Qwelljbot_: you suck
04:08.54Qwell~nanp
04:09.00Corydon76-home~nanpa
04:09.04Qwelljbot_: nanp is North American Numbering Plan
04:09.08JamesDotComi'm looking at it from entirely a programmer writing billing software kind of perspective
04:09.21ManxPowerJamesDotCom, so what is the specific problem?
04:09.21JamesDotComif that helps you understand
04:09.31Corydon76-homeNope, still doesn't
04:09.36JamesDotComnothing specific, just commenting in general
04:09.39Corydon76-homeUnless you're trying to figure out LATA
04:09.44JamesDotComi'm just reading through the spec of it now, heh
04:09.51JamesDotComahh well, nevermind me
04:09.56Corydon76-homeYou can blame the regulators for LATA
04:10.16Corydon76-homeNANP had nothing to do with LATA fees
04:10.38Corydon76-homeintra-LATA connection fees
04:10.59JamesDotComso mobiles within the US still dont have a specific prefix do they? they're all given geographic numbers?
04:11.11Corydon76-homeThat's correct
04:11.17Corydon76-homeWhat's the problem with that?
04:11.18ManxPowerJamesDotCom, Correct.  HOWEVER, calls to movile are billed EXACTLY like a landline
04:11.38ManxPowerThere is NO different rate between calling a landline and a mobile
04:11.56_Thorhelp!! -- anybody familiar with PrepaidCall?
04:11.57*** join/#asterisk digime (n=drooth@ip68-111-235-172.sd.sd.cox.net)
04:13.08JamesDotComdoes the reciever of a call on a mobile still pay for it?
04:13.10ManxPowerCorydon76-home, JamesDotCom is prolly a non-USAian
04:13.19ManxPowerJamesDotCom, Usually
04:13.20*** part/#asterisk opus__ (n=opus@dahphish.org)
04:13.24JamesDotComyeah, i'm in australia
04:13.28JamesDotComsee
04:13.32Corydon76-homeDepends upon the calling plan of the mobile user
04:13.40tclineksunitedstatesian*
04:13.57ManxPowerFree incoming calls is still pretty uncomman in the USA/Canada, but getting more common
04:13.58JamesDotComthe whole reciever-pays idea is foreign to this country
04:14.19Corydon76-homeI get a number of minutes per month, and I never use that many... so all calls are essentially free
04:14.20JamesDotCombecause there's no way to differentiate mobile calls
04:14.24Corydon76-homeincoming or outgoing
04:14.28JamesDotComthat fact alone makes me think it's awful
04:14.28JamesDotComhaha
04:15.08tclineksany advice on `setup_zap: Ignoring switchtype`?
04:15.26Corydon76-homeYeah, we have this thing in the US called privacy...  maybe you've heard of it?
04:15.51JamesDotComeh?
04:15.55Corydon76-homeIt means that you don't know where I am, even if you have my phone number
04:16.05JamesDotCom...
04:16.16Corydon76-homeIt's wonderful...
04:16.30JamesDotComis this sarcasm?
04:17.04Corydon76-homeNo, it's the benefit of callers not knowing what they're calling
04:17.11Corydon76-homeHOWEVER
04:17.14JamesDotComhaha
04:17.24Nuggetmaybe we used to have something called "privacy".  Not so much these days, though.
04:17.31JamesDotCommy mobile number is 0421448XXX
04:17.36JamesDotComtell me where in australia i am :)
04:17.53NuggetYou must be in Canberra, otherwise you'd have better things to do than argue on irc.
04:17.57Corydon76-homeIf you know the NPA and NXX of the number, you can look up what telco originally had the number, and from there, be able to know if it's likely whether the number you're dialling is mobile or not
04:18.53Corydon76-homeOf course, number portability now makes that less certain, nowadays
04:18.59JamesDotComi'm actually discussing it because i don't know much about NANP
04:19.11JamesDotComand trying to see how people dont think it's a mess
04:19.11JamesDotComheh
04:19.18Corydon76-homeGet yourself a copy of the LERG
04:19.46Corydon76-homeJamesDotCom: all of our phone numbers are exactly the same length... that's a feature...
04:20.24nestArFEATURE!
04:20.26nestArnot a bug
04:20.46JamesDotComi guess that's one benefit
04:21.13Corydon76-homeIt means you know, straight off, whether someone gave you a complete phone number or not
04:21.34Corydon76-homeWith variable length, do you ever really know if that beautiful blonde shorted you two digits?
04:23.09JamesDotCombecause unless she's not really the type of girl you want to be calling up, her number will either be a 10 digit mobile number or an 8 digit geographic number, with a 2 digit area code if she's from out of the state
04:23.16JamesDotComheh
04:23.29JamesDotComeven if she did slip you a 1900 number, it's still 10 digits
04:23.37JamesDotComit will just cost you 4.95/m to talk to her
04:25.31Corydon76-homeNow that's something I miss... I wish I could get a 900 number, without having to pay administrative costs of $500/mo
04:26.40Corydon76-homeCould you imagine a mortgage company paying me $5/minute to try to convince me to refinance?
04:29.54Nuggetthat would be great
04:30.25Dr_RayI bet you could find an 809 number
04:31.43*** join/#asterisk lehel (n=asd@82.79.20.17)
04:31.54lehelhey
04:32.00Jam1eexten => s,3,BackGround(press-1)
04:32.00Jam1eexten => 1,1,Goto(sales-context,s,1) ; Pressed 1 for sales, lets go to sales department
04:32.04Jam1ewhy isn't that working?
04:32.08*** join/#asterisk michael1234 (i=michael1@escazu-a236.racsa.co.cr)
04:33.37michael1234how someone help me configure
04:34.28michael1234pri for australia
04:34.36tclinekswoohoo
04:34.42JamesDotCommichael1234: what about them?
04:35.34*** join/#asterisk digime (n=drooth@ip68-111-235-172.sd.sd.cox.net)
04:36.54*** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com)
04:48.31FuriousGeorgeyou know what would be pretty snazzy.  i dunno if asterisk does it already, but could my peers be allowed to make 1 ulaw call and any concurrent call that leaves the network be gsm?
04:49.44*** join/#asterisk maik (n=maik@bfs.cs.uni-sb.de)
04:49.46*** join/#asterisk rajo_ (n=rajo@bfs.cs.uni-sb.de)
04:51.47*** join/#asterisk justinnnnn (n=justinnn@61.95.68.85)
04:51.51justinnnnncan someone help me again pls
04:51.54justinnnnni changed my zaptel.conf
04:51.55justinnnnnfrom this
04:51.55justinnnnnbchan=1-10
04:51.55justinnnnndchan=16
04:51.56justinnnnnto this
04:52.02justinnnnnbchan=1-15,17-21
04:52.03justinnnnndchan=16
04:52.14justinnnnnand now asterisk wont startup.. it crashes when it has to load chan_zap.so
04:52.24justinnnnnwhen i change it back to 1-10 it works fine though :(
04:52.33justinnnnnbut then we only get 10 out of our 20 lines :(
04:54.13Corydon76-homeYou think about running ztcfg ?
04:54.16L|NUXwell
04:54.30L|NUXremove = bchan=1-10
04:54.33*** join/#asterisk loud (n=ariel@cypher.punk.net)
04:54.48justinnnnnya i did run ztcfg... and i did a rmmod/modprove
04:54.49justinnnnnprobe
04:54.57justinnnnnlinux what do  umean ?
04:55.03Corydon76-homeDid you also change your zapata.conf?
04:55.06justinnnnnya
04:55.12justinnnnnits 1-10 at the moment
04:55.19justinnnnni changed it to 1-15,17-21
04:55.24justinnnnnbut that doesnt seem to crach asterisk
04:55.31justinnnnni can make that evan like 1-100000 and its fine
04:55.39justinnnnnits the bchan bit in zaptel.conf that it doesnt like :(
04:55.48L|NUXhmm
04:56.02Corydon76-homeYou're running an E1?
04:56.05justinnnnnya
04:56.08justinnnnnbut we only have 20 lines
04:56.11justinnnnnwe used to have 10..
04:56.35Corydon76-homeYou know you have to shut down Asterisk before you run ztcfg, right?
04:56.37NiHoassuming its a EuroPRI 1-15,17-21 as b 16 as b and 22-31 as unused
04:56.58justinnnnncorydon yip..
04:57.09justinnnnni shutdown asterisk.. at did ztcfg (and sometimes manualy unloaded/loaded module)
04:57.21justinnnnnits definetly loading/unloading ok
04:57.26NiHodoes ztcfg -vvv complain?
04:57.28justinnnnncause it nos when i change zaptel.conf thats why its crash
04:57.29justinnnnnno
04:57.42outtoluncdid you ever stop the system? power off?
04:57.42justinnnnnmy telco said 16 is definetly the D channel
04:57.46justinnnnnno
04:58.02justinnnnnthe system is up and working at the moment.. just with only 10 lines configured :(
04:58.05outtoluncsadly, sometimes it helps
04:58.15justinnnnnu serious :(
04:58.22outtoluncumm YEAH
04:58.28justinnnnnservers only been up for like 2 days tho
04:58.41justinnnnnill try reboot 2nite tho :)
04:58.44justinnnnnany other ideas >
04:58.48justinnnnn?
04:58.57Corydon76-homeYou could also try posting your zaptel.conf and zapata.conf on http://pastebin.ca/
04:59.55outtoluncwell there you have it
04:59.57Corydon76-homepowerdown sometimes fixes things, but I'd prefer to know what the problem was
05:01.56outtoluncand getting info from a user that didn't at least try that is 'that much better, how"
05:01.58justinnnnnok ppls
05:02.00justinnnnnhttp://pastebin.ca/24559
05:03.06NiHoscrew OO
05:03.10Corydon76-homeTry posting the ENTIRE zaptel.conf
05:03.31Corydon76-homeAbbreviating doesn't help... sometimes you have a typo that you miss when you abbreviate
05:04.08Corydon76-homeand the ENTIRE zapata.conf
05:04.13*** join/#asterisk syle (n=blag@unaffiliated/syle)
05:04.25justinnnnnthat is the entire file...
05:04.29justinnnnnit is...
05:06.49Corydon76-homewhat happens when you set it to 1-15?
05:07.44QwellI'm finally gonna be setting up a * box at work.
05:07.57Qwellgonna be using a...umm...that other crappy PBX
05:08.15QwellNortel maybe?  The voicemail system is called Octel
05:08.53Qwellhopefully I can get them to give us a T1, heh
05:08.57tclineksahh taht sweet demo voice =)
05:11.39*** join/#asterisk mthem (n=merlin@64.235.245.133)
05:12.30mthemhey guys, have a little problem, im trying to detect a fax call via a context and not zapata.conf, is this possible?
05:12.39*** join/#asterisk KaBewM (n=kabewm@66-215-7-106.dhcp.psdn.ca.charter.com)
05:13.04outtoluncsure, use another app like backgrounddetect or nvfaxdetect
05:13.06mthemi have to termination options, IAX for voice, and g2 (POTS) for fax,
05:13.29*** join/#asterisk LostFrog (n=reallyno@dsl093-100-093.wdc2.dsl.speakeasy.net)
05:13.41mthemi cant use the Answer() and then a way to send the call to either termination group
05:14.00outtoluncwell, then i think you have a serious prob
05:14.25*** join/#asterisk debeast (n=icechat5@rrcs-24-39-201-178.nys.biz.rr.com)
05:14.27mthemthat was a question ... ;)
05:14.31debeasthello
05:14.34mthemsry
05:14.37LostFrogI just plugged in my snom 360, and it has no way to enter authentication information for the sip registry.
05:14.47LostFrogAnyone run into this before?
05:15.08JamesDotComthe admin interface doesnt work?
05:15.39LostFrogvia the web, yes.. But, it doesn't list Line1 Line 2.. etc like it show in the manual.
05:15.40mthemdoes it have a mediatrix approch? with text files and TFTP only?
05:15.45debeastcan someone please explain pbxbox*CLI> Oct  4 01:09:36 NOTICE[1260]: chan_iax2.c:3918 register_verify: No registration for peer 'guest' (from 192.168.1.130)
05:15.55JamesDotComLostFrog: which version of firmware?
05:16.00LostFrog4.1
05:16.05JamesDotComyou need to do some stupid registration thing on snom.com
05:16.11JamesDotComput in the mac address and your email
05:16.14JamesDotComit will email you a key
05:16.17JamesDotComyou upload to phone
05:16.21JamesDotComit unlocks lines
05:16.44LostFrogHmm.. the website says if it came with 4.1 you don't need to, but, thanks.. I will try it.
05:17.34debeastwhy cant i get this softphone to register with the asterisk server
05:18.00LostFrogYou are not required to install a license after upgrading to the 4.x firmware as the phone has already a license build in. If you phone shows 'SIP disabled' on the screen after upgrading to 4.x please write a mail to phonelicense@snom.com
05:18.15LostFrogThat's what I get if I try to get a license from the webpage.
05:18.18outtoluncmthem, if you actually meant you 'want' to use answer.. then do that with another app
05:18.41outtoluncthe way i first read it was you 'can't use answer'
05:20.04outtolunc[22:13] <mthem> i cant use the Answer() and then a way to send the call to either termination group .. was the reason i read it that way
05:20.04LostFrogCome to think of it, there is no "Software Update" link on the web interface. :(
05:20.37outtoluncthe point is you CAN, and you can reset the cdr's along the way (IF that is your issue)
05:21.02*** join/#asterisk neil_ablang (n=neil@202.124.128.39)
05:21.02mthemouttolunc: sorry bout that, ya i was hopeing there was a way to make asterisk detect fax calls from extensions.conf by throwing all fax and fax/voice calls into a "faxdetect" context that would either send the call to IAX or POTS for termination depending on if it detected fax tone
05:21.20outtoluncwhat devices?
05:21.33mthemi dont wanna use fax detect on all lines since it adds 3-4 sec to termination time
05:21.49outtoluncare these zap?
05:22.07mthem4 CBs with lines to customers, termination through IAX for voice and TDM400P for fax
05:22.25mthemthe CBs are connected to a TE405P
05:22.25outtoluncso the detection isn't in zap
05:22.39mthemi would like it not to be
05:22.45debeasthow do i get the demo config working i have idefisk running and it wont allow me to register with the asterisk server ??
05:22.45outtoluncwell it is
05:22.53outtoluncand thats actually good
05:23.02mthemso the Answer() will do me no good?
05:23.09outtoluncin zapata.conf when you are setting you channels
05:23.14_Thorhelp
05:23.35_Thoranybody knows how to compile asterisk add-ons???
05:23.40outtoluncjust set the faxdetect for the channels you want, then turn off
05:24.19mthemi cant make Answer() output something that will let me change termination path based on fax tone or not... that sucks
05:24.51mthemit would be so much easier to only set the context for fax or non fax
05:25.43debeastwho designed this switch software to be so headache causing
05:25.44mthembut if i do it through zaptel, then how do i send it oneway or the other?
05:25.59LostFrogok.. I'm stoopid
05:27.26outtoluncmthem: for that you need to do MORE than answer, but if you enable faxdect=incoming/outgoing on just those channels, thats a diff story
05:27.33justinnnnnwhen i do 1-15 asterisk still crashes ?
05:27.52outtoluncre faxdetect
05:28.27*** join/#asterisk wolfson` (n=hehe@usr-kdh-208-6-58-26.beachlink.com)
05:28.30mthemouttolunc: that is a seperate module and not included in the CVS HEAD, right?
05:29.05outtoluncmeaning, group=1, faxdetect=incoming, yadda group=2, faxdetect=none, yadda
05:30.10outtoluncthat is cvs-head stuff
05:30.37outtoluncnote you would define channels in between
05:31.13mthemouttolunc: thanks that was very helpful! ya ill have to have an option for all channels since they are all mixed between eachother
05:32.25outtoluncjust remember for it to come together you need an exten => fax,1,yadda in the context for those zapata channel ranges
05:32.39mthemouttolunc: just wishing it could be done through a context.... would be so much easier to handle in my management software
05:32.51outtoluncread ^
05:33.32JerJerdoes faxdetect need to have the line answered for it to work?
05:33.35*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
05:34.08outtoluncnot to my understanding, but when did that ever come into play <G>
05:34.13*** join/#asterisk syle (n=blag@unaffiliated/syle)
05:34.33JerJersomeone asked a couple days ago, nobody had an answer
05:34.42outtolunci'm assuming it MUST
05:34.54JerJerwhat about early media on say PRI /
05:34.55JerJer?
05:34.58outtoluncelse how would a dsp get on it
05:35.35outtoluncthat i have to look at
05:35.38justinnnnnanyone got any ideas ?
05:35.53justinnnnnwhy doing 1-15,17-21 in zaptel.conf would make asterisk die ?
05:36.04debeasti cant get asterisk to register my iax soft phone  can anyone take a second to help me please
05:37.08outtoluncfyi:
05:37.11outtoluncdsp.c:           int digitmode, int *writeback, int faxdetect)
05:37.11outtoluncdsp.c:                  (faxdetect)) {
05:37.20mthemjustinnnnn: what does your zapata.conf look like?
05:37.27mthemis 16 defined?
05:37.35outtoluncso, the answer would be, yes, it needs to be answered
05:38.09outtoluncunless someone came up with a way to get a dsp rockin on a NON-answered asterisk channel
05:38.13*** join/#asterisk dudes (n=dudes@12-215-34-84.client.mchsi.com)
05:38.41outtolunc16 as a 'cause code' is an 'normal answer'
05:39.02LostFrogDoes STUN deal with registration or RTP?
05:39.06outtolunci have no clue what you meant<G>
05:39.09mthemouttolunc: ya was just trying to help the other guy
05:39.14outtolunck
05:39.42debeastheh i can wait
05:41.09mthemouttolunc: so im convinced i have to do the detect in zaptel.conf, done! now i hit extensions.conf with a detected fax call, then it will hit the context defined in zaptel.conf and go either exten => fax,1,Dial,Zap/g2/${EXTEN} or ???
05:41.15debeastlooking for a win32 sip softphone any suggestions
05:41.28outtoluncstop
05:41.34mthemexten => voice,1,Dial,IAX2/termin/${EXTEN}
05:41.43debeastok besides stop
05:42.01outtoluncthose channels in zapata.conf that 'should look' should be diff context
05:42.06outtoluncin extensions.conf
05:42.27outtoluncthat context needs a exten => fax,1,yadda
05:42.30outtoluncthats it
05:43.05mthemright, but how does it know where to send the calls detected as voice?
05:43.15outtoluncjust remember the faxdetect=yadda is directional/both/none
05:43.30mthemis the whole purpose of faxdetect just to turn off echo cancel
05:43.39outtoluncit doesn't, and that was never one of your Q's
05:44.05mthemi know, i was just wondering if i was asking the wrong way
05:44.16outtoluncif you want that kind of info maybe you should read the agentlogin doc
05:44.26mthemthe POTS are always echo(all) = no
05:44.48outtolunchttp://www.dynx.net/ASTERISK/DOCS/RTF/agentlogin3.rtf
05:44.57debeasti didnt edit anything from the default configuration iam using idefisk to connect and i get this from asterisk register_verify: No registration for peer 'guest' (from 192.168.1.130)
05:45.21outtoluncbrb, drink
05:46.22mthemouttolunc: so that is not the problem, maybe instead of me just yabbing away, could you show me an example of a fax detect context. where faxes are send one way and voice another
05:47.05outtoluncthe point is if you read that doc
05:47.19mthem<outtolunc: ok i will
05:47.53outtoluncyou would have learned how 'backgrounddetect' works, then if you searched on nvfaxdetect (i stated what over half hour ago), then this would click
05:48.56outtoluncbackgrounddtect only triggers to nothing or 'talk', nvfaxdetect does 'fax' and 'talk'
05:49.03*** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye)
05:49.04debeastugh all i want to do is run the demo
05:49.18outtoluncwhich are what you want as a 'dialplan' progression
05:49.27outtoluncare/is/whatever
05:49.46outtolunci'm going for that drink now
05:49.53mthemhehe, well thanks
05:49.57debeastill buy you one
05:50.11mthemdebeast: what was your question?
05:50.32debeasti cant get my soft phone to register with asterisk
05:50.51debeasti didnt edit anything in the conf files
05:51.00mthemdebeast: what SC?
05:51.09debeastexcept for the codecs section
05:51.34debeastiax with idefisk
05:52.00debeasti get a message every time on the cli
05:52.07debeastregister_verify: No registration for peer 'guest' (from 192.168.1.130)
05:52.35*** join/#asterisk MGSsancho (n=user@adsl-67-125-157-68.dsl.irvnca.pacbell.net)
05:52.37mthemidefisk: have u created a peer/user in IAX.conf for your client
05:53.11debeastthere is a peer calles asterisk in the demo section of iax
05:53.17debeasterr guest
05:53.41debeastwith no secret
05:53.41mthemidefisk: how bout the secret
05:53.51mthemidefisk: k
05:54.30mthemoops
05:54.43mthemdebeast: what does it look like
05:54.54debeast1 sec
05:56.08debeast[guest]
05:56.19debeasttype=user
05:56.30debeastcontext=default
05:56.34mthemdebeast: do type = friend
05:56.58JerJerso a guest is going to register or something then?
05:56.58mthemdebeast: then asterisk -rx 'reload'
05:57.01debeastok reloading
05:57.50debeastahh much better
05:57.58debeastnow i get a bunch of errors
05:58.19mthemdebeast: well better that than none at all ;)
05:58.32*** join/#asterisk ptblank (n=MURDER1@68-169-161-61.lmdaca.adelphia.net)
05:58.36debeastat least it appears to have registered
05:58.48mthemread: http://www.asteriskguru.com/tutorials/idefisk_softphone.html
05:59.15debeastfunny thats where i dwonloaded fisk from
06:00.00debeastcool it playing it
06:00.29mthemdebeast: thats not even english.. ;)
06:00.38debeastshure it isnt
06:00.57*** join/#asterisk BMSS (n=bharatsa@210.211.246.47)
06:01.11BMSShello there
06:01.46*** join/#asterisk MrMAGO (n=mglucksm@pdpc/supporter/sustaining/MrMAGO)
06:02.29MrMAGOnight everyone... how can I make TEMP= the 3 first digits of ${EXTEN}?
06:02.42QwellTEMP= is 5 characters
06:02.59Qwellor do you mean $TEMP?
06:03.11QwellSet(TEMP=${EXTEN:3})  perhaps?
06:03.21twistedno
06:03.28mog_homenope
06:03.30mog_homeget the length
06:03.31twistedSet(TEMP=${EXTEN:0:3})
06:03.36mog_homethen cut off last three
06:03.39mog_homeor that
06:03.39Qwellumm, right
06:03.41mog_homei think
06:03.50twistedmog_home, that starts at zero and reads 3
06:03.54mog_homeyeah
06:04.07mog_homei forgot about that
06:04.10twisted;)
06:04.11mog_homebut README.variables
06:04.13mog_homeis my friend
06:04.14twistednitenite
06:04.26MrMAGO=)
06:04.32MrMAGOthx
06:04.47*** join/#asterisk cybertank (n=todd@CPE000dbd0f269c-CM00111ae6ff9c.cpe.net.cable.rogers.com)
06:05.12outtolunctwisted:0:3 <G>
06:05.27mog_homegnite
06:05.29*** part/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
06:06.18debeastcool when i hang up asterisk gives me a segmentation fault
06:06.30JerJerthen don't hangup
06:06.30outtoluncsweetness
06:06.48LostFrogAlways nice.
06:06.56LostFrogMake for long convos.
06:06.57outtolunc'end of call' (any of them) <G>
06:07.09debeastall of them at once
06:07.36outtoluncat least you don't have to worry about 'overages' <G>
06:07.36debeastalot of latency in the echo test though
06:08.39debeasthow much cpu horse power does asterisk need for 1 line anyway
06:09.06JerJerall of it
06:09.33debeastso i guess a 333 is to slow then
06:10.17debeastyes databases have been known to be process intensive
06:10.38debeastsigh
06:11.43debeastwouldnt it be nice to have a cable modem that uses tdm
06:11.45*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
06:12.14debeastyou missed
06:12.46debeastcrap got hit by ricochet
06:12.56outtoluncmissed, my ASSterisk
06:13.01outtolunchehe
06:13.39outtolunc<- king of 'old crap'
06:13.43*** join/#asterisk KaBewM (n=kabewm@66-215-7-106.dhcp.psdn.ca.charter.com)
06:13.51debeastis it my imagination or does nobody believe in tight coding anymore
06:14.21outtoluncexample?
06:15.11debeasti downloaded a couple of debian packages for x11 over 130 megs for gui
06:15.12outtoluncreason i ask is that since there are alot of other programmers in the mix, most of the time it's 'playing nice' that rules the code
06:15.40outtoluncok, so this is NON-asterisk related
06:15.52outtoluncand you are complaining about X
06:15.55outtolunchmm
06:16.03debeastasterisk is reasonably small for the features
06:16.17michael1234i do I configure pris
06:16.26michael1234for australia
06:16.30outtoluncget in line
06:16.50debeastwow you just let me cut ahead 23489873789738974798783587 spots thanks
06:17.07outtoluncthose were lost in y2k <G>
06:17.23outtoluncdidn't you get the memo <G>
06:17.35debeasti put it in the email
06:17.50lehelouttolunc: Zap/g1/XXXXXX < what exactly means this? (in group 1 of the zap channels.. and that number? - i'll use for sending faxes)
06:17.56outtoluncit means
06:18.45outtoluncchannel: type "Zap", group "1", xxxx= "number/exten to dial"
06:18.52debeastwell goodnight all
06:18.57outtoluncgnight
06:19.24debeastseya tommorow when we explore segmentation faults and you
06:19.40outtoluncsimply put, you'll probably get a congestion tone <G>
06:19.53justinnnnncan anyone here me with this zaptel.conf stuff ?
06:19.57justinnnnnaccording to the examples
06:20.04justinnnnnbchan=1-15,17-21
06:20.07debeasthow bout a no service tone
06:20.08justinnnnnshould do the trick...
06:20.12outtolunccan you hear me now?
06:20.14justinnnnnbut alas it crashes asterisk :(
06:21.08outtoluncdebeast: take the joke and 'run-away' with it <G>
06:21.17outtolunchehe
06:21.23lehelouttolunc: i'll use, but i dunno yet how to.. you could give me a hint? i know that i have to create a a separate file to send faxes.. how do i insert in my extensions? and XXX is equal with my fax extension?
06:21.26mthemjustinnnnn: what kinda span is it?
06:22.18*** join/#asterisk oej (n=Olle@apollo.webway.se)
06:22.21mthemjustinnnnn: some kind of euro ISDN?
06:22.34outtolunci'm wondering how 20 'channels' become 21
06:22.53outtoluncnevermind
06:22.57justinnnnnouttolunch.. because 16 is the d channel
06:23.09outtolunchad 6 fingers on my left hand
06:23.13outtoluncmy bad
06:23.17justinnnnn:)
06:23.21*** join/#asterisk wolfson (n=hehe@usr-kdh-208-6-58-26.beachlink.com)
06:23.31justinnnnnguess u had to be there :P
06:23.48outtoluncit was meant as a joke
06:23.54justinnnnnya
06:24.08lehelnobody using asterisk to send faxes?
06:24.09justinnnnndid i ruin the joke :P ?
06:24.19outtolunci came up with 'this is my luck' earlier
06:24.44*** join/#asterisk jeffik (n=Jeff@CPE0011505c92d3-CM014350000760.cpe.net.cable.rogers.com)
06:25.07lehelk..np;)
06:25.27JerJerasterisk can deal with faxing, no problem
06:25.39outtoluncyou find out you atm card is messed up, so you goto the bank and to have them test it on their reader, they say their reader is also messed up, this is my luck
06:25.42JerJerits just all how you deal with the faxing
06:26.13BMSSAnybody in the room working on the development of the SIP protocol?
06:26.14*** join/#asterisk Knight_DKN (n=knight_d@61.95.68.85)
06:26.23BMSSthat is on the SIP stack..
06:26.27JerJerdefine working
06:27.02BMSSok Jerjer
06:27.14Knight_DKNHas anyone ever heard of the leckiebus module?
06:27.28BMSSi mean have you worked on sIP stack , like buildding a server
06:27.45BMSSthe NIST Jain SIP stack is in Java..
06:27.55BMSSso have you workeed on that
06:28.26justinnnnnleckiebus module ?
06:28.26outtolunci can hoestly say i have not
06:28.29justinnnnnisnt that like sex over ip ?
06:28.33outtoluncer +n
06:28.46Knight_DKNYeah, brand new stuff...
06:28.50BMSSjerjer are you there
06:28.55outtoluncit's 'incubus module' <G>
06:29.01outtoluncsheesh
06:29.04BMSSanswer me please
06:29.39outtoluncbmss, i already have
06:29.53outtoluncthought others might have more to say
06:30.19lehelJerJer.. i can receive faxes; i want to send to..
06:30.37outtolunclehel, over what
06:30.46outtoluncto what
06:30.54outtoluncin what gx
06:31.02JerJerlehel:  then send them
06:31.25outtoluncplease people, just a TADBIT of specifics
06:32.36lehelouttolunc, JerJer: chan_capi.. (fritz! isdn), tdm400p (4 fxs)..
06:32.43outtoluncpleasssssse?
06:32.48leheli know about that fax.call file..
06:33.23lehelbut i dunno how to use txfax.. yet
06:33.42JerJerlehel: i'm sorry
06:34.04outtoluncok, that last line cleared it up, i know if i send a txfax to rxfax on the same box, hhehe those channels are up till i drop asterisk on its face
06:34.04lehelyou do?
06:34.47outtoluncseriously
06:35.15outtoluncwhy, probably because txfax doesn't gen a faxtone
06:35.34outtoluncbut anyways
06:35.48digimeanyone here use a sipura spa-841?  my TOS is not working and I can't find a setting to turn it on
06:35.50lehelhow to make txfax to get faxtone?
06:36.01outtoluncget is not gen
06:36.07outtoluncit WILL get one
06:36.17lehelzzz
06:36.24outtoluncso if the other end IS a fax, it will see it
06:37.13outtoluncbut can it do the rest of the nasty
06:37.38outtoluncwith g4, my tests say yes
06:37.47outtoluncwith g3, not really
06:38.07outtoluncwith g2, hahha
06:38.18outtoluncso
06:38.43outtoluncthe point is, if you want to do faxing finish t.38
06:39.23lehelouttolunc: if i'm correct t.38 is about voip faxing.. (?)
06:39.49outtoluncit's truely sad when a 'hasbeen' such as me can control the daily happening here
06:40.30outtoluncyes, t.38 is about Foip
06:40.40outtoluncfax over ip
06:41.12lehelyea;) do i need foip to send fax? i don't think so..
06:41.38outtoluncok, why don't you think do
06:41.40outtoluncer so
06:41.55outtolunclets take this from a user perspective
06:42.26outtoluncplease, give me your reasons why this is BS
06:42.41outtolunci'm gonna go get a drink in the interium
06:43.59lehelwhere did you see VoIP?.. it is about asterisk:fax:isdn - PSTN ..
06:44.23tclineksLogging into database with user test, password test, and database vmdb
06:44.28tclinekswhere are those values held?
06:47.57outtolunclehel, it was probably due to the 'actual' questions you 'did' ask that lead me to believe this is a total new thing to you
06:48.14tclineksah, they're defaults
06:48.38outtoluncbut since the questions were out there and everyone is in such a good mood, i opened the floor
06:49.22Qwellbed
06:49.27lehelouttolunc: did heard about ast_fax?
06:49.33outtoluncgnight qwell
06:49.56outtolunci've heard of a lot of things
06:50.07lehelme either,,
06:50.18outtoluncdid you also heard that it is just a topical app based on spandsp
06:50.45outtoluncuse app_rxfax and app_txfax
06:51.02outtolunc(which i'd already mentioned realier)
06:51.08outtoluncer earlier
06:51.19outtolunchmmm?
06:51.41lehelof course.. how do you think i receive faxes?..
06:52.05outtoluncnote: you do NOT want me to honestly answer that
06:53.00outtoluncwould probably go something like 'i guess you 'heard' it through the grape vine'
06:53.29lehel..i rather stop now talking with you, and read more.. zz
06:54.02outtoluncwell if you ask a question that 'can' be answered, i'd be more than happy to do so
06:54.27outtolunci am 'slightly' capable of doing so
06:56.05*** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net)
06:56.06outtoluncsuggestion: instead of asking 'if i heard of something' why not 'i've used xyz, and i think this that and the other thing need serious help, what are your thoughts?
06:56.29Qwellokay...wtf
06:56.31lehelmaybe i'm not that good to put questions
06:56.34outtolunchaha
06:56.37QwellSomebody paid $10 to get an answer to this question.
06:56.38Qwellhttp://tinyurl.com/cgddo
06:56.48outtoluncqwell WE KNOW your thoughts about ME
06:57.56outtolunchonestly, i think 'cups' should be shared, whats 'gained or lost' by one, should be shared by all' <G>
06:58.00outtolunchahah
06:58.18outtoluncor didn't you think i could read
06:58.36outtoluncand if so, so fast
06:58.54outtoluncor was it my aftershave
06:59.11outtolunci guess we'll never know
06:59.22mthemouttolunc: arent you just a bit harsh?
06:59.32outtolunctoo whom?
06:59.39outtoluncto qwell
06:59.45mthemouttolunc: in general
06:59.51outtolunche did it as jest to ME
07:00.14Qwellyep, ME sucked
07:00.27outtoluncto others here, i give 'sound' advise, i ALWAYS say keywords that will help you
07:00.40outtoluncALWAYS
07:00.45JerJernobody has seen harsh until you've pissed me off
07:00.48outtoluncall you have to do it 'listen'
07:00.56QwellJerJer: meh, you're a freaking teddy bear
07:01.21outtoluncso, am i 'harsh' sometimes, do i 'help' ALWAYS
07:01.39outtoluncand what lesson did you learn from this?
07:02.34outtoluncgees
07:02.36*** join/#asterisk Asylum (n=Asylum@dsl-58-6-126-60.qld.westnet.com.au)
07:02.39outtolunci guess nothing
07:03.12mthemouttolunc: i aggree u do give sound advice, but if ppl can understand it dont get all "u dont get anything, go read what i told you to read 1 hour ago" maybe better just to say u think they sould find more info on their own
07:03.30mthemits just the tone, nothing wrong with your answers
07:04.00*** join/#asterisk inspired (i=mikael@213.197.167.61)
07:04.06*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no)
07:04.06leheli'm afraid i agree.. with mthem,
07:04.20outtoluncmthem, i've trained people for over 25 years
07:04.33outtoluncsome of those people learned
07:04.37lehelbut anyway i'm still not a good questioner..
07:04.54inspiredanyone managed to get attended transfers working?
07:04.56outtoluncthe ones that didn't, weren't able to to taught
07:04.57leheli learned once from you.. but this fax thing ..
07:05.22inspiredI'm sure I can do it - if call parking would just return a variable with the parking extension
07:05.34outtolunc(meaning i gear training to the person, i do NOT have that personal aspect over 'bits')
07:05.42outtoluncso
07:05.58outtoluncyou are here because you want to learn about asterisk
07:06.03outtolunc(i'm assuming)
07:06.16outtoluncso much so you ask questions
07:06.24outtolunc(this is fact)
07:06.51outtoluncso, when i respond, i expect you to listen, because YOU invoked this)
07:06.59outtoluncthis is me
07:07.00lehelcorrect, but maybe today i put the wrong question.
07:07.15outtoluncand i'll never hold it against you
07:07.28outtoluncyou asked, that is the first step
07:08.10mthemouttolunc: im sure u are that good, it was my 2 cents, i like itif ppl get interested in asterisk and get a sense that you should only post here what you cannot figure out on your own by a simple google search, but ppl should not be discurraged to ask questions because you think they are too stupid. now lets move on to something more asterisk like :)
07:08.35outtoluncfirst off, i'm just me
07:09.05outtolunci've gleened alot of shit because of the TIME i've been doing shit like this
07:09.10AsylumCan anyone help me with a slight problem I have. When i caller calls, it goes to the IVR when they choose an option.. for somereason it goes silent for a bit then it dials through to the extentions.. Why is it pausing before dialing the extentions?
07:09.31tclineksblast: Logging into database with user test, password test, and database vmdb
07:09.38tclineksnot pulling from cdr_mysql.conf
07:09.53outtoluncand yes, 95% of the things that are asked here, COULD be learned within MINUTES, by a simple google search
07:10.14outtoluncbut, life isn't logical
07:10.29outtoluncsimply put, why am i here
07:10.50outtolunci'm here( to me) because i actually want to help
07:11.06mthemouttolunc: good, then thats settled
07:11.09outtoluncto i want to 'hand over' simple answers, not really
07:11.42outtolunci'll give you all teh 'words' you need, and SOMETIMES just spill it
07:11.53outtoluncbut i want you to learn
07:11.59outtoluncso
07:12.11outtoluncyou say asshole, i say tomato
07:12.18outtoluncit's that simple
07:13.07outtoluncdoes anyone want to contest this?
07:13.18outtoluncdidn't think so
07:13.29outtoluncso, anyone with a question?
07:16.27outtoluncthere are those i envy for being able to take in all these questions, without a snide word, tho, i'm not here often (the other sad part is more gets answered when i am)
07:16.36*** join/#asterisk vlrk (n=vlrk@59.93.72.238)
07:17.03vlrkis the chanspy not coming with asteirsk-1.0.9 ?
07:17.37outtoluncfrom earlier tests (another with a failed compile) i'd say no
07:18.19outtoluncwhat worked for that user was a complete flush of the the /usr/src/asterisk dir and a regrab
07:18.40outtoluncthen recompile obviously
07:19.29*** join/#asterisk Goshen (n=Goshen@c-67-172-238-57.hsd1.ut.comcast.net)
07:20.04*** join/#asterisk Mike_TK (n=Mike_@193.164.95.254)
07:20.30Jam1elol
07:20.40Jam1eI finally got everything working the way I wanted
07:21.06Jam1eafter many hours of config writing and trial and error
07:21.21outtoluncsadly, everyone has a question that (9 out of 10 times) is easily answered, and when you give them the answer they just tilt their heads
07:21.38outtoluncthat's it in a nutshell
07:22.10tclineks<PROTECTED>
07:22.11tclineks<PROTECTED>
07:22.26tclineksbut then it uses test/test/vmdb
07:23.34outtoluncit 'would only 'then use'' another because it had priority, did you try loading the vm stuff before the mysql stuff
07:23.56tclineksprobably, one moment
07:24.01outtoluncand vice versa
07:24.36tclineksmoved the cdr_mysql line in modules.conf up, same result
07:24.56tclineksouttolunc: can you explain the 'vm stuff'
07:25.31outtoluncyour alarm clock is going off, and your phone, do you answer the phone and listen to the alarm clock going off during the conversation, of do you answer the phone and say hold one, i need to turn the alarm clock off'
07:25.52outtoluncof=or
07:26.04tclineksheh
07:26.42outtoluncthe vm stuff is whatever crap you have occuring that is voicemail stuff
07:27.17outtoluncif you load asterisk as /usr/src/asterisk/asterisk -vvvvvvgc
07:27.19tclineksi see my problem
07:27.34outtoluncyou'll see how (and what order) things are loaded in
07:28.03outtolunck
07:29.47mthemok, think i have the simple stuff about the detect fax stuff down, here is the extensions.conf (when i have configured zapata.conf for the channels i wnat to fax detect with the [faxdetect] context and detectfax=yes)
07:29.49mthem[outbound]
07:29.49mthemswitch => Realtime/@realtime_route
07:29.49mthem[faxdetect]
07:29.49mthemexten => _1NXXNXXXXXX,1,Answer
07:29.49mthemexten => _1NXXNXXXXXX,2,Goto(outbound,${EXTEN},1)
07:29.49*** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek)
07:29.51mthemexten => fax,1,Dial,Zap/g2/${EXTEN}
07:30.01mthemouttolunc: see any problems?
07:30.24outtoluncwhat IS truely strange to me, is that it never amazes me how people with a question will steer away from asking one if they thing something adverse might happen
07:30.51outtoluncwell you are switching to realtime_route
07:30.52mthemjust checking, i might not just be as smart as i think
07:31.05outtoluncand the shit you want is in faxdetect
07:31.12outtoluncmeaning
07:31.24outtoluncthis is CONTEXT based
07:31.31mthemright, coz in case it is a voice i want to route based on area code
07:31.34outtoluncnot switch based
07:31.52outtoluncso put your stuff in a context, THEN do a switch
07:32.44outtoluncif ONLY the ones you want are those that qualified
07:33.14mthemi think thats what i did, the Answer() determines if it is a fax call, and sends it to the fax group, incase it is not it jumps to the realtime context right?
07:33.14outtoluncthen, move the exten => fax to that context
07:33.18outtoluncmeaning
07:33.30outtolunc[00:29] <mthem> switch => Realtime/@realtime_route
07:33.30outtolunc[00:29] <mthem> [faxdetect]
07:33.30outtolunc[00:29] <mthem> exten => _1NXXNXXXXXX,1,Answer
07:33.30outtolunc[00:29] <mthem> exten => _1NXXNXXXXXX,2,Goto(outbound,${EXTEN},1)
07:33.42outtoluncthats 3 diff contexts happening
07:34.08*** join/#asterisk \PsyKo\ (n=xxxxxxxx@golia.caltanet.it)
07:34.11outtoluncthe switch, the faxdetect (which was never meant to be one, and the outbound)
07:34.43outtoluncthink 'simple'
07:35.00outtoluncdo your switch (if you have too
07:35.10tclineksgreat! AMP is up and running - pretty bad install docs
07:35.14Ikarushmmmmmmm, *ponder*, what phones other then the Sipura and BudgetTone are there in the around EUR 100/workplace cost
07:35.34outtoluncon each context/box have the context, in that/those contexts have the exten => fax
07:35.55*** join/#asterisk voipguy (n=voipguy@196.200.26.42)
07:36.11mthemright but u cannot mix static and switch in realtime as far as i know
07:36.27mthemthat is the reason for the goto
07:36.34outtolunchmm
07:36.59outtoluncwhy can't you make it part of the realtime
07:37.09outtoluncit's dialplan flow
07:37.24outtoluncregardless where that dialplan is
07:37.49outtoluncut oh <G>
07:38.13mthembecause then all lines in [outbound] (voice only lines) would also be Answered and i dont wanna add 4 sec to my termination time for those lines
07:38.46vlrkwhen i use the chanspy it gives me beep on snooping channel nothing else ?
07:38.50vlrkany clues?
07:38.51outtoluncbecause THEN you are forced to put all in one place and not the other?
07:38.55mthemif someone calls out on a fax line with voice i have no problem with them waiting
07:39.09*** join/#asterisk Delvar (n=irc@host-83-146-53-34.bulldogdsl.com)
07:39.25outtoluncit's realtime, you can put damn near everything in there nowdays
07:39.42outtoluncexcept the pointer to it
07:40.20outtoluncwhat asterisk version are you running again?
07:41.11*** join/#asterisk TK9 (n=Administ@p54B29575.dip0.t-ipconnect.de)
07:43.02outtoluncside note: [00:38] <mthem> because then all lines in [outbound] (voice only lines) would also be Answered and i dont wanna add 4 sec to my termination time for those lines
07:43.20outtoluncall that, it's a 'group' right
07:43.33mthemmain point is that i dont want voice only calls answered, if i put all in a single context that will happen
07:43.57mthemno, the fax detect are not in a single group
07:44.31mthemvoice and fax are mixed in the same group, but i only enable faxdetect = yes for the fax lines
07:44.34outtoluncok, so what are they?
07:44.45mthemZAP
07:44.53outtoluncnow we are getting to the issue
07:45.01outtoluncok
07:45.08outtoluncthat makes it easy
07:45.24outtoluncwhere on the zap train do they lay
07:45.47mthemno comprende
07:46.03mthemin zapata.conf?
07:46.28mthemall over, channel 1,6,17-56,95... etc
07:46.38mthemand voice only in between
07:46.46mthemall under group 1
07:46.57outtolunci'm only assuming, but since this all came up, i'm ASSUMING that only a select few, say ports xx to xx on span 1, ports xx-xx on span2 etc
07:47.11mthemno, all over
07:47.18justinnnnnmthem any ideas on my little issue :) ?
07:47.21outtoluncno, EXACTLY
07:47.38Ikarusanyone here ever used phones from http://www.ipchitchat.com/
07:47.45mthemthe exact channels that are used for fax?
07:48.05outtolunci can honestly say no, no ipchitchat <G>
07:48.11*** join/#asterisk Goshen (n=Goshen@c-67-172-238-57.hsd1.ut.comcast.net)
07:48.34outtoluncmthem, the exact structure you want
07:48.40outtoluncexample
07:48.49outtoluncyou have a te410p
07:48.59*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
07:49.14outtoluncit's pri based on all 4 ports
07:49.49*** join/#asterisk pussfeller (n=todd@12.150.129.171)
07:49.53outtoluncyou only want ports 0-3 one EACH to be in this method
07:49.58outtoluncshit like that
07:50.12outtoluncer one=on
07:51.11outtoluncwhich might also equate to why i'm not being as nice as i should
07:51.21Ikarushmmm, then I think my choice is down to BudgetTone vs Sipura
07:51.30mthemright, 4 CBs with customer phones/faxes connected to a TE405P (96 channels = group 1), in the box is also a TDM400P for fax termination (no echochancel) that is group 2. calls will be comming in on the TE405P and depending on the context they are in and if they detect as fax they will be terminated through the TDM400P or IAX to a coreserver
07:51.58outtolunccb's meaning 'channel banks' thats my assumption
07:52.05mthemya\
07:52.15outtoluncone te405p
07:52.24mthemya
07:52.43outtoluncall in ONE freakin group (defined by zapata.conf)
07:52.54IkarusOr is there something else in that price range ?
07:53.57outtoluncit's one device
07:54.17outtoluncthat just happens to naturally be devided in 4
07:54.18mthemok, but then when i want to terminate a fax call and the first POTS line is busy what do i do? it is so much easier to terminate to a group when i want first availible line
07:54.50outtoluncand for petesake, you do NOT thing you can divide it more with zapata.conf
07:54.53outtoluncYOU CAN
07:55.25outtoluncthe prob you are having is you want the best of both worlds
07:55.36outtoluncmeaning, all ports for this
07:55.43outtoluncbut only these for that
07:55.58mthemok, still i dont get the big issue, maybe i use a few more cycles goto other contexts, but that is not really a concern
07:56.01outtoluncwell, umm get over it
07:56.12mthemi cannot group zapata for fax and voice
07:56.19outtoluncdecide what you want to do and config to fit
07:56.40outtoluncmeaning, select SOME for fax
07:56.57outtoluncselect (and group) the rest for other
07:57.06mthemi cant do that
07:57.16outtoluncwel then you are shit out of luck
07:57.33outtoluncNO voip board will do what you want
07:57.54outtolunc(note: that called SUB-select)
07:58.10mthemso i cant configure fax detect per channel?
07:58.35outtoluncooo these ports do this, until i say do that, then ONLY a SUB selection does the other
07:58.48outtoluncsure you can
07:59.03outtoluncbut they will HAVE that regardless
07:59.07outtoluncmeaning
07:59.19outtoluncyou can 'group' channels
07:59.33outtoluncyou cannot sub-group channel
07:59.34outtoluncs
08:00.10outtoluncso select channel span1 1-20, span2 1-20, etc.
08:00.16*** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1)
08:00.31outtoluncthen select span1 21-23, span2 21-23
08:00.35outtoluncetc
08:00.36mthembut fax detect is not a group wide feature or span wide for that sake
08:00.45outtoluncand group them diff
08:00.54outtoluncbut you canNOT
08:01.30outtoluncdo group 1 channel 0-96 but only ports 3, 56, 94 do this
08:01.42outtoluncunderstand?
08:02.43outtolunc'this isn't burger king' comes to mind, you can't have it 'your way'
08:03.06mthemsure, u want to group all fax lines in one single group, i just dont get why? just so i dont have to have 2 contexts for outbound fax/maybe voice calls?
08:03.32outtoluncnow, if you want to get creative about it, AND understand the limitations with it... then fine
08:03.50outtoluncwhen did i say single group
08:04.10outtolunconly referring to YOUR wanting
08:04.13*** join/#asterisk tobiasWolf (n=konversa@195.162.255.10)
08:04.40mthemok, tell me were u are going?
08:04.46outtoluncwhen i all comes down, you'll need as many group as you want sub'ed
08:04.52mthemam i making something that will not work
08:04.54*** part/#asterisk Mike_TK (n=Mike_@193.164.95.254)
08:04.59outtolunci'm not doing that <G>
08:05.11outtoluncmost of my stuff is outbound
08:05.28wunderkinumm  i have a feeling someone at my telco fucked up my pri.. im just now beginning to get it set back up again.. i installed it on another pc with another t1 card and the install went perfectly.. now its saying provisioned in alarm up.. after they do their testing it comes out of alarm, back into red alarm and gives me an error that both sides think they are cpe.. hmmmmmm??????
08:06.05outtoluncyou 'installed your pri on another pc' <G> hmm
08:06.18outtolunci'm not even gonna go there
08:06.46wunderkin...
08:07.16wunderkini was setting up a new server, it wasnt ready when they were to install it so i used someone else's 1 port card in the old machine to get the line installed
08:07.18mthemill set it up tomorrow, thanks for all the help
08:07.32outtoluncgoodluck
08:07.37mthemthanks
08:07.40outtoluncseriously
08:08.01mthemdont worry
08:08.04wunderkini setup 2 pcs back to back between a te410p each and they worked same config.. now when i put it into prod, one line is ok and the other isnt
08:09.05outtoluncso the te410p is just hangin out in the open
08:09.33outtoluncyou said 'a te410p' that means 1
08:09.41outtolunc2 boxes
08:09.46wunderkini had it setup between 2 boxes yes for testing
08:09.49outtoluncone is without
08:09.54wunderkin1 in each
08:10.10outtoluncthen its a 'pair'
08:10.26wunderkinthe each meant 1 in each box
08:10.39outtoluncand you obviously used a crossover (t1) cable to connect them
08:10.45wunderkinyes and now im not :D
08:10.54outtoluncwhy not?
08:10.57wunderkinim using the same cable that i used during the install
08:11.11wunderkinnow its hooked up to the telco and i use a regular cable to them
08:11.12outtoluncwhich you found where?
08:11.28wunderkinan ethernet cable yayya dont go there
08:11.44outtoluncwell i have to do there
08:11.47wunderkinhehe
08:12.00IkarusBLEAH, I am going OUT OF MY MIND, comparing VoIP phones without being able to just try them
08:12.08wunderkinand the cable would cause each side to be cpe?
08:12.43ZeeekIkarus what budget?
08:12.44outtoluncactually, it's after 1am here
08:12.51wunderkinhere too :D
08:12.57Zeeekhere too. It's 10:12 AM
08:13.24outtoluncsearch the wiki for t1 crossover, if you think your cable suffices, then go for it
08:13.42wunderkinit did before
08:13.46IkarusZeeek: around 100 euro/phone
08:13.46outtoluncif not, make one that conforms
08:13.47wunderkinand it works for the other one
08:13.56wunderkini have 2 lines to them
08:13.58outtoluncwhat cards?
08:14.06*** join/#asterisk darkskiez (n=darkskie@194.247.78.146)
08:14.20wunderkinon the same te410p that i was doing testing with before
08:15.04outtoluncokk on a te410p, if you use a 'real' t1 crossover cable when you boot up and load the drivers those ports will be green
08:15.15outtoluncif they are NOT, your cable has issues
08:15.31outtoluncgoodnight
08:15.34wunderkinim not going between the computers now
08:15.37wunderkinnite
08:15.37ZeeekIkarus you sure you want a phone and not an ATA+analog phone?
08:16.50ZeeekIkarus some peoplehave better luck with their analog phones and for that budget you could buy a converter
08:17.34IkarusZeeek: we would need to buy new analog phones then :)
08:17.40Delvaraye in my experiense an ATA like a handytone or sipura is much better than a TDM board.
08:17.45IkarusZeeek: so not a good option
08:17.52ZeeekI have heard good things about SIpura (now Linksys)
08:18.04Delvarsipura rock :)
08:18.17ZeeekI have several BT102 (Grandstream)  and they work great for what we need
08:18.35IkarusZeeek: none of the echo issues, etc some people reported ?
08:18.37ZeeekTHe best phone I've used so far is the Polycom ip500 but it's twice what you want to pay
08:19.53ZeeekThere is an occasional echo on voip/voip calls (me-->asterisk-->voip service) but usually only for a couple seconds
08:20.04Zeeekand I think that happens with all phones
08:22.23Zeeekor could happen
08:24.21IkarusZeeek: hmmmmm, I'll guess I should get a single one for a test run here
08:24.54IkarusCan get them for EUR 63.50 a piece, which is quite a price saver and will make me the hero of the IT dept
08:25.33ZeeekIkarus a lot of people complain about the old BT100 series. I have had very good luck with them, but for $70, it ain't an enterprise phone :) It works well daily for us to connect home and office and use with all kinds of voIP services
08:25.45ZeeekDefinitely try ONE first
08:26.03IkarusZeeek: this is a school, so it is a really low budget deal here
08:26.04Zeeekfor a company though, my opinikn would be to try for the $200 ip501
08:26.06*** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de)
08:26.10Zeeekok, understood
08:26.33Zeeekin the US the school could finance the phones with a junk food vending machine
08:27.15IkarusThe only alternatives are the Sipura SPA-841 and the Grandstream GXP 2000 but if I go for those I really will have to justify my budget
08:27.40ZeeekI've heard very miced reviews about the GXP - Sipura is better spoken of
08:28.09IkarusZeeek: I am hearing total crap about the Sipura, mainly due to stupid menu systems, etc
08:28.13Zeeeks/miced/mixed
08:28.32Zeeekyou want to see the worst web config system, look at Polycom
08:28.48ZeeekThe grandstream web config interface is 100 times better
08:28.51IkarusZeeek: this is on phone menu
08:28.57Ikarusbbias, need to do some helldesking
08:29.06Zeeekbut Polycoms aren't "made" to be configured using the web server
08:30.29fugitivomorning
08:30.36Zeeekhi
08:30.46fugitivogod, it's 5:30am here
08:31.37Zeeeksleep(8*60*60)
08:32.08Zeeeksystem(MORNING_PEEE,COFFEE)
08:32.22fugitivocoffee, good idea
08:36.05IkarusZeeek: right, back, the problem with the Sipura is that transferring calls takes 6 button presses (excluding the number)
08:36.19Zeeeknot if you use # and asterisk
08:36.39Zeeekbut that would be unannounced
08:36.47Ikarushmmmm, true
08:37.14*** join/#asterisk Badenser (n=user@141-19-124-83.dsl.3u.net)
08:37.38Zeeekpersonally, and this is meant to be open for comment by one and all, I'm reflecting on how our small business can best use asterisk, which we've had for over a year now
08:37.49*** part/#asterisk Badenser (n=user@141-19-124-83.dsl.3u.net)
08:37.54Zeeekwe have three internal ZAP phones
08:38.04*** join/#asterisk darkskiez (n=darkskie@194.247.78.146)
08:38.51Zeeeksome calls are routed by CID, others ring several phones and I'm thinking of having SIP incoming services for each person, now that numbers are cheap here
08:39.42IkarusZeeek: I am going to order a single BudgetTone and hook it up to the second line or something
08:40.16IkarusAny suggestions on what generic brand ISDN card to pickup (eventually 2 need to be used) ?
08:40.53ZeeekI don't use ISDN, so, no
08:41.26Ikarusah well, I'll just use an old card I still have first should work
08:42.06ZeeekI believe the BT now does attended transfer with the later firmware
08:42.06*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
08:42.08puzzledmorning
08:42.25Zeeekhi puzzled; something like 5.18 and up?
08:43.34puzzled5.18?
08:43.43*** join/#asterisk \PsyKo\ (n=xxxxxxxx@golia.caltanet.it)
08:43.57Zeeeksorry, that was the firmware version in the other subject
08:44.10puzzledah right :)
08:44.13vlrkdo we have any application in the asterisk where it will play beep in between rtp flowing (apart from chanspy)
08:49.52*** join/#asterisk FuzzyCat (n=ScaredyC@j25251.upc-j.chello.nl)
08:54.26*** join/#asterisk Delvar (n=irc@host-83-146-53-34.bulldogdsl.com)
08:56.22*** join/#asterisk olivier_ (n=olivier_@obs92-4-82-239-116-113.fbx.proxad.net)
08:58.49olivier_well i knew it should be in the docs but is * sip rfc3261 compliant ?
08:59.27*** join/#asterisk RoyK (n=roy@80.239.107.70)
09:02.05RoyKgninrom doog
09:02.35Ikarussilly randomness, anyone here running fax services using something other then an ATA or a CAPI compaitible card
09:03.26*** join/#asterisk folsson (n=filip@h82n1fls32o985.telia.com)
09:03.35RoyK~seen inspired
09:03.39jbotinspired <i=mikael@213.197.167.61> was last seen on IRC in channel #asterisk, 1h 58m 17s ago, saying: 'I'm sure I can do it - if call parking would just return a variable with the parking extension'.
09:03.39Zeeeknow that I talked up the BT100, I can't get it to stay registered inside the network :)
09:04.21*** join/#asterisk ZX81 (n=ZX81@222-153-100-242.jetstream.xtra.co.nz)
09:04.32ZX81mailing lists down?
09:04.41ZX81:) hi all
09:05.09ZX81~ping
09:05.11jbotpong
09:05.15ZX81~pong
09:05.16jbotwheeeeeeeeeeeeeeeeeeeeeeeeee!
09:05.18Zeeekyou're there
09:05.25ZX81hehe
09:05.31ZX81you getting mailing list posts?
09:05.41Zeeekhaven't looked for a couple days
09:05.48ZX81kk
09:06.06ZX81~wheeeeeeeeeeeeeeeeeeeeeeeeee
09:06.07jbotextra, extra, read all about it, wheeeeeeeeeeeeeeeeeeeeeeeeee is ping
09:06.20ZX81~ping
09:06.22jbotpong
09:06.25ZX81:)
09:10.05*** part/#asterisk FuzzyCat (n=ScaredyC@j25251.upc-j.chello.nl)
09:11.42RoyK~pong
09:11.43jbotwheeeeeeeeeeeeeeeeeeeeeeeeee!
09:15.21Zeeek???
09:18.52*** join/#asterisk phpboy (n=shane@196.34.242.154)
09:19.18phpboyhi, how would I go about answering another extentions number if it's ringing from my extention?
09:19.49*** join/#asterisk Delvar (n=irc@host-83-146-53-34.bulldogdsl.com)
09:20.24lehelphpboy: pickup-group;? but it's not so clear 4 me what do you mean..
09:21.14*** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net)
09:22.17phpboysay now that my phone is extention 1 and the phone ringing is extention 2... I want to be able to answer extention 2 from my phone if I hear it ringing in the office
09:22.35Zeeekdial *8
09:23.01phpboyBut I want to specify which extention I wanna answer
09:23.12Zeeekthen you're screwed
09:23.31phpboyit is possible, I just dunno how to do it
09:23.35Delvaryou will need to modify asteridsk
09:23.52phpboy:<
09:24.08Delvaror maybe use agi script... that bridges a call
09:24.09puzzledhmm, that makes it a pretty useless feature
09:24.50Zeeekit answer a phone that's ringing. Maybe it proposes a choice if several are ringing
09:25.00Delvarno it doesnt
09:25.06Delvaror at least iv not seen it
09:25.36phpboyso will it just pick any phone if more than one phone's ringning?
09:25.49Delvarit will pickup the fiorst to start ringing
09:25.54puzzledso this can be solved with some agi magic?
09:26.02Delvarprobably
09:26.13puzzledi can not imagine any company accepting the current *8 behavior
09:26.25phpboypuzzled: correct
09:26.33Delvarfor most situations it works fine
09:26.37puzzledyou need the ability to specify the extension you want to pick up
09:26.47phpboyDelvar: not for big firms
09:27.04Delvarsuppose, ost our custmers are small-medium
09:27.20phpboyI c
09:27.39puzzledit's like functionality matching. if the old system has it than asterisk should have it
09:28.17Delvarim sure iv seen a patch or agi script for this somewhere...
09:29.15*** join/#asterisk montag___ (n=montag@195.223.103.50)
09:29.24montag___<PROTECTED>
09:29.45Delvarare htey registered?
09:29.48Delvarsip show peers
09:30.02montag___yes
09:30.30montag___both clients can receive call from non sip extension, only thing that does'nt working is SIP to SIP
09:30.59Delvarcanreinvite=yes/no?
09:31.11Delvarboth setup with supported codecs?
09:31.21montag___canreinvite=no
09:31.23olivier_<montag___> seem to be a context issue no ?
09:31.35Delvarthen yes its a context thing
09:31.47*** join/#asterisk Firestorm-voip (n=Firestor@dilbert.mysoft.se)
09:32.12Delvarwhen you dial from one ext to anotehr do you see the Dial at teh * cli?
09:32.19RoyKsdrawkcab elttil a smees siht
09:32.46montag___context it's the same for al sip client....
09:32.55montag___nat=yes
09:33.34Delvarcheck cli and make sureyour dialplan is doign the corect Dial
09:36.18Ikarusbleah, someone here wants me to use a traditional PBX
09:36.45Ikaruswhich is ofcourse a Bad Idea (tm) when you want a IT techy to set it up
09:36.49*** join/#asterisk opus__ (n=opus@dahphish.org)
09:36.54opus__hello ppl
09:36.55Zeeekheh
09:36.59opus__who here is from london
09:37.13puzzledonly the little mice
09:37.17opus__sup zeeek
09:37.30IkarusI never figured the current PBX out which is half the reason for this switching
09:37.47opus__hahaha
09:38.16fishboy1669anyone got any idea why i get this when i try compile asterisk
09:38.19fishboy1669chan_sip.c:9319: internal compiler error: output_operand: invalid expression as operand
09:38.54opus__remove your local include/asterisk
09:39.27fishboy1669?
09:40.10fishboy1669where is that? in the c code or is it in a directory or something
09:40.18fishboy1669or the make file?
09:41.13*** join/#asterisk mmmToop (n=chatzill@196.31.11.194)
09:41.15fishboy1669hi opus
09:41.41opus__lemme find it real quick
09:41.51fishboy1669cheers
09:41.59opus__rm -rf /usr/include/asterisk
09:42.08opus__make clean
09:42.27opus__make sure you have the latest zaptel and libpri, and make sure they are 'make && make install' for each dir
09:43.11fishboy1669i have latest dl from cvs repos
09:43.20fishboy1669i dont have /usr/include/asterisk
09:43.26opus__what OS?
09:43.59fishboy1669suse 9.3
09:45.45fishboy1669make[1]: Entering directory `/usr/local/src/development/asteriskv1-0_1.0.9/asterisk/channels'
09:45.45fishboy1669gcc -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g  -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE  -O6 -march=i686   -DZAPTEL_OPTIMIZATIONS  -DASTERISK_VERSION=\"CVS-v1-0-09/30/05-22:31:05\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\" -DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\" -DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" -DA
09:45.45fishboy1669chan_sip.c:9319: internal compiler error: output_operand: invalid expression as operand
09:45.45fishboy1669Please submit a full bug report,
09:45.47fishboy1669with preprocessed source if appropriate.
09:45.49fishboy1669See <URL:http://www.suse.de/feedback> for instructions.
09:45.51fishboy1669{standard input}: Assembler messages:
09:45.53fishboy1669{standard input}:123824: Warning: partial line at end of file ignored
09:45.55fishboy1669is more detail
09:46.00opus__hmmmm.....
09:46.12fishboy1669cant find anything in google
09:46.25fishboy1669zaptel and libpri seem to compile ol
09:46.26fishboy1669ok
09:46.29RoyK~pastebin
09:46.31jbotpastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca/
09:46.39fishboy1669sorry
09:46.48opus__fuck pastebin
09:46.55fishboy1669but not many people were chatting so thought id get away wit it
09:46.58fishboy1669;-)
09:47.01opus__yeah no shit:)
09:47.06phpboyexten => s,5,Dial(SIP/6940&SIP/6941,17) <--- what exactly would that line do?
09:47.10opus__NOW they are chatting
09:47.15fishboy1669lol
09:47.26Zeeeki'm not chatting
09:47.27fishboy1669do something wrong and u get jumped on
09:47.28fishboy1669he he
09:47.34opus__phpboy dial two channels at once, timeout in 17 seconds
09:47.39Zeeeki'm eating
09:47.44opus__first channel that answers gets the call
09:47.50phpboyI see
09:47.54fishboy1669so royk any idea on my issue?
09:48.05*** join/#asterisk MuppetMaster (n=MuppetMa@62.37.168.255)
09:48.09MuppetMasterHello
09:48.26opus__hello
09:48.30fishboy1669hello
09:48.37phpboyopus__: how would I go about doing the following say now I had a number(external pstn) 011 993 4260-69
09:48.41opus__cdrecord: Notice: Use -overburn option to write more than the official disk capacity.
09:48.47opus__hell yeah
09:48.52fishboy1669guess im on my own with this one
09:48.56fishboy1669thanks for trying opus
09:49.25phpboyif I dial 011 993 4260 then it'll go through to exten 4260 and if I dial 011 993 4261 it'll go through to extention 4261
09:49.30phpboyhow would I go about doing this?
09:50.03Zeeekcontexts
09:50.05opus__phpboy - the channel driver ( zap, sip, iax) will have a default context, usually specified in [general] \n context= at the top of the file. If not then its always default.  Then, do exten => 0119934260-69,1,Answer ... followed by standard dial plan logic for an extension to answer a call
09:50.33opus__[default]
09:50.45phpboyI see
09:50.46opus__exten => 0119934260-69,1,Answer
09:50.50opus__exten => 0119934260-69,2,Wait,1
09:50.58opus__exten => 0119934260-69,3,Dial(SIP/softclient-bob)
09:51.10opus__exten => 0119934260-69,4,Congestion
09:51.33phpboyok... but how does it know which extention it's coming through on from the pstn?
09:52.09opus__you have to use variables
09:52.29opus__the dial plan logic is like QW Basic/BASIC, there is an if statement and goto statement
09:53.01opus__chances are, there is a variable to denote where its coming from -- else wise you can separate everythin by context
09:53.08opus__[iax-from-inet]
09:53.20opus__[zap-from-phone-company]
09:53.27opus__[sip-softclinets]
09:53.44opus__{$CALLERID} etc..
09:54.04phpboyI see
09:54.39phpboyso I've gota look up how-to statements
09:54.44phpboygoto even
09:55.43Zeeekif you look up how to you'll know how to got
09:56.16MuppetMasterI have just upgraded from v1 to v2 of PHPAGI (http://phpagi.sourceforge.net/) and am looking into using the php-fastagi.php script with xinetd.
09:56.38opus__phpboy it might be users to use hierarchies of [context]'s..
09:56.47opus__might be easier UGH.. bleh
09:57.37MuppetMasterIf one calls an AGI app with agi://localhost/script.agi and there is no listener running, will Asterisk automatically try to get the script out of the /var/lib/asterisk/agi-bin directory?
09:57.40opus__damn it takes like 3 hours of just sitting to start work:)
09:57.57Zeeekphpboy what kind of channels are these? ZAP ?
09:57.58opus__agi://localhost/script.agi? how do you do that?
09:58.17MuppetMasteropus__  Not sure I follow your question?
09:58.36opus__<MuppetMaster> If one calls an AGI app with agi://localhost/script.agi
09:58.38opus__explain that
09:58.55MuppetMasteropus__ http://phpagi.sourceforge.net/
09:58.58MuppetMasterOops
09:59.00MuppetMasterwrong link
09:59.20phpboyZeeek: yeah, ZAP...
09:59.27Zeeekwhat hardware?
09:59.50phpboyJurghanns Quad ISDN card
09:59.58fishboy1669is there a definitive list of dependencies that i need?
10:00.01fishboy1669for a install?
10:00.02Zeeekeach line is a ZAP channel?
10:00.29Zeeekphpboy I mean each incoming number is a ZAP channel?
10:00.42*** join/#asterisk FABRIZIOxxx (n=FABRIZIO@81-208-26-86.ip.fastwebnet.it)
10:00.49MuppetMasteropus__ http://www.voip-info.org/wiki/index.php?page=Asterisk+FastAGI
10:01.22phpboyZeeek: correct... 4 ISDN lines = 8 channels and 8 numbers
10:01.26opus__lame!
10:01.58Zeeekphpboy so all you need to do is give a different context for each channel, then at that context, route the calls
10:02.18Zeeekif there is common code, put it in macros or goto(it)
10:02.34phpboyproblem is... in my country... the calls 'hunt' through the channels looking for an open line
10:02.36phpboy:<
10:02.42Zeeekaha
10:02.50FABRIZIOxxxcan a tdm400 card fit on the 5,5 V PCI slot? if so is it better on that type or on tyhe classic 3,3V??
10:03.00Zeeekand the number cazlled is transmitted into a variable so...
10:03.19Zeeekis this link still good?
10:03.20ZeeekThe dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls.
10:03.20Zeeekhttp://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650
10:03.34Zeeekyes it is
10:04.34phpboyZeeek: that doc will help me through?
10:04.56Zeeekwell at least it'll explain most of the basic dialplan info that you don't seem to know yet
10:05.12ZeeekAll about channels
10:05.12Zeeekhttp://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN510
10:05.15Zeeekthis too
10:05.47opus__time for BEER
10:05.56opus__BEER
10:06.27Zeeekphpboy you may be looking for DNID
10:07.14phpboy:/
10:07.32Zeeek<PROTECTED>
10:07.45phpboyah ok
10:07.49phpboyso a goto statement?
10:07.51phpboywith that
10:07.59Zeeekif that is what I think it is (and it may not be)
10:08.12Zeeekyou would use goto yeak
10:08.18phpboylike if(${DNID} = '119936942')
10:08.53ZeeekI'd suggest doing this first: NoOp(${DNID})   call in and see what it shows
10:09.33RoyKphpboy: gotoif($[ '${DNID}' = '119936942' ]?asdf)
10:09.35RoyKi guess
10:09.36*** part/#asterisk MuppetMaster (n=MuppetMa@62.37.168.255)
10:09.37nfi|ermeshi m8s
10:09.46Zeeekyou could then GoTo(${DNID})
10:10.00RoyK~lart Zeeek
10:10.14nfi|ermeswhich codec should i use in europe ?
10:10.22*** join/#asterisk ful|work (n=fulgas@213.58.130.46)
10:10.25CleanerX711 a.law
10:10.30Zeeekbah
10:10.33nfi|ermeswhen i call from my analog phone a listen a lot of noise
10:10.34CleanerXit's the standard isdn codec
10:10.51CleanerXno transcoding needed if you use this codec
10:11.02RoyKCleanerX: s/europe/most of the sane world/
10:11.07phpboygotoif($[ '${DNID}' = '119936942' ]?SIP/6942)
10:11.08opus__nfi|emes me to
10:11.10phpboylike so?
10:11.16opus__nfi|emes i found out i had the wrong Zap channel in my logic
10:11.29opus__nfi|emes try your stuff
10:11.39opus__phpboy i believe so.
10:12.01Zeeekphpboy no, you won't dial a phone that way
10:12.07opus__where is everybody from in here?
10:12.28phpboy:/
10:12.29CleanerXthe kingdom of far far away
10:12.41phpboyPretoria, South Africa
10:12.47Zeeekphpboy it's time to read the link I sent so you know what dial and the dialplan are
10:12.48swm_oregon who is in oregon?
10:12.54phpboyok
10:13.01opus__who is asking?
10:13.05swm_opus where are you in oregon?
10:13.10opus__portland
10:13.10swm_I'm in oregon too
10:13.12CleanerXswm_, it's a spice
10:13.14swm_I am in salem
10:13.17opus__nice
10:13.22CleanerXmostly used on pizza
10:13.28opus__hmmmm... pizza
10:13.32swm_opus what do you do with asterisk in portland?
10:13.43opus__swm_ i sell a turn key solution
10:14.02swm_turn key? elaborate please, interested
10:14.03opus__and voip ip-pbx consulting
10:14.10CleanerXpacked in nice plastic you may think it's weed ;-)
10:14.15opus__you put in a cd, eat a pizza, and you come back its done.
10:14.33swm_Dont talk about weed, i'm on the grand jury till friday
10:14.50opus__don't smoke the evidence hehe
10:14.51swm_lol opus you should come down to salem and check out my technology
10:15.37gordonjcpCleanerX: now all I need to do is stop my Mum posting unusual spices to me
10:15.50gordonjcpshe buys things in her local wholefood shop and posts them down to me
10:16.17gordonjcpdespite the fact that I live in a city with the second-highest proportion of Asians in the UK
10:16.29gordonjcpthus shops selling all manner of good stuff are everywhere
10:17.50BMSSHEllo
10:18.56BMSSany IRC channel for SIP protocol development
10:19.07BMSSdoes anybody have a knowledge
10:19.17BMSSof the IRC channel
10:19.44swm_What about it?
10:20.16BMSSi am speaking about the SIP protocol
10:20.56swm_Well asterisk implements the sip protocol so umm if you look at the structure of it, you can figure it out
10:23.23opus__swm What are you using Asterisk for?
10:23.26opus__what company are you with?
10:24.14swm_I own my company
10:24.34swm_I use asterisk for everything from security system technology to queue's and other stuff.
10:24.48opus__security systems?
10:25.10BMSSwow
10:25.29swm_I have a DHL Security system that is intergrated into asterisk, when it goes off, it notify's everyone that there is a certian problem (it's specific on the problem)
10:25.49BMSSthats great
10:25.50lehelpeople why is possible that i hear myself (zap channel) double when i'm talking with an IAX user? , and after a while the connection interrupts
10:26.02BMSSwell; are you also working on the IMS?
10:26.11BMSSIP Multimedia Subsystems
10:26.17opus__DHL security, never heard of them
10:26.38lehelthis is the output CLI: http://pastebin.ca/24569
10:26.42johnmopus__: British Fedex :)
10:26.54swm_They make alarm panels and equipment, I can monitor tempature, fire, burg, I even have meters in the ground to monitor movement.
10:27.18opus__no way :)
10:27.29opus__do you use it for video , or have that planned?
10:27.35opus__i wanted to write a motion detection plugin
10:27.49lehelany ideas?
10:27.57swm_Oh yes way, it's basically a waterproof sisemomiter set to a certian intensity (foot traffic) and if something happens it sets off a silent alarm
10:28.21swm_It knows where somone is going around my property and notify's me on each event
10:28.35lehelwhat could cause such thing?
10:28.47opus__lehel i don't know, I don't use Zap channels
10:29.06lehelk
10:29.14opus__swm_ thats awesome.
10:29.27opus__i wouldn't mind checking out your install
10:30.02swm_Yeah I modified the app_alarmreceiver program to tailor it to advanced needs, it supports all various protocol's now other than thier single protocol, 5 Protocols to be exact.
10:30.38swm_I have 14 computers, 3 laptops, 4 desktops and the rest are Intel 2.0 x 2 Dual Hyperthreading servers
10:31.06BMSSthats really awsome swm_....
10:31.31swm_LOL dont ask how much it all costs from buying it off ebay heh
10:31.42BMSSLOL
10:33.17swm_I think each server cost me... 650 for processors, $75 for 2u fans, server board was $35, cases were $120/each (rackmount), memory was about 140 for two 512mb sticks of registered ecc memory
10:33.37BMSShmm
10:33.53BMSSpretty nice investment...:)
10:34.09swm_Yeah got surplus stuff really cheap, and it adds up
10:34.30BMSScan you speak to me on the msn
10:35.38swm_The server boards have full diagnostics, i can manage them from my desktop (literally see it from bios bootup), and it has 2 1000mbps nic's on board, oh yeah scsi cards cost me about $45 for a 64-bit card each, and the drives were cheap... bought a surplus of 10gb sca2 hard drives
10:35.56opus__IBM?
10:36.04swm_Intel
10:36.28swm_Intel SE7500WV2 board and Intel 2.0 Ghz XEON Processors (Dual 604-bit processors)
10:36.43BMSSgreat
10:37.03opus__i'm building some P4 3.2Ghz EMT64 servers with raid for about $600.. only 40gb though
10:37.07*** join/#asterisk NoRemorse (n=bah@202.161.68.2)
10:37.10NoRemorsehi all
10:37.15BMSSswm, can i have your msn ID
10:37.57swm_admin@digitaldatabits.net
10:38.12BMSSis this your chat ID
10:39.11swm_I got 140 gb amongst 14 drives on most servers and some have 40 gb ... I also have NCD Thinstars with the Xclient software hooked into my televisions. I'm hoping to get my livingston portmaster working w/ LCD control pads around my house to access other enhanced features.
10:39.47BMSSswm, please add me to your chat , my msn ID is bharatsarvan@hotmail.com
10:40.00flok420doesw anyone know a SIP phone (voip) (software) that uses the artsd daemon of kde under linux?
10:40.02opus__BMSS who the hell ar eyou
10:40.20*** part/#asterisk TK9 (n=Administ@p54B29575.dip0.t-ipconnect.de)
10:41.40swm_It's amazing what a high school dropout is capable of
10:43.37*** join/#asterisk Nix (n=Nix@81.214.255.57)
10:44.04swm_They just changed oregon law so that Marijuana and Methamphetamine is now a seperate statute, It's no longer a Schedule 1 or 2 drug anymore. Pos meth is a felonly and pos marijuana is a ticket
10:47.30swm_~molest NoRemorse
10:47.37swm_~lart NoRemorse
10:47.44opus__marijuana has always been just a ticket where i am at
10:47.45swm_They just changed oregon law so that Marijuana and Methamphetamine is now a seperate statute, It's no longer a Schedule 1 or 2 drug anymore. Pos meth is a felonly and pos marijuana is a ticket
10:47.51swm_~lart NoRemorse
10:48.36swm_~lart NoRemorse
10:50.14fishboy1669is there any specific version of gcc to use to compile asterisk ver 1.0.9?
10:50.22fishboy1669how do i find which i have?
10:50.28opus__fishboy, better then 3.4
10:50.39opus__gcc --version -V or -v ?
10:52.10fishboy1669Reading specs from /usr/lib/gcc-lib/i586-suse-linux/3.3.5/specs
10:52.10fishboy1669Configured with: ../configure --enable-threads=posix --prefix=/usr --with-local-prefix=/usr/local --infodir=/usr/share/info --mandir=/usr/share/man --enable-languages=c,c++,f77,objc,java,ada --disable-checking --libdir=/usr/lib --enable-libgcj --with-slibdir=/lib --with-system-zlib --enable-shared --enable-__cxa_atexit i586-suse-linux
10:52.10fishboy1669Thread model: posix
10:52.10fishboy1669gcc version 3.3.5 20050117 (prerelease) (SUSE Linux)
10:52.29fishboy1669would this be ok
10:52.30fishboy1669opus?
10:52.38opus__yes
10:52.51opus__although I've never ran SUSE before
10:53.08opus__try Centos 4.1 if you keep running into problems is my suggestion
10:53.22opus__where can I download a SUSE iso..
10:53.35opus__and does it includ evolution and Mono? :)
10:54.06fishboy1669it works fine from the distro version but its only
10:54.08fishboy1669Asterisk CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a built by abuild@Owens on a i686 running Linux
10:54.33fishboy1669suse iso are on there site i think
10:54.38fishboy1669get the pro version
10:54.44fishboy1669or just get from ftp
10:54.48Nixopus__: www.opensuse.org
10:54.59fishboy1669ill get
10:56.14opus__url for pro version?
10:58.31opus__hell
10:58.32opus__yeah
10:59.35fishboy1669http://www.novell.com/products/linuxprofessional/downloads/ftp/int_mirrors.html
11:00.14*** join/#asterisk pr0 (n=pr0@ndn-165-146-90.telkomadsl.co.za)
11:00.45fishboy1669it is the best distro at the mo
11:00.57fishboy1669for someone that wants least hassle
11:01.02opus__i sware by RHEL 4
11:01.18fishboy1669red hat el?
11:01.20opus__... or know as Centos 4.0
11:01.25opus__red hat enterprise linux
11:01.27fishboy1669enterprise level?
11:01.29fishboy1669cool
11:01.34opus__yes
11:01.43fishboy1669red hat is ok i used to use ver 9
11:01.47opus__but I use Mono .NET exclusively, supported by Novell/SUSE
11:01.55opus__version 9 sucked.
11:02.11opus__Centos 4.X has built in SELinux and some really good 2.6 kernel tweeks
11:02.15fishboy1669they change so quick these days its hard to get settled
11:02.24fishboy1669not heard of centos
11:02.32fishboy1669first learned on mandrake
11:02.35opus__you can make a safe bet on centos 4, all it is is RHEL
11:02.49fishboy1669and had a dable in slackware
11:02.58fishboy1669but it got too admin intensive
11:03.05fishboy1669so went for cop out of suse
11:03.06fishboy1669:D
11:03.23opus__arge, i can't downlaod 10.1 ??
11:03.41opus__i learned on slackware
11:03.48opus__back when linux was .98
11:03.49opus__:)
11:03.59opus__then i used openpbx, then freebsd
11:04.10opus__sunos, solaris.  then back to linux for the last 5 years
11:04.18*** join/#asterisk tobiasWolf (n=konversa@195.162.255.10)
11:04.37opus__I think linux is the best operating system, overall.
11:04.55lehelme2
11:05.27fishboy1669it is for low number of processors
11:05.28Zeeekfrom security sys to smoking weed we've evolved to OS ?
11:05.38fishboy1669but i heard bsd is better for 4 + processors
11:05.39ZeeekI must have missed a lot :)
11:05.40gordonjcpopus__: it's ok for a desktop
11:05.42opus__i used openbsd sorry not openpbx
11:05.50fishboy1669lol openpbx
11:05.52fishboy1669i likei t
11:05.53fishboy1669it
11:06.04fishboy1669i have many times
11:06.09fishboy1669even in big isps
11:06.17fishboy1669lastminute.com use linux
11:06.20fishboy1669and microsoft
11:06.22fishboy1669he he
11:06.25johnmgordonjcp: that almost seems completely backwards.
11:07.08johnmfishboy1669: then why does linux run on super-cluters? were talking, thousands of nodes.
11:07.34fishboy1669yes but thats on multiple boxes with 1 or 2 processors
11:07.41johnmfishboy1669: I've actually developed linux on 48-way mammoths :)
11:07.47fishboy1669but if u have one box with say 16 processors then bsd is better
11:07.49johnmfishboy1669: no, even supercomputers.
11:08.01gordonjcpjohnm: what seems backwards?
11:08.11johnmgordonjcp: not wanting linux on a server, but OK for a desktop.
11:08.24gordonjcpwell, it's not really mature enough for using on a server
11:08.24fishboy1669rumor i heard was bsd was better on multiple but am open to new info
11:08.31RoyKjohnm: may I have some of that you're smoking, please?
11:08.31fishboy1669its all subjective anyway really
11:08.39gordonjcpit's a great desktop OS though
11:08.40johnmgordonjcp: it's much more mature than many other OS's and it's software is often much more secure
11:08.47gordonjcpjohnm: yeah, right
11:08.51johnmheh
11:08.56johnmIm going to be biased anyways
11:09.05fishboy1669subjective as i said
11:09.06fishboy1669he he he
11:09.08johnmbut I would liek to see anyone prove me wrong :)
11:09.10gordonjcpever heard of an OS called "Solaris"?  Used to be called SunOS?
11:09.15johnmgordonjcp: yes.
11:09.23gordonjcpever heard of an OS called "NetBSD"?
11:09.24johnmgordonjcp: it's been made open fairly recently.
11:09.46johnmgordonjcp: Ah, the one which has been reknown for terribly security flaws and the authors are on crack.
11:09.54fishboy1669oh here we go argumenst
11:09.55fishboy1669lol
11:10.23johnmgordonjcp: I love BSD. Always have. But it's by no means anywhere near as good in some areas as, lets say, linux.
11:10.23fishboy1669wish you guys would put as much feeling into helping my asterisk compile issue ;-)
11:10.34johnmFBSD opposed to *BSD though.
11:10.39gordonjcpjohnm: if you're conecrned about security, and you want to use Linux, you need your head examined
11:10.40fishboy1669well i still say oric os is the best
11:10.50gordonjcpfishboy1669: ah, Oric 1 or Oric Atmos?
11:10.58fishboy1669oric 1
11:11.08fishboy1669i have both black and white and color versio
11:11.13johnmgordonjcp: heh, you're misinformed. And possibly refering to Linux as GNU.
11:11.15gordonjcpany BASIC with the commands ZAP, PING, SHOOT and EXPLODE has got to be good
11:11.18iDunnoFreeBSD is the least crack addled BSD.
11:11.22fishboy1669was way better computer than spectrum
11:11.26iDunnoSolaris is just plain crack addled...
11:11.30fishboy1669but sinclare marketed better
11:11.34gordonjcpfishboy1669: except in reliability
11:11.41gordonjcpjohnm: if you want secure, use VMS
11:11.47iDunnoand GNU/Linux needs a bit of polishing.
11:12.03iDunnogordonjcp: that's not secure, that's *obscure* ;)
11:12.07johnmgordonjcp: Even that has downsides, security is dependant requirement anyways.
11:12.07opus__<gordonjcp> johnm: if you're conecrned about security, and you want to use Linux, you need your head examined
11:12.18opus__gordon, you don't know about SELinux
11:12.22fishboy1669so wich is better ps2 or gameboy
11:12.36fishboy1669lol how many arguments can i start in here on what is better
11:12.42fishboy1669tomato tomaaatooooo
11:12.52johnmopus__: gordonjcp: I actually work a lot on grsec/selinux/pax and thigns like RBAC etc are a much better ACL system to most.
11:12.56iDunnofishboy1669: that seems like a stupid argument ;)
11:13.31iDunnofishboy1669: obviously the PS2 is better than the gameboy from certain perspectives, but the gameboys portability and battery life make it better for going on the road ;)
11:13.33opus__johnm cool
11:13.58fishboy1669arnt all arguments pointless
11:13.59gordonjcpjohnm: yes, but you're just sticking a padlock on a biscuit tin
11:14.02fishboy1669there all subjective
11:14.05iDunnofishboy1669: now, you should compare the 2 new handhelds for real justice, the PSP and the DS
11:14.06johnmiDunno: unless you literally mean on the road. THen a PS2 with a car-kit is much better :)
11:14.11fishboy1669anyway how do i fix this ;-)
11:14.14fishboy1669make[1]: Leaving directory `/usr/local/src/development/asteriskv1-0_1.0.9/asterisk/res'
11:14.14fishboy1669make[1]: Entering directory `/usr/local/src/development/asteriskv1-0_1.0.9/asterisk/channels'
11:14.14fishboy1669gcc -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g  -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE  -O6 -march=i686   -DZAPTEL_OPTIMIZATIONS  -DASTERISK_VERSION=\"CVS-v1-0-09/30/05-22:31:05\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\" -DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\" -DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" -DA
11:14.15fishboy1669chan_sip.c:9319: internal compiler error: output_operand: invalid expression as operand
11:14.16johnmgordonjcp: thats not true at all.
11:14.17fishboy1669Please submit a full bug report,
11:14.19fishboy1669with preprocessed source if appropriate.
11:14.21fishboy1669See <URL:http://www.suse.de/feedback> for instructions.
11:14.23fishboy1669{standard input}: Assembler messages:
11:14.25fishboy1669{standard input}:123824: Warning: partial line at end of file ignored
11:14.29gordonjcpfishboy1669: pastebin
11:14.33gordonjcp~pastebin
11:14.34jbotrumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/
11:14.52*** join/#asterisk lvp (n=lpressl@interner.SerNet.DE)
11:15.01fishboy1669but if i put it in paste bin u wont look at it ;-)
11:15.03fishboy1669he he
11:15.41opus__gordonjcp - it actually a kernel level check
11:16.28fishboy1669oh silence again
11:16.35fishboy1669has someone died
11:16.36fishboy1669he he
11:16.39fishboy1669oh shit
11:16.42fishboy1669someone has
11:16.45fishboy1669just seen on news
11:16.47fishboy1669:((((
11:17.16fishboy1669i know im being a pain but has anyone got any leads for me im getting really frustrated here
11:17.24fishboy1669please help!
11:18.27johnmfishboy1669: paste chan_sip.c to pastebin for me
11:18.41johnmfishboy1669: this isn't CVS I take it?
11:19.39*** join/#asterisk areski (n=areski@polar.es6.egwn.net)
11:19.47opus__goatse made front page on slashdot
11:20.19fishboy1669yes its from cvs
11:20.22fishboy1669v1_0
11:20.49johnmfishboy1669: paste to pastebin please.
11:20.58johnmfishboy1669: or... since it's going to be big.. send it this way
11:22.10fishboy1669which way?
11:22.33fishboy1669its the standard chan_sip.c from cvs no changes
11:22.38fishboy1669ver 1.0.9
11:22.45fishboy1669stable release v1_0
11:23.05johnmfishboy1669: not got anything to pull it down with atm.
11:23.11johnmfishboy1669: dcc please
11:24.26*** join/#asterisk MicC_ (n=sum1@CPE000c419ce901-CM000a7363f92c.cpe.net.cable.rogers.com)
11:24.30MicC_sup guys
11:24.42MicC_dumb question...how do I clear a stuck zap channel?
11:25.33pr0there was a way
11:25.40pr0type zap and press tab
11:26.37MicC_zap destray channel ...but what is the syntax on the channel? zap/5-1 ?
11:27.05pr0I think so...
11:27.18pr0the same as dialling syntax I presume
11:29.00MicC_man it jus twon't go away
11:29.04fishboy1669hi john
11:29.11fishboy1669dcc?
11:29.40fishboy1669do u have pastebin url?
11:29.44fishboy1669so i can put there
11:30.19ZeeekMicC try soft hangup
11:30.34MicC_zeeek?
11:31.03Zeeekverb noun noun
11:31.14Zeeekwhat part did you not get?
11:31.25MicC_define soft hangup?
11:31.42Zeeektype it into CLI two words
11:32.09MicC_lol...k. Misunderstood
11:32.15MicC_yah...did that one first
11:32.19Zeeek<PROTECTED>
11:32.28iDunnotry soft hangup:
11:32.36iDunno<PROTECTED>
11:32.39iDunnoexcept:
11:32.40Zeeekthere's a serious echo inhere :)
11:32.45iDunno<PROTECTED>
11:33.02MicC_yah...did that again...its still there...stupid queue.
11:33.35Zeeekhow did it get in that state?
11:33.54MicC_that will be my next step.
11:34.06Sk3tChwhat is pager_email?
11:34.21fishboy1669johnm can i email it to u
11:34.23Zeeekbefore cellphones there were these silly gadgets called pagers
11:34.37fishboy1669there well cool i still have one
11:34.51Zeeekcells are cheaper than pagers
11:34.53Sk3tChemail or the pager email where the asterisk sends voicemails ?
11:34.58fishboy1669in my collection with my orric
11:34.59fishboy1669lol
11:34.59pr0youre both right, cool, silly little devices
11:35.24pr0page a voicemail... interesting
11:35.25Zeeekpager: "bring home milk" - cellphone in restaurant: "where are you?"
11:35.26fishboy1669johnm r u about?
11:36.00fishboy1669i always wanted to set mine up so my linux box was my home alarm system and if i got broken into my pager would bleep n tell me
11:36.14fishboy1669but alas god didnt give us 10 day weeks and 30 hour days
11:36.16fishboy1669;-)
11:36.45pr0fishboy1669, thats simpler than you may thing
11:36.45gordonjcpgosh, pagers
11:36.46pr0think
11:37.00gordonjcpisn't it easier just to sms someone?
11:37.05Zeeekexactly
11:37.11pr0I rigged a motion sensor directly to the parralel port once...
11:37.17pr0it actually worked.
11:37.21fishboy1669it isnt when u are a slack git like me
11:37.29fishboy1669and easyer using sms these days
11:37.32johnmfishboy1669: yes
11:37.34gordonjcpOrange let you set up an email account, and mail from certain email addresses willl have the subject line sent to you as an SMS
11:37.42johnmfishboy1669: just doing one quick thing in work tho. bare with me
11:37.46gordonjcphow easy does it need to be?
11:37.46fishboy1669hi johnm
11:37.53pr0or it could just call you, and even playback whats happening in your house live
11:37.54fishboy1669thats ok
11:38.11fishboy1669johnm whats your email and ill email it for u to look over while im at lunch
11:38.18johnmfishboy1669: pm ;)
11:40.15*** join/#asterisk voipguy (n=voipguy@196.200.26.42)
11:41.36*** join/#asterisk Tili (i=Tili@203.101.166.65)
11:43.05MicC_got call to clear
11:43.21MicC_by restarting asterisk :(
11:43.34iDunnoawww.
11:43.43*** join/#asterisk sivana (n=sivana@mixdown.ca)
11:43.48MicC_yah...it was up for a whole month without a restart.
11:44.47Weezeypr0: I use the SPA-841 auto-answer to spy on my home and office.
11:45.37opus__NOW I crash
11:45.43*** join/#asterisk wizzup (i=wizzup@161.200.90.25)
11:45.56opus__my brain hurts anyways
11:47.27MicC_question: for my external voicemail extension, should I be using a queue to handle incoming ?
11:48.03*** join/#asterisk zotz (n=zotz@24.231.36.100)
11:50.22*** part/#asterisk wizzup (i=wizzup@161.200.90.25)
11:56.21nfi|ermesi have a hfc isdn card
11:56.43nfi|ermesand i use i4l with hisax driver
11:57.10nfi|ermesbut i have not zaptel module
11:57.26nfi|ermesis it a problemor i can use asterisk the same ?
12:00.57pr0:(
12:10.56*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
12:12.00Delvaranyone have asterisk and ser setup so you have multiple registrations on ser and are able to call though from asterisk? i seem to get weird calls setup..
12:12.23*** join/#asterisk expat_iain (n=expat_ia@194.204.96.54)
12:12.52*** part/#asterisk Maksim (n=max@213.142.207.20)
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12:30.34KeX_WorXhi
12:30.43KeX_WorXanyone using srtp with asterisk?
12:30.55KeX_WorXor ever looked at srtp support ?
12:31.11*** join/#asterisk nextime (n=nextime@213-140-6-96.ip.fastwebnet.it)
12:33.07azziedoes it support srtp ?
12:33.53*** join/#asterisk dwmw2_gone (i=ctrlprox@baythorne.infradead.org)
12:37.20KeX_WorXazzie, not really
12:37.37KeX_WorXbut it should be possible to bring srtp into asterisk via libsrtp
12:37.44KeX_WorXsrtp.sourceforge.net/srtp.html
12:38.32*** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka)
12:39.09azzieKeX_WorX, the first link on google is a $100 bounty for implementing SRTP for Asterisk ;-)
12:46.38IkarusOnly 100
12:46.41Ikarusthat is zip
12:46.49*** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com)
12:47.42drumkillaSRTP was one of the google summer of code projects
12:48.09drumkillaso, in theory, it should be coming down the road ...
12:48.34X-RobKeX_WorX - submit it as a wishlist to openbpx.org
12:48.38KeX_WorXazzie, perhaps i should think bout that ; )
12:48.44X-Robno isses with gpl stuff then.
12:48.58KeX_WorXX-Rob, k, ill look at that
12:50.29KeX_WorXazzie, can u post that link pls? cause i can't find it : / (non us user)
12:50.50nfi|ermesshould i use zaptel 1.2 with asterisk 1.2 ?
12:51.30X-Robyes
12:53.47Kattysleepy.
12:56.29*** join/#asterisk RoyK (n=roy@80.239.107.70)
13:00.10*** join/#asterisk enyc (n=enyc@ip126.0.whitehorse.co.uk)
13:01.55*** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk)
13:05.08*** join/#asterisk pr0 (n=LANmower@ndn-165-146-90.telkomadsl.co.za)
13:05.53pr0needless to say its not a straightforward process
13:06.14pr0the fact that it has to be the bristuff versions going on doesnt make it any easier
13:07.56*** join/#asterisk jonathh (n=asd@host81-154-159-222.range81-154.btcentralplus.com)
13:08.21jonathhcan anyone give my any pointers as to how to get 'RxFax' working?
13:08.39pr0ooh sorry there, no idea
13:08.39jonathhi recognise now it isn't built into (my) asterisk, but i cant seem to identy which fax app it uses
13:09.10*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
13:11.44Zeeekjonathh what have you done so far?  downloaded it ?
13:13.37*** join/#asterisk k31th (n=Kevin@flashtek-uk.com)
13:14.14*** join/#asterisk cio (n=na@adsl-072-149-159-016.sip.bhm.bellsouth.net)
13:15.50*** join/#asterisk hotgrits (n=hotgrits@192.160.238.156)
13:15.54jonathhso far.. i have found some demo code, and identified it isn't a feature of asterisk and more of an addon
13:16.05Zeeekok
13:16.08jonathhi now believe it is app_rx app_tx whcih is part of  spandsp
13:16.17Zeeekgo to spandsp.org (? let me check that)
13:16.25pr0um
13:16.29Zeeekbut you don't have spandsp?
13:16.40pr0isnt asterisk supposed to support faxes out of the box?
13:17.02Zeeekdepends on where you get the box :)
13:17.09pr0oh yeah
13:17.18Zeeekjonathh : go here http://spandsp.sourceforge.net/spandsp/
13:17.28jeffikAll: can i get some help setting up H323?
13:17.48Zeeekor maybe even here : http://spandsp.sourceforge.net/spandsp/spandsp-0.0.2pre20/
13:17.48jonathhcheers guys.. i can prop suss it now i know it aint in the box..
13:17.54jonathhi do have 1 more question tho
13:18.05jonathhthe example i saw.. used 'n' as a priority
13:18.08Zeeekget the apps, do a make and you're there
13:18.28*** join/#asterisk dudes (n=dudes@12-215-34-84.client.mchsi.com)
13:18.43jonathhafter initially setting the first one to priorty one.. is 'n' as a priorty a new thing? or maybe is it a 'inser your priorty here'
13:19.08Zeeekit used to mean 'next' but I think it got yanked at opne point
13:19.24Zeeekcan't remember if it still works
13:19.25jonathhwell i replaced it.. with 1,2,3, etc
13:19.31Zeeekgood
13:19.33pr0yeah, I've never seen it being used in a production environment...
13:19.35jonathhand i get sensible results now
13:19.52ZeeekI know it worked, then it didn't ;)
13:19.59dudesit's n+101
13:20.25dudesso if your extension goes to 6 ... you'd have 106 as the next priority
13:20.45Zeeekthat's not what he was referring to I don't think
13:21.10jonathhno..
13:21.16dudesor use t,i,o
13:21.30jonathhwell i dont thin anywayts
13:21.38jonathhi had this...
13:21.38jonathhexten => 448449861136,1,Answer
13:21.38jonathhexten => 448449861136,n,SetVar(FILE=/etc/asterisk/${UNIQUEID}.tif)
13:21.38jonathhexten => 448449861136,n,RxFax(${FILE})
13:21.38jonathhexten => 448449861136,n,System(/etc/asterisk/emailfax.sh ${FILE} info@ipdanmark.dk "${CALLERID}")
13:21.56dudesI take it not
13:21.58jonathhwhich didn't complain.. but didn't work either.. so i replaced it with actual pririty
13:22.10wunderkinn works
13:22.13wunderkinon head
13:22.19Zeeekso you have a version where n didn't work. I think it is only in HEAD
13:22.27*** join/#asterisk szer (n=Miranda@217.116.36.22)
13:22.27Zeeekya, there ya go
13:22.33szerhi all
13:22.34jonathhthats fine.. i dont think i want to be using HEAD
13:22.43dudesyou should
13:22.47dudesHead works just fine =)
13:22.51Zeeekgetting HEAD is part of life
13:22.59bendy24can someone give me HEAD?
13:23.06Zeeekusing it is another matter
13:23.09jonathhcant roll it in production.. once i know it works.. i'll stick with whatever
13:23.22dudesWe use it in production
13:23.30jonathhi believe im on 1.0.7 currently
13:23.35pr0hmm
13:23.37Zeeeksome people enjoy jumping out of planes, too
13:23.38jonathhwhich i admit is getting alittle dated
13:23.53jonathhso i aint that far behind then!
13:23.54Zeeekjonathh I just upgraded from 0.6 to 0.9
13:23.56dudesHead from yesterday
13:24.05RoyKhmmmm
13:24.11jonathhwhy upgrade to 0.9? stabiliy?
13:24.17RoyKdoes app_queue support multilanguage?
13:24.21ZeeekI don't see any diff to be honest
13:24.31pr0The echo cancellation is better in 1.0.9
13:24.36ZeeekRoyK there is only one language, English
13:24.39pr0less intensive too
13:24.52Zeeekall the others are imitations
13:24.55jonathhf or now i am gonna stick with 1.0.7 :)
13:24.58*** part/#asterisk jeffik (n=Jeff@CPE0011505c92d3-CM014350000760.cpe.net.cable.rogers.com)
13:25.00Kattyhmm
13:25.11Zeeekit was already at 40x24
13:25.21pr0RoyK, its basically the language of the recordings you use...
13:25.32pr0by default, its english
13:25.48RoyKpr0: yes, of course, but Playback and Background support searching in subdirs first......
13:26.18pr0we run a calling card platform in chinese here...
13:28.18pr0Royk, you mean like language switching?
13:28.22nfi|ermesi need someone explains me something i can t understand in the documentation
13:28.52nfi|ermesi have an hfc isdn card
13:29.01*** join/#asterisk gambolputty (n=gambolpu@72.240.241.108)
13:29.03nfi|ermesit works with isax driver (i4l)
13:29.24nfi|ermesdo i nedd zaptel module ?
13:29.33*** join/#asterisk rg1_ (n=rg1@mail.airlinksystems.com)
13:29.34ZeeekI need to download the deniro revision
13:29.35RoyKpr0: show application setlanguage
13:29.44Zeeek"taxidriver"
13:29.48pr0I see what you mean...
13:30.09pr0Sorry royk, never thought about making a box support multiple languages at once...
13:30.24pr0I would most likely use macros for that or something...
13:31.06jonathhwhy is nothing simple?
13:31.08*** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com)
13:31.52pr0jonathh, its simple compared to an analogue switchboard, or those crossbreed systems.
13:35.09jonathh:)
13:35.12jonathhim sure..
13:35.16pr0I've found that asterisk handles all standard pbx tasks very easily, and the difficulty only comes in where youre trying to do something the everyday pbx doesnt
13:35.40jonathhbut installing the fax stuff isn't just a 'do this' it is a list as long as my arm of things it needs also (ok, so maybe it is just 2~3things)
13:35.40*** join/#asterisk struct2 (n=struct@81-17-62-133.dsl.uwadslprovider.nl)
13:35.52jonathhyeah true..
13:36.02jonathhand time invested getting it working once is usable over and over.
13:36.28dudeswe've got asterisk doing Mass Faxing
13:36.29pr0jonathh, I think astlinux has the fax stuff in, its 30mb and its got 1.0.9, as well as all the drivers you can think of.
13:37.10*** join/#asterisk jimmy_deanPB (n=jhodapp@adsl-70-228-242-126.dsl.ipltin.ameritech.net)
13:38.52*** join/#asterisk Moc (n=mochouin@modemcable173.101-70-69.mc.videotron.ca)
13:40.05*** join/#asterisk jonathh1 (n=asd@host81-154-159-222.range81-154.btcentralplus.com)
13:40.16jonathh1it is i.. just known under a slighty different name
13:40.33jonathh1i have 1.0.7 i just need ot add the fax stuff
13:44.00struct2how to make my extensions.conf react on SIP response numbers
13:44.11struct2such as 408 Temporary not available
13:44.30struct2i tried to use DIALSTATUS/ CAUSECODE but both do not indicate an error significantly
13:44.34struct2and there are dubblicates
13:45.21pr0omg, I think i'm actually getting close to using asterisk on slax
13:45.24*** join/#asterisk IPmonger (n=ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net)
13:45.35*** join/#asterisk devonst17 (n=devonst1@dsl092-032-215.lax1.dsl.speakeasy.net)
13:45.51pr0that will be excellent, slax is the perfect appliance-os if you ask me
13:48.56*** join/#asterisk Raceman (n=bla@cust-02-E169.adsl.scarlet.nl)
13:49.02Racemanhi all
13:49.08jonathh1pr0
13:49.24jonathh1can i ask an impartial question? why do you think that? is it because it is lean?
13:49.26*** join/#asterisk casio_ (n=Cris@200-126-118-252.bk8-dsl.surnet.cl)
13:49.35Kattylean?
13:49.56pr0jonathh1, I guess thats also true, its more about module management to me though.
13:49.58jonathh1no fat that isn't required
13:50.04Kattyi suddenly feel like in a cooking channel.
13:50.11olivier_<struct2>not sure but you shuld take e look to HANGUPCAUSE
13:50.17jonathh1how do you feel about gentoo say?
13:50.24Kattygentoo takes forever.
13:50.30olivier_s/shuld/should
13:50.33jonathh1forever to do what?
13:50.36Kattycompile
13:50.39jonathh1ahh
13:50.41jonathh1that is true
13:50.44cioUbuntu(tm) - a MAKE bases OS.
13:50.44pr0gentoo users never use their os for something usefull, theire too busy compiling.
13:50.52cioheh
13:50.54jonathh1lol
13:50.56Kattyi run asterisk on debian.
13:51.01ciodebian rocks.
13:51.08Kattytwisted runs his on gentoo
13:51.29jonathh1i am looking for a distro to run production asterisk on.. i use gentoo currently cos it has only what i want.. and give me atleast the impression it is lean..
13:51.44jonathh1what do peeps here use for production environs?
13:51.50pr0I am quite fond of both debian and gentoo, I have some inhibitions about compiling everything in gentoo though.
13:51.52Kattyjonathh1: if you're comfortable with using gentoo, then use it.
13:52.03Kattyjonathh1: it doesn't really make much difference.
13:52.11jonathh1i spose not.
13:52.14Kattyjonathh1: asterisk is just an over glorified answering machine that sits on a desk
13:52.17Kattyjonathh1: use the desk you like
13:52.30jonathh1:)
13:52.44Racemani've a question, and can't find help by google/voipinfo/digium . I've a fresh new installation of Asterisk, installed by ports on my FreeBSD 5.4 server , and did some default sip.conf and extensions.conf configuration. But when i trying to call my budgetphone (dutch sip provider) number, something goes wrong when it ring's
13:53.03Zeeekwhat goes wrong?
13:53.09pr0gentoo is perfect, slackware too, debian is very well suited, and the others (fedora, suse etc) I wouldnt reccomend as much, but they work.
13:53.10Racemani've pasted the errors i got from the asterisk console to http://www.raceman.nl/asterisk.txt
13:53.37Kattyi know plenty of people that run asterisk on fedora core 4
13:54.05pr0sure, many people use it, it does work.
13:54.07Kattyi've not used Suse before...or slackware......or gento
13:54.08Racemanthe same errors happen when i pick up softphone, extension 10, I can't hear myself
13:54.08Kattygentoo
13:54.17Kattyi've only used mandrake and debian.
13:54.28azzieKatty, don't use slackware...
13:54.28olivier_<Raceman> where do you see an error ?
13:54.37Kattyazzie: i'll do what i want, kthx.
13:54.50Racemanolivier_, on the asterisk console, see http://www.raceman.nl/asterisk.txt for the error
13:54.50pr0my experience with astlinux was extremely pleasant, that runs on gentoo.
13:54.58*** join/#asterisk zeedo (n=zeedo@www.reboot-robot.net)
13:55.18KattyHmmhesays: are most of your asterisk boxes running on debian?
13:55.20azzieKatty, from my experience - most unmanagable system after windows :)
13:55.22mmlj4Katty: use whatever linux (or BSD) distritbution you want
13:55.27jonathh1im gonna give this astlinus thing a go.
13:55.29Kattyazzie: so?
13:55.35Kattyazzie: that's you. i'll do whatever i want.
13:55.38*** join/#asterisk ^sandro^Zzz (n=M@67.55.28.3)
13:55.40^sandro^Zzzhello
13:55.43Kattyhihi
13:55.44^sandro^Zzzi need some help please.
13:55.45Zeeekif asterisk is set up behind NAT can a SIP phone on the same side of the NAT router work? I can't get mine reachable
13:55.48azzieKatty, sure you will ;)
13:55.48^sandro^Zzzim going crazy
13:56.04jonathh1we are all going crazy
13:56.05Katty^sandro^Zzz: :<
13:56.08Katty^sandro^Zzz: we're already there.
13:56.08mmlj4Zeeek: yes, of course... ping it, is it there?
13:56.13^sandro^Zzzyup... i have about 500 users on a box
13:56.17Zeeekyep and it calls out
13:56.22^sandro^Zzz1/2 register 1/2 dont.. they register then fall into unreachable
13:56.26^sandro^Zzzi dont knwo what to do to change it
13:56.28ZeeekI've with registering, I've tried with fixed ip
13:56.34pr0er you'll probably see the majoraty foolhardely running on redhat based distros, at least over here its so...
13:56.40^sandro^Zzzon other boxes.. no problem... ( older boxes ) .. the new ones i can't understand why its doing that
13:56.54jonathh1i have to admit.. i used dead-rat initially
13:56.56Racemanolivier_ any idea?
13:57.25^sandro^Zzzanyone have any idea..
13:57.32olivier_<Raceman> just read it.
13:57.34^sandro^Zzzi have nat = yes . qualify = 5000
13:57.37Racemanokay thanks
13:57.48^sandro^Zzzstill customers are unreachable. its somethin to do with their linksys or dlink router
13:57.54^sandro^Zzzhowever on the other servers they dont have that issue.
13:58.00^sandro^Zzzit works just fine.. i can't understand why
13:58.25*** join/#asterisk simprix (n=simprix@gw001.cdsoc.org)
13:58.32simprixCan asterisk hook up to vonage ?
13:58.41dudesno
13:58.55dudesIt could but I don't think they'll let you
13:59.01*** join/#asterisk jonathh (n=asd@host81-154-159-222.range81-154.btcentralplus.com)
13:59.05jonathhback again
13:59.09jonathhbloody thing
13:59.14simprixhow is voicepulse ?
13:59.37dudesWhat are you look for
14:00.00*** join/#asterisk tangel (i=tangel@208.239.77.234)
14:00.06simprixi want to get a voip service that i can hook up to asterisk with out a ata
14:00.21^sandro^Zzzplease please please.. i will pay anyone who can help me... i just need to get this running
14:00.29pr0a pstn service?
14:00.31^sandro^Zzzwaiting on asterisk technical support right now
14:00.33jonathhwhat seems to be the problem
14:00.44simprixyes
14:00.45^sandro^Zzzsip peers go unreachable
14:00.57jonathhhow many peers?
14:00.58dudesturn off qualify
14:01.00^sandro^Zzzalot
14:01.02pr0we use diamondcard mainly
14:01.10jonathhthey dhcp?
14:01.13^sandro^Zzzyes but then they can't be reached... the problem seems to be nat
14:01.26dudesdo you have nat=yes or nat=1
14:01.29pr0where we dont we use mci
14:01.29jonathhNAT?
14:01.32^sandro^Zzzthey are dynamic .. nat=yes
14:01.39jonathhare they connected to a box on the other side of a router?
14:01.51^sandro^Zzzwell the situation is.. i have 5 boxes.. all on the same network
14:01.58jonathhok
14:02.05*** join/#asterisk viLeR (i=1000@gw.mutualasterisk.com)
14:02.06^sandro^Zzzclients have no problems with 2 boxes.. but the new 3 boxes i put in half can't register
14:02.17^sandro^Zzztheir linksys or dlink doesn't allow the traffic back in
14:02.21simprixdudes: are there any other services besides voicepulse
14:02.22Katty^sandro^Zzz: be careful when you mention you will pay people.
14:02.27^sandro^Zzzbut if we change the ip to the other box.. boom rightaway it works
14:02.30*** join/#asterisk Stoker (n=stoke@static-141-149-254-136.buff.east.verizon.net)
14:02.36Katty^sandro^Zzz: there are plenty of jackasses which will take advantage of you in here.
14:02.39dudessimprix - I don't know
14:02.48^sandro^Zzzkatty sorry just need to get this fixed thats all
14:02.51^sandro^Zzz:) but thank you
14:02.55Katty^sandro^Zzz: i understand (=
14:03.04Katty^sandro^Zzz: you will get help here. just be patient.
14:03.04^sandro^Zzzi been at it for days
14:03.14^sandro^Zzzya but i have about 250 people not working right now
14:03.14dudessimprix - I'd suggest commpartners or txlink myself
14:03.17^sandro^Zzzi need to get it fixed :(
14:03.18jonathhit isn't some fancy router intrusion detection or some such shit?
14:03.26jonathhyou tried resetting the router/modem?
14:03.29*** join/#asterisk jeffik (n=Jeff@CPE0011505c92d3-CM014350000760.cpe.net.cable.rogers.com)
14:03.32^sandro^Zzzdid that... been there done that
14:03.44^sandro^Zzzno lucky.. that + proxy auth errors
14:03.51^sandro^Zzzother users have no issues
14:04.01^sandro^Zzzon the old box most register successfully
14:04.29jonathhcan you ping the handsets
14:04.34jonathhfrom the new boxes?
14:04.49jonathhit isn't a shit network cable or switch that it is connected to?
14:04.54dudesif their router blocks ICMP's (as most do) you won't be able to ping it
14:05.04^sandro^Zzzi can ping the ips yes
14:05.15^sandro^Zzzbut they still say unreacable
14:05.24^sandro^Zzzfunny thing is.. my other routes are doing that problem
14:05.27*** join/#asterisk voipguy (n=voipguy@196.200.26.42)
14:05.31^sandro^Zzznot router sorry .. asterisk box
14:05.49^sandro^Zzzdifferent softswitches behind the exact same network with same firewalls etc. no problem
14:05.49dudesAssuming asterisk isn't behind a restrictive router
14:05.55^sandro^Zzznope.. not at all
14:06.01jonathhit isn't a shit cable that is dropping packets or anything like that?
14:06.10jonathhhmm
14:06.10^sandro^Zzznope... i had problems with DSL too
14:06.31^sandro^Zzzmy home was doing that.. all of a suddent on its down it started working ok... im slowly going to hang myself :P
14:06.48jonathhi need to understand your model more.. you have x handsets connecting to an asterisk box on the other side of a router/dsl line?
14:07.10*** join/#asterisk |cleric| (n=dacleric@p548293AD.dip0.t-ipconnect.de)
14:07.20^sandro^Zzzno.. i have asterisk boxes on my network ( static ips )
14:07.40^sandro^Zzzi have x handsets behind their own routers ( cable / dsl ) whatever.. connecting to my X box
14:07.57^sandro^Zzzthey all register and connect to the box
14:08.08^sandro^Zzzbut minutes later qualify tells me they are unreachable and they can't make calls or receive
14:08.41jonathhany handsets on the same side of the box as the asterisk machine is?
14:08.41^sandro^Zzzwhen they first register they can send and receive calls
14:08.49dudesif you do iptables -L on the asterisk box
14:08.54jonathhi think i had this problem once.
14:08.56^sandro^Zzznothing iptables is empty
14:09.23^sandro^Zzzthey can send receive.. but after a min or two.. boom line on their phone goes dead
14:09.23*** join/#asterisk MuppetMaster (n=MuppetMa@62.37.168.255)
14:09.26MuppetMasterHello
14:09.43jonathhis it a fixed amount of time?
14:09.44MuppetMasterIn Asterisk-1.2 what has happened with MusicOnHold capabilities?
14:09.46jonathhwhat if you are in a call?
14:09.48^sandro^Zzz?
14:09.51jonathhdoes it still die?
14:10.08^sandro^Zzzjanathh not sure what you mean by fixed amoutn of time
14:10.20^sandro^Zzzif you are in a call its perfectly fine
14:10.20jonathhfrom when it registers.. to when the lines dies..
14:10.21dudesMuppetMaster - look at the sample configs
14:10.24dudesit will become clear
14:10.24^sandro^Zzzit wont drop the call
14:10.34^sandro^Zzzno idea.. its about the same for everyone yes
14:10.35jonathhdefinately a routing issue.
14:10.46^sandro^Zzzoh ya.. i know.. just can't figure out why
14:10.57MuppetMasterdudes I see them and seems to be entirely different.  Is it no longer possible to use format_mp3?  Or to play from a Shoutcast stream?
14:11.10jonathhand you say if you change the IP of the new machine to that of one of the older ones.. it starts to work?
14:11.12dudesI'd imagine it is
14:11.22^sandro^Zzzyup.. exactly
14:11.24dudesthe syntax though different, functions very similiar
14:11.26^sandro^Zzzveyr strange
14:11.39MuppetMasterdudes My config no longer works and when I do as the Wiki says I get an error that the config file is deprecated and to look at the example.
14:11.50MuppetMasterdudes But then there is nothing on the example or wiki and I am stuck.
14:11.53^sandro^Zzzasterisk version is older on one other box.. but the other box i have .. ( i have 2 that works fine .. 3 that dont )
14:12.04^sandro^Zzzit works on box 1 & 2
14:12.13^sandro^Zzzbox 2 has a new version of asterisk ( latest ) and it works
14:12.18^sandro^Zzzbox 3,4,5 dead :(
14:12.23^sandro^Zzzsame network same routers etc.
14:12.27jonathhmaybe the router has a physical limit on what it can route and you have reached it?
14:12.29^sandro^Zzzon this end.. client same on the other end
14:12.36^sandro^Zzzno way..
14:12.42jonathhalthou.. if that was the case.. as reboot would make the new box work.. for a time..
14:12.46azzie^sandro^Zzz, did you try running a sniffer to see whether anything is still coming from "dead" phones after they go dead ?
14:12.47^sandro^Zzzsame routes we use to handle Gig E traffic
14:12.50dudesin the 1.2 folder there is a folder called configs
14:13.10dudesand it's very clear what todo to resolve your issue
14:13.20MuppetMasterYes, I am looking at that but it is not helpful unless you want to use mpg123, which I do not.
14:13.29jonathhi think you need ot spend some time packet sniffing
14:13.37Stokerhmm, no one has a suggestion to what hard phone to get?  usually such inqueries starts a flood of "the one I got is the best.." comments
14:13.39^sandro^Zzzits sooo strange
14:14.03^sandro^Zzzdriving me crazy
14:15.27azzie^sandro^Zzz, did you try running a sniffer to see whether anything is still coming from "dead" phones after they go dead ?
14:15.46dudesMuppetMaster - how did you play mp3's before
14:16.08^sandro^Zzzya they constantly try again however it shows retransmissions.. all the way to 7
14:16.11^sandro^Zzzthen it exits
14:16.20^sandro^Zzz7 retries and then dead
14:16.30^sandro^Zzzit tries every say 20 secs or whatever i think
14:16.37^sandro^Zzzwhatever i set registration to
14:17.00azzie^sandro^Zzz, do you register phones into asterisk ?
14:17.07^sandro^Zzzyup.. its running mysql realtime
14:17.33azziedo you see registrations coming from phones after they go dead ?
14:17.43MuppetMasterdudes I used format_mp3 as specified in the Wiki here:  http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf
14:17.46MuppetMasterworked great before
14:19.25^sandro^Zzzim going crazy.... so dahm crazy.. :( im loosing my mind... wooo wooo wooo.. hey.. im going crazy... so dahm crazy.. im loosing my mind
14:19.51dudesmake; make install of /usr/src/asterisk-addons/format_mp3.
14:20.10*** join/#asterisk dos000 (n=dos000@CPE00119572fd49-CM00137186e53a.cpe.net.cable.rogers.com)
14:20.12dudesIf they have addons for 1.2
14:20.26MuppetMasterdudes  Did that of course when I installed 1.2, but I keep getting the deprecated warning at startup and silence when I call MusicOnHold.
14:20.32MuppetMasterdudes Yes, 1.2 has addons.
14:20.35expat_iainIs there a way I can limit the number of active lines to and from a softphone?
14:20.54*** join/#asterisk samueltc (n=samuel@laid.izisolution.com)
14:20.58dudesSo format_mp3 depricated
14:21.48expat_iainFor example, Xten has three lines per phone. I need to stop all but one line from being used.
14:21.49^sandro^Zzzalot of my usrs are showing errors retrans_pkt: Maximum retries exceeded on call
14:21.52MuppetMasterdudes I do not know, as the example config seems to elude to it but does not give a clear idea of how to use it.
14:21.54^sandro^Zzzon the sip CLI>
14:22.12RoyKexpat_iain: show application setgroup
14:22.18MuppetMasterthe exampe config gives [native] mode=files directory=/var/lib/asterisk/moh-native
14:22.19dudesI asked what warning you get when you load asterisk
14:22.29MuppetMasterwhich was similar to the config before but then it says it is missing a [default] section.
14:22.35expat_iainThanks. Will take a look at that.
14:22.42RoyKWARNING: extreme user error! please shoot at sight
14:22.50MuppetMasterNow I am not getting a warning, but an error message that there is 'No class: default'
14:23.19MuppetMasterbut if I do [default][native]mode=files directory=whatever, then I get the fact that [default] needs a directory.
14:23.30MuppetMasterThe config file example is not very useful, or I am just being completely dense.
14:23.36dudesthen put [default] on the first line directory=/usr/lib/null
14:23.53dudesor use default for your files
14:25.10Kattydo do do
14:25.24Kattyanyone heard of the PRESCRIBE programming language?
14:25.37*** join/#asterisk fordvoice (n=channelv@rrcs-70-61-133-91.central.biz.rr.com)
14:26.20dudesMuppetMaster - make sure you point default to a real dir that's empty and setmusiconhold(class)
14:26.26bjohnsonda da da
14:26.28bjohnsonno
14:26.45bjohnsonMuppetMaster: my default mode is "co class" too
14:26.48bjohnsonerr
14:26.53bjohnsonscrewed that up
14:26.56bjohnsonMuppetMaster: my default mode is "no class" too
14:27.03MuppetMasterokay, will give it another shot here, very confusing.
14:27.10*** join/#asterisk brent21 (n=Brent21@70.88.149.221)
14:27.24brent21Anyone have any experience with bandwidth.com's Flex T?
14:27.29MuppetMastercd
14:28.16fordvoicehow are we this morning
14:28.48fordvoiceI have heard of them never heard anything bad
14:30.10fordvoicehey brent21
14:30.18brent21hey
14:30.24fordvoiceI called a friend of mine
14:30.27fordvoicehe has that service
14:30.34brent21yeah?
14:30.42fordvoicehe said it is ok
14:30.47fordvoicewhat is your question
14:30.59brent21I was just curious with SLA's and stuff
14:31.04*** join/#asterisk psyco-obiwan (n=cschnee@2001:4060:4419:b1:0:0:0:2)
14:31.07brent21How reliabile it is?
14:31.07MuppetMasterWell no error message on load, the channel answers, no error message says it is starting class [native] which is what I used in my musiconhold.conf, but I get silence.
14:31.17psyco-obiwanGood afternoon...
14:31.21brent21we are considering a PRI from Verizon, and other creative options such as bandwidth.com
14:31.31brent21fear is w/ bandwidth.com QOS
14:32.12fordvoicehe says that BW.com and Verizon should work but he never had any issues with QOS
14:32.16psyco-obiwanDoes anybody know how to find out what Revision a PCI bus has, given that most dmesg always report 2.1 (but digium says I need a 2.2) ?
14:32.35brent21cool, thanks
14:32.49fordvoicenp
14:33.54nfi|ermesZT_SPANCONFIG failed on span 1: No such device or address (6)
14:36.59*** part/#asterisk MuppetMaster (n=MuppetMa@62.37.168.255)
14:37.45fordvoice?
14:37.53fordvoiceI have an ibound 866 number
14:38.02fordvoicemy account with sellvoip is IAX
14:38.30fordvoicehow do I forward that number to a pstn number with voipjet as I call 9+1+###-####
14:39.04fordvoiceexten => 18662317407,1,Answer
14:39.04fordvoiceexten => 18662317407,2,Dial(SIP/201,30)
14:39.04fordvoiceexten => 18662317407,3,Hangup
14:39.14fordvoicerigth now it dials my ext 201
14:39.25fordvoiceI want to have that number call a pstn instead
14:39.34RoyK~pb
14:39.36jbotmethinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/
14:39.51fordvoiceok
14:39.53fordvoicesorry
14:40.09struct2how to express only the first digit of an exten
14:40.18struct2${exten:1} strips the first char
14:40.23struct2how to only hold the first digit
14:40.38RoyKer
14:40.39RoyKOct  4 16:38:48 WARNING[2265]: res_musiconhold.c:870 local_ast_moh_start: No class: jangarbarek
14:43.40*** join/#asterisk albi (n=albi@dslb-084-056-132-079.pools.arcor-ip.net)
14:50.08*** join/#asterisk }cytrak{ (n=kvirc@208.63.19.172)
14:51.46jake1932jangarbarek? is that default in Norwegian?
14:52.14*** part/#asterisk brent21 (n=Brent21@70.88.149.221)
14:52.57*** join/#asterisk funxion (n=nunya@mtnuser.icgws.com)
14:53.30funxionhi
14:53.39*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
14:53.39*** mode/#asterisk [+o anthm] by ChanServ
14:54.57*** join/#asterisk jhiver (n=jhiver@AStDenis-105-1-4-4.w193-253.abo.wanadoo.fr)
14:55.12jhiverguys, are there any IAX hardphones out there that actually work?
14:55.31jhivermost of what I see on the web seems like total hassleware
14:56.24*** join/#asterisk robb_ (n=robb@matrix.netsoc.tcd.ie)
14:56.27robb_hi all
14:56.36jhiverhi
14:56.48BeirdoI'm not all, but hi
14:57.02robb_is it alright to ask questions here? (he asks ironically)
14:57.24blitzragerobb_: don't ask to ask -- just ask
14:57.30robb_coola
14:57.38^sandro^Zzzdahmit....
14:57.42*** join/#asterisk eivindtr (n=wingnut-@062016241059.customer.alfanett.no)
14:57.50^sandro^Zzzhey whats a good way to set it so the keepalive is on
14:57.56^sandro^Zzzqualify right ?
14:58.00^sandro^Zzzis there another way
14:58.04blitzragequalify=yes
14:58.14^sandro^Zzzthis has to be a server issue :(
14:58.14*** join/#asterisk phpboy (n=shane@c1-86-2.tbnb.isadsl.co.za)
14:58.17phpboyhi friends
14:58.47phpboywhat's the extention for '*8' (answering a ringing extention)
14:58.54robb_having a problem getting fax detection to work. have followed scott laird's tutorial but it's still not working. when i run zap show channel it says that fax handled is set to no. what does that mean? is there some configuration option somewhere i'm missing? faxdetect in zapata.conf is set to both
15:00.35*** join/#asterisk Abbas (n=Abbas@203.81.225.107)
15:01.29*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
15:02.45Beirdomorning, blitzrage
15:02.52Beirdohow's life in the burbs?
15:03.16Abbashi all
15:03.39Abbashow can i force the asterisk to bridge the calls and  for non bridged calls
15:04.06Ariel_blitzrage, nice book. Just got it today.  (Took a some time to get it but it's here).
15:06.00*** join/#asterisk AsteriskNoob (i=BoredBoz@207-114-232-10.gen.twtelecom.net)
15:06.06AsteriskNoobmorning everyone!
15:06.07MeatyAbbas : Use same codec on both sides !
15:06.20MeatyMorning !
15:06.58blitzrageAriel_: wooohoooo!  Glad it made it. People ordering now should get it within' a couple of days now that the places have them in stock
15:07.19Ariel_blitzrage, great to hear that.
15:07.34blitzrageAriel_: I didn't realize how long it usually takes to get books distributed from the publisher to the stores. They even told us that it usually takes some time... so thats a normal process
15:07.39Ariel_blitzrage, now I need time to go to the bathroom and so some reading
15:07.44blitzragehehehe :)
15:07.50blitzrageAriel_: you heading to Anaheim for AstriCon ?
15:08.11Ariel_blitzrage, no can't afford the ticket and room stays.
15:08.27blitzrageBeirdo: oh things are ok... I gotta get to doing some programming today, and clean up my room since my accountant is coming over to show me the QuickBooks ropes :)
15:08.40Beirdooh fun :)
15:09.05filemy cable is out!
15:09.07BeirdoI hope to get a copy of your book sometime, but not immediately, too preoccupied with engagement rings, etc.
15:09.20tzangerfile: put your cable back in your pants
15:09.21blitzrageBeirdo: but the book is so cheap! :)
15:09.25blitzragetzanger: LOL
15:09.27filetzanger: nooooooo
15:09.46Beirdoblitzrage: and I have so little time to read it, so I'll wait until I have a WEE bit more time :)
15:10.10blitzrageBeirdo: fair enough :)
15:10.18blitzrageBeirdo: or just go to Astricon and get a free copy :D
15:10.22Beirdohehe
15:10.29blitzrage(free is a relative term :D)
15:10.34Beirdosorry, I have my October flight already booked
15:10.42BeirdoYYZ-SJU via MIA
15:10.50blitzrageMissing In Action!
15:10.51file2 more days till my Birthday!
15:11.04blitzrageBeirdo: San Jose?
15:11.05Beirdoyeah, that airport was MIA for a day or two
15:11.12BeirdoSan Juan, PR
15:11.15blitzrageoh right
15:11.17BeirdoSan Jose is SJC
15:11.23blitzrageBeirdo: finally getting to see the g/f eh?
15:11.24pygrammerAsteriskNoob: I assume you use Asterisk.
15:11.30Beirdosecond time down
15:11.32AsteriskNooboh yeah
15:11.34Beirdobringing ring
15:11.36Beirdoand mom
15:11.38Beirdo:)
15:11.42blitzrageBeirdo: good luck :)
15:11.42pygrammer:)
15:11.47fileoh cable's back
15:11.55Beirdothanks.  I'm not worried
15:11.58Beirdoshe'll say yes
15:12.00blitzrageBeirdo: coolio
15:12.04blitzrageBeirdo: always good to know :)
15:12.10Beirdoher father will give me a LONNG lecture, but it
15:12.14Beirdoit's worth it
15:12.14*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfj3b.dialup.mindspring.com)
15:12.27blitzrageBeirdo: what, you didn't ask his permission first? :)
15:12.46Beirdokinda hard to do when she already agreed before I had a chance
15:12.47Beirdohehe
15:12.58Beirdonot like I'm down there all the time
15:13.17Beirdoit was pretty much a sure thing before I left there the first time
15:13.30Beirdoit's not like she's 18 anyways
15:13.35AsteriskNoobalthough i'm not exactly a noob anymore
15:13.39Beirdohe'll be fine with it
15:13.42*** part/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
15:14.00*** join/#asterisk casio_ (n=Cris@200-126-117-11.bk8-dsl.surnet.cl)
15:14.01Beirdohehe
15:14.04pygrammerAsteriskNoob: Do you use Asterisk as a personal or business tool?
15:14.24AsteriskNoobso, problem... maybe somebody has a fix.... my 4 fxs ports off of a TDM400 are not doing call waiting OR message waiting
15:14.38AsteriskNoobpygrammer: i have a 20 extension business system set up
15:14.44*** join/#asterisk Blackthorn (i=blacktho@ws-10.smyth.net)
15:14.51pygrammerAh, okie-dokie then.
15:15.02AsteriskNoobpygrammer: all with cisco 6970's and a pri off the back end
15:15.57Beirdoblitzrage: I'm not gonna ask for his PERMISSION anyways.  only his blessing
15:15.57pygrammerNice. I'd like one of those 6970s, but I'm stuck with a GXP-2000 by Grandstream. Oh well, it works, and I don't rely on it.
15:16.00blitzrageBeirdo: thats a better way of going about it :)
15:16.03AsteriskNoobany idea's on the analog problems?
15:16.06jake19327960?
15:16.10BlackthornHello, I am moving providers from a provider that has no 911 servce on there voice pri's to one that does.  I have dedicated #'s that are still registered across town to the corespodnign address.. when an ata dial's 911 sends out the pri you think it will work?
15:16.13pygrammerErr, yeah jake1932
15:16.19pygrammerI think we're both dyslexic :D
15:16.21AsteriskNoobjake1932: yes, typo
15:16.30Beirdoif he says no, to quote her: "He'll have to think again, we ARE getting married" :)
15:16.31AsteriskNoobit's early
15:16.34pygrammerI'm not; I just had a senior moment.
15:16.35Blackthornthe callder id will come up correctly i do know that.. but will 911 get the right address/ani info or will they get the local pri locaiton?
15:16.40*** join/#asterisk mbranca (n=matteo@host-210-mi.espia-net.net)
15:16.45phpboywow, this is so cool... asterisk greps the last 4 digit's dialed from the PSTN, very nice
15:16.55pygrammerYou know, at 15.75 years old, you tend to forget a lot of things quickly.
15:17.09jake1932only downhill from there
15:17.13AsteriskNoobi'm 22 but been at this crap for 14 years
15:17.13pygrammerYeah.
15:17.20phpboy:/
15:17.21AsteriskNoobmy memory doesnt work anymore
15:17.26pygrammerheh
15:17.34AsteriskNoobi have to make tons of text files to make notes with
15:17.35pygrammerTime to collect our Social Security benefits, eh?
15:17.37blitzragepygrammer: you're that young? you should be absorbing things like a sponge
15:18.04pygrammerblitzrage: Well, more than 15.75 -- 15.94 or so.
15:18.11blitzragelol
15:18.13pygrammerBirthday's in a few weeks.
15:18.18blitzrageeither way ... :)
15:18.34AsteriskNoobso none of the couple hundred people in here has any clue why my caller id and Message Waiting Indicator wont work on my FXS ZAP channels?
15:18.34pygrammerWhen I turn 16, jeez, I'm going to be in terrible shape...
15:18.47blitzrageenjoy it while you can
15:18.48pygrammerI'm not going to remember my name.
15:18.51Beirdoand we'll stay off the sidewalks for a while
15:18.59pygrammerhah
15:18.59AsteriskNoobthey worked before... wait call waiting not caller id
15:19.03AsteriskNooback cant think
15:19.17jake1932usually at 21 you don't remember your name - but it goes away a couple days after
15:19.19pygrammerYeah, it's 11:17; too early for any of this Asterisk shit.
15:19.33blitzragepygrammer: 11:19
15:19.36pygrammerjake1932: yeah.
15:19.41AsteriskNoob9:21
15:19.53AsteriskNoob(MST - Synced with time.nist.gov)
15:19.57pygrammerblitzrage: Oh, excuse me... I don't have NIST time on my Winblows setup. ;)
15:19.59blitzrageBeirdo: thank goodness I'm not leaving for Astricon until Thursday... I get to watch the game
15:20.07blitzragepygrammer: you should!  I do :)
15:20.14AsteriskNoobbe back later
15:20.15pygrammerHeh.
15:20.16jake1932pygrammer:you can sync to nist
15:20.32pygrammerI have it on my phone right next to me... but I suppose I should.
15:20.33blitzrageas long as you're using something newer than Win98SE
15:20.38pygrammerTwo minutes makes a huge difference.
15:20.45blitzragepygrammer: it does
15:20.46jake1932pygrammer: on xp click on Internet Time
15:20.48blitzragehehe
15:21.02Beirdoblitzrage: I'll be cheering for the Sens.
15:21.03pygrammerHmm, yeah, I did. It's synced to time.windows.com.
15:21.04Beirdo:)
15:21.07pygrammerNo wonder it's off. ;)
15:21.17blitzrageBeirdo: we are no longer friends
15:21.47Beirdoblitzrage: sorry, dude, I lived in Ottawa for a long time. :)
15:21.58pygrammerAh, it says 11:21... I guess it's just my hardware clock.
15:22.01blitzrageBeirdo: you're still talking to me?
15:22.03Beirdonow live 2 blocks from the ACC and still wore the Sens jersey to work today
15:22.07blitzrageBeirdo: :)
15:22.08pygrammerThis is an old PC, so it's not surprising it can't keep time.
15:22.20BeirdoI like pissing off people who take hockey too seriously
15:22.27pygrammer4 years old this month.
15:22.34pygrammerAnyway, bbl -- omelette.
15:22.36blitzrageok... I've procrastinated too long -- off to start working for the day
15:22.36Beirdowhat would be nice is if Toronto and Ottawa AREN'T paired up in the first round.
15:22.52Beirdothat way at least one of them has a chance, they keep wearing each other down
15:23.02blitzrageBeirdo: true
15:23.17*** join/#asterisk mbranca (n=matteo@host-210-mi.espia-net.net)
15:23.18Beirdolet them meet in second round when they have momentum
15:23.30Beirdoor even better... in the finals
15:23.32Beirdoheh
15:23.38bendy24go leafs go
15:24.11*** part/#asterisk robb_ (n=robb@matrix.netsoc.tcd.ie)
15:24.14Beirdobendy24: you better not miss the game tomorrow
15:24.15Beirdo:)
15:24.20bendy24oh dont worry
15:24.35bendy24i'll be there to see sundin kill that red haired fairy
15:24.47*** join/#asterisk TrevorSHarrison (n=trevorsh@24.49.36.218)
15:25.10jonathhis it home time yet?
15:25.10Beirdohehe
15:25.15*** join/#asterisk da_monumental_1 (n=da_monum@rrcs-24-172-102-198.midsouth.biz.rr.com)
15:25.36Beirdohey, at least we HAVE hockey to fight about this year
15:27.18bendy24its the best game you can name
15:27.26bendy24the good ol' hockey game
15:27.47*** join/#asterisk klasstek (n=nunyobiz@sta-206-168-218-202.rockynet.com)
15:28.23synthetiqis it possible to have background music play instead of a ring going back to the dialer?
15:28.32*** join/#asterisk mbranca (n=matteo@host-210-mi.espia-net.net)
15:29.19*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
15:29.30mogormanyes
15:29.33mogormanuse the m flag
15:29.36mogormanin dial statement
15:30.12*** join/#asterisk Repuation (n=root@p5090A896.dip0.t-ipconnect.de)
15:32.20RepuationHi there. I need help. Have problems compiling zaptel.
15:32.49mmlj4you're in good company
15:33.03shido6he
15:33.05shido6heh
15:33.09shido6so whats wrong, Repuation ?
15:33.32synthetiqill go read up on the m flag
15:33.40shido6for music?
15:33.45shido6what do you want to do?
15:33.53shido6just add |m
15:34.01Repuationwhen i run make i get an error
15:34.07Repuationit says make[1]: *** No rule to make target `modules'.  Stop.
15:34.11shido6what error do you get, Repuation
15:34.13Repuationand i don't like that
15:34.21shido6what kernel?
15:34.31Repuation2.6.11. and so on
15:34.40shido6synthetiq, add ||m to te end of your dial statement
15:34.58shido6or |20|m
15:35.06shido6for te length of 5 rings
15:35.13shido6but it plays music
15:35.15Repuation2.6.11.4-21.9-default
15:35.19shido6or 20k milliseconds
15:35.25synthetiqwhat kind of music default music?
15:35.30shido6Repuation, you have to follow the readme for 2.6
15:35.39shido6synthetiq, if you want.... or you can upload your own music
15:35.49Repuationwhere can i find it? asterisk.org?
15:36.01synthetiqwell where do i specify the music to play
15:36.20*** join/#asterisk mbranca (n=matteo@host-210-mi.espia-net.net)
15:36.24shido6were do you keep your music files?
15:36.46shido6"/var/lib/asterisk/mohmp3"
15:36.59shido6upload music you want into /var/lib/asterisk/mohmp3
15:37.19shido6then edit "/etc/asterisk/musiconhold.conf"
15:37.32shido6if you choose default
15:37.47*** join/#asterisk SwK (i=bbgewy@12-219-144-126.client.mchsi.com)
15:37.51shido6it will play the first mp3 in that folder using alphabets and numbers for sort order
15:38.02*** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm)
15:38.16shido6got any fxs / fxo interfaces on this box of yours?
15:38.25shido6if so, you also need to edit "/etc/asterisk/zapata.conf"
15:38.32*** join/#asterisk ms345 (n=mike_sim@64.74.198.10)
15:38.38shido6and add musiconhold=default /random/ whatever u want
15:38.46shido6see how this works?
15:38.54*** join/#asterisk syle (n=blag@unaffiliated/syle)
15:39.20*** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm)
15:39.23*** join/#asterisk cripito (n=ncripito@ip67-154-143-190.z143-154-67.customer.algx.net)
15:39.33shido6Repuation, you need to follow the /usr/src/zaptel/README.udev
15:39.47Repuationoh thanks :)
15:39.53fiber0ptiCan the four softkeys on the bottom of the LCD of a Cisco 7960 be programmed for specific things in asterisk?
15:40.06Repuationthanks for your help
15:40.07shido6not that I know of unless u can rewrite firmware
15:40.27shido6i mean call forward all works
15:40.28shido6new call
15:40.29shido6redial
15:40.33shido6te blind transfer
15:40.35shido6transfer
15:40.37shido6conference
15:40.39shido6they all work
15:40.49shido6but to make tem DO asterisk only things
15:41.01shido6dunno how :0
15:41.16doughecka_shido6: do you ever have issues with 2100hz tones disabling your echo can?
15:41.21doughecka_and when thats not a fax?
15:42.17shido6LOL!
15:42.22shido6LOL!!
15:42.41shido6thats crazy.
15:42.44*** join/#asterisk _Thor (i=Christia@user-vc8fl7l.biz.mindspring.com)
15:42.49doughecka_... :P
15:43.01shido6faxing, eh?
15:43.02shido6wow
15:43.11shido6I dont know anything about NuFone's fax service
15:43.30doughecka_I read somewhere that telco LD's echo cans will send out that tone to turn off other echo cans
15:43.50doughecka_this is unrelated to nufone and faxing =D
15:43.54shido6tell me you are not using spandsp
15:44.01doughecka_nope
15:44.07doughecka_I am not recieving faxes
15:44.25shido6so u need an echocan force or something
15:44.29doughecka_yea
15:44.39doughecka_but I was just wondering if you have ever run into this issue
15:44.48doughecka_I know nufone has a billion PRIs.... =D
15:45.21*** join/#asterisk Attila_Kovacs (n=kovacsat@dsl51B678B4.pool.t-online.hu)
15:46.10*** join/#asterisk smarta (n=smarta@198.65.201.34)
15:46.13synthetiqi know hwo music on default works but i wanna play a specific music on hold for the phone call isntead o the ring
15:46.20synthetiqill read up on v-i
15:46.37smartahello?
15:47.00doughecka_?olleh
15:47.05shido6SetMusicOnHold ?
15:47.22drishow i can verify that my x100p works now?
15:48.01shido6make a call
15:48.02shido6throug it
15:48.06shido6or receive a call through it
15:48.21phpboyguys, what's the extention for answering a call ringing at a remote extention?
15:48.41*** join/#asterisk gambolputty (n=gambolpu@72.240.241.108)
15:49.13synthetiqhmm looks like  you can do  m(yourmusic)
15:51.40*** join/#asterisk argos73 (n=mike@adsl-70-228-109-5.dsl.akrnoh.ameritech.net)
15:52.33Qwellconnecting an iaxy to the second NIC in a box would be a crossover cable, right?
15:53.28wunderkinya
15:54.01*** join/#asterisk fugitivo (n=ajf@201.255.102.160)
15:54.10Qwellk, heh
15:55.04fugitivoi used a pendrive to install gentoo and asterisk and I realized that my machine doesn't boot from USB
15:55.15Qwellfugitivo: awesome
16:00.54*** join/#asterisk pegger (i=pegasus@66.92.40.210)
16:02.00jonathhpop question for the willing.. on the spandsp installation page.. where it says
16:02.04jonathhext, put app_rxfax.c, app_txfax.c and Makefile.patch in your Asterisk apps directory. Use the command:
16:02.16jonathhdoes that mean in the src dir for asterisk?
16:02.17jonathhpresumably
16:04.54*** join/#asterisk santiago (n=santiago@208.195.215.98)
16:07.35*** join/#asterisk Tili (n=Tili@202-133-67-167-dialup.sat.net.pk)
16:08.30*** join/#asterisk Timoti (n=asqsa@pool-71-110-46-117.lsanca.dsl-w.verizon.net)
16:08.36Hmmhesaysdamn web stuff, why don't html forms return the full file path
16:08.41TimotiHi everybody
16:09.11distortionis there an easy way to set caller-id depending on what extension is dialed?
16:09.42*** part/#asterisk Attila_Kovacs (n=kovacsat@dsl51B678B4.pool.t-online.hu)
16:09.45*** join/#asterisk yaaar (n=chatzill@12-216-227-226.client.mchsi.com)
16:09.49yaaarword
16:10.02KattyHmmhesays: mew.
16:10.02mutilatorHmmhesays return full path?
16:10.11*** part/#asterisk santiago (n=santiago@208.195.215.98)
16:10.21mutilatorfor what
16:10.46mutilatorit only returns what ya tell it to.. no more no less
16:11.06jake1932distortion: yes
16:11.27jake1932distortion: use SetCallerID(callerID)
16:12.04mutilatoringenius name eh
16:12.08Timotia question .. I know that asterisk has no limits related fxo and fxs connection .. But if you this about connecting 50 FXS ... it is not very cheap to build it .. I think for that cheap PBX can be found .. or am I wrong ?
16:12.37mutilator.. ya wrong
16:12.43Timotiwhy ?
16:12.52Timotihow much does 50 FXS cost ?
16:13.03mutilatorget a te210 for like.. $900
16:13.04distortionjake: perfect, thx
16:13.05Timotior am I still wrong ?
16:13.11mutilatorand 2 ta624's for $1300/ea
16:13.12jake1932np
16:13.20mutilatorthen 50 $2 phones
16:13.26TimotiI am not talking about FXO .. I am talking about FXS
16:13.54Timotior am I still wrong ?
16:13.57mutilatorum
16:14.03mutilatorwhat do ya think fxo/fxs are?
16:14.10*** join/#asterisk Mauro__ (n=mauro@oliver.altascumbres.cl)
16:14.15Mauro__Hi
16:14.36Mauro__I have some asterisk noobs questions :P
16:14.37TimotiFXS is the end poing for phone to call .. FXO is for PSTN or GSM gateway or ISDN or etc or ?
16:14.55mutilatorya..
16:15.15mutilatorta624 is a channel bank
16:15.24mutilatorturn your pri into 24 fxs's
16:15.39Mauro__my company wants to do voip stuffs where should I begin reading? :P
16:15.48jake1932~docs
16:15.49jbothmm... docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
16:15.49NuggetYou plug a phone into an FXS and you plug a dial tone into an FXO
16:16.27Timotiwhat is the brand name of it ? TA624
16:16.31mutilatoradtran
16:16.47Mauro__thanks :P
16:16.49mutilatoryou can get them in quite a few configs
16:16.56mutilator16fxs and 8 fxo
16:16.58mutilatoror 24fxs
16:17.04mutilatorand some other things i think
16:17.13Timotihimm .. the strange stuff is .. here PRI is very very expensive
16:17.34*** part/#asterisk darkskiez (n=darkskie@194.247.78.146)
16:17.40jake1932Timoti: in the US?
16:17.50mutilatoryou can still use pots for incoming/outgoing
16:18.03mutilatorbe kinda retarded but ya can do it
16:18.15*** join/#asterisk hellagony (n=egutierr@tvn95-3-82-237-157-188.fbx.proxad.net)
16:18.26Timotino in my country ... Turkey ... only connection fee is something like 1500 USD
16:19.04mutilatoryou could get a 4 port fxo card with the te210
16:19.06Timotimontly connection fee is 300 USD .. I think they are crazy .. I hate monopols
16:19.16mutilatorand then you'de have 2 for incoming calls and 2 for outgoing calls
16:19.18mutilatoror something
16:19.29mutilatordunno how you'de want to handle 50 people on pots lines tho
16:19.34Timotiwell but I need 50 FXS
16:19.43mutilatorfor?
16:20.03TimotiI have 8 ports vegastream FXO ..
16:20.11jake193250:8
16:20.13jake1932hmm
16:20.23mutilator50:8 is almost reasonable
16:20.33mutilatorif you can't afford a pri
16:20.37jake1932i'd go a little higher
16:21.07mutilatorheh we only have 1 pri for like 10 school districts
16:21.10nestArlol. i'm so over capacity.. i have like 30 phones deployed and 2 pris
16:21.10mutilatorand they almost fill it
16:21.11nestAr:)
16:21.34Dr_Rayand have some sort of voip iax termination if you go over capacity
16:21.50Timotiwell most of the calls will go through Voip ...
16:22.05mutilatori'de say do that then
16:22.10jake1932yeah
16:22.11mutilatorget a quad t/e card tho
16:22.24mutilatorto hook all ya chan banks to the box
16:22.52TimotiBut I would love to know more about that TA624  ..
16:22.58Timotistuff ...
16:23.00mutilatorwww.adtran.com
16:23.08mutilatorbasically
16:23.18Timotiand also How much does PRI cost in US ?
16:23.21mutilatorya don't even have to configure it if you don't want
16:23.37mutilatorplug in your T and plug in a phone
16:23.45mutilatorconfigure it for loop start(the adtran default)
16:23.49mutilatorand be on your merry way
16:23.49phpboyexten => _694X,2,Dial(SIP/${EXTEN},20) <--- that will make it ring for 20 secs and then move to the next step, right?
16:24.05mutilatorya phpboy
16:26.30*** join/#asterisk zeedo (n=zeedo@obsidis.org)
16:27.19*** join/#asterisk myiagy (n=myiagy@200.138.215.78)
16:29.06*** join/#asterisk _tekati_ (n=captain@cpe-66-75-215-63.bak.res.rr.com)
16:30.11odie_floconpri in the us = $$$$
16:30.30nestAri pay about $400/mo for each PRI
16:30.57mutilatorcogent was doin $250/mo for a t
16:31.03mutilatorabout 3 weeks ago
16:31.12mutilator'tis too bad it's not in my area
16:31.27jonathhanyone got any idea what
16:31.28jonathh[app_rxfax.so]Oct  4 17:30:03 WARNING[29307]: loader.c:258 ast_load_resource: libspandsp.so.0: cannot open shared object file: No such file or directory
16:31.28jonathhOct  4 17:30:03 WARNING[29307]: loader.c:440 load_modules: Loading module app_rxfax.so failed!
16:31.30nestAra basic pots line for a business is about $65 here.
16:31.35jonathhmeans? i just tried to get the fax stuff working
16:32.00nestAryou don't have span dsp installed (or installed correctly) ??
16:32.02mutilatori'm glad we're a semi telco
16:32.03yaaarso what do you guys think about this openpbx business? does that have legs?
16:32.10mutilatori get that stuff cheap
16:32.13jonathhyes.. i think
16:32.22jonathhi added the entry into ld.so.conf
16:32.31jonathhdo i need to restart anything to get that checked?
16:32.43nestAryou run ldconfig ?
16:32.48nestArmight try that
16:32.50jonathhno...
16:33.05nestArhate fedex
16:33.08nestArhate hate hat
16:33.09nestAre
16:33.12nestArbring me my phone
16:33.13nestArbitches
16:34.36rayvdOk!
16:34.42rayvdI am collectively known as "bitches"
16:36.27iDunnoyou are more than one bitch?
16:37.11Darwin35look all you kunts when I say jump you ask how hi SIR
16:37.12rayvdI am three bitches.
16:37.16rayvdBut I speak for 17 bitches.
16:37.54*** join/#asterisk ChaotY2k (i=juergen@p54A7C347.dip.t-dialin.net)
16:38.07ChaotY2kanybody found how dtmf on sipgate is running? - i hav dtmfmod=info and allow=gsm - but it won't work
16:38.25Timotican someone tell me the PRI cost in US ? per month and installation fee
16:38.25*** join/#asterisk Sedorox (n=Brandon@smartserv/cna/Sedorox)
16:38.30Darwin35why info
16:38.50Darwin35I htought default for dtmfmode and dtmf = is rfc2833
16:39.12Dr_Rayit varies on which city Timoti
16:39.36Timotiroughly ?
16:40.21Dr_Rayif I had to guess between $500 and $1200
16:40.33Dr_Raywe live in seattle so ours is closer to $500
16:40.54jonathhhmmm it dont seen to recieve faxes too well
16:40.56jonathhbooger
16:41.00ChaotY2kit also dont work with rfc2833 ...
16:41.11Dr_Rayit would surprise me if you lived in podunk town it could be $2000
16:41.18Dr_Rayer, would not surprise
16:41.35Timotiso in my country ... installation fee is 1500 USD and monthly 300 USD ... so it is very cheap or ?
16:41.40ChaotY2kthe extension is okay - on anonther line i jump to the same extension and there it is running
16:42.09dos000Timoti, where are you ?
16:42.17Timotiin Turkey
16:42.34dos000that is per ds0 or ds1 ?
16:43.26TimotiHimm good question ... but to wrong person :-))
16:43.36Timotilet me check ..if I can such info ...
16:43.45yaaarTimoti: that's extremely inexpensive
16:43.56dos000Timoti, a ds1 monthly cost is what you mentioned here
16:44.00yaaarTimoti: monthly that is......the installation is quite high
16:44.23Hmmhesaysugh thank god, no more mitel repairs
16:44.33dos000yaaar, the Installation i was quoted was around $CDN 1k
16:44.42Timotiit is only written
16:44.44TimotiPA (Primary Access)
16:44.45yaaarTimoti: here (Missouri) we pay around $220/m for a DS1 local loop, plus about $550 for the PRI on top of it. But the installation is only $450
16:45.11*** join/#asterisk hypnox (n=dan@cornelyn.force9.co.uk)
16:46.08yaaarhowever, muxing a DS3 can save a ton....here it's around $1700 for the 28 DS1 local loops it provides
16:46.12Timotiit is written here ... ISDN PA has 30B/ 1D
16:46.26Timotiwhich one is that ?
16:46.36Sedoroxyaaar, that a month... or install?
16:46.40yaaarSedorox: monthly
16:46.47Sedoroxfor all lines usuable?
16:46.50Sedorox-u
16:46.51yaaarSedorox: i'm not sure of the install on one of those
16:46.53yaaarSedorox: yeah
16:46.55Sedoroxah
16:46.58Sedoroxhats cheap
16:47.00Sedoroxthats*
16:47.14*** join/#asterisk syle2 (n=blag@unaffiliated/syle)
16:47.15yaaarparticularly cheap in light of our ds1 loop rates
16:47.23dos000yaaar, we have 1388 for a ds3 and 285 per pri
16:47.30Sedoroxyea...
16:47.33yaaarnot bad
16:47.39Sedoroxdang...
16:47.47Sedoroxwhats the difference between ds3 and t3?
16:47.52yaaarSedorox: nothing
16:47.54Sedoroxor nothing besides naming?
16:47.57mutilatorfiber = ds3
16:47.59*** join/#asterisk Micc (n=dotirc@c-24-19-175-112.hsd1.wa.comcast.net)
16:48.00SedoroxI figured as much
16:48.07Sedoroxooooooooo... and t3 is coax...
16:48.09Sedoroxgotcha
16:48.09dos000Sedorox, nothing .. just telco guys trying to confuse us
16:48.19*** join/#asterisk huslage__ (n=huslage@c-67-169-200-122.hsd1.or.comcast.net)
16:48.20Sedoroxlol do
16:48.21mutilatoroc1 = same thing too
16:48.21Sedoroxdos000,
16:48.24yaaarmutilator: well, that's the old-school difference.....according to my ILEC's tarrifs, they define them as the same
16:48.26Sedoroxdarn completion :p
16:48.27phpboyhow can I setup custome voicemail msg's?
16:48.32Sedoroxoc1=ds3?
16:48.44yaaarmutilator: i suppose for their purposes though they just want to do that for ease of writing a contract
16:48.51Timotimany many thanks everybody .... have a great day there .. byeee
16:48.56mutilatorprobly
16:48.59*** join/#asterisk chidex (i=richard_@82-45-239-141.cable.ubr01.enfi.blueyonder.co.uk)
16:49.01*** join/#asterisk dca[laptop] (n=dca[lapt@sta-206-168-218-202.rockynet.com)
16:49.07dos000mutilator, noo  .. t3 is another name for ds3 .. except t3 infers wires specifically
16:49.27mutilatorand ds3 infers fiber specifically
16:49.38mutilatorbecause it's just a broken oc line
16:49.42Racemanphpboy, setup voicemail messages? the sound file that has to play when you ring your extension?
16:50.17*** part/#asterisk dca[laptop] (n=dca[lapt@sta-206-168-218-202.rockynet.com)
16:50.17phpboyRaceman: yeah!
16:50.30*** join/#asterisk dca[laptop] (n=dca[lapt@sta-206-168-218-202.rockynet.com)
16:50.30mutilatorthey're all the same anyway
16:50.34mutilatort3 = ds3 = oc1
16:50.36dos000mutilator, you mean oc1=t1 ... in volume ???
16:50.39RacemanIll tell you in a private box
16:50.46phpboyta :D
16:51.33mutilatorlike i said
16:51.35mutilatort3 = ds3 = oc1
16:51.42mutilatoras far as what they do
16:51.56SedoroxOC-1 51.84 Mbps
16:51.57*** part/#asterisk ChaotY2k (i=juergen@p54A7C347.dip.t-dialin.net)
16:51.59SedoroxT3 is 45...
16:52.22hypnoxis it correct to say that a sip client must be registered to * for it to be callable via ts extension?
16:52.35Sedoroxyes
16:52.36mutilatords3 is also 52mbit
16:52.38Sedoroxor static IP...
16:52.58SedoroxI thought the max for t3 was 45mbit...
16:53.15dos000mutilator, is right !
16:53.17SyncrosT3 is 45 mbps
16:53.22dos000hugh
16:53.29mutilatorbbfew
16:53.32mutilatori need coffee
16:53.39*** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net)
16:53.42Sedorox45 != 51
16:53.46SyncrosT1 T3 OC1 OC192 is only a speed indication not a technology
16:53.47Sedoroxso how are they the same?
16:54.05Sedoroxwell if you put it that way.. yea
16:54.13jhivergargl, PRI / GSM gateways are so damn expensive :)
16:54.45dos000"OC-1 service is the equivalent of DS3 or T3. It provides 672 voice channels (or 64K data channels) and runs at 52 Mbps to include the extra bits needed for the optical service."
16:55.11Sedoroxahhh ok
16:55.11mutilatorbak
16:55.14mutilatorya..
16:55.24mutilatorthat lil box they mount on your wall when ya get a ds3
16:55.24jhivermhhh that's a lot of bandwith :)
16:55.31mutilatorit's actually an oc3 mount
16:55.34mutilatorya just use one channel
16:55.43mutilatorfor most cases i've seen
16:55.44*** join/#asterisk T-Squared (n=ted@hidden.serreyn.com)
16:55.46Sedoroxhehe
16:56.08mutilatori'll be happens the it's gbps worldwide
16:56.57Sedorox40gbps
16:56.59Sedoroxyummmm
16:57.12sahafeezOC is from the SONET spec. T1 and such are old school and was jury rigged in.
16:57.13mutilatorbout the only way that'de happen is I2
16:57.46Sedoroxlol
16:58.05mutilatorI2 bandwidth is mehhhh
16:58.14sahafeezi am building a new production box. what version should i use?
16:58.15mutilatorgot to play with it helpin grad students at mich tech
16:58.28Corydon-w1.2 beta, of course
16:58.50sahafeez<=- built an USA backbone. had 142 OC-3s and gig to the desktop
16:59.07SedoroxI would like our school to get on I2..... we're affliated with PSU.. who's on it.. so... dunno
16:59.14Sedoroxlol
16:59.23mutilatorwe were doin some physics crap
16:59.25sahafeezis 1.2 stable for production?
16:59.54Corydon-wWe don't use that word around here
17:00.11Corydon-w"stable" is too ambiguous
17:00.35sahafeezok, would you run your company on a 1.2 setup
17:00.41wunderkini do
17:00.46dudesI use head
17:00.48wunderkintest it
17:00.49Dr_Raylots of people do
17:00.52Dr_RayI use head
17:00.53Corydon-wThank you, Debian, for making the word "stable" completely ambiguous
17:00.58Sedoroxlol
17:01.06sahafeezdabain stable = 2 years behind
17:01.08Sedoroxin debian terms... stable == 5 years old
17:01.11Sedoroxlol
17:01.13sahafeezs/dabain/debian
17:01.15Dr_RayI come into the channel before an upgrade to make sure that head is not broken
17:01.31dudesI'm using head from today
17:01.37Dr_Rayyou can tell that head is fine when people are not screaming in here
17:01.39sahafeezwell i am installing slackware now. going to pull cvs in about an hour so..
17:01.48*** join/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl)
17:02.22pooh_hi all, is there a way to pump up the volume output of asterisk in general?
17:02.25*** join/#asterisk jtodd (n=jtodd@sip3.edge.voip.coloco.com)
17:02.38sahafeezis there some sort of min-faq that say, touch these 5 files and go...voip.org has tons of crap
17:03.04sahafeezi have a pri card and sip phones
17:03.05chidexso are you guys saying head is better thant *@home?
17:03.23sahafeezno, head is better in a car while driving
17:03.35sahafeez@home is bording
17:03.40SedoroxEverything > home
17:03.45Sedorox'cept if your just starting out
17:03.46Sedorox:p
17:04.03sahafeezjust starting out and quite clueless
17:04.17Sedoroxhehe
17:04.20chidexdoes the head come with all the macros?
17:04.28*** join/#asterisk marshall (n=test@S0106000f3dcdd088.wp.shawcable.net)
17:04.55*** join/#asterisk Astinus- (n=abba@213.167.111.138)
17:05.12Astinus-Hello, a little bit offtopic, but are there softphones which can receive faxes?
17:05.21marshallMy outbound call display is not working, the telco is claiming I am sending the record as a display element and they need it as an information element, ideas?
17:05.32RacemanIf you find one, let me know ;)
17:05.41Racemanasterisk can receive faxes, and convert them, and mail them
17:05.43*** join/#asterisk AgentBlueUK (n=agentblu@premierit-66.premierit.com)
17:05.52Dr_RayI would put the extra effort to learn asterisk propper instead of asterisk@home
17:06.00Astinus-Raceman: nice ok.
17:06.23RacemanI've not try it for myself, but read it can. I've you've a moment, i'll find a url for you
17:07.10chidexDr_Ray: so which head should I download?
17:07.45RacemanAstinus- http://www.voip-info.org/wiki-Asterisk+Fax+to+email
17:07.51*** join/#asterisk devonst17 (n=devonst1@dsl092-032-215.lax1.dsl.speakeasy.net)
17:08.46Astinus-Raceman: and the other way? :P
17:09.22Racemanfor receiving, also take a look at this tool
17:09.23Racemanhttp://tafm.sourceforge.net/
17:09.34Racemanfor sending, http://www.voip-info.org/tiki-index.php?page=Asterisk+Email+to+Fax
17:13.44Dr_Raychidex - I think there is only one head
17:17.56*** join/#asterisk miah (n=miah@chia-pet.org)
17:18.05miahanybody used galaxyvoice and have opinions?
17:18.29phpboysay exten 2 is ringing and I'm sitting at extention 1 and I want to answer exten 2... how would I go about doing this?
17:18.43marshallDoes anyone know of a company who provides paid second level support?
17:19.45Dr_Raydigium does
17:19.56*** join/#asterisk dalabera (n=Dalabera@pmr.pmrtechnologies.com)
17:19.58Dr_Rayand lots of people do
17:20.32marshallIm looking on Digiums site and didn't see the link
17:20.36marshallI'll give them a call
17:20.49Dr_Rayin there store they sell prepaid support
17:20.51Dr_RayI think
17:21.42RacemanCommercial Asterisk Support
17:21.42RacemanIn addition to the free installation support for Digium hardware, Digium provides full support for the entire Asterisk software suite, including hourly rates with no commitments. For more information on commercial Asterisk support or any other Digium professional service, please contact express@digium.com or call us toll free at 877-LINUX-ME (877-546-8963).
17:21.53Dr_Rayunder the services tab
17:22.10marshallthanks
17:22.51*** join/#asterisk pifiu (n=myassisb@208.205.181.170)
17:23.25pifiuwow thevoice.digium.com checkout, registering website blows
17:23.33phpboyexten => 6940,1,Dial(SIP/6940,30,rtT)               ; Ring for 20 seconds
17:23.33phpboyexten => 6940,2,Goto(SIP/6942)
17:23.34pifiufreakin password problems login problems constantly
17:23.38phpboydoes that look right?
17:24.14miahwouldn ,30,rT ring for 30 seconds? =)
17:24.27phpboyit would...
17:24.32phpboy:P
17:24.41phpboybut it should work, right? because it isn't :/
17:24.45miahwrong comments aren't very helpful :P
17:25.25phpboytrue true
17:25.25phpboy:P
17:25.27*** part/#asterisk T-Squared (n=ted@hidden.serreyn.com)
17:25.36enderhey, I'm having a problem where * will stop allowing calls through after the first SIP call, message it stops at is Oct  4 10:16:19 DEBUG[2616] chan_sip.c: Checking SIP call limits for device 5713
17:25.38pifiuanyone ever order from allison?
17:25.59phpboyBut it should work, no?
17:27.54phpboy:/
17:29.52*** join/#asterisk toddf (n=toddf@ns0.fries.net)
17:30.03*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
17:31.21pifiuis anyone here?
17:31.28enderyes
17:31.31jarrodno
17:31.35jarrodwe are all bots
17:31.38pifiulol
17:32.33pifiuanyone ever had allison do a recoring for them?
17:32.46jarrodwhy not find your own chick
17:32.59pifiucuz its not so expensive and she does it professionally?
17:34.23jarrodbecause you dont know any girls? :(
17:34.36pifiumaybe so =P
17:34.37pifiulol
17:34.47pifiuno man seriously their site SUCKS
17:34.59pifiui registered, tried to buy the credits and said password incorrect
17:35.03pifiui reset it and nothing
17:35.12pifiucalled and they bs'ed me about saying it works
17:35.16pifiui tried from 3 computers and nothing
17:35.27pifiui hang up and i re-register in another section of the site and try to login
17:35.28pifiuit works
17:35.31AgentBlueUKI personaly hate their site, and i'm sure no matter how professional she is, the custom ones won't sound quite the same as the ones that ship with asterisk
17:35.56shido6where'd reputation go?
17:36.01pifiubut then i logged in to make the recordings and it says i have 0 credits when i just bought 5!
17:36.34NuggetI've purchased prompts from allison via digium's thing.  worked great for me
17:36.39pifiuoh yeah?
17:36.44pifiuwhich site did you use to register yourself at?
17:36.50pifiuit seems easy enough but its not
17:36.52Nuggetthevoice.digium.com I guess.
17:36.58pifiuinteresting
17:37.32pifiubecause thats what i did
17:37.36pifiuand keeps saying wrong password
17:37.39pifiuall this BS
17:37.41pifiuits all bs
17:37.41pifiulol
17:37.50pifiupiece of shit site
17:38.16NuggetI just logged in.
17:38.18Nuggetworked fine.
17:42.29*** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.cust.bezeqint.net)
17:43.02pifiui know but it seems that they sync their passwords from thevoice.digium.com to their yahoo store
17:43.07pifiuits a mess technically
17:43.52watchyi want alison to say sexy things
17:43.58pifiuok so buy a credit
17:44.13tzafrir_homeanybody knows how to contact the author of the perl module Asterisk::AGI? please contact tzafrir.cohen@xorcom.com
17:44.14watchywill she say whatever i want?
17:44.20tzangerpretty much
17:44.25mutilatori'm streaming some streaming audio on a teamspeak server is anyone wants to listen :P ts1.stonedinvaders.com
17:44.26watchysweet
17:44.29tzangerbkw says she won't say the word 'cunt' but pretty much everything else is fair game
17:44.57watchyi wonder why she won't say that
17:45.03Dr_Raywomen are funny about that
17:45.03pifiui dunno
17:45.04Beirdoit's offensive?
17:45.14watchyyea but its $
17:45.19Beirdoso?
17:45.29bendy24"Hi there, press 1 to hear me moan"
17:45.41watchyhaha
17:45.43}cytrak{hehe
17:45.52Beirdo"press 2 to kick bendy in the nuts"
17:45.54enderhave there been problems w/ voicemail?  seems once I call voicemail then I can't call anymore.
17:46.00Beirdohehe
17:46.02watchyhaha
17:46.04bendy24:O
17:46.14bendy24damn sens fans
17:46.27Beirdonot sure how you'd implement that in an AGI, but it could be funny
17:46.53}cytrak{is it true that if I connect an voicemail asterisk box to a legacy PBX using analog lines .. asterisk may now understand the PBX ?
17:47.04}cytrak{may not
17:47.44*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfi1d.dialup.mindspring.com)
17:48.04enderah HAH
17:48.41enderwhat priority is reached if password is incorrect in voicemail?
17:48.44enderI know whats happening.
17:51.26iDunnon+101, I'd guess ;)
17:51.31iDunnothat's the usual way.
17:52.05*** join/#asterisk gonzo- (n=gonzo@195.140.246.50)
17:53.17*** join/#asterisk techass (n=alt@Toronto-HSE-ppp3736759.sympatico.ca)
17:53.50techasscan anyone help me with a sidetone issue on a x101p clone card
17:55.48Damintechass: Perhaps you should call Digium..err.. no.. that isn't an option is it now? :)
17:56.49Dr_Raywell, you could call digium and pay for support
18:00.21pifiuwhat causes there to sound like static when music on hold?
18:00.28pifiuon the other end people hear some static randomly
18:04.08miahsolar flares
18:05.09*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
18:06.40Cresl1n:-)
18:06.41Cresl1nhey!
18:06.57Corydon-wYes?  ;-)
18:07.06enderif I hang up before putting in a bad password, the call is never ended.
18:07.46enderThe Spawn extension never ends.
18:08.32*** part/#asterisk techass (n=alt@Toronto-HSE-ppp3736759.sympatico.ca)
18:10.10*** join/#asterisk RickTick (n=rpulido_@c-67-191-89-108.hsd1.fl.comcast.net)
18:13.39enderhrm, can anybody help me w/ some VoiceMailMain issues?
18:14.31*** join/#asterisk viLeR (i=1000@gw.mutualasterisk.com)
18:15.04*** join/#asterisk iCEBrkr (i=icebrkr@24.129.130.158)
18:15.24*** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com)
18:16.30Jam1edo you have a Hangup entry in your VoiceMailMain exten?
18:17.08enderJam1e: yes, now I have a few of them.
18:17.24Jam1eHmm
18:17.32enderoh strange.
18:17.41enderit may also happen if I just hang up while in voicemail.
18:17.50Jam1eweird
18:17.53RickTickcan anyone instruct me/where I can get info  ...on how to get my HT488 ...to make outbound calls via the fxo port?
18:18.09enderinstead of using # to exit.
18:21.59RacemanI also have a question about voicemail
18:23.31phpboydoes anybody use the PickUpChan command?
18:23.33Racemanafter the beep, it's recording what the caller says (ofcourse, duh), but it writes to 3 files!, why 3? I use one for e-mail (i guess the smallest .wav), and one for voicemailmain (i guess the .gsm file), but what about the big .wav file?
18:25.15hardwireblah
18:25.17hardwiretoday is another day
18:25.40*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
18:25.58hardwirehi harryvv
18:26.03harryvvhello
18:26.15enderRaceman: that is configured in voicemail.conf
18:26.17harryvvnice to be done with this move now that im back on.
18:26.43enderRaceman: basically different methods for different usages.  You can set it to record in only 2 codecs or even just one if you like.
18:26.50nestArgot my wifi phone in
18:26.55nestArso far, seems to work ok
18:26.57*** join/#asterisk rabelais (n=blank@ca-stmnca-cuda4-gen2c1-171.vnnyca.adelphia.net)
18:26.59hardwireharryvv: back on the crack?
18:27.00harryvvWhat are some of the best outdoor wifi antennas that can be installed. Thinking of installing one outside my townhouse.
18:27.02hardwireback on the horse?
18:27.09harryvvhardwire, no back online.
18:27.18Beirdoback on the wagon?
18:27.21harryvvBeen doing lots of physical work moving.
18:27.25hardwireharryvv: go to hyperlinktech.com
18:27.31Racemanokay, thank you, that's the format=wav49|gsm|wav line in voicemail.conf right?
18:28.05enderRaceman: yep
18:28.35gordonjcpharryvv: homebrewed ones
18:28.41enderis there anyway to see at the CLI what priority a call is trying?
18:29.12FuriousGeorgeany way to control what codec is used for concurrent conversations.  i.e. 1 outbound call is ulaw, but a second concurrent outbound call will be gsm
18:29.35harryvvdord, yea I can make yagi and omni antennas and have the formulas but the commercial ones are more durable and also if I start to sell them then no place to mfg them.
18:29.35Dr_Rayleast cost routing
18:29.37Racemanokay thanks. Is there also a way to delete the sound files after mailling for some extensions? (not all extensions)
18:29.46shido6yes
18:29.55hardwireharryvv: pvc glue and copper :)
18:30.01hardwirewhats there to manufacture :)
18:30.04Racemanhow?
18:30.42harryvvhardwire, say a customer wants 50 of them in a week.
18:30.51harryvvfaster just to order them.
18:31.26hardwireor hire canadians to do the labor
18:31.38file[laptop]eh?
18:31.42hardwire:)
18:31.53hardwirecanadians are good for cheap labor.. since they are so hard up for cash all the time.
18:32.04netsurferyooo harryvv u got moved ok?
18:32.12hardwireok
18:32.16hardwireI need to hire a voice.
18:32.31hardwiremy at&t natural voice major breach of agreement for use isn't cutting it
18:32.34hardwireto monotone
18:32.42hardwiretoo even
18:34.24shido6plug? what plug? :)
18:34.34*** join/#asterisk ahigerd (n=root@65.66.90.197)
18:35.11ahigerdHey, what's generally considered to be the "best" snapshot of Asterisk to run? 1.0.9 won't launch for me (illegal instruction in chan_modem.so)
18:35.51Dr_Rayhead
18:36.03netsurferyup.. head is good
18:37.13ahigerdProduction stable?
18:37.25*** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1)
18:37.26shido6ahh, I love head
18:37.53shido6digium gives good cvs
18:38.01netsurferlmao
18:38.33hardwirewhast new in head today?
18:38.36hardwirewhats new
18:38.39hardwireWHATS ASDFK:ASDJF NEW
18:38.58hardwirehow do you get a sum up of cvs up before it ups.
18:39.37*** join/#asterisk thammer (n=thammer@proxy.rtccom.net)
18:40.01shido6call x 2000, and she'll tell you all about it.
18:40.17*** join/#asterisk acidfoo (n=acidfoo@66.11.160.156)
18:40.21enderahigerd: I use 1.2.0beta1
18:41.08hardwireshido6: you trying to afford an expensive car :)
18:41.23shido6naaah
18:41.30shido6funding a visit to the moon.
18:41.33hardwireahigerd: you should disable chan_modem
18:41.38hardwireshido6: heh
18:41.42hardwireI should pimp out my bitch too
18:41.51Beirdo??
18:41.51hardwireI can see the money coming in now
18:42.00shido6$199,999,999,998 to go
18:42.00hardwireBeirdo: nm :)
18:42.01mutilatorO_o
18:42.04Beirdocarrying on the story from yesterday?
18:42.08*** join/#asterisk funxion (n=nunya@mtnuser.icgws.com)
18:42.09hardwireBeirdo: no
18:42.12mutilatori was about to say
18:42.12Beirdowhew
18:42.14hardwirebut things did get better
18:42.16hardwireso there
18:42.20Beirdogoood
18:42.21mutilatorand we take off where we left off!
18:42.33Beirdopimping her out would get you more money though
18:42.37hardwireBeirdo: I just have to go work out with her mondays and fridays.. and she will probably be happier than usual
18:42.39Beirdobut likely a LOT more hell
18:42.50hardwireBeirdo: no.. el shido6 is pimping some mad vocals
18:42.56hardwirewow
18:42.59hardwirethat was lame
18:43.08mutilatori was about to do something
18:43.10mutilatorwhat was it...
18:43.15mutilatorbeen thinkin about it all day
18:43.22hardwirepimping
18:43.22mutilatoroh yea... work... hmm
18:43.24Beirdoahh
18:43.27phpboypickupchan, help PLEASE :<
18:43.29hardwirefordvoice: can I help you?
18:43.38hardwirephpboy: it will destroy you
18:43.52mutilatori was reading bash for the good part of 3 hours this morning
18:43.59hardwiremutilator: read doodie.com
18:44.00mutilatorsomeone pasted a quote and i just couldn't stop
18:44.36filebash.org is like addictive
18:44.49Qwellfile: qdb.us. when you run out :p
18:45.08fileexactly
18:45.09funxionwhats the reason I would get an error with ast_config_load when trying to load cdr_addon_mysql.so module
18:45.15funxionwith latest cvs
18:45.24hardwirebash is great fun
18:45.30hardwirebut you need to be carefull when you use it
18:45.44hardwireHHGTG
18:45.57hardwireI love growisofs
18:48.01marshallIm going bug eyed from all these technical documention but correct me if I'm wrong. There is no point in sending outbound Call Display Names since they are looked up on the callees telco database.
18:48.12Qwellmarshall: true
18:48.17hardwiremarshall: true
18:48.17*** join/#asterisk syle (n=blag@unaffiliated/syle)
18:48.33fileUNLESS your telco passes it internally on their infrastructure
18:48.35marshallthanks guys
18:48.36hardwireQwell: however for numbers not in the DB... thats not true
18:48.44hardwireit just passes it if the telco supports it right?
18:49.07filealways looked up on the end no matter what, unless they do what I said above...
18:49.12*** part/#asterisk ahigerd (n=root@65.66.90.197)
18:49.15enderfile: hey there.... can you spare a minute to help me debug voicemail?
18:49.18marshallasterisk seems to be passing it as a display element, not an information element
18:49.20fileender: what's it doing?
18:49.25marshallso it wouldnt work anyways
18:49.35filemarshall: Cresl1n knows aboot that stuff
18:49.43hardwiretechdata is slowly failing
18:50.20enderfile: when I hang up w/out putting in a password or just hang up inside voicemail w/out pressing #, the call is never killed.  * never gets teh -1 return.
18:50.23hardwireI really hate sites that don't put img width height tags in their img statements
18:50.25hardwirerealyl
18:50.34fileender: hrm lemme try something
18:50.35enderfile: Oct  4 11:41:09 DEBUG[3282] pbx.c: Extension 999, priority 3 returned normally even though call was hung up
18:51.06netsurferhardwire - yeah, though sites that add code to stop u backclicking really piss me off
18:51.15hardwireyeh
18:51.15fileender: I get that but my channel disappears
18:51.25*** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
18:51.31enderfile: my cahnnel dissapears too, but my phone is no longer able to make/get any calls
18:51.38enderfile: until I restart asterisk.
18:51.51fileender: I'd debug the phone side then
18:52.14enderfile: this is the last message in debug:
18:52.15enderOct  4 11:43:30 DEBUG[3372] chan_sip.c: Checking SIP call limits for device 5713
18:52.26filesip debug when thoust trys to place a call
18:52.28endernever gets past that.
18:52.48enderconsole never shows anything
18:53.02enderit's like the voicemail somehow stuffs * and the console.
18:53.27*** join/#asterisk SarahEmm (n=sarahemm@Toronto-HSE-ppp3682154.sympatico.ca)
18:53.41filedoesn't do it for me I'm afraid, how odd
18:53.57SarahEmmwoooooo
18:54.00hypnoxwhat would cause * to segfault with "Spawn extension exited non-zero" when *hanging up* a sip call. Doesnt happen during call establishment.
18:54.05SarahEmmmy phone is here *giggles excitedly*
18:54.14hypnoxum, insert a question mark there somewhere.
18:54.16shido6which phone?
18:54.17enderfile: I have to 'stop' asterisk for a few seconds before I can make calls again.
18:54.23fileSarahEmm: eep
18:54.52enderOct  4 11:46:16 WARNING[3484]: app_voicemail.c:4922 vm_authenticate: Unable to read password
18:54.55ender<PROTECTED>
18:54.59SarahEmmfile: eep?
18:55.03SarahEmmshido6: polycom ip501
18:55.05endersee, that Call accepted, it shouldn't accept a call, it's hung up by then.
18:55.05yaaarso what do you guys think about this openpbx fork? is it going to go anywhere?
18:55.16fileender: why would it be calling something to begin with...
18:55.25fileare you using an IAX2 switch or something?
18:55.34enderfile: nope
18:55.45filewhat does thoust dialplan logic perceive itself to be?
18:55.46enderfile: this is straight sip -> Asterisk
18:56.17enderext '999' is in context 'gamehouse', which is included in context 'default'.
18:56.22ender999 goes to VoiceMailMain
18:57.08fileseeing it in totallity would help
18:58.02enderfile: http://pastebin.com/382957
18:58.42fileyou are using switches... now lemme do this out
18:59.25enderyes, switches are in use for dialing 9+ and _2XXX
18:59.39fileit still does a lookup no matter what
18:59.57enderok.
19:00.16fileit *might* be that, as that's the only thing different between my dialplan and yours at that level
19:00.29enderlet me comment those out for now.
19:00.54fileI need a Beatles fanatic to identify a song for me...
19:01.08mmlj4shoot #not that i'm still a fan
19:01.15MikeJ[Laptop]file, perhaps?
19:01.37enderfile: well I'll be a sonofabitch.   Commenting out switch seems to have done it.
19:01.41fileall I have is it done by a marching band...
19:01.47mmlj4or just paste lyrics into google
19:01.59fileender: I think I know what it's doing... just wondering why it would cause that to happen
19:02.46enderfile: I included one switch and still working, moving on.
19:03.05*** part/#asterisk clint_ (n=clint@snap.helixsystems.com)
19:03.07mmlj4file: google shows nothing that I can find
19:03.22enderfile: second switch working.
19:03.51*** join/#asterisk gaiadog (n=josh@m815f36d0.tmodns.net)
19:03.56mmlj4well, there's a medley of sgt pepper, etc.
19:03.59Qwellokay, Qwell needs a new job
19:04.01Qwellanybody hiring?
19:04.17enderok, WTF?
19:04.18QwellI'm gonna be quitting in about 4 hours here
19:04.20enderfile: now it's working....'
19:04.32harryvvQwell, what kind of work ?
19:04.33fileender: yeah you'll find that just happens sometimes
19:04.34SarahEmmQwell: meep!
19:04.39Qwellharryvv: I do it all
19:04.39gaiadoganyone using a dell sc400 or 800 with 1 or more tdm400 cards?
19:04.47harryvvQwell, where are you at?
19:04.53Qwellharryvv: southern CA
19:05.03phpboyanybody got any PickUpChan or PickUp experience?
19:05.06harryvvgaiadog Arial uses them all the time.
19:05.20enderfile: thats just f'd up.
19:05.24harryvvQwell, well you have any trade experaince like electrical behind you?
19:05.26filephpboy: yes, I have experience with app_directed_pickup in CVS
19:05.46*** join/#asterisk antoniofcano (n=antonio@5.Red-80-32-90.staticIP.rima-tde.net)
19:05.54antoniofcanoHello
19:06.08phpboyfile: could u please help me :<
19:06.09Qwellharryvv: the only "real world" experience I have, is programming
19:06.20filephpboy: I can't help if you don't say what you need help with
19:06.30phpboyOct  4 21:06:04 NOTICE[4215]: chan_sip.c:7496 handle_request: Nothing to pick up
19:06.41phpboythat's the most relivant error
19:06.44filethat's the normal pickup stuff, have you set callgroup and pickupgroup?
19:06.45phpboythis is the *8 exten
19:06.56phpboyexten => *8,1,PickUpChan(1)
19:07.08phpboyin features.conf ?
19:07.13file*8 is the normal pickup stuff built into chan_sip and other channel drivers
19:07.17fileit doesn't go through the dialplan
19:07.37harryvvQwell, well dont know about that. But if you like working with your hands and have some electrical can be a electrician. Severe shortage of these trades here. Qwell msg me for a second.
19:07.41*** join/#asterisk brc_ (n=brc@pdpc/supporter/basic/brc)
19:07.54phpboyfile: should I set that in sip.conf
19:07.55phpboy?
19:08.08filephpboy: you need to have callgroup and pickupgroup set, go look on voip-info.org for information on this
19:08.14gaiadogharryvv...who exactly is Arial? And do you know if they are having interrupt trouble with an SC400?
19:08.17fileit's documented.
19:09.02harryvvgaiadog, dont know about irq problems. what other pci cards do yu have installed.
19:09.29*** join/#asterisk CrazyYoss (n=nobody@69.236.44.222)
19:09.29filephpboy: http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups
19:09.59*** join/#asterisk BuckRogers (n=steve@ool-44c29ac5.dyn.optonline.net)
19:10.04BuckRogershello all
19:10.05antoniofcanoHI I'm getting a little problem with uniqueid generated by asterisk. The VICIDIAL doesn't work because it gets "uniqueid = asterisk-2282-1128443453.0" and should be something like "uniqueid = 1128443453.0". My system is a Debian Sarge and the Asterisk installed is one 1.0.7 from apt sources
19:10.05syleever tried working with your wife hahah
19:10.13gaiadogWell I have the TDM400 with 4 fxo modules and now I am being asked to add an addition card w/ two more fxo modules. I am concerned that there will be interrupt issues.
19:10.14*** join/#asterisk Ganlron (n=Ganlron@omega.csolve.net)
19:10.26antoniofcanoIs that a error config or problem with asterisk version?
19:10.33harryvvsyle, my wife always challenges me..even in a suttle way like her opinion
19:10.34harryvv:)
19:10.42sylehaha
19:10.56syleohh god dude, i can;t tell my wife anything or i get a battle
19:11.23harryvvsyle, has she ever made a comment like " I can stand it when im not in control!"
19:11.52phpboytahnks file... got it working... thanks a MILL :D
19:11.56toddfis there anywhere to get the hw specs (aka if I want to write a driver from scratch, bsd licensed, not gpl, for OpenBSD) for digium hw like the TDM400P
19:11.57*** join/#asterisk Assid (n=assid@203.115.64.57)
19:11.58Assidheya
19:11.59syleno more like "your not listening" as a defence for me trying to tell her something
19:12.09wunderkinyup
19:12.23wunderkinNO! YOU'RE NOT LISTENING!@#@#
19:12.29harryvv:)
19:12.34BuckRogersquick question I recently purchased a IAXy (the new more rounded model) and im tring to connect from my house through nat to my office *server also through nat using dynamic dns service does anyone have any insight on how to get this working?
19:12.38phpboyexten => 6940,1,Dial(SIP/6940,20,rtT)               ; Ring for 20 seconds
19:12.39phpboyexten => 6940,2,Goto(SIP/6942)
19:12.44phpboythat should work right?
19:12.48Dr_Raywomen do not want you to solve problems for them, they want you to listen to how they feel about it
19:12.50Assidgoto ??????
19:12.51Assidno
19:12.59phpboyfuck :<
19:13.00Assidgoto isfor a context/method
19:13.17antoniofcanoI don't know where to look for a solution, after one day debuging tHE VICIDIAL that's the only question to test... but i don't know if we can change the uniqueid prefix from the config files
19:13.18phpboyso what should I use?
19:13.25phpboyjust str8 dial?
19:13.31Assideah
19:13.34BuckRogersalso we have a vonage box hooked to the * fxo port behind nat
19:14.56BuckRogersive set up the IAXy internally and it worked fine and made the appropate changes (or so i thought) but it does not make a connection
19:15.02phpboyAssid: yeah?
19:16.01Assidyeah
19:18.22Kattyscooby do.
19:18.36iDunnoscooby *doo*.
19:18.40iDunnosurely.
19:18.44enderfile: I got it to happen again!
19:18.45Katty...
19:18.58KattyiDunno: you obviously don't speak kat.
19:19.17iDunnoKatty: but it is Scooby Doo! not scooby do! ;)
19:19.29Assidhrmm
19:19.31Kattylies.
19:19.33Assidi gotta figure this one out..
19:19.34iDunnoKatty: and yes, I have problems with anything that's not fairly broken english ;)
19:19.37Assidit doesnt let me traceroute
19:19.38KattyiDunno: you forgot your filter.
19:19.39Assidbut i can ping
19:19.46Assidstupid router
19:21.05*** join/#asterisk Spacebar (n=stingray@stingr.net)
19:22.58RickTickcan anyone instruct me/where I can get info  ...on how to get my HT488 ...to make outbound calls via the fxo port?
19:23.10*** join/#asterisk chidex (i=richard_@82-45-239-141.cable.ubr01.enfi.blueyonder.co.uk)
19:24.41*** join/#asterisk funxion (n=nunya@mtnuser.icgws.com)
19:25.46fordvoice?
19:26.05fordvoicewho wants to make some beer money and give me a hand with ASTCC
19:28.48fordvoicehello
19:28.57fordvoiceNo takers
19:29.21fordvoicemsg me if anyone is interested....
19:31.49*** join/#asterisk bartpbx (n=bartpbx@ip-80-226-232-211.vodafone-net.de)
19:32.43bartpbxhello
19:33.46bartpbxI'm searching for an ack call solution. I've seen one somewhere in the wiki but i cant find it
19:33.57acidfoobpx:1467:ast_log(LOG_DEBUG, "Function result is '%s'\n", cp4 ? cp4 : "(null)");
19:34.25acidfoo*printf and cie doesnt already hangle (null) ?
19:34.48bartpbxsomeone calls an extension. the call gets forwared to my mobile any i need to accept the call by pressing the # key or something
19:35.00bartpbxanyone has a working example for this?
19:35.04acidfoos/hangle/handle
19:35.11filecp4 is a pointer to the function result... if it exists, it prints it out, otherwise it prints out (null)
19:35.39fileif you tried to do %s with a null pointer, I dunno what it would do - never tried... might crash... might just have no value
19:35.46acidfooeuuu
19:35.53acidfooit will print (null) by default
19:36.03acidfoochar *s = NULL; printf("val: %s\n", s);
19:36.06acidfoowill print (null)
19:36.46filerealllllllly
19:36.50acidfooyup
19:37.40fileast_log  might be different
19:38.29acidfooit's base on vfprintf()
19:38.38acidfoovoir - void ast_log(... ) _
19:39.16filesubmit a bug note if you want to do optimizing stuff
19:39.20acidfooMaybe if Mark Spence would be here, he could answer ;)
19:39.44fileI believe Kevin wrote the function stuff
19:40.06jarrodwhois the author "SC" in the chan_mgcp driver that is on the development team?
19:40.39acidfooill post , maybe someone will come with an unbelivable answer hehe, would be nice, im very intrigued
19:42.24*** join/#asterisk razu_ (n=razu@ip58.cab60.mus.starman.ee)
19:42.27razu_hi
19:42.46enderih
19:43.54razu_is there any wildcard for .conf files to include a whole dir of .sip files ?
19:43.59fileyeah sorry overlord
19:44.17Corydon-wMight also be a cross-platform issue.
19:44.36Corydon-wfor example, does passing NULL also work on FreeBSD, Solaris, and Mac OS X?
19:45.08acidfooC standard
19:45.16anthmhappily printing NULL would be daft
19:45.38Corydon-wwhich C standard?
19:45.39enderoper systems love to do daft things...
19:45.42Corydon-wThere are multiple.
19:46.15JerJeri wish people would stop randomly wire transfering us thousands of dollars
19:46.26QwellJerJer: damn them!
19:46.33JerJeri've been sitting on a grand from someone for over a month and i do not know who sent it
19:46.36anthmit was never acceptable to print NULL till a bunch of ppl with buggy code convinced the compiler to let you get away with it it's an ongoing battle.
19:46.36enderJerJer: accounting nightmare.
19:46.42*** part/#asterisk thammer (n=thammer@proxy.rtccom.net)
19:46.47opus__jerjer, that was me
19:46.52opus__I need it back :(
19:46.57QwellJerJer: He meant to send it to me...the names are so close, ya know?
19:47.01Corydon-wIf it's ANSI C, then fine... However, if it's C99, there might be a valid objection.
19:47.13acidfooCorydon-w: let me 1min
19:47.26JerJeropus__:  my bank tried to reverse the transaction and they couldn't
19:47.30JerJerthat's the crazy thing
19:47.32*** join/#asterisk arguile (i=user224@66.38.201.234)
19:47.41JerJerand it was from a London bank
19:48.00anthmdilligence to avoid printing a null pointer can not possibly be a bad thing
19:48.05opus__send an receipt to the queen!
19:50.01*** join/#asterisk michael1234 (i=michael1@pavas-a507.racsa.co.cr)
19:50.36michael1234I am trying to setup asterisk@home to work with a quad e1 card can anyone helpme. I just dont knwo the config for the e1 cards
19:50.56*** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)
19:51.04generalhanwhats going on everyone ?
19:52.07*** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn)
19:52.29_Thoranyone knows to compile asterisk add-ons?
19:52.55*** join/#asterisk spackle (n=spackle@209.234.83.19)
19:52.57Qwell_Thor: should just be `make && make install`
19:53.01Dr_Raymake add-ons?
19:53.03bartpbxhello. anyone has an idear how to solve this dialplan issue? I konw there is a way. But I can't find it in the wiki...
19:53.08Qwellin the addons dir
19:53.36generalhanQwell: you familar with Cisco 7960s ?
19:53.44*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) [NETSPLIT VICTIM]
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19:53.45Qwellgeneralhan: yeah, but I've gotta head out...sorry ;/
19:53.46_Thoryes, it is a bigger problem than that
19:53.56generalhanQwell: its cool ill catch up with you next time !
19:54.28_Thorhere is the problem Qwell, I donÂīt know how in the world, but after I compile, the system behaves as if it still reading the old code
19:54.49*** join/#asterisk alexhopper (n=a27386@CPE000103d29ae2-CM001225dfdfe0.cpe.net.cable.rogers.com)
19:55.26_ThorI even erase the so file in  /etc/lib/asterisk/modules, but the old executable does not desappear
19:55.49_ThorI donÂīt know what the heck is going on
19:56.02_Thor....regarding that behavior :)
19:56.31*** part/#asterisk spackle (n=spackle@209.234.83.19)
19:57.49_Thorhello
20:01.37Kattyhmm.
20:02.04*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) [NETSPLIT VICTIM]
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20:03.22harryvvSo what states or hosting companies have the lowest pri hosting rates?
20:03.56nestArhosting rates?
20:04.02harryvvyes
20:04.13harryvvin other words
20:04.16nestAryou talking about how much they charge to cross connect from your equipment to the demarc?
20:04.23harryvvdont have the pri and equipment on your own site.
20:04.43mthemi have a PRI in yellow, with tons of HDLC errors... any ideas besides irq confiicts
20:04.43nestArcolocation with pri's?
20:05.04Dr_RayI called several colo's in Seattle and none of them could/would do a PRI to them
20:05.23nestArwe do colocation, if a customer wants a PRI, they contract that from the phone company of their choice.. we only charge them a one time fee for wiring from the demarc to their equipment.
20:05.35harryvvnestAr at the rate PRIs are charged here in bc, its better to just have the equipment located in a site that charges alot less in the states.
20:06.03nestArpri's here run about $350-400/mo
20:06.05nestArUSD
20:06.26acidfooCorydon-w: well,  after some research, I've come to conclusion that it's glibC  and freeBSD-libc standard ton output (null)
20:06.55acidfooneither c99 and c98 specify something about null pointer in *printf function
20:07.14Corydon-wwhich means it's not a standard we can depend upon
20:07.49acidfooI think both method are good
20:08.08acidfoobecause it's widely used in C library to output (null)
20:08.15Corydon-wI think it's a fine candidate for the next revision to the C standard
20:08.16acidfooso, I dont see importance to change code
20:08.23Beirdoonce should never depend on a library to do null checks for you when it's not officially documented that it must be done
20:08.56Corydon-wHowever, since it is not now standardized, we should leave the code the way it is
20:09.31acidfooyup
20:09.39acidfooanyway, it was a good discussion about standard ;)
20:09.44nestArMy initial review of the UTStarCom F1000 is that it makes and takes calls
20:09.50hardwiremoof
20:10.09mutilatorfoom
20:12.48SarahEmmnestAr: woo! ;)
20:12.58SarahEmmmy initial review of the Polycom IP501 is that it does not yet make or take calls. ;)
20:13.37hardwirehar
20:13.38Beirdomy initial review of an IAXy on a crappy remote cable modem connection is:  SMASHY SMASHY!
20:13.53hardwirehere here
20:13.54SarahEmmheh
20:14.02hardwireI have our house set to using one
20:14.04nestArlol
20:14.09nestArI'm done with Polycom
20:14.10hardwireI am going to have to host an asterisk server at my house
20:14.15Beirdoheh
20:14.17nestArI don't think I'll be buying any more of them
20:14.21harryvvI like my polycom ip500
20:14.33hardwirewhats a 501?
20:14.44hardwirespecial?
20:14.47harryvv501 is with ehanced security and memory
20:14.50harryvvthats it.
20:14.51Beirdowell, I put an IAXy in PR, but her cable modem connection is craptacular, so I get mucho dropouts, glitches, etc
20:14.52*** join/#asterisk wolfson (n=hehe@usr-kdh-208-6-58-26.beachlink.com)
20:15.08hardwireBeirdo: yeh
20:15.10hardwireI had to enable adpcm
20:15.21BeirdoI think I'll enable an SPA3000
20:15.22bendy24Beirdo: like you guys say much with all the moaning and whatnot
20:15.22hardwirewhich works fine until a porn site comes along
20:15.32hardwireBeirdo: I have a few of those so I will go that way
20:15.35Beirdobendy24: you little wanker
20:15.40Beirdo:)
20:15.43hardwirebut we have a linux router.. so I might as well make use of that
20:15.48Mauro__Can I connect an asterisk box to a siemens PBX?
20:15.56Beirdoyeah, I have one SPA3000 that I could bring down there
20:16.01mutilatorhow much power does a mini itx board take?
20:16.02hardwireMauro__: can you connect a T1 to it?
20:16.15Beirdoand suffer the indignity of SIP over the internet (through NAT!)
20:16.16Mauro__I think that yes
20:16.17Mauro__:P
20:16.20hardwirethen probably
20:16.43hardwireMauro__: the real question is what do you want to do when thats all said and done?
20:17.11Mauro__I want to manage some numbers from asterisk and other with the siemens
20:18.51*** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com)
20:19.52hardwireMauro__: sounds like a pain in the ass
20:20.06Mauro__yeah :P
20:20.13hardwireand not really all that reliable :)
20:20.40Mauro__maybe I should just get a new conection for my testing asterisk :P
20:21.02hardwirewhile your at it get me one
20:21.50opus__hi hardwire
20:22.03hardwirehi opus__
20:22.04hardwirewhats new
20:22.36hardwireanybody here have experience w/ EVDO under linux?
20:22.37opus__Not much
20:24.27*** join/#asterisk darkskiez (n=darkskie@host86-133-148-56.range86-133.btcentralplus.com)
20:27.43_Thoranybody here knows compiling add-on packages?
20:29.20hypnoxis it normal for things to end with 'spawn extension exited non-zero' ?
20:31.04acidfooCorydon-w: hey dude
20:31.06acidfoochar *ast_callerid_merge(char *buf, int bufsiz, const char *name, const char *num, const char *unknown)
20:31.07acidfoo{
20:31.07acidfoo<PROTECTED>
20:31.07acidfoo<PROTECTED>
20:31.19acidfoodo you understand the purpose of that ?
20:31.44Corydon-wWhat's wrong with that?
20:31.51cpatryif unknown is NULL it sets it to <unknown> ?
20:32.22Corydon-wso you take a NULL pointer and make it into a string that indicates unknown...
20:33.27_ThorI keep compiling successfully the add-on, but the old code keeps coming up
20:33.56*** join/#asterisk ahigerd (n=root@65.66.90.197)
20:33.58Beirdowell, you declared unknown as "const" and you are changing it.  bad
20:34.02acidfooCorydon-w: erm I just wonder where the "<unknown>" will be stored if the value point to null
20:34.29Corydon-wYou're not a C guy, are you?
20:34.36BeirdoI am
20:34.46Corydon-wIt doesn't point to NULL, the pointer itself IS null
20:34.51*** join/#asterisk brookshire (n=matt@gateway.digium.com)
20:34.51Beirdo:)
20:34.59ahigerdThree different versions of Asterisk -- 1.0.9, a CVS snapshot from February that I took off of our production server, and 1.2.0b1 -- and all three are dying with SIGILL.
20:35.06Beirdoit's illing
20:35.09acidfooCorydon-w: maybe the way he did that is puzzling me
20:35.18acidfooI never had to do that ;)
20:35.24*** join/#asterisk spackle (n=spackle@209.234.83.19)
20:35.29ahigerdBut I'm running basically the same hardware/software setup on the machine right beside me and it ran fine the first time.
20:35.30Corydon-wacidfoo: so after that, the pointer is no longer NULL.  It now points to a static string
20:35.42ahigerdIdeas, anyone?
20:38.06anthmconst char *unknown implies the pointer always points at read-only data "unknown" certianly is read-only
20:38.25acidfooCorydon-w: http://rafb.net/paste/results/pHOg0m49.html
20:38.28acidfoothat's what I mean
20:38.42cpatrycause its const, so its true, it cant be changed.
20:39.10*** join/#asterisk CrazyYoss (n=nobody@adsl-69-236-44-222.dsl.pltn13.pacbell.net)
20:39.29*** join/#asterisk _sleepy_ (n=sleepy@p549D970F.dip0.t-ipconnect.de)
20:39.55*** join/#asterisk drumkilla_laptop (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
20:39.55*** mode/#asterisk [+o drumkilla_laptop] by ChanServ
20:40.11anthmconst pointers can change what they point at including to null all they want but you cannot change *what* it's pointing at it
20:40.31Corydon-wacidfoo: that's because you're passing the wrong thing
20:40.43Corydon-wacidfoo: you're passing the VALUE null...
20:40.57anthmyour test func cannot modify s unless you prototype char **s and pass &s
20:41.01*** join/#asterisk casio_ (n=Cris@200-126-118-191.bk8-dsl.surnet.cl)
20:41.12Corydon-wacidfoo: you'd need to pass a POINTER to the POINTER in order to modify it in the original function.
20:41.14anthmthen you need to deal with *s not s in the func
20:41.14acidfooim my previous past, the value isnt **
20:41.20acidfoo<PROTECTED>
20:41.20acidfoo{
20:41.20acidfoo<PROTECTED>
20:41.20acidfoo<PROTECTED>
20:41.44Corydon-wRight, because it's being handled all INSIDE that function
20:42.04Corydon-wIt's not passing the "unknown" string back out
20:42.17*** part/#asterisk spackle (n=spackle@209.234.83.19)
20:42.24Corydon-wnot in the unknown variable, anyway
20:42.39acidfooso the scope of that is the bloc { } ?
20:42.45ahigerd... You DO know that doesn't work, right?
20:42.46ahigerdYou have to use strcpy()
20:42.58Darwin35why
20:43.05Corydon-wNote that buf is getting copied into, not getting reset to a different pointer
20:43.43tzangerok wtf
20:43.44tzangerOct  4 16:43:44 NOTICE[10365]: app_dial.c:1109 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3 - No route to destination)
20:43.59tzangerI can ping switch-1.nufone.net just fine from the box, but can't route a call there?
20:44.04acidfooCorydon-w: now I fully understand the purpose
20:44.06acidfoo;)
20:44.10tzangerer rather "no route to destination" ?
20:44.29anthmthe func promises to return a pointer to a char if it is pointing at a constant you dont need to copy anything
20:44.51*** join/#asterisk kuj (n=kuj@c-67-174-106-30.hsd1.co.comcast.net)
20:44.55anthmlike if the func was int func() and you return 12
20:45.04anthmyou are safe cos 12 is a constant
20:45.17anthm"hello" is the char equiv
20:46.35Primerthis is so strange. I can't get festival work work consistently. It'll gladly say "Mary had a little lamb", but it won't say "Who is buried in Grant's tomb"
20:47.13enderPrimer: the apostrophe
20:47.29enderPrimer: try it as "Who is buried in grants tomb"
20:47.49mutilatorulyses?
20:49.11wunderkinthats classified
20:49.38Mauro__pfff small world Primer :P
20:49.46Primerender: naw, that's not the real sentence I was using
20:49.52Primerit does not have an apostrophe
20:49.56PrimerMauro__: ÂŋQue passa?
20:50.46RickTickanyone using a HandyTone Ht488 ata fxs/fxo?
20:51.10drumkilla_laptopwunderkin: can you comment on the chanspy crash bug, please?
20:51.48jarrodwhy is MGCP generating DTMF TONES in the middle of my CONVERSATIONS when the REMOTE PARTIES are not pressing DIGITS
20:52.35michael1234please help me i need to setup /etc/zapata.conf but cant remember the settings required
20:53.08*** join/#asterisk afrosheen (n=test@txprotoa2.august.net)
20:54.42*** join/#asterisk funxion (n=nunya@mtnuser.icgws.com)
20:55.02funxionhello
20:56.22funxionanyone have an idea why I would not be able to make asterisk-addons its cvs v1-0-9?
20:56.24funxion<<<<<<< Makefile <is the line in questgion
20:56.26funxionMakefile:62: *** missing separator. Stop. <is what Im getting
20:57.40Beirdoif it has <<<<<<<<<<
20:57.46Beirdothat's a CVS conflict
20:57.56*** join/#asterisk tclineks (n=tclineks@ppp-70-243-238-201.dsl.tpkaks.swbell.net)
20:58.18Primerbah, now all asterisk can do is block on festival
20:58.29funxionBeirdo but both asterisk and asterisk-addons are both v1-0-9
20:58.40funxionBeirdo how could I go about fixing that
20:58.57Beirdoedit the Makefile, look for the <<<<< and >>>>>
20:59.03Beirdoresolve the conflict
21:00.15funxionso the conflict would be between the <<<<<<< and >>>>>>>
21:00.41Beirdoyes.  <<<<<<<  then ======= then >>>>>> or something like that
21:00.43Beirdobeen a while
21:01.23InfraRedfunxion: what os
21:01.30funxiondebian
21:01.32*** join/#asterisk santiago (n=santiago@63.245.87.62)
21:01.46InfraRedno idea then
21:01.53funxionBeirdo what is the ======= for
21:01.57*** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk)
21:02.16MRH2hi everyone - does hangup() take options?
21:02.24Beirdoit's the separator between the two sections, one is the original, one is the new contents that conflict with the original
21:02.36SarahEmmhttp://voip-info.org/tiki-index.php?page=Asterisk+cmd+Hangup
21:02.40SarahEmm^-- hangup command
21:02.54funxionBeirdo once I remove one of the lines do I need to remove >>> ===== <<<<< ?
21:02.57MRH2yep
21:03.02cpatryMRH2: just type show application Hangup
21:03.17Primerseems that festival forks a process to handle new connections, and that process just zombies right away
21:03.18Beirdoyes
21:03.23funxionBeirdo thnx it werk
21:03.24Beirdojust leave behind the line
21:03.29Beirdono prob
21:03.43funxionI appeciate the help man
21:03.49MRH2can I hangup(Zap/g1) for example
21:04.04cpatryMRH2: no, its for the current chan
21:04.20hardwirethe disconnect tone on our phones is now known as the "beep beep beep of doom"
21:04.38MRH2anyway to hangup up 1 or all zap channels from the dialplan - not the current channel
21:04.38hardwireugh
21:04.53*** join/#asterisk soleil713705 (n=blalba@62.220.135.115)
21:04.53cpatrygoing home.
21:05.25MRH2scenio: all zap lines in use, someone wants to dial an mergency call
21:05.30MRH2scenerio
21:05.40MRH2emergency
21:05.45MRH2(sp!)
21:06.34iCEBrkrAnyone else having problems with Voicepulse connect?
21:06.39MRH2was thinking to clear all channels prior to dial would be the way to do it
21:06.47soleil713705hello. I'm using asterisk to listen to radios via phone. It was working well with version 1.0.1 (debian woody backport) but now it doesn't work anymore with version 1.0.7.
21:06.48Beirdobetter watch it, Katty, some guys might start eying you :)
21:06.55soleil713705has anybody any clue ?
21:07.10iCEBrkrsoleil713705: We're all clueless :)
21:07.13Beirdohehe
21:07.16soleil713705Iget this message in the logs "app_mp3.c:91 timed_read: Poll timed out/errored out with 0"
21:07.24soleil713705:-)
21:08.00hardwirehmm
21:08.25hardwirethis receptionist things attended transfers requires you to hang up.. instead of ever hitting transfer
21:08.33hardwirelots of lost calls.. blame me..
21:08.59_ThorHi everyone, if I am compiling successfully, but it still runs the old code, what do you think it can be?
21:09.16iCEBrkryou didn't 'make install'
21:09.20watchyyoou didnt install it
21:09.30hardwireeverybody
21:09.32hardwire1
21:09.32hardwire2
21:09.33hardwire3
21:09.35hardwireYou didn't install it
21:09.39iCEBrkrKatty, SarahEmm Get a room :P
21:09.41watchyYou didn't install it
21:09.45iCEBrkryou didn't 'make install'
21:09.50hardwireiCEBrkr: don't you wish..
21:09.54iCEBrkrhehehe
21:09.54KattyiCEBrkr: pfft.
21:09.56iCEBrkrNot really.
21:10.08hardwireiCEBrkr: just because they are women.. and they mew at eachother doesn't mean their hot lesbians with videocameras
21:10.12hardwiredoes it?
21:10.17Kattyno
21:10.22hardwiredidn't think so :)
21:10.23watchywe can wish thoughh right
21:10.34hardwirewatchy likes to watchy
21:10.35iCEBrkrhardwire: don't you wish!
21:10.42hardwireiCEBrkr: on this dull day
21:10.45iCEBrkrhaha
21:10.47soleil713705there are 2 mpg123 process when doind http streaming but only one when playing local files
21:10.51hardwirea little lesbian porno could do me a world of good
21:11.09iCEBrkrsoleil713705: Sounds familiar.. Like the old MoH stuff.
21:11.14hardwiresoleil713705: do you have two contexts for moh?
21:11.18iCEBrkrhardwire: lesbian p0rn is overrated.
21:11.20_ThorHi everyone, if I am compiling successfully, but it still runs the old code, what do you think it can be?
21:11.28hardwireiCEBrkr: its just what the doctor ordered
21:11.31watchyhot teen lesbian porno
21:11.33iCEBrkr_Thor: 'MAKE INSTALL' for the 10th time.
21:11.37*** join/#asterisk diclophis (n=diclophi@adsl-69-238-124-226.dsl.pltn13.pacbell.net)
21:11.37hardwirewatch it there watchy
21:11.47diclophishello, anyone using realtime configuration?
21:11.52hardwire_Thor: did you make install yet?
21:11.54diclophisalso, anyone a AGI guru?
21:11.56watchyi'm not talking kids
21:11.59hardwire_Thor: did you make install yet?
21:12.13iCEBrkr_Thor: did you make install yet?
21:12.16hardwire_Thor: did you make install yet?
21:12.16KattyHmmhesays: can i get a little yum yum kitty kitty
21:12.22iCEBrkr_Thor: did you make install yet?
21:12.32hardwirealright.. I don't feel like being a brat anymore
21:12.38KattySarahEmm: s'ok.
21:12.41soleil713705icebrkr, hardwire: only one context I guess, my local files are in mohmp3
21:12.42hardwirethe both of you are teases.
21:12.47_Thoryes, the problem is weird
21:12.50hardwiresoleil713705: weird.
21:12.54watchyare theey atleast hot hardwire?
21:13.01hardwirewatchy: how the hell should I know
21:13.03Kattyhardwire: and your problem with this is.......>?
21:13.06hardwireKatty's nick is all straight
21:13.07wunderkinlesbian porno? guess you got mine, katty and sarahs attention
21:13.08tclinekssoleil713705: is your musiconhold.conf pointing to valid files?
21:13.09hardwireand
21:13.11_Thorthank you, here is the problem: it all complies well
21:13.14hardwireSarahEmm's is a little curvy
21:13.18_Thorit all compiles well
21:13.19hardwireso I am gonna go with yay
21:13.20iCEBrkrsoleil713705: I think that's just the nature of the beast.. I've always had 2 mpg123 processes
21:13.24Kattyhardwire: straight?
21:13.27watchyhaha
21:13.28hardwireK
21:13.28hardwireA
21:13.29hardwireT
21:13.33Kattyoh
21:13.34hardwirepick a curve
21:13.35Kattythat
21:13.35_Thorbut I donÂīt know how the heck it keeps running the old code
21:13.36*** join/#asterisk LostFrog (n=reallyno@dsl093-100-093.wdc2.dsl.speakeasy.net)
21:13.39hardwirenow S
21:13.41hardwirethats where its at
21:13.51iCEBrkr_Thor: are you sure you compiled a newer version?
21:13.59_ThorI made sure I am erasing the .so file in the modules folder
21:14.00iCEBrkr_Thor: Did you stop and restart asterisk?
21:14.05_ThorI am sure
21:14.08iCEBrkr'make clean'
21:14.09_Thorno, i didn
21:14.12Kattywhat a bunch of weirdos.
21:14.15_ThorI haven
21:14.20_ThorhavenÂīt
21:14.23hardwireKatty: SarahEmm sorry for being a bastard :) you can go lick eachother and what not now w/o harassment from me.
21:14.25soleil713705ice: yes, I guess but I don't really understand why it is stuck now...sometime it works
21:14.29afrosheenlol
21:14.30Kattyhardwire: kthxbi
21:14.30iCEBrkr_Thor: Well, you're gonna have to stop the old asterisk and start the new one
21:14.30Corydon-wYeah, Sarah's hot
21:14.33blitzrage_Thor: and make sure you erase the headers in /usr/include/asterisk/ as well
21:14.35BeirdoKatty: you didn't expect an oddball or too?
21:14.43KattyBeirdo: of /course/ not
21:14.44soleil713705ice: in extensions.conf I have got exten => 005,1,Macro(radio,http://ruisseau.ctrlaltdel.ch:9000/stream/2.mp2)
21:14.57KattyCorydon-w: not just hot, but hotttt
21:15.03_Thorbut I canÂīt stop *, it is in production
21:15.04Corydon-wKatty, too
21:15.12iCEBrkrsoleil713705: Personally, I would't worry about it, unless you have like 10 mpg123 processes :P
21:15.21blitzragelol
21:15.23Kattykit-kats are not vegan.
21:15.27hardwireKatty: you have some issues
21:15.28iCEBrkr_Thor: Umm.. Then how do you expect to run the new executable???
21:15.32Kattyhardwire: obviously.
21:15.44watchyyea deff lesbian. the vegan part gave it away
21:15.49wunderkinkatty can only eat things from the earth - dirt water and vegetables
21:15.53Corydon-wMmmm, tofu dogs...
21:15.56Kattywunderkin: nodnod.
21:16.00diclophis/quit
21:16.02_ThorI thought it would pick it up automatically on reload
21:16.08diclophisarg
21:16.09Kattywunderkin: and fritos and oreos and things.
21:16.10soleil713705ice: I looked in the source code in apps_mp3.c and it seems that the implementation of timeout is new from version 1.1 to 1.07
21:16.12iCEBrkr_Thor: Um.. no.
21:16.16*** part/#asterisk diclophis (n=diclophi@adsl-69-238-124-226.dsl.pltn13.pacbell.net)
21:16.23soleil713705ice: sorry 1.0.1 to 1.0.7
21:16.24wunderkinkatty: umm oreos.. cream... ? milk? oh no
21:16.31iCEBrkr_Thor: The server is running in memory.. You have to restart the process.
21:16.31fordvoiceDamn what did I miss
21:16.39iCEBrkr_Thor: There's nothing weird about that.
21:16.41Kattywunderkin: go check the ingredient list silly.
21:16.46Kattywunderkin: they're pure sugar and vegetable shortening
21:16.46wunderkinok :P
21:16.46_ThorMy friend, that probably is the answer to my question
21:16.57syleif an mp3 is already converted to 8000hz , if your telling mpg123 or madplay to process the file in 8000hz with command line arguement to that program, does it recognize its actually 8000hz and not try to transcode it?
21:17.04wunderkinsince im sure like a nice vegan you did
21:17.06Corydon-wHmmm, app_oreos ?
21:17.08iCEBrkr_Thor: Most definitely.
21:17.15wunderkinat least i got a pat
21:17.17KattyCorydon-w: app_mew
21:17.21_Thorblitztrage:  why shoul I also erase the headers?
21:17.34soleil713705ice: anyway it was fun listening to radio through phone...I had about 50 radios available
21:17.40Corydon-wBack in the 50s, Oreo creme was partially composed of lard
21:17.44iCEBrkrsoleil713705: neat.
21:17.49LostFrogMmm.. lard.
21:18.04_ThorOK icebrkr, thank you very much
21:18.22hardwireKatty: issues with most people.. or am I a target?
21:18.28LostFrogI am having huge problems with my Snom 360. the audio on every call sounds muffled.
21:18.34*** join/#asterisk huslage (n=huslage@c-67-169-200-122.hsd1.or.comcast.net)
21:18.40SarahEmmi'm semihere btw
21:18.50Kattyhardwire: mew?
21:18.51soleil713705ice: what is strange is that it is unstable, sometime it works and I cannot explain why but is is very rare
21:19.01hardwireKatty: I am so happy looking.. see.. http://www.beringsea.com/staff/shane/headshot.jpg
21:19.03soleil713705ice: I should have stayed to debian woody :-)
21:19.05hardwireyou can't hate a man that looks so happy
21:19.16hardwiresoleil713705: did you oops and get gcc 4.0?
21:19.23Kattyhardwire: k
21:19.35hardwireyou can't beat me with candy..
21:19.39hardwirewhen I look so cgosh darn cute
21:19.41soleil713705ice: no
21:19.48soleil713705hardwire: no
21:19.53Kattyhardwire: watch me >:)
21:19.58SarahEmm:)
21:20.05hardwirehttp://www.beringsea.com/communities/Saint_Paul/goingwireless2002/gfx/shanecheckingsignal.jpg
21:20.13hardwireKatty: me and my dew rag are gonna kick your arse then
21:20.20Kattyhardwire: lies.
21:20.25LostFrogHmm.. seems to be g726..
21:20.26Kattyhardwire: you should never kick a lady.
21:20.33hardwireI kick my woman all the time
21:20.38hardwirebut she asks for it
21:20.50Kattyi see.
21:20.51BeirdoKatty: and ladies should never kick us, especially not in the nads :)
21:20.53hardwiremainly when she says things like.. "Give me a high kick"
21:20.57fordvoiceDamn I am gettigng an education here
21:21.06hardwireBeirdo: indeed..
21:21.11hardwireyou can get an infection that way
21:21.13*** join/#asterisk zoo (i=nobody@ip-98-16.travedsl.de)
21:21.14Kattyheh. and people wonder why i have androphobia...
21:21.15zoohello
21:21.24Corydon-wGotta love the S&M that comes out in this channel...
21:21.26hardwireKatty: you are afraid of that Kevin Sorbo series?
21:21.28Cresl1nhey!
21:21.30fordvoiceHell ya
21:21.35fordvoiceI missed some of it
21:21.40BeirdoKatty: I'm sorry.
21:21.45fordvoicehave to look back at the logs
21:21.47BeirdoI should just shut up :)
21:21.50KattyBeirdo: (=
21:21.55soleil713705hardwire, icebrkr: normally mpg123 shouldn't fork in 2 as if we look in the mp3play() definition there's a dup2() so that the descriptor are passed to the child
21:22.05hardwiresoleil713705: I dunno why it does that
21:22.08hardwireits weird eh
21:22.17hardwireKatty: maybe my g/f has this
21:22.21hardwirewhats penis phobia?
21:22.21Corydon-wKatty: besides, most guys aren't into S&M
21:22.36KattyCorydon-w: S&M doesn't bother me.
21:22.40hardwireCorydon-w: I cry like a little bitch.. probably why most women like it
21:23.12Corydon-whardwire: if you cry on me, I'll beat your ass... :-P
21:23.21hardwireis that a sexual invite..?
21:23.22Kattylet's move on.
21:23.24hardwirenot really on topic here
21:23.27KattyNEXT
21:23.33hardwirelets keep it clean Corydon-w
21:23.37hardwireseriously.
21:23.39Corydon-wUh huh
21:23.41hardwireMeow
21:24.04Corydon-wThat depends.  Do you like being submissive to another man?
21:24.06hardwirethe Meow scene in Super Troopers seems to show up whenever I am idly thinking of nothing
21:24.31hardwireCorydon-w: this was not in my day planner under "Things to talk about today"
21:24.49Corydon-wIt never is.
21:25.27fordvoiceneed need a little help with ASterisk
21:25.37hardwirefordvoice: why were you pinging me eariler?
21:25.38fordvoicethink somebody can give me a an ear
21:25.44fordvoiceI was
21:25.47fordvoice>>
21:25.47hardwirewhy are you pinging me now?
21:25.55fordvoiceIm not
21:25.56hardwire<fordvoice> think somebody can give me a an ear
21:25.56hardwire-fordvoice- Ping Coming Provided y
21:25.56hardwire-fordvoice- [Fast][(1)][Slow] [1secs] [From-lem.freenode.net] [Pg-#579]
21:25.57hardwire-fordvoice- - (Estimated Ping Reply Time 1.3940 Seconds...) - [No Lagg Detected]
21:25.57hardwire<fordvoice> I was
21:26.04hardwire--- Received a CTCP PING 1128472051 from fordvoice
21:26.05hardwire--- Received a CTCP PING 1128472052 from fordvoice
21:26.22hardwireyeh.. now you are flooding me with them
21:26.23fordvoicedamn irc
21:26.23*** join/#asterisk FuriousGeorge (n=bri@pool-70-111-115-131.nwrk.east.verizon.net)
21:26.27hardwiredamn irc?
21:26.27FuriousGeorgehey all
21:26.33KattyFuriousGeorge: FuriousGeorge FuriousGeorge
21:26.38hardwiredamn really crappy IRC client
21:26.41fordvoicemirc program
21:26.43FuriousGeorgehey Katty
21:26.43fordvoicesorry about that
21:26.45fordvoiceinteresting
21:26.48hardwireCorydon-w: thats enough now
21:26.51hardwireno more pingy
21:26.51fordvoiceneed to d/l a diferent one
21:27.01fordvoicesorry about that hardwire
21:27.01Corydon-wOh, come on...
21:27.03hardwirefordvoice: I am setting you to ignore now.
21:27.12SarahEmmwooo there is a config option to make the phone not #$@^ing reset the handset volume after every call
21:27.13Kattylet's grow up boys.
21:27.19fordvoicewhat does that do
21:27.21fordvoice??
21:27.21Kattywe're not in 1st grade.
21:27.24FuriousGeorgecan anyone recommend a codec the sounds better than gsm and doesnt take as much bandwidth ulaw
21:27.27Corydon-w[16:27:07]  Sending a CTCP PING -9.9E511 to hardwire
21:27.28Corydon-w[16:27:08]  CTCP PING reply from hardwire: inf seconds
21:27.33KattyFuriousGeorge: lbc maybe?
21:27.38hardwireCorydon-w: yeh.. I hear you loud and clear
21:27.46FuriousGeorgeKatty:  ill give it a whirl
21:27.50KattyFuriousGeorge: k
21:27.50hardwireFuriousGeorge: with CVS I use speex
21:27.58hardwirew/ vbr and all sorts of other fun options
21:28.03hardwirebut iLBC is fun too
21:28.04FuriousGeorgehardwire:  these are in production so i shy away from cvs
21:28.14Corydon-wgotta love those inf second replies
21:28.17hardwireFuriousGeorge: I had more problems with 1.0.9 and 1.2.0 than I do with CVS
21:28.19afrosheenFuriousGeorge: g729, end of story
21:28.37FuriousGeorgeill look into ilbc vs g729
21:28.39PrimerIs there a variable that's set with the IP address of a sip caller?
21:28.44FuriousGeorgehardwire:  interesting
21:28.48Corydon-wStrange, how did you get a copy of 1.2.0 ?
21:28.49hardwireFuriousGeorge: g729 is great stuff.
21:28.54hardwireCorndawg_: www.asterisk.org
21:28.56hardwireerr
21:28.59hardwireCorydon-w: ^^
21:29.13Corydon-wThat's not 1.2.0, that's 1.2.0b1
21:29.47hardwireI can argue that 1.2.0 is not 1.2.0-final-release-happy-mookshit too
21:31.20*** part/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl)
21:31.23FuriousGeorgeg729 looks like itll be worse quality than gsm
21:31.25arguileI have a few issues with g729s sound, it has a tinny (not quite echo) to it. Like you're in a large very bright (accoustically) room
21:31.32hardwireFuriousGeorge: you should test it out
21:31.40hardwirego buy some $300 headsets
21:31.55FuriousGeorgehardwire:  yeah but we use zap channels too
21:32.15hardwireFuriousGeorge: whats that got to do with g729 ?
21:32.20opus__arguile, check QoS
21:32.29hardwireFuriousGeorge: zap == ulaw.. thats it..
21:32.53FuriousGeorgehardwire:  300 dollar headsets wont do anything for the quality
21:32.58*** join/#asterisk rking (n=rking@ip68-105-231-56.lu.dl.cox.net)
21:33.06hardwireFuriousGeorge: it was just an audiophile joke
21:33.07FuriousGeorgei gotcha
21:33.18afrosheenFuriousGeorge: g729 is tons better than gsm quality wise
21:33.56FuriousGeorgeafrosheen:  im sold, ill give it a whirl
21:34.04FuriousGeorgehardwire:  i get it now
21:34.05hardwireFuriousGeorge: you wanted something less cpu intensive as GSM?
21:34.10hardwirethats not g729
21:34.13hardwirethats more like.. adpcm
21:34.17hardwireits smaller than ulaw
21:34.17LostFrogHow much are g729 licenses going for?
21:34.20hardwirefaster than gsm
21:34.24hardwire32kbps
21:34.34FuriousGeorgeg729 is proprietary huh
21:34.42arguileopus__: That was my first thought, doesn't appear to be the issue however
21:34.45afrosheenyeah licenses are like $5 per channel I think
21:34.46hardwireand very cpu intensive
21:34.49SarahEmm.ckear
21:34.51SarahEmmoops
21:34.52afrosheenyeah it eats cpu..so what.. :)
21:35.07LostFrogHow does gsm compare to g729 bandwidth-wise?
21:35.09hardwireafrosheen: he didn't want that
21:35.17afrosheenok
21:35.20hardwireLostFrog: gsm is 12kbps g729 is 9.6kbps
21:35.23rkingso i'm in a company of 20 people, we have weekly all-hands conferences where people sometimes have to dial in via PSTN, and daily all-dev (~8 people) conferences, and all-day 1-to-1 or 3 person calls.  right now we're using skype, and i've cooked up an asterisk proof-of-concept using iax running on my localhost.  i'd like to go forward with doing it myself, but others think it would be better for me to not spend time on it and hire a consul
21:35.23rkingtant.
21:35.27rkingwhat do you all think?
21:35.32afrosheenyou gotta decide on your trade offs
21:35.33hardwireLostFrog: but the nominal ethernet bandwidth is almost the same
21:35.41hardwireuse IAX trunking w/ g729 and you will see amazing results
21:35.45afrosheeng729 trunks _really_ well though, saves bandwidth that way
21:35.48FuriousGeorgeis it possible to test g729 w/o a licence.  is it an honors system deal?
21:36.10hardwireFuriousGeorge: sure.. call somebody using g729 :)
21:36.17*** join/#asterisk snitt (i=snitt@snitt.info)
21:36.19LostFrogFuriousGeorge: there are "free" g729 libraries. :)
21:36.28LostFrogThey aren't legal for commercial use.
21:36.31hardwireLostFrog: for transcoding off-line streams?
21:36.33hardwireooh
21:36.35hardwire?
21:36.39snitthi
21:36.42*** join/#asterisk Attila_Kovacs (n=kovacsat@spool3-31.gatesgroup.hu)
21:36.47hardwiresnitt: hi
21:36.51FuriousGeorgeLostFrog:  isnt that "free" licence the pass-thru stuff im reading about.  i.e. no voicemail
21:36.52snittAttila_Kovacs: szia
21:36.57LostFrogno.
21:37.05LostFrogpass-thru doesn't require a codec to be installed.
21:37.18FuriousGeorgeLostFrog:  do i need to do anything ebsides allowing it for my clients?
21:37.53snittif i register a g729 codec, the registration program identifies only the real physical nic's, or virtual ethernet interfaces too, like bonding, bridge, tuntap, and so on?
21:37.54arguile"Free" g729 references: http://www.voip-info.org/tiki-index.php?page=ITU+G.729
21:38.10rkingafrosheen: if we do go with a consultant deal, it seems to me like we shouldn't have to have /that/ many hours put into it for such a straightforward setup (i.e., no physical wiring except maybe a few PSTN-in lines, etc.)  who would you recommend we contact to get a square deal?
21:38.36*** join/#asterisk AJ-Mpls (i=DJAJay@63.231.252.9)
21:39.06AJ-MplsHey guys when i have Astman running I get  "Ignoring unknow event 'PeerStatus' come up every few mins.. Whats making that happen?
21:39.52afrosheenrking: what are you wanting to do exactly, implement a 100% asterisk system?
21:39.59rkingafrosheen: yep, 100%
21:40.08afrosheenrking: and that includes sip phones and whatnot?
21:40.44LostFroghttp://kvin.lv/pub/Linux/Asterisk/
21:40.46rkingafrosheen: i was thinking the main deal being iax, with sip for the vonage users
21:40.52LostFrog^--- g279 codec
21:41.20afrosheendoes vonage sip integrate with Asterisk? I haven't even looked into that
21:41.48afrosheenrking: anyway..good consultants are usually very regional, I'd pick a guy off voip-info.org that's local to you
21:41.56FuriousGeorgeafrosheen:  i think i heard someone jerry rigged it somehow
21:42.04rkingafrosheen: ok
21:42.16*** join/#asterisk LoRez (i=lorez@freenode/staff/lorez)
21:42.29rkingafrosheen: we're not sure re using SIP to bring Vonage to Asterisk, it was just a guess/hope.
21:42.47*** join/#asterisk casio_ (n=Cris@200-126-75-100.bk5-dsl.surnet.cl)
21:43.03SarahEmmuhh
21:43.09SarahEmmvonage doesn't let your bring your own device last i saw
21:43.26Beirdovonage sucketh
21:43.38rkingworst case is we'd let vonage pay the long-distance and use PSTN for that.
21:43.38brookshireyeah.. they want to lock you into their device
21:44.14afrosheenthere are alot better providers than vonage, trust me
21:44.47afrosheenour iax trunk provider just got bought out by commpartners, we hope they stay good
21:44.47rkingwe've already got a bunch of people using vonage, i could suggest they switch, but it would have to be a real slam-dunk or else they'll whine
21:45.11afrosheenanyone hear anything about commpartners quality or lack thereof?
21:45.11LostFroglusers whine.. get used to it. :)
21:45.20*** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net)
21:45.38SplasPood'lo all...  Is there a channel variable that contains the IP address of either end of the connection?
21:46.05afrosheenno
21:46.35rkingLostFrog: well, the deal is, it's not like i'm responsible for providing a telecom solution, i'm just a developer who happens to think that * is the right solution, so i'm footing the bill with my own time until we all get rid of Skype.
21:46.55rkingLostFrog: so each step along the way i have to think of how disruptive my suggestions will be.
21:46.56LostFrogI was joking, rking... well.. mostly so.
21:47.07afrosheenrking: the only thing I like about vonage is their cheap qos-enabled routers, they make a big difference for remote extensions and home users
21:47.20AJ-Mplswhen i have Astman running I get  "Ignoring unknow event 'PeerStatus' come up every few mins.. Whats making that happen?
21:47.41LostFrogI wish asterisk supported SIPS..
21:48.11LostFrogWell.. I think I wish..
21:50.49*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
21:50.53afrosheensecurity on sip will be an issue eventualy
21:51.04afrosheenif it takes a bounty to get it done, so be it
21:51.30LostFrogSIP over TCP has to be done first.
21:52.57wunderkinafrosheen: my provider uses them too.. and was using txlink..  i havent really used them very much lately so i dont know
21:53.41afrosheenwunderkin: yeah we're on txlink here, love them so far
21:53.43hardwireIEUEUEEE
21:53.56hardwiremust remember not to put cup of soupe in front of desktop cd burner set to eject on complete
21:54.13afrosheenhardwire: you must have a really tiny computer
21:54.27hardwiredesktop.. cd burner..
21:54.29hardwireIEE1394
21:54.34hardwire<PROTECTED>
21:54.50arguileHehe, I did that with coffee and my old external SCSI one
21:54.51afrosheenoh..external
21:55.05FuriousGeorgeare lower bandwidth codecs generally worse for echo surpression or vice versa?
21:55.05LostFroghardwire: that is your cup holder.. how dare you use to burn cds..
21:55.18fordvoicewhat is the cmd in the CLI that will show how many IAX calls are being placed at the current time or how may active calls there are at the current time
21:55.26hardwirecup-o-soup holder
21:55.27AJ-MplsWhy am i getting  "Ignoring unknow event 'PeerStatus' come up every few mins.. When i run astman?
21:55.31michael1234anyone able to helpme me configure a new box
21:55.32enderfile: ping
21:55.36afrosheenAJ-Mpls: just ignore it
21:55.55afrosheenmichael1234: cardboard or plastic
21:55.58enderanthm: ping
21:56.17file[laptop]eh
21:56.51*** join/#asterisk wundaboy (n=uo@67.189.123.103)
21:57.02*** join/#asterisk SimonR (n=SimonR@CPE001310092352-CM001371142e78.cpe.net.cable.rogers.com)
21:57.11wundaboywhat is the default username/password for the web control panel on a polycom ip500?
21:57.16enderfile: so my problem w/ sometimes bridging failing, I seem to have solved it.  During one of my other debug periods, I _had_ put in some wink timings and these seemed to have caused the problem.
21:57.21enderfile[laptop]: ^^
21:57.25file[laptop]silly!
21:57.40afrosheenwundaboy: Polycom 456
21:57.49wundaboyafrosheen: thanks
21:57.51afrosheencaps matter
21:57.53afrosheennp
21:57.57enderfile[laptop]: I know, silly me.
21:58.23enderfile[laptop]: I"ve also killed off one of the switches, that will be going away soon anyway, and no more dead lines either.
21:58.54hypnoxis there a free sip client for linux that doesn't suck? :|
21:59.05enderhypnox: kphone isn't your thing?
21:59.07*** join/#asterisk nagl (n=nagl@213.235.241.6)
21:59.11blitzragehypnox: X-Lite
21:59.32hypnoxender it sends out warbly dtmf that asterisk doesn't understand
21:59.42endernice
21:59.54hypnoxi'll try x-lite
21:59.59wundaboycan i have my polycom connected to more than one pbx?
22:00.10wundaboywill that mess anything up?
22:00.23*** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com)
22:00.53Attila_KovacsHi All! Anybody faced the incomming call - no voice problem with misdn_chan?
22:01.17af_I am wondering: in a bri telco line, I have a Dial statement at first priority. the callee spend money with telco even if the noone answer to the dial statement?
22:01.18endermmmmm poppyseed
22:02.12*** join/#asterisk Pr0ph37 (n=pr0ph3t@m615e36d0.tmodns.net)
22:02.13generalhananyone here familar with the cisco 7960s ?? i need some help with the Date/Time display on the phone
22:02.20*** part/#asterisk Pr0ph37 (n=pr0ph3t@m615e36d0.tmodns.net)
22:02.35mizticanyone have a fractional T1, say 12 voice lines+the rest data? I don't think i'd like my phone system be internet accessible as it would be doing firewall duty too, how do you handle that ?
22:03.45SarahEmmanyone here have experience with polycom configs?
22:04.02syleyou make no sense miztic
22:05.08*** part/#asterisk tclineks (n=tclineks@ppp-70-243-238-201.dsl.tpkaks.swbell.net)
22:05.12wunderkinhe doesnt want people to hax0r his box
22:05.35miztici was thinking of using the wildcard T100P - single T1 card
22:05.44mizticso asterisk would get its phone lines from that
22:05.54mizticbut my internet comes in on that same line
22:05.55*** join/#asterisk Pete_Largo (n=PeteLarg@225-196.35-65.tampabay.res.rr.com)
22:06.28Pete_Largohey y'all
22:06.42Hogiemiztic: wanna know what I did in the same situation?
22:07.09mizticHogie, thats what i'm asking, lets hear how you did it
22:07.34*** join/#asterisk ManxPower (i=eric@1Cust3522.an7.dfw28.da.uu.net)
22:07.37HogieI had a 4port card, and just passed my data channels through to our adtran router (which was doing our channel bank stuff before we did asterisk)
22:07.37zoohello, i have got two sip phones, both use nat=yes. one is a sipura and one is a xlite. the xlite cannot hear the sipura talking. the RTP port is forwarded in the routers firewall, but no success. do i need nat=route?
22:07.44ManxPowerIf there are any Cisco DDR gurus out there, please /msg me
22:07.54HogieDance Dance Revolution?
22:08.02SarahEmmDial on Demand Routing, hogie
22:08.03ManxPowerHogie, Dial on Demand Routing
22:08.07SarahEmmbeat you ManxPower ;)
22:08.08Hogieah, ok
22:08.18ManxPowerI hate cisco dialout.
22:08.21SarahEmmstupid acronym is used for like 5 things
22:08.24HogieSorry Manx, never even dealt with that
22:08.34zooand the echo test works. that is interesting
22:09.00HogieManx: you shouldn't run an isp on an isdn connection
22:09.11mizticHogie, so i'd need to forward my data channels to another T1 card in another Pc
22:09.25HogieI didn't do it that way, and I dont know if it will work
22:09.34HogieI have a 4 port card
22:09.56mizticmaybe i can have a smaller asterisk install just forward the voice lines to the main asterisk from the firewall
22:10.13miztici'll have to read up
22:10.26mizticjust found out about the possible frac T1/voice
22:10.34HogieI just did a:   dacs=7-24:31
22:10.54Hogiein my zaptel.conf, to forward channels 7-24 on span1 to channels 7-24 on span2
22:11.06Hogiethen made my own wire:)
22:11.11miztichehe
22:11.31enderthat works
22:11.35Hogiehttp://gallery.cyberjunky.net/Work_Pictures/P0009072
22:11.40Hogiethere's when it was wired in
22:12.11Hogiehttp://gallery.cyberjunky.net/Work_Pictures/P0009075   <-- that's our adtran
22:12.19Hogieender: its better now
22:12.30mizticnice work
22:12.40Pete_LargoHogie, you should do something about the rats
22:12.44Hogiehah
22:12.51Hogieits fine now, it was just a mess then
22:13.10enderWe get to re-do our data closet soon
22:13.11Hogienow we have a 12 channel PRI, and a full T1 just for data, and no more wiring hacks
22:13.25mizticthat helps, appreciate it
22:13.33enderdid a remodle of our office that included all new data runs (replacing existing voice runs w/ data too), so ours gets to be CLEAN soon.
22:13.45HogieI can't wait until we move
22:14.05Hogieright now we have 7960's on everyone's desks, with their computers behind them
22:14.19endereeeeew
22:14.25Hogie(we use poe to power most of the 7960's)...  well, I'll be able to break them out to their own data drops
22:14.28Hogieat our new place
22:14.30enderWe installeda completely segregated network for our phones.
22:14.36Hogiewe will too
22:14.43enderit's the only way to go (:
22:14.47Hogiewe have been here 8 years.  They never forsaw it in this building
22:15.01endergot a nice pair of overpowered managed SMC 10/100 switches for the network.
22:15.13enderyeah, this building was built in the 30's I think.
22:15.37Hogiethey remodeled this building when they moved in, and pulled enough cable for network, but didn't think to pull double what they wanted
22:15.43FuriousGeorgebbl, all
22:16.06syleyou call anything under gigabit switches nice, /me gives ender more weed
22:16.07HogieWhen we move, Im having them pull 4 cat5 drops per workstation/phone area
22:16.23endersyle: oh no, we use GigE for our server network and to feed the office switches.
22:16.40endersyle: however, wasting GigE for IP phones is a bit more overkill than I can justify
22:16.47endersyle: especially managed ones.
22:17.01syleyeah true
22:17.07sylejust not a waste for database servers
22:17.13enderhell no
22:17.39enderwe're actually going to trunk gigE lines together at the switch for 2G links to our DB server.
22:17.39*** join/#asterisk Nyvar (i=Ot@216.14.14.48)
22:19.15ManxPower~docs
22:19.17jbotdocs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
22:19.17ManxPower~mailinglist
22:19.18jbotmethinks mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
22:19.50endersyle: ever mess w/ the high end SMC managed switches?
22:20.10syleunfortunately no
22:20.26sylebest i;ve played with is the cisco switches
22:20.39sylei like their SMP support on them
22:20.57syleerr
22:20.59syleSNMP
22:21.01SarahEmmSMP support on switches?
22:21.04SarahEmmoh lol. *nods*
22:21.04enderwell, some (most) will argue that Cisco is better than the SMCs, but the price is about 50% lower on SMC.  And you can SSH to the damn thing for configuration.
22:21.20Hogieisn't smc what dell retags?
22:21.24endersyle: yeah, the SMCs have SNMP
22:21.24Hogieer, relabels?
22:21.27Nyvarlol.. cisco v1 ssh..
22:21.30enderHogie: possibly...
22:21.38enderNyvar: CIsco uses V1?!
22:21.42enderno wonder nobody turns it on.
22:21.43HogieI have a dell 24port sitting on my desk
22:21.48Nyvaryep
22:21.51*** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com)
22:22.26Nyvarwe're about to buy one of those smc l3 24 port things
22:23.31enderNyvar: We've got 2x TigerSwitch 10/100 48ports and 2x 10/100/1000 24port ones.
22:23.33sylei agree completely
22:23.37sylecisco is overpriced
22:24.28Nyvarender, are the 24 port ones the new layer 3 "managed" switches?
22:24.45enderNyvar: um, let me find the model number.
22:25.03fordvoicecan somebody help me set up h323 on my asterisk I am completely lost
22:25.28enderSMC6750L2
22:26.13Nyvarahh, im talking about this guy: SMC8748ML3
22:26.16enderNyvar: by the 'L2' at the end, probably not the L3.
22:26.22Nyvaryep
22:26.43enderNyvar: it does support some features at Layer 3 though.
22:27.06enderCoS support, IP / Port priority, etc...
22:28.25Nyvarnice, well at $2100 for a 24 port 10/100/1000 copper, 4x gbic, and 10Gig-e expansion module, with l3 and ospf, looks amazing for the money
22:28.48enderyep
22:28.54enderI've been very satisfied w/ the SMCs thus far
22:29.34enderhighly configurable, great feature set, sturdy, and no flakyness thus far.
22:30.02Nyvargood to hear, I'm looking forward to playing with it, beats the hell out of the $300 cisco 5500's we are using now :)
22:30.36enderand it comes w/ a 600 page book on how to manage the thing.
22:30.54enderours even has a redundant power supply port.
22:32.54enderhrm, damn.
22:33.12enderI just realized that the SMC ssh is v1 as well.  Looks like i need to setup some access rules.
22:34.10Nyvarzoiks
22:35.04hardwiremy trash can needs to be much larger
22:35.33Nyvarbigger target after drinking?
22:37.13*** join/#asterisk Johnsie (n=john@acs-24-154-53-217.zoominternet.net)
22:38.38enderhrm, our Gig smc uses SSH v2
22:38.43enderseems the 10/100's only use V1
22:38.57hardwirewe have one too
22:39.00*** join/#asterisk SarahEmm (n=sarahemm@Toronto-HSE-ppp3682154.sympatico.ca)
22:39.03Nyvarthe datasheet for the one we are getting just says SSH
22:39.04hardwirebut our PoE SMC uses v2
22:39.25enderbut if people can get through our outside network, through our Asterisk Box, and finally to the 10/100 switches, I think I Have bigger problems then people messing w/ my switches.
22:39.26SarahEmmcurrent ciscos do that too (ssh2) :)
22:39.40hardwireender: heh
22:40.02enderNyvar: I'm imagining that if they are a newer model (and th emodel # is higher than our gigE switch) that they use v2
22:40.04hardwirehow long until ssh2 is comprimised
22:40.05hardwire?
22:40.05hardwireheh
22:40.06Nyvarcisco put ssh2 into ios?  thought they still only had v1 support
22:40.27SarahEmmi can ssh2 into switches, i know.
22:42.27Nyvaryep, looks like cisco is putting it in, it has been in the testing trains, the 7200 router got v2 in 12.3
22:45.24*** join/#asterisk fordvoice (n=channelv@rrcs-70-61-133-91.central.biz.rr.com)
22:46.31Attila_KovacsHi All! misdn_chan help needed!
22:48.52SarahEmmask the question :)
22:52.22SarahEmmanyone know what a T in a Polycom dialplan is?
22:52.30SarahEmmlike [2-9]xxxT or 0T
22:52.41JerJerself destruct
22:52.47doughecka_timeout
22:52.59SarahEmmtimeout? meaning?
22:53.06JunK-YSarahEmm: theres a RFC for that.
22:53.09doughecka_sit in the corner
22:53.10SarahEmmoooh
22:53.16SarahEmmthanks JunK-Y, iddn't know
22:53.21doughecka_and pout
22:53.22SarahEmmi figured each mfg had their own thing
22:53.25Nyvarterminate?
22:53.37doughecka_toosh?
22:54.36Nyvarit _might_ mean terminate listening for digits and dial once this pattern has been matched
22:54.44*** part/#asterisk Attila_Kovacs (n=kovacsat@spool3-31.gatesgroup.hu)
22:54.51*** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net)
22:55.11Nyvarsipura S0 at the end of it's dialplan entries
22:55.14SarahEmmahh
22:55.21SarahEmmhrm
22:55.22Nyvar^ sipura uses
22:55.56MRH2digit timeout?
22:56.31SarahEmmokay, random guesses aren't so much helping ;)
22:56.40SarahEmmuhh
22:56.42SarahEmmfordvoice?
22:56.44SarahEmmcould you please not do that?
22:56.56Nyvaryep, manuals are for definative answers :)
22:57.17kujSarahEmm, check out RFC3435, at http://www.faqs.org/rfcs/rfc3435.html
22:57.41SarahEmmaha!
22:57.44Nyvari just configured a polycom soundpoint 4000 sip speakerphone the other day for asterisk, don't recall reading the dialplan stuff though
22:57.52SarahEmmthat's the RFC i was looking for, didn't expect it to be in part of the MGCP one tho :)
22:58.06enderI read through the dialplan stuff as I modified our polycom phones.
22:58.07SarahEmmNyvar: ahh.. this is a polycom ip501 :) prolly the same think i got it now :)
22:58.11enderer the dialplan.
22:58.16kujyep, funny that
22:58.30enderSarahEmm: yeah, I have 501s and 301s, with my own custom dialplan.
22:58.34SarahEmmpolycom configs sure are.. thorough.
22:58.47Sedoroxgot your poly SarahEmm ?
22:58.55enderdigitmap="911|9,011xxx.T|9,[2-9]xxxxxxxxx|9,1[2-9]xxxxxxxxx|7,xx|666|999|998|411|8xx|9,[1-9]11|[2-6]xxx"
22:58.57SarahEmmhmm. T 'matches a timer expiry'
22:59.12SarahEmmmakes sense that 0 would have it then
22:59.17SarahEmmbecause you could be typing 011
22:59.21SarahEmmgah
22:59.21enderSarahEmm: yeah, basically instead of grabbing the digits as soon as a map is met, it waits a second before grabbing them.
22:59.29SarahEmmfordvoice: quit it please.
22:59.37SarahEmmmakes sense ender :)
22:59.43SarahEmmSedorox: yep!
22:59.48Sedoroxhow you like it?
22:59.51SarahEmmSedorox: sooooooo nice! (coming from a budgetone ;) )
22:59.57MRH2do i win?
23:00.03SarahEmmhehe )
23:00.18Sedoroxlol.. I have a Bt100 right now :p
23:00.43wunderkincan i get some snuggling?
23:00.53SarahEmmmrrew?
23:00.57enderI tested the gstream, and the sipura.  Polycom was MILES ahead in quality and sound quality
23:01.04SarahEmmyep :)
23:01.55*** join/#asterisk bjohnson (n=bjohnson@i216-58-58-237.cybersurf.com)
23:02.36*** join/#asterisk mcn (n=mcn@ext-gw.newtoncomputing.co.uk)
23:02.46wunderkinyes to me sarah?
23:03.05SarahEmmwunderkin: erm, what?
23:03.13wunderkinhaha
23:03.28*** join/#asterisk fordvoice (n=chrisf0r@rrcs-70-61-133-91.central.biz.rr.com)
23:03.54SarahEmm*confused*
23:04.04enderSarahEmm: he's looking for some snuggling.
23:04.07RickTickhas anyone gotten a grandstream HT488 working on the fxo port?
23:04.10ender* ender snuggles his IP501
23:04.12wunderkinshes slow
23:04.12ender<wunderkin> can i get some snuggling?
23:04.14ender<SarahEmm> yep :)
23:04.17wunderkinhaha
23:04.19fordvoiceSorry about the Flood
23:04.27fordvoiceit was teh IRC program I was using
23:04.50wunderkinwow..
23:05.08SarahEmmmew?
23:05.32fordvoicechanged my IRC software to the plain one off the web My applogies Guys.....
23:12.09*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
23:21.32hardwireanybody here use K6OPT for GSM optimization?
23:21.41Nyvari do
23:21.55hardwireNyvar: how do you feel about it
23:22.41Nyvarnever really thought about it, if you're using a pentium pro or higher x86 you should probably enable it
23:22.48hardwireok
23:22.56hardwireusing a p4 HT
23:22.58Nyvarif powerpc or some other proc then don't :)
23:23.02hardwireyeh
23:23.05hardwireweird
23:23.13Kattypaging twisted[asteria] to the front desk.
23:23.16hardwireand do you spec PROC=i686
23:23.29hardwireor does the build figure it out
23:23.35Nyvarthe asterisk audio files tend to be in gsm, so it might get a little bit of use
23:23.50*** join/#asterisk MrMAGO (n=mglucksm@pdpc/supporter/sustaining/MrMAGO)
23:23.58Nyvari never specifiy it, but doesn't hurt to if that's your hardware
23:24.37SarahEmmgah. anyone know if it's possible to make polycoms save volume settings through a warm boot?
23:24.43SarahEmmi suppose i could just stop rebooting it, but..:)
23:25.00SarahEmmmrrew?
23:25.03hardwirehttp://bugs.digium.com/view.php?id=5338 .. I don't think anybody cares
23:25.06MrMAGOhi, anyone knows if the search engine has been disabled in the wiki? I only have google as search option...
23:25.12SarahEmmit's disabled, yeah
23:25.14SarahEmmjust google right now
23:25.26MrMAGO=S
23:25.31SarahEmmuhh.. that bug was just put in a few days ago
23:25.48MrMAGOall righty, thx
23:25.51SarahEmmmeep!
23:27.24SarahEmmmeep!
23:27.25SarahEmmokie :)
23:28.03*** join/#asterisk Ash (i=aaron@outofband.org)
23:28.27Kattyaaron.
23:29.45AshHELLO KATTY
23:30.05enderhello capslock.
23:30.05KattyAsh: let's not use caps, kthx.
23:30.13*** join/#asterisk heath__ (n=root@12-215-34-84.client.mchsi.com)
23:30.13AshWHAT'S CAPS
23:30.21AshI AM ON A VAX SYSTEM AND DON'T HAVE LOWERCASE SORRY
23:30.30Nyvarlol
23:30.31Ashhee
23:30.32Ash;-)
23:30.45Kattysilly rabbit. caps is for kids.
23:30.57rayvdsoft caps or hard caps?
23:31.06Nyvarunlimited
23:31.06Ashkneecaps
23:31.29fiber0ptidoes anyone know how to handle a separate voip DID number within asterisk?
23:32.12Nyvarfiber, please elaborate

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