IRC log for #asterisk on 20080815

00:01.39jeevif dtmfmode isn't set, what is the default method ?
00:02.32chkngumboam i right in thinking that with asterisk i can have 4 phones talking to 4 different people through a single phone line?
00:03.04jeevthat, i dont know how many channels are available..
00:03.10*** join/#asterisk Linker3000L (n=chatzill@78.32.25.201)
00:16.09*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-62e8e7a482a96901)
00:18.34jeev[Aug 14 17:17:36] DTMF[22791] channel.c: DTMF begin ignored '4' on SIP
00:18.35jeevhmm
00:30.02jeevok, so apparently i can't get incopming DTMF if i'm set to inband, not sure if bug or provider but provider will check it out.. i will try teliax later, so i added SIPDtmfMode(rfc2833) into the call for accepting and booya! oh, i disabled rfc2833compensate too :)
00:30.04jeevwooooooooo
00:30.10chkngumbocan anyone tell me if asterisk can accept calls on a single phone line, and carry multiple conversations over that single phone line at the same time?
00:33.33*** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id)
00:34.22chkngumboi haven't seen that functionality clearly described as a function of a pbx or asterisk in particular. closest i've found is that its one of a pbx's most basic functions to "Establish connections (circuits) between the telephone sets of two users. (e.g. mapping a dialled number to a physical phone, ensuring the phone isn't already busy)"
00:47.11chkngumbonobody?
00:47.44jeevi guess not now
00:49.59*** join/#asterisk coppice (n=chatzill@175.202.17.210.dyn.pacific.net.hk)
00:52.13*** join/#asterisk qorky (n=qorky@202-65-72-37-wireless.bbnet.com.au)
00:52.35qorkyneed some help please... with a legacy PABX and PRI card.
00:53.08qorkywhen dialing from the PABX. asterisk picks up the extension and start trying to use it before I dial all my digits on the PABX system.
00:53.21qorkyI believe i need to set the TIMEOUT digit etc.
00:53.27qorkybut it doesnt seem to be working.
00:54.03qorkyif i use a speed dial on the PABX it works fine. I guess that is because it is passing all the digits to asterisk within the time it is expecting.
00:54.16qorkycan anyone help?
01:01.51*** join/#asterisk Johnakabean (n=Johnakab@pool-72-82-113-183.nrflva.east.verizon.net)
01:02.14Johnakabeanhey guys the interdigit short timer and interdigit long timer are what?
01:02.35Johnakabeani know they are the timout before sending the SiP message but what is the difference
01:13.53*** join/#asterisk andrewy (i=andrewy@209.126.180.153)
01:23.35*** join/#asterisk chendy (n=chatzill@58.61.196.154)
01:32.01*** join/#asterisk macros73 (n=cs@c-67-163-224-69.hsd1.pa.comcast.net)
01:32.01Johnakabeanhey guys the interdigit short timer and interdigit long timer are what?
01:32.03Johnakabeani know they are the timout before sending the SiP message but what is the difference
01:34.13*** join/#asterisk [netman] (n=netman@203.Red-83-37-168.dynamicIP.rima-tde.net)
01:39.27*** join/#asterisk _-Jon-_ (n=jonmiron@CPE000d8861e8f7-CM00080d290642.cpe.net.cable.rogers.com)
01:39.39_-Jon-_Hey all, I have a call forwarding question..
01:40.03_-Jon-_Basically, I'm wondering it it's possible to activate call forwarding remotely
01:47.04*** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id)
02:01.13*** join/#asterisk flujan (n=flujan@201-42-71-186.dsl.telesp.net.br)
02:01.19flujanhello guys
02:01.31flujani am having a problem with the blindxfer and the # key.
02:01.48flujanWhen i use the transfer button from the softphone or the hardphone it works
02:02.02flujanbut it does not work when i use the # key and type the extension/phone number
02:02.06flujanhere goes the output
02:02.14flujanhttp://pastie.org/253375
02:02.16flujanany ideas?
02:05.55jayteewow, no solution in 3 minutes = quit
02:06.06jayteewith patience like that you'll never fix anything
02:12.57*** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun)
02:16.52*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
02:26.14*** join/#asterisk l2trace99 (n=jr@static-71-251-65-16.tampfl.fios.verizon.net)
02:32.39*** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com)
02:33.36*** join/#asterisk Paige (n=Paige@2001:470:1f05:531:21f:c6ff:fe48:8988)
02:33.58*** part/#asterisk Paige (n=Paige@2001:470:1f05:531:21f:c6ff:fe48:8988)
02:54.31*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)
02:55.13*** join/#asterisk moy (n=moy@189.169.61.29)
03:01.01*** join/#asterisk apollonx (i=kit@193.19.189.38.STATIC.ISP.KZ)
03:01.29*** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
03:01.29*** mode/#asterisk [+o lmadsen] by ChanServ
03:02.08*** join/#asterisk hittop (n=nevercar@CPE001839bd0eb2-CM0014f8c23dbe.cpe.net.cable.rogers.com)
03:02.28lmadsenhowdy!
03:10.58jeevmy body shut down
03:11.00jeevi was so tired
03:11.59jeevi just woke up from a wack nap
03:12.49jeevdamn
03:12.55jeevalmost had a full on conversation with myself
03:13.22*** join/#asterisk AndyML (n=alauppe@pool-71-185-74-86.phlapa.fios.verizon.net)
03:13.34*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)
03:14.43AndyMLWhat is the default behavior for a blind transfer in asterisk 1.4? does it eventually go to voicemail?
03:15.21*** join/#asterisk hoegaatit (n=laa@c-67-160-226-244.hsd1.ca.comcast.net)
03:15.27_ShrikEAndyML: that entirely depends on what you have your dialplan configured to do.
03:16.33AndyMLok - that is what i was thinking as I was typing the question.
03:17.17*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net)
03:18.30*** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
03:18.30*** mode/#asterisk [+o lmadsen] by ChanServ
03:22.50*** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net)
03:22.57AndyMLany idea if there is a channel variable that you could use to get the channel of the device that transfered a call into a particular extension? for example - you have a custom pattern matching extension in the dialplan that rings a device for 20 seconds, then you want it to sent the call back to the person that initially transferred it (like how call parking timeouts go back to the original parker.)\
03:25.01*** part/#asterisk AndyML (n=alauppe@pool-71-185-74-86.phlapa.fios.verizon.net)
03:26.34*** join/#asterisk AndyML (n=alauppe@pool-71-185-74-86.phlapa.fios.verizon.net)
03:26.46AndyMLoops - did anyone answer me? i got disconnected...
03:26.54jeevnop
03:27.18AndyMLdarn. i was optimistic...
03:27.24jeevif you transfer a call
03:27.28jeevit'll come back anyway
03:27.31jeevi dont get what you're trying to do
03:27.44AndyMLwhy would it come back anyway?
03:28.09AndyMLwhen you blind transfer a call, its gone.
03:28.42AndyMLsure - if you do an attended transfer you can get it back anytime, but if you just want to transfer and get out, its essentially gone.
03:29.31jeevis call parking considered a blind transfer?
03:29.33AndyMLthese people want to transfer the call, move onto another call, and if the person they transfered it to isn't there, they want the call back, rather than it going to voicemail.
03:29.47*** join/#asterisk terracon (n=greisky@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com)
03:30.17jeevmy call parking does that
03:30.35AndyMLproblem with call parking is that you then have to tell the person that the call is for that someone called them and is waiting on line 701. blind transfer takes them to a different part of the dialplan - to ring a different phone for X seconds.
03:31.05AndyMLbut i want to configure that part of the dialplan to send the call back 20 seconds or so later - JUST like call parking.
03:31.15AndyMLor rather, just like call parking's timeout response
03:31.46jeevso if you dont want to do call parking
03:31.50jeevwhat do you want to do?
03:31.55jeevjust send it over and never see it again?
03:31.57AndyMLtransfer
03:32.17jeevahh
03:32.27AndyMLif you get a call on extension 101, and want to send it to the person at extension 102, you transfer it.
03:32.47jeevman, my friends office is full of the most pretentious people. if i can manage to make them work with call parking, anyone else can
03:33.22AndyMLusually the dialplan is configured for that call to ring for a while, then go to voicemail. This customer absolutely demands that, instead of going to voicemail, the call ring back to 101
03:33.52*** join/#asterisk HaMYaI (n=LAMER@ppp-58-8-6-136.revip2.asianet.co.th)
03:33.57jeevhttp://forums.whirlpool.net.au/forum-replies-archive.cfm/680210.html
03:34.37AndyMLmy thought was to create a section in the dialplan for like #XXX or 9XXX - something other than the regular extension - that if transfered to, would ring XXX for a while, then callback the person that transfered the call, but I can't seem to figure out what channel variable the original person was at to return the call to them...
03:35.05AndyMLwow jeev - nice find.
03:36.01jeevhope it works for you
03:36.32AndyMLthat is pretty much exactly what I was looking for. Looks like instead of using a channel variable that is already defined, he creates one before making the transfer.
03:36.43AndyMLthanks man.
03:36.46*** join/#asterisk ppyy (n=ppyy@58.216.74.139)
03:36.56jayteeAndyML, check in the asterisk docs, there is a file called channelvariables.txt that explains the use of variables in the dialplan and has a list of them towards the end of the file.
03:37.00ppyy~centos52bug
03:37.00jbot[~centos52bug] There is a bug compiling Zaptel  up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver  in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889
03:37.10jeevkills jaytee with Fender's katana
03:37.16*** join/#asterisk xenonex (n=xenonex@89.218.237.133)
03:37.22*** part/#asterisk AndyML (n=alauppe@pool-71-185-74-86.phlapa.fios.verizon.net)
03:37.39jeevjaytee, i fixed my dtmf issues.
03:38.42jayteethought he heard something but he must have been imagining it cuz he's dead.
03:39.15jeevtakes a crap on jaytee's body to give flies incentive.
03:40.13*** join/#asterisk MaliutaLap (n=nikolai@kiev.lusan.id.au)
03:43.42*** join/#asterisk hoegaatit (n=laa@c-67-160-226-244.hsd1.ca.comcast.net)
03:47.19jaytee[TK]D-Fender, PING?
03:48.20*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
03:49.15HaMYaII 'm trying to detect hangups on my TDM400 + 4 FXOs using "hanguponpolarityswitch=yes ; busydetect=yes ; busycount=2 ; busypattern = 500,499 ; callprogress = yes"
03:50.15HaMYaIbut it just doesn't work for me, is it because the amplitude of the hangup tones is low?
03:50.55HaMYaII also tried busypattern = 500,500 but still doesn't work
03:51.22hoegaatitI would like to call AddQueueMember when a voip client registers with my pbx.  I see you can trap the PeerStatus AMI event with PyStar and then issue an AddQueueMember call through that, but is there a cleaner 100% asterisk config way of doing this?
04:10.42*** join/#asterisk ElSonico (n=tav@dyn35-116.vpn.utu.fi)
04:11.55jeev~book
04:11.55jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
04:15.32*** join/#asterisk ZX81 (n=matt@202.20.97.211)
04:21.01*** join/#asterisk macros73 (n=cs@c-67-163-224-69.hsd1.pa.comcast.net)
04:29.38*** join/#asterisk LoOoD (n=gman@64.201.247.2)
04:32.28*** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net)
04:33.42LoOoDThere any sip tool/client I can run from the command line?..
04:35.32*** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70)
04:41.25*** join/#asterisk inv_arp (n=junya@b07s03mr.corenetworks.net)
04:50.48*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net)
05:02.06*** join/#asterisk hi365_m (n=hi365@213.151.61.43)
05:03.31*** join/#asterisk hoegaatit (n=laa@c-67-160-226-244.hsd1.ca.comcast.net)
05:07.58*** join/#asterisk igorw (n=dasboot@d209-121-138-115.bchsia.telus.net)
05:08.09oilinkiis there an restiction of the default context name? can it include dots? sip.conf:context=foo.bar.com
05:14.33*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
05:17.31*** join/#asterisk salzh (n=chatzill@58.247.194.227)
05:23.26*** join/#asterisk Defraz (n=T0tal@70.102.238.228)
05:36.01*** join/#asterisk route (i=skarecro@64.62.195.91)
05:37.06routeHi, I have a brand new PIAF install, and Asterisk isn't playing any system recordings...  Well, if you stay on the line for over 4 minutes you'll hear about a quarter of a second of the main greeting, and about 4 minutes later another quarter of a second of it, and so on.
05:39.25jeevi dunno what a PIAF install is
05:39.32jeevhave you watched the console with debugging on ?
05:40.15routePBX in a Flash
05:40.20routeYes I have.
05:40.32routeIt says it's playing, but it doesn't play correctly.  No errors.
05:41.21routeFiles are .wav recorded at 8k 16bit just like they should be.  In fact, they are the exact same files from our old Asterisk box.
05:41.37routeTHey worked fine on there...  Even the pre-recorded system sounds don't play back correctly.
05:43.14jeevso you've checked the codec
05:43.16jeevit's the same codec?
05:43.28jeevor at least the proper codec?
05:43.47routeWhere do I check that?
05:43.49jeevpastebin.com your allow and disallow
05:43.53jeevshow me your sip.conf in pastebin.com
05:43.59jeevhide your passwords and crap
05:44.01*** join/#asterisk tengulre (n=tengulre@125.71.208.16)
05:44.43routeUhm, every line in sip.conf is commented out.
05:44.56*** join/#asterisk oej (n=olle@ns.webway.se)
05:45.27routesip_additional.conf has a lot of stuff in it, but I can't edit that with FreePBX.  It doesn't list that file at all.
05:45.51jeevgrep allow sip_additional.conf
05:46.00jeevpastebin.com
05:46.01jeevdont paste it here
05:46.44routedisallow=all, and allows are ulaw alaw gsm and g729...  Not a paste, I typed that.  :-P
05:47.11jeevfind your sounds directory and see what files are in it
05:47.12routeAll are there twice.
05:47.21jeevwhat are the extensions
05:47.22jeevare you behind nat ?
05:47.36routesounds directory has all the pre-recorded stuff, sounds/custom has the 2 custom greetings we recorded, and that worked on the old box.
05:47.50routein sounds/custom there are closed.wav and main.wav
05:48.13routeThe box isn't behind a nat, but the phones are.
05:48.13jeevwhat are the extension
05:48.15jeevextensions
05:48.22route.wav .wav
05:49.07jeevhmm, i'm not too sure how this works.. but when i was having audio issues.. i had only gsm files in my sounds directory and ulaw was passing it.. so it was sounding nasty
05:49.15*** join/#asterisk chendy (n=chatzill@58.251.110.69)
05:49.17jeevso installed the sounds and got the .ulaw extension sounds and all is well
05:50.55routeI don't understand why the pre-recorded sounds won't work on a fresh install though.
05:51.07jeevwell, i had built mine from source and had only installed .gsm and i didn't realize that
05:51.24routeI even did update-scripts and update-fixes
05:51.34jeevwell, i dunt know how that works..
05:52.13jeevasterisk-core-sounds-en-ulaw-1.4.9.tar.gz
05:52.15jeevthat may be your issue man
05:52.21routeok, my custom recordings are .wav, but the rest of the sounds are .gsm.  I just looked at them again.  I thought it had both in there.
05:52.25jeevi'm not that great with asterisk.. but look for that and empty it into the sounds folder
05:52.39jeevdebug it and see what codec it's playing it at
05:52.48jeevi'm sure ulaw would play wav though.. that's how my voicemail is recorded
05:53.45route-- <Local/1908XXXXXXX@from-sip-external-1ed9,2> Playing 'custom/closed' (language 'en')
05:54.58jeevah
05:55.03jeevso it's your custom that's being stupid ?
05:55.11jeevnot the default sounds?
05:55.41routeNo, all sounds.
05:56.03route-- <SIP/1337-0c98d190> Playing 'ss-noservice' (language 'en')
05:56.28route-- Executing [s@from-pstn:4] Playback("SIP/1337-0c98d190", "ss-noservice") in new stack
05:57.17routeAfter 1 minute and 7 seconds it said "The Nu"
05:57.38routeI believe it's supposed to say "The number you have dialed....."
05:57.53jeevjust *try* getting those files, untarring it in the folder
05:57.55jeevit wont hurt.
05:58.06jeevi'm in no position to help anyone anyway but that's what my sound issue was.
05:59.57*** join/#asterisk astassistant (n=jtknapp@h-72-244-204-146.sttnwaho.dynamic.covad.net)
06:02.02*** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com)
06:03.31routeok, ulaw files are in the directory now.
06:04.39routeDo I need to change anything anywhere for it to use those files?
06:06.08HaMYaII've unsuccessfully detected hangups on my TDM400 + 4 FXOs using "hanguponpolarityswitch=yes ; busydetect=yes ; busycount=2 ; busypattern = 500,500 ; callprogress = yes"
06:06.51HaMYaIwhat else should I consider?
06:11.47routejeev ?
06:12.28*** join/#asterisk sergee (n=serg@voip1.west-call.com)
06:13.07jeevah sorry
06:13.09jeevwell
06:13.19jeevjust change a sound to that
06:14.14jeevtry setting one to 'privacy-your-callerid-is'
06:14.16jeevand see what happens
06:14.33*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
06:15.41jeevor vm-goodbye
06:16.50*** join/#asterisk mvanbaak (n=michiel@vanbaak.xs4all.nl)
06:18.43*** join/#asterisk obnauticus_ (n=obnautic@about/windows/regular/obnauticus)
06:18.46routeStill not playing.
06:19.26jeevbut it shows it ?
06:19.41jeevsip debug and pastebin the whole thing
06:19.50jeevincluding the call
06:23.32routegenzaptelconf -svdM
06:23.41routeThat's all I needed to do to solve the problem.
06:23.56routeNow all sounds seem to be playing correctly.  YAY!
06:24.31*** join/#asterisk ElSonico (n=tav@nat/ibm/x-057e88ec7e2e8c0c)
06:26.38*** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-283d2df17e798cfb)
06:27.06jeevoh heh
06:27.39*** join/#asterisk VaNNi (n=VaNNi___@38.98.61.143)
06:27.53jeevsleep time. night
06:32.34*** join/#asterisk oilinki3 (n=oil@ppp-124-120-15-56.revip2.asianet.co.th)
06:33.11pputmanHas anyone heard of the G.CLEAR codec, and know if it has another name that would be supported by asterisk?
06:33.42Corydon76-digpputman: sounds like a marketing gimmick
06:34.17pputmanCorydon76-dig, a customer is telling me it's the codec used for an ISDN line, so I assumed he meant ulaw, but now I'm thinking he means the equivalent of setting clear=1-23 in zaptel.conf
06:36.12Strom_Mlol.
06:36.57*** join/#asterisk sack (n=sack@212.Red-81-44-114.dynamicIP.rima-tde.net)
06:38.26Corydon76-digulaw, alaw, or G.721 (ADPCM)
06:39.02Corydon76-digThose are the only codecs specified in the ISDN spec
06:39.05pputmanright, I think he's just confused though
07:07.38*** join/#asterisk korihor (n=korihor@190.78.32.60)
07:10.24*** join/#asterisk MSone (n=MSone@196.203.51.52)
07:24.29*** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola)
07:35.43*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)
07:39.02*** join/#asterisk blogbasti (n=blogbast@calypso.planet-ic.de)
07:41.05*** join/#asterisk ragix (n=rage@125-238-132-29.broadband-telecom.global-gateway.net.nz)
07:42.13obnauticusCorydon76-dig: >:|
07:47.29*** join/#asterisk bl4q (n=Bl@dslb-088-067-047-201.pools.arcor-ip.net)
07:47.36*** join/#asterisk dominic1 (n=dob@213.221.82.242)
07:47.42dominic1hi folks
07:48.57dominic1can anybod tell me why the second channel in the asterisk parkaction ist not dialed after giving up the call?
07:52.23*** join/#asterisk E-bola (i=psybnc@ip181.rev112.brygge.net)
07:52.45E-bolaWhere there any big changes in regards to conferences/meetme from Asterisk 1.2 to 1.4?
07:53.40pollerIs there any cmd that lets me listen in on ongoing calls?
07:57.55*** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net)
08:04.16E-bolaHmm can you change the pincode for a meetme room with a command?
08:07.25mort_gibE-bola: not without reloading
08:08.15mort_gibYou want to write a small ivr that only allows the user to enter based on a key fetched from a  database
08:08.16E-bolaHmm i just found the Web-MeetMe util
08:08.22E-bolathat seems to make it possible, via the web
08:08.37E-bolamort_gib: Ahh your right, that would work just as well
08:09.10E-bolaI was just a bit worried that if the password stayed the same always, anybody who had ever been on a conference could join any future conferences
08:09.35mort_gibYes that is true...
08:10.14*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
08:10.58mort_gibYou have different options, a perl script that changed the pwd every night, and reloads, finally sending the new pwds to a reception...
08:12.07E-bolaYep, but web meetme seems as the most userfriendly
08:12.13mort_gibsure :-)
08:12.16E-bolaif it works etc....
08:12.26E-bolamort_gib: have you ever tried it? Web meetme
08:12.32mort_gibYeah,. but you want it to be automatic right
08:12.51mort_gibI must admit I haven't had a look at web-meetme
08:13.01E-bolaNah, optimally the conference organiser can set it up before the conference and give it to the attendees
08:13.13E-bolaWhich seems preicely what web meetme allows you to do
08:13.39mort_gibSo after the conference has stopped the password is revoked??
08:15.51E-bolaThe whole conference room is deleted
08:16.11E-bolaor maybe the password is just left untill next time a room is needed, and its changed then
08:19.27*** join/#asterisk zydoon (n=zydoon@213.150.170.26)
08:19.39*** part/#asterisk zydoon (n=zydoon@213.150.170.26)
08:20.25*** join/#asterisk tkbeat (n=tk@80.64.182.204)
08:21.29*** join/#asterisk xenonex (n=xenonex@88.204.197.139)
08:26.25*** join/#asterisk PepOSX (n=angeldav@190.72.129.75)
08:34.20*** join/#asterisk jivco (n=jivco@85.187.217.6)
08:49.18*** part/#asterisk gones (n=gones@203.193.37.251)
08:53.31*** join/#asterisk wonderworld (n=ww@ip-62-143-216-14.hsi.ish.de)
08:59.04*** join/#asterisk obnauticus_ (n=obnautic@about/windows/regular/obnauticus)
09:01.34*** join/#asterisk miloux (n=miloux@static-213.88.173.45.addr.tdcsong.se)
09:03.56*** join/#asterisk oej (n=olle@ns.webway.se)
09:06.09HaMYaIin the Asterisk's Makefile, there's an option BUSYDETECT+= #-DBUSYDETECT_COMPARE_TONE_AND_SILENCE
09:06.50HaMYaII tried to uncomment that and compiled but it shows errors
09:10.44dominic1hi, short question. Why do I need the extension on the manager redirectaction, when I bridge two channels?
09:12.27_E-bolamort_gib: You still here?
09:13.08*** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net)
09:13.46mort_gibYeah, I'm here :-)
09:14.26_E-bolaI drove out to the client needing the conference setup. And they actualy had something else in mind. The boss wants to be able to dial each participant in the conference 1 by one to "let them into the room"
09:14.35_E-bolaIS this sort of setup possible at all with asterisk?
09:17.14mort_gibEh, my clients do that, they ask the receptionist to do it :-)
09:17.28mort_gib-You want to have a look at creating dial files...
09:17.34dominic1how can I redirect a parked call to my current active call?
09:17.43dominic1does anybody know that?
09:18.08mort_gibdominic: you want a three way join??
09:18.13_E-bolamort_gib: no i think u missunderstand. Lets say u got a conference going with 2-4 people
09:18.46_E-bolamort_gib: And suddenly need the advice of a 5'th person. How can you ring this person, and get him joined to the conference call?
09:18.51dominic1no I want that my call to a specific person is hung up and the person I called to speaks to a person which was in parkposition
09:18.54mort_gibYes, so when J.Doe.1 calls 8000 (meetme) * automatically calls 4 other numbers and join them to 8000
09:19.27dominic1bridgeaction is only available in 1.6
09:19.35dominic1is there anything else in 1.4
09:19.59_E-bolamort_gib: cant it be done more manualy? Like dial something while in the room, and then you "privately" talk to the 5th guy, and if he agrees to join the room. You press a button and he's joined to the room
09:20.51mort_gibAh, You could 1. Leave the conference, call the 5th participant, transfer to the conf romm and join yourself 2. put the conf on hold on your handset, call 5th and hava a little prep talk transfer to meetme and go back to meetme
09:21.28_E-bolaahh right
09:21.47_E-bolathats what i want. What happens if you transfer a call to a conf room? They get asked for the pin code or?
09:22.59mort_gibNormally I don't put PIN codes on conf rooms
09:23.25_E-bolaI guess if i set it up so all conf rooms are only accisible locally i dont have to either
09:23.35_E-bolaGreat, thanks alot for your help mort_gib :)
09:23.41mort_gibBut if you do the person calling the conf room gets the request, in this case the you would get the request as you are dialing BEFORE transferring
09:24.01mort_gib;-) Your welcome, where are yo in the world??
09:24.12_E-bolaDenmark (North europe)
09:24.32obnauticusdrmessano: are you there -- honey?
09:24.41mort_gib-This is why my clients ask their PA's or reception staff to handle conf calls
09:24.51mort_gib-Yeah :-) Hvordan er vejret??
09:24.58_E-bolalol
09:25.12_E-bolaFint fint Morten :P
09:25.20mort_gibHer er det 30+
09:25.28_E-bolabastard :)
09:25.32_E-bolaWhere's here?
09:25.43mort_gibSorry...
09:28.22mort_gibGibraltar/Spain
09:42.15*** join/#asterisk gones (n=gones@203.193.37.251)
09:48.12*** join/#asterisk jazzmann (n=chatzill@cpc1-lutn9-0-0-cust163.lutn.cable.ntl.com)
09:49.47obnauticusbridgeaction?
09:49.50jazzmannUnable to read config file mobile.conf       I am trying to add chan_mobile
09:49.50obnauticuswhat is this business dominic1 ?
09:50.04obnauticusjazzmann: have fun with that, i hate chan_mobile, it was mean to me :|
09:50.07obnauticuscrosses arms
09:51.16jazzmannhttp://www.geek-pages.com/articles/asterisk/howto_build_and_configure_chan_mobile_on_trixbox.html
09:51.17jazzmannthis is the tutorial I was following in mandriva
09:51.51*** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb)
09:52.09jazzmannI really need some help in installing it
09:59.03dominic1obnauticus: it's atxfer but initiated by a software
10:09.19*** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com)
10:11.51*** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net)
10:14.10*** join/#asterisk Jenna (i=JJ@gateway/tor/x-71af9c9123fac3b2)
10:33.32*** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279552046.dsl.bell.ca)
10:42.18*** join/#asterisk chevap (n=chevap@89.201.199.176)
10:42.33*** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br)
11:00.32*** join/#asterisk Linker3000L (n=chatzill@host81-133-250-152.in-addr.btopenworld.com)
11:21.11*** join/#asterisk shinao1 (n=shinao1@smtp.gtbank.com)
11:23.35*** join/#asterisk mocker (n=kyle@206.55.118.90)
11:23.50mockerI'm having a problem with a Rhino channel bank that I'm getting configured.
11:24.12mockerI have the T1 sending signal and framing is good, but when I press digits on the phone nothing seems to be detected.
11:24.28mockerSo.. it's analog phone connected to channel bank which is connected to asterisk
11:33.10*** join/#asterisk Great_Anta_Baka (i=c4219f6b@gateway/web/ajax/mibbit.com/x-5494840d5532a385)
11:37.17*** join/#asterisk bartpbx (n=bartpbx@217.24.210.202)
11:38.00bartpbxhello
11:40.41*** join/#asterisk [netman] (n=netman@203.Red-83-37-168.dynamicIP.rima-tde.net)
11:42.27*** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk)
11:49.48*** join/#asterisk scampbell (n=scampbel@199.105.195.156)
11:55.04*** join/#asterisk J4zen (n=jeroen@a82-95-153-17.adsl.xs4all.nl)
11:59.15*** join/#asterisk washburnello (n=johnsy@h23.238.30.71.dynamic.ip.windstream.net)
12:01.47J4zenDoes anyone happen to have implemented HylaFax into Asterisk?
12:01.58*** part/#asterisk work-burnello (n=johnsy@h23.238.30.71.dynamic.ip.windstream.net)
12:01.58bartpbxys
12:02.11J4zenDid you ever get an error such as "Failure to train remote modem at 2400 bps or minimum speed" ?
12:02.15J4zenresulting in fax failure
12:02.24J4zeni cant recieve faxes because of that
12:02.31bartpbxno, never seen this before
12:02.38bartpbxbut looks like a problem on the remote side
12:02.53J4zenyes, but in this case the remote modem is my PBX
12:05.52*** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163)
12:07.47[netman]so , u should try to send a fax to a different number to check it out
12:09.01*** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla)
12:09.01*** mode/#asterisk [+o russellb] by ChanServ
12:10.12Wayhighsup all
12:11.56J4zen[netman]: Yeah i have, outgoing fax works fine to all numbers
12:12.07J4zenits the incoming faxes on my PBX that don't work
12:12.27J4zenif someone faxes to me, in this case i used HylaFax to fax myself.. they will get a repsonse stating "Failure to train remote modem at 2400 bps or minimum speed"
12:12.40J4zenso outgoing works just fine
12:12.43J4zenincoming is broken
12:14.38mockerJ4zen: Welcome to faxing w/ asterisk :)
12:16.07[TK]D-FenderJ4zen: incoming and outgoing over WHAT?  And last I recall 9600 was the minimum, no?
12:17.08mockeropens a support ticket w/ Rhino.
12:18.05*** join/#asterisk smellynoser (n=ashley@87-194-183-38.bethere.co.uk)
12:18.15*** part/#asterisk smellynoser (n=ashley@87-194-183-38.bethere.co.uk)
12:22.18keith49600 used to be the maximum
12:22.28keith4i hear tell of some fancy ones that can do 14400, though
12:22.48J4zen[TK]D-Fender: In more detail >  Fax going out through SIP-trunk(sending) -> PSTN -> SIP-trunk(recieving) -> PBX -> Inbound Route -> HylaFax
12:23.03keith4ugh
12:23.11keith4it's a miracle it works at all
12:23.32[TK]D-Fenderkeith4: 33.6kbps
12:23.35J4zenit sounds edgy, it might just be due to the fact that im both sending and recieving the fax on the same SIP-trunk
12:23.44[netman]J4zen: so *everybody* who wants to fax you generetes that error msg, I see
12:23.46keith4[TK]D-Fender: incredible. soon we'll have flying cars!
12:24.15keith4(shows when the last time I used a modem was, I guess)
12:24.24Wayhigh?? P-Asserted-Identity
12:24.27[netman]J4zen: hylafax doesn't see a SIP trunk..., only a iaxmodem, I don't think that could be the reason
12:24.38J4zen[netman]: Well i tested it on one external analogue fax machine and from my own SIP-trunk.. seeing as we're not activley using FAX thru VoIP right now; Yes, *everyone*.
12:24.52J4zeni see
12:25.16J4zenill run some more tests, gotta love faxing
12:25.49Wayhighis still looking for multiple trunks that pass P-Asserted-Identity
12:26.30[TK]D-FenderWayhigh: flowroute.com
12:26.59Wayhighyeah.. I know.. I want to find more of them that pass it
12:27.01keith4is that the one that was used to demonstrate "unmasking" hidden CID?
12:27.34[TK]D-Fenderkeith4: Yup.  All the kiddies want to be the shiznit y0
12:27.41WayhighI figure they can't be the only one and it is an interesting technique. I know that a number of the wholesalers are requiring p-asserted-identity headers now
12:28.15keith4I just don't accept private calls. problem solved
12:28.34WayhighI want it for something different though.. I'm testing something I discovered yesterday
12:28.56keith4[TK]D-Fender: maybe someday I'll get a flowroute.com account, so I can h4x CID too!
12:29.03[TK]D-FenderWayhigh: Namely?
12:30.26Wayhighfender: a method of screenpopping
12:30.39*** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
12:30.39*** mode/#asterisk [+o lmadsen] by ChanServ
12:32.10[TK]D-FenderWayhigh: ummm, you want to get what from the telco exactly?
12:35.03*** join/#asterisk TurboTech (n=Sono@katy-adsl-dhcp-64-92-26-35.consolidated.net)
12:37.01*** join/#asterisk aiksa[LV] (n=aiks@mx.fiveplus.lv)
12:37.07aiksa[LV]hi everyone.
12:37.12*** join/#asterisk linuxstb_ (n=linuxstb@rockbox/developer/linuxstb)
12:37.36aiksa[LV]where could I find full list of asterisk events (sent to manager interface)
12:37.51[TK]D-Fender~book
12:37.51jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
12:37.53[TK]D-Fenderaiksa[LV]: ^^^^^^^^
12:38.22aiksa[LV][TK]D-Fender: hi. somehow i overlooked that
12:38.44aiksa[LV]http://tfot.leifmadsen.com,thanks
12:38.54[TK]D-Fenderaiksa[LV]: thats like asking where you jacket is with your eyes closed.
12:40.56*** join/#asterisk _nn (n=nn@unaffiliated/nn)
12:45.30aiksa[LV]:)
12:46.11aiksa[LV]iiiiihj
12:48.07*** join/#asterisk brodiem (n=brodiem@rrcs-24-227-171-186.sw.biz.rr.com)
12:50.34*** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130)
12:51.53aiksa[LV]thats more like looking for the glasses when you have them on your eyes already
12:52.09aiksa[LV][TK]D-Fender: thanks nevertheless
12:52.29[TK]D-Fenderaiksa[LV]: I was being kind...
12:53.42aiksa[LV]i just somehow thought that was not covered in the book
12:54.35aiksa[LV]dont know why
13:01.55*** join/#asterisk luxxx77 (n=lutz1@f051064033.adsl.alicedsl.de)
13:02.17zambawhat can i do on my router (software router running linux) to give voip higher priority?
13:02.27zambai'm talking about the outgoing stream now
13:02.49zambai'm perfectly aware there's little i can do with throttling the incoming
13:03.24[TK]D-Fenderzamba: http://www.google.ca/search?hl=en&q=linux+bandwidth+management&btnG=Google+Search&meta=
13:04.40zamba[TK]D-Fender: but what software tool do you guys recommend?
13:05.02zambai've heard about wondershaper
13:05.13zambabut i need something -simple- to set up
13:05.28zambai see m0n0wall, but that's a complete distribution, isn't it
13:05.29zamba?
13:05.39[TK]D-Fenderzamba: I have also heard of Wondershaper, and it was simple when I looked at it back then
13:05.42mockerzamba: Outgoing to the Internet?
13:05.45zambamocker: yeah
13:05.52[TK]D-Fenderzamba: Yes, m0n0wall is a distro
13:05.59mockerSo high priority until it leaves your home/office, then nobody cares. :)
13:06.00zambaso that's not an option
13:06.25zambamocker: yeah, exactly.. but i need a way to shape it on my router and am thus looking for a software package that can help me fix this
13:07.32keith4monowall is a mean and lean little bugger
13:07.44zambawell, reinstalling the router isn't exacly an option
13:07.50keith4i had it running for years, on an 8M kodak CF card from an ancient digital camera
13:08.27luxxx77Hello! Can anyone tell me if i can configure asterisk 1.4.21.2 with an SVN install of asterisk-gui/branches/2.0/? Is 2.0 compatible with 1.4.?
13:08.32zambaespecially not since the router is around 400 km away from me :)
13:08.41zambakeith4: i'm running the router on a 1 GB CF atm :)
13:08.41*** join/#asterisk shinao1 (n=shinao1@smtp.gtbplc.com)
13:08.48zambakeith4: did you use flashybrid then?
13:08.54keith4what what?
13:09.11zambakeith4: to prevent writes to the flash?
13:09.25keith4monowall doesn't really work that way
13:09.34zambahow does monowall work?
13:09.37keith4it only writes if you change the config
13:09.44keith4(as long as you log so a syslog server)
13:09.57*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
13:10.09zambaah, so it's embedded per design?
13:10.13keith4yep
13:10.15zambacool
13:10.40keith4doubtful that it will help your issue. but, whatever
13:11.28luxxx77does somebody konw here a bit about asterisk-gui?
13:13.44keith4points at the topic
13:15.31*** part/#asterisk luxxx77 (n=lutz1@f051064033.adsl.alicedsl.de)
13:18.48*** join/#asterisk Nate187 (n=Nate187@gw.bigrivertel.net)
13:19.08*** join/#asterisk linuxstb__ (n=linuxstb@rockbox/developer/linuxstb)
13:24.56Cheap_tdm400:)
13:26.36*** join/#asterisk moy (n=moy@nat/ibm/x-b9ff527d51020876)
13:26.42*** join/#asterisk JenniferAkemi (n=akemi@76-10-152-16.dsl.teksavvy.com)
13:26.47JenniferAkemiMorning!
13:30.15mgromanomg hi2u!!!!
13:31.13[TK]D-FenderWayhigh: Just F'N ebay it
13:31.33Wayhighfender: I may do that.. I just didn't wanna go through that kinda hassle :P
13:31.47mgroman... stoner ...
13:33.56*** part/#asterisk E-bola (i=psybnc@ip181.rev112.brygge.net)
13:34.36WayhighI think I may sell my S100think I may sell a S100-FX ATA as well
13:34.45Wayhigherr.. heh..
13:37.21*** join/#asterisk fogo (n=Paul@rs-69-169-132-200-0003.broadweave.net)
13:39.51*** join/#asterisk aiksa[LV] (n=aiks@mx.fiveplus.lv)
13:40.11aiksa[LV][TK]D-Fender: are you sure The Book had a list of asterisk manager events?
13:40.25[TK]D-Fenderaiksa[LV]: Yes.
13:40.45aiksa[LV]all i found was manager actions
13:42.28[TK]D-Fenderaiksa[LV]: And have you tried the WIKI?
13:42.29aiksa[LV]the unlink as a word was mentioned in three places in the book, but none of them was even closely related to that specific event
13:42.37aiksa[LV]you mean voip-info?
13:43.05*** join/#asterisk JayTee52 (n=jforde@unaffiliated/jaytee)
13:43.26[TK]D-Fenderaiksa[LV]: Clearly.
13:43.32aiksa[LV]of course I have. blame on me - voip-info is the first resource i go to for a quick reference
13:43.46aiksa[LV]but I have a suspicion that their event list is not complete
13:43.50[TK]D-Fenderaiksa[LV]: Seems to be a list there.
13:44.08[TK]D-Fenderaiksa[LV]: Why don't you leave a listener on AMI and log the events it sees?
13:44.55aiksa[LV][TK]D-Fender: what I was looking for - a documentation of specific events and circumstances under which they are triggered
13:45.20*** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com)
13:45.31aiksa[LV]i have already logged the events and i am not sure about one thing - thats was reason why I wanted to read up on it
13:46.03[TK]D-Fenderaiksa[LV]: Maybe you should have come right out with this specific thing in the first place.
13:46.50aiksa[LV][TK]D-Fender: I will come out - but I first wanted to know if there is any other information resource which I didnt consult in the first place
13:47.09*** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
13:47.09*** mode/#asterisk [+o blitzrage] by ChanServ
13:48.04*** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com)
13:48.44aiksa[LV][TK]D-Fender: the question was - If I have a queue served by a couple of SIP extensions as queueMembers. then why doesnt dial event happens before link event for an incomming call from queue
13:49.37aiksa[LV]and are there alternatives to detect that specific call member have been called by a person waiting in the queue.
13:50.10*** join/#asterisk Iptime (n=chatzill@119.82.102-84.rev.gaoland.net)
13:50.14Iptimehi
13:50.32aiksa[LV]as far as I can see for now - the Dial event gets launched only when there is a Dial command in the dialplan, but not in the applications
13:51.12*** join/#asterisk iulius (n=iulius@mail1.technologieshq.com)
13:52.39[TK]D-Fenderaiksa[LV]: Guess thats just the way it is..
13:53.11Iptimehello
13:53.17*** join/#asterisk luxxx77 (n=luxxx77@f051064033.adsl.alicedsl.de)
13:54.22*** part/#asterisk luxxx77 (n=luxxx77@f051064033.adsl.alicedsl.de)
13:54.45IptimeI am very new to the Asterisk IP telephony. i got a small and stupid question like you can say, boff my question is simple, i got asterisk server at home, and create lots of sip extensions, the problem is all my sip extension users not "online" all the time, if i want to join one my sip user, i have to check  each time from my server by doing this command line : " sip show peers " and from...
13:54.46Iptime...my work i can't check if the user is online or not, to understand well my question, i just need the same functionality as skype,( when a user come online u heard a  bip ) like i want my phone ring twice time. In hope of gotting best answer here
13:55.45[TK]D-FenderIptime: Go read about "presence" on the WIKI.
13:55.48[TK]D-Fender~wikis
13:55.49jbot[~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.  http://www.voip-info.org (c) Arte Marketing Inc / CommPartners
13:55.50[TK]D-Fender^^^^^^^^^^^
13:56.36*** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org)
13:57.50*** join/#asterisk Assimilate (n=Assimila@216.83.78.108)
13:58.08Iptimeis it this one the answ<wer ?
13:58.09aiksa[LV][TK]D-Fender: - so you dont know of any other source for more detailed explanation on asterisk manager interface events?
13:58.10Iptimehttp://www.voip-info.org/wiki/view/Asterisk+presence
13:58.31*** join/#asterisk axisys (n=axisys@155.70.141.45)
13:58.37[TK]D-Fenderaiksa[LV]: Probably on Mantis where it is explained.  Just Goolge whatever evernt you're wondering about.
13:59.12[TK]D-FenderIptime: Does it LOOK like what you want?
13:59.33*** join/#asterisk luxxx77 (n=luxxx77@f051064033.adsl.alicedsl.de)
14:00.23Iptimewhat i want is exactly same as msn messenger
14:00.39Iptimewhen a user present on my server
14:00.53Iptimewant to have alert
14:01.28*** join/#asterisk oilinki (n=oil@ppp-124-120-251-104.revip2.asianet.co.th)
14:03.52[TK]D-FenderIptime: use presence on a soft-phone, or FOP, or any of the other pile of status viewers out there.
14:06.36*** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view)
14:15.34*** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view)
14:18.02mgromanDo people still use nano to code at digium?
14:18.18Nuggetoh dear.
14:18.54Nuggethttp://macnugget.org/photos/strange/curves
14:19.04mgromanvirus?
14:19.08[TK]D-Fenderfires up the old flammenwerfer
14:19.13russellbhas never used nano to code
14:19.13russellbheh
14:19.28russellbpoints at Qwell
14:19.30russellbonly he does
14:19.48russellbmost are using vim or emacs ...
14:19.55mgromanI remember a long time ago, like 6 months, one of the admins here suggested I try vim, and i have been using it since, and im starting to get fast at it, so whoever it was, thanks
14:20.01Nuggetrussell uses a magnetized needle and a steady hand.
14:20.08russellb<3 vim !!!
14:20.13*** join/#asterisk km- (n=pgrace@fmservices.v6.fierymoon.com)
14:20.15d3wayneQwell is silly with his nano
14:20.29Nuggetusing nano to code is like using scissors to mow the lawn.
14:20.37*** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130)
14:20.44Nuggetsure, the UI is simple, but it's certainly not easier.
14:20.49km-ooh did I join into a vi/nano penis contest?!
14:20.51*** join/#asterisk ManxPower (n=manxpowe@45.sub-75-248-217.myvzw.com)
14:20.54russellbkm-: yes
14:21.10km-lemme throw it in that I love nano until I need vi's feature set.
14:21.20russellbalright fence sitter
14:21.22Iptimethcs Tk
14:21.26russellb:-p
14:21.29Nuggetheh
14:21.45Iptimethe problem is : i use linksys adaptor
14:21.56Iptimeand there there is no that option
14:22.03seanbrightemacs <3
14:22.29seanbrighti have to admit though... the *only* reason i don't use vim is because...
14:22.31russellbemacs is for noobs
14:22.33km-russel: gotta play the politics, man!
14:22.35seanbrighti have a penis and testicles
14:22.40Iptimeok
14:22.42russellbkm-: not on IRC you don't!
14:22.48Iptimei got an other question
14:22.55mgromanis sorry for starting a text editor crusade
14:23.04russellbmgroman: ;)
14:23.04km-russellb: bah!
14:23.06Iptimei got an asterisk server
14:23.21km-Me too!!!
14:23.27russellbdoesn't
14:23.28russellb:(
14:23.34Iptimeis it possible to listen what r they talking using my server for example ?
14:23.43km-In fact, I'm about to have an entire ESX cluster of them
14:23.49russellbIptime: *CLI> core show application ChanSpy
14:23.53[TK]D-FenderIptime: "core show applications like spy"
14:24.05russellbawww, [TK]D-Fender had to be more complete than me
14:24.07russellbI see how it is
14:24.22km-that was pretty sneaky with that "like" thrown in there.
14:24.22seanbrightExtenSpy is for noobs
14:24.33Iptimemy question is : asterisk is it secure or not .?4
14:24.46*** join/#asterisk kensuke_ (i=be0210a1@gateway/web/ajax/mibbit.com/x-28b76095ab4b22b4)
14:24.48km-Yeah, so get this.  I'm getting promoted at my company to R&D, I get to hack on asterisk all day every day!
14:24.49russellbIptime: depends how you set it up, and what you mean by secure, i guess
14:24.50[TK]D-Fenderseanbright: Yeah and its not like we have any of THOSE around here....
14:24.56seanbright[TK]D-Fender: :)
14:25.06*** join/#asterisk Rico29 (n=Rico@ARennes-257-1-128-156.w86-210.abo.wanadoo.fr)
14:25.32Iptimeis it possible to listen what ppl  r talking ?
14:25.32km-now the question is whether I can give patches back to the community for stuff I write.
14:25.44Iptimewith 2 extension ?
14:25.53russellbdamnit, Iptime, two people already answered that question
14:25.57russellbgo read about what those apps do
14:26.11russellbkm-: a question of your employer, you mean?
14:26.18km-yeah.
14:26.24russellbkm-: well I sure hope so.  :)
14:26.33Iptimeif i type this from asterisk cli
14:26.34km-I'm already signed up as a committer for asterisk, the question is whether I'll be allowed to commit it
14:26.36Iptimecore show applications like spy
14:26.37seanbrightmost employers have this nagging concept of "intellectual property"
14:26.39russellbkm-: If you have any questions about how the process works, feel free to contact me directly.  russell@digium.com
14:26.52Iptimeit can only show they are online or not
14:26.54russellbalready signed up as a committer?
14:26.56km-russellb: I think I was actually one of the first couple hundred people to sign mark's agreement
14:27.02[TK]D-FenderIptime: READ AGAIN
14:27.08Iptimemy question is how to pick a line and listen
14:27.08km-russellb: you know the, disclaim all rights to what you submitted
14:27.16Iptimewhat they r talking
14:27.21russellbkm-: Oh, that.  Yeah, we have a new license agreement in place now that is electronic.
14:27.23km-russellb: they were doing that wayyy back in the day
14:27.29[TK]D-FenderIptime: And we just gave you the command list to use to DO IT
14:27.29russellbnods
14:27.30km-oh, yeah, I had to fax mine in
14:27.45russellbkm-: You'll have to electronically agree to the new one via your account on bugs.digium.com
14:27.53km-ahhh
14:27.57*** join/#asterisk Arck-FR (n=Arck-FR@cvl92-2-82-228-145-232.fbx.proxad.net)
14:27.58km-lemme go see if I remember my password
14:28.09russellbkm-: once you log in, you should see "Sign License" at the top, I think
14:28.14Iptimelogically for me there must be have a number to connect the line
14:28.15km-I havent had much time to hack lately, my job has been 60 hours a week of operations
14:28.38russellbkm-: if you get into coding, hang out in some of the dev channels ... #asterisk-dev, -bugs, -commits
14:28.38Iptimeand the command not help
14:29.11[TK]D-FenderIptime: Go read each applications instructions
14:29.12km-russellb: yeah, I occasionally pop into -dev to harass corydon, et al
14:29.15*** join/#asterisk Dr-Linux|home (n=Nothing@119.63.130.34)
14:29.19russellbcool.
14:29.44km-wow, up to 11 windows in irssi heh
14:29.59km-hmm, how do I hit alt+11. :P
14:30.04Dr-Linux|homeI'm using Queues with asterisk, but i don't want it should play the message "You are caller number 2 ..." what should i do to disable on this message?
14:30.11Dr-Linux|homei mean position
14:30.19Iptimeok thx
14:30.19km-there's probably a flag on the app for it
14:30.29Nuggetalt-q through alt-p will get you windows 11-19 or whatever.
14:30.35km-nugget: hot!
14:30.45km-nugget wins for fun fact of the day
14:30.51Dr-Linux|homekm-: talking to me?
14:31.07km-dr-linux: yeah, I'd have to assume there's gotta be an option on the app for it.
14:31.23km-since way back it didn't used to do that, so it'd be a feature you should really allow to be turned on and off
14:31.30Dr-Linux|homekm-: yeah but i'd like to know that option
14:32.01Dr-Linux|homekm-: I made search on the web, but can't get appropirate solution for get if off
14:32.07Dr-Linux|homes/if/it
14:32.38km-curious
14:32.53[TK]D-FenderDr-Linux|home: Read the sample config.
14:33.17Dr-Linux|home[TK]D-Fender: already did
14:33.37Qwellrussellb: pfft, you kids these days and your regex and syntax highlighting
14:33.37[TK]D-FenderDr-Linux|home: then keep reading it till your eyes bleed
14:34.11seanbrightnano has syntax highlighting, hater.
14:34.49Dr-Linux|home[TK]D-Fender: hhm... can you give me hint for that option where i can disable the position?
14:35.26km-[tk]d-fender: are you referring to voip-info or something?  I'm curious on this now and want to read up :)
14:35.29fileDr-Linux|home: http://pastebin.com/m3f919d14 copy and pasted right from the queues.conf.sample file
14:35.54[TK]D-FenderDr-Linux|home: READ THE DAMN SAMPLE CONFIG FILE
14:35.57km-file: wow, that is extremely obvious.
14:36.05km-dr linux should have his phd stripped
14:36.22seanbrightinvents a way to stab people in the face over the internet
14:36.49*** part/#asterisk jivco (n=jivco@85.187.217.6)
14:37.10*** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view)
14:37.21[TK]D-Fenderseanbright: License it to me!
14:37.30km-yeah I'd totally license that patent.
14:37.44seanbrighton a per use basis
14:37.46seanbright:)
14:37.51km-finally a software patent I can agree with.
14:37.58km-though something as useful as that really should be open-sourced.
14:38.04seanbrightpfft
14:38.05Dr-Linux|homefile: I aready read that on wiki but when i do announce-frequency=0  it disable time announcement as well
14:38.08Dr-Linux|homewhich i don't want
14:38.49filedoubt you can get that granularity
14:39.15Dr-Linux|homeas i mentioned before i only want to disable the the message like "you are caller number ..."
14:39.36seanbrightDr-Linux|home: going to have to edit the source
14:39.39ManxPowerDr-Linux|home: it's an all or nothing thing.
14:39.51seanbrightDr-Linux|home: what version of asterisk are you running?
14:39.53ManxPowerWhy don't you just record an file and use it as MoH for those calls.
14:40.19Dr-Linux|homeseanbright: the server where i want to deploy it is 1.2.17
14:40.28seanbrightjeeeeeeeeeeeeeebus
14:40.47seanbrightit's going to be one of those days, huh?
14:40.55km-hahaha
14:41.10km-I'm running asterisk v0.9.8 and I hear there's this app called valet parking that I MUST HAVE.
14:41.18ManxPowerseanbright: every day is like that here, why do you think I hang out on #asterisk-cli mostly?
14:41.26Dr-Linux|homeseanbright: rest of all servers having 1.4.xx .... however never tried 1.6
14:41.40seanbrightDr-Linux|home: vanilla 1.2.17?  or do you have custom patches and such floating around in there?
14:42.04km-the documentation in that config file is wrong.  and/or is wrong, it should just be "and" if it turns off both.
14:42.33ManxPowerIn any case, the answer to Dr-Linux|home's question has been answered.
14:43.24Dr-Linux|homewhat is vanilla
14:43.31seanbrighta flavor
14:43.32Dr-Linux|homeVanilla is IceCream here
14:43.44ManxPowervanilla = no chances made
14:43.50seanbright^^^
14:43.51*** join/#asterisk ElSonico (n=tav@xdsl-179-7.nblnetworks.fi)
14:43.51ManxPowerno changes, that is.
14:43.53km-dr-linux: un-bespoke
14:44.00ManxPowervanilla also means "not kinky"
14:44.39Dr-Linux|homeseanbright: I don't knwo C .. so can't modify app_queue
14:44.52Dr-Linux|homehowever i'd like to know can i do that using 1.4 what i want?
14:45.03ManxPowerDr-Linux|home: then you are out of luck if my suggestion is not workable for you.
14:45.45Dr-Linux|homeManxPower: sorry sir, but your suggestion was?
14:46.05ManxPowerDr-Linux|home: 9:39:52, go read the scrollback
14:46.13*** join/#asterisk ichverstehe (n=harry@67-207-147-205.slicehost.net)
14:46.46ichversteheDial(Zap/g2/12341234||o) doesn't send the original callerid to 1234 1234
14:46.50*** join/#asterisk sakajawebe (n=chazz@nat/digium/x-9e4dc16c9671847f)
14:47.18*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
14:47.41Dr-Linux|homeManxPower: i don't get time in my irc client but:
14:47.45Dr-Linux|home<ManxPower> Dr-Linux|home: it's an all or nothing thing
14:47.56Dr-Linux|homeis that?
14:48.01ManxPowerichverstehe: your carrier lets you set the callerid?
14:48.02ManxPower(9:39:52 AM) ManxPower: Why don't you just record an file and use it as MoH for those calls.
14:48.20ManxPowerI don't know why I get so sick and tired of me making suggestions and people not even READING them.
14:48.21seanbrightDr-Linux|home: i have a patch for you
14:48.49ManxPowerichverstehe: also 12341234 is not a valid callerid
14:48.59ManxPowerat least in the USA, since you didn't specify.
14:49.14ichversteheManxPower: it was an example as i don't want to reveal my id's etc .. and it's denmark.
14:49.18Dr-Linux|homeManxPower: good suggestion, i'm already using that method, but my question was just to understand/learn
14:49.31ManxPowerichverstehe: the more you mask the harder it is to help you.  mask only PASSWORDS
14:49.41[TK]D-Fenderichverstehe: NoOp your callerID in your dialplan and pastebin the complete CLI output of the call with PRI debug enabled.
14:49.42Dr-Linux|homeseanbright: can you share?
14:49.45ichversteheManxPower: hm. might be they don't .. there goes a great deal of using asterisk to redial incoming calls to out of the house cellular .. it's too annoying not to be able to see who's calling
14:49.46[TK]D-Fender~pb
14:49.47jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
14:49.48[TK]D-Fender^^^^^^^^^^^^^^
14:49.54seanbrightDr-Linux|home: no, i'm just going to tease you with it ;)
14:50.24ichverstehe2min
14:50.28Dr-Linux|homeseanbright: okey no problem,
14:50.29ManxPowerFor example if the callerid has a leading 0 then your carrier might reject the call or the callerid
14:50.38seanbrightDr-Linux|home: cd /path/to/1.2.17/source/
14:50.40seanbrightDr-Linux|home: wget -q -O - "http://pastebin.ca/raw/1173004" | patch -p0
14:50.46ManxPowerbut until I see some real data I can't do ore.
14:50.51seanbrightDr-Linux|home: recompile and reinstall and such
14:52.08*** join/#asterisk spokra (n=spokra@host093-179-187.sea0.speakeasy.net)
14:53.10ichverstehe[TK]D-Fender: http://pastie.org/253586
14:53.18Dr-Linux|homeseanbright: you had it already?
14:53.25seanbrightDr-Linux|home: no, i just wrote it
14:53.30seanbrightit's a 2 line change :)
14:53.52Dr-Linux|homeseanbright: great that what i was thinking :)
14:54.37ichverstehe[TK]D-Fender: and that is some other phone calling 96528888 which is then dialing 21437412
14:54.39seanbrightDr-Linux|home: once you apply that and install that, you won't hear the position part.
14:55.01Dr-Linux|homeseanbright: this will work only with 1.2?
14:55.26ichversteheasterisk 1.4.18
14:55.30seanbrightDr-Linux|home: it will probably only work with 1.2.17
14:56.20seanbrightDr-Linux|home: i didn't even test it, so it might not work.  but it should.
14:56.26seanbrightgoes to smoke and such
14:57.11Dr-Linux|homeseanbright: I also want to apply it on Asterisk 1.2.24
14:59.58ManxPowerichverstehe: What info did [TK]D-Fender ask for?
15:00.16[TK]D-Fenderichverstehe: What do they see?
15:00.28ManxPowerah, he's back.
15:00.41[TK]D-Fenderichverstehe: And what happens if you don't use "o"?
15:00.54ManxPower[TK]D-Fender: I suspect his carrier is not allowing custom callerid.
15:01.15ManxPoweror he is adding a leading 0 or 00, but with all that crap he posted it's hard to tell.
15:01.38ichverstehe[TK]D-Fender: the caller id shows as '99400300' which is the base number in the series we've got ..
15:01.42ichverstehesame without 'o'
15:01.48[TK]D-FenderManxPower: No, its actually pretty clear and super minimalist
15:02.12[TK]D-Fenderichverstehe: a thought : Answer the call first, then issue Dial.
15:02.28ManxPower[TK]D-Fender: maybe to a Q.931 guru.
15:02.58[TK]D-Fenderichverstehe: If that fails, it could be that your telco blocks #'s that they have not assigned to you.
15:03.16[TK]D-FenderManxPower: well... it IS exactly waht we need to see :)
15:03.21ManxPowerichverstehe: "(10:01:43 AM) ichverstehe: same without 'o'" contact your carrier then and tell them to allow to send the callerid info.
15:03.42ManxPower[TK]D-Fender: no what you wanted to see.  I could have seen the same thing with a simple Noop.
15:04.32[TK]D-FenderManxPower: but here we should see if it was indeed being passed on, a confirmation that this IS PRI we're looking at, and response accept/deny, etc.
15:04.42[TK]D-FenderManxPower: Don't get all crotchety on us now!
15:04.53[TK]D-FenderManxPower: Fo once they comply quickly and accurately!
15:05.58*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
15:06.07ManxPower[TK]D-Fender: well other than there not being a single Noop in that paste....
15:06.36ManxPowerwhich you did ask for.
15:06.42[TK]D-FenderManxPower: Yeah, I know.... but the PRI debug showed it all anyways...
15:06.55[TK]D-FenderManxPower: Not going to complain...
15:07.03ManxPower[TK]D-Fender: after 5 mins of reading it instead of the 5 seconds for a noop 8-|
15:08.06[TK]D-FenderManxPower: Took you 5 minutes?  Convalescence is upon you!
15:08.19ManxPowerBut you solved his problem, I guess that's what's important.
15:08.29ManxPower[TK]D-Fender: I never need to look at PRI debug.
15:08.59[TK]D-FenderManxPower: well... I've SUGGESTED something... no confirmation of "solved" yet.
15:09.09[TK]D-FenderManxPower: Credit where credit is due
15:09.09*** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb)
15:09.24*** join/#asterisk bad_duck (n=bad_duck@dynamic44-3.MAN-STDOM.cablenet.com.ni)
15:09.27ManxPowerGotta love the mailing list:  My Astra MWI doesn't work unless I use qualifty=yes".
15:09.59*** join/#asterisk scampbell (n=scampbel@199.105.195.156)
15:10.02ManxPower[TK]D-Fender: when the answer is "contact your carrier", as far as I'm concerned that is the end of needing support here.
15:10.54bad_duckHi, I have just installed asterisk on debian linux, and asterisk-gui, the installation seems to be ok but the web interface refresh itself all the time when i log me in
15:11.16ariel_has anyone worked on a load balance for the app_queue.so, Seems asterisk has a hard time with having more then 100 agents to a box.
15:11.17[TK]D-FenderManxPower: That was your answer.  Was more of a "response" than an answer actually since you didn't go through other avenues and have no confirmation.
15:11.50[TK]D-FenderManxPower: and that ML post was understandable (minus the typo)
15:13.49seanbrightDr-Linux|home: you can try using the same patch, you'll get an error if it won't apply cleanly
15:15.54x86http://farm4.static.flickr.com/3283/2762173260_b67720e0d0_o.jpg
15:15.57x86this is the best "toilet humour" i think I've ever seen ;)
15:16.39Iptime( ¨ | ¨ )
15:18.41x86hah
15:18.51x868===D ~~
15:19.45seanbrightwow
15:19.54seanbright_slightly_ OT
15:19.58*** part/#asterisk Iptime (n=chatzill@119.82.102-84.rev.gaoland.net)
15:20.54x86perhaps
15:20.56x86:p
15:21.53*** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org)
15:21.57*** join/#asterisk bsdwarrior (n=bashguar@70.44.84.87.res-cmts.sm.ptd.net)
15:22.50bsdwarrioronce a parked call has timed out it rings back to the ext that parked the call. If they do not answer, it hangs up on them. Is there any way to have the call return to park if this happens (I know I can increase the timeout setting)
15:23.58*** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler)
15:24.30*** join/#asterisk Chotaire (i=chotaire@chotaire.net)
15:26.02Chotairehi guys, I have a question, how can I have a command executed after a user hangs up in a meetme? I don't seem to have an idea how to do continue in an extension when user exits non-zero.
15:26.33Chotairei am sure there is a trick, I am just stuck on this. would be so great if you give me a tip.
15:27.17ManxPowerbsdwarrior: that is all dialplan stuff
15:27.25Chotairehey there manxpower... ;)
15:27.29Chotaireltns
15:27.37ManxPowerhello Chotaire
15:27.41*** join/#asterisk chkngumbo (n=warble@adsl-69-232-228-26.dsl.pltn13.pacbell.net)
15:27.42chkngumbohi
15:28.07bsdwarriormanxpower ok, I have no idea where to even start
15:28.25ManxPowerbsdwarrior: start with the extension the call gets sent to on a timeout.
15:28.47[TK]D-FenderChotaire: "h" <- * Standard extension.
15:28.59Chotaireexten => h,* ?
15:29.13[TK]D-FenderChotaire: exten => h,1,.....
15:29.31Chotairefender: give me a second, I will try it out...
15:29.36Chotairedoes that work with asterisk v1.2?
15:30.06Chotaire(just don't ask, I haven't found time to port all my code to v1.4)
15:30.11[TK]D-FenderChotaire: Yes
15:30.14Chotaireok sec.
15:30.15ManxPowerbsdwarrior: different versions handle parking timeouts differently
15:30.34bsdwarriorthe problem is I want the call to go back to park if the person that parked it doesnt answer.
15:30.36chkngumbousing asterisk, can i carry on multiple phone calls at once through a single phone line from my phone company?
15:30.45Chotairefender: also.. hi fender, long time no see... good to see most old faces still here.
15:30.52ManxPowerbsdwarrior: that can be done too
15:31.12Chotairemark and kapejod still around?
15:31.16bsdwarriormanxpower, I know very little, would appreciate a kick in the right direction
15:32.38ManxPowerbsdwarrior: start with the cli output of a failed call on pastebin
15:33.17Chotairefender: perfect...
15:33.18*** part/#asterisk bad_duck (n=bad_duck@dynamic44-3.MAN-STDOM.cablenet.com.ni)
15:33.44Chotairefender: thanks so much... I was just totally slammed against my head.. I've been coding all night and just had a mental outage... couldn't have been easier.
15:34.20chkngumboi'm confused about whether its such a basic feature of a pbx that nobody's mentioning it or if its an exotic feature that nobody's doing
15:34.30bsdwarriormanxpower, this will be hard, these people are trained to pickup the parks (but yet they bitch about this problem) lol
15:34.41Qwellchkngumbo: No
15:34.47Qwellit's not possible
15:34.52ManxPowerbsdwarrior: I'm sorry, but unless you do what I ask I cannot help you.
15:35.04bsdwarriormanxpower, ill see what I can do
15:35.07ManxPowerI'm waiting for a pastebin of a failed call.
15:35.14bsdwarriorgimme a few
15:35.24heedlychkngumbo: no, one phone line = one call
15:35.32ManxPowerbsdwarrior: If it's too hard to do we don't have to work on this.
15:35.33heedlybut you can put several calls over an interenet link.
15:35.53ManxPowerheedly: unless the line it a T-1/E-1/PRI
15:36.01Qwellbut then it's 23 lines
15:36.07heedlyright
15:36.13*** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net)
15:36.13ManxPowerwell, 23, 23, 30, or 31
15:36.17Qwellwhatever
15:36.21heedlylol
15:36.28chkngumbohow do call centers redirect a single phone number to a large number of seperate conversations?
15:36.29Qwellyou know what I mean
15:36.40Qwellchkngumbo: a phone number is NOT the same as a phone line
15:36.44ManxPowerchkngumbo: they use multiple lines.
15:36.53heedlychkngumbo: line hunting it's called.
15:36.56Qwella single phone number can go to multiple lines
15:36.59Qwellheedly: no it isn't :p
15:37.12Qwell(and a single line can also have multiple numbers)
15:37.19heedlyhow about you two just answer his questions instead of correcting me.
15:37.19bsdwarriormanxpower trying now
15:37.28heedlyI'm sure your both very smart.
15:37.30heedly*you're
15:37.38QwellI did answer his questions.
15:38.09heedly\o/
15:38.22ManxPowerheedly: how about we keep correcting you so the 279 people on this channel don't get wrong information?
15:38.32heedlyManxPower: how about not.
15:38.39heedlysince I've really said nothing wrong.
15:38.46ManxPowerheedly: I guess you could start using the right terms.
15:38.54heedlyor I could just ignore you!
15:39.03ManxPower(10:36:53 AM) heedly: chkngumbo: line hunting it's called.
15:39.27ManxPowerheedly: all people are welcome to /ignore anyone they want.
15:39.31Qwellthis is just silly...
15:39.53chkngumboall this on behalf of poor little me? heavens!
15:40.02ManxPowerBut I'm not going to participate in some poor sod asking for "line hunting " on a PRI because he heard it here.
15:40.06Qwellchkngumbo: no, ask away if you've got more questions :D
15:40.11heedly#asterisk is full of ass hats, so it's not just because of you.
15:41.16Kattymore dots more dots more dots!!!
15:41.39Kattynow stop dots!
15:41.46Qwellmutters at Katty
15:41.56Chotaireok I'll be busy coding extensions, I'll stay tho. good luck with your projects.
15:41.58jeevManxPower, thanks for answering my question yesterday. i had gone afk.
15:42.01Kattyhugs Qwell
15:42.07*** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek)
15:42.09chkngumbomy dream is to be able to have a customer call a single phone number, for a system to ring all the phones in the office (4 of them)  which aren't currently busy, and be able to maintain 1 individual conversation per phone
15:42.11Kattyhai Zeeek!
15:42.27ManxPowerchkngumbo: then you must have more than 1 line.
15:42.39ZeeekPreconference seats available right now at #voip-users-conference and call us http://bit.ly/voip
15:42.42KattyQwell: hit it like you mean it!
15:42.49Zeeek{{{Katty}}
15:42.58KattyQwell: now stop dots!!!
15:43.18ZeeekDOnner, Blitzen,Russell,Qwell,File, MOH pleaseeeeeeese join us now
15:43.24ManxPowerchkngumbo: the best would be to have a PRI line, but it would be too expensive for just 4 calls
15:43.27QwellZeeek: topic?
15:43.35Zeeekoh, that again?
15:43.39KattyQwell: there is no aggro reset :<
15:43.40ZeeekUmmmmmm
15:43.46KattyQwell: except for invisible
15:43.54KattyQwell: can you invisible in raid?
15:44.01Zeeekthe the topic is uh.... it's... tooo sensitive to be mentioned aloud hre
15:44.08QwellKatty: 5 seconds.
15:44.20KattyQwell: disgusting.
15:44.20Zeeekbut there are FREE Astricon passes to be given away, I can mention that
15:44.28KattyQwell: i have a solution.
15:44.31KattyQwell: bring a better tank.
15:46.50*** join/#asterisk jazzmann (n=chatzill@cpc1-lutn9-0-0-cust163.lutn.cable.ntl.com)
15:47.41jazzmannmobile search coming as no suc command in chan_mobile
15:47.45chkngumboa T1 line is a sort of PRI?
15:47.58Qwella PRI is over a T1
15:48.05jazzmannplease help
15:48.40Qwelljazzmann: is it loaded?
15:48.53jazzmannyes
15:48.57chkngumbohow much is a T1 line likely to cost, and how many voice connections would i let me have?
15:49.16jazzmannhttp://www.geek-pages.com/articles/asterisk/howto_build_and_configure_chan_mobile_on_trixbox.html
15:49.17Qwellit varies wildly..  between ~$300 and...~$800
15:49.30ManxPowerchkngumbo: $300 - $800/month, up to 23 calls
15:49.34ManxPowerlooks at Qwell
15:49.38Kattychkngumbo: you can have lots of DIDs tho
15:49.39QwellManxPower: was I actually..right?
15:49.43jazzmannI followed this walkthrough .I do not have tribox it is mandriva linux
15:49.49jeevchkngumbo, 300-800 bux.. about 23- MAYBE 24 calls.
15:49.51KattyDIDs++
15:49.54jeevlol
15:50.09QwellYou can also get partial PRIs - I think the min you can usually get is about...8 channels?
15:50.13jeevrusselb?
15:50.13ManxPowerQwell: no way to know about prices, I was just surprised you estimated the same as me.
15:50.22QwellManxPower: yeah, we must be right then
15:50.30russellb~8ball does jeev have a good reason to be messaging me privately?
15:50.30jbotI'm sure of it.
15:50.30Kattyour company is paying 500 something a month
15:50.33russellbdamn
15:50.37jeevhahahah
15:50.40jeevand you aw, it's a damn good one
15:50.41Qwell~8ball does jeev want his +q yet?
15:50.42jbotAbsolutely.
15:50.51Qwellrussellb: You heard the bot!
15:50.56heedlyKatty: do you get free local calls wiht that?
15:51.10Kattyheedly: i believe so.
15:51.38Qwelljazzmann: how do you know it's loaded?
15:51.39russellbQwell: k!
15:51.40ariel_Prices for PRI and T1 are depended on locations.  I have 12 PRI lines setup and I am paying 3,500 for all of them. But I also have 3 DS3's for data with same provider
15:51.40jeev~8ball if qwell goes to astricon, should i give him a real +q ?
15:51.41jbotAbsolutely.
15:51.56*** mode/#asterisk [+b %jeev!*@*] by russellb
15:52.06Kattyhehehe
15:52.07*** join/#asterisk coppice (n=chatzill@175.202.17.210.dyn.pacific.net.hk)
15:52.30jazzmannmodule load chan_mobile.so when I enter this command asterisk disconnects
15:52.40Zeeekhttp://bit.ly/voip to join us. See you all later. Have a nice, California-like day!
15:52.46*** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek)
15:52.48Qwellso what's the topic ze
15:52.49Qwell...
15:52.51Qwellwhatever
15:53.03heedlyariel_: what do you pay then for usage on the PRI's?
15:53.05*** join/#asterisk astassistant (n=jtknapp@h-72-244-204-146.sttnwaho.dynamic.covad.net)
15:53.35jazzmannNw when I trying to connect it is giving this error.Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
15:53.45jazzmanni.e asterisk
15:54.07ariel_heedly, all inbound traffic is included, I don't use them for outbound. I pay per 100 did's permonth at a rate of $ 3.20 per 100
15:54.24russellbQwell: I'm working on a bribe to get the +q removed
15:54.43Qwellpallet of sugarfree redbull?
15:54.48Qwellsomething along those lines?
15:54.50chkngumboright now i have 2 voice lines, 1 fax line, and 1 dsl line. something like $160/mo. does that sound like a reasonable setup unless i feel like spending a bit more for a nicer system?
15:54.51jazzmannwhen I close the command prompt and open it again asterisk is connecting but again mobile search not working
15:55.12jazzmannat cli command
15:55.19russellbQwell: i told him i'd settle for a single case (24 cans)
15:55.21Qwellchkngumbo: that's pretty expensive, though I'm sure the DSL line is a large part of that
15:55.44chkngumboi'm in a pretty expensive area, so that's probably part of it
15:56.03*** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com)
15:56.24JayTee52which is a better choice with Asterisk for an ATA adapter, Linksys PAP2T or Linksys SPA2102?
15:56.35[TK]D-FenderJayTee52: Clearly the latter
15:56.59JayTee52[TK]D-Fender, then if that's your opinion that's what I'm going to order :-) Thanks!
15:59.24*** mode/#asterisk [-b %jeev!*@*] by russellb
15:59.28jeevgasps for air
15:59.31jazzmannI need to connect my cell to asterisk ,please help me in providing some solution
15:59.59jazzmannit is now a week since trying  to achieve this
16:00.41*** join/#asterisk draygon-w (n=draygon@gateway5-pnap.exigo.com)
16:00.44draygon-wHeyHey
16:00.45draygon-wHey*
16:00.58draygon-wDoes anyone have Allison's phone number? I remember she used to have it on the site but I dont see it there anymore
16:02.19chkngumbocan i make it so that a land line and a few cell phones ring at the same time, except the ones that are already busy?
16:03.04*** join/#asterisk ajohnson (n=ajohnson@63.147.46.186)
16:03.13draygon-wthat's possible
16:03.23draygon-wchkngumbo look up follow me
16:03.25draygon-wi think
16:04.34jazzmannlocalhost*CLI> mobile search
16:04.35jazzmannNo such command 'mobile search' (type 'help' for help)
16:04.54jazzmannlocalhost*CLI> module load chan_mobile.so
16:04.56jazzmannlocalhost*CLI>
16:04.58jazzmannDisconnected from Asterisk server
16:05.09jazzmannthis is the output
16:05.35ManxPowerjazzmann: chan_mobile is crashing your Asterisk
16:06.00jazzmannso what should I do
16:06.07ManxPowerDon't ask me why.  I just know it is based on your paste of the 5 lines
16:06.35jazzmannI mean do you have any suggestions
16:06.39draygon-wManxPower do you know a quick way to get in touch with allison or a number i can reach her at?
16:06.42JayTee52did anyone hear the news that Nortel bought Pingtel?
16:06.51brodiemdraygon-w: theivrvoice.com
16:06.53ManxPowerjazzmann: No.
16:07.06ManxPowerdraygon-w: No.
16:07.40jazzmannanyone on this channel has any suggestion please
16:07.42draygon-wI know she used to have a number on her site
16:07.44draygon-wI guess she took it out
16:08.11*** join/#asterisk moy (n=moy@nat/ibm/x-998d8b2c4d5f3c52)
16:08.19*** join/#asterisk Assimilate (n=Assimila@216.83.78.108)
16:08.47*** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net)
16:08.52jazzmannany forum or other channel who can help me in chan_mobile
16:08.53jazzmannthanks
16:10.05chkngumboit sounds like what i want is "Multi-line hunting"
16:10.50heedlyhehe
16:11.00heedlythey don't call it that anymore I guess.
16:11.08Qwellnot on PRI they don't
16:11.18heedlyit's just channels now?
16:11.26*** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net)
16:11.29heedlyand DID's associated with them?
16:11.33chkngumboi got the term from wikipedia: http://en.wikipedia.org/wiki/Hunting_(telephony)
16:13.16draygon-wanyone here can recommend someone who does good voice overs?
16:13.36x86Allison
16:13.43x86I forget her last name....
16:14.06draygon-wAllison smith
16:14.10draygon-wyes, other than her
16:14.14*** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net)
16:14.18draygon-wShe seems hard to get through with these days
16:14.35x86http://store.digium.com/productview.php?category_id=8&product_code=8IVRPROMPT&main_category_id=8
16:15.19x86draygon-w: there is also June
16:15.31draygon-wahh
16:16.05draygon-wwhats the turn around with these?
16:16.26*** join/#asterisk bad_duck (n=bad_duck@dynamic44-3.MAN-STDOM.cablenet.com.ni)
16:16.32x86not sure, i closed the page
16:16.36*** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir)
16:16.47draygon-wwhats june's last name?
16:16.52draygon-wany idea?
16:17.02[TK]D-FenderWallach
16:17.02x86nope
16:17.17*** join/#asterisk Nate187 (n=Nate187@gw.bigrivertel.net)
16:17.23[TK]D-Fendersomething like that
16:17.24x86Note: Orders, not prompts, are typically processed within 24 hours of receipt, on the next business day. Allow 2-3 days for voice prompts to be created.
16:17.29QwellJune Wallack
16:19.25draygon-wthanks qwell
16:19.32draygon-wanyone in here do voice overs? heh
16:19.41Qwelldraygon-w: sure, I'll do voiceovers
16:19.45*** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com)
16:19.53QwellI offer a 0% quality guarantee
16:19.57draygon-wno thanks
16:19.59draygon-wheh
16:20.01Qwell2%
16:20.08draygon-wgetting there..
16:21.36*** join/#asterisk Firass-z0r (n=Firass@ead224-222.housing.wwu.edu)
16:22.10*** join/#asterisk scampbell (n=scampbel@199.105.195.156)
16:23.29*** join/#asterisk steliosk (n=Stelios@athedsl-285035.home.otenet.gr)
16:27.56*** join/#asterisk ar3dam (n=ar3dam@189.156.217.142)
16:28.22ar3damhello there, how i can see what channel is busy with the cli?
16:28.36ariel_core show channels
16:29.12ar3damthks, ariel_
16:31.00chkngumbodoes anybody know how this might be done: all cell phones in a group of 4 cell phones ring at the same time (unless if they're busy)
16:31.25Qwellchkngumbo: sure, the Dial application in Asterisk can dial multiple devices simultaneously
16:31.35Qwellwhichever device answers, gets the call
16:32.01ManxPowerWell, unless any of the calls go via a FXO port, of course
16:32.14chkngumbowould that require 1 phone line per simultaneous call you want to make to a cell phone?
16:32.23Qwellchkngumbo: yes
16:33.19ManxPowerchkngumbo: You should expect to pay about $700 for all the cards and devices you need
16:36.32seanbrightDr-Linux|home: did it work?
16:36.57Dr-Linux|homeseanbright: i didn't try yet it's production time
16:37.03Dr-Linux|homemany calls bridged
16:37.06seanbrightah
16:37.12seanbrighti say try anyway
16:37.14seanbrightthey'll call back.
16:37.41Dr-Linux|homeok
16:38.30chkngumbodo any phone companies offer the ability to have one number ring a group of numbers at the same time? (whether or not each number is busy might not matter)
16:39.37ManxPowerchkngumbo: maybe you stop asking and start reading The Book
16:39.39ManxPower~book
16:39.40jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
16:39.48ManxPowerthis is not a telecom tutorial channel
16:40.15*** join/#asterisk beek (n=klinebl@65.211.106.242)
16:41.44*** join/#asterisk discHead (n=larry@wsip-70-183-82-162.sd.sd.cox.net)
16:42.17Dr-Linux|homeseanbright: when i can see you here again
16:42.24chkngumboi'm not sure if asterisk is the way to go if a phone company can ring 4 phones at once for cheaper than they'll sell me 5 phone lines so that i can do it myself
16:42.48Qwell~101
16:42.49jbotmethinks 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony.  You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf
16:42.56Qwellchkngumbo: That may help you a little bit as well
16:43.39[TK]D-Fenderchkngumbo: NO.
16:47.02*** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb)
16:48.25*** join/#asterisk Assimilate (n=Assimila@216.83.78.108)
16:50.53jameswf-homeI think i shal start a new blog... I will post on craigslist as an underage girl and post all the replies to said blog...
16:52.38Strom_Myou could be extra clever and call it "The Pedometer"
16:54.00*** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
16:54.00*** mode/#asterisk [+o lmadsen] by ChanServ
16:55.08*** join/#asterisk matt_keys (i=smatt@usaregs.com)
16:55.17*** join/#asterisk luckyaba (n=lucky@ip68-6-98-146.sb.sd.cox.net)
16:57.07chkngumboQwell: that book was very helpful. thank you
16:57.15Qwellchkngumbo: thank Strom
16:57.21chkngumbothank you, strom
16:57.36Stromyou're welcome
16:58.31chkngumbodid you suggest the book or write it or?
16:58.53Qwellhe hosts it and made it available
16:59.19chkngumbocool
16:59.35lowtekWell so much for satisfiying both my inner-geek and inner-child, apparently clone-wars is a bust ...
16:59.47Qwelllowtek: boo
16:59.57luckyabaI just got a DID and want to make sure 5060 is the only port i need to forward on my router to my asterisk box?
16:59.57matt_keysI'm trying to create a MixMonitor macro (named wiretap) so I can monitor by extension when needed. The macro has the line "exten => _X.,1,MixMonitor(${EXTEN}.wav|a)" , and I'm trying to include it in the regular extensions list (i.e, exten => 0154,1,Macro(wiretap,stdexten,0154,sip/0154) )
17:00.11matt_keysWhen I do so and place a test call, the line is busy... what am I doing wrong?
17:00.16*** part/#asterisk korihor (n=korihor@190.78.32.60)
17:03.31matt_keys<-- asterisk newb.
17:03.39luckyabahaha, ditto
17:03.53*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
17:04.08matt_keysanyone?
17:04.40lowtekmatt_keys: pastebin your console output ...
17:04.58matt_keysok just a sec..
17:05.23*** join/#asterisk jayrod422 (n=jayrod42@node2.164.136.64.1dial.com)
17:06.34jayrod422does anyone know how i can read the number from where a call was forwarded from using astiersk?  i thinking about installing a pri card into a * box and use the asterisk to read the ani of the originall caller not the did that the call came in on...
17:06.55Stromjayrod422: lol.
17:07.08Stromyou don't get separate ANI and CPN on a PRI
17:07.54matt_keyslowtek : http://pastebin.com/d6f245cf6
17:09.16lowtekmatt_keys: Ok, now please pastebin your macro ...
17:09.59[TK]D-FenderContext 'macro-wiretap' for macro 'wiretap' lacks 's' extension, priority 1 <- like this doesn't say it all?
17:10.13jeevdamn, nakee pics of swedish olympian leaked, she's pretty good.
17:10.38jameswf-homeokay first ad up :)) lets see if it gets flagged off
17:10.40lowtekjeev: url?
17:11.29matt_keyslowtek: http://pastebin.com/d3b17ebfc
17:11.45lowtekReplace your _X. with s
17:12.23lowtekAlso, just advice, if this is a business, you may want to change the word 'wiretap' to something else .. just in case
17:12.52chkngumbo"QualityAssurance"
17:13.24*** join/#asterisk MrNaz (n=naz@ppp121-44-245-95.lns4.mel4.internode.on.net)
17:13.48matt_keyslowtek: http://pastebin.com/d5301b6ea
17:14.02lowtekI personally don't see anything wrong with the naked olympian pics .. she's worked hard to look that way, they are tasteful, and she is beautiful.
17:15.10Wayhighnaked olympian pics?
17:15.13jeevyea
17:15.19jeevi'm not gonna paste here
17:15.21lowtekmatt_keys: Did you reload?
17:15.23jeevget my ass kicked
17:15.30matt_keyslowtek : yes. something to do with queues?
17:15.33jeevher last pic is nice.. nice boobs and pootang
17:15.51[TK]D-Fendermatt_keys: because your macro has 1 line in it, thats all it does, and then it stops
17:15.54lowteklol, very nice grooming ...
17:16.21matt_keys[TK]D-Fender : ok, so what else should be included to keep it going throug hthe other macros?
17:16.55[TK]D-Fendermatt_keys: what "other macros"?  "s" runs out!  If you wanted to do more go make more priorities
17:17.40matt_keys[TK]D-Fender : I want it to continue on to stdexten|0154|sip/0154
17:18.01matt_keys[TK]D-Fender : I'm assuming that's why it's not ringing?
17:18.02lowtekmatt_keys: Do an s,n,GoTo()
17:18.14Wayhighok.. I got one complaint about the pics..
17:18.24Wayhighthem tan lines are totally blinding..
17:18.25lowtekWayhigh: ??
17:18.28*** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net)
17:18.28jeevlol
17:18.37[TK]D-Fendermatt_keys: Macro does not jsut call a bunch of other macros.  Who said you could next them like that?
17:18.46[TK]D-Fendernest*
17:19.11[TK]D-Fendermatt_keys: You want to call 2 macros, call them back to back.  You can't cram them onto the same line
17:19.17jameswf-homeholy crap NSFW
17:19.39lowtekjeev: I'm sure it's some ugly chicks making a big deal out of it... always is...
17:19.48jeevyea
17:20.40matt_keysok, so the best thing would be to remove the macro all together, then put it above the origional extension list line (exten => 0154,1,Macro(stdexten,0154,sip/0154) )?
17:20.41lowtekYowzers! Did you look at the third section down o nthat page??? I need to go shower ...
17:21.08lowtekmatt_keys: Or just call the mixmonitor first, second do a Goto() to your stdexten macro ...
17:21.10[TK]D-Fendermatt_keys: Of call the other macro from within this one, but that can get messy.
17:21.45jameswf-homehere lizard lizard lizard
17:22.19matt_keys[TK]D-Fender : so how is stdexten,0154,sip/0154 being called all at once?
17:22.27matt_keysor are they arguments?
17:23.05luckyabaanyone know where i can look to get info on mapping my DID to ring to all phones?
17:23.19luckyabagoogle isn't being nice to me at the moment
17:23.33lowtekluckyaba: Dial(SIP/peer&SIP/peer&SIP/peer|options)
17:23.47jeevjameswf-home, not safe? you'd poke that till it was unsafe to move
17:24.16jameswf-homeis married therefore doesnt poke anything
17:24.19lowtekI wouldn't touch that with your d...
17:24.28matt_keyssorry guys, i'm pretty new at this!
17:24.32lowtekI had to empty my browser cache ...
17:24.36lowtekFor fear of disease
17:24.40coppicecome on. olympic athletes are supposed to complete naked. its traditional
17:24.42jeevi iwould hit that.. but i love my girl so i wouldn't poke anything either. she doesn't have diseases!
17:25.05*** part/#asterisk bad_duck (n=bad_duck@dynamic44-3.MAN-STDOM.cablenet.com.ni)
17:25.17jameswf-homehow can you truely reach the olympic spirit with clothes on
17:25.18*** join/#asterisk jayrod422 (n=jayrod42@node2.164.136.64.1dial.com)
17:25.18lowtekjeev: Are we talking about the same pic?  Third section down?
17:25.37jayrod422<PROTECTED>
17:25.40matt_keyslowtek : I've got the Mixmonitor line above the stdexten line, should I remove the s and just have 0154,1,MixMonitor(....
17:25.41jeevoh, no idea who you're talking about
17:25.43lowtekjeev: I'm not talking about the hot olypian..
17:25.49jeevohhhhhh
17:25.51lowtekLook at the third section down
17:26.10jeevi can't look now
17:26.48lowtekThere's the olypian, then some other girl who looks pretty mean, then the third section there's some ragged looking thing ..
17:27.39jameswf-homei must be looking at the wrong site...
17:27.39[TK]D-Fendermatt_keys: stdexten,0154,sip/0154  <- stdexten is the macro, the rest ar arguments
17:27.46lowtekcoppice: agreed
17:28.01matt_keys[TK]D-Fender : thank you, that's what I was confused about
17:28.58coppicegymnasium comes from the greek word meaning naked. the original olympians competed naked. folk are just so screwed up these days
17:29.14jeevjames, are you looking at that swede with the stockings?
17:32.37matt_keys[TK]D-Fender : Ok great it works (and no one liner macros!) but there's a new problem. It only works on inbound and not outbound
17:33.00[TK]D-Fendermatt_keys: Show us that you're USING it on "outbound"
17:33.20[TK]D-Fendermatt_keys: "monitor" works... your DIALPLAN... well that you don't my opinion on ;)
17:33.45matt_keysoic, so it needs to be on the trunk also
17:33.55matt_keyslistening for outbound on 0154
17:34.31[TK]D-Fendermatt_keys: "trunk"?  No, this is DIALPLAN.
17:34.40[TK]D-Fendermatt_keys: And never use that word again.
17:36.03jayrod422anyone have any idea what it is needed to read a pri's q931 debug info for call forwarding info and then gen a sip call with the info prepened on the invite? im trying to build a vmail app using asterisk but cant figure out the call forward part so asterisk knows what phone forwarded the call
17:36.46*** join/#asterisk Strom_M (n=strom@208.127.172.112)
17:37.53*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)
17:37.59luckyabawhen i call my number i don't hear anything. Doesn't ring but it does say its connected
17:38.26*** join/#asterisk mercutioviz (n=chatzill@freeswitch/developer/msc)
17:38.52luckyabaohhh
17:39.03luckyabanow it says the call didn't go through...
17:39.05luckyabalol
17:43.24matt_keys[TK]D-Fender so I'm assuming I need to place the MixMonitor line before the Dial command, but if that's included in the stdexten macro, how can I narrow it down to the extension of my choice?
17:46.49*** join/#asterisk talntid (n=eric@66.208.251.170)
17:47.48[TK]D-Fendermatt_keys: Change your macro so you pass it a parameter that will indicate if it should record or not and just do it all in one macro
17:52.31*** join/#asterisk Entr4nced (n=IMG001@cpe-76-190-141-153.neo.res.rr.com)
17:52.54*** join/#asterisk til_ (n=til@ns5.okg-computer.de)
17:53.57til_hello all
17:54.10til_i am new to asterisk
17:54.19matt_keys[TK]D-Fender : I created another stdexten macro and placed the MixMonitor line before Dial in priority, but it didn't kick it off. Do I need to change something in sip.conf, or am I on the right track?
17:54.23[netman]hi til_
17:54.34til_and i have troubles with the dial plan
17:54.36matt_keys[TK]D-Fender : we're going sip -> pri for outbound
17:55.01til_i managed to activate the sip account of my voip provider
17:55.19til_and i can connect via softphone to asterisk
17:55.38til_but how do i route the calls?
17:55.43[TK]D-Fendermatt_keys: Dialplan is dialplan.  If it didn't kick off, you did it wrong.
17:56.14[TK]D-Fendertil_: Make a peer entry for your ITSP and you'll probably be wanting to register to them as well
17:59.26til_[TK]D-Fender: well sounds so easy ;)
17:59.47til_but i am a complete asterisk noob
18:00.57til_and if im right, i only need a connection between my sip account at my provider and my local sip accounts for the soft phones
18:02.30[TK]D-Fendertil_: there is no connection betwen your softphone & your ITSP
18:03.00[TK]D-Fendertil_: Each connects to * and your DIALPLAN tells * to take a call from A and call out to B.
18:03.13[TK]D-Fendertil_: Time to go read the BOOK
18:03.23[TK]D-Fendertil_: and here's a "sample" for you
18:03.25[TK]D-Fender~jerjerguide
18:03.26jbot[~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * :  http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/
18:03.28[TK]D-Fender~book
18:03.29jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
18:07.27til_[TK]D-Fender: i try my best, but the syntax is very confusing when you look at it for the first time :(
18:07.36*** join/#asterisk lanning (n=lanning@66.151.128.195)
18:08.13[TK]D-Fendertil_: Go try stuff.
18:10.49til_i do, i am really happy, that i can talk between my two pc
18:12.08jeevhmm
18:12.12jeevtil.... same til i know ?
18:12.27mercutiovizare there any fonality users here? I was wondering if there's a discussion forum or IRC channel for things like PBXtra, HUD, etc.  Thoughts?
18:13.33til_jeev: dont think so
18:14.02[TK]D-Fendermercutioviz: I think that if there were you'd have seen it on their respective web-sites
18:14.32mercutiovizI thought so... all I could find was a wiki
18:14.44mercutiovizWell, I'll ask my question here, just in case anyone knows...
18:14.54matt_keys[TK]D-Fender : can you point me to some conditional parameter examples? I'm having a hard time figuring out what to do here
18:15.54seanbright[TK]D-Fender: thought you might appreciate this -> http://www.penny-arcade.com/comic/2008/8/1/
18:17.06ManxPowerI wish the devs would make "cat /path/to/src/asterisk/UPGRADE.TXT" as the last line of "make install".  NOBODY seems to read the damn thing.
18:19.11jayrod422any know how to read the redirecting number in a dial plan for a pri using libpri? so when i a call come in i can assign the redirecting num to a var...
18:19.36[TK]D-Fendermatt_keys: "core show application gotoif".  Go read about expressions and how to use variables on the WIKI
18:20.22[TK]D-Fenderjayrod422: I know its visible in debug, but I don't believe * imports it or give you access to it like you can with SIP headers
18:20.51jayrod422can you buy sip orig with that info in the header?
18:22.06*** join/#asterisk rs^ (n=irc@84-75-214-211.dclient.hispeed.ch)
18:22.12rs^hello there
18:22.37rs^does anybody have an idear how to configure, how long it should ring until voicemail comes up ?
18:23.03matt_keys[TK]D-Fender thanks
18:23.17*** join/#asterisk angom (n=angom@201.170.65.143)
18:23.28[TK]D-Fenderjayrod422: Depends on the ITSP
18:23.51[TK]D-Fenderrs^: extensions.conf.  its YOUR dial statement, go change it
18:25.06rs^[TK]D-Fender: great one second
18:26.17jameswf-homeawesome http://rocketgirl1993.blogspot.com
18:26.56rs^[TK]D-Fender: is it this ? "4:5)"  for 45 seconds ?
18:26.59draygon-wWow.
18:27.08draygon-wJune has a pretty good turn around prices
18:27.13draygon-wtime*
18:27.14*** join/#asterisk Paige_ (n=Paige@208.89.241.31)
18:27.39[TK]D-Fenderrs^: thats 6 chars without any proof that it is your DIAL COMMAND.
18:27.58[TK]D-Fenderrs^: does it say that its calling the DIAL application?
18:28.41rs^hmm where exactly do i configure it ? its saying
18:29.04ManxPowerrs^: you edit extensions.conf, copy the part you have a problem with, copy it up to pastebin.ca
18:29.05rs^exten => asterisk,3,GotoIf .... /voicemail/${PHONE}/vm1) 4:5)
18:29.22rs^may i query one of u ?
18:29.29rs^[TK]D-Fender: or ManxPower  ?
18:29.47[TK]D-Fenderrs^: that is GOTOIF, not DIAL
18:29.47ManxPowerrs^: Only if you have a credit card with a high limit.  I don't do personal consulting for free.
18:30.00ManxPower~manxpower
18:30.01jbotextra, extra, read all about it, manxpower is NOT an employee of Digium.  He is looking for a training/teaching job in networking and/or Asterisk.  Contact: eric@fnords.org
18:30.03talntidanyone want to buy a 3 month old R1T1?
18:30.13jameswf-home~me
18:30.13jbotno u
18:30.21rs^[TK]D-Fender: ok so it has to be DIAL command
18:30.22[TK]D-Fender~i
18:30.23jbot[TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy.
18:30.40jameswf-home~i
18:30.41jbotrumour has it, jameswf-home is a tool
18:30.42[TK]D-Fenderrs^: How many more times should I have to say it?
18:30.47jameswf-homedamn bot
18:30.50rs^[TK]D-Fender: sorry
18:30.53fogo~i
18:31.02jameswf-homedude wtf
18:31.07[TK]D-Fender~ou812
18:31.20rs^I dont find the word "DIAL" in extensions.conf
18:31.28rs^[TK]D-Fender: there is none
18:31.47rs^It's always getting into voicemail after to time ringing [TK]D-Fender
18:31.52talntidHad to replace the ever-defective r1t1 with a sangoma card... anyone had good luck with these r1t1 cards?
18:31.55ManxPowerrs^: I'm starting to think that you did NOT set up this Asterisk server or that you used a GUI to set it up.
18:32.12[TK]D-FenderManxPower: that's what I'm thinking.
18:32.20ManxPower~gui
18:32.21jbothmm... gui is (Graphical User Interface) Any system that uses graphics to represent the functions of a program. All Windows operating systems are GUI. or point-and-drool, double-click, drag-and-drop, ... or at http://pla-netx.com/linebackn/guis/index.html.  Of course Real Programmers use the command line interface.  See cli
18:32.27ManxPowereeek!  That was not what I wanted.
18:32.30ManxPower~zeeek
18:32.31jbotextra, extra, read all about it, zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff."
18:32.35*** join/#asterisk oej (n=olle@ns.webway.se)
18:32.36ManxPowerThat's the one I wanted.
18:32.51rs^ManxPower: [TK]D-Fender Thats right I have only basic knowledge up to now
18:33.16[TK]D-Fenderrs^: What exactly are you using to configure your system?
18:34.24ManxPower~employees
18:35.14ManxPowerrs^: I'm sorry, but you don't even know enough about Asterisk for us to help you.  You might consider taking some time right now to read the asterisk book.
18:35.16ManxPower~book
18:35.16jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
18:36.28jblackso, Sangoma cards have this really neat feature that the Rhino cards lack. They work.
18:36.33rs^[TK]D-Fender: Im using the config files directly
18:36.57talntidjblack: be fair, the r1t1 worked........ for about 2 days.... at a time....
18:37.01ManxPoweremployees
18:37.23rs^ManxPower: so point me at least to where i have to read for solving the problem this would be fine for me too
18:37.35talntidwhat really makes the sangoma cards worth it though, is they actually have documentation.
18:38.05ManxPowertalntid: what in the world made you go with someone other than Digium or Sangoma?
18:38.45ManxPowerrs^: You're going to make me go find the book, find the section and give you a page number, right?
18:38.49jblackjameswf-home in the world.
18:38.57jameswf-homehuh?
18:39.28talntidsupposedly, ManxPower, the rhino r1t1 was a good product.
18:39.41[TK]D-Fenderrs^: only reason not to see a dial command in extensions.conf is A )  devices are configured with users.conf (flaming piece of shit) B ) it is #include from another file.
18:39.51[TK]D-Fenderrs^: or C ) from AEL
18:39.51talntidbuy it seems poorly engineered.. at least, the driver side of it...
18:39.56ManxPower?4:5 means "Jump to priority 4 if true or priority 5 if false (I would have to do "core show application gotoif" to know for sure).  Of course we don't use priority numbers in Asterisk anymore, so who knows even if changing it would work.
18:40.25jameswf-homethinks the r1t1 is good of course I am biased :)
18:40.32jblackor drugged.
18:40.35[TK]D-FenderManxPower: Speak for yourself... explicit line-numbering FTW!
18:40.56ManxPower[TK]D-Fender: you luddite you, move in to the 1.2 world from your 1.0 world!
18:41.09talntidjameswf-home: why? have you had good luck putting LOTS of calls through it? like in a call center enviorment?
18:41.15talntidwhats the trick to making it reliable?
18:41.16[TK]D-FenderManxPower: Ok, I'll be the pot this time, you can be the kettle.
18:41.22ManxPoweror in shorter terms "pot.  kettle.  black." 8-)
18:41.28*** join/#asterisk nighty^ (n=nighty@host217-37-109-17.in-addr.btopenworld.com)
18:41.28*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
18:41.30[TK]D-FenderManxPower: Beat you to it :p
18:42.40talntidI was putting 2k calls per day through it, and it was as unreliable as a windows ME machine.
18:42.46ManxPowertalntid: I'm tempted to try Digium cards again.  They have been totally redesigned at least once, maybe twice since I used them.  However, the last Digium card I used cost me $1,200 of my own money and almost cost me a client.
18:43.18ManxPowerthe $1,200 was for a server that did not generate HDLC Abort errors anytime the disk was used.
18:44.54*** join/#asterisk seanmh (i=seanmh@216.31.101.25)
18:45.11ManxPowerBut I was using 2nd generation cards, not the 4th generation cards they have now
18:47.00jblackDoes digium do the same object file blob that rhino does?
18:47.27ManxPowerjblack: you mean for the kernel module?
18:47.36jblackCorrect.
18:47.37ManxPoweror firmware to upload to the card?
18:47.55ManxPowerNeither Digium nor Sangoma have binary blob kernel modules.
18:48.07QwellI would highly doubt Rhino does either.
18:48.08jblackNope, not firmware. Object files. The bulk of the driver comes as an object file.
18:48.13ManxPowerSangoma has a binary blob for firmware that is field upgraded.
18:48.33jblackNo, no, not firmware. That's a universal thing.
18:48.37ManxPowerI assume Digium does to if they have any firmware upgradable cards.
18:49.29*** join/#asterisk MarkJenks (n=MarkJenk@yosemite.cellcom.com)
18:50.13MarkJenksAnyone out here have knowledge of getting ss7 working w/ dahdi?
18:50.15jblackLet me grab their tallball, now that they have a working ftp site
18:50.56lowtektallball?  Some kind of drink?  Like a 40?
18:51.01tzafrir_laptopMarkJenks, I suspect libss7 will require dahdi (or rather: versions of asterisk that support libss7)
18:51.11jblackum, whoops. I'm sorry. I retract that statement about the rhio.
18:51.37jblackAll I see there now is firmware
18:51.39tzafrir_laptopMarkJenks, I saw your(?) changes in the dahdi page on voip-info
18:51.52tzafrir_laptopTo unload all modules:  /etc/init.d/dahdi stop
18:51.53MarkJenksyes, I have libss7 and dahdi loaded already.     If I put a loopup back on it, I can see it come up with ss7linktest.
18:52.27tzafrir_laptopYou can't tell in advance which modules will be loaded. e.g: which echo canceller module, xpp module, etc.
18:52.30MarkJenksI never saw that out there.  I will take a look at it and remove the wiki if it's the same.
18:52.34*** join/#asterisk bram247 (n=bram@96.28.114.46)
18:52.48*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
18:53.12rs^[TK]D-Fender: what's AEL ?
18:53.21MarkJenkshey, it's about the same!  :)
18:53.25[TK]D-Fenderrs^: Google-able
18:53.41rs^[TK]D-Fender: can you show me or point me to an example conf ?
18:53.56[TK]D-Fenderrs^: http://www.google.ca/search?hl=en&q=asterisk+AEL&btnG=Google+Search&meta=
18:53.56MarkJenksIf we set the PRI to AMI, all I get from the DMS500 is HDLC Abort.    When it'
18:54.02ManxPowerrs^: example conf included in /path/to/src/asterisk/configs
18:54.12jblackwonders how he screwed up thinking the rhino driver was loaded with sourceless object files. there's clearly none there now.
18:54.12*** join/#asterisk Op3r (n=Edwin@ded-139-109.eglobalreach.net)
18:54.23MarkJenkss set for B8, it just doesn't want to come up on the switch side.    But the switch sees that I'm up.
18:54.57Op3rhello, Is it possible to create a caller id on per area code on extensions.conf? does anyone knows any script that does this?>
18:55.00MarkJenksWhat it looks like I need, is a way to change the dchan to 56k and leave the PRI at 64.
18:55.04ManxPowerMarkJenks: SS7 and PRI require cleanchannel, and ESF/B8ZS provides that.  AMI/D4 does not
18:55.19rs^ManxPower: sorry but your no help for me you dont even want to I'm happe that there are not so many people around like you ... and thank you very much [TK]D-Fender for trying to help me !!
18:55.40ManxPowerrs^: Best of luck with that.
18:56.04MarkJenksokay, that explains why AMI isn't working.     But something must be missing..
18:56.08*** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun)
18:56.17MarkJenksshould it be mtp2, dchan, or ?
18:56.49Kobazjblack: i've recently dumped the last of my rhino cards
18:56.52QwellMarkJenks: there is a README in the libss7 source...
18:57.00Qwellmtp2=24, bchan=1-23
18:57.15[TK]D-Fenderrs^: its the 1st link is a silly Google search that I handed you.
18:57.16QwellMarkJenks: take a look at that - it's got a bunch of config options there
18:57.23[TK]D-Fenderrs^: You sho no evidence of trying at all
18:57.25MarkJenksokay, I'll take a quick look.
18:57.34MarkJenksI'll be right back in 5......
18:57.40tzafrir_laptopso it's like pri but with mtp2 instead of dchan/hardhdcl, right?
18:57.46Qwell*shrug*
18:57.52tzafrir_laptopSame on E1-s?
18:58.06Qwellprobably replace whatever the D channel would be, with mtp2
18:58.27Qwellhell, might even work on BRI...who knows
18:58.42Qwellerr
18:58.52*** join/#asterisk balzac (n=chatzill@199.99.96.227)
18:58.53Qwellnot BRI, because SS7 would replace BRI
18:59.03tzafrir_laptopthe HFC bri chips do the hdlc encoding in hardware. mtp2 uses hdlc as well?
18:59.09Qwelltzafrir_laptop: no idea
18:59.36*** join/#asterisk jpastore (n=jpastore@69.65.65.40)
18:59.46coppicemtp2 uses HDLC, but to do it properly you need a couple of features most HDLC channels don't have
19:01.24tzafrir_laptopwell, I figure using ss7 over bri is not such a great idea anyway
19:02.07coppicewhat's wrong with SS7 over BRI? it seems a pretty good idea for a lot of people
19:02.40*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
19:03.09jeevif you use a company like callcentric and some other providers with unlimited calling.. what do they usually provide? 2 outgoing calls at once max?
19:03.10*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
19:04.12Op3rjeev, really depend on how many channels they will open up for you
19:04.18jeevhmm
19:04.24ManxPowerjeev: there are no flat monthly rate services or providers, their "unlimited" always has limits
19:04.49jeevyea, i know manx.. im thinking hrmf.. what's the best way to make sure that if those two channels (assuming they give 2), if they're being used, to use another ITSP?
19:05.32[TK]D-Fenderwas wondering what that burning smell was
19:06.09jeevit's your feet, i set them on fire.
19:06.35km-you can limit it on your side I think
19:06.43km-set your account to call-limit=2
19:06.56km-and then attempt to dial out on it and then if it fails you can make extension 101 or whatever be your next itsp
19:07.01Op3ror just use congestion goto another provider
19:08.07rs^[TK]D-Fender: thats the only dial I have -> exten => _22X.,1,Dial(IAX2/blah/${EXTEN},,T)
19:08.22km-ah that reminds me
19:08.27[TK]D-Fenderrs^: Funny last time you said you didn't have a dial.
19:08.40km-I want to write a transfer-to-blackhole thing for my wife for these damned 800 number callers
19:08.51[TK]D-Fenderrs^: perhaps you should pastebin an entire call.
19:08.52rs^[TK]D-Fender: I said i have no DIAL but a Dial ...
19:08.53[TK]D-Fender~pb
19:08.54jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
19:08.56[TK]D-Fender^^^^^^^^^^^^^^^^
19:09.09jeevthey only speak of incoming channels here: http://www.callcentric.com/faq.php?s_go=1&search=channel&go=Search#172
19:09.11[TK]D-Fenderrs^: Don't play capitalization games
19:09.38rs^[TK]D-Fender: sorry I dont play games I mean it like that ...
19:09.54rs^[TK]D-Fender: thast was meant seriously
19:10.25[TK]D-Fenderrs^: pastebin the complete call attempt
19:11.37*** join/#asterisk Gershwin (n=fake@63.250.233.162)
19:12.49km-d-fender: didn't yaknow?  Clinton had "Relations" with monica, but not, RELATIONS.
19:13.03MarkJenksSorry about that, had to go talk to the electrician for a project.....
19:13.42tzafrir_laptopkm-, dialplan apps are case insensetive
19:14.22tzafrir_laptop(as opposed to dialplan functions, whose names are case sensitive and are always CAPITAL)
19:14.23ManxPowertzafrir_laptop: But FUNCTIONS are not.
19:14.23km-tzafrir: I think that's bound to rs^, I was being snarky ;)
19:14.40km-haha
19:14.46km-so, does nufone still exist?
19:14.52*** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com)
19:15.05[TK]D-Fendertzafrir_laptop: personally I find that a complete waste.  Why make functions case sensitive?  Stupid parser....
19:15.27seanbrightmaybe for BC
19:15.46seanbrightless likely to collide with existing variable names
19:15.47MarkJenksI just looked at README for dahdi, and it looks like I'm all set.    I am trying to to it outside of asterisk for testing, using just the ss7linktest.   Should I just do it inside of asterisk with chan_dandi?
19:19.36*** join/#asterisk n3hxs (n=HAMming@151.196.87.132)
19:20.08MarkJenksThe guys at the switch say that the pri has to be b8..  but the ss7 mtp2 has to be 56
19:20.22MarkJenksis there a place to set it up like that?
19:24.02*** join/#asterisk MrNaz (n=naz@122.110.231.8)
19:27.13*** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus)
19:29.46*** join/#asterisk nny_2 (n=Scott_My@64.203.244.146)
19:30.33nny_2Anyone who has experience with building in call center requirements to an asterisk system lemme know. May have some work coming up, and I have only set up asterisk for soho
19:32.10Op3rHello, here's a question. For example I have 100 did onmy disposal that I want to be used as a Caller ID, then I want it to reflect when I dial a certain area code for example 212, I will also show that my caller ID starts with 212 too. Is there an easy way to do it?
19:33.05[TK]D-FenderOp3r: its your dialplan, do whatever you want with it
19:34.31Op3r[TK]D-Fender, yeah I know but its 100 Caller ID numbers I want to reflect. Is there any easy way to just implement it than having to type exten => 91212NXXXXXXXX,1,SetCallerID per caller ID and area code?
19:34.35*** join/#asterisk lanning (n=lanning@66.151.128.195)
19:35.02[TK]D-FenderOp3r: Base your CID on the # dialed.
19:36.04Op3r[TK]D-Fender, any idea on how? I have 0 clue when it comes to creating macros on asterisk :(
19:36.52*** join/#asterisk Jacco (n=root@unaffiliated/jacco)
19:36.53JaccoHey guys.
19:36.55[TK]D-FenderOp3r: I never said anything about "macros".  And you'd better learn to master the dialplan, but that's 95% of Asterisk
19:37.07JaccoI have two zap lines. One of them will show the caller's number but the other won't. :(
19:37.09[TK]D-FenderOp3r: Go read the WIKI page on variables, and expressions.
19:37.15JaccoHow do I figure out what's wrong?
19:37.52Op3r~wiki
19:37.56[TK]D-FenderJacco: Physically verify with a separate analog phone that both lines DO have CID functional
19:37.58[TK]D-Fender~wikis
19:37.59jbot[~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.  http://www.voip-info.org (c) Arte Marketing Inc / CommPartners
19:38.01[TK]D-Fender^^^^^^
19:38.22[TK]D-FenderJacco: Then if that checks out, pastebin your zapata.conf.
19:38.24[TK]D-Fender~pb
19:38.25jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
19:38.27[TK]D-Fender^^^^^^^^^^
19:40.17nny_2I am working on setting up a quote for an * system with advanced call queue features, for a call center. Their old system was just* a phone system, so I don't need to engineer the complete solution, and at first only want to offer them the basics. (no experimental bs)
19:40.17Jacco[TK]D-Fender: so... I get an analog phone with callerid and call both lines?
19:40.19JaccoHokay, thanks.
19:40.25JaccoI'll go test that.
19:40.33*** join/#asterisk korihor (n=korihor@201.211.168.130)
19:40.38nny_2They only need 30 phones fwiw
19:40.43Jaccoalthough.. oh crap. The phone server might collapse if I disconnect the lines. Probably not though.
19:41.05[TK]D-Fendernny_2: Clarify "advanced"
19:41.15[TK]D-FenderJacco: nope.
19:41.51mgromanvimperator
19:44.52nny_2[TK]D-Fender: hmm comparing this to their old system, so advanced would be queue control with round robin, etc strategies as options. I am still researching what other options would be good to offer them, vs here's everything available.
19:45.10nny_2[TK]D-Fender: agent log in / out etc, (afaik this is all native to asterisk)
19:45.52[TK]D-Fendernny_2: thats actually all basic stuff so far
19:46.50nny_2[TK]D-Fender: yeah i need to go poke around and figure out what features a small call center would expect
19:48.28*** join/#asterisk blq (n=Bl@dslb-088-066-229-038.pools.arcor-ip.net)
19:48.39*** join/#asterisk korihor (n=korihor@201.211.168.130)
19:49.14*** join/#asterisk heedly (n=heedly@purplehaze.lamedomain.net)
19:50.09*** join/#asterisk VaNNi (n=VaNNi___@38.98.61.143)
19:50.51obnauticus[TK]D-Fender: know of any projected release date of 1.6 stable?
19:50.53obnauticusor 1.6rls
19:51.39*** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola)
19:53.41*** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net)
19:54.09jeevwill the passthru on a polycom 330 also distribute PoE? i REALLY need it to!
19:55.39JayTee52jeev, NO
19:56.45*** join/#asterisk washburnello (n=johnsy@h23.238.30.71.dynamic.ip.windstream.net)
19:57.02[TK]D-Fenderobnauticus: "when its ready"
19:57.20obnauticusWho said that?
19:58.11[TK]D-Fenderobnauticus: that the stock answer for anyone crazy enough to ask for dealines on an open-source project.
19:58.26obnauticus[TK]D-Fender: I know there are no deadlines.
19:58.28obnauticuslol
19:58.44obnauticusI'm just wondering if you may have some idea...well maybe with bug reports and stuff it's hard to predict
19:58.44obnauticusnm
19:58.47obnauticusI'm just being retarded
19:58.57[TK]D-Fenderobnauticus: I'm guessing 2008, but don't quote me on it.
19:59.06JayTee52if you want a reliable release date on open source software try Ubuntu. The release date is reliable but usually the software isn't.
19:59.20obnauticusOh god
19:59.22bram2471.6.0-beta9 works pretty well for me ;)
19:59.25obnauticusI was just makign fun of ubuntu
19:59.27bram247as is
19:59.30obnauticusI made this quote:
19:59.32obnauticus`Make this channel more sexy. `Trying to gain Linux experience on Ubuntu is like trying to gain BSD experience on a Mac.`
19:59.39obnauticusoh shi-
19:59.45obnauticuswell... it did make this channel more sexy.
19:59.54JayTee52I'm running Gutsy which works very well for me but I've been really disappointed with Hardy on the two machines I've tested so far.
20:00.01obnauticuseh
20:00.04obnauticusI'm running Slackware 12.1 :\
20:01.12*** join/#asterisk deeperror (n=deeperro@76.226.177.255)
20:02.02jeevcrap JayTee52.
20:02.12JayTee52I got really spoiled by Ubuntu but I'm now using RHEL 5 64 bit for production * and I much prefer that to running * on 'buntu. If I was going to choose a different distro for production other than RH or one of it's offshoots I'd stick with Debian itself.
20:02.35JayTee52jeev, they have pills for that now. Hope everything comes out alright
20:02.48*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
20:02.50jeevobnauticus, you can't tell people your own quote
20:02.58obnauticusjeev: how do you know it's mine.
20:03.01jeevJayTee52, i've actually been having pooping problems.. but oh well
20:03.06obnauticusI can quote myself
20:03.07jeevcause you said i made this quote.
20:03.14JayTee52jeev, TMI
20:03.17obnauticusmade/found
20:03.18obnauticuswhatever.
20:03.29jeevyou have less credibility than the average gardner
20:03.34obnauticusI'm quoting myself from another channel.
20:03.36obnauticusrofl.
20:03.56jeevJayTee52, know any unlimited outgoing providers with lots of channels ?
20:04.22jeev~itsp-us
20:04.23obnauticusThere are none :\
20:04.26obnauticusheh
20:04.27jeev~us-itsp
20:04.28JayTee52jeev, told you before I don't do ITSP's
20:04.28obnauticuswouldn't they loose money?
20:04.30jeev~itsp
20:04.31jbot[~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs
20:04.33[TK]D-Fenderjeev: "Would you like fries with that, sir?"
20:04.33jeevbastards
20:04.39JayTee52PRI to PSTN FTW!!!!
20:04.42jeev~itsplist-us
20:04.42jbot[~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net
20:04.48obnauticusya I have a PRI
20:04.51obnauticushehehehheh
20:04.52jeevfender, dood, seriously. go to astricon, i will send you a shemale hooker
20:05.09obnauticusjeev: i think if I goto astricon Corydon76-dig will molest me
20:05.16JayTee52jeev, how old are  you? like 14 or something?
20:05.26bkrusejeev: you kind find them all over the forums :/
20:05.27jeevlol
20:05.33jeevthanks bkruse
20:05.37jeevbkruse, you work at digium?
20:05.42bkruseNo.
20:05.43obnauticusROFL
20:06.20jeevwhat's so funny
20:06.28obnauticusLFOR
20:06.46obnauticusRicers racing on my street just got in a big car accident
20:06.51obnauticusfail.
20:07.04bkrusewonders if it's collin
20:07.40jeevwhat the hell kind of street do you live on
20:07.43JayTee52"Dude! watch my Tokyo drift!!! Oooops!!!! [CRASH!!!!] Waaahhh, waaaahhhhh, MOMMY!!!!"
20:07.55obnauticusjeev: im in a neighborhood, but close to the outside jeev.
20:07.57bkruseraces on the track
20:08.05obnauticusso there is a like 4 lane street outside my neighborhood
20:08.06jeevwhere do you live
20:08.07obnauticusthat people alwas race on
20:08.09jeevvegas?
20:08.12obnauticusSW Washington.
20:08.19jeevah
20:08.29JayTee52the state or the district
20:08.35obnauticusit's like a 2.5mi long semiflat road
20:08.39jeevi was in the district 2 weeks ago.. was cool
20:08.48jeevhumid like a mofo
20:08.49obnauticusWashington...state.
20:08.58JayTee52near Camas or Vancouver?
20:09.05obnauticusIN vancouver.
20:09.16JayTee52used to go there alot on business.
20:09.22obnauticusheh
20:09.27obnauticusTo centrytel?
20:09.33JayTee52got to stay at the Residence Inn on the company dollar
20:09.35obnauticusI would guess because you are in #asterisk-dev
20:09.39obnauticusasterisk***
20:09.47Corydon76-digobnauticus: I already told you you aren't getting into my pants
20:09.53obnauticusCorydon76-dig: ...
20:09.59JayTee52no, this was back when I worked for US Cellular
20:10.02obnauticusCorydon76-dig: I told you, i wasn't expecting you to not put up a fight.
20:10.14obnauticusJayTee52: ya this area is pretty nice
20:10.16jeevJayTee52, residence innn sounds like it's 10 bux a night
20:10.28obnauticusjeev: the operative word is `dollar`
20:10.30obnauticuscompany DOLLAR
20:10.34obnauticus$1 a night.
20:10.57JayTee52nope, it's Marriot upscale. It's like getting an apartment. Comes with living room, kitchen and they'll even shop for groceries for you while you're working.
20:11.01obnauticusEh, the residence inn by PDX is pretty nice, if that's where you stayed.
20:11.10jeevahh
20:11.18*** join/#asterisk pascal_alm (n=pascal_a@static-66-11-93-129.ptr.terago.net)
20:11.19jeevthought you said their PBX is pretty nice
20:11.19bkrusejeev: I am waiting on my redbull
20:11.24jeevit's on the way
20:11.25JayTee52It's just down the street from Tony Roma's
20:11.26jeevassmunch
20:11.32obnauticusPDX is an airport..
20:11.33jeevyou guys are extorting me
20:11.42bkrusenotices the star next to his name, which has the power to ban.
20:11.52obnauticusJayTee52: If you are firmilliar with the salmon creek area. that's whre I am
20:11.53obnauticus:\
20:11.57pascal_almis there known problem for asterisk voicemail accepting dmtf from blackberry?
20:12.00obnauticusCorydon76-dig: is probably gmapsing it right now.
20:12.03jeevi dont see a star, i see an at sign though and i know the capabilities.. so i'll shut my mouth!
20:12.04JayTee52been through it.
20:12.10Corydon76-digobnauticus: do what?
20:12.14obnauticusrofl.
20:12.24jeevbkruse, i told russell that it may come with some anthrax.
20:12.25obnauticuspascal_alm: not really.
20:12.39obnauticuspascal_alm: rfc2833 is pretty darn clear
20:12.39bkrusejeev: I will personally, kill you.
20:12.46jeevhaha
20:12.53russellbjeev: I have that logged for reference by the FBI as needed
20:12.57JayTee52there's a nice jazz club in downtown PDX called Jazz Uno (at least that's what I think it's called, been 8 years) that had good food too.
20:12.57jeevhahahah
20:13.03jeevbastages
20:13.16obnauticusJayTee52: that's where you pick up your bitches when on business eh?
20:13.16pascal_almoh... blackberry seems to go through my app ok, but once it gets inside voicemail, app no longer recognizes key presses.
20:13.35russellbjeev: I'll make you a deal.  I'll get you a copy of this _awesome_ telephony toolkit for _FREE_ if you hook me up with a case of sugarfree redbull
20:13.45jeevhaha
20:13.48JayTee52obnauticus, no but I did see alot of hookers in that neighborhood. think it's right on the outskirts of Portland's Chinatown
20:14.00bkruseheck, you know what russellb, I'll even throw in zaptel
20:14.05obnauticusJayTee52: probably, Portland is pretty fscked up.
20:14.06russellboh snap
20:14.07bkruseout of _my_ pocket
20:14.12bkruserussellb: not that though :X
20:14.19jeevrussellb, i'll go hijack a truck containing pallets of red bull and bring it to digium
20:14.20bkruselol
20:14.27jeevinfact, i'll drive it straight through the doors.
20:14.33bkrusek
20:14.41*** join/#asterisk Assimilate (n=Assimila@216.83.78.108)
20:14.42JayTee52obnauticus, I probably wouldn't be picking up hookers even if I could afford them since I just found out from jeev the other day that I'm gay because I rollerblade.
20:14.47obnauticusSpeaking of redbull I went to the portland Flugtag.
20:14.55jeevyou guys will probably be so messed up on redbull, wont even realize what happened
20:14.58obnauticusROFL you rollerblade>?!
20:14.59JayTee52all this time I thought I preferred women
20:15.06obnauticusya me too.
20:15.07jeevhahahaha
20:15.09russellbrollerblading is fun
20:15.10obnauticusWhy do you rollerblade?
20:15.16russellbi haven't done it in years ...
20:15.16JayTee52low impact
20:15.17jeevwhere is that link that fender sent!!!
20:15.20JayTee52I can't jog
20:15.25jeev"the hardest thing about rollerblading is telling your parents you're gay"
20:15.32JayTee52almost lost a leg once in an accident
20:15.33obnauticusROFL
20:16.01russellb#asterisk is off the hook right now
20:16.03*** join/#asterisk GhOnDiE (n=ghondie@dsl78-143-210-38.in-addr.fast.co.uk)
20:16.04bkruseok losing a leg in a motorcycle accident, tight, in a rollerblade accident, lame
20:16.08obnauticusJayTee52: how do you rollerplade with one leg?
20:16.15jeevhahahah
20:16.18obnauticusrollerplade?
20:16.20obnauticuswhat the hell
20:16.22obnauticushow;'d i even do that
20:16.23JayTee52It's a good workout and I really like it because it reminds me of ice skating. I used to ice skate and play hockey all the time growing up.
20:16.35obnauticusDo you wear hot shorts too?
20:16.43obnauticuswith large headphones
20:16.44obnauticusin miami
20:16.50JayTee52obnauticus, I still have both legs. I just don't jog because of the constant impact bothers my right leg and knee.
20:16.59russellbJayTee52: they're all jealous
20:17.03obnauticusJayTee52: WD40
20:17.16russellbtoo much time on their asses at the compter, rollerblading is too much physical exertion :-p
20:17.19Corydon76-digI don't jog, because it's nauseatingly hot outside
20:17.26JayTee52I wear earbuds with a SanDisk R250 MP3 Player and I wear cargo shorts. I have very sexy legs.
20:17.33jeevlol
20:17.34JayTee52except for the scar
20:17.35jeevew.
20:17.36russellbi don't jog because it's boring as hell
20:17.43jeevrunning is gay too
20:17.45JayTee52it's bad for your ankles
20:17.49jeevi can't even breathe anymore, i've broken my nose a lot.
20:17.59jeevi just play basketball and i'm not in any tournaments right now
20:18.02russellbjeev: huh, i wonder why
20:18.10jeevactually, all sports.
20:18.18JayTee52jeev, everything is gay to you because........well.....because you're from california and you're in denial.
20:18.20jeevi've taken elbows and basketballs in my face
20:18.24obnauticusI jog because usually when I jog ridiculous amounts of wemon come out of their houses wanting to get with me
20:18.40russellb"I've taken ... balls in my face". --jeev
20:18.44obnauticusROFL
20:19.32obnauticusthe same thing happens to JayTee52 when he rollerblades... except with....men
20:20.04*** part/#asterisk deeperror (n=deeperro@76.226.177.255)
20:20.51JayTee52ok, joke if you will but at 40 I was living with a 21 year old hottie and that lasted 3 1/2 years before I dumped her.
20:21.00obnauticus...
20:21.04obnauticusyou were 19 when she was born
20:21.09obnauticusthat makes you almost a pedophile.
20:21.21JayTee52almost, but still within the legal boundary
20:21.25obnauticus...
20:21.30obnauticusthat is disturbing.
20:21.49obnauticusI am still living with my hand
20:21.54obnauticusIf that's what you wanna call it
20:21.57JayTee52no, she was an adult. hell in Arkansas she'd have been considered a spinster
20:21.58jeevha ha ha
20:22.23Corydon76-digobnauticus: not nearly as disturbing as people who sleep together who are wide enough in age to be grandfather/grandson
20:22.27obnauticusJayTee52: I bet thta's what you said about all the other 13yr. olds
20:22.31obnauticusno they were adults!
20:22.47bkrusewhat does jeev like about 28 year olds?
20:22.53bkrusethose 20 of them!
20:23.02bkrusethere's * dangit I messed that up.
20:23.02jeevwhatcha mean
20:23.13[TK]D-Fenderlol
20:23.19*** join/#asterisk blebleble (i=godie@caesar.godie.net)
20:23.28blebleblei'm looking to have in our ivr ask the user for their account # and if they have it have it intergrate with some type of screen pop to a csr's desk when they take the call, has anyone done something like this or can point me in the correct direction
20:24.07[TK]D-Fenderblebleble: "core show application read" , "core show application dial"
20:24.18[TK]D-Fenderblebleble: "core show application system"
20:24.54bleblebleTK: thank you, ill start reading
20:25.04[TK]D-Fenderok, checkout time.  heading home.
20:25.30obnauticusbkruse: that is a verbal pun
20:25.46obnauticusit doesn't work when you type 28 and therefore is clearly a differetn number than 20 & 8
20:26.46jeevi'm lost
20:26.56obnauticusjeev: in reality
20:26.58obnauticuswhen someone says
20:27.11obnauticusWhat does jeev  like about `twenty-eight` year-olds
20:27.15obnauticusand you say there is 20 of them
20:27.23obnauticusit changes the last statment so that it means
20:27.32obnauticusWhat does jeev like about twenty, eight year olds
20:27.40jeevoh
20:27.42jeevlameeeeeeeeeeeeeeeeeee
20:27.44jeevl4m3
20:27.46bkruseobnauticus: Right, but I thought since most people knew the joke, they would get it.
20:27.47obnauticusdoesn't work on IRC.
20:27.57obnauticusbkruse: canadians dont joke about that sort of thing
20:27.59obnauticusthey actually do it.
20:28.02bkrusejeev: the fact the you spelled lame "l4m3" says a lot.
20:28.05obnauticusexplains jeev.
20:28.11jeevhaha
20:28.17jeevi can't believe i did that. sorry
20:28.29jeevi dont send text messages with 'u' and stuff
20:28.40Corydon76-digJust wait until you get old and crothety.  You'll understand.
20:34.31*** join/#asterisk deeperror (n=deeperro@76.226.177.255)
20:36.53mgromanjeev: how old are you?
20:37.08jeevi'm not sure, like the chinese olympians
20:37.40*** part/#asterisk nny_2 (n=Scott_My@64.203.244.146)
20:39.14russellbthe chinese woman gymnasts looked like they were 9 years old
20:40.00jeevseriously
20:40.20mgromanguys, there might be some java programmers here that are offended by that
20:40.32jeevheh
20:44.41*** part/#asterisk mgroman (n=miles@adsl-68-252-62-180.dsl.wotnoh.ameritech.net)
20:45.10*** join/#asterisk nighty^ (n=nighty@host217-37-109-17.in-addr.btopenworld.com)
20:50.12*** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net)
20:52.28*** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194)
20:56.27exothermcWhere do I start to debug when my music on hold starts to breakup
20:57.22[TK]D-Fenderexothermc: What source?  Where is the caller?
20:57.49exothermc[TK]D-Fender: Just called into an empty conf bridge (99.999% sure it isn't network related)
20:57.54exothermcso on net
20:58.00[TK]D-Fenderexothermc: DETAILS DAMMIT
20:58.18exothermcSIP to SIP g.711u
20:59.35*** join/#asterisk alancio (n=Alancio@10.Red-80-38-197.staticIP.rima-tde.net)
21:01.03kfifeSpeaking of MOH, is there a way to have the caller be placed RANDOMLY into a point within one of the music files in the MOH class?  If we have 3 professional recordings in the class, a regular customer always hears one of three musicified sales pitches, and never hears the rest.  Ideas?
21:01.27kfife...hears the BEGINNING and never the END
21:01.29*** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com)
21:01.41EmleyMoorHas support for rotary phones on FXS ports been dropped?
21:02.01*** part/#asterisk deeperror (n=deeperro@76.226.177.255)
21:02.18kfifeRotary as in "turn the magneto crank?" :-) Just kidding
21:02.28*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
21:02.48Strom_MEmleyMoor: not AFAIL
21:02.50Strom_Mer, AFAIK
21:02.53EmleyMoorRotary as in loop disconnect signalling by rotary dial
21:02.58EmleyMoorMy dial no longer works
21:03.05Strom_MEmleyMoor: what kind of card?
21:03.11[TK]D-FenderStrom_M: looks like a "fail" to me :)
21:03.15EmleyMoorTDM400P
21:03.19Strom_M[TK]D-Fender: I'm captain fail
21:03.32Strom_MEmleyMoor: you may need to tweak the debounce setting in wctdm.c
21:03.37[TK]D-FenderStrom_M: Aye aye cap'n!
21:03.56bkruseStrom_M knows all
21:04.02EmleyMoorStrom_M: Tweaked it. Dial is now "dead", rather than going wrong
21:04.06bkrusealong with [TK]D-Fender, of course
21:06.35EmleyMoormakes sure the tweaked version is installed
21:08.19jeevi wonder who changes extensions and sip files at digium
21:08.21jeevdo people fight over it ?
21:09.15bkruseno.
21:09.15jeevahh
21:09.15russellbwe used to have pretty open access to it, heh
21:09.15russellbi had root on the pbx for a long time ...
21:09.17russellbbut not anymore :(
21:09.40EmleyMoorHmmm... dial is now going "wrong" rather than not doing anything
21:09.42russellbIT manages it.
21:10.17EmleyMoorTone dial still works#
21:10.21jeevahaha
21:10.27jeevdoes it talk shit or what
21:10.28jeevIT
21:10.29bkruserussellb: ohrly? I thought you still had root
21:10.38EmleyMoorwill look into this one further
21:10.38russellbnegative
21:10.52russellbbkruse: our PBX is switchvox now, d00d
21:10.59russellbweb admin FTW
21:11.03bkruserussellb: Oh right, nvm
21:11.34[TK]D-FenderEEK
21:12.00russellb[TK]D-Fender: :-p
21:12.35[TK]D-FenderThat's like commissioning Picasso to draw stick-people in a "colour by numbers" book.
21:12.47russellbblinks
21:13.44*** join/#asterisk lzhang (n=lzhang@rrcs-67-78-33-170.sw.biz.rr.com)
21:13.59*** join/#asterisk Zemmad (n=calibisi@cpe-66-69-164-90.sw.res.rr.com)
21:14.05EmleyMoorI've also lost caller ID on my FXO line
21:14.23ZemmadI'm in need of a solving a AGI problem that i am having
21:14.52lzhangquestion: I have 2 pri's from my provider, and they have been trunked together. Does this mean I need to make 1 channelgroup instead of 2?
21:16.16*** join/#asterisk washburnello (n=johnsy@h23.238.30.71.dynamic.ip.windstream.net)
21:18.12*** part/#asterisk washburnello (n=johnsy@h23.238.30.71.dynamic.ip.windstream.net)
21:18.28*** join/#asterisk Entr4nced (n=IMG001@h160.145.31.71.dynamic.ip.windstream.net)
21:18.58*** join/#asterisk wonderworld (n=ww@ip-62-143-216-14.hsi.ish.de)
21:19.29[TK]D-Fenderlzhang: Clarify your definition of "trunked
21:20.15EmleyMoorI don't seem to be able to call festival any more
21:22.10EmleyMoorAny clues on that?
21:22.38lzhang[TK]D-Fender: apparanted they are merged in some way, so they can roll over to each other
21:23.17[TK]D-Fenderlzhang: "in some way".  Doesn't say much.  You can have roll-over on otherwise separate PRI's.
21:23.24EmleyMoorJust done a Debian upgrade and it's now failinng to configure zaptel - what should I do?
21:25.19EmleyMoorCaller ID is coming in, but not being processed correctly, so that's OK
21:25.37EmleyMoorWhat's gone wrong with festival, though?
21:27.27*** join/#asterisk justmehere (n=justmehe@24-176-158-178.dhcp.kgpt.tn.charter.com)
21:28.19*** part/#asterisk justmehere (n=justmehe@24-176-158-178.dhcp.kgpt.tn.charter.com)
21:28.29lzhang[TK]D-Fender: the actual problem I am having is that some outbound calls are hanging (dead air)... then I end up seeing this in the error log chan_zap.c: Ring requested on channel 0/10 already in use on span 2. Hanging up owner.
21:28.52lzhangthis only started happening after this morning, when my provider came in and added a second pri
21:31.10lzhangwhen I check the active channels while one of these problem calls are going on, I see Local Extension -> (the outbound number) and then a call in from (the outbound number) -> internal
21:31.39lzhangthis leads me to believe that there is some sort of mixup when it's trying to rollover, or that I've misconfigured my channel groups somehow
21:32.19*** join/#asterisk barakuda (n=baraka@78.158.192.203)
21:35.18*** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir)
21:38.09jeeva
21:46.08jeevdamn, cdrtool is a little headache isn't it. i have to start that tonight
21:49.05jeevhttp://www.gamma.ru/~avk/ anyone use that?
21:49.07jeev~cdr
21:49.08jbotmethinks cdr is Call Detail Record, a log of what happens to the call at each step through its traversal of the PBX, details like from, to, time, duration, number dialled etc, useful for billing also - it could also be Compact Disc Recordable, see cdrw
21:51.07*** join/#asterisk Entr4nced (n=IMG001@h160.145.31.71.dynamic.ip.windstream.net)
21:51.25TJNIIHey, according to Moscow the US started the war in Georgia!  Go USA!
21:52.53*** join/#asterisk duncan_16 (n=richard@pool-141-158-248-35.phil.east.verizon.net)
21:53.36drmessanoI live in Georgia, and I have seen no war
21:53.40duncan_16hi all.. was wondering if you could assist me with a quick query.  I'm trying to install an RPM into the asteriskNOW system, and cannot... Any suggestions?  It was installed from the ISO that we downloaded at the asteriskNOW site.
21:53.45drmessanoI'm calling it Shenanigans
21:53.50errrlol
21:54.03jeevi believe russia TJNII.
21:54.07drmessano"unable to reproduce"
21:54.28jeevmaybe russia should've warned the US not to invade iraq
21:54.55drmessano"user reported war in georgia"
21:55.04duncan_16to be more specific, we are trying to get the HUDLite server running on the asteriskNOW system.
21:55.04drmessano"went outside, observed no war."
21:55.12drmessanoTicket Closed: Unable to reproduce
21:55.37drmessanoHudlite sucks
21:55.40drmessanoUse astassistant
21:55.56duncan_16ok.  We looked at HudLight... seemed like what we wanted.
21:56.07drmessanoSuit yourself
21:56.15drmessanoYou'll be sorry
21:57.08duncan_16hrm.  Why would we be "Sorry"?  Need to ask the noob questions on this... so that I can get learned on the system ;)
21:58.41drmessanoHudlite uses some pseudo XML crap spewed forth by a perl bot running on an IRCd.. your clients are connecting to the IRC backend and attempt to interface to it
21:58.54drmessanoIt's some shit I would write on a bored weekend, not professional software
21:59.08drmessanoIt's buggy as hell, almost unsupported, and GREEN
21:59.45drmessanoWhen a feature breaks in Hudlite, fonality removes the feature
21:59.49drmessanoForget fixing it
22:00.15drmessanoIts a crap application that the devs know is crap that even they won't use
22:00.19duncan_16ok... I just downloaded the ASTassist.  Looking at how it configs.  What would we need to do with a stock install of the asteriskNOW server to run ASTassistant?
22:00.40heedlywow.... I thought you were joking at first.
22:00.52heedlyand then I See ircd-hybrid as a requirement
22:00.53drmessanoSet up manager accounts in manager.conf, assign them user rights, let the client connect
22:00.56drmessanoYes
22:01.01drmessanoIts a HORRID hack
22:01.12heedlygui looks pretty
22:01.30duncan_16we were partially interested because of the integration with Outlook.
22:01.48drmessanohudtapi freezes outlook
22:01.53drmessanoDoesnt work half the time
22:02.03drmessanoOr sends crap dial strings
22:02.05duncan_16ahh.  what about ASTAssistant?
22:02.13drmessanoIt's rock solid
22:02.41drmessanoWorks with AMI, so no need for some crap middleware to make it work
22:03.08*** part/#asterisk Assimilate (n=Assimila@216.83.78.108)
22:03.09duncan_16oh... ok.  So you can do the same functionality that you could with their piece, minus the outlook integration then, correct?
22:03.44drmessanoAstAssistant works with your Outlook Address book
22:03.53drmessanoJust doesnt have a tapi connection
22:03.58drmessanotapi sucks for the most part
22:04.04drmessanoWell I take that back
22:04.11drmessanoTheres few good implementations
22:04.49drmessanoBut most work for one specific Outlook version.. and then the app becomes buggy when a new MS Office is released
22:05.13drmessano"Oh, it installs on Outlook 2009.. Hmm.. but it send my credit card number to thailand"
22:05.16drmessanoShit like that
22:06.01drmessanoI seriously doubt the Outlook functionality in Hudlite works worth a crap in Outlook 07
22:06.17drmessanoIt barely worked in 2003.. I doubt they put the resources into updating it
22:06.22duncan_16heh.
22:07.12drmessanoAstAssistant has chat functionality built in that can use an IRC server.. but thats all it uses it for.. the chat backend
22:07.22drmessanoNot as the API
22:07.39duncan_16Here's the deal.  I do consulting for a P.O.S. company... they are implementing it in their office, and we are working close with them, since some of our product offerings are in-line with theirs.  We all just wanted to have a decent way of doing call switching/routing... it seemed as if the HUD piece would have done what we wanted.
22:07.40drmessanoNot sure I have ever seen a Telephony Internet Relay Chat API before
22:08.05duncan_16but... as I'm evaluating this piece a bit... I think it may be the solution.
22:08.27drmessanoAstAssistant is a great app, and the developer is very responsive
22:12.28duncan_16I guess I'll have to drop a line to the dev guy and get some info on how to connect with the outlook contacts.
22:12.49duncan_16we were also considering configuring up with LCS '07, since we're a M$ Gold Partner, but dunno if we want to go that far.
22:15.52alanciowhat is LCS?
22:16.15scooby2Whats the best place to get * documentation? The wiki is quite out of date as is the documentation on digium.com.
22:17.17ManxPowerscooby2: /path/to/src/asterisk/doc
22:17.22ManxPowerIt's a secret, so don't tell anyone!
22:17.33scooby2lol
22:17.38alanciohttp://downloads.oreilly.com/books/9780596510480.pdf
22:17.59ManxPowerTha asterisk book is a littel out of date, but still the best reference outside of the doc dir.
22:18.11ManxPowerAlso, I can't seem to type today.
22:18.33alanciowhy? you only misstyped a littel
22:18.50ManxPowerBut ANY time docs conflict with the info in /doc then the info in /doc should be considered correct.  It is THE place for Asterisk doc.
22:19.15scooby2I was just looking for some more explanation on the new queue/agent system. the doc dir explanation is pretty tough to understand
22:19.54scooby2queues-with-callback-members.txt
22:20.00ManxPowerscooby2: The queue/agent system is hell to understand regardless of the Asterisk version 8-|
22:20.17duncan_16alancio: Live Communications Server 2007 from Microsoft.
22:20.23barakudascooby2, what version ?
22:20.35ManxPowerWe have almost totally eliminated queues from my systems, but we are not a call center and don't have complex queue needs.
22:21.18scooby21.4.21.2 I'm getting the warning about "AgentCallbackLogin is deprecated and will be removed in a future release."
22:22.42scooby2so i figure that I might as well fix it now. Trying to upgrade from 1.2
22:23.35barakudadoes your agents sip peers ?
22:24.26scooby2yes
22:24.35barakudawe wrote down in our configs that the members of the queue is directly sip peers of operators extension
22:24.39barakudalike tha
22:24.40barakudat
22:24.42barakudamember => SIP/402
22:25.13barakudaso we have extension registered as queue member, just right after it's registration on asterisk
22:25.14scooby2ahh, no
22:25.27scooby2member =  Agent/535
22:25.52barakudawe wiped this out of queue.conf
22:26.14scooby2can they move to different locations that way?
22:26.21barakuda<PROTECTED>
22:26.44barakudayes, they just enter their sip login each time at differnet places
22:26.58scooby2that sounds nice
22:27.24barakudai've had same warnings
22:28.00barakudaand i've had to interact with * when i had to register as agent
22:28.09barakudanow this problems went away
22:28.43barakudai register extension, and autmatically, asterisk pushes queue calls for me
22:32.10scooby2sounds like what I need
22:32.26*** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net)
22:33.28*** join/#asterisk Greek-Boy (n=email@41.221.58.7)
22:34.40EmleyMoorI'm in the middle of applying a patch to make UK caller ID work
22:37.06EmleyMoorwonders how long this package is going to take to build
22:39.38*** join/#asterisk icel (n=mail@c-76-113-115-132.hsd1.nm.comcast.net)
22:40.13EmleyMoorI wanted to test my upgraded asterisk out but found a showstopper I needed to patch for
22:40.25*** part/#asterisk lzhang (n=lzhang@rrcs-67-78-33-170.sw.biz.rr.com)
22:41.52EmleyMoorIs there a list anywhere of what Asterisk 1.4 allows but considers deprecated?
22:42.11[TK]D-FenderlzhangUPGRADE.TXT
22:42.15[TK]D-FenderEmleyMoor: rather
22:42.31EmleyMoorI will have a look
22:43.29EmleyMoorI know my dialplan is 1.4 safe, and some "deprecated in 1.4" stuff that wasn't even needed in 1.2 is already clear
22:44.18*** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq)
22:45.21*** part/#asterisk mercutioviz (n=chatzill@freeswitch/developer/msc)
22:47.18*** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net)
22:52.35ManxPower[TK]D-Fender: make install needs to just cat UPGRADE.txt
22:59.35*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
23:06.52*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)
23:08.53*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
23:13.50*** join/#asterisk Greek-Boy (n=email@41.221.58.7)
23:17.20*** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net)
23:19.17EmleyMoorFound my first deprecated thing
23:22.15EmleyMoorMy FXO line is staying open if someone calls it and a SoftHangup occurs during that time
23:22.33EmleyMoor(and they don't hang up in consequence)
23:31.16*** join/#asterisk deeperror (n=deeperro@d149-67-49-94.try.wideopenwest.com)
23:51.19*** join/#asterisk Johnakabean (n=Johnakab@pool-72-82-114-148.nrflva.east.verizon.net)
23:52.18Wayhighis having a really hard time finding a local milliwatt line
23:52.47Johnakabeanwonders why wayhigh needs 1000 killowatts
23:54.02drmessanomilliwatt is not 1000 kilowatts
23:54.05Johnakabean<PROTECTED>
23:54.08seanbrightit's 1002
23:54.31Wayhighjohn: a milliwatt test line is a line that puts out 1004Hz tone at 0dBm.. it's used for testing line loss
23:54.33drmessanoAlways high on my list of qualifications for a tech
23:54.37drmessano"Bad at math"
23:54.47seanbright1011 i mean
23:55.04Wayhighthey're otherwise known as type 102
23:55.21Wayhigh11?
23:55.33seanbright17, then.
23:55.36Johnakabeanpulls out a crash cart, charges to 500, and hands it to wayhigh
23:55.48Johnakabeanyou want two to make 1000?
23:56.23deeperror1000?
23:57.15Johnakabeananyway, do i have to specify the DiD along with the cid or can i just use the cid
23:57.32Johnakabeanit straight up ignores everything except for DID's

Generated by irclog2html.pl Modified by Tim Riker to work with infobot.