irclog2html for #asterisk on 20060703

00:01.22*** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net)
00:03.01*** join/#asterisk JackEStorm (n=thinkthi@ip68-225-72-125.no.no.cox.net)
00:05.16*** join/#asterisk h0 (n=h0@ool-44c69453.dyn.optonline.net)
00:06.40*** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1)
00:12.17*** join/#asterisk iq|mobile (n=iq@unaffiliated/iq)
00:18.09*** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net)
00:25.59*** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net)
00:39.00*** join/#asterisk mjh001 (n=mjh001@c-68-37-78-102.hsd1.nj.comcast.net)
00:42.37*** join/#asterisk userdefined (n=jross@cpe-24-169-142-23.rochester.res.rr.com)
00:43.15*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220)
00:48.56*** join/#asterisk mpruett (n=mpruett@24-240-203-82.static.stls.mo.charter.com)
00:51.56mpruettanyone here?
01:03.03Hmmhesaysnot really
01:28.01*** join/#asterisk test34 (n=test34@unaffiliated/test34)
01:36.06*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
01:36.25*** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg)
01:42.15*** join/#asterisk Brijn (n=bas@S0106004063c0fa1f.vn.shawcable.net)
01:48.00BrijnAnyone seen the D-Link DPH-540  already?
01:50.25*** join/#asterisk dec_ (n=tom@ppp230-148.lns2.adl4.internode.on.net)
01:55.22*** join/#asterisk d-tech (n=dtc@72.245.233.107)
02:16.41*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-125-116.red.bezeqint.net)
02:20.30*** part/#asterisk Tier_1 (n=tier@c-67-176-28-65.hsd1.co.comcast.net)
02:33.37*** join/#asterisk d-tech (n=dtc@72.245.233.107)
02:38.08TheCopsAgent can have multiple call at the same time ?!
02:38.54*** join/#asterisk bhearsum (n=bhearsum@durhamlug/pdpc.basic.bhearsum)
02:38.57*** join/#asterisk inv_Arp (i=junya@c-67-191-62-53.hsd1.fl.comcast.net)
02:39.20bhearsumanyone here using a nortel i2002 (fw 0603B60) with asterisk?
02:39.51bhearsumi'm not getting any sound from my phone
02:39.58bhearsumbut when i place calls the phone on the other end rings
02:44.10VeNoMouS_lol i just had a rant on eyebeams forum
02:44.17*** join/#asterisk mspruett (n=mpruett@24-240-203-82.static.stls.mo.charter.com)
02:47.47*** join/#asterisk mpruett (n=mpruett@24-240-203-82.static.stls.mo.charter.com)
02:50.54*** join/#asterisk tengulre11 (n=tengulre@222.90.66.4)
02:54.07tzafrir_laptopanything wrong with the digium mailing lists?
02:55.58*** join/#asterisk P-NuT (n=nut@fw.office.unitedip.net.au)
02:57.00*** join/#asterisk _murf_ (i=murf@216.166.159.235)
02:59.58_murf_hmmmm. Late Sunday night(US) early monday morn (Europe) must be a slow time on this channel....!
03:00.24*** join/#asterisk bhearsum (n=bhearsum@durhamlug/pdpc.basic.bhearsum)
03:03.55tengulre11Hi,all, which channel discuss H.323 protocol?
03:09.04mrdigitalanyone use Zoom
03:12.43*** join/#asterisk L|NUX (n=linux@202.5.145.56)
03:36.41*** join/#asterisk mogorman (n=mogorman@adsl-220-179-200.mob.bellsouth.net)
03:37.38*** join/#asterisk Klydal (n=Klydal@ip68-226-15-98.nc.hr.cox.net)
03:38.47Klydalanyone available to help a newb?  Im trying to setup FWDout on my pbx and Im not sure how to make out going calls
03:46.47P-NuTHey all.
03:47.03P-NuTMaking outbound calls to other asterisk boxes...
03:47.19P-NuTis that done by making an extension and then dialing it?
03:48.25P-NuTlike this?
03:48.26P-NuTexten => 750,1,Dial(IAX2/s@ipadresshere,30,t)
03:48.26P-NuTexten => 750,3,Hangup
03:48.39P-NuTis that right?
03:51.39[TK]D-FenderP-NuT : missing step 2
03:58.16*** join/#asterisk masked (n=masked@ppp66-113.lns1.mel4.internode.on.net)
03:58.52DimitripietroAnyone could help me compiling the sangoma drivers ?
03:59.32mogorman./configure ; make ; make install ^_^
04:00.08DimitripietroYou never compiled sangoma driver for sure :-)
04:00.34mogormanbut you are right
04:00.39mogormani dont have any sangoma hw
04:01.03*** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net)
04:02.18websae_sagnoma is the BEST :)
04:02.47mogormanbut i am biased
04:02.49[TK]D-Fenderif mogorman were to say Sangoma 3 times his head would explode ;)
04:02.54mogormanworking for their competitor
04:03.05mogormanlol [TK]D-Fender
04:03.13mrdigital[TK]D-Fender: ever hear of Zoom?
04:03.44*** join/#asterisk JamesDotCom (i=jamesdot@creep.bur.st)
04:04.10Dimitripietro<websae_> You are using sangoma driver ?
04:04.53DimitripietroHave you ever had any error concerning udev ?
04:05.57*** join/#asterisk Eggplant (i=No@dsl-216-155-213-228.cascadeaccess.com)
04:06.09mogormanyour rules are probabl bad Dimitripietro
04:06.26DimitripietroI don't know anything about udev
04:06.51mogormanwell i dont know anything about sangoma, but you probably have /etc/udev/rules.something
04:06.59mogormanand it needs to have correct entries
04:07.35DimitripietroI do have /etc/udev/rules.d/ but I don't know what the correct entries look like
04:07.54DimitripietroI was ruinning my system with a TDM400 and everything was fine
04:07.57mogormanthere is probably sample in sangoma source
04:09.37DimitripietroSee as there is a samples, i'M gonan take a llok
04:09.39Dimitripietrolook
04:10.14mogormanokies
04:10.20mogormanhope you can get it working
04:11.04[TK]D-Fendermrdigital : Yup
04:12.09DimitripietroEverything from the sample is already present in the udev rules file :-(
04:12.48mogormanonly thing i can think of is your udev is yuck
04:12.52mogormanbut id have to see
04:14.31russellbyou will not see!
04:14.37mogormanwell yeah
04:14.41mogormanim about to go to bed
04:14.43russellb:)
04:14.44mogormantired
04:16.56*** join/#asterisk Gamercjm (n=chris@pool-71-254-175-66.lsanca.fios.verizon.net)
04:17.53P-NuThey all,
04:18.10P-NuTwho can dial my FWD number for me?
04:19.58QwellP-NuT: the fwd callback app can
04:21.33*** join/#asterisk NotJohnDavid (i=dave@c-68-47-199-178.hsd1.tn.comcast.net)
04:21.54*** join/#asterisk silly_ (n=silly@cpe-70-112-173-32.austin.res.rr.com)
04:28.15*** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net)
04:28.45P-NuTyeah, I tried that, but am getting nothing..
04:29.02P-NuTso I was wondering if somebody could dial me and tell me what error they get.
04:29.12P-NuTmaybe I've not got soemthing right.
04:30.33NotJohnDavidhm
04:30.56P-NuThm?
04:31.59NotJohnDaviddial you for what?
04:32.36littleballhello, i am looking for small size box for linux, who can suggest?
04:38.16P-NuTNotJohnDavid: Well, I tried it from the website, and it never called. So I was wondering if someone could try dialing my number and tell me what result they get.
04:38.24P-NuTNotJohnDavid: Make sense?
04:39.30NotJohnDavidwhat is your #
04:39.46*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
04:41.46*** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie)
04:44.16*** join/#asterisk variable_office (n=variable@Adv-Proprietary-Systems.s7-0-0.2-15-0.ar4.CHI1.gblx.net)
04:44.44variable_officehow can you change the voicemail message?
04:50.08littleballhello, anyone using spandsp to send out fax?
04:54.25VeNoMouS_<PROTECTED>
04:54.25VeNoMouS_Jul  3 16:48:08 WARNING[10138]: app_getdtmf.c:95 getdtmf_exec: GET_DTMF_DIGITS was set to [54654]
04:54.25VeNoMouS_<PROTECTED>
04:54.25VeNoMouS_<PROTECTED>
04:54.25VeNoMouS_<PROTECTED>
04:54.30VeNoMouS_err shit wrong window
04:55.30mpruettRusselb I have an odd one for you
04:55.30Sedoroxdarn.. no passwords :p
04:55.46VeNoMouS_Sedorox : lol
04:56.39mpruettI have two boxes setup (I believe) exactly the same
04:57.33mpruettOn one box when I make a sip connection you can't hear the other side when they speak but they can hear you
04:57.49mpruettThe other box works fine
04:58.01VeNoMouS_checked your firewall rules/
04:58.05VeNoMouS_rtp allowed in both ways?
04:58.24mpruettIf both callers use Meetme both sides can communicate fine
04:58.32VeNoMouS_sip would've established the call, but if only one rtp is going
04:58.39VeNoMouS_that would say to me your blocking a stream
04:58.43mpruettI turned of my firewall - problem still exist
04:59.00VeNoMouS_yes but meetme is a conf
04:59.15VeNoMouS_it dont try neg the streams
04:59.39*** join/#asterisk pingywon (n=mike@c-71-230-221-39.hsd1.pa.comcast.net)
05:00.22mpruettVeNoMous: So your thinking a firewall is blocking me somewhere?
05:00.53mpruettI don't believe my CoLo is filtering me but I could check
05:01.19mpruettThe box that is working fine is local the one that isn't is at CoLo
05:03.56VeNoMouS_rtp debug ip <hostnamehere>
05:04.32*** part/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg)
05:06.16VeNoMouS_well
05:06.21VeNoMouS_err wrong window ffs
05:07.03mpruettVeNoMous: You didn't mean that for me correct?
05:07.52VeNoMouS_the rtp
05:07.52VeNoMouS_yes
05:07.59VeNoMouS_the "well" no
05:08.14mpruettok let me try that
05:15.56mpruettOK now the stupid question
05:16.08mpruettHow do I use this command
05:16.40mpruettAll I get is the following
05:16.41mpruettUsage: rtp debug [ip host[:port]]
05:16.41mpruett<PROTECTED>
05:17.53mpruettIn the ip host:port - I put the IP of the ATA I am using and 10000 as the port
05:22.50mpruettVeNoMous: I turned on rtp debug
05:23.45mpruettVeNoMous: then I made a call - I can see RTP packets to and from host
05:34.28VeNoMouS_on both sides?
05:34.40VeNoMouS_i mean on both boxes?
05:34.44mpruettYes
05:36.18mpruettAnother clue - I just made a call and for a couple seconds it worked fine then it went back to how I described b4 in mid call
05:36.25*** part/#asterisk variable_office (n=variable@Adv-Proprietary-Systems.s7-0-0.2-15-0.ar4.CHI1.gblx.net)
05:37.59VeNoMouS_err in mid call?
05:38.21VeNoMouS_weird
05:38.35mpruettYes we could hear each other fine for a couple seconds then I couldn't hear him talking but he could still hear me
05:39.38VeNoMouS_<mpruett> In the ip host:port - I put the IP of the ATA I am using and 10000 as the port
05:39.40VeNoMouS_no
05:39.46VeNoMouS_u should use the ip of the remote machine
05:40.46mpruettOK - will it give me different info that just "rtp debug"?
05:40.54mpruetts/than/that
05:41.25mpruettLet me try
05:42.58muppetmaster<PROTECTED>
05:43.02muppetmaster<PROTECTED>
05:43.03muppetmaster<PROTECTED>
05:43.03muppetmasterazz
05:43.05muppetmaster<PROTECTED>
05:44.26muppetmasterApologies, my son got a hold of the keyboard
05:45.54russellbthat's awesome :)
05:46.01mpruettlol - and he is already saying dirty words!! A techy in the works!
06:03.40*** join/#asterisk Assid (i=assid@203.115.83.215)
06:05.07*** join/#asterisk stephane_ (n=stephane@merlin.cabale.net)
06:05.21stephane_jour
06:09.28P-NuTbonjour.
06:12.48*** join/#asterisk nagl (n=nagl@86.59.54.237)
06:19.38*** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr)
06:24.17muppetmasterAnyone else here have the Nokia E61?
06:25.01filein an alternate universe I do
06:25.36*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220)
06:25.46muppetmasterWe all do in that case
06:26.42Corydon76-homebut only in separate parallel universes
06:27.00*** part/#asterisk P-NuT (n=nut@fw.office.unitedip.net.au)
06:28.08VeNoMouS_<PROTECTED>
06:28.08VeNoMouS_Jul  3 18:21:40 WARNING[16012]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x8143078', 10 retries!
06:28.08VeNoMouS_<PROTECTED>
06:28.08VeNoMouS_Jul  3 18:21:48 WARNING[16029]: app_getdtmf.c:73 getdtmf_exec: Digits Entered Were [123456987**789***000**7]
06:28.08VeNoMouS_Jul  3 18:21:48 WARNING[16029]: app_getdtmf.c:78 getdtmf_exec: Digit Limit set of [3]
06:28.10VeNoMouS_Jul  3 18:21:48 WARNING[16029]: app_getdtmf.c:81 getdtmf_exec: GET_DTMF_DIGITS was set to [123]
06:28.12VeNoMouS_arse
06:28.28*** join/#asterisk P-NuT (n=nut@fw.office.unitedip.net.au)
06:28.28VeNoMouS_thats the prob with having to bitchx's open
06:28.44*** part/#asterisk P-NuT (n=nut@fw.office.unitedip.net.au)
06:37.50*** join/#asterisk pa (n=paolo@unaffiliated/pa)
06:38.43*** join/#asterisk blkremedy (n=ur3rdeye@193M23.oasis.mediatti.net)
06:39.41blkremedydoes anyone here know of a simple program that can be used to clone a asterisk install.
06:39.43*** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it)
06:41.39VeNoMouS_dd
06:41.41VeNoMouS_:P
06:45.03*** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it)
06:45.08*** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg)
06:46.19*** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135)
06:50.09SplasPooddoes AEL1 support any type of #include style syntax?
06:50.17*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no)
06:51.36*** join/#asterisk P-NuT (n=nut@fw.office.unitedip.net.au)
06:51.39*** join/#asterisk L|NUX (n=linux@202.5.145.56)
06:53.54P-NuTHey all, if I try to call anyone from my IAX extension, (external to my network) I get congestion and it just hangs up.. What am I doing wrong?
06:54.23P-NuTI have port 4569 open, do I need anything else?
06:54.39L|NUXallow guest
06:54.39L|NUX:)
06:54.58P-NuThmm..
06:55.02P-NuTI thougth I did...
06:55.04P-NuTlets see..
06:55.10P-NuTof
06:55.13P-NuToh ok,
06:55.31P-NuTso in iax.conf under general, I want allow=guest ?
06:55.32L|NUXokay
06:55.50P-NuTyeah?
06:56.05L|NUXna
06:56.07L|NUXsip.conf
06:56.12P-NuToh...
06:56.17P-NuTum..
06:56.20*** join/#asterisk X-Gen (n=X-Gen@dsl-145-219-164.telkomadsl.co.za)
06:56.24P-NuTwhy sip.conf?
06:56.36L|NUXbecause sip.conf have it
06:56.40P-NuToh ok.
06:56.42P-NuTthsanks
06:56.45L|NUXbasically you can do like this
06:56.51L|NUXadd a user in your iax.conf like this
06:56.55L|NUX[linux]
06:57.00L|NUXtype = friend
06:57.04L|NUXusername = linux
06:57.12L|NUXsecret = ******
06:57.16L|NUXhost = dynamic
06:57.26L|NUXcontext = default
06:57.35L|NUXcallerid = "Linux Calling"
06:57.45L|NUXand when you want to dial to your network
06:57.48L|NUXyou can dial using this
06:57.53L|NUXin your extensions.conf
06:58.37L|NUXexten => 1,1,Dial(linux:******@host/exten,60,tr)
06:58.44L|NUXsimple :>
06:58.48P-NuThmm....
06:58.51P-NuTok then...
06:58.57L|NUXreload
06:58.59L|NUXand then call
06:59.54P-NuTdo I need secret in there?
06:59.58L|NUXyeah
07:00.00L|NUXyou need
07:00.01L|NUX:)
07:00.23*** join/#asterisk infinity1 (i=foobar@208.184.76.100)
07:01.11P-NuTok, so in ,Dial(linux:******@host/exten,60,tr)   linux:****** is the username and password.
07:01.31L|NUXyeah
07:02.04*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220)
07:02.14P-NuTso if I wanted to call digium lets say, it would be exten => 1,1,Dial(linux:******@misery.digium.com/6000,60,tr)
07:02.17P-NuTis that right?
07:04.07P-NuTand also, where in sip.conf do I put the allow=guest. [General] or [authentication] ??
07:04.43L|NUXyeah
07:05.00L|NUXwell see general section of your sip.conf
07:05.24SheriF_WorKwhat kind of softphone supports G729 or G726 or G723.1 and also supports SIP INFO in DTMF mode ?
07:06.09VeNoMouS_eyebeam
07:07.02L|NUXSheriF_WorK : but g723 is not for sale
07:07.22L|NUXSheriF_WorK : you can get it if you buy 250 + copies
07:07.24SheriF_WorKL|NUX: there is an opensource G729 and G723
07:07.26*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220)
07:07.40L|NUXSheriF_WorK : codec is different thingy
07:07.48L|NUXSheriF_WorK : but softphone is different
07:08.06SheriF_WorKL|NUX: yes :-s good point ok i want a phone supports G729 :-s
07:08.19L|NUXSheriF_WorK : eyebeam
07:08.31SheriF_WorKL|NUX: any other options ?
07:08.34muppetmasterLINUX:  G729 is proprietary and licensed.
07:08.37muppetmasterNo way around that
07:08.56muppetmasterYou may only get a binary from Digium or others
07:09.08SheriF_WorKhttp://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
07:09.31L|NUXmuppetmaster : i know but there is some open source thingy arround with intel ipc ....
07:09.42L|NUXas SheriF_WorK given link
07:09.51muppetmasterLINUX I don't think so, if so, it would be a violation of licensing for G729
07:10.06L|NUXSheriF_WorK : use X-pro
07:10.16SheriF_WorKPlease note that this code is available for you to download for education purposes only
07:10.23L|NUXmuppetmaster : might be but no one keep this sites down
07:11.17SheriF_WorKL|NUX: no other options for a softphone ?
07:11.35L|NUXwell there are some opensource :0
07:12.39L|NUXSheriF_WorK : http://www.portsip.com/
07:12.50L|NUXthis have g723 + g729 :)
07:14.42DrkShdwL|NUX: basically,  you are getting a multi-thousand dollar PBX system for free.   don't go pirating the g729 codecs.   hell,  it's only $10/channel.
07:15.07*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
07:15.15DrkShdwSorry.  I stayed quiet long enough.  :)
07:15.52L|NUXDrkShdw : did i said any thing :(
07:16.06L|NUXDrkShdw : i am not the person i just said yes i have seen
07:16.23L|NUXDrkShdw : i my self have digium g729 license
07:16.31*** join/#asterisk Dico_ (n=niko@60.51.217.61)
07:16.34*** join/#asterisk CleanerX (n=nix@p54A38827.dip0.t-ipconnect.de)
07:30.10SplasPoodCan anyone point me in the right direction as to how I can use Asterisk realtime static meetme.conf ?
07:33.02Bert-I bought one but as I'm student, I could have used G729 without paying for taht, no ?
07:33.06*** join/#asterisk angom_h (n=angom@red-corp-200.76.251.26.telnor.net)
07:33.47*** join/#asterisk tparcina (n=tparcina@lns01-0003.dsl.iskon.hr)
07:34.07tparcinahi channel!
07:34.21Bert-and another question : If I want to put my asterisk on a new computer, what about the license I bought ?
07:34.24Bert-is it portable ?
07:34.28Bert-hi tparcina
07:35.01tparcinai head beautifull weak in brussel. it's realy a nice town. hopefully i'll go back thate again...
07:35.25*** part/#asterisk angom_h (n=angom@red-corp-200.76.251.26.telnor.net)
07:35.30DrkShdwBert-: you can transfer the license to another machine once.  after that, you have to contact digium
07:37.22Bert-okay
07:37.31Bert-anyway, it is only 10$ :)
07:37.57DrkShdwper channel,  yes
07:38.03Bert-yep
07:38.11*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
07:38.18Bert-and I bought one for the compagny I'm working for (I'm a trainee)
07:38.31Bert-but I don't want to give them my own license
07:39.31*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
07:44.05tparcinahi Bert, it's quite today - everybody goone to vacation?
07:47.05Bert-??
07:47.17Bert-from my side, I've to go to work :(
07:47.26Bert-have a good day here :)
07:49.29*** join/#asterisk Splat (n=Splat@220-253-134-28.TAS.netspace.net.au)
07:50.03*** join/#asterisk kay2 (n=ashdown@sd-420.dedibox.fr)
07:56.38*** join/#asterisk hads|home (n=hads@mail.nice.net.nz)
07:57.48muppetmasterSo, I posted this on the E61 @ Nokia:  http://discussion.forum.nokia.com/forum/showthread.php?t=83985
07:57.57muppetmasterAbout to take it back to FNAC if I can not get inbound working on this thing.
08:03.37*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
08:11.43*** join/#asterisk bmg505 (n=leon@196.207.32.253)
08:13.27*** part/#asterisk bmg505 (n=leon@196.207.32.253)
08:18.16_4d4m_hi all.. anyone able to offer some advice on billing app's for *?
08:19.23_4d4m_have looked at plenty of options but am not sure which is right for me
08:19.40_4d4m_couple of hundred pre-pay accounts to handle
08:20.09*** join/#asterisk faberk (n=faberk@80.181.228.104)
08:20.17_4d4m_nothing too flash.. not bothered about UI.. something lightweight, flexible, and known to work well.
08:20.38_4d4m_am running * realtime static, flat csv cdr at the mo
08:27.12*** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org)
08:36.53*** join/#asterisk stephane_ (n=stephane@merlin.cabale.net)
08:36.55*** join/#asterisk AltnTab (n=ecs@nrjsoft13.networx-bg.com)
08:41.26hwthas anybody managed to compile spandsp rxfax on recent asterisk?
08:43.59*** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net)
08:58.13*** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net)
09:01.52*** join/#asterisk tsurk0 (n=tsurko@85.187.160.157)
09:02.01*** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net)
09:09.14*** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net)
09:09.15Bert-hop
09:09.47*** join/#asterisk TeePOG (n=1234@dsl-145-143-190.telkomadsl.co.za)
09:09.53TeePOGgood morning
09:10.04TeePOGHi FuriousGeorge!
09:10.11*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
09:11.05*** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com)
09:11.42*** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net)
09:12.41*** join/#asterisk Joe__11 (n=Miranda@host217-114-154-220.pppoe.mark-itt.net)
09:14.02*** part/#asterisk Joe__11 (n=Miranda@host217-114-154-220.pppoe.mark-itt.net)
09:23.39*** join/#asterisk ionix (n=ionix@p1200-ipbfp05miyazaki.miyazaki.ocn.ne.jp)
09:24.21*** join/#asterisk bmg505 (n=leon@c1-199-16.rndf.isadsl.co.za)
09:42.41Bert-according to you guys, what is the "best" gui to use asterisk ??
09:43.16dpryovim
09:43.21dpryoand asterisk -r
09:43.39Bert-hahaha
09:43.42Bert-I totally agree
09:44.08Bert-but I want a kind f graphical interface to add/remove a phone, etc ...
09:44.25Bert-beacause I not sure to work for my company for ever :)
09:46.51hwtBert-: there aren't any good, IMHO.
09:47.00hwtBert-: for stock *.
09:48.46*** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
09:50.34Bert-I tried freepbx, but it never worked wih Free account
09:50.53Bert-and Trixbox ... how to say ... it sux
09:50.56Bert-:)
09:51.43*** join/#asterisk dlynes_laptop (n=dlynes@zz212094.cipherkey.net)
09:54.17*** join/#asterisk TeePOG (n=1234@dsl-145-143-190.telkomadsl.co.za)
09:54.34dlynes_laptopGood morning, peeps
09:57.57*** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se)
10:00.03*** join/#asterisk postel_ (n=jp@unaffiliated/postel)
10:05.15*** join/#asterisk mosty (i=mostynm@60-241-198-194.static.tpgi.com.au)
10:06.05*** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org)
10:08.14*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
10:10.14*** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au)
10:16.10*** join/#asterisk d-tech (n=dtc@72.245.233.107)
10:16.32*** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
10:23.53*** join/#asterisk mrtwister (n=manopulu@107.250.broadband5.iol.cz)
10:26.09*** join/#asterisk damned (n=vpol@prior.lanck.net)
10:26.35*** join/#asterisk frenzy (n=frenzy@196.46.104.62)
10:26.39frenzyhey all
10:26.57frenzywhere can I get standard pbx voice overs?
10:27.05frenzythe free stuff
10:27.29MrChimpyget a microphone and speak into it
10:27.40frenzyhahaha
10:28.04MrChimpythere's the standard asterisk-sounds set
10:28.12MrChimpysame place you got asterisk
10:29.09*** join/#asterisk Bert- (n=bert@LAubervilliers-151-12-81-84.w193-252.abo.wanadoo.fr)
10:30.47damnedcan anybody suggest if asterisk is technically able to initiate voice call and if it would be taken - connect person to another number?
10:31.27dlynes_laptopdamned:  read up on the wiki about how to do call files
10:32.44mostydamned: i have seen click to call firefox extenions that use asterisk
10:32.56mostyso i guess so
10:33.04damneddlynes_home: the actual question is about ability to force asterisk to call from a command or a script.
10:33.28dlynes_laptopdamned: yeah...like I said...use call files
10:33.38dlynes_laptopdamned:  you can use a script to generate them
10:33.46damneddlynes_home: ok. thnx.
10:33.55dlynes_laptopdamned:  asterisk checks the spool directory for the call files periodically
10:34.00VeNoMouS_damned : perl script
10:34.12dlynes_laptopdamned:  when it finds one, it automates a process based on the info in that call file
10:34.21VeNoMouS_http://search.cpan.org/~jhiver/Asterisk-LCR-0.08/lib/Asterisk/LCR/Dialer.pm
10:36.42MrChimpyaye, i've used call files. easy.
10:38.07MrChimpylike the music on hold feature, but a looped sample which starts when the dial starts
10:38.28*** join/#asterisk P-NuT (n=P-Nut@CPE-60-227-93-75.nsw.bigpond.net.au)
10:39.14*** join/#asterisk fulgas (n=fulgas@82.102.2.30)
10:43.12dlynes_laptopMrChimpy:  why not write it yourself?
10:49.40*** join/#asterisk pingywon (n=mike@c-71-230-221-39.hsd1.pa.comcast.net)
10:54.06*** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net)
10:58.03*** join/#asterisk P-NuT (n=P-Nut@CPE-60-227-93-75.nsw.bigpond.net.au)
11:09.35*** join/#asterisk psk (n=psk@golia.caltanet.it)
11:09.37MatsKIs there anyone that has experience with pattern matching with letters instead of numbers ?
11:10.02MatsKexample: exten => _smith,1,Answer
11:12.33MatsKand how is dotts interpreted in letter pattern, example: exten => _john.smith,1,Answer
11:14.43*** join/#asterisk P-NuT (n=P-Nut@CPE-60-227-93-75.nsw.bigpond.net.au)
11:14.57dlynes_laptopMatsK:  probably \.
11:16.42MatsKyou mean that it's possible to escape it ?
11:17.01*** part/#asterisk P-NuT (n=P-Nut@CPE-60-227-93-75.nsw.bigpond.net.au)
11:18.28kay2u can't
11:19.19MatsKkay2: Thx I thought so, I have to ask for that feature then ;-)
11:22.11*** join/#asterisk beyond (n=beyond@200.192.160.100)
11:25.41*** join/#asterisk __chris (n=chris@unaffiliated/redlined)
11:28.49MrChimpydlynes_laptop: that's what I meant when I said "MrChimpy suspects he will be writing it himself"
11:29.19MrChimpywhy not write it myself? it'd be a bit silly to re-do it if someone has done it already.
11:31.36__chriswhen storing callerID names/numbers in * using database put cidname how are these actually stored?  I'm trying to back them all up but can't seem to find them anywhere, grep knownstorednumber doesn't return anything either
11:34.08vgsterastdb
11:38.08*** join/#asterisk zotz (n=zotz@24.244.133.115)
11:38.54*** join/#asterisk kmilitzer (n=km@office-gw.westend.com)
11:40.14*** join/#asterisk yxa (i=lonari@cm121.gamma228.maxonline.com.sg)
11:45.51*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
11:57.30knoboI have a diva 2.02 PCI card. lspci shows it to me, but I'm not shure which tools to use to configure it
11:58.00knobocapiinfo says "capi not installed"
11:58.25knobocapi modules are loaed. I can se them with lsmod
11:58.53knobodivacapi is loaded and in use by the kernelcapi module
12:04.23MrChimpywtf?
12:04.38MrChimpy1.2.9.1 doesn't build!
12:04.47MrChimpymake[1]: Leaving directory `/root/asterisk-1.2.9.1/channels'
12:04.55MrChimpyxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
12:05.05MrChimpychan_zap.c: In function `pri_dchannel':
12:05.05MrChimpychan_zap.c:9038: error: structure has no member named `call'
12:05.07muppetmasterMrChimpy - Build for me
12:05.07*** join/#asterisk cjk (n=cjk@80.92.64.103)
12:05.15muppetmasterI do have a problem with SVN TRUNK though....
12:05.18MrChimpyi grabbed the latest zaptel in case that fixed it
12:05.19muppetmasterPretty unstable
12:05.28cjkhi, does anyone know a good website to do reverse number lookups?
12:05.31MrChimpy1.2.9.1 unstable?
12:05.45muppetmasterMrChimpy - No, 1.2.9.1 is fine, SVN TRUNK is unstable
12:05.52MrChimpyah, ok
12:06.18MrChimpyquick google said it builds without zaptel and iax trunking
12:06.29muppetmasterI was at Astricon last week and Kevin Flemming said they are trying to get v1.4 beta out this week.  Seems a stretch.
12:06.30*** join/#asterisk fenlander (n=fenlande@82.152.81.57)
12:06.33MrChimpywhich are the two major portions that my application uses
12:07.23MrChimpyi think i may end up sticking with a crusty old version and forgetting about submitting stuff to digium
12:08.18*** join/#asterisk rogier (n=rogier@83-67-dsl.ipact.nl)
12:08.36MrChimpyah
12:08.39MatsKMrChimpy: And after that will it be more stable ?!
12:08.46MrChimpyi need to update libpri too.
12:08.56MrChimpymatsk: it's stable with what i'm using now
12:09.37*** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka)
12:11.09*** part/#asterisk muppetmaster (n=jasongoe@169.red-81-184-73.user.auna.net)
12:11.36MrChimpyat least if i screw it up i know what i've done :)
12:12.28MrChimpylive platform is too critical to go playing upgrade frenzy on
12:17.14MatsKWell, I have a "dogfood" platform that is upgraded first and then after a test period is the production platfrom upgraded.
12:17.19*** join/#asterisk e1mer (n=root@58.71.14.245)
12:17.52*** part/#asterisk \etc\bin (n=root@58.71.14.245)
12:21.21*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
12:25.48knoboanyone has a eicon Diva card?
12:26.37MrChimpytesting for me is something of a problem, as we have very heavy call volumes which I can't replicate - so any upgrade is a case of trying it and having a spare pair of trousers for when it goes wrong
12:30.55*** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br)
12:31.18MatsKI just use SIPP to simulate some load and we have also some inhouse usage on the dogfood platform so it's not impossible to emulate your heavy call volume ;-)
12:31.45*** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com)
12:32.07mosty__chris: they can be stored in sip.conf or iax.conf for those types of clients, or set on the clients themselves, i think
12:34.52*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
12:39.53*** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane)
12:40.04*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)
12:44.36*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
12:56.35*** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net)
12:58.17*** join/#asterisk Katty (n=aisaacs@64.82.232.54)
12:59.20[TK]D-FenderKatty: Mew
13:00.33*** join/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net)
13:04.38*** join/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net)
13:08.20Kattymorning mister fender (=
13:16.15*** join/#asterisk BertZ (n=bert@bas33-1-82-66-4-198.fbx.proxad.net)
13:16.52trelane_I'm having severe IRQ problems, is there a way to manually steer interrupts via insmod?
13:17.22VeNoMouS_only if the module supports it
13:17.23MooingLemurlemme guess.. a dell server :P
13:17.28VeNoMouS_ie its been written into the code
13:17.40VeNoMouS_MooingLemur comming it has to be a ibm!
13:17.53VeNoMouS_ibm == its buggered mate
13:18.14VeNoMouS_hp dl 140 > *
13:19.05MooingLemurI think that's biting us too.  Our asterisk boxes are dell 1750s, and they get pretty choppy with disk access
13:19.24trelane_MooingLemur, how'd you know?
13:19.33VeNoMouS_what kernel u guys running?
13:19.41*** join/#asterisk pengyong (n=lala@218.93.71.200)
13:19.57MooingLemurdells tend to route everything to one IRQ
13:20.02trelane_MooingLemur, I'm runing on 830's
13:20.05MooingLemurpain in the butt
13:20.13*** join/#asterisk skraelings001 (n=skraelin@201.230.209.141)
13:20.23skraelings001hi everyone
13:20.27trelane_MooingLemur, I've got most of the bios steering disabled and am manually assigning IRQ's however network and smbus and raid and ide seem to want the same IRQ's
13:20.30VeNoMouS_better yet, time to go smoke a fat one
13:21.49trelane_MooingLemur, I can get asterisk working or I can get a stable system (the other option is to put the wctdm24xxp against the ide controller
13:22.18skraelings001i'm trying to do an agi application and so far i don't know what format channelname should have. anyone know? is this like the output in show channels ?
13:22.44skraelings001the command is CHANNEL STATUS [channelname]
13:22.44VeNoMouS_well what u writing it in genious?
13:22.50VeNoMouS_c,perl, python?
13:22.54MooingLemurI don't really have the answer.  The only problem I'm having is with rxfax and IRQs screwing that up
13:22.56skraelings001python
13:23.09VeNoMouS_dont know python ure out of luck
13:23.11MooingLemurshared IRQs that is
13:23.38skraelings001i didn't give reference, cause i don't think is language-dependent
13:23.39VeNoMouS_and the channel is the rtp stream
13:23.45trelane_MooingLemur, what's sharing for you?
13:25.01MooingLemurdigium card and scsi controller
13:25.04VeNoMouS_skeffling
13:25.05VeNoMouS_http://home.cogeco.ca/~camstuff/agi.html#CHANNELSTATUS
13:25.12skraelings001i'm thinking is like for everycall channelname's format is SIP/101-234w3fs or IAX2/102-239fsd
13:25.13VeNoMouS_^^ pyython agi
13:25.17trelane_MooingLemur, both are discrete cards? (physical hardware and not onboard)
13:25.21VeNoMouS_CHANNEL STATUS Zap/9-1
13:25.21VeNoMouS_Return the status of channel Zap/9-1
13:25.30MooingLemurscsi is onboard
13:25.48VeNoMouS_its amazing when you type in "asterisk  CHANNEL STATUS python" into google
13:25.49trelane_MooingLemur, there's options in bios to move irq's
13:26.17MooingLemurI'm on vacation anyway.. I'll deal with that when I get back home :P
13:26.17*** join/#asterisk pbx1 (n=pbx1@58.69.92.39)
13:26.51trelane_MooingLemur, I'M NOT!
13:26.52trelane_:-p
13:27.53skraelings001VeNoMouS_ : i guess so, thanks
13:34.46VeNoMouS_np
13:35.16*** join/#asterisk coppice (n=chatzill@61.197.17.210.dyn.pacific.net.hk)
13:36.08skraelings001VeNoMouS_ : it's what i expected, it would work with zap, but sip or iax assign a random set of numbers and letters to tech/ext, this is the problem
13:37.39VeNoMouS_skraelings001 not really
13:37.41VeNoMouS_because if you had read
13:37.44VeNoMouS_u would have seen
13:37.44*** join/#asterisk vgster (n=vgster@217.78.147.238)
13:37.45VeNoMouS_Return the status of the specified channel. If no channel name is specified, return the status of the current channel.
13:38.12VeNoMouS_unless u want a certain channel
13:38.35skraelings001VeNoMouS_ : certain channel
13:40.30VeNoMouS_skraelings001 so call the var # ${CHANNEL}
13:40.33VeNoMouS_skraelings001 so call the var  ${CHANNEL}
13:40.48*** join/#asterisk Spy000007 (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net)
13:55.38BertZhow to see which codec is used for an active call please ?
13:57.44a1faAnybody know a provider that sells $1did + 1c/min?
13:59.18*** part/#asterisk kmilitzer (n=km@office-gw.westend.com)
13:59.26BertZhmm
13:59.34BertZwho works with G79 codecs here plz ?
13:59.45BertZI have a lot of errors about VAD
14:00.18malcolmdturn off VAD on your endpoint or the connecting gateway
14:00.23malcolmdVAD isn't supported by Asterisk
14:01.36BertZhmm
14:01.55BertZVAD is not supported by me , as I don't know what is it :)
14:02.17coppiceVoice Activity Detection
14:02.19*** join/#asterisk Pepse (n=pepse@ip68-109-169-37.ph.ph.cox.net)
14:02.23BertZokay
14:02.24BertZthx :)
14:04.04BertZhm
14:04.40BertZthere is no options about VAD on my endpoint. It is a Nextone SoftSwitch. Does someone ever worked with asterisk, G729 and Nextone plz ?
14:06.06coppiceits usually selectable. maybe they call it something else in the config
14:09.21BertZno sip option to specify no VAD in asterisk ??
14:09.35*** join/#asterisk Hmmhesays (i=negative@66.173.103.110)
14:10.47*** join/#asterisk ariel_ (n=Ariel@70.46.87.158)
14:11.07BertZsupoprt said to me that I've to disable VAD on the GW, not on the Nextone
14:11.18BertZbut I see no option about VAD in Asterisk :
14:11.20BertZ:(
14:11.36*** join/#asterisk HuSoft (n=Hoo@227stb47.codetel.net.do)
14:12.22coppicethere's no option for VAD in Asterisk, as it doesn't have support for VAD
14:12.55BertZI use a Grand Stream maybe there is an option ...
14:13.24*** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com)
14:13.37Hmmhesayslikely assumption
14:13.57HmmhesaysIf you "need" to use VAD, here is a clue... don't use VOIP
14:14.07HuSoftIs there a way I can make asterisk recognize calls using context names?  for example, a user normally calls with the url, sip:1234@10.0.0.3, Is it possible to call: sip:husoft@10.0.0.3 ? (husoft is the context name for the extension 1234).
14:14.22coppicethere's nothing wrong with true VAD
14:14.23Hmmhesaysof sorts
14:14.24*** join/#asterisk Qb3rt (n=jhgjkgui@kyle.colba.net)
14:14.40Qb3rtmy litle problem (question) ---> http://pastebin.ca/77848
14:15.26Hmmhesaysyeah but generally people who are asking about it have something like a 14.4k connection with about 1500ms ping between endpoints
14:15.32BertZhmmm on voip-info.org, G723 and G729 must be disabled :(
14:15.53*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
14:16.00BertZwhen I disable G723/G29, all works fine :(
14:16.16BertZbut I want to use it, as it need less bandwith
14:16.27HmmhesaysThen you need to read more
14:16.34BertZsure
14:16.46Hmmhesaysfor g.729 you either need a license, or use it  passthru only
14:16.51Hmmhesaysg.723 is passthru only
14:17.30rob0Qb3rt: if [ -z "$1" ] ; then exit 1 ; fi
14:17.50rob0you could also test to be sure it's a directory
14:17.56rob0"help test"
14:17.58BertZHmmhesays : Grand Stream has G723/G729, so no problem for passthrough
14:18.07BertZI bought a G729 license from digium
14:18.30*** join/#asterisk kay2 (n=ashdown@sd-420.dedibox.fr)
14:18.31BertZbut I can test G723 and G729, as I'm only testing asterisk.
14:18.42HmmhesaysThat last sentence made no sense
14:18.50BertZ?
14:19.11BertZI'm a student, I'm trying to see how asterisk works, then make a summary of it
14:19.20BertZI don't have to pay for G729
14:19.30BertZwritten on digium's website if I remember
14:19.36BertZanyway
14:19.40BertZI bought my own license
14:19.44BertZI want it to work
14:19.52BertZthe problem is VAD, not G729 license
14:19.59BertZI'll find :)
14:20.02HuSoftIs there a way I can make asterisk recognize calls using the extension description?  for example, a user normally calls with the url, sip:1234@10.0.0.3, Is it possible to call: sip:husoft@10.0.0.3, sip:voicemail@10.0.0.3, etc...?
14:20.08*** join/#asterisk onweald_tim (n=onweald_@c-67-173-213-205.hsd1.tx.comcast.net)
14:21.32*** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net)
14:21.41Qb3rtrob0 thanks
14:22.42mostyhusoft: yes, via the dialplan
14:23.37HmmhesaysI love it when people ask the same question over and over again
14:23.41Hmmhesayseven though I answered it
14:23.47HmmhesaysHuSoft: yes
14:23.54trelane_Hmmhesays, it's called generating extra content :)
14:24.02mostyhusoft: use Goto to jump to a different context and Dial to dial a particular device
14:24.15trelane_Hmmhesays, it's called saying the same thing over and over again because the old grey matter upstairs died years ago
14:24.22trelane_;)
14:24.32HmmhesaysI have some friends like that, too many drugs in college
14:24.59HuSoftmosty, ok, thanks.
14:24.59*** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx)
14:25.04iqHi
14:25.34*** join/#asterisk tdonahue (n=tdonahue@207.138.151.58)
14:28.22*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
14:28.22*** mode/#asterisk [+o anthm] by ChanServ
14:28.31rene-hi, i am looking for steve totaro of totaro tech and asteriskhelpdesk, has anyone seen it? his web pages have disappeared this side of the internet? i am hoping someone knows his irc handle
14:29.19*** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com)
14:29.30*** join/#asterisk lorinc (n=ang@caracas-0996.adsl.interware.hu)
14:29.33*** part/#asterisk HuSoft (n=Hoo@227stb47.codetel.net.do)
14:29.46*** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net)
14:32.00*** part/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com)
14:33.02*** join/#asterisk cytrak (n=kvirc@adelphi.geofocus.com)
14:33.26cytrakis there a way to review the recorded grettings without having to always re-record them
14:35.42a1faAnybody know a provider that sells $1did + 1c/min?
14:36.06Nivexno, but I bet the wiki might
14:36.14a1facouldnt find it
14:36.25*** join/#asterisk tgrman (n=jcmoore@picard.ojc.nuvio.com)
14:37.39*** join/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com)
14:37.45trelane_do the zaptel modules support IRQ steering, I'm still working around an IRQ conflict on this damnable dell server
14:37.54*** part/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com)
14:39.40*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
14:39.52*** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br)
14:40.34cytraktrelane: check which devices are using IRQs and disable them on the BIOS
14:41.03a1famaybe he needs those
14:41.04cytraktrelane: that what I did on mine, I got no usb , serial, parralel , they are all disabled
14:41.20a1fatrelane_ : hey
14:41.25a1fajust switch irqs in bios
14:41.27a1faetc
14:42.00cytrakwell my bios didn't give me the option to actually assign irqs
14:42.05cytraki wish I had
14:42.07a1faupdate bios
14:42.18cytrakeven with that
14:43.02cytrakdo you know if there is a way to review the recorded grettings without having to always re-record them ?
14:43.08*** join/#asterisk nfi|ermes (n=nfi_erme@217.220.121.62)
14:43.15nfi|ermeshi all
14:43.22cytrakmy users bitch so much about all these little things
14:43.26*** join/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com)
14:44.02[TK]D-Fendercytrak : You mean VM announcements?
14:44.16trelane_alfa: been there done that, I'm suffering from braindead hardware
14:44.37cytrakyeah I guess so , I'm talking about the busy and unavailable messages that you record
14:44.44trelane_alfa: this dell likes moving the smbus with the zap card
14:45.11cytrak[TK]D-Fender: was that what you meant ?
14:45.35[TK]D-Fendercytrak : You could jsut make your own little dialplan script to listen to them
14:45.37a1fatrelane_: it doesnt have a jumper on it
14:45.55trelane_a1fa, no
14:46.12*** part/#asterisk bhearsum (n=bhearsum@durhamlug/pdpc.basic.bhearsum)
14:46.15a1fahehe
14:46.18a1fasorry dude
14:46.18trelane_it's a server board, I can steer IRQ's in bios but regardless of what I do the smbus controller follows the pci slot that the zap card's in
14:46.21a1fadisable something in bios
14:46.32trelane_everything that can be disabledi s
14:46.33a1fadisable that esd
14:46.37a1faor whatchmaqall it
14:47.04*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
14:47.08cytrak[TK]D-Fender: ok , so the voicemail app won't be able to
14:47.10*** join/#asterisk RoyK[at] (n=roy@80.109.196.173)
14:48.03cytrak[TK]D-Fender: I thought about doing that but then I know I'm gonna hear a lot of bitching about "oh we got call another extension to be able to do that .."
14:48.17a1fa[TK]D-Fender : you are the one who recomended that $1 did + 1c a minute
14:48.53[TK]D-Fendera1fa : nope.
14:48.58a1fayeah it was you
14:50.16*** join/#asterisk hohum (n=dcorbe@mail.interceltelecoms.com)
14:50.19a1fai no lie, no lie
14:50.21*** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com)
14:50.54[TK]D-Fendera1fa : Sorry, I don't do PSTN termination like that...l don't have anyone to recommend to anyone really, so couldn't be me.
14:51.16*** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.232.51.Dial1.SanJose1.Level3.net)
14:52.06*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)
14:52.19*** join/#asterisk ben_d (n=ben@cpe-66-67-129-4.rochester.res.rr.com)
14:53.07*** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.232.51.Dial1.SanJose1.Level3.net)
14:53.12nfi|ermesi would like to enable the caller to transfer the call, but this doesn t happen
14:53.31Hmmhesayswhat a fantastically vague statement
14:54.02nfi|ermeslol
14:55.40RoyK[at]~nickometer RoyK[at]
14:55.53*** part/#asterisk tdonahue (n=tdonahue@207.138.151.58)
14:55.58*** join/#asterisk tdonahue (n=tdonahue@207.138.151.58)
14:57.14knoboanyone knows how asterisk 1.0.7 works with isdn4linux?
14:58.16knoboor if it is better to use capi
14:58.52*** join/#asterisk DarKnesS_WolF (n=wolf@82.201.232.126)
14:59.20RoyK[at]at least
14:59.43*** join/#asterisk Skarmeth (n=Skarmeth@201009012196.user.veloxzone.com.br)
15:00.40*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
15:00.47RoyK[at]knobo: any particular reason you want to use 1.0.7? or isdn4linux?
15:01.19*** part/#asterisk RoyK[at] (n=roy@80.109.196.173)
15:01.21knoboRoyK[at]: People force me to use debian stable
15:01.33*** join/#asterisk RoyK[at] (n=roy@chello080109196173.3.graz.surfer.at)
15:01.45RoyK[at]knobo: any particular reason you want to use 1.0.7? or isdn4linux?
15:02.10RoyK[at]knobo: 1.0.7 is quite old, and i4l is perhaps some of the worst shite I've ever touched
15:02.12*** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com)
15:02.20RoyK[at]use bristuff or visdn instead. perhaps capi
15:02.29*** join/#asterisk salviadud (n=ralfalfa@201.135.2.210)
15:02.48knobowell, i4l is the the only way to get the eicon Diva 2.02 card going
15:02.56knoboand 1.0.7 is in debian stable
15:02.58RoyK[at]then get another card
15:03.26knobowhich one do you recomend?
15:03.53RoyK[at]http://www.komplett.no/k/ki.asp?sku=119006
15:03.55RoyK[at]that one
15:04.06RoyK[at]works well with bristuff and visdn
15:04.26RoyK[at]i've heard some people have even managed to make it work with capi, but I don't have any idea how
15:07.11RoyK[at]you can also use more of them in a system, but they generate a bloody storm of interrupts, so it might not be so good
15:07.28*** join/#asterisk nf1 (n=nf1@vpn-pppoe-213-240-242-81.megalan.bg)
15:07.30RoyK[at]two or three should work, though
15:08.35*** join/#asterisk smackus2 (n=smackus2@c-67-169-248-217.hsd1.ut.comcast.net)
15:09.38smackus2has anyone had experience with an e&m winkstart T1 em_w? I have set one up, but it does not show caller ID. Is there some other setting that has to be used to send it? I can get it to work with my PRI's
15:15.12*** join/#asterisk marv0997 (i=marv0997@190.4.2.83)
15:15.26*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
15:17.44*** join/#asterisk Synthe (i=Synthe@odo.synthe.net)
15:18.10*** join/#asterisk ben_d (n=ben@cpe-66-67-129-4.rochester.res.rr.com)
15:24.30*** join/#asterisk PakiPenguin (n=Junaid@linuxpakistan/admin/pakipenguin)
15:25.19trelane_are digium's offices closed? the pbx is being evil and noone's available at the operator extensions
15:25.44Hmmhesaysbeing evil?
15:25.49Hmmhesaysfeed it some children
15:25.59trelane_Hmmhesays, I would but it's in atlanta iirc
15:26.06trelane_that would require driving
15:26.09Hmmhesayswhat type of pbx?
15:26.20trelane_I'd hope they're using asterisk
15:26.26Skarmethhi all
15:26.32HmmhesaysHello
15:27.35malcolmdwe're effectively closed; there are only 3 people here today.
15:27.51SkarmethI am trying to set up a E1 ISDN PRI with Asterisk and TE110P, I can call mobile phones (local and long distance), I can call tollfree numbers (0800), I can call long distance
15:27.57Sonderbladehow long does it usually take for softphones to reregister to a server when you reboot that server?
15:28.00Skarmethbut not local 8 numbers
15:28.10HmmhesaysSonderblade: whatever the timeout is?
15:28.21Hmmhesayslooks like a DP problem Skarmeth
15:28.31SonderbladeHmmhesays: that is why i said *usually*....
15:28.43Hmmhesaysthere is no "Usually"
15:28.54Hmmhesayseach endpoint is going to likely have a different timeout
15:29.28SonderbladeSkarmeth: you need to add the area code to local numbers
15:29.31Hmmhesaysif there is a specific softphone you are looking for, just download it and look at the default
15:29.33Qb3rtSonderblade: let say after 30minutes if it is not registering you have a problem
15:29.51SonderbladeQb3rt: thanks
15:30.12Hmmhesaysheh
15:30.30HmmhesaysI see what I get for giving decent advice, instead of a generic answer
15:31.54*** part/#asterisk mosty (i=mostynm@60-241-198-194.static.tpgi.com.au)
15:32.12rob0I have children and am in driving range from Digium!
15:32.41Hmmhesays30 minutes, I don't think I've ever seen an endpoint have 1800 seconds for a registration time out
15:33.03Hmmhesaysyou see a lot of 60, 180 and 3600's
15:33.54NotJohnDavidsalviadud: i got the SPA3k working
15:34.15*** join/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net)
15:34.28Qb3rtHmmhesays: yeah! true
15:34.31SpaceBassanyone using linksys phones? Anyone get auto answer working
15:34.50Hmmhesaysmodel number FOO!
15:35.03Hmmhesayshaha
15:35.11SplasPoodDoes anyone know if it's possible to use RealTime to store meetme.conf ?
15:35.13SpaceBassSPA-941 and 942
15:35.45Qb3rtbye A+
15:36.36salviadudNotJohnDavid, how you do it?
15:38.29HmmhesaysSpaceBass: you still just messing around with voip at home?
15:38.54NotJohnDavidsalviadud: posted to a forum and someone explained it to me.  I knew that the SPA3k handled PSTN like Voip.  the way it does pass thru is encode the PSTN and then place a call to the FXS
15:39.11SkarmethSonderblade, not here in brazil
15:39.15SpaceBasspretty much
15:39.28salviadudNotJohnDavid, so what did you tweak at the end?
15:39.31NotJohnDavidsalviadud: what I didn't know that the only codec that it can use is g711.  If you set Line1 to something else it just doesn't work.  doesn't warn you about it... no documentation anywhere
15:39.44SpaceBassgot some voip related projects for work up my sleeve...but I have to sneak that stuff in
15:40.02*** join/#asterisk _problem_ (n=lokesh_k@estrela.nortenet.pt)
15:40.10NotJohnDavidslaviadud: so for passthru to work both PSTN and Line1 have to be set to g711.  AND when I emailed tech support (Sipura/linksys) they didn't mention this in the 4-5 emails that were exchanged
15:41.00SkarmethSonderblade, may be something related to pri dial plan
15:41.45NotJohnDavidsalviadud: thanks for the help.  I just think it comes down to poor documentation on their part
15:42.00HmmhesaysSpaceBass why is that?
15:42.24HmmhesaysSend them my way, I'm in between projects at the moment
15:42.42SpaceBassim not in IT at all....its hard to relate VoIP to healthcare :)
15:43.17Hmmhesayswhat do you do in healthcare?
15:44.03SpaceBasswe have a TON of conference calls...I'm on like 5 a day...so I'd like to have a * box recieve the e-mail invite for the conf call, schedule a job that dials into the call and records it, then converts to MP3 and uploads to a sharepoint site
15:44.31HmmhesaysThat can be arranged
15:44.41Hmmhesaysimap or pop3?
15:44.41SpaceBassI'm a revenue cycle consultant...fancy way of saying I help hospitals with patient processing, registrations, billing, etc
15:44.43salviadudI hate linksys support
15:44.49SpaceBasssalviadud, oxymoron
15:45.04SpaceBasslinksys doesnt support anything...they just put you on hold and transfer you to india
15:45.14SpaceBassHmmhesays, imap
15:45.22HmmhesaysThat can be arranged
15:45.47*** join/#asterisk eKo1 (n=bernd@190.4.7.90)
15:45.51HmmhesaysI have an asterisk box that retrieves email and plays mp3's based on subject line
15:45.55Hmmhesaysits not a far leap to what you want
15:45.55SpaceBassi was working on it...got hung up trying to get postfix to check the mail and parse it
15:46.03SpaceBassnot far at all
15:46.19Hmmhesaysits pop3 but the switch to imap is trivial
15:46.47HmmhesaysI can get it done for you for a bit of monetary compensation
15:46.51SpaceBassthese are all outlook/exchange invites... we'd (company) have to establish a standard for putting the phone number and access code into the meeting request
15:47.19Hmmhesaysyou could have a script parse the email easily enough
15:47.27Hmmhesaysor have a form to autogenerate the email
15:47.35Hmmhesaysa web form type deal
15:47.56*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
15:47.56*** mode/#asterisk [+o anthm] by ChanServ
15:48.22SpaceBasswell, its got to be seamless....my co-workers arnt going to use outlook and a web form...im the only office geek :)
15:49.01a1falool
15:49.05Hmmhesaysthen they'd have to put the access number and time into the email
15:49.14Hmmhesaysyou could have a script parse just bout any format they could think of
15:49.28Hmmhesaysthat would be no big deal
15:49.46SpaceBasstypically we put the number and access code into the location field in the metting request
15:50.01SpaceBasswhen you get one via imap or pop (IE not outlook) all of that is there in cleartext
15:50.27SpaceBassand if you can fix my site-to-site VPN while you are at it..... :)
15:50.44Hmmhesayswhat vpn?
15:50.59HmmhesaysSpaceBass that would be cake to pull info out of that email
15:51.06SpaceBassusing freeswan via IPcop
15:51.18Hmmhesayswhat are your two vpn endpoints?
15:51.58SpaceBassboth are IPcop routers...the problem is one endpoint gets a private IP for its WAN address (but has a Public IP that points to it)
15:52.25HmmhesaysI would use a couple wrt routers with openvpn
15:52.27SpaceBassand Im going to build a freenas box today...if I can get out of this hangover :)
15:52.38SpaceBassI think OpenVPN probably works a lot better
15:52.42Hmmhesaysprobably?
15:52.52SpaceBasswell, I havent played with it to confirm :)
15:52.55*** part/#asterisk NotJohnDavid (i=dave@c-68-47-199-178.hsd1.tn.comcast.net)
15:53.11HmmhesaysI run all kinds of network traffic over openvpn
15:53.16Hmmhesaysit...just...rocks
15:53.34SpaceBassI need to play with it
15:54.08Hmmhesayswell the fact that you can run it on an $80 router and have it be stable..
15:54.09SpaceBassto access my home network I use windows 2003 as the VPN server...works fine...but for this net-to-net that I'm trying to set up OpenVPN might be the way to go
15:54.22*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
15:54.31Hmmhesaysyes, it would be
15:55.01*** join/#asterisk ben_d (n=ben@cpe-66-67-129-4.rochester.res.rr.com)
15:55.50*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
15:55.53Hmmhesaysopenvpn has some kickass features too
15:56.01[TK]D-FenderHmmhesays : OpenWRT?
15:56.05SpaceBassbut I'm pretty partial to IPcop....although I think Endian Firewall (which is an IPcop clone) uses OpenVPN
15:56.12Hmmhesaysi've used it on sat connections before that have a really small MTU
15:56.29Hmmhesaysyou can fragment the packets and send them across
15:56.36Hmmhesays[TK]D-Fender: yeah
15:57.14SpaceBasstrying to figure out the most cost effective way to do this FreeNAS box.... a few 300gb drives mirrored, or several 150gb drives spanned and mirrored
15:57.35Hmmhesayswhat are you trying to accomplish?
15:57.58SpaceBassmassive and redundant storage :)
15:58.14jbalcombSpaceBass Raid 10 (one-zero)
15:58.32[TK]D-FenderHmmhesays : I'd think that a small MTU would absolutely kill Sat because of latency and packet overhead....
15:59.11SpaceBassVPN over a sat connection sucks!
15:59.47Hmmhesays[TK]D-Fender: its a guess and check type deal
15:59.54Hmmhesaysand SpaceBass: not really
16:00.54SpaceBassmaybe its just that satellite connections suck in general :)
16:01.04Hmmhesaysmostly yeah
16:01.30Hmmhesaysbut when you're in a grass hut and you have phone service where you used to use that dude down the road that could run fast...
16:01.51SpaceBasstrue
16:02.03SpaceBasswe have a family farm and sat is all we can get there...and it does beat dialup
16:02.15SpaceBassbut its very expensive and still not that fast...
16:02.29Hmmhesaysand you don't need use binoculars and hope the hut down the roads window is open to see boobies
16:02.34Hmmhesayssatellite is pretty damn nice
16:02.41SpaceBasslol
16:04.43*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
16:05.03salviadudi have pansat
16:05.10salviadudi wish they had a fix...
16:05.11HmmhesaysSo anyhoo SpaceBass, if you want some help on that project let me know. I'm going to be available for awhile
16:06.36SpaceBassI will!
16:07.02SpaceBassI'm not working today...had to fedex my laptop to Ga to be reimaged...but when I get it back thats on my list of stuff to paly with
16:07.32Hmmhesaysyou use googletalk at all?
16:07.47*** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net)
16:07.56SpaceBassi had it loaded on my XP box...but haven't played with it in a while
16:09.11*** join/#asterisk neoXite (n=neoxite@xdsl-87-78-77-206.netcologne.de)
16:09.41Hmmhesaysahh ok, I have mine going most of the time
16:09.46Hmmhesayspretty kickass messenger
16:09.51neoXitehi, i have problems building asterisk 1.2.9.1 on dapper drake
16:10.05*** join/#asterisk Bert- (n=bert@LAubervilliers-151-12-81-84.w193-252.abo.wanadoo.fr)
16:10.14Bert-hello again :)
16:10.24*** join/#asterisk smacku1 (n=smackus@c-67-169-248-217.hsd1.ut.comcast.net)
16:10.33neoXitei installed all the necessary packages (build-dep from version 1.2.7.1) but gcc gives me an error during building
16:10.37Bert-can I run Idefisk on a windows CE plateform please ?
16:10.48Bert-or only on windows XP/2K ?
16:10.51Hmmhesaysis there a build for windows CE?
16:10.54neoXiteanyone here experienced similar problems?
16:11.21HmmhesaysI've had gcc errors before, however if you don't tell us the error then your question will never be answered
16:11.38Bert-what is the gcc error ?
16:11.52neoXitejust a sec, i had it copied somewhere
16:11.59smacku1I am using inbound digits from an 800 number to route calls, and I am capturing CDR Data into mysql database. But I am not finding any data to identify those calls by incoming digits. does cdr trap incoming digits? if so, by what name?
16:12.00Hmmhesaysthere is an echo in here, lol
16:12.29SpaceBassHmmhesays, it is great...unfortunatly I dont know anyone else using it :) .... aslo on a mac most of the time
16:12.39Hmmhesaysahh ok
16:12.43SpaceBassI know iChat can connect to google talk...just been lazy about it
16:12.58Hmmhesaysit uses jabber protocol
16:13.33Bert-hey it's 6 p.m !! I'm still at work
16:13.36Bert-i'm crazy ... :)
16:13.46Bert-see ya here ;)
16:14.01smacku1does the cdr not store dnis?
16:14.59eKo1smacku1: not by default
16:15.12smacku1how can I change that?
16:16.19neoXiteokay, should i paste in here or is that against channel rules?
16:16.25eKo1Hack the cdr_* module that you're using.
16:16.35Hmmhesaysheh you *could* do that
16:16.40Hmmhesaysor use custom cdrs
16:17.02smacku1ok, I am even newer to databases than I am asterisk... is there a newbie answer?
16:17.07*** join/#asterisk Ludo_ (n=Ludo@obelix.zoxx.net)
16:17.16eKo1I've never used custom cdrs...
16:17.31Hmmhesayspay me money and me give you answer
16:17.32Hmmhesayslol
16:17.42neoXitegcc says 'chan_zap.c:9160: error: ‘pri_event_setup_ack’ has no member named ‘call’'
16:18.16neoXitei tried building the same source on debian/testing and it compiled fine
16:18.41smacku1I am open to that... but with a different approach. I have offered others payment for work. I need a quote. The end result is I need to trap dnis in my cdr, how much for you to do it?
16:18.54neoXitebut on ubuntu/dapper it fails with that error
16:19.05Hmmhesayswhere is the call coming from?
16:19.21smacku1i have 10 different 800 numbers coming in from all over.
16:19.28smacku1sending the last 4 digits as dnis
16:19.42smacku1i need to bill companies based on the 800 number dialed
16:19.50Hmmhesaysi'm guessing if you're a n00b to this you're probably using asterisk@home distro?
16:19.56smacku1nope
16:20.07smacku1this is a 80+ agent call center
16:21.10Hmmhesayswhat shows up in the SRC field of your CDR?
16:21.34smacku1hang on... gotta connect to vpn to get it
16:22.00*** join/#asterisk Qwell[] (i=north@unaffiliated/qwell)
16:22.52Ludo_Hi, I have one asterisk PABX with one public ip, if I connect directly to this server from outside or inside of the network it's ok
16:23.11Ludo_now I would like try to connect to this server in using a client behind a nat
16:23.22Ludo_do I need a stun server or not?
16:23.24Ludo_to do it?
16:23.37Hmmhesaysnot
16:24.47Ludo_ok just to activate nat=yes in my sip.conf?
16:25.56Hmmhesaysworth a shot
16:26.07*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
16:26.24Ludo_what do you mean?
16:27.57*** part/#asterisk smacku1 (n=smackus@c-67-169-248-217.hsd1.ut.comcast.net)
16:28.03*** join/#asterisk smacku1 (n=smackus@c-67-169-248-217.hsd1.ut.comcast.net)
16:28.15*** join/#asterisk muppetmaster (n=jasongoe@169.red-81-184-73.user.auna.net)
16:28.20muppetmasterHello
16:28.26*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
16:28.37muppetmasterHas something changed with the codec disallow/allow in terms of ulaw/alaw as of v1.2.9.1?
16:29.10muppetmasterI have disallow=all and allow=ulaw, and yet when I call with Xlite with G711u enabled on its side I get a notification that no codecs are available.
16:29.23RoyK[at]try alaw
16:30.01muppetmasterOkay
16:30.16muppetmasterI did, but will do again to be sure
16:30.19*** part/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx)
16:30.20muppetmasterAs it had the same problem
16:31.21Qwell[]Corydon-w: ;)
16:31.39*** join/#asterisk Gamercjm (n=chris@pool-71-254-175-66.lsanca.fios.verizon.net)
16:31.44Corydon-w:-P
16:33.12Corydon-wQwell[]: 7 days
16:33.28Qwell[]6
16:33.36Qwell[]until I get there, anyhow
16:33.39RoyK[at]knobo: ping
16:33.41RoyK[at]~ping
16:33.43jbotpong
16:33.43Corydon-wQwell[]: you're coming to Nashville on Sunday?
16:33.47RoyK[at]muppetmaster: i'm running x-lite with alaw
16:33.50RoyK[at]works splendidly
16:33.50Qwell[]Corydon-w: nope, HSV
16:33.53RoyK[at]and we run 1.2.9.1 with several thousand customers
16:33.55RoyK[at]so if alaw didn't work, our customer support centre would kill me
16:33.57RoyK[at]:)
16:34.05Qwell[]Corydon-w: You'll have to drive up, to harrass me :P
16:34.10Qwell[]or..down, as it were
16:34.12*** join/#asterisk BugKham (i=CKGLOB@125.24.0.136)
16:34.15Corydon-wQwell[]: I meant until you'd have me to spoon
16:34.18Qwell[]:P
16:34.19*** part/#asterisk BugKham (i=CKGLOB@125.24.0.136)
16:34.19jbalcombmuppetmaster: i'm running x-lite with ulaw
16:34.27muppetmasterNot sure why it is not working...
16:34.29jbalcombworks splendidly
16:35.08jbalcombanyone using gnophone?
16:35.18muppetmasterulaw nor alaw is working
16:35.29muppetmasterIf I do an allow all, then it works
16:35.41jbalcombmuppetmaster what codec does it use when it works?
16:35.46muppetmasterulaw
16:36.04jbalcombwhats the trouble then really?
16:36.16muppetmasterThe fact that I want to select that it only uses that codec
16:36.19jbalcombgranted its queer but disallowing codecs isnt the biggest issue ever
16:36.27muppetmasterWhen I try that, then it tells me no go, as no supported codec is available
16:36.49Qwell[]in which order are you allowing/disallowing codecs?
16:36.52jbalcombits like worrying about whether the TP unrolls from the front or the back
16:36.58muppetmasterdisallow=all
16:37.00muppetmasterallow=ulaw
16:37.03Qwell[]jbalcomb: sorry, but that DOES matter
16:37.13jbalcomb=)
16:37.17*** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx)
16:37.28muppetmasterThen get:  Jul 3 18:15:56 NOTICE[13210]: chan_sip.c:3691 process_sdp: No compatible codecs!
16:37.29jbalcombQwell[]: I prefer over-the-top
16:37.34Qwell[]as you should
16:37.55jbalcombQwell[]: It only makes sense. ;)
16:38.18neoXiteok nevermind, fixed it. my zaptel-source package apparently was too old
16:38.30Qwell[]I could go into detail on why, but it's incredibly offtopic, and disturbing :P
16:38.59RoyK[at]muppetmaster: as i said, i use disallow=all,allow=alaw and it works with all clients i've tried
16:39.11muppetmasterRoyK[at] I have no idea what is going on here.
16:39.14muppetmasterVery strange
16:39.23muppetmasterI have used it before no problem, but puking as you can see
16:39.26RoyK[at]what does sip debug say?
16:39.34*** join/#asterisk Eggplant (i=No@dsl-216-155-213-228.cascadeaccess.com)
16:39.47muppetmasterRoyK[at] Too much traffic at the moment to run a debug, need to isolate it
16:39.52RoyK[at]pastebin your config, then
16:39.52RoyK[at]and debug output
16:39.52RoyK[at]s/debug/verbose/
16:39.56jbalcombmuppetmaster: run the command in asterisk and see what codecs it think are available. Anyone know what that command is?
16:40.06muppetmastershow codecs
16:40.06RoyK[at]sip debug peer xxxxx
16:40.14RoyK[at]muppetmaster: won't help much
16:40.37RoyK[at]muppetmaster: show translation will show what codecs are actually loaded
16:40.46jbalcombmuppetmaster: too much traffic for debug? HA! I run debug with 46 zap channels and 150 sip users at midday on wednesday1
16:40.54RoyK[at]shite. this internet connection is bad.....
16:40.56RoyK[at]100 packets transmitted, 76 packets received, 24% packet loss
16:40.56RoyK[at]round-trip min/avg/max/stddev = 59.958/121.233/880.028/107.601 ms
16:41.08muppetmastershows ulaw and alaw both present and thea bility to translate
16:41.21RoyK[at]then it's a config problem. pastebin your config and verbose output
16:41.23RoyK[at]~pb
16:41.24jbotit has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/
16:41.38RoyK[at]if you can't isolate it, setup another box and try there
16:41.46jbalcombthe jbot entry for pb should be shorter
16:42.08RoyK[at]~jbalcomb
16:42.27jbalcomb~TP RoyK[at]
16:42.50RoyK[at]jbot: tp?
16:42.51jbot[tp] I AM THE GREAT CORNHOLIO! I NEED TP FOR MY BUNGHOLE!
16:42.52jbalcombhaha.. dude, i just threw toilet paper all over you, remotely.
16:43.10jbalcombFeel the pang.
16:43.16jbalcomb(tm)
16:43.31jbalcombjbot: yermom?
16:43.45*** part/#asterisk smacku1 (n=smackus@c-67-169-248-217.hsd1.ut.comcast.net)
16:43.49neoXitejbot: orly?
16:43.50jbotYARLY
16:44.00jbalcombjbot: wtf?
16:44.01jbotwho?
16:44.23jbalcombjbot: Zombie Jesus?
16:44.24jbotACTION summons an army of the undead to eat Jesus
16:44.32jbalcomboh damn....
16:44.45RoyK[at]:)
16:44.58RoyK[at]not too much left, though
16:45.21skraelings001what does this mean ?  Jul  3 10:51:53 NOTICE[13638] channel.c: Dropping incompatible voice frame on Local/202@INTERNACIONAL-5175,2 of format alaw since our native format has changed to slin
16:46.24RoyK[at]skraelings001: 1.2?
16:46.43skraelings001yes
16:46.54RoyK[at]what codec?
16:48.38skraelings001alaw | ulaw | gsm
16:48.50RoyK[at]skretry disallow=all, then allow=alaw, and try again
16:49.09RoyK[at]~codecs
16:49.10jbotextra, extra, read all about it, codecs is http://snipurl.com/wiki_codecs
16:49.10RoyK[at]~codec
16:49.12jbothmm... codecs is http://snipurl.com/wiki_codecs
16:51.38RoyK[at]jbot: codecs is also If you have audio/codec problems, first try to 'disallow=all' and 'allow=all' and see if that works
16:51.40jbotokay, RoyK[at]
16:53.24*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)
16:54.23RoyK[at]~codecs
16:54.25jbot[codecs] http://snipurl.com/wiki_codecs.  If you have audio/codec problems, first try to 'disallow=all' and 'allow=all' and see if that works
16:54.30RoyK[at]~codec
16:54.31jboti heard codecs is http://snipurl.com/wiki_codecs.  If you have audio/codec problems, first try to 'disallow=all' and 'allow=all' and see if that works
16:56.05*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
16:56.48*** join/#asterisk Bullseye_Network (n=info@216.143.192.69)
16:58.34*** join/#asterisk HuSoft (n=apo@227stb47.codetel.net.do)
16:58.42*** join/#asterisk ToyMan (n=stuq@74-32-9-135.dsl1.mdl.ny.frontiernet.net)
16:59.49skraelings001RoyK[at] : i found that is related with MoH, seems can't get back to previous format
17:03.22HuSoftIs this the way to support both num/alpha dialing, e.x.: sip:husoft@localhost ?:   exten => husoft, 1, dial(SIP/1234)
17:04.55*** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net)
17:06.48*** part/#asterisk HuSoft (n=apo@227stb47.codetel.net.do)
17:08.16Ludo_I have nat=yes activated, I try to call my asterisk server from a sip client behind nat, I see in log some activity, but I don't heard nothing on my sip client
17:08.28Ludo_my sip client is behind a fon router
17:08.57RoyK[at]skraelings001: what sort of MoH format?
17:09.00skraelings001RoyK[at] : having diferents formats for same file seems to be the answer and also low-down cpu usage
17:09.16skraelings001mp3
17:09.24Ludo_do you think it's the fon router the problem?
17:09.33Ludo_I get a 408 timeout
17:10.29SpaceBassanyone useing the linksys phones (SPA-941 and 942) gotten auto answer to work?
17:11.06jbalcombSpaceBass I have three but i dunno if the auto answer works, sorry.
17:13.10*** join/#asterisk Tsop (n=Tsop@69.155.81.24)
17:13.41jbalcombSpaceBass: ours do not auto-answer.
17:13.45Tsopgood morning all
17:15.26SpaceBasshummm
17:17.41Tsopi am new to linux world, i have just finish installing debian and i already love it ;), i would like to install asterisk and use it at home just for my personal use? do i need to subscribe to any service or this is compeletly free?
17:18.14Qwell[]Tsop: It's free as in beer (and libre), for any usage
17:18.22Qwell[]feel free to use it in a corp, or whatever
17:18.28Qwell[]just...I suggest you read the GPL
17:19.37TsopQwell, so with just my dsl connection and my linux box i will be able to make phone calls?
17:19.45Qwell[]not quite
17:19.59Tsopwhat do i need to get started
17:20.08Qwell[]~docs
17:20.09jbotfrom memory, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
17:20.11Qwell[]~wikis
17:20.13jbotrumour has it, wikis is http://www.voip-info.org
17:20.13*** join/#asterisk Sponge_bob (n=None@cpe-66-27-162-13.socal.res.rr.com)
17:21.07Tsopi had a linksys RT31P2 but i couldn't get it to work , they Vonage told me it's lock to them and i called linksys and they also told me i'll have to call vonage
17:21.08*** join/#asterisk syle (n=blah@unaffiliated/syle)
17:21.14Sponge_bobwhat's the best way to test out a meetme room for echo?
17:21.25Qwell[]Sponge_bob: call it, and listen for echo
17:21.35Sponge_bobi need 23 calls
17:21.41Qwell[]call it 23 times?
17:21.52Qwell[]or use a call file for 22 of them
17:22.32*** join/#asterisk file2 (n=file@out.clearnet.com)
17:22.39Qwell[]eeps
17:22.41Sponge_bobwell i found that with 2-5 users the calls seems to be ok.  if i get a couple more some hear their own voice
17:22.45file2moon
17:22.50file2er mooo
17:23.22skraelings001Can anyone shed light on what these warnings mean >channel.c: Avoided initial deadlock for '0x829c350', 10 retries!<, how they can be avoided, and if they are something to worry about?
17:23.57Sponge_bobQwell[]: how can i test a meetme with a call file?
17:24.19file2magic
17:24.40*** join/#asterisk argos73 (n=mike@w010.z208036240.chi-il.dsl.cnc.net)
17:24.40Sponge_bobtell me how the magic works
17:25.01TsopQwell,  thanks, i know i'll def. need to read , but as i told u i'm a n00be, can u just point me to a link that you'll give to someone that know nothing about asterisk and want to set it up?
17:26.34file2ooh nice outside
17:26.48Qwell[]file2: sorry, you aren't allowed outside
17:27.15Qwell[]close your window too...no peeking
17:27.18file2why not :(
17:27.25Qwell[]because I'm mean
17:27.33file2ah ic
17:27.41RoyK[at]~rtfs
17:27.42jboti guess rtfs is probably read the f*cking source...
17:27.42file2makes sense
17:28.25*** join/#asterisk Ellegon (n=sbryant@72.164.50.72)
17:28.54*** join/#asterisk evisu (n=hIRC@bzq-88-152-238-38.red.bezeqint.net)
17:29.48SkarmethSonderblade, I am back. I was trying to call a number of my E1 and it was sucessful (I can see that the call goes out), but the outgoing and the inconming channels was hang up. I have a TE110P on IRQ 24 and a TDM04B on IRQ 25... it make any sense?
17:29.55trelane_is there a document somewhere listing options to pass to insmod on zap modules
17:30.00Ellegonquick question... I have the Zap dev kit. When ever I try to make a call from Zap/1 (FXS) to Zap/4 (FXO) I can hear the person on the call fine but they complain that I am muffaled
17:30.15Ellegonthis problem is not there if I call from one of my pap2's
17:30.21Ellegonto Zap/4
17:31.15EllegonAlso the call is fine from Zap/1 to any port on the Pap'2
17:33.40*** part/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net)
17:38.15*** join/#asterisk donpaolo (n=donpaolo@pri-214-b7.codetel.net.do)
17:39.06Tsopi don't know where to get this information please?
17:39.46donpaoloHi guys! I get this error when calling through mutualphone:  WARNING[5015]: channel.c:2691 ast_channel_make_compatible: No path to translate from SIP/mutualphone-5e3b(256) to SIP/oficinamision-a47a(4)
17:39.48Tsopi need to know what do i need in addition of installing asterisk to make phone calls , do i have to sign with a voip provider
17:40.03donpaoloand then: WARNING[5015]: channel.c:2691 ast_channel_make_compatible: No path to translate from SIP/oficinamision-a47a(4) to SIP/mutualphone-5e3b(256)
17:40.03donpaoloJul  3 13:38:25 WARNING[5015]: app_dial.c:1572 dial_exec_full: Had to drop call because I couldn't make SIP/oficinamision-a47a compatible with SIP/mutualphone-5e3b
17:40.22file2can not transcode between g729 and ulaw
17:40.48donpaoloWhat have I to change? I followed the instructions on mutualphone site
17:40.56RoyK[at]perhaps you haven't bought any g.729 codecs.....
17:41.12file2you have to buy a license if you want to transcode
17:41.28RoyK[at]$10 per concurrent transcode
17:41.31RoyK[at]iirc
17:42.07*** join/#asterisk smacku1 (n=smackus@c-67-169-248-217.hsd1.ut.comcast.net)
17:42.17donpaolofile, are you speaking to me?
17:42.26file2yes
17:43.03smacku1is there any way for a queue once a call has timed out and been delivered to another extension to be able to detect that the other extension did not answer and then transfer the call to a different extension?
17:43.29RoyK[at]donpaolo: yes, it's $10 per concurrent transcode
17:44.05file2so nice outside
17:44.13donpaoloRoyK[at], do you mean that I can't communicate with them without buying a g.729 license?
17:45.04*** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka)
17:46.37*** part/#asterisk tgrman (n=jcmoore@picard.ojc.nuvio.com)
17:46.45RoyK[at]donpaolo: yes
17:46.52RoyK[at]donpaolo: except in passthrough
17:46.57donpaoloRoyK[at], how?
17:47.02RoyK[at]how what?
17:47.05RoyK[at]how to buy?
17:47.11RoyK[at]or how to use passthrough?
17:47.50*** join/#asterisk jcmoore (n=jcmoore@picard.ojc.nuvio.com)
17:47.54donpaolohow to use pasthrough
17:48.19RoyK[at]just setup client a to use g.729, client b to use g.729, disallow=all,  allow=g729
17:48.44RoyK[at]asterisk doesn't understand much of the audio, and thus cannot playback() or playtones() or anything, but the users can communicate
17:49.23RoyK[at]trelane: man modinfo :)
17:49.54donpaoloRoyK[at], I set up * disallow=all allow=g729, but mutualphone gives me those error I put above...
17:50.53RoyK[at]excactly what are you trying to do?
17:51.14*** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie)
17:52.48Ellegonquick question... I have the Zap dev kit. When ever I try to make a call from Zap/1 (FXS) to Zap/4 (FXO) I can hear the person on the call fine but they complain that I am muffled. this problem is not there if I call from one of my pap2's
17:52.53smacku1if I wanted to replace time: GotoIfTime(9:00-17:00| with *, would it be GotoIfTime(*| or GotoIfTime(*-*|?
17:53.00EllegonAlso the call is fine from Zap/1 to any port on the Pap'2
17:54.03donpaoloRoyK[at], I want to connect to mutualphone, and I set up sip.conf according to their instructions in http://www.mutualphone.com/asterisk.htm , I can register with them but I can't place a call, I get errors: "ast_channel_make_compatible: No path to translate from SIP/mutualphone-01fa(256) to SIP/oficinamision-fbf6(4)" - "chan_sip.c:2542 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 256/256)" - "chann
17:54.03donpaoloel.c:2691 ast_channel_make_compatible: No path to translate from SIP/oficinamision-fbf6(4) to SIP/mutualphone-01fa(256)" - "app_dial.c:1572 dial_exec_full: Had to drop call because I couldn't make SIP/oficinamision-fbf6 compatible with SIP/mutualphone-01fa"
17:54.37donpaolosmacku1, the 1st
17:54.47smacku1thank you much
17:55.59RoyK[at]donpaolo: then asterisk is trying to transcode, and not to do passthrough
17:56.04RoyK[at]dunno why
17:57.29*** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie)
17:58.24*** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net)
17:58.24TripleFFFF: hey guys.. got something weird.. got a pap2 client.. his calls drop every 300 seconds.. no firewall , hes dmz'ed.. im wondering if asterisk could be thinking that call nevr asnwered and still ringing and drops after 300 ?
17:59.17*** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie)
17:59.20TripleFFFFanyone ?
18:00.47cypromisnone
18:01.02dlynes_homeTripleFFFF: 300 seconds every single time?
18:01.05TripleFFFFnonewhat ?
18:01.06TripleFFFFyeah
18:01.09TripleFFFFwell 300 after pickup
18:01.15TripleFFFFso 301 to 325
18:01.20TripleFFFFdepending on how long it rang
18:01.25TripleFFFFbut its that im sure
18:01.31TripleFFFFcant be always 300 sec
18:01.40TripleFFFFand got people doing 3000-4000 secs no prob
18:01.46TripleFFFFits the linksyspap2t na im sure
18:01.53dlynes_homeTripleFFFF: is he on dialup, or broadband?
18:01.58TripleFFFFbroadband
18:02.04*** join/#asterisk |marv0997| (i=marv0997@190.4.2.83)
18:02.16dlynes_homeTripleFFFF: ppoe, or regular?
18:02.28TripleFFFFcable
18:02.28*** join/#asterisk dec_ (n=tom@ppp206-151.lns1.adl2.internode.on.net)
18:02.38dlynes_homeis it ppoe, though?
18:02.41dlynes_homeerm
18:02.47dlynes_homepppoe i mean?
18:02.50TripleFFFFno
18:03.09RoyK[at]pppoipoatmoatoe
18:03.28dlynes_homepoint-to-point-protocol-over-ethernet
18:03.54TripleFFFFvonage is ok
18:03.58TripleFFFFanyother is ok
18:03.59TripleFFFFjust me
18:04.00TripleFFFFlol
18:04.05dlynes_homeah, ok
18:04.06TripleFFFFbut using the linksys pap2t na,
18:04.34*** part/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)
18:04.39dlynes_homeis he the only one you're connecting to in the same manner?
18:04.52donpaoloRoyK[at], increible, I can connect to mutualphone with gsm codec!
18:04.53TripleFFFFno
18:05.21dlynes_homeTripleFFFF: is he the only one that you have that's not portmapped?
18:05.36TripleFFFFwhat you mean portmapped ?
18:05.49TripleFFFFhe just tried to dmz since it was making no sense
18:06.16TripleFFFFits the 300 sec that worrying me
18:06.21dlynes_homeport forwarding udp port 5060 on your router to your machine
18:06.21TripleFFFFmakes me think its on my side
18:06.28TripleFFFFudp or tcp ?
18:06.38dlynes_home^^^^^^^^^^^^^^^^^^^^
18:06.41dlynes_homeread :)
18:06.56TripleFFFFi mean i tought was tcp
18:06.58EllegonIs there a way to make a Zap fxo port wait 2 rings before picking up?
18:07.01TripleFFFFbut hes DMZ so no matter
18:07.09TripleFFFFwait(2)
18:07.09dlynes_homeTripleFFFF: i'm asking about yoru side, not his
18:07.23TripleFFFF? my box is direct on net
18:07.30TripleFFFFim runnign asterisk 1.2.9.1
18:07.32dlynes_homeTripleFFFF: is it firewalled, though?
18:07.33TripleFFFFon wan
18:07.35TripleFFFFno
18:07.38TripleFFFFdirect
18:07.56dlynes_homeis it running in a vm on a windows boxen?
18:08.02TripleFFFFcents 4.2
18:08.04TripleFFFFcentos
18:08.14TripleFFFFdual xeons 2 giger
18:08.22TripleFFFFbasically as it should
18:08.23TripleFFFF;)
18:08.35dlynes_homeand when you installed it, and it asked about a firewall, you told it none, right?
18:09.10TripleFFFFno idea
18:09.15TripleFFFFbut iptables is off
18:09.24dlynes_hometype iptables -nL
18:09.38TripleFFFFbash: iptables: command not found
18:09.46dlynes_home/usr/sbin/iptables -nL
18:10.05TripleFFFF<PROTECTED>
18:10.19TripleFFFFno iptables in there
18:10.22dlynes_homewhat do you mean which iptables?  there is only one iptables
18:10.23TripleFFFFlocated none neither
18:10.33TripleFFFFlol
18:10.38Tsopi need to know what do i need in addition of installing asterisk to make phone calls , do i have to sign with a voip provider?
18:10.42TripleFFFF<PROTECTED>
18:10.57dlynes_homeTsop: either that, or have an analog line, and get a tdm card
18:11.10dlynes_homeTripleFFFF: i know that...I'm not stupid
18:11.12TripleFFFFhmm
18:11.13TripleFFFF<PROTECTED>
18:11.13TripleFFFFChain INPUT (policy ACCEPT)
18:11.13TripleFFFFtarget     prot opt source               destination
18:11.13TripleFFFF<PROTECTED>
18:11.13TripleFFFFChain FORWARD (policy ACCEPT)
18:11.14TripleFFFFtarget     prot opt source               destination
18:11.18TripleFFFFChain OUTPUT (policy ACCEPT)
18:11.20TripleFFFFtarget     prot opt source               destination
18:11.22salviadudpastebin
18:11.22TripleFFFFweird
18:11.25salviadudfor the love of god
18:11.25TripleFFFFsorry
18:11.26a1fagod damn you
18:11.28dlynes_homeTripleFFFF: but sbin directories are not usually in your path
18:11.35a1fafucking ah!!!!!!!!
18:11.40Tsoplol
18:11.45a1fawtf. paste bin you jack the fuck ass
18:11.58dlynes_homea1fa: dood...get it straight
18:12.02TripleFFFFa1fa sirry
18:12.02dlynes_homea1fa: it's triple fuck ass :p
18:12.21salviadudyeah, nique ta mere, pastebin
18:13.24Tsopdlynes_home, thank you, from my understanding if i got a tdm card , and used asterisk and called internationl will this be charged on my analog line?
18:13.30a1fasome croatian hax0r?
18:14.15dlynes_homeTripleFFFF: is he running a firewall on his end?
18:14.24TripleFFFFhes dmz';ed
18:14.28TripleFFFFdemilitarized
18:14.29dlynes_homeTsop: yes
18:14.40dlynes_homeTsop: doesn't matter...still going through a firewalll
18:14.50dlynes_homeTsop: erm...mistell
18:14.59dlynes_homeTripleFFFF: doesn't matter...still going through a firewall
18:15.00TripleFFFFtrough
18:15.08dlynes_homeTripleFFFF: is it a real firewall, or a router?
18:15.10TripleFFFFcould be a router table flush of something
18:15.33Tsopdlynes_home, so yes i will be charged? or it was a yes for TripleFFFF ?
18:15.40dlynes_homeTsop: that was a yes for you
18:16.10dlynes_homeTsop: assuming your outgoing calls were going out on analog, and not voip
18:16.21Tsopdlynes_home, i thought i could use asterisk to make free phone calls
18:16.34dlynes_homeTsop: not free, necessarily
18:16.39dlynes_homeTsop: more like inexpensive
18:16.56TripleFFFFlol
18:17.01Tsopdlynes_home, my friend is calling me from africa and he told me he doesn't pay anything using this system
18:17.16TripleFFFFcaus he hacked someone lol
18:17.27TripleFFFFnothing free in life.. even death .. gota pay the toll bro
18:17.28dlynes_homeTsop: until everyone starts using dundi and enum, and all calls go voip
18:17.34dlynes_homeTsop: voip will not be 100% free
18:17.49dlynes_homeTsop: he could be calling for free
18:18.02dlynes_homeTsop: if he's doing sip direct calling, and not dialing a pstn number
18:18.41Tsophe is using his pc and calling land #
18:19.05dlynes_homeTsop: then either he's bullshitting you, or he's stealing long distance from someone
18:19.24TripleFFFFalso dlynes_home.. hes call trough vonage are ok
18:19.25smacku1i am trying to make it so that when a queue times out it tries an extension not logged into the queue. If that extension does not answer, I want it to go to another extension's voicemail. Here is what I tried. It did not work. Can anyone tell me if this is possible, and if so how to adjust my dial plan? http://pastebin.ca/78012
18:19.28TripleFFFFso the router should be ok
18:19.29dlynes_homeTsop: or he's found one of those extremely rare free call places
18:19.29*** join/#asterisk file2 (n=IrcNet@out.clearnet.com)
18:20.02dlynes_homeTripleFFFF: he's using a pap2, or a pap2-na?
18:20.09TripleFFFFpap2 t -na
18:20.19Tsopdlynes_home, he mentiond something about gizmo project and use it to get sip #
18:20.41_problem_smacku1: use n option to exit from queue to dial an extension not in that q
18:20.43dlynes_homewtf is a pap2 t-na?
18:21.01TripleFFFFbah
18:21.02smacku1ok, let me adjust and have you double check me
18:21.02dlynes_homeoh nvm...the 't' is short for toast?
18:21.05TripleFFFFupgrades twice ram etc
18:21.13TripleFFFFFirmware Version: 3.1.10(LSc)
18:21.14dlynes_homeoh
18:21.22TripleFFFFProduct Name:     PAP2T
18:21.44TsopTripleFFFF, is this is a linksys router
18:21.51TripleFFFFyes
18:22.12smacku1_problem_: it exits the queue fine, and dials the extension fine, but then goes into that extensions voicemail rather than trying the next extension. does the n option change that?
18:22.13TsopTripleFFFF, luck i am stuck with a TR32P2 can't get it to unlock
18:22.22TripleFFFFRegister Expires: 300.. could it be after 300 it cant reregister ?
18:22.35_problem_smacku1:no
18:22.53_problem_u already understood what i means..
18:24.05Tsopdlynes_home, so is there is a cheep voip service? and what is the advantage to use asterisk ?
18:24.26smacku1_problem_: yeah, it exits to the first extension fine. What I was hoping is that if it did not answer, it would try a second extension rather than leave a voicemail at the first extension. Kind of like a hunt group, but I did not want to go that way if there was another option.
18:24.52*** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net)
18:25.52_problem_smacku1: no dear i dont know other way ..i was also facing the similar problem...which i little bit solved with n option...perhaps somebody else could help
18:26.30*** join/#asterisk brc_ (n=brc_@pdpc/supporter/basic/brc)
18:26.32smacku1ok
18:26.33*** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br)
18:27.36smacku1_problem_: so in your scenario using the n option, how does your call flow go? Once you exit the queue, what happens?
18:29.01_problem_smacku1: once it exits from a q it goes to an extension..after timout at that extension it goes to another q
18:29.42smacku1can you pastebin what that looks like?
18:30.04_problem_smacku1: sure hold on
18:30.17Tsopdlynes_home, i am just wondering what will be the use for asterisk? if i sign up for a voip service i could use my phone rihgt?
18:31.39_problem_smacku1: http://pastebin.ca/78026
18:32.58Tsopanyone please cleaify , i am just wondering what will be the use for asterisk? if i sign up for a voip service i could use my phone rihgt?
18:39.04TripleFFFFoh
18:39.05TripleFFFFdlynes
18:39.20TripleFFFFi see a REGISTER after the 300 sec and a 401 unauthorized
18:41.50TripleFFFFmaybe not
18:43.38*** join/#asterisk evisu (n=hIRC@bzq-88-152-238-38.red.bezeqint.net)
18:46.09smacku1_problem_: thanks for the suggestion, I am going to try the n option and see what it does.
18:46.23smacku1based on what I am reading about it, it makes sense that this is the correct way to do what I need.
18:46.50_problem_smacku1: ok best of luck...ya
18:47.56*** join/#asterisk clive- (n=pirch@dsl-145-18-135.telkomadsl.co.za)
18:55.14*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
18:55.59*** part/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br)
19:05.07Tsopis it possable i can just use cisco ip phone 7940  instade of tdm card ?
19:05.35clive-tsop, yes, with ztdummy for timming
19:06.18TripleFFFFINVITE Expires:
19:07.52Tsopclive-, please guide me? why will i use asterisk? what is the benifit ?
19:09.37clive-tsop there are many reasons...for fun, for business, for ivrs, voicemails, etc etc
19:10.25nortexTsop, THe reason/benifit of asterisk is to have a full pbx connected to the voip service. If you just want a SIP device to ring when the calls come into you voip service you don't need asterisk.
19:11.33Tsopi am really very confused right now?
19:12.35clive-tsop if you just want to call your girlfreind with your ip phone and thats all, you dont need asterisk
19:12.38clive-lol
19:12.55nortexclive-, hahaha
19:13.07Tsopclive-, she is over sea thu
19:13.08TripleFFFFanyone know how to factory reset a pap2 ?
19:13.32nortexTsop, What do you want your IP phone to do?
19:14.16TsopTripleFFFF, i found the info u want last night in google, try vonage forum they have a user name and password that will rest to factory
19:14.30TripleFFFFdarn.. the shit pap2 t na is sending me byes at exactly 300 sec.. .every time
19:14.45Tsopnortex, basiclly to be able to call my familly over sea, and for fun too
19:17.15nortexTsop, You won't "need" Asterisk for calling overseas if you have an IP phone/ ATA and a voip service
19:17.36TripleFFFFtsop not vonage
19:17.41TripleFFFFits a unloked pap2 t na
19:17.50*** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net)
19:17.52nortexIf you want to have features like clive- mentioned then Asterisk is pretty cool.
19:17.56*** join/#asterisk DarKnesS_WolF (n=wolf@82.201.232.126)
19:18.04clive-tsop I hope she is not here in south africa, the adsl here sucks for voip
19:18.30Tsopclive-, no north africa
19:18.38Tsopand yes it suck there too
19:18.38nortexTsop, Have you set your IP phone up for the voip service yet?
19:18.59clive-tsop welcome to african internet
19:19.02Tsopnortex, not yet i just started yesturday
19:19.26Tsopclive-, i know man the last time i was there they only used dial up , lol
19:19.47Tsopnormsteel, thats why i need ur help guys to be directed on the right track
19:20.09Tsophow can i shop for a voip
19:23.44Tsop???
19:24.08clive-tsop what do you want to buy
19:24.58Tsopclive i want to be able to call north africa for the cheapest price what do i need to get , i am in usa now
19:26.00clive-tsop well you either need to find a voip provider who can offer you cheap calls to the country you need, or you need a callingcard
19:27.08Tsopclive-, which voip u have
19:27.40clive-tsop or if theer is decent internet in the place, you can get another voip phone and call hers with yours
19:28.15rob0FWD doesn't like me. :( I'm certain the FWD number and password are correct, but my IAX2 registration is being rejected.
19:29.21clive--I forgot to "make insta;;"...scary
19:29.29rob0I even reset the password!
19:29.46TripleFFFFso no one know why the pap2 sends a bye every friggin 300 seconds ?
19:31.25*** join/#asterisk mpruett (n=mpruett@24-240-203-82.static.stls.mo.charter.com)
19:32.10Tsopclive-, i am haveing a hard time to show her how to open skype ? do u think i'll be able to show her how to setup voip
19:32.13Tsoplol
19:33.37TripleFFFFshit
19:33.43nortexTsop, You may have a problem there :)
19:33.51*** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie)
19:34.21rob0Hmmm, maybe I should try FWD with SIP.
19:35.41*** join/#asterisk postel_ (n=jp@unaffiliated/postel)
19:36.07Tsopnortex, yeah so thats why i was trying to setup up somethig for cheat
19:37.55smacku1so in my cdr i have a cdr record for the extension, as well as the agent. so it has essentially doubled the call record. is there a way around this?
19:38.20Tsopclive-, can i pm u
19:39.25clive-sure
19:42.14Tsopclive-, did u get my pm
19:43.04*** part/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com)
19:43.42rob0Hmmm, after a bunch of rejections it just started working ...
19:45.09*** join/#asterisk docelmo (n=docelmo@55-65.126-70.tampabay.res.rr.com)
19:45.26smacku1also, do exten => values have to be numeric? or can they be # and * also?
19:45.45docelmoYou can use # or * in the extensions
19:45.54docelmoAnything sip can pass including alpha values
19:46.17smacku1can you see if there is anything that would keep this from working?: exten => #1,1,VoicemailMain(@progrexion)
19:46.57docelmoya the @progrexion
19:47.46docelmoshould be user@context
19:47.49docelmonot @context
19:48.19smacku1exten => 8500,1,VoicemailMain(@progrexion) works just fine. The @progrexion specifies which voicemail context to log into
19:48.20[TK]D-Fenderdocelmo : No, it should be valid as a way to seperate #'s
19:48.37[TK]D-Fenderdocelmo : As he's running amnulti-tennent system
19:49.13docelmoHmm..
19:49.36[TK]D-Fenderdocelmo : that way it still promts for the box but in a specific VM context.
19:49.54docelmoya..  I thought asterisk didnt handle VM that wa
19:49.55docelmoway
19:50.05smacku1yeah, works great.
19:50.25[TK]D-Fendersmacku1 : only reason it might not work would be dialplan context based.
19:50.29smacku1I have 5 different companies all with their own context. working smooth after a weekend of overhaul
19:50.37[TK]D-Fendersmacku1 : Or something stopping the phone from dialing that exten.
19:51.02smacku1hmmm. interesting thought, because I get absolutely no CLI output when dialing it.
19:51.18smacku1maybe the phone its self
19:51.29smacku1xlite soft phone
19:51.48docelmoI know linksys would kill it.  You have to fudge the dialplan in the phone to make that work
19:52.22[TK]D-Fendersmacku1 : either the phone of context setup
19:52.49[TK]D-Fenderdocelmo : Linksys shouldn't have anything to say about that at all... its all * dialplan...
19:53.32docelmo[TK]D-Fender check the Linksys ATA's trust me they will screw with it.  I believe the supura's would also.  I had a hell of a time getting *98 to work correctly
19:54.03[TK]D-Fenderdocelmo : *.T|X.T|#.T
19:54.29[TK]D-Fenderdocelmo : I never had any kind of dialplan issue with them....
19:54.29*** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com)
19:54.29docelmosmart ass
19:54.31[TK]D-Fender:D
19:54.42docelmoI should have asked you 6 months ago
19:54.45[TK]D-Fenderdocelmo : You had me at "smart" ;)
19:55.01docelmoits ok..  I dont work there anymore..   :)
19:56.10docelmoI work for another ITSP now
19:56.33docelmoANYWHO..  any news on astricon yet?  speakers and such?
19:57.10skraelings001what mean that dchan is provisioned?
19:57.14[TK]D-Fenderdocelmo : I can tell you Olle WON'T be there....
19:58.25filetoo early to know aboot speakers...
19:58.46Corydon-waboot?
19:58.50[TK]D-Fenderfile : Silly Canuckian!
19:58.53file:D
19:58.58file[TK]D-Fender: it's beautiful here today!
19:58.58*** part/#asterisk muppetmaster (n=jasongoe@169.red-81-184-73.user.auna.net)
19:59.17[TK]D-Fenderfile : Mississauga : If you don't like the weather, wait 10 minutes.
19:59.27fileha
19:59.28filenice
20:00.07[TK]D-Fenderfile : And dispicably true throughout my vacation.  We had EVERYTHING.  In a day : heavy fog, clear skies & sun, rain, and .... HAIL.
20:00.17*** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net)
20:00.36[TK]D-Fenderfile : I was just waiting for the walls to start bleeding...
20:00.46file[TK]D-Fender: sacrifice a goat!
20:00.47docelmo[TK]D-Fender yes sad..  I heard there was a fall out
20:01.23docelmoSounds like the weather in Tampa
20:01.36docelmoWere in the Hurricane/Rainy season right now
20:04.56*** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org)
20:06.50*** join/#asterisk jcmoore (n=jcmoore@picard.ojc.nuvio.com)
20:08.04*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-246-145.buckeyecom.net)
20:08.46gambolputtyHi.  I am trying to install the mysql addons for * trunk and they don't get selected for compiling.  Could someone help?
20:09.33TripleFFFFany idea on pap2 false hangups ?
20:09.43*** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com)
20:11.37*** join/#asterisk Krush (n=Krush@213.9.109.181)
20:13.48Krushdoesnt zaptel support DESTDIR on make install?
20:14.15docelmolook in the make file
20:14.34docelmoand why not say hi and chill a bit before hammering with questions..
20:14.37docelmobe polite..  :)
20:14.44smacku1is there a quick easy to install app out there that an agent on the floor can see if they are logged into a queue or not?
20:15.00docelmoNope..
20:15.06docelmoCould script one
20:15.10docelmofairly easily
20:15.24smacku1it would just use the manager interface?
20:15.25Krushdocelmo, ok. sorry
20:16.06websaeKrush: are you going to ClueCon?
20:16.19websaeand what about you docelmo are you planning to attend?
20:16.35[TK]D-Fendersmacku1 : AMI would work, or just dump 'asterisk -rx "show queues"' and parse it out.
20:16.35docelmoCluecon?   Doubt it.  When is it?
20:16.56Krushbut make install with destdir ends up in: /bin/sh: /tmp/package-zaptel//etc/udev/rules.d/zaptel.rules - file does not exist
20:16.59smacku1you said dump :-D
20:17.29websaeAug 1-3
20:17.47[TK]D-Fendersmacku1 : Yes, like I did to the bodies of those people who pissed me off last week :F
20:18.39smacku1ahhhh yes, one of those may have been me?
20:19.08[TK]D-Fendersmacku1 : No, you're still talking, but I give you 5 mins tops ;)
20:19.16smacku1ah
20:19.18smacku1run
20:20.10smacku1thats just what happens when you log in while on vacation ;)
20:21.05[TK]D-FenderNo longer on vacation... back with a mountain of work ready forme.
20:21.08websaedocelmo: i better see you there
20:21.10Krushthe makefile contains INSTALL_PREFIX:=$(DESTDIR)
20:24.46docelmoI dunno we will see..  Depends mainly on the new company
20:25.14docelmoThey are all about sending me out from what I have seen so far..  Im going somewhere in august then october and possibly VON in september
20:25.42smacku1is there a command that shows why an agent has been logged out?
20:25.50dpryolol
20:26.08skraelings001pls, help with directed pickup application
20:26.08docelmono
20:26.16smacku1figured as much
20:26.16dpryoI would like a command that shows why an agent didn't answer the call.
20:26.22smacku1i see the error
20:26.25smacku1Jul  3 13:54:08 NOTICE[12176] chan_agent.c: Agent 'Dan Black' didn't answer/confirm within 15 seconds (waited 16)
20:26.35*** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it)
20:26.40smacku1just wondered if anything showed that like the -rx "show agents"
20:27.02dpryogrep "didn't answer" /var/log/asterisk/messages
20:27.02dpryo:D
20:27.04[TK]D-Fenderdpryo : "Agent Busy Masturbating in Opposite Sex's Bathroom, Back Later"
20:27.09docelmohay [TK]D-Fender you got any experience with post dialing?  like into a PBX or something?
20:27.20[TK]D-Fenderdocelmo : "post dialing"?
20:27.32docelmolike the call is established then you dial thru a IVR to reach someone
20:27.55[TK]D-Fenderdocelmo : as in scripted?
20:28.23*** part/#asterisk a1fa (n=a1fa@207.210.210.202)
20:28.32docelmoyes
20:28.54rob0oh wait, that would be an app, not a chan
20:28.56dpryohehe
20:29.02Krushhmm. the zaptel makefile does not seem to work correctly on destdir, as it does not create the necessary subdirs
20:30.13smacku1asterisk seems to be running, but I cannot get any output when i type in a cli command? any idea what is up and how to fix it?
20:30.28smacku1ie i type: slk-apbx-01*CLI> show agents
20:30.28smacku1slk-apbx-01*CLI> show channels
20:30.28smacku1slk-apbx-01*CLI>
20:30.31docelmoIm not sure if its possible.. I keep running shit thru my head and nothing seems to click to make it work
20:31.59smacku1without restarting asterisk is there a way to fix it
20:32.09docelmoisnt there something in the dial command to send DTMF after the call is established?
20:32.57[TK]D-Fenderdocelmo : There is an option for post DTMF dial, but nothing "too bright".  Not really viable.
20:33.07gambolputtyHi.  I am trying to install the mysql addons for * trunk and they don't get selected for compiling.  Could someone help?
20:33.27[TK]D-Fenderdocelmo : its just dumps the DTMF instantly, not waiting or detecting connect no really allowing paus between IVR options.
20:33.45[TK]D-FenderBBIAB, heading home.
20:37.51*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
20:38.19anthmyou *can* insert w's for 500ms pauses if you wish
20:41.02smacku1ok, so if i call my extension where I have AgentCallbackLogin, suddenly it is not working 100%. Nothing has changed since it did work a few minutes ago except for when I ran asterisk -rx "show queues"
20:41.07smacku1now when you call it:
20:41.29smacku1you get the prompt to log in, but then it never gets to the point where it asks for pin or extension.
20:41.36*** join/#asterisk nagl (n=nagl@86.59.54.237)
20:41.37smacku1the call does not die, it just does not do anything.
20:41.56smacku1any help?
20:43.22*** join/#asterisk prh (n=paul@X80.mjr.org)
20:44.51smacku1anyone?
20:46.41smacku1is there a stop command to this? asterisk -rx "show queues"
20:47.25smacku1reload app_queues.so gives me the error "the previous command didnt finish yet"
20:47.34smacku1I think that may be causing my issue
20:48.30smacku1now I am getting the previous reload did not complete yet.
20:48.33smacku1wtf
20:48.36smacku1any ideas?
20:50.59mrtwister*CLI> stop nowJul  3 11:50:17 ERROR[11949]: res_config_mysql.c:615 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on . Check debug for more info.
20:51.11mrtwisterhow to read debug info
20:51.46smacku1that is the output you got from a stop now command?
20:52.00Corydon-wmrtwister: please update your -addons source
20:52.54mrtwisteri installed mysql-dev and compiled addons
20:53.10Corydon-wWhen?
20:53.35smacku1anyone know how to stop a reload?
20:53.51Corydon-wsmacku1: 'stop now'
20:53.59smacku1does that not stop asterisk?
20:54.03Corydon-wYes
20:54.13smacku1cant stop asterisk 49 calls on the lines
20:54.19smacku1i need to stop the reload
20:54.22smacku1is it not possible?
20:54.51smacku1that has to be a way
20:54.51Corydon-wOkay, then.  You cannot stop the reload without shutting down Asterisk.
20:54.57Corydon-wSorry.  It wasn't designed to be stoppable.
20:55.09Corydon-wThen again, it's not supposed to take that long, either
20:55.12smacku1crap
20:55.30skraelings001mrtwister: are you working with real time ?
20:56.05mrtwisteryes
20:56.15mrtwisterwith realtime
20:56.29*** join/#asterisk Greek-Boy (n=Greek-Bo@193.220.93.162)
20:56.47Corydon-wmrtwister: that particular error message was addressed last week.
20:57.36*** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it)
20:57.41Corydon-wSo update your addons to the latest SVN 1.2.
20:58.38Greek-Boywhats a good solution for someone that uses multiple phones? to quickly set which phone you're at so someone just has to remember one extension?
20:59.15mrtwisterwhy i cannoit use stable addons
20:59.16mrtwister:)
20:59.22*** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it)
20:59.38Corydon-wmrtwister: there is no such thing as "stable addons"
20:59.43mrtwisteri use 1.2.9.1
21:00.05Corydon-waddons are not at release 1.2.9.1
21:00.48*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
21:00.54Corydon-wThat's why I'm telling you to run the release branch of addons
21:01.27skraelings001mrtwister: what is your res_mysql.conf ?
21:01.44clive-what does chan_phone do, my asterisk keeps failling to laod up on that
21:01.54mrtwister[general]
21:01.54mrtwisterdbhost = localhost
21:01.54mrtwister;dbname = asterisk
21:01.55mrtwisterdbname = billing
21:01.55mrtwisterdbuser = root
21:01.55mrtwisterdbpass = amoeba0819
21:01.57mrtwisterdbport = 3306
21:01.59mrtwisterlike that
21:02.15PakiPenguinhello there , i have a tdm2400p , and i need to mark a specific callerid to an incoming fxo channel ( like whenever a call comes in on channel1 , i need it to say 7085441700 on caller id by default , so i can differentiate between calls ) , i did this in zapata-chanels.conf -> http://pastebin.ca/78171 , but its not sending me the number
21:02.25skraelings001clive : standard linux telephony API
21:02.28Corydon-w~pb
21:02.30jboti guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/
21:02.53clive-skraelings what can cause it to crash on that?
21:02.58PakiPenguincan anyone help me please
21:03.12[TK]D-FenderPakiPenguin : Set each FXO channel to have its own incoming context and in the "s" exten, set it at the start.
21:04.15PakiPenguin[TK]D-Fender, amp :( .. i wanted to do that .. but we have amp here and its needed since the user managing wont be capable at all
21:04.57[TK]D-FenderPakiPenguin : if you ahve to ask, the answer is "ditch AMP"
21:05.07PakiPenguin:) yeah i know
21:05.47[TK]D-FenderPakiPenguin : I presume you can define each channel seperately.  From there you should be able to define a "custom script" at the end of which you can "goto" a defined IVR
21:06.12skraelings001clive: what does it say when fails?
21:06.32*** join/#asterisk juanjoc (n=juanjoc@200.73.189.82)
21:06.52skraelings001mrtwister:  socket exists? /var/run/mysqld/mysqld.sock
21:07.58clive-skraelings : Jul  3 22:31:45 WARNING[3134]: loader.c:554 load_modules: Loading module chan_phone.so failed!
21:08.05*** join/#asterisk marv0997 (i=marv0997@190.4.2.83)
21:08.18mrtwistersolved.
21:08.31mrtwisterneeded to define socket in res_mysql.conf
21:08.35skraelings001mrtwister : good!
21:08.43mrtwisternot working witout :)
21:09.19skraelings001mrtwister : what is your modules.conf ??
21:10.12mrtwisterfrom make samples
21:10.23mrtwisterit is installed just 20 minutes ago
21:10.32mrtwisterat ubuntu dapper
21:11.34skraelings001mrtwister: you haven't touched anything since?=+
21:12.34skraelings001mrtwister: how do you start asterisk?   asterisk or asterisk -vvvvv
21:13.24*** part/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com)
21:13.26mrtwisterasterisk, then asterisk -vvvdddr
21:14.13skraelings001killall asterisk and try asterisk -vvvvv , what's the output?
21:14.29mrtwisterno, all ok now :)
21:14.44skraelings001ok
21:15.10*** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
21:15.10*** mode/#asterisk [+o russellb_] by ChanServ
21:16.58docelmo~seen trixter
21:17.01jbottrixter <n=trixter@65-165-167-217.du.volcano.net> was last seen on IRC in channel #asterisk, 2d 15h 40m 14s ago, saying: 'but the fact that its there and can be spread by bites means that this year they are likely to  have more cases where people get infected'.
21:22.59*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
21:24.23*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
21:24.33*** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com)
21:26.26*** join/#asterisk Mattwj2005 (n=Matt@user-12l3n74.cable.mindspring.com)
21:26.59Mattwj2005hey guys...what do you think of net neutrality?
21:27.10fileI'm neutral on the issue.
21:27.20*** join/#asterisk blebleble (i=godie@caesar.godie.net)
21:27.24dlynes_homeis that where your net gets neutered?
21:28.24Mattwj2005the telephone company wants a two tier Internet
21:28.38*** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net)
21:29.08Mattwj2005on with high priority traffic for their traffic and companies that pay extra and one for everything else
21:29.09gambolputtyNo ISP should block or degrade any service that competes with theirs
21:29.18rob0FWD: IAX or SIP? I think I have it working with IAX but it seems flaky, someone on FWD forums said SIP is better.
21:29.40Mattwj2005I just wonder how this is going to affect VoIP?
21:30.03dlynes_homeMattwj2005: *shurg*  aol thinks they can get away with it for email
21:30.08gambolputtyImagine an ISP blocking all UDP 5060 traffic.
21:30.34Corydon-wThat's why God made port 5070
21:32.07Mattwj2005I work in Networking.....I think it is just going to be more headaches
21:32.54dlynes_homeMattwj2005: phone up the fcc...maybe they'd like to hear about a phone company that wants preferential network treatment
21:33.17rob0Maybe they've already been paid.
21:33.22dlynes_homethat, too
21:33.32Mattwj2005it America....yeah probably
21:33.34Mattwj2005*in
21:34.47Mattwj2005The job of the phone company should be to get my voice or data from point A to B.....and then people like myself should be concerned with everything else
21:35.37nortexBut that might cut into their precious profits :0
21:36.21Mattwj2005good point
21:39.32*** join/#asterisk mooodi (i=mooodi@bouncer.ikhost.com)
21:40.18Mattwj2005I think in the end...even if it does pass it will eventually bit them in the behind....I am sure they will get a lot of complains from users and companies like the one I work for.....in the end the good will win
21:40.48mpruettHello Everyone!
21:41.01Mattwj2005hi mpruett :)
21:41.10mpruettI have a tricky one for you guys
21:41.47mpruettPlease take a look at http://pastebin.ca/78191 - I put the problem here with everything I have done to this point
21:42.05*** part/#asterisk mooodi (i=mooodi@bouncer.ikhost.com)
21:42.07mpruettDidn't want to clutter up everything in here ;)
21:43.16fileI assume you set nat=yes?
21:43.21mpruettYes
21:43.46Mattwj2005mpruett where are you from?
21:44.00mpruettMissery
21:44.19mpruettor Missouri whichever
21:44.32filempruett: well, a packet has to come from the NATted device before the media can be sent to it's non-private IP address and it doesn't look like it sent one
21:44.35Qwell[]You can't spell your state?
21:44.40filempruett: silence suppression on? VAD?
21:44.49Mattwj2005oh okay....you have a similar name to one of my old college teachers.....just thought I would check :)
21:44.52*** join/#asterisk gorauskas (n=gorauska@66-224-20-131.atgi.net)
21:45.01mpruettPlay on words - obviously not a funny joke
21:45.55mpruettfile: I do not believe I have silence suppression on
21:46.27mpruettfile: I know how to do this from ATA - Is there a way to do at Asterisk level?
21:46.42fileAsterisk doesn't support it
21:47.01filebut in order for your audio to travel, both sides have to send at least one packet of audio to Asterisk
21:47.09mpruettfile - I have this set to "No" on ATA
21:47.32fileit will then change the destination IP address and port for both sides from it's private IP/port to the NATted public IP/port, and audio should flow
21:48.51*** part/#asterisk gorauskas (n=gorauska@66-224-20-131.atgi.net)
21:50.17smacku1is anyone here using e&m wink start T1s?
21:51.01smacku1I cannot get the caller id to work with them
21:51.12mpruettfile: How can I tell if this is happening? Does this behavior change with MeetMe?
21:51.12smacku1so i am wondering if there is any special configuration that i need to consider.
21:51.25smacku1it all works with my pris but not e&m
21:52.19filempruett: you look at rtp debug, and when a packet comes in from the phone that is NATted... all outgoing packets should then switch to that source IP address and port
21:52.46filempruett: do you have canreinvite=no ?
21:53.29mpruettfile: I have canreinvite=yes
21:53.52filethat's going to cause issues with NATted devices... weird things can happen
21:53.56*** join/#asterisk paolob-parroquia (n=paolob-p@pri-214-b7.codetel.net.do)
21:54.05mpruettLet me switch and retest
21:56.05paolob-parroquiaGuys, I have my asterisk server working, but I can't understand how can I do the following: I'm the secretary, after answering a call for my boss I must ask him whether to pass him the call or not, but if I perform a transfer I can't ask the boss the permission to pass him the call. How can I do it?
21:57.10filepaolob-parroquia: attended/supervised transfer?
21:57.43mpruettfile: YOU ARE THE MAN!!!!!
21:57.53paolob-parroquiafile, no, if I do an attended transfer, when my boss ansers he get directly the call
21:58.10filepaolob-parroquia: that's a blind transfer
21:58.14mpruettfile: 4 tests all good - will continue to test but thanks for the help!!!
21:58.24*** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net)
21:59.20mpruettfile: This is the second time you have helped me out with an issue - I seriously appreciate your help
22:02.15paolob-parroquiafile, ok, I understand. But after the boss answer my attended call, what are the options?
22:02.45smacku1does anyone know about callerid on E&M? there references out there that suggest it is done differently than normal, but the link I found to the details is dead
22:02.56*** join/#asterisk jcmoore (n=tgrman@c-71-199-75-134.hsd1.ks.comcast.net)
22:03.29filepaolob-parroquia: I've never done that kind of transfer lol
22:03.56bleblebleanyone ever see 'Unable to pass the full buffer onto the device file. -1 bytes of 2 written: Resource temporarily unavailable' from iaxmodem?
22:04.12smacku1are you referring to doing a "warm transfer"? where you announce the transfer, then hang up the call?
22:04.21smacku1paolob-parroquia:
22:04.26filewarm... attended... supervised...
22:04.30[TK]D-Fenderpaolob-parroquia : Attended transfer means you call the 3rd party, they see YOUR caller ID, they answer, you ask if they want the call, if so you typically press the transfer button again and the call gets passed off
22:04.54smacku1that is what i was typing
22:06.37*** join/#asterisk BZBW (n=wlwzhang@ip67-153-142-109.z142-153-67.customer.algx.net)
22:07.56paolob-parroquia[TK]D-Fender, ok, thnx
22:09.29smacku1after a while of asterisk running, I get the issue where stuff just doesnt do anything when i type it. Like:
22:09.29smacku1slk-apbx-01*CLI> reload
22:09.29smacku1slk-apbx-01*CLI> show channels
22:09.29smacku1slk-apbx-01*CLI>
22:09.45smacku1when an agent tries to log in at this point it only prompts them for there agent id, and then does nothing.
22:09.49smacku1what could be happening.
22:09.54smacku1second time in two hours
22:10.03smacku1no one is doing anything on the system except using it
22:17.09*** join/#asterisk bjohnson (n=bjohnson@i216-58-63-230.cybersurf.com)
22:17.19*** part/#asterisk mpruett (n=mpruett@24-240-203-82.static.stls.mo.charter.com)
22:20.14*** join/#asterisk |marv0997| (i=marv0997@190.4.2.86)
22:20.18smacku1http://pastebin.ca/78217
22:20.34smacku1are the calls that are down supposed to still be there?
22:21.33*** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com)
22:22.00smacku1can i kill calls from the cli?
22:24.00smacku1did everyone leave?
22:24.09clive-smaku soft hangup
22:24.34smacku1ok, and do i specify a channel? or extension?
22:25.08clive-yes
22:25.15clive-try it until it works:)
22:25.17smacku1looks like channel
22:28.46smacku1ok, so i have done soft hang up on all three lines that show down, ring and down. if that did not kill them, whats next to try?
22:29.12[TK]D-Fendersmacku1 : "restart now"
22:29.26smacku1suck! second time in 2 hours.
22:29.35smacku1any idea as to why this keeps locking up?
22:29.40smacku1I think it is that same extension
22:33.11*** join/#asterisk luke-jr_ (n=luke-jr@2002:1891:f657:0:20e:a6ff:fec4:4e5d)
22:33.48dlynes_home[TK]D-Fender: a101 is working nice and smooth on our main softswitch now :)
22:34.10dlynes_home[TK]D-Fender: no more spurious hdlc framing errors
22:34.58[TK]D-Fenderdlynes_home : Good to hear.
22:35.14[TK]D-Fenderdlynes_home : Hints on the cause?
22:39.51smacku1this looks to me like the calls are all locking up on the queuing and agent logins. am I seeing this correctly? http://pastebin.ca/78237
22:42.37*** join/#asterisk bigmac4444 (n=mtur2848@CPE-124-177-67-147.qld.bigpond.net.au)
22:43.09bigmac4444good morning/evening all
22:45.18bigmac4444i have a quick question
22:46.35bigmac4444i need to redirect an inbound pstn call to another number.
22:50.39bigmac4444*grabs a number and standings in line
22:50.53bigmac4444stands*
22:52.30*** join/#asterisk eipi (n=eipi@139-213-126-200.fibertel.com.ar)
22:56.03Bullseye_Networkbigmax4444: like this? http://pastebin.ca/78254
22:56.12Bullseye_Networkmac not max
22:56.13Bullseye_Networklol
23:00.49*** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net)
23:07.24dlynes_home[TK]D-Fender: probably shitty hardware, defective card, or slow cpu
23:07.29dlynes_home[TK]D-Fender: or a combination thereof
23:10.59[TK]D-FenderAll hail the C3! ;)
23:11.24dlynes_home[TK]D-Fender: not that bad :)
23:11.55[TK]D-FenderDunno... C4 is at least explosive ;)
23:12.01dlynes_homeCeleron 2Ghz
23:12.18dlynes_homerunning 10 g729 channels
23:12.49[TK]D-Fendernot terrible...
23:13.06dlynes_homeand even then most of the time
23:13.25rob0Oh man, I had a helluva time trying to get * working on a C3.
23:13.33dlynes_homeI was usually maxing out at three channels (7 of those licences I don't think were ever used)
23:13.58dlynes_homerob0: I've had quite good experience getting it to work on a C3
23:14.23dlynes_homerob0: but try to use any digium hardware on it, forget it
23:14.34dlynes_homerob0: the shared interrupts kill the digium hardware
23:14.38rob0I failed because the TDM400 card didn't work on that motherboard. Not truly PCI 2.2 compliant.
23:14.44rob0yes
23:15.32rob0My newest * box is x86_64 ... heaven!
23:19.57*** join/#asterisk bryanfe (n=chatzill@c-24-8-177-94.hsd1.co.comcast.net)
23:20.23*** join/#asterisk Kis (i=vlad@p5080FC98.dip.t-dialin.net)
23:20.54bryanfeQuestion - after installing addons, is there something more I need to do to get musiconhold to use format_mp3? I have confirmed that the module is being loaded but musiconhold seems to be complaining that it can't launch the thread
23:21.43[TK]D-Fenderbryanfe : What "mode" is it using?
23:22.13bryanfemode=quietmp3
23:22.37[TK]D-Fenderbryanfe : then you're not set for using Native MoH.  That uses MPG123.  You should be using "mode=files"
23:22.46*** part/#asterisk smacku1 (n=smackus@c-67-169-248-217.hsd1.ut.comcast.net)
23:23.06dlynes_homempg123 is eeeeeeeeeeeeevilllll!!!!
23:23.09bryanfegot it, will try that. (sample musiconhold.conf didn't really say that)
23:23.28[TK]D-Fenderdlynes_home : not as bad as.... *coughs* telnet
23:23.28Nuggettelnet is eeeeeeevil!
23:23.46[TK]D-Fenderbryanfe : Yes it did ;)
23:23.52dlynes_hometelnet doesn't go into a zombie process :)
23:24.01bryanfe(where? I may be dense)
23:24.10[TK]D-Fenderdlynes_home : Even starting telnet means you're already a zombie ;)
23:24.24dlynes_home[TK]D-Fender: nah
23:24.38dlynes_home[TK]D-Fender: when you need to fill your craving for mud, and a mud client isn't handy
23:24.42bryanfestill getting this error, hmph... Jul  3 18:24:08 NOTICE[10985]: res_musiconhold.c:511 monmp3thread: Request to schedule in the past?!?!
23:24.43dlynes_home[TK]D-Fender: you pretty much have to use telnet
23:25.10dlynes_homebryanfe: restart when convenient
23:25.11[TK]D-Fenderbryanfe : Thats because its still trying to use MPG123... shut * down completely and restart.
23:25.18bryanfe10-4
23:25.28[TK]D-Fender"restart with reckless abandon"
23:25.38[TK]D-Fender"restart with extreme prejudice"
23:25.43dlynes_homerestart NOW
23:25.55[TK]D-Fender":D
23:26.02Qwell[TK]D-Fender: I like that last one
23:26.30[TK]D-FenderQwell : I figured you would :)
23:26.36bryanfeliftoff, we have MOH.
23:26.41bryanfethank you ;)
23:26.50[TK]D-Fenderbryanfe : quite welcome
23:26.59Qwell[TK]D-Fender: on a semi-related note...
23:27.07Qwellon this sunfire, you can do `poweroff -fy`
23:27.10[TK]D-Fenderdlynes_home : That was well rought up... too bad you weren't :)
23:27.19QwellI've decided that the 'fy' stands for "fuck you"
23:27.22[TK]D-FenderQwell : cute
23:27.27Qwellforce, yes :D
23:27.32[TK]D-FenderQwell : that wa my interpretation as well
23:27.35Qwellheh
23:27.37dlynes_homeQwell: i'll have to let l-fy know that :)
23:27.44Qwelldlynes_home: like-fuckyou
23:27.47Qwell?
23:27.59dlynes_homel-fy, the moderator in #yate
23:28.06Qwelldlynes_home: yes, I know
23:28.08dlynes_homeshe often comes in here
23:28.10dlynes_homeah
23:28.14Qwelloh...she...oops :D
23:28.36dlynes_homeshe just comes in here, when she wants to stir up shit :)
23:28.38bryanfeis SetMusicOnHold(none), where "none" is a class pointing to /dev/null, really the correct way to shut off MOH? Seems a little brutish to me ;)
23:29.40[TK]D-Fenderbryanfe : Kill the class
23:29.47*** join/#asterisk m_a_g_o (i=maxgluck@201.243.102.189)
23:30.14bryanfei'm looking at the example here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MusicOnHold
23:31.09[TK]D-Fenderbryanfe : You don't normally turn off MoH....
23:31.24m_a_g_ogood evening folks, I'm trying to install *'s oh323 channel, but keep getting: H.323 listener creation failed... any idea/advice on this particular error please? =)
23:31.29Juggiebryanfe, do you want Moh for NO users?
23:31.31Juggieor just not for some.
23:32.02[TK]D-Fenderok, time for volleyball, later all.
23:32.03*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
23:32.16bryanfehere's what I need: 1) turn on MOH, 2) Dial() with 30 sec timeout (it won't answer). 3) Wait(20), 4) Turn off MOH, and (5) Dial again (it will answer this time)
23:32.42bryanfelegacy stuff needs to be triggered with that first dial, I'm looking for MOH while the user waits
23:33.36dlynes_homem_a_g_o: turn on more logging so you can see what the real error is?
23:34.44*** join/#asterisk Brijn (n=bas@S0106004063c0fa1f.vn.shawcable.net)
23:34.52m_a_g_othks, let me do that... brb
23:36.30m_a_g_oactually that is all there is to it, with debug error and warning enabled
23:39.36bryanfeI may be misunderstanding how this all works but shouldn't the Wait() command play MOH while waiting, if the MOH class was previously set with SetMusicOnHold()?
23:41.17bryanfeor maybe this here dummy should try the WaitMusicOnHold() command.
23:42.40wunderkinbryanfe, the dial is probably screwing with it.. you can dial with moh, after that yeah you will want to do waitmusiconhold
23:43.58bryanfetrying it now..
23:46.21bryanfeDumb question maybe, but the -addons music on hold mp3 files -- are they, like, donated to the Asterisk project royalty-free (the music, I mean), by the artists?
23:47.14Qwellbryanfe: there is a license file
23:47.43Qwelldoc/README.fpm I think it is, in 1.2
23:48.33hadsdoc/musiconhold-fpm.txt
23:48.36bryanfeno such file is in the current addons distry
23:48.37bryanfedistro
23:48.42Qwellbryanfe: It's in asterisk
23:48.45Qwellhads: That's trunk
23:48.54hadsAh, my bad.
23:53.29RoyK[at]<PROTECTED>
23:54.01RoyK[at]~lart Qwell
23:57.35*** join/#asterisk Klydal (n=Klydal@ip68-226-15-98.nc.hr.cox.net)
23:59.08*** join/#asterisk paolob-parroquia (n=paolob-p@pri-214-b7.codetel.net.do)

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