IRC log for #asterisk on 20070620

00:16.37*** part/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net)
00:16.42*** join/#asterisk Corydon76-home (i=five@pdpc/supporter/sustaining/Corydon76-home)
00:16.42*** mode/#asterisk [+o Corydon76-home] by ChanServ
00:17.38*** join/#asterisk Mavvie (n=edwin@ppp121-44-63-246.lns2.syd6.internode.on.net)
00:23.07*** join/#asterisk rsd99 (n=rsd095@c-71-224-187-182.hsd1.pa.comcast.net)
00:23.36rsd99does anyone know if there is a configuration utility out there for asterisk configs?
00:23.38*** join/#asterisk paolob (n=donpaolo@196.3.84.214)
00:23.45*** part/#asterisk paolob (n=donpaolo@196.3.84.214)
00:23.48JTi use vi myself
00:24.29rsd99i was hoping there would be something out there that generated the config files.  i just don't know where to start
00:25.02rob0Used to be, "make samples", but now I think "make install" does that.
00:25.47rob0IOW, the default config files give you a pretty good start.
00:26.24JT~thebook
00:26.24jbotthebook is probably a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
00:26.27JTis quite useful too
00:28.04rsd99just a little overboard for me.  i am just doing this for something to do at home.  a simple ivr where someone dials in, and chooses an extension and it rings one of my ip phones
00:28.51rsd99i have the calls working internally.
00:31.58JTso how is the book overboard?
00:32.05JTit applies just fine to at home
00:32.15*** join/#asterisk _DAW (n=chatzill@adsl-222-41-108.msy.bellsouth.net)
00:34.35[TK]D-Fenderrsd99, go to www.trixbox.org and use that.  If you have any questions they have their own IRC channel & support forums.
00:35.15Qwell[TK]D-Fender: Who are you, and what have you done to [TK]D-Fender?
00:35.31mihinomenestsomeone gave up the ghost.
00:36.11[TK]D-FenderQwell[], Preemptive strike to rid us of a person who doesn't actually want t learn anything.
00:36.19JTStrom_M: how much longer are you there for?
00:36.21[TK]D-FenderQwell[], that is ENTIRELY me :)
00:36.25Strom_MI leave saturday
00:37.59JTStrom_M: ah, i thought you were just doing a plane change
00:38.17Strom_Mno no; i'm in melbourne for a week
00:38.31Strom_Mi only change planes in sydney
00:39.02Jon335Does anyone know of a site that sells VoIP DECT phones in the US?
00:42.14*** join/#asterisk docelmo (n=vircuser@c-76-99-157-112.hsd1.de.comcast.net)
00:42.27docelmoanyone know anything about automon?
00:43.03JTStrom_M: ah ok
00:43.11JTStrom_M: business or fun?
00:48.56*** join/#asterisk Strom_C (n=strom@60-241-200-26.static.tpgi.com.au)
00:52.21[TK]D-FenderBBIAB
00:52.27*** join/#asterisk SwK (n=SwK@m090e36d0.tmodns.net)
00:55.03pllaJT: thanks, it worked.
00:55.26*** join/#asterisk Strom_C (n=strom@60-241-200-26.static.tpgi.com.au)
00:55.28plla12 hours and the pri hasn't failed.
00:55.29*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
00:56.35JTplla: all you changed was the span line?
00:56.52pllayep
00:57.11JTnice
00:57.16JTgood to see it helped
00:57.23pllaStill it makes me wonder how has it been working in the other setup for 4 months.
00:57.34JTchance
00:58.59*** join/#asterisk spaceinvader (n=server@unaffiliated/spaceinvader)
01:00.07spaceinvaderis it possible to connect an client to an iax channel from the console?
01:00.41spaceinvaderi.e. something like dial but that affects a remote sip connection and not the console
01:02.43*** join/#asterisk GlobeTrotter (n=eric@ip22-28-10-190.ct.co.cr)
01:02.45JTi really don't understand what you're asking
01:02.47*** join/#asterisk Caplain (n=shayne@adsl-75-45-253-93.dsl.sfldmi.sbcglobal.net)
01:02.55samy_b1hello
01:03.16Caplainwhats a good fxo fxs card?
01:03.24spaceinvaderJT: i have a client connected via sip
01:03.28GlobeTrotterhi i get this error when i start asterisk
01:03.36GlobeTrotter-/usr/lib64/asterisk/modules/chan_sip.so: undefined symbol: ast_osp_terminate
01:03.37samy_b1can some one tell me why soem times DISA works and some time don't
01:03.45spaceinvaderJT: the ui wont accept a IAX/ addr as its sip
01:04.02spaceinvaderJT: can i somehow initiate the IAX call from the console and route it to that sip client?
01:04.08samy_b1like 3 of the times i call from my cell i get it working and one not]
01:04.18samy_b1is realy strange and i don't see any errors
01:04.29samy_b1in the erros logs files
01:04.30JTspaceinvader: that makes no sense, why wouldn't you just use sip?
01:04.50samy_b1any one head that issue too ?
01:06.00spaceinvaderJT: i want to make a call to an IAX channel, but its only a one-off so i dont want to make an extension for it
01:06.43JTi assume you mean an iax channel somewhere other than your network
01:07.02spaceinvaderyes
01:08.04JTno, i don't know an easy way to do that
01:08.09JTan iax softphone would be easy
01:08.53Strom_Cwhy not make a temporary extension that you delete later?
01:08.56Strom_Cor would that be too easy? :)
01:09.29*** join/#asterisk Infested (n=infested@24.148.112.10)
01:11.22JTStrom_C: so you here on business?
01:11.45Strom_Cyes
01:12.48GlobeTrotteri get this error when i try to start astereisk   ::load_modules: Unable to open modules directory /usr/lib/asterisk/modules
01:13.33_VoiceMeUp_COMls -la /usr/lib/asterisk/modules/*.so
01:13.37_VoiceMeUp_COManything in there ?
01:13.54_VoiceMeUp_COMif no modules ./configure --enable-shared
01:13.55_VoiceMeUp_COMand redo
01:14.35JTStrom_C: do you have to change planes in sydney, or just get off and reboard the same flight?
01:14.36GlobeTrotterwhere do i run this command?
01:14.42GlobeTrotterthere is nothing in there
01:14.54GlobeTrotterwhre do i run ./configure --enable-shared
01:14.55_VoiceMeUp_COMin asterisk src dir
01:15.01_VoiceMeUp_COMhow you get asterisk ? rpm ?
01:15.03_VoiceMeUp_COMor src ?
01:15.21_VoiceMeUp_COMif rpm .. then type updatedb &
01:15.50GlobeTrotterBEcd
01:15.56GlobeTrotterbussiness edition cd
01:16.25*** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id)
01:16.28_VoiceMeUp_COMonce done locate pbx_functions.so
01:16.29Strom_CJT: it's a completely new plane in sydney
01:16.32_VoiceMeUp_COMand youll find where htey are
01:16.42_VoiceMeUp_COMthen check asterisk.conf fo rlibdir and see if they match
01:16.48_VoiceMeUp_COMim out luck to you
01:17.12JTStrom_C: so would you have time to meet at the airport (would it even be possible? heh)
01:17.13_VoiceMeUp_COMbut you paid 1000 for this , so it includes support i hear.. you should have an 800 support # to bitch to
01:17.40_VoiceMeUp_COM<PROTECTED>
01:17.44_VoiceMeUp_COMthat means DIR not there
01:17.46_VoiceMeUp_COMbtw
01:17.53_VoiceMeUp_COMnot that it cant find the modules
01:17.56_VoiceMeUp_COMso my bad
01:18.09_VoiceMeUp_COMstill locate pbx_functions.so
01:18.46Strom_CJT: um, i'd have to check my schedule
01:19.05*** join/#asterisk zotz (n=zotz@24.244.163.157)
01:19.05JTStrom_C: i'm thinking it might not even be possible
01:19.13JTif you're in the secure area
01:19.26Strom_Cyeah
01:19.48JTif it's the international terminal, you need to clear customs to get in, and you need a boarding pass for that :/
01:19.55JTi did that last week
01:19.57JTwas in japan
01:21.28GlobeTrotterastmoddir => /usr/lib/asterisk/modules is what is in asterisk.conf..  but the files are in astmoddir => /usr/lib64/asterisk/modules
01:22.14*** join/#asterisk mindCrime (n=chatzill@65.190.188.124)
01:25.33Strom_CJT: well, you could always pop down to melbs for dinner or somesuch :)
01:25.37Strom_Chow long is the drive?
01:26.21JT8.5 hours
01:26.50JTthat's with a max of an hour break, obeying the speed limit mostly :)
01:27.06JTit's cheaper to fly for one person travelling, due to fuel costs
01:27.32*** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net)
01:27.32*** mode/#asterisk [+o mog] by ChanServ
01:27.41GlobeTrotteryeah i paid but the support is now closed
01:27.43GlobeTrotter:(
01:30.56*** join/#asterisk Cyber-Dogg (n=jphelps@24-178-240-97.dhcp.stls.mo.charter.com)
01:31.15Cyber-DoggI have zaptel and asterisk setup on my system... but I'm having a couple isses
01:31.34Cyber-DoggI have zaptel.conf, zapata.conf and extensions.conf all setup the way I want...
01:32.01Cyber-Doggi do not have a dial tone on my phones though... and when I load up asterisk... I don' thave any zap commands available
01:32.05Cyber-Doggany thoughts of what to check?
01:32.13Cyber-Dogglsmod shows that zaptel is loaded...
01:33.09JTis the card's driver loaded?
01:33.25Cyber-DoggI think so...
01:33.46Cyber-Doggwct4xxp and wctdm are both loaded according to lsmod
01:33.54Cyber-Doggbut don't I need wctfxo too?
01:34.30JTerr
01:34.40JTyou only need the driver for your card loaded
01:34.45Cyber-Doggoh ok
01:34.51Cyber-Doggwell i think it is then... I have a 400p
01:35.03JTyou have 2 drivers loaded
01:35.20Cyber-Doggshould I kill the wctdm then?
01:38.57Cyber-Dogghow do I unload it?
01:39.39*** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar)
01:41.46*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
01:43.05Cyber-Doggin my modules.autoload.d I only have zaptel and wct4xxp
01:43.10Cyber-Doggso I'm not sure how to undo the other one
01:43.51JTrmmod
01:44.13Cyber-Doggok... let me see if that helped any
01:44.54Cyber-Doggthat breaks it worse...
01:44.59Cyber-DoggI think I need wctdm
01:45.54Cyber-Doggwhen I do ztcfg -v the channel map looks right... so I think everything is ok with my channels
01:48.06*** part/#asterisk Cyber-Dogg (n=jphelps@24-178-240-97.dhcp.stls.mo.charter.com)
01:50.11*** join/#asterisk tengulre (n=tengulre@222.90.66.10)
01:52.03*** join/#asterisk Cyber-Dogg (n=jphelps@24-178-240-97.dhcp.stls.mo.charter.com)
01:52.14tengulrenobody here?
01:52.16Cyber-DoggI am
01:52.27Cyber-Doggbut I'm pretty worthless... :-)
01:52.31Cyber-DoggI'm trying to get help myself
01:53.57[TK]D-Fenderwww.drphil.com
01:54.07Cyber-Dogghe's a crock...
01:57.10*** join/#asterisk fujin (n=aj@unaffiliated/fujin)
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02:00.43*** join/#asterisk ehaupt_ (n=ehaupt@unaffiliated/ehaupt)
02:02.50*** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp)
02:03.02Cyber-Doggso... anyone have any ideas why I'm having problems?
02:03.14Cyber-Doggmy mapping looks right...
02:03.21Cyber-Doggbut no dialtone...
02:04.02*** join/#asterisk jmacz (n=jmacz@190.24.103.191)
02:05.25*** part/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net)
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02:08.13Cyber-Doggwhen I am in the cli... shouldn't I have some "zap" commands available to me?
02:08.20Cyber-Doggliek... zap show status
02:09.08*** join/#asterisk flujan (n=flujan@201-42-103-137.dsl.telesp.net.br)
02:09.51*** part/#asterisk flujan (n=flujan@201-42-103-137.dsl.telesp.net.br)
02:10.40JTyes, unless chan_zap failed to load
02:11.03*** join/#asterisk lmoreira (n=lmoreira@201009076233.user.veloxzone.com.br)
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02:12.21Cyber-Doggchan_Zap.... how do I chekc that?
02:13.13*** join/#asterisk Jabroni (n=Jabroni@red-corp-200.76.249.142.telnor.net)
02:13.15JTthere's probably some error that prevented it from loading
02:13.24Cyber-Doggwhere can I check to see?
02:13.50Cyber-Doggdmesg doesn't show anything...
02:13.57*** part/#asterisk Jabroni (n=Jabroni@red-corp-200.76.249.142.telnor.net)
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02:15.01JTthe startup of asterisk
02:15.14Cyber-Doggis ther ea log file for that?
02:16.04JTif you have enabled full log, yes
02:16.19Cyber-Doggand where might that log file be?
02:17.29Cyber-Doggwhen I type /etc/init.d/asterisk start it shows !! so I'm assuming thre is an error
02:18.23*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
02:18.37JTwhere all asterisk logs are kept /var/log/asterisk
02:19.56*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
02:20.18Cyber-Doggusr/lib/asterisk/modules/chanzap: file not found
02:20.51JTis this asterisk from source or a package?
02:21.30Cyber-Doggemerge in gentoo
02:21.34Cyber-Doggsource I believe
02:23.04JTwould seem you compiled it in the wrong order
02:23.24Cyber-Doggoh... so... I need to do zaptel... then asterisk I assum?
02:23.25JTcompile zaptel, then asterisk
02:23.27JTyes
02:23.29Cyber-Doggok
02:23.44Cyber-Doggso... since  have both already installed... can I just redo asterisk?
02:24.26JTi guess so
02:24.34Cyber-Doggok... here goes nothing
02:25.11*** join/#asterisk n00dle (n=ccraft@ip-249-27.springsips.com)
02:27.19*** join/#asterisk flujan (n=flujan@201-42-103-137.dsl.telesp.net.br)
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02:33.39Cyber-Doggok... I did a rebuild...
02:35.09Cyber-Doggnow chan_zap.so is ther
02:35.21Cyber-Doggshould I do a rboot?
02:35.33Cyber-Doggor is stopping asterisk and zaptel... then restarting them enough?
02:37.06JTno need to rebook
02:37.09JTreboot
02:37.14JTyou didn't change kernel or hardware
02:37.17Cyber-Doggnope
02:37.25Cyber-Doggwell... I restart zaptel
02:37.35Cyber-Doggand then I tried to restart asterisk... but I don't think it is...
02:37.40Cyber-DoggI can't connect to the CLI
02:37.51Cyber-Doggunable to connect
02:38.09Cyber-DoggI checked the log and there isn't anything there
02:42.43*** join/#asterisk ^rocket^ (n=rocket@c-71-235-48-164.hsd1.ct.comcast.net)
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02:45.16xhelioxNOTICE[19527]: chan_sip.c:5250 process_sdp: No compatible codecs, not accepting this offer!
02:45.35xhelioxI'm seeing this a lot lately -- is there anyway for it to be more verbose?
02:45.45xhelioxE.g. The IP it's rejecting?
02:46.59*** join/#asterisk nephfl (i=nephilim@wsip-70-184-144-158.ga.at.cox.net)
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02:49.43snuffy22sip debug ;)
02:50.08snuffy22only problem is then u get the kitchen sink as well
02:50.28xhelioxyeah
02:50.32xhelioxit's only happening every few hours
02:50.52xhelioxI've allowed all codecs.. and I'm still getting that, so I think it's pretty bogus, I'm just curious :)
02:53.20*** join/#asterisk rsd99 (n=rsd095@c-71-224-187-182.hsd1.pa.comcast.net)
02:53.42rsd99when i go to listen to a voicemail, all i hear is static
02:53.48*** join/#asterisk Bryan93108 (n=chatzill@131.muf30.hrfr.wswdc01r18.dsl.att.net)
02:54.21Bryan93108can anyone tell me if SRTP is usable in Asterisk yet, with the Linksys ATAs?
02:55.32JTxheliox: do you have g.729 licenses?
02:56.11Bryan93108no but will obtaini them if needed (kinda new to SRTP) -- whom do I license it from?
02:56.48JTBryan93108: i wasn't talking to you there :)
02:56.54JTunless you're xheliox
02:56.55Bryan93108doh
02:57.40[TK]D-FenderBryan93108, No.
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02:58.15rsd99i just finished setting up voicemail.  when i go to play a new vm, all i hear is static
02:58.18rsd99any ideas?
02:58.24^rocket^I installed asterisk opn Xubuntu, and port 5060 isn't open when I do an nmap scan, what am I doing wrong?
02:58.26xhelioxJT: Hmm. I might not on that box...
02:58.46Strom_C^rocket^: why are you running asterisk on a box with x windows?
02:58.58JTxheliox: perhaps a client is requesting a codec with no transcoder available
02:59.08JTwhich'd most likely be g.729
02:59.11^rocket^it's running on an iMac, someone gave me a boot disk for Macppc
02:59.20Strom_C.....
02:59.27xhelioxYeah, I'm willing to accept that ;) Checking now...
02:59.27rsd99is the mac running OSX
02:59.28^rocket^I had poor results booting other distros
02:59.44^rocket^rsd99 no
03:00.07^rocket^It's Linux muffin 2.6.15-26-powerpc #1 Fri Sep 8 19:51:33 UTC 2006 ppc GNU/Linux
03:00.09rsd99just curious
03:00.20[TK]D-Fender^rocket^, SIP uses *UDP*, not *TCP*.  You'd have to adjust your scan.
03:00.53rsd99i used to tun macppc back in the day. ;-)
03:00.56xhelioxHmm, yeah, I have g729 licenses installed, good thought though :)
03:01.01*** join/#asterisk HockeyInJune (n=HockeyIn@pool-70-18-14-219.nycmny.east.verizon.net)
03:01.25JTg.723, iLBC, or a few others could be culprits
03:01.37xhelioxjust should be nice to know where it's coming from
03:01.50xhelioxshould/would
03:02.03^rocket^[TK]D-Fender: nmap -PU shows no port 5060 open
03:02.19^rocket^I can't connect a P2000W to it
03:02.41*** join/#asterisk Cyber-Dogg (n=jphelps@24-178-240-97.dhcp.stls.mo.charter.com)
03:02.46Cyber-Doggok... I'm back again... still no more progress
03:02.54JTa P2000W, yes, that
03:03.01Cyber-Doggchan_zap exists... but I can't load asterisk
03:03.23^rocket^nor my IAXy
03:03.32JTCyber-Dogg: is zaptel.conf and zapata.conf setup?
03:03.35Cyber-Doggyes
03:03.47[TK]D-Fender^rocket^, your failure to properly setup a specific device does not mean much...
03:03.47JTdid you run ztcfg?
03:03.49Cyber-Doggand ztcfg -v yields the channel map appropriately
03:04.11Cyber-Doggwell.. ztcfg -vv
03:04.13^rocket^well, shouldn't a port scan show open ports?
03:04.25^rocket^they have in the past with other setups I did
03:04.36^rocket^I had no problems with asterisk on debian or BSD
03:04.41JT^rocket^: check netstat
03:05.10Cyber-Doggzapata.conf is setup as well
03:05.17^rocket^JT: I do see open ports with tha
03:06.30JTso clearly you have a firewall/network issue
03:06.50^rocket^JT: k
03:06.59*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)
03:06.59*** mode/#asterisk [+o anthm] by ChanServ
03:07.49Cyber-DoggI am supposed to edit the zapata.conf in /etc/asterisk right?
03:09.56[TK]D-FenderJT, nothing is clear right now.  we have no idea if his configs are sane
03:11.40^rocket^I'm gonna reboot and see if that helps
03:11.48JT[TK]D-Fender: well he said that netstat had the sip port as open, if he was right about this, the issue is networking related :)
03:12.05^rocket^I had two different IPs listening on UDP 5060
03:12.18Cyber-DoggI'm certian that zaptel.conf and zapata.conf are right
03:12.26Cyber-Doggthe issue has to be with asterisk somewhere...
03:12.27[TK]D-FenderJT, or the scan was done wrong.
03:12.34Cyber-Doggor some other config issue
03:12.49JT[TK]D-Fender: yes that's possible, but assume he also can't connect with some sip stuff
03:12.52^rocket^nmap -sU shows only DHCP listening
03:13.46[TK]D-FenderCyber-Dogg, never assume your configs are right and DEFINITELY never think for a second they we assume it.
03:13.59Cyber-DoggLOL... ok
03:14.26[TK]D-FenderCyber-Dogg, first rule... if everything was right... it would WORK.  So clearly somethings ^&%#ed up.
03:14.36Cyber-Doggright...
03:14.57Cyber-Doggwell... zaptel.conf is pretty straight forward...
03:14.58[TK]D-FenderCyber-Dogg, Next tip : Never nag about a problem without providing a DETAILED pastebin including everything relevent to the problem.
03:15.16*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
03:15.33JTthird rule: too many dots make the baby jesus cry
03:15.48[TK]D-FenderCyber-Dogg, next tip : Never try to explain why you think your configs are right without SHOWING THEM.  Again it is MY automatic assumption that you have in all likelyhood screwed up EVERY file you may have gotten your hands on :)
03:15.52*** join/#asterisk iratsu (n=iratsu@modemcable090.239-59-74.mc.videotron.ca)
03:16.29Cyber-Doggalright alright alright,,, I'll go make a pastebin
03:16.45[TK]D-FenderCyber-Dogg, good, because I was running out of rules :p
03:16.57Cyber-DoggI was hoping I hadn't done much more wrong...
03:16.59JTcommas are basically dots for the purposes of my rule ;)
03:17.02Cyber-Dogg;-)
03:17.10Cyber-DoggLOL sorry it's a habit
03:18.20nephflDon't forget the tip that since people in IRC are not paid to help, they have no incentive not to be an ass, so always expect to get what you pay for and consider yourself lucky when you get more. (not to say there aren't a ton of very helpful people in irc, but there are alot of asses too)
03:19.40nephflShoot, even when you pay for support you get mixed results.
03:20.00*** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com)
03:20.44Cyber-Dogghttp://pastebin.com/932351
03:22.02Cyber-Doggas mentioned, ztcfg displays my channel map appropriately but after I execute asterisk, I am unable to connect to the CLI
03:22.11Cyber-Doggand the log has nothing in it
03:22.16*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
03:22.25blitzragezup hombres
03:22.55[TK]D-FenderCyber-Dogg, how are you starting *?
03:23.05Strom_CCyber-Dogg: what happens when you start asterisk by typing "asterisk -cvvvvvvvg"?
03:23.34[TK]D-FenderCyber-Dogg, indeed, start it exactly as shown above
03:24.04Cyber-Dogghey! now that freaking helpful
03:24.12Cyber-DoggI wish I would have known that before :-)
03:25.31Cyber-Dogga few notices
03:25.34Cyber-Dogg2 warnings
03:25.44blitzragedoes it die at some point?
03:25.58Cyber-Doggyes, loading module chan_oss.so
03:26.56*** join/#asterisk apardo (n=deal@26.144.217.87.dynamic.jazztel.es)
03:27.18Cyber-Doggit just says failed
03:27.20Zion800Does anyone have any experience setting up Broadvoice with Asterisk?
03:27.53*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
03:28.51[TK]D-FenderCyber-Dogg, add "noload => chan_oss.so" to modules.conf
03:29.22Cyber-Doggwhere is the modules.conf?
03:29.48Cyber-Doggn/m
03:29.48JTwhere all your asterisk configuration files are
03:29.51[TK]D-FenderCyber-Dogg, same folder as the rest of *'s configs
03:30.25JT[TK]D-Fender: keep your windowsisms away ;)
03:30.43Cyber-Doggok, I added that
03:31.05Cyber-Doggretry the asterisk -cvvvvvvvvvg
03:31.06Cyber-Dogg?
03:31.11*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
03:31.18JTthat'd be the idea
03:31.27JTi'm sure you can work out what these steps are doin
03:31.29JTdoing
03:31.33Cyber-Doggyes
03:31.55Cyber-Doggwell the reason I asked was because I had already tried the asterisk -cvvvvvvvvg and it still says the same thing
03:32.03Cyber-Doggso I was hoping there was an additional step yet ;-)
03:35.12[TK]D-FenderCyber-Dogg, pastebin the failure so we can see
03:35.18[TK]D-FenderCyber-Dogg, and your modules.conf
03:35.37Cyber-Doggetc/modules.conf right?
03:35.49JTno
03:35.52JT<PROTECTED>
03:36.08Cyber-Doggwell... that would be a problem
03:36.21Cyber-DoggI don' thave a modules.conf ther
03:36.44Cyber-DoggI have one in /etc/asterisk though
03:36.47Cyber-Doggerr... /etc
03:37.24*** join/#asterisk nowork (n=jfu2808@216.254.141.97)
03:37.26Cyber-Dogghttp://pastebin.com/932362
03:37.51noworkhi, in asterisk CLI, what command I can check if the call is proxy mode or not? thanks
03:38.10Strom_CCyber-Dogg: pastebin.ca seems to be sucking less than pastebin.com, fyi :)
03:38.25JTCyber-Dogg: wrong modules.conf
03:38.32JTCyber-Dogg: remove the line from that file
03:38.35Cyber-DoggI did
03:38.47JTit sounds like you didn't make the sample asterisk configurations
03:39.10Cyber-Doggdoesn't ring a bell :-)
03:39.37JTif there's nothing in /etc/asterisk, it's likely you didn't make them
03:39.39[TK]D-FenderJT : I was right behind you on that :)
03:39.47Cyber-DoggI have some files, just not that one
03:40.17[TK]D-FenderCyber-Dogg, backup the configs you know you did by hand (extensions.conf, sip.conf, zapata.conf, etc....) and do a "make samples"
03:40.18JTsounds defective
03:40.47Cyber-Doggwhere do I need to be to do the make samples?
03:40.50Cyber-DoggI'm on gentoo
03:40.54noworkhi,TK>.. in asterisk CLI, what command I can check if the call is proxy mode or not? thanks
03:41.03Strom_CCyber-Dogg: asterisk source directory...
03:41.06[TK]D-Fendernowork, * is NOT a proxy.
03:41.44noworkTK..hm, I just don't want my client see my SIP DID provider 's ip address.
03:42.06noworkTK..I don't know what term to express this..
03:42.12[TK]D-Fendernowork, "canreinvite=no"
03:42.18Cyber-Doggwould that be /usr/portable/net-misc/asterisk
03:42.24Cyber-Doggcause it doesn't wrok ther
03:42.27Cyber-Doggit says no target
03:42.32[TK]D-Fendernowork, use gratuitously throughout sip.conf
03:43.04JTCyber-Dogg: if you can't work out gentoo, i suggest you download the source from asterisk.org
03:44.19[TK]D-FenderPackaged * = ASS
03:44.27noworkTK..thank you..
03:44.52*** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net)
03:44.53JT[TK]D-Fender: it was compiled in gentoo from a gentoo package
03:45.24[TK]D-FenderJT : should have been labeled "Fruit Loops".
03:45.40JTheh
03:47.21Cyber-DoggI'm trying to find it
03:47.38JTasterisk.org > download
03:47.59rob0Cleverly hidden there!
03:48.51Strom_Ctotally impossible to find!
03:50.24Cyber-Doggeh... I just made my own modules.conf and it works now LOL
03:51.10JTrofl, i guess
03:52.33*** join/#asterisk bmg505 (n=leon@196.209.177.223)
03:52.37Cyber-Doggwoot
03:52.58Cyber-Doggok!
03:53.14Cyber-Doggso, should I use SIP or IAX for two systems?
03:53.33JTup to you
03:53.50Cyber-Doggwhich would you recommend?
03:54.10JTdepends on what you're trying to do
03:54.35Cyber-Doggjust be able to call from one system to the other and use voicemail across them too
03:56.18blitzragenicer to use SIP across all of them, but it's a bit of a pain with SIP
03:56.25blitzrage(at least until TFoT2 comes out)
03:56.33rob0If doing NAT, IAX is easier.
03:56.34blitzragesince i documented that scenario this time around
03:56.42*** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net)
03:56.59*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
03:57.13rob0And both are pretty well covered at the Wiki ... Asterisk+dual+servers
03:57.16JTif doing a big multiserver cluster, SIP has advantages
03:57.27JTlike being able to use OpenSER
03:57.37Cyber-Doggjust two
03:57.45Strom_Cblitzrage: is a draft of TFoT2 available, or will we just have to wait? :)
03:58.00blitzragejust have to wait at this point -- we're in copyedit mode right now
03:58.07Strom_Calright
03:58.23blitzrageI have about 500 pages to go through (and is about 2/3s of the book :))
03:58.38Cyber-Doggwell, I'm goign to save that task for tomorrow
03:58.48Cyber-DoggI really appreciate the help and the hints and tips too! :-)
03:59.03Cyber-DoggI'll be sure to use the pastebin in the future
03:59.26Cyber-Doggok time to crash...
03:59.31Cyber-Doggneeded to get that last ... in there ;-)
03:59.34Cyber-Doggg'night all
03:59.37blitzragenight
04:01.37Strom_Ctomorrow afternoon!!
04:01.43Zion800strom!
04:01.52Strom_Chi?
04:01.58Zion800its michael!
04:02.04Strom_Coh hey! :)
04:02.14Zion800u still in au?
04:02.26Strom_Cyeah
04:02.36Zion800hows ur class?
04:03.52Strom_Cgoing well
04:04.10Zion800r they smart?
04:05.32rue_mohrdoes asterisk support call waiting lights on analog phones?
04:05.47[TK]D-Fenderrue_mohr, Yes
04:05.55rue_mohrI been told its done with dtmf tones while on hook...
04:05.57rue_mohrcool
04:06.11[TK]D-Fenderrue_mohr, More specifically Digium's analog cards and jsut about every ATA I've ever seen
04:06.27rue_mohrhave to dig up some docs for that and mod my phones
04:06.30[TK]D-Fenderrue_mohr, Oh wait.. CW lights... well thats the PHONE's job
04:06.46Strom_Crue_mohr: it's not DTMF tones
04:06.49Strom_Cit's FSK data bursts
04:06.55rue_mohrno, message in voicemail thing
04:06.56Strom_CZion800: yeah, they're having a good time
04:07.03rue_mohrdidn't mean call waiting
04:07.11rue_mohrwhat you thought I meant was what I did
04:07.19rue_mohrhmm
04:07.25Strom_Crue_mohr: the analog card sends an FSK data burst that says "turn the MWI on"
04:07.31Strom_Chow the phone actually handles that is up to the phone
04:07.37rue_mohrsame fsk as call display?
04:07.44Strom_Cwell, similar
04:07.47Strom_Cbut not identical :)
04:07.54rue_mohrsame baud rate?
04:07.59Strom_Cyeeesssssss
04:08.10Strom_Cjust different data encoded
04:08.15rue_mohrsame freq for 1/0?
04:08.19rue_mohryea
04:08.21rue_mohrhmm
04:08.35rue_mohrmight have a use for the parts from those call displays after all
04:08.39rue_mohroh wait
04:08.41rue_mohridea
04:08.54rue_mohryou dont have to RING to send call display data do you
04:08.55Strom_Crue_mohr: jesus, can you please stop pressing enter every three words?
04:09.05rue_mohrsorry
04:09.12[TK]D-Fenderrue_mohr, Ditch that analog crap and buy a decent SIP phone :)
04:09.26rue_mohrok, whats your credit card number again?
04:09.55Strom_C4417 0131 1555 2368
04:09.57blitzrageI DON'T WANT TO MEET YOUR MOM
04:10.05rue_mohrI have a bunch of call display modules, I'm thinking with a few tweeks, I could send messages with them
04:10.22Strom_C"call display" -- you must be canadian
04:10.26*** join/#asterisk centrex (n=mythtv@c-68-62-167-203.hsd1.al.comcast.net)
04:10.31rue_mohr?
04:10.34Strom_Chere in the rest of the universe, we call that "caller ID"
04:10.40rue_mohryea that
04:10.58rue_mohrhow do i do this enter key reduction thing?
04:11.06rue_mohrI know, I go to bed
04:11.46[TK]D-Fenderblitzrage, I JUST WANT
04:11.52blitzrage! ! !
04:11.59*** join/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net)
04:12.01[TK]D-Fenderz0mg
04:12.06blitzrageanyone know what the MS Exchange replacement software is for Linux?
04:12.13blitzragez0mghawt
04:12.32kiscokidblitzrage Postfix?
04:12.38[TK]D-Fenderblitzrage, Ximian Evolution
04:12.48[TK]D-Fenderblitzrage, Oh, exchange, not Outlook
04:12.55blitzrageexchange :)
04:12.57kiscokiddepends on what functions of Exchange you want
04:13.11blitzragewell, I'm just curious to look
04:13.12[TK]D-Fenderblitzrage, There is always OpenExchange.
04:13.16blitzrageaha
04:13.27*** join/#asterisk justdave_ (n=dave@unaffiliated/justdave)
04:13.30blitzrageI'm just curious of the list of features it supports right now
04:13.35[TK]D-Fenderblitzrage, semi-free or SomethingOrAnother....
04:13.53[TK]D-Fenderdon't fully recall....
04:14.04blitzrageeh?
04:14.09[TK]D-FenderHead office... heck MY office poos on OSS these days...
04:14.21blitzrageah
04:14.22kiscokidalso check out Postfix mail server
04:14.45blitzragewhat I'd really like is push mailing to my cell phone (Nokia E61i)
04:15.11JTkiscokid: postfix is great for smtp, it is not an exchange replacement, however
04:15.40kiscokiddoesn't postfix do imap and pop3?
04:16.09blitzrageand I'm not even using Exchange now (thank god), I use Google for my Calendar and Email... just looking at some other things for interest
04:16.16blitzragealthough I should go to bed ... got a flight in the morning
04:16.17blitzragenight all!
04:16.21JTkiscokid: no
04:16.33JTimap and pop3 are also NOT exchange replacements
04:17.00JTpostfix can work together with stuff which does imap and pop3 access
04:17.42kiscokidok, I stand corrected
04:18.27JTi'm thinking you've never used outlook/exchange
04:18.43rob0I know what he's talking about but can't think of the name now.
04:19.11rob0Someone (novell?) built a groupware suite based on Postfix.
04:19.37kiscokidI'm trying to understand the physical wiring aspects of my current Norstar/Nortel PBX and what I would need to replace the PBX with *...
04:19.47JTi think there have been a couple of solutions to try and replace exchange functionality
04:20.36rob0Suits seem to crave it. Geeks don't seem to care enough to make a free software solution.
04:20.43kiscokidcurrently have 5 analog lines with "hunt" and one DID analog trunk (which provides access to 20 extensions)
04:21.24JTkiscokid: i'd be tempted to replace all the lines with a fractional PRI
04:21.50kiscokidIf I looked at the wiring of my PBX would I expect to find 5 cables terminated by RJ-11 plugs ?  Would there be one for the DID?
04:21.53*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)
04:22.20JTdepends, the more professional stuff is terminated by punchdown blocks
04:22.30*** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net)
04:22.37kiscokidwell there is a punchdown block
04:23.25rob0Zimbra!
04:23.58kiscokidwhat about the DID?  Does it have a separate cable?
04:24.48kiscokidor does the DID just control which extensions can be called on the 5 analog lines?
04:25.11JTsince it's analogue, probably seperate
04:25.17JTas analogue has crap signalling
04:25.30JTdistinctive ringing could also do the trick, but it's hackish
04:26.39kiscokiddon't think we're using distinctive ringing since we have 20 extensions
04:28.43JTdoes each extension have a different did?
04:29.20kiscokidnot sure I understand the question.  Each extension has a different number
04:29.58JTthen the answer is yes
04:30.09JTif that's analogue, i have no idea how it's done
04:30.14JTmaybe some strange line signalling
04:30.35[TK]D-FenderJT, DTMF on initial connect prior to bridge
04:30.37*** join/#asterisk Splat (n=splat@home.heehawhills.com)
04:30.46JT[TK]D-Fender: nasty
04:30.56JT[TK]D-Fender: does it have a proper name? and does ast support it?
04:30.58[TK]D-FenderJT : ANALOG ;)
04:31.13kiscokidhow would fractional PRI help?
04:31.20JTanalog is a horrible abomination of "analogue" :)
04:31.25[TK]D-FenderJT : yeah * would support it.  basically your IVR has got to grab it fast
04:31.31JTkiscokid: digital signalling is simple to work with and reliably
04:31.54[TK]D-FenderJT : it's called a homonym ;)
04:32.42Strom_Chomophones are words which sound alike but suck each other off in the men's restroom
04:34.09[TK]D-Fenderkiscokid, like I told you yesterday, diconnect 1 line from the PBX and hook an analog line on and LISTEN.
04:34.41kiscokidok, what would I listen for?  DTMF?
04:34.52JTjesus
04:34.54[TK]D-Fenderkiscokid, exactly
04:35.27Strom_Chow to work with asterisk
04:35.28Strom_Cstep 1:
04:35.32Strom_Cturn your brain on
04:36.02kiscokidFender: what would I have to connect to the line?  An analog phone?
04:36.05rob0I *knew* I forgot something.
04:36.14*** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku)
04:36.20waKKuhi folks.. good evening
04:36.46waKKucan someone explain to me: wtf a new exten i had create doesnt work with Playback app ?
04:37.20waKKuthis says me that doesnt locate carried-away-by-monkeys in any format... (i use this file in ALL my dialplan and works fine)
04:37.20rob0Because you did something wrong. :)
04:37.26waKKuno rob0 ;/
04:37.38rob0paste the line here
04:37.45waKKuok.. pastebin
04:37.52rob0just one line, can put it here
04:38.03rob0exten =>
04:38.40rob0exten => 0,1,Playback(tt-monkeys)
04:39.37rob0That's the name, "tt-monkeys". Do you have one called "carried-away-by-monkeys"?
04:39.54waKKurob0 http://pastebin.ca/578061
04:40.06waKKui already try another file too
04:40.07rob0My "carried-away-by-monkeys" was eaten by weasels.
04:40.15[TK]D-Fenderkiscokid, Yes
04:40.41waKKuthe only playback line is: exten => 19062007,2,Playback(health-center)
04:40.57waKKu2 lines down has other playback that not work too
04:41.36JTtry Playback(tt-monkeys) instead?
04:41.48[hC]waKKu: by the looks of it that sound file does not exist or has incorrect permissions in your sounds directory
04:41.49waKKuno.. but try beep, good, goodbye
04:41.59waKKulet me check
04:42.05kiscokidFender: this is an unknown area for me.  When I plug the phone into this line will it act like a regular analog line like I have at home?  Will I get dialtone?
04:42.08[hC]waKKu: which should be /var/lib/asterisk/sounds by default i believe
04:42.44[hC]you can check what directory its using in asterisk.conf, look for 'astvarlibdir' it should be /var/lib/asterisk
04:42.54[hC]then the sounds directory is 'sounds' after that
04:42.58[hC]and asterisk needs to be able to read the file.
04:43.20JTkiscokid: yes, just plug it in already
04:44.17*** join/#asterisk JT_ (n=jon@unaffiliated/jt)
04:44.46kiscokidJT: well, I'm not at work so I can't.  Also, I have to figure out how to make a cable that goes from a punchdown block to an rj-11 socket
04:45.40waKKuyeah.
04:45.42*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
04:45.47JTkiscokid: it's only 2 wires
04:47.06kiscokidJT: ok lets say I get it connected to a phone.  Then I call a DID with my cellphone.  What will happen with this analog phone?  Will it ring?  When I pcik it up will I hear the DTMF?
04:47.51JTthat's what you'll find out
04:48.12JTif it's all the one group of lines, you'll probably want to try line 1
04:48.51*** join/#asterisk gardo (n=gardo@121.97.198.153)
04:48.55kiscokidJT: I guess you mean line 1 of the hunt group
04:50.11JTyes
04:50.31JTi shouldn't need to spell out every little obvious detail :)
04:51.11kiscokidJT: If I hear the DTMF would that imply that a digium or other analog card would properly handle the DID ?
04:52.10kiscokid(hopefully once I learn about this I'll be able to help other folks on here)
04:52.55JTkiscokid: [TK]D-Fender was saying you should drop it into an IVR and the IVR would be able to pick up the dtmf, if that's how it's done
04:53.13JTi'd be inclined to get rid off the analogue crap and replace it with digital myself
04:53.18JTs/off/of/
04:53.47JTi really hate analogue phone lines for pbxes, especially if it's for a business
04:54.29Strom_Cfor small numbers of lines, ISDN BRI is the way to go if you're not in north america :)
04:56.12JTStrom_C: fractional 10ch PRI is cheaper than about 5 channels (2-3BRIs) worth of connectivity here, if you can get an Optus PRI :)
04:56.27Strom_Cah, ok
04:56.31JTbut yes
04:56.42Strom_C....I have an Optus SIM
04:56.44JTBRI is an excellent filler for the lower numbers
04:56.47Strom_Ctheir customer service is crap
04:56.55JTheh
04:56.55Strom_Cfor prepaid anyway :)
04:57.09JTthey're better than telstra, for pri
04:57.19JTtelstra is the only one offering bri, hence the expense
04:57.26Strom_Cfrom what I'm told, Telstra is the GTE of Australia
04:57.29waKKudamn..
04:57.45JTnot sure what gte is, but the answer is probably yes :)
04:57.48waKKuthis is using /usr/share/asterisk/sounds instead of /var/lib/asterisk/sounds (defined varlibdir = /var/lib/asterisk)
04:58.10JTalso, call costs through telstra are substantially higher than optus
04:58.19waKKuthe only entry for /usr/share/asterisk is agidir = /usr/share/asterisk/agi on asterisk.conf
04:58.21waKKuweird
04:58.28Strom_CwaKKu: you installed from a package, didn't you
04:58.29*** join/#asterisk matsk (i=matsk@h110n2fls32o882.telia.com)
04:58.42waKKuStrom_C dunno.. wasnt me
04:58.47Strom_Clame
04:59.03JTStrom_C: gte?
04:59.13waKKubut i guess no.. /usr/src/asterisk* exists
04:59.20Strom_CJT: General Telephone & Electronics
04:59.30Strom_CJT: the crappy north american phone company
04:59.30JTah
04:59.55JTwere they an incumbent provider?
04:59.58*** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net)
05:06.14Strom_Cthey were the independent telco
05:06.22Strom_Cs/the in/the largest in/
05:07.07*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
05:08.33*** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca)
05:09.34JTStrom_C: telstra was the national government run telco, but has gradually been privatised
05:09.44Strom_Cyes, I know :)
05:10.14JTok :)
05:11.59*** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net)
05:13.32JTStrom_C: optus do fractional PRIs starting at AUD$20/ch/mo, minimum 10 channels
05:13.39JTit's quite competitive :)
05:18.54*** join/#asterisk iratsu (n=iratsu@modemcable090.239-59-74.mc.videotron.ca)
05:25.50*** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com)
05:27.33*** join/#asterisk bbryant_ (n=Brett@user-24-214-124-177.knology.net)
05:28.04SplatJT, you do know you can get BRI and PRI from Verizon in Australia too don't you? I don't know how they compare to Optus pricing.. but they are another option I like their local call rate.. since it's cheaper then the cheapest VoIP rates I've ever found.. heh
05:28.43*** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com)
05:35.19JTSplat: bri, it's probably resold telstra bri
05:35.37*** join/#asterisk oej (n=olle@apollo.webway.se)
05:35.55*** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com)
05:37.04*** join/#asterisk RyanW (n=cableguy@ge0-0-15-lns0.207alg.qx21.net)
05:37.40*** join/#asterisk inv_arp[work] (n=junya@c-71-229-122-61.hsd1.fl.comcast.net)
05:37.53RyanWwhat does "Got a FRAME_CONTROL (5)" mean ?
05:40.30waKKuargh.. i cant use System(); ... i solve two possible problems: asterisk is running with -U asterisk, i give a shell bash to user asterisk - and i set permissions to asterisk runs these shellscript... can someone help me with it ?
05:41.06JTSplat: cheaper than the cheapest voip rates, with what sort of minimum spend? :)
05:41.17waKKuthis shows me on CLI running... but scripts just make a "echo teste > /tmp/works.log"
05:41.40waKKuand doesnt work.. i ran it manually and works ok
05:41.49inv_arp[work]hmm how to slow this registration down...   sip.inphonex.com:5060           6762000             20 Registered
05:41.59*** join/#asterisk steliosk (n=Stelios@62.169.217.209)
05:42.08inv_arp[work]every 20sec..
05:43.53*** part/#asterisk RyanW (n=cableguy@ge0-0-15-lns0.207alg.qx21.net)
05:45.46waKKuforget.. again..
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06:18.16*** join/#asterisk Nuitari (n=Nuitari@142.46.207.230)
06:18.44NuitariHi
06:19.10NuitariIs there a way to have SIP hints w/o limiting the number of calls someone can have?
06:24.31*** join/#asterisk bintut (n=bintut@203.125.63.150)
06:26.20*** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl)
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06:30.11Splatjt: on a BRI around $62.52 +GST per month with 8.8 cent local calls and the qualifier was the cheapest VoIP rates that *I* have found.. the main thing was I needed the lines and they had better rates then Telstra so I ended up only needing to put STD calls through VoIP.. heh
06:30.45Nuitarinaughty...
06:31.04JTnaughty?
06:31.17NuitariSTD...
06:31.17JTSplat: that's telstra resold
06:31.27JTSplat: expensive line charges
06:31.37JTNuitari: subscriber trunk dialling
06:32.49creativxor...
06:32.50creativxsexually..
06:32.52creativxehh nevermind
06:32.59Nuitarifinally someone gets it
06:33.18SplatJT: I figured as much.. for the simple fact that no one other then telstra actually has anything in the Collinsvale exchange.. so that left Verizon as a better deal then telstra.. heh
06:33.21Nuitariis there any way to make sip presence work in 1.4 w/o limiting how many calls someone can do ?
06:33.26JTi knew what you were implying, it just wasn't very funny :P
06:33.47JTSplat: i assume you don't need that high a call volume
06:33.54JTsince it's just rebilled
06:34.16JTSplat: optus is still king of line rental costs, if you can get them
06:34.52Splatnah.. that was basically just so I could play with asterisk on isdn.. heh
06:35.21*** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl)
06:35.32JTSplat: 10chs with optus is cheaper than 6 with telstra
06:36.44Splatyeah.. most things are cheaper then telstra.. it's just that optus doesn't really have anything outside of the CBD in Hobart..
06:37.34JTah damn
06:38.02JTyou could chuck a pri into a datacentre there, but then you're betting on your Internet link
06:38.35JTSplat: what are the costs like for other verizon calls?
06:38.43*** join/#asterisk saftsack (n=oliver@p54A7D363.dip.t-dialin.net)
06:39.47SplatI wouldn't need a PRI in my house.. well not anytime soon.. maybe if I had 3 teenage girls in the house I'd need a PRI to be sure I can make or receive a phone call.. :P
06:40.57*** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-194-128.tx.res.rr.com)
06:41.12JTah, i didn't know this was just for your house :P
06:41.50IOscannerI am having an issue with asterisk if I just run ps -ef or ls -al on a large directory the audio stream breaks up.
06:41.56IOscannerI tried to renice to -20
06:42.00IOscannerstill does the same thing
06:42.09IOscannerany ideas what is causing this
06:42.15IOscannerI have plenty of memory
06:42.18IOscannerand CPU
06:43.32webmanIOscanner: what filesystem and HDD are you using, and what parameters do you start asterisk with?
06:43.44SplatJT, didn't have anywhere else to play with it at the time.. of course now there's on in the office where I supposedly work but haven't been to for ages.. heh and I got rid of the one I had here.. for now.. until I have a real use for one.. heh
06:44.06IOscannerUsing sata II
06:44.15flenderssplat: I have 7 POTS line with tesltra, which cost us 34.50/line
06:44.17IOscannerSegate 250 .NCQ
06:44.28flenderssplat: a 10ch PRI with optus is 200
06:44.34IOscanner-f
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06:45.09webmanIOscanner: try to add -p to asterisk startup.... you still didn't specify what filesystem you use ...
06:45.34IOscannerext3
06:45.35IOscannersorry
06:45.37IOscannerwhat is -p
06:46.08Splatyeah.. $100 a month cheaper then Verizon for 10 channels.. what are their local call rates like?
06:46.13webmangives asterisk realtime priority (if you start it as root), so if asterisk screwwwws up then your whole box can be hosed
06:46.58webmanbut it should also give better performance for asterisk, which is usually what you want :)
06:47.46IOscannernope still the same thing
06:49.29webmanIOscanner: how many files in the directory
06:50.04IOscanner50 or so
06:50.09webmanflenders: any idea what optus call their PRI service, I can't see it on the website
06:50.10IOscannereven when I start top
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06:50.40IOscannerpretty much anything I do on the command line causes the audio stream to have problem
06:50.48JTSplat: depends on your deal, but around 10-13ex GST untimed
06:50.49IOscannerI am using ZAP lines only
06:50.54IOscannerI have 8 lines in this box
06:50.59JTSplat: Multiline
06:51.02IOscannerso it is not a bandwidth issue
06:51.17IOscannerI thought it might be, but it seem to be on the system level.
06:51.36webmanIOscanner: 50 isn't a lot ... I have a directory with over 5000 files, and don't have a problem (but I do use reiserfs )
06:51.44JTwebman: i meant that for you, Multiline
06:51.46IOscannernope
06:51.53IOscannerI have never seen this before
06:52.02IOscannerI am running 1.2.18
06:52.06webmanIOscanner: possibly CPU related then.... or IRQ problems (missing IRQ's)
06:52.26webmanIOscanner: what kernel version? have you checked with zttest ?
06:52.33webmanJT: thanks....
06:52.42IOscanner2.6.18
06:52.53IOscannernope maybe I will try that
06:53.03IOscannerI am rebuilding the kernel for full 64bit support
06:53.25justdavesince asterisk 1.4.x has this nice menuselect thing now, what file do I need to move from one source tarball to the next when upgrading to preserve my config choices?
06:54.46IOscannerzttest shows all 100% and a few 99.98%
06:54.50webmanIOscanner: something is being starved of resources, you need to find out which resource, either disk IO, memory, CPU, or something, and then fix it.... I've never had that problem, so I'm not entirely sure how to fix it....
06:55.07webmanIOscanner: what card are you using?
06:55.24IOscannerTDM400p
06:55.39IOscanner2 with 4 modules each FX)
06:55.45IOscannerFXO
06:56.09webmanIOscanner: try removing one card and see what happens
06:56.33IOscannerI can't the box is not with me.
06:56.37*** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net)
06:56.37IOscannerI am doing this from remote
06:57.08webmanIOscanner: I thought you said it was all ZAP channels, how do you connect to the ZAP if you are remote?
06:57.11JTwebman: to get a quote, basically ring optus business direct, punch in your post code, and speak to the channel partner for your area
06:57.34IOscannerI am testing with a single IAX client
06:57.42IOscannersame results with a local phone
06:57.58webmanJT: am waiting to speak to them now... thanks.... I mainly just want to move the line rental to them, as all my calls are either preferred or prefix dialled to another company already
06:58.27JTwebman: how many lines do you have?
06:58.41webmanJT: 10 channel PRI with telstra at the moment
06:58.52JTwebman: ah, $365/mo?
06:59.06webmanJT: something like that
06:59.23JTwebman: optus pris have free installation on 24mo contracts
06:59.37JTand they will probably give you a credit if you negotiate one
06:59.47flendersJT: optus hates you mate
06:59.48flenders:D
06:59.52JTthe main disadvantage is the lead time
06:59.57JTflenders: :P
07:00.06flendersbut on his case, PRI is already there
07:00.10webmanJT: can they "take" a telstra PRI?
07:00.31JTwebman: it'd need to be reterminated to their own exchange gear for the line rental savings
07:00.42JTi'm sure they can rebill, you just won't save on line rentall
07:01.26flendersJT: why the cabler who installs the E1s for optus is from telstra?
07:01.50webmanoh well, I can just wait for the new install I suppose and then cancel the telstra one later
07:01.59flendersoptus equipament on telstra exchange?
07:02.02JTflenders: it's telstra copper
07:02.07JTtelstra own all the copper
07:02.23JTthey just tag the line really, from what i can tell
07:02.44JTflenders: yes, optus equipment inside the telstra exchange
07:02.54flendersgotcha
07:02.57JTwebman: well optus will port your numbers for free
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07:03.24webmanJT: good, gotta keep my numbers, it ends in 0000 which is nice :)
07:03.37JTheh
07:03.37flendersJT: optus told me today that if I wanted to switch our ADSL2 lines to them, we wouldn't pay for line rental on those lines
07:03.45JTit can take up to 10 weeks though, webman
07:03.49JTfrom signing of contract
07:03.51JTto porting
07:04.02JTtelstra drag their feet on porting
07:04.07KpoHis it correct to use chan_local in IVR?
07:04.13KpoHto forward calls
07:04.17KpoHlike this
07:04.20KpoHexten => 1,1,Dial(Local/7094@local/n|30|tTm)
07:04.38KpoH@local is one of my contexts
07:05.11KpoHi did so because i need to bill users for callforward
07:05.16webmanJT: yeah, I really don't mind about the timeframe, as long as actual downtime would be fairly minimal..... I've been paying lots for a couple of years now, so a few more months probably won't matter
07:05.33*** join/#asterisk tsurko (n=tsurko@77.70.24.142)
07:05.34KpoHbut it seems this is not working
07:05.52webmanJT: neway, optus say they will callback within 48 hours.... so I'll see what they have to offer :)
07:06.03flendersKpoH: why don't you use account codes?
07:06.36flenderswebman: after 5PM
07:06.38KpoHflenders: what you mean? Set(CDR(accountcode))?
07:06.44JTwebman: 5mins down time for each phone number
07:06.51JTthey port them one at a time
07:06.57flendersKpoH: yeah, wouldn't it work for you?
07:07.17flendersthen on your Master.cvs file you would get all calls related to that account code
07:07.46webmanJT: hmmm, a 100 number range, thats 500 minutes... looks like it will take them all night
07:07.53*** join/#asterisk SwK (n=SwK@user-69-73-37-99.knology.net)
07:08.09*** join/#asterisk jmacz (n=jmacz@190.24.97.36)
07:08.12JTwebman: maybe they can more a 100number range faster
07:08.18JTi was talking about individual numbers
07:08.24JTmove
07:08.45flendersour porting (100 numbers) will take minutes
07:08.51*** join/#asterisk SwK (n=SwK@user-69-73-37-99.knology.net)
07:08.57flendersit'll happen next thursday night
07:09.01webmanJT: yeah, I'd like to think so, but I'm sure telstra will manage to slow things down or stuff something up :)
07:09.13flendersstart at 5PM and they reckon it'll be finished by 5:30PM
07:09.14JTflenders: were they on bri?
07:09.18flendersyeah
07:09.19webmanflenders: you are in the process of moving to optus?
07:09.33flenderswebman: yeah, 4 BRIs to 1 PRI
07:09.40JTflenders and I both have multiple sites getting optus pris in ;)
07:09.45JT<3 optus
07:09.53JTok, maybe that's a bit strong there :P
07:10.43webmanI wonder if optus can move my optus cable analog service onto the same PRI (101 numbers on the PRI) ??
07:10.43flendersJT: have you heard from them on yours?
07:11.23JTflenders: next couple of weeks is what i was told
07:11.28JTwebman: yes they can
07:11.51flenderswebman: they can port any number to the PRI
07:11.56JTthey move up to 1 number per channel for free, then you have to start paying for 100 number ranges
07:11.58flenderswell, not mobiles, I think
07:12.11JTflenders: heh, that'd be awesome if they could move mobiles ;)
07:12.19webmankewl, I can dump my old analog line as well :)
07:12.48flendersJT: yeah, I was already pretty happy that they can change the ownership of mobile numbers at anytime, so you can have you personal number on the company account
07:13.07JTwebman: i'd always keep one as a backup
07:13.17JTremember a pri only runs over 1 or 2 pairs
07:13.20webmanhmmm, I wonder if that means they will charge me to port the extra 90 numbers from my 100 number range....
07:13.26JTif you lose it, buy buy phones
07:13.32KpoHrealy strange...why IVR instead of some reaction write's in cli  -- Attempting native bridge of SIP/7325928432-b7222120 and SIP/USA Route-160-082de1b0
07:13.34*** join/#asterisk corpcomp (n=IceChat7@125-238-120-174.broadband-telecom.global-gateway.net.nz)
07:13.40JTwebman: $36.50inc gst, 100number range
07:13.40KpoHsomethink wrang with DTMF
07:13.53KpoHbut what it chould be?
07:14.07JTs/buy buy/bye bye/
07:14.18webmanJT: true... though I have a cable that runs direct from the exchange into the office... (they ran out of pairs in the street, so they ran a new cable for me)
07:14.36JTa lot of buildings have that, webman
07:14.38JTours does
07:14.44JTthey still manage to f*ck it up
07:14.54JTidiot techs at the exchange dejumpering shit
07:15.20JTi've had if happen at least twice
07:15.36JTalso, POTS is the most reliable thing in a blackout
07:16.12flendersJT: PRIs arent?
07:16.17JTflenders: nup
07:16.25flendersJT: why not?
07:16.29JTespecially since most of us stick them in power hungry PCs
07:16.38JTbecause they aren't powered from the exchange
07:16.43webmanJT: well, if we really lost the PRI, analog lines wouldn't help, would just use mobile phones :) we don't have very high call volumes anyway
07:16.51JTand you can't use simple telephones on them
07:16.59flendersJT: ok, but as long as you're on a decent UPS, you're fine, no?
07:17.07JTwebman: i'm thinking more for an emergency situation too
07:17.21JTwebman: also if you have a fax, or modems, it's easier
07:17.33JTflenders: i bet your ups can't keep up with a telstra exchange
07:17.55flendersJT: at the travel agent, we'll have a TDM400 with FXS modules on it for fax and eftpos
07:18.06flendersJT: I bet it too
07:18.10JTbanks of 2V @ 500-1000Ah lead acid batteries
07:18.11webmanJT: I want to move all my fax to iaxmodem and hylafax
07:18.19JTand backup generators that kickin within minutes
07:18.27JTflenders: hopefully it will work....
07:18.40flendersJT: I can't see why it wouldn't
07:18.50JThaven't heard eftpos success stories yet
07:18.53JTin theory it should
07:19.06flenderseftpos on FXS?
07:19.06JTbut i assume everything doesn't work with asterisk until proven otherwise
07:19.07*** join/#asterisk dijungal (n=chatzill@209.59.110.5)
07:19.07webmanflenders: I use a TDM card with a fax machine (outbound) + eftpos, works pretty well
07:19.35flenderswebman: couple of FXS modules?
07:19.51webmanI configured a new context for the eftpos/fax so that you didn't need to dial 9 to get a external dialtone :)
07:20.10flendersover here we don't dial anything to dial out
07:20.19webmanflenders: yeah, I think it is one FXO and 3 x FXS
07:20.23*** join/#asterisk SwK_ (n=SwK@user-69-73-37-99.knology.net)
07:20.35dijungalhas anyone successfully connected a inbound H.323 to asterisk?
07:20.39flenderspeople like to dial like they do at home or their mobiles
07:20.45JTflenders: also, if anything in your setup fails, bye bye phones
07:20.51JTanalogue phone = very simple
07:20.52flendersJT: true
07:21.06JTpx based pbx + csu/dsu + digital = complex ;)
07:21.14JTs/px/pc/
07:21.32webmanflenders: but how do you deal with internal extension numbers and escaping them from the public dialplan/numbering scheme?
07:21.36flendersa single pots line wouldnt hurt
07:21.37flendersI know
07:21.58flenderswebman: different contexts?
07:22.18webmanflenders: but your users only exist in a single extension
07:22.30flendersall public dialplans have to have at least 8 digits
07:22.31webmans/extension/context/
07:22.39*** join/#asterisk Diablus (n=bth@217.115.85.18)
07:22.46dijungali am trying to connect the H.323 calls from a cisco 3661 router to an asterisk box... i know asterisk can interconnect with cisco but it's not working out for me... any ideas?
07:22.53webmanflenders: so you rely on a digittimeout
07:22.55JTwebman: you don't need outbound dialling prefixes if your internal extensions don't interfere with pstn numbers
07:22.58flendersinclude the outgoing context
07:23.15JTflenders: 13XXXX is 6 digits, not 8+
07:23.23JT000 is 3 digits
07:23.25webmanJT: but how can you be sure that you won't conflict
07:23.26flendersdamn you JT!
07:23.42JTwebman: you only allow certain prefixes out to pstn
07:23.43flendersthose ones are on the dialplan
07:23.49webmanthere are also some weird special numbers like 1xxx
07:24.03flenders1234?
07:24.08JTwebman: emergency, 13/1300, 1800, local, national, international
07:24.12webmanflenders: yeah, that crap
07:24.23JTwebman: yes but if your staff don't need to dial them, don't add them to the pbx dialplan
07:24.29flendersno one ever complained here.
07:24.29*** join/#asterisk syneus (n=syneus@syneus.aemcom.net)
07:24.42flendersand you know they cost you money too?
07:24.42JTwebman: and you can be sure if you use the national numbering plan
07:24.50JTi always consult the national numbering plan
07:24.53flendersso, they can google for the number they want
07:24.56JTit's what the telcos refer to
07:25.49flendersJT: damn, the other guys are late for the meeting
07:25.52webmanhow about 1802288 or 12552 or 1223
07:26.08JTwebman: do they do anything useful?
07:26.11JTflenders: ah well
07:26.18flendersI don't even know what they're for!
07:26.23webmanone of those is the telstra number for reporting ADSL faults
07:26.36flenderswebman: we don't use telstra ADSL
07:26.37JTadd it to the dialplan then ;)
07:26.41webman1223 is directory assistance (I think local call cost/free if it still works)
07:26.57flenderswebman: google/yellow pages
07:27.12*** join/#asterisk tenzind (n=tenzind@202.144.144.11)
07:27.22JTwebman: most businesses these days don't allow unrestricted raw access to the pstn to staff
07:27.28webmananyway, the question remains, what prefix is not used by the PSTN.... AFAIK, every number has a use
07:27.32JTwebman: anyway, you can write override prefixes too
07:27.40JTto send stuff raw
07:28.01JTwebman: national numbering plan, everything is detailed there
07:28.13webmanJT: we don't allow raw access... well, except for me :)
07:28.17JTyou have to match by prefix anyway if you want to implement any form of least cost routing
07:29.09webmanJT: I read that ages ago... from memory, there wasn't an available range.... thats why I am wondering how you have managed it?
07:29.23mvanbaakheya all
07:29.25mvanbaakapp_queue.c: No one is answering queue
07:29.30mvanbaakwhat does that mean ?
07:29.41JTwebman: i use 0 for an outside line personally
07:29.49JTbut it doesn't give raw pstn access
07:29.50webmanmvanbaak: that there is someone in the queue who is not getting answered
07:29.59JTfor that you need to dial a special prefix and type in a pin :)
07:30.16flendersJT: funny that on the national numbering plan, they drop the '0' from most numbers
07:30.32JTflenders: ?
07:30.34flendersand say '9 digit number' for a sydney local number, for example
07:30.37mvanbaakwebman: this user will stay in the queue waiting for an agent to pickup ?
07:30.41webmanJT: OK, so that is basically the same as here... btw, does "000" work, or do you need "0000"
07:30.41JTah
07:30.45Shaun2222anybody use the iaxy s101 unit, can i program it with iaxyprov from a remote network?
07:30.48JTwebman: 0000
07:30.58JTwebman: everyone knows you need 0 for an outside line
07:31.07JTit's the same as hitting a line key
07:31.08flenders2 9 (NDC)9 digits9 digitsGeographic number for fixed network telephone serviceMetro: Sydney
07:31.11JTwe use an isdn key system too
07:31.16mvanbaakI just want to know wether this can be a problem
07:31.20mvanbaaklike loosing calls
07:32.08webmanmvanbaak: it depends on your configuration (timeout options configured) and how long they are left in the queue, I think the default is for them to wait forever
07:32.34webmanJT: what about a guest/visitor?
07:32.39mvanbaakwebman: thanks
07:32.41JTwebman: hmm?
07:32.55mvanbaakthey will stay forever in that queue indeed (if I look at the configs)
07:33.06JT0 is the standard in australia for getting an outside line
07:33.10mvanbaak$some_company hired me to find the fuckups in their asterisk setup
07:33.13JTthey can hit a line key instead anyway
07:33.29mvanbaaksome polish dudes configured this machine but they are missing many calls and they have some other trouble
07:33.40mvanbaakthis logline I never saw on any of our own boxen
07:33.43mvanbaakthat's why I'm asking
07:33.55webmanJT: some people have asked me to use 9 as the prefix .... some want 0.... derpends on what they had before usually
07:34.09JT1heh
07:34.32JTyou can't make 0 then 00 call 000 anyway
07:34.36JT0011 is the idd code
07:35.32webmanJT: yeah, I think the way around that one was to setup a 'internal' extension that required a password, before you could access DISA to dial your 0011 numbers :)
07:35.40*** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com)
07:35.57JTthere may be some other 00 prefixes too
07:36.17JTthe solution is to not let idiots use your phone system :)
07:36.31webmanjt: AFAIK, all the 00xx prefixes are for international access ....
07:36.49JTthere used to be 0055, maybe there's other non idd codes
07:37.15JTour phone system is heaps easy to use compared to say japan
07:37.30webmanJT: personally, I might argue that if you are too dumb to dial 0000 or 9000 in an emergency, then you don't deserve to survive (darwin theory) but the problem is I might be the one in trouble and needing someone else to call 000 for me :(
07:38.03JTheh, they can use a mobile surely
07:38.15JTor you have instructions next to your phones on getting an outside line
07:38.17webmanJT: I think the other codes like 0055 were converted to 1500 xxx xxx or something...
07:38.24JT1900 mostly
07:38.53webmanyeah, those too... but I always ban those on the telstra side as well :)
07:38.57JTin japan to call overseas, you need to dial 011 <carrier access code 010
07:39.13JTso to use NTT, 011 033 010 + country code + number
07:39.32JTit's harder to work out when the error messages are in japanese ;)
07:40.28webmanJT: but wouldn't the PBX 'auto-dial' the carrier access code and the 010 for you?
07:40.46JTwebman: pay phones aren't on a pbx :/
07:41.00JTwhich is the type of phone a foreigner is likely to try
07:41.11webmanJT: payphones?? who still uses those things :)
07:41.20webmanJT: true
07:41.28*** join/#asterisk angryuser (n=aster@df01t2-213-44-148-16.d4.club-internet.fr)
07:41.55JTwebman: umm, most australian mobile phones will not work in japan
07:42.34*** join/#asterisk jmls (n=jmls@62.49.235.130)
07:42.40JTi brought 2 different phones, i knew they would not work
07:42.43JTand they didn't :P
07:44.59snuffy22long as they are tri-band generally u wont have any problems
07:45.12JTtri band gsm would be completely useless
07:45.31JTit would need to be 3G AND have roaming to japan enabled by your carrier before you leave
07:46.16*** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk)
07:46.19zeeeshhi all
07:48.14*** join/#asterisk CelticSoul (n=CelticSo@bne75-1-81-57-10-55.fbx.proxad.net)
07:53.21JT(you can't actually buy a phone sim in japan as a non resident)
07:53.54*** join/#asterisk vgster (n=vgster@h146106.navonline.net)
08:02.05dijungalanyone has any experience with ooh323 on asterisk
08:02.19dijungali'm trying to get asterisk to pickup incoming H323 calls
08:02.48dijungalbut when the calls come in the CLI does not even budge... no debug... it's like asterisk didn't even see the call
08:02.51Strom_Cthe H stands for Headache
08:03.27JTonly in astland ;)
08:03.41dijungalnow a tcpdump shows me that the call packets are actually getting to the box... asterisk just sits there like a big dummy doing nothing to take the call :)
08:04.30dijungali've been wondering over the net for days... looking at all different h323 configurations... still no help
08:05.31JTasterisk really sucks with H.323
08:06.05dijungaloooh that's great news... i know that... but i also know in certain cases it works
08:06.45*** join/#asterisk oej (n=olle@136.240.13.217.in-addr.dgcsystems.net)
08:10.12dijungalso i guess that's it.. "asterisk really sucks with H.323"
08:12.03*** join/#asterisk yassaccan (n=yassacca@admin146.hgo.se)
08:12.33sergeedijungal: i have asterisk in production converting SIP to H.323... no problems,
08:13.15dijungali need to do that.. SIP <-> H.323 for outgoing and opposite for incoming
08:13.36dijungali'm more interested in the incoming right now... H.323 -> SIP
08:14.06sergeedijungal: which instance of h323 did you try?
08:14.23dijungalooh323
08:14.30dijungalthe one that come with asterisk-addons
08:14.55dijungalasterisk-addons-1.2.6_1.2.18-1
08:15.25sergeedijungal: chan_h323 from asterisk (trunk or 1.4) works well, it was fixewd by PCadach
08:15.51sergeedijungal: asterisk-trunk/channels/h323/
08:16.39dijungalhow do i get that installed?
08:17.36sergeedijungal: cd /usr/src
08:17.54walhalaio
08:17.55dijungaluhuh
08:19.05*** part/#asterisk jmls (n=jmls@62.49.235.130)
08:19.17sergeedijungal: svn co http://svn.digium.com/svn/asterisk/trunk asterisk-trunk
08:19.31sergeedijungal: /* you need to have subversion installed */
08:20.02dijungalk.. i should be able to YUM it
08:20.02sergeeyes
08:20.21sergeedijungal: svn co ..., will download trunk version of asterisk
08:20.52dijungalhmm.m... will that replace the asteirsk i already have installed...?
08:21.37sergeedijungal: then go to /usr/src/asterisk-trunk/channels/h323/README (or http://svn.digium.com/view/asterisk/trunk/channels/h323/README?view=markup)
08:21.43sergeedijungal: yes
08:21.54*** join/#asterisk tenzind (n=tenzind@202.144.144.11)
08:22.14dijungalk thanks
08:23.21sergeedijungal: you will need to install 2 libraries - PWLIB and OPENH323, you can download them here: http://sourceforge.net/project/showfiles.php?group_id=80674
08:24.22dijungalhmmm... my DNS not configured... :(
08:24.25sergeedijungal: i recommend you to use the following versions: Pwlib 1.10.0 and openh323 1.18.0 - these are the most stable versions
08:25.18sergeeecho "nameserver 195.94.224.3" >/etc/resolv.conf
08:27.43dijungali fixed the resolv.conf... apparently the ISPs DNS is not working so i added one i knew off hand
08:27.57dijungalwho is 195.94.224.3 ?
08:28.20sergeedijungal: so, you need to compile pwlib (only "make", not "make install"), then you need to configure and compile openh323, when you are done, go to /usr/src/asterisk-trunk, configure it (./configure --with-pwlib=/... --with-h323=/...) and then build and install your asterisk
08:28.26sergeedijungal: my DNS
08:28.29snuffy22i have a nifty one u can't forget.. its in australia though.. 61.88.88.88 how easy huh.. 3 digits :)
08:28.57snuffy22if only all dns ips could be that easy to remember
08:29.10sergeesnuffy22: nice :)
08:29.14*** join/#asterisk purplet (n=purplet@010.041.dsl.concepts.nl)
08:29.19dijungalindeed
08:30.07dijungali c alot of reference to H323 in yuh compile line will this asteirsk-trunk also support SIP,IAX and such?
08:31.04sergeedijungal: yes
08:31.34dijungalk... welll that's alot of instructions.. i better get started
08:31.52sergeedijungal: you can expect some issues, but nothing lethal, asterisk-team keeps trunk in a very good condition, big respect to them
08:32.06*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
08:32.15dijungalso what exactly is this asteirsk-trunk?
08:32.43*** part/#asterisk corpcomp (n=IceChat7@125-238-120-174.broadband-telecom.global-gateway.net.nz)
08:32.46sergeedijungal: it is also known as CVS_HEAD, it is a kind of bleeding edge release...
08:32.52JTtrunk is beta
08:33.12dijungalk
08:33.20dijungalahh makes sense now...
08:34.00HarryRseems quite stable at the moment though
08:34.48dijungallol @ HarryR
08:35.09sergeewell, sometimes this beta much better then so-called-stable :)
08:35.47sergeeanyway
08:35.54snuffy22depends what features u use really.. jabber/gtalk have been receiving lot of love in trunk
08:35.55sergeeh323 works pretty wel in trunk
08:36.20sergeesnuffy22: you are right i have no exp with those channels,
08:36.41*** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu)
08:37.07*** join/#asterisk Dibbler_ (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com)
08:37.13snuffy22i dont use them either just yet.. have enough trouble with cdr's etc 1st :)
08:37.21*** join/#asterisk ptblank (n=MURDER1@cpe-75-84-221-219.socal.res.rr.com)
08:37.57sergeeyes cdrs have issues too :)
08:38.14*** join/#asterisk waptaxi (i=kvirc@45.151-224-87.telenet.ru)
08:38.25sergeebut trunk has normal transfer, not those nasty blindxfer/atxfer
08:40.44*** join/#asterisk qdk (n=qdk@213.150.62.32)
08:40.54snuffy22mm true either way atm i'm dreading redoing the cdr tests to check if 1.4.5 hasn't broken the way we cater for the broken cdr
08:41.19*** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch)
08:42.40dijungaltrunks has issues with cdrs?
08:42.48dijungalso how do i track my calls?
08:43.10snuffy22dijungal, they dont have issues.. if your making asterisk do simple stuff
08:43.16dijungalhey u can connect gtalk to asteirsk.. sweet :)
08:43.51snuffy22if you are doing lots of complex dialplans etc.. the cdr's generally dont come to the party as you hope
08:44.38dijungallol.. k
08:45.31*** join/#asterisk clinfix (n=haa@85.98.175.207)
08:45.54dijungali accidentally did a yum upgrade... and i've been waiting for the past 30 mins for it to finish
08:45.59dijungalcan i cancel it?
08:47.11snuffy22you can.. the rpm's are generally saved in cache so it won't re-dl them
08:47.12HarryRdijungal: press control C
08:47.32dijungalk
08:47.36snuffy22if u want to kill yum while its dl'ing ctrl+c is useless.. open another terminal and kill yum :)
08:48.01snuffy22ctrl+c while dl'ing just makes it skip to the next mirror (at least in centos)
08:48.09clinfixHi , I am a computer engineer( new graduated),I want to do a voip project. Do I need a PBX like asterisk?
08:48.19JTprobably
08:48.22HarryRsnuffy22: ctrl+z then kill :)
08:49.01*** join/#asterisk elzapp (n=chatzill@bkkb-gw.voop.net)
08:50.34dijungalthe ctrl+c worked
08:50.38dijungalits stopped... lol
08:50.56dijungalnow i'm getting the asterisk-trunk :)
08:50.58dijungaldownloading
08:51.30sergeeguys, btw, anybody from europe here at moment? (UK,NL,DE,CY)?
08:51.48dijungalwow does the trunk come with g729, i c a .c fiile in formats for g729
08:52.02sergeedijungal: it is not the codec
08:52.03dijungal"formate_g729.c"
08:52.11dijungaloooh :(
08:52.40dijungalSergee: no but i'm gonna be going to the UK for vacation :)
08:53.07dijungaland prolly look for a job.. if i'm so lucky
08:53.07*** part/#asterisk clinfix (n=haa@85.98.175.207)
08:53.16sergeedijungal: i have some questions about legal issues related to voip in EU, trying to get any info :)
08:53.52dijungalahh... can't help u.. i berely know how to get around ... lol
08:54.01dijungalok trunk downloaded
08:54.40CelticSoulCan someone explain why there is SIP in IAX?
08:55.06*** join/#asterisk drray (n=drray@c-67-170-9-176.hsd1.wa.comcast.net)
08:56.24*** join/#asterisk floppp (n=flop@nat-staff.b3g-telecom.com)
08:57.25dijungalsergee: u suggest i use openh323 1.8 instead of 1.9
08:57.50sergeedijungal: yes, 1.18.0 and pwlib 1.10.0
08:57.58dijungalhmm.. k
08:59.26*** join/#asterisk Dr-Linux (n=asfdf@DSL-202-59-73-131.nexlinx.net.pk)
08:59.40dijungalpwlib does not have a 1.10.0 on sourceforge
08:59.47dijungali will do the 1.10.3
09:00.13Dr-Linuxtrying trying .. but no solution for this zaptel issue
09:00.14Dr-LinuxZT_CHANCONFIG failed on channel 1: No such device or address (6)
09:00.38*** join/#asterisk Strom_M (n=strom@dsl-202-173-183-69.vic.westnet.com.au)
09:00.43Dr-Linuxany clue?
09:02.02sergeedijungal: find 1.10.0, you can have very bad experience with versions other then i stated
09:02.14Strom_M1.10?!
09:02.14dijungalk
09:02.33JTpwlib
09:02.34Strom_Mthat's not an asterisk version, I hope :)
09:02.36Strom_Mah, ok
09:02.55sergeedijungal: http://sourceforge.net/project/showfiles.php?group_id=80674&package_id=89974
09:03.46dijungalsergee: thanks
09:03.55Dr-Linuxsergee: any clue on my question?
09:04.19dijungalsergee: ok so now compile pwlib and openh323
09:04.40sergeedijungal: pwlib first then openh323
09:04.42Dr-Linuxhttp://readlist.com/lists/lists.digium.com/asterisk-users/10/51936.html << i'm having the exact issue
09:04.55dijungalk
09:04.59sergeeDr-Linux: no idea, sorry. Don't have a lot of exp with zaptel.
09:05.29sergeeDr-Linux: i have only 1 card (TE207) and it works well, no problems at all..
09:05.54Dr-Linuxmine was also working
09:06.01Dr-Linuxbut suddenly stopped
09:06.05Strom_MDr-Linux: try re-seating the card.
09:06.44Dr-LinuxStrom_M: do you mean unplug and re-plug the card/
09:06.45Dr-Linux?
09:06.57Strom_Mi mean remove the card from the PCI slot and then re-insert it
09:07.03Strom_M(after you turn the system off, of course)
09:07.14*** join/#asterisk zapp-branigan (n=zapp-bra@84.79.33.1)
09:08.18Dr-Linuxsure i'll try this as well
09:08.28Dr-Linuxit's very serious issue
09:08.37sergeeDr-Linux: http://readlist.com/lists/lists.digium.com/asterisk-users/10/51940.html
09:08.43*** join/#asterisk casix (n=casix@edifici-pub.adam.es)
09:08.45casixhello
09:08.52sergeehey
09:11.17Dr-LinuxStrom_M: is it a known issue? :S
09:12.23Strom_MDr-Linux: beats me; i'm just suggesting you do what I'd do
09:12.39Strom_Mif re-seating it doesn't help, try a different slot; if not, you probably have to RMA the card
09:12.42Dr-LinuxStrom_M: ok
09:13.13zapp-braniganhi i have a problem compiling speex :  ave a porblem compoiling speex
09:13.13zapp-branigan<zapp-branigan>
09:13.23Dr-Linuxbut i've 2 cards in this server
09:13.29zapp-branigan[codec_speex.so]Can't modify /usr/lib/asterisk/modules/codec_speex.so's text section. Use GCC option -fPIC for shared objects, please.
09:13.44sulanis it possible to listen for dtmf-digits in a bridged call (an AGI that answers an incoming calls and Dials an outgoing) and distinguish the party that pressed it?
09:14.13Strom_MDr-Linux: well, swap the cards then.  does the problem follow the card or does the problem stay with the slot?
09:15.03Dr-LinuxStrom_M: right now i'm at home, so i want al least one card should work
09:15.07*** join/#asterisk extr3m (n=nexilus@213.134.125.3)
09:15.21Dr-LinuxNOC guys told me that one card's leds are green
09:15.34Dr-Linuxbut the 2nd one has nothing
09:15.38Strom_MDr-Linux: so wait, you're asking for help and you're not even in front of the system>
09:15.39Strom_M?
09:15.54Strom_Mget the NOC guys to do it then
09:16.01Dr-Linuxyeah, i'm infront of the system now
09:16.12JTwow, fast transportation
09:16.25Strom_MDr-Linux: physically in front of the box, or just SSH'd in?
09:16.31Dr-Linuxbut remotely
09:16.37Strom_Mthat's not "in front of the system"
09:16.40Dr-Linuxssh
09:16.47Dr-Linuxok sorry
09:16.55Strom_M"in front of the system" would imply that you can physically manipulate it
09:16.56*** join/#asterisk tenzind (n=tenzind@202.144.144.11)
09:17.06Dr-Linuxyeah
09:17.29Strom_MDr-Linux: so first bring the system down and get the NOC guy to re-seat the card
09:17.31CelticSoulCan someone explain why there is SIP in IAX?
09:17.43Strom_MCelticSoul: um, there is no SIP in IAX
09:18.16Dr-LinuxStrom_M: i think i should go to work myself 40 min drive
09:18.23Strom_Mok
09:18.33Strom_Mgo to work and come back online when you're in front of the computer
09:18.36Dr-LinuxStrom_M: basically for now i want one card should work
09:18.43Dr-Linuxcurrently even asteisk is down
09:18.47Dr-Linuxservice i mean
09:18.53Strom_Mwell, stop typing then
09:18.55Strom_Mand GO TO THE MACHINE
09:18.55dijungalsergee: what does this mean "Also make sure you have added the $PWLIBDIR/lib directory to your
09:18.56JTwhat version of ast and zaptel?
09:18.57dijungal<PROTECTED>
09:18.57Strom_Mjeez
09:18.58dijungal<PROTECTED>
09:19.42Dr-LinuxStrom_M: okey thanks
09:19.46Dr-Linuxbye from where
09:20.09dijungali'm reading the install instructons for PWLIB
09:20.18dijungalwhat does this mean: "Also make sure you have added the $PWLIBDIR/lib directory to your
09:20.20dijungal<PROTECTED>
09:20.21dijungal<PROTECTED>
09:20.32Strom_Mdijungal: don't spam please
09:20.41Strom_Myou pasted the same thing one minute ago
09:21.01sergeedijungal: here are the instructions: make && make install
09:21.02dijungalk
09:21.34dijungalfor pwlib i only need to use the make
09:21.45CelticSoulStrom_M: In the example in AsteriskTFOT in extensions.conf for aix, they put: exten => 10001,1,Dial(SIP/john)
09:22.00Strom_MCelticSoul: which page?
09:22.39CelticSoulStrom_M: Page 73
09:22.43Strom_Mhang on
09:25.49Strom_MCelticSoul: you're completely misinterpreting it :)
09:25.52sergeedijungal: yes, make would be enough, but then you'll need to configure openh323 smartly, (./configure --with-pwlib=/usr/src/...)
09:26.05Strom_Mthe inbound iax connection in that example will bridge to a SIP phone called "john"
09:26.10CelticSoulStrom_M:  How?
09:26.16dijungalk
09:26.22Strom_Mthat does not mean there is "SIP in IAX"
09:27.16CelticSoulthat means IAX device -> Asterisk -> SIP phone?
09:27.21Strom_Myes
09:27.22sergeeCelticSoul: Think of Asterisk as of transport Hub, you can get there by different transport - train, airplane, ferry, etc
09:27.42sergeeCelticSoul: and you can get out of there by different transport to,
09:28.01Strom_Mooh boy, yet another pointless confusing analogy :)
09:28.11CelticSoulgot it
09:28.19sergeeCelticSoul: so if you'll get there by ferry, and get out by plane, it doesn't mean that ferry in a plane, it means that you changed transport
09:28.41CelticSoulthank you Strom_M, sergee
09:28.58sergeeStrom_M: you see? he's got it, so not so pointless and not confusing for sure
09:29.13Strom_Msergee: I was joking, hence the smiley
09:29.17Strom_MDUH :)
09:29.18sergee:)
09:29.34CelticSoul:)
09:29.37Strom_M:)
09:29.39JTbut if you have a helicopter in a plane, the plane must be a C-17
09:29.49casixanyone uses ast-rad-acc.pl for billing?
09:29.55Strom_Mand if you have cheese in a helicopter, you must also add mayonnaise
09:29.56JTand if you have a plane on a boat, it must be an aircraft carrier
09:29.56sergeeyeah, and don't forget to take a tank with you :)
09:30.15Strom_Mand let's not forget what happens when you put gasoline in the dog
09:30.33JTyou get a hot dog when an ignition source is near
09:30.55*** join/#asterisk CBU[^_^]M`` (n=love@210.213.145.71)
09:32.07sergeemmmmm hot dog...
09:32.15creativxmmm lunchtime
09:32.23sergee(c) Homer Simpson
09:36.41e-ddiemmm cold dog
09:37.50*** join/#asterisk Dovid (n=Dovid@bzq-88-155-205-93.red.bezeqint.net)
09:39.15walhalasomeone use 7960 and chan_skiiny here ?
09:39.41*** join/#asterisk andyd (n=andyd@host90-152-23-30.ipv4.regusnet.com)
09:40.32zapp-braniganthere is any way to detect the linksys spa3102 answer supervision ? by software or something else ?
09:41.39walhalano chan_skinny users ?
09:57.23dijungalwhen compiling openh323 do i have to compile g729 with it or can i just add the already compiled .o file the asterisk modules directory after asterisk install?
10:00.58tzafrir_laptopopenh323 is not related to g729
10:01.01Dovidcompile g729 = ? u mean installing the licence ?
10:01.09Dovidnm. tzafrir answered it
10:11.54*** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com)
10:12.17EmleyMoorIs there any way I can assign a caller ID name to a call when there is no number, and have X-lite detect it?
10:12.45creativxset callerid(name)
10:12.58creativxdont know how xlite has implemented blank callerid(num)s
10:13.15EmleyMoorcreativx: X-lite ignores it if the caller ID number isn't there
10:16.10creativxthen set the calleridnum to 000000
10:17.23EmleyMoorThat fails too
10:17.51*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
10:18.10EmleyMoorX-Lite still displays "Unknown"
10:18.21EmleyMoorIt displays the right name if the number is present
10:19.54EmleyMoorIs this Asterisk's problem or X-Lite's?
10:20.25*** part/#asterisk spaceinvader (n=server@unaffiliated/spaceinvader)
10:24.34creativxeh i dont think i understood your problem correctly
10:26.06EmleyMoorcreativx: X-Lite is not interested in displaying the name unless the number was actually presented
10:26.18EmleyMoor(either that or Asterisk is failing to send it)
10:26.28creativxwell
10:26.35creativxwhat if you force the cid num to something other than 0000
10:26.39creativxdoes xlite show it then
10:27.38EmleyMoorNo
10:28.23EmleyMoorWould "SetCallerPres" influence it?
10:31.57EmleyMoorAh, yes
10:32.18EmleyMoorSetCallerPres(allowed) presents "asterisk" as the "number"
10:32.30zeeeshhow to set up call conference without using zap and etc?
10:32.51*** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net)
10:34.20EmleyMoorThat's what it was all along - the withheldness was passing on
10:41.55zeeeshdialing access number through mobile . after a beep usr can dial their destination number . i need after dial the access number they can dial destination number or conference room .. bcoz if they will able to join conference room they can talk free of cost ?
10:42.59*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)
10:44.09EmleyMoorzeeesh: Do you mean you are allowing them DISA and don't want to?
10:44.11Strom_Mzeeesh: you're not making a whole lot of sense...or anything reselbling sense :)
10:44.14Strom_Mer
10:44.17Strom_Mresembling
10:44.42*** join/#asterisk cy303 (n=cy@is.trapped.in.themetaverse.org)
10:49.37*** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net)
10:51.07*** join/#asterisk codey (i=codec@iglu.paranoid-penguin.de)
10:51.12codeyhit here
10:51.15codey*hi there
10:51.19Strom_Mhit there
10:51.23Strom_Mhit everywhere
10:51.28codeyhrhr
10:51.46codeyi've got a problem with my dialplan, maybe someone can help: http://slexy.org/paste/3167
10:52.03codeyonly the first rule is working (_0NXZ)
10:52.18codeybut as soon as i call 0089whatever it doesn't match any rule
10:52.47Strom_Mcodey: um, that's kind of not how you should be setting up your dialplan
10:52.56Strom_Meach new extension should start with priority 1 :)
10:53.05Strom_Mfurthermore, SetCallerID() is very very deprecated
10:53.16codeyerr, its the outgoing "dialplan"
10:53.50Strom_Mand also, never never never use the r, t, or T options for Dial() unless you know /exactly/ why you're doing so
10:54.28codeythe gsm card requires this
10:54.30dijungalto upgrade asterisk after downloading the cvs version... do i do a "make upgrade" ?
10:54.44Strom_Mdijungal: CVS?  is it 2005 again?
10:54.55dijungalTrunnk
10:55.01dijungaltrunk
10:55.11Strom_Mplease tell me you're not using the SVN Trunk branch in production
10:55.28dijungalok i wont tell u :)
10:55.41*** join/#asterisk tenzind (n=tenzind@202.144.144.11)
10:55.52Strom_Mdon't be a moron - run the 1.4 branch
10:55.54dijungalnah it's mostly for testing.. proof of concept ..i'm trying to interconnect a cisco router to asterisj
10:55.55dijungalasterisk
10:56.22dijungalcisco (H.323) -> asterisk
10:56.43Strom_Myeah...so like I said, run the 1.4 branch
10:56.44dijungalonce i have it running and got the right configs etc.. i can install the 1.4
10:57.17dijungali'm currently running 1.2, and i'm just compiling the /channels/h323
10:57.57dijungalso i'm wondering if i should do a straight "make && make install" after or "make upgrade"
10:58.22Strom_Min two and a half years of running asterisk, i've never run "make upgrade" if that's any indication
10:58.31dijungalk
10:58.36dijungalso make install it is
10:58.43Strom_Minstaboners!
10:58.55dijungalthat should replace my 1.2
10:59.10Strom_Myou're asking for trouble, man
10:59.25Strom_Mbut, hey, if you don't want to listen to me, that's your fault :)
11:00.54dijungali'm listening
11:01.11dijungalthe compile's been going for some time now... i think i have time.. i'm listening
11:01.18Strom_Mdon't
11:01.19Strom_Mrun
11:01.19DrAk0dijungal, yes, make ; make install
11:01.21Strom_Mtrunk
11:01.32Strom_Munless you're actually developing for asterisk
11:01.49dijungalwhat's wrong with doing the trunk? (yes i was a "WHY" child)
11:01.56dijungalhmmmm
11:02.01Strom_Mdijungal: it's broken quite often
11:02.04dijungalk
11:02.16dijungalok so do the 1.4?
11:02.20Strom_Myes
11:02.36dijungali have the 1.4 installed on another server.. it seems be working fine :)
11:03.12dijungalwill do... let me go grab the download :)
11:06.18dijungalthis means i have to sit and watch the h323 channel drivers compile for another 2 hrs
11:06.59extr3mare * and # usable in dialplans?
11:07.23extr3mi.e if i wanna have like a short extension: *30 to go to some ext?
11:07.32extr3mor is that considered a wildcard?
11:08.24EmleyMoor* is perfectly usable, # is usable with caution
11:08.51EmleyMoor(naturally, they are no good with rotary phones)
11:09.07Strom_Mextr3m: where are you located?
11:10.08Strom_M* is usually used to indicate special service codes, and # is usually used to indicate the completion of dialing...so, long story short, don't use them unless you know how to work around the specific numbering plan requirements in your locality
11:12.26*** join/#asterisk Ethon (i=arne@Oldman.steinkamm.com)
11:14.27*** join/#asterisk zotz (n=zotz@24.244.163.157)
11:18.07*** join/#asterisk lwh (n=lwh192@66.212.165.24.tor.pathcom.com)
11:18.15*** join/#asterisk pj_ (n=pj@happycoders.org)
11:18.41pj_Hello ppl
11:19.46pj_I'm setuping a * backup server and copied my working conf on it, but I can't seem to call 2 sip phones connected to it, or get any sound when accessing the voicemail. Any idea ?
11:20.13pj_RTP packets don't seem to come out of asterisk server at all (from tcpdumping)
11:22.25*** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au)
11:27.26*** join/#asterisk jubei (n=jubei@147.27.47.165)
11:30.23jubeiguys
11:30.43*** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se)
11:30.51Dovidhello jubei:
11:30.59jeremy_gthe peer tag in sipp is not working
11:31.22jeremy_gi am using it for * testing, any one else using sipp v2.0 ?
11:31.22*** join/#asterisk spaceinvader (n=server@unaffiliated/spaceinvader)
11:31.23jubeiDoes anybody know of an alternative to asterisk but one with a db back end and decent management UI? even a non free one?:)
11:31.33jeremy_gjubei:yes many
11:31.38Dovidgoole it
11:31.41Dovidthere are lots out there
11:31.44jeremy_gjubei:depends how non free u wana bee
11:32.11spaceinvaderHi, I have downloaded the alaw voices from http://www.enicomms.com/cutglassivr/ and put them into the correct place, but * is still looking for gsm and ulaw files. Is there a configuration option or something?
11:32.18jubeii was wondering if there was one which is considered like.. industry standard
11:32.33jeremy_gis it possible that if i write a gui software for asterisk and include asterisk source with it as a seperate package and sell the whole thing to someone. would that be legal?
11:32.37Strom_Mspaceinvader: what file extension do the files have?
11:32.53jeremy_gto distribute asterisk with ur own software as one bundle. (not linked)
11:33.04Strom_Mjeremy_g: read the license :)
11:33.08spaceinvaderStrom_M: .g711a
11:33.17jeremy_gStrom_M:it says yes
11:33.23*** join/#asterisk psk (n=psk@golia.caltanet.it)
11:33.24Strom_Mspaceinvader: change that to .alaw
11:33.27spaceinvaderok
11:33.31Strom_Mjeremy_g: well there you go tghen
11:33.33Strom_Mer, then
11:34.20spaceinvaderStrom_M: Is there a quick way of renaming files en masse? :P
11:34.36jubeijeremy_g: I think it would be legal, yes
11:34.39Strom_Myou can do it easily enough in bash
11:34.49mostyspaceinvader, google "mmv"
11:35.10spaceinvaderok
11:35.40dijungalhello when i do "make opt" from channels/h323 i get "make: *** No rule to make target `opt'.  Stop."
11:35.50dijungaland it does nothing
11:35.53Strom_Mfor i in ./*.g711a; do mv $i ${i%%g711a}alaw; done
11:35.55Strom_Mor something
11:35.59Strom_Mi dont remember the exact syntax
11:36.26dijungalbut the instructions say "cd /path/to/asterisk/channels/h323
11:36.27dijungal<PROTECTED>
11:36.29dijungal<PROTECTED>
11:36.30dijungal<PROTECTED>
11:36.37dijungalwhat i'm i missing?
11:36.55Strom_Mare the instructions for that specific version of h323?
11:36.59tzafrir_laptopfor which version of asterisk?
11:37.13tzafrir_laptopin asterisk 1.4, it is even more stupid:
11:37.24tzafrir_laptopmake || true; make
11:37.26*** join/#asterisk bintut (n=bintut@203.125.63.150)
11:37.52tzafrir_laptopthat is: run make and expect it to fail, but hope it will fail because of h323, and then run make again
11:37.54dijungalasterisk 1.4  h323 1.18
11:38.02dijungal1.18.0
11:38.21dijungalhuh?!?
11:39.03dijungali've compiled the PWlib, and openh323 ... but now asterisk is not compiling the channel driver
11:39.09dijungalwell it seems like i'm missing something
11:39.20tzafrir_laptop./configure
11:39.28tzafrir_laptopagain, maybe?
11:40.09jubeidijungal: i've had similar problems in the past, what distro are u running?
11:40.36dijungalcentos
11:40.56dijungalthe full instructions are here: http://svn.digium.com/view/asterisk/branches/1.4/channels/h323/README?view=markup
11:41.01Jubeidijungal: you have to be absolutely certain that you have the correct sources for your running kernel
11:41.10Doviddijungal: i personally use the h323 drivers in the asterisk add ons
11:41.30*** join/#asterisk okinsey (n=bal@58.85-200-224.bkkb.no)
11:41.50sergeedijungal: cisco runs sip perfectly, i'm using sip to interconnect with Cisco AS5350 and 5400
11:41.52dijungalDovid: and does that work, have u actually got H.323 enpoints connected and working?
11:42.16DovidI have had lots of issues with asterisk h323 + Cisco
11:42.21dijungalsergee: unfortunatly in my scenario i'm stuck with a H.323 supported Cisco 3661
11:42.26Dovidsome box's work well and oters make issues
11:42.29*** join/#asterisk DragoraN (n=dragoran@217.67.19.74)
11:42.30DragoraNhi
11:42.38Strom_Mcocks
11:42.39Dovidif u have a choice go with SIP
11:42.49Strom_Mthe hi/cocks protocol (rfc 4373)
11:43.01Dovidi am looking at getting some patton boxs for SIP to H323 conversio
11:43.05Dovidconversion*
11:43.36dijungalhmmm...
11:43.57dijungalokstill does not answer why this thing not compiling
11:44.11dijungalnow i know if i go back to the asterisk root and do a "make" it will compile asterisk
11:44.18Doviddijungal: I had many issues with the nufone h323 driver compilation
11:44.26dijungalbut i'm afraid it will omit the h323 drivers
11:44.29Dovidtry the dirver in asterisk add ons
11:44.45*** join/#asterisk DragoraN (n=dragoran@217.67.19.74)
11:44.57DragoraNplease say again if someone said something..
11:45.01sergeedijungal: did you configure asterisk?
11:45.10Strom_MDragoraN: all we saw you say was "hi"
11:45.17DragoraNso
11:45.21DragoraNclients are connected to their own sip server on their local network, there servers are connected with SIP via port 5060, singaling works fine, but RTP packets destination IP address is as the other user is on same network... how to solve this?
11:45.25dijungalsergee: not as yet.... just the openh323 and pwlib
11:46.40sergeedijungal: cd /usr/src/asterisk-trunk
11:46.45sergeedijungal: ./configure --with-pwlib=/usr/src/pwlib_v1_10_0 --with-h323=/usr/src/openh323_v1_18_0
11:46.58sergeedijungal: replace paths with yours
11:47.58dijungalk
11:48.32sergeehmm, so many people has an issues with compiling asterisk + h323 :) there should be a kind of service: "express asterisk + h323 installation only $74.95" :))
11:49.36sergeedijungal: when configure script finished, tell me, i will tell you what to do next
11:49.42okinseyok, I have a problem that suddenly appeared. For some reason when one of my xten softpone answers the phone it takes about 5 sec before asterisk connects the call. When trying with idefisk, the connection is instantaneous. Any ideas?
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11:50.04santibioticohi
11:50.20dijungalsergee: it's done
11:50.32dijungaland i'm doing the 1.4 instead...not the trunk :)
11:50.32DragoraNsomeone can help me?
11:50.44sergeedijungal: any errors during script's running?
11:51.14sergeedijungal: why?
11:51.15dijungalnope
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11:51.27sergeedijungal: do you have any problems with trunk?
11:51.28Strom_MDragoraN: I could probably help you if your question made any sense :)
11:51.38dijungalTrunk seems a little too cutting edge :)
11:51.52mostyokinsey, which softphone is it that's slow?
11:51.52sergeedijungal: ok, anyway
11:52.08okinseymosty: xten.
11:52.16okinseyor x-lite
11:52.18dijungalsergee: checking OpenH323 build option... opt
11:52.19dijungalchecking OpenH323 installation validity... yes
11:52.22mostyokinsey, you do realise that x-lite is lousy, right?
11:52.32sergeedijungal: run "make menuselect" from asterisk root directory, and make shure that chan_h323 is selected (not empty and not XXX)
11:52.32dijungalthat was part of the output.. so i guess that's a good thing
11:52.46dijungalk
11:53.15okinseymosty: it has worked so far. What is the preferred one, idefisk? I havent found to many free AND good clients
11:53.31*** join/#asterisk DragoraN (n=dragoran@217.67.19.74)
11:53.36DragoraNsomeone said something?
11:53.41DragoraNmy internet connection is bad :(
11:53.50mostyokinsey, i have no preferred softphone, they all suck in my opinion. especially the linux softphones
11:54.22okinseyI know.. but they are practical..
11:55.21sergeedijungal: any progress?
11:57.34dijungalsergee: what do i do when i'm done selecting the options i want... and yes there's a * by chan_h323
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11:58.14sergeetype "x"
11:58.32sergeedijungal: don't change anything except chan_h323
11:58.38DragoraNplease can someone help?
11:58.46sergeedijungal: make sure that it is selected
11:58.58dijungalk
11:59.16dijungalsergee: it is
11:59.58sergeedijungal: when done, hit "ESC" to exit without saving anything (if you didn't change ) or hit "x" to save changes..
12:00.13sergeedijungal: then, run "make" and wait
12:00.15dijungali used x
12:00.19dijungalk
12:00.42dijungalok it's going
12:00.58sergeedijungal: tell me when it's done
12:01.03cpmDragoraN, You are just going to have to state your issue, and stop asking to ask
12:01.21dijungalit's doing the chan_h323.o now
12:01.22DragoraN<PROTECTED>
12:01.33DragoraNi asked 4 time
12:01.35DragoraNs
12:01.55dijungalDragoraN: lol
12:02.20dijungalsergee: ***************************************************************
12:02.22dijungal********** Re-run 'make' to pick up H.323 parameters **********
12:02.23dijungal***************************************************************
12:02.25dijungalmake[1]: *** [h323/libchanh323.a] Error 1
12:02.26dijungalmake: *** [channels] Error 2
12:02.31mostydijungal, don't paste in here
12:02.38dijungalohooo
12:02.38sergeedijungal: run make again,
12:02.56sergeedijungal: and please use something like http://pastebin.ca
12:03.07dijungalirie
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12:05.25*** join/#asterisk basty (n=basty@212.218.65.223)
12:05.26bastyHi
12:05.50dijungalsergee: still goin.......
12:06.01bastyWhat does BLINDTRANSFER:0:6 mean ? -> "exten => _XX,103,Set(tx=${BLINDTRANSFER:0:6})"
12:08.16*** join/#asterisk matsk (n=mk@194.68.102.174)
12:10.30*** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk)
12:10.32zeeeshhi all
12:10.35sulanis it possible to listen for DTMF in an ongoing bridged call? (Asterisk 1.4)
12:11.12dijungalsergee: it's done
12:11.42dijungalit's says Asterisk has been successfully built and can be installed by "make install"
12:11.51*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
12:12.02sergeeyes, run make install
12:12.13dijungalyaaaay!
12:13.22*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)
12:14.05mostybasty, it's a variable
12:14.40mostyor more precisely, it's a substring of a variable
12:14.48bastymosty: I know - but what does 0:6 mean? Cutting the number from 0 to 6 ?
12:15.08zeeeshmy asterisk server users dial access number through mobile . simplest in extensions.conf i did ... exten=> _x.,1,Answer ...exten=> _X.,2,WaitExten(15)
12:15.24mostybasty, see the asterisk variables page on voip-info.org
12:15.50bastyheheh
12:15.51bastywell thanks
12:18.20zeeeshmy asterisk server users dial access number through mobile . simplest in extensions.conf i did ... exten=> _x.,1,Answer ...exten=> _X.,2,WaitExten(15)... exten=_X.,3,Dial(SIP/{EXTEN}@carrier) ... where can i make enter an extension for call conferencing .. like i tried the way exten=> 5557,4,MeetMe(54321) but failed .. without using zap etc .. ?
12:19.36dijungalsergee: asterisk: error while loading shared libraries: libh323_linux_x86_r.so.1.18.0:
12:19.48dijungalsergee: that's after the compile
12:20.04sergeedijungal: check channels/h323/README
12:20.19dijungalyea i think the library is just in the wrong place
12:20.28[TK]D-Fenderzeeesh: Your _X. will match 5557 as well.  that pattern match is dangerous and highly inadvised.  IVR's should be run off of "s".  Also always include CLI output of the failed call at verbose 10, and relevent configas
12:21.19*** join/#asterisk guillote_GNU (n=guillote@host155.200-117-35.telecom.net.ar)
12:21.32sergeedijungal: yes, update your ENV vars to point to them
12:26.02zeeesh<[TK]D-Fender>:using _X., i hv about 6 different countries DID. waitexten  bcoz getting dtmf of their destination number ..  now will "s" can help for this ?
12:26.45zeeesh<[TK]D-Fender>: not using zap ..
12:27.14[TK]D-Fenderzeeesh: FORGET ZAP!  You don't understand the basics of *'s Standard Extensions.
12:27.38[TK]D-FenderAnd _X. will match whatever they dial ALSO and jsut send them in a loop!
12:28.02dijungalsergee: ahh finally a good isntall :)
12:28.05[TK]D-Fenderzeeesh: You can't escape a menu when all roads lead back to the menu!.  Change it now.
12:28.13sergeedijungal: enjoy
12:28.20zeeesh<[TK]D-Fender>: ok sir
12:28.32dijungalsergee:so do i make sure the h323 channel drivers is installed correctly
12:28.42dijungalis there a CLI command for it?
12:29.08sergeemodules show like 323
12:30.13santibioticoi want asterisk to use the first zap channel available when i make a call. i use dial(zap/g1... and if channel 1 from group1 is busy, it uses channel2...until this point it's ok...but if for example, channel1 is down (no link, etc.) it doesn't use channel2...it says there is a failure in channel1...is there any way to use first available channel?
12:31.59*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)
12:31.59*** mode/#asterisk [+o anthm] by ChanServ
12:32.53JTsantibiotico: your channels aren't meant to be down
12:34.10[TK]D-Fendersantibiotico: Go make some dialplan to dial out each consecutively checking the DIALSTATUS between each to se if it should try another channel.
12:36.35JTa simple way is to dial sequentially ascending, then if that fails dial sequentally descending
12:40.34pj_I'm setuping a * backup server and copied my working conf on it, but I can't seem to call 2 sip phones connected to it, or get any sound when accessing the voicemail. Any idea ?
12:41.08pj_(voice packets don't get out of asterisk, from tcpdumping)
12:42.00*** join/#asterisk jetlagmk2 (n=jetlag@pool-70-106-120-83.pskn.east.verizon.net)
12:42.05[TK]D-Fenderpj_: and where is it located relative to the phones, and relative to the other serv?
12:42.12pj_on the same vlan
12:42.17pj_so no nat magic :/
12:42.35pj_And the phone voice packets come alright to the * server
12:42.41pj_(gs bt200 btw)
12:42.48*** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk)
12:43.01pj_the sip session looks alright, and no error whatsoever :/
12:43.08[TK]D-Fenderpj_: pastebin all the backup that shows networking isn't in the way, along with CLI output w/ SIP debug enabled.
12:43.12[TK]D-Fender~pb
12:43.14jboti guess pb is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org
12:43.15[TK]D-Fender^^^^^
12:44.07JTquality phone i see
12:44.40pj_Thanx
12:47.49*** join/#asterisk Cyber-Dogg (i=Cyber-Do@24-178-240-97.dhcp.stls.mo.charter.com)
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12:56.42pj_[TK]D-Fender: http://pastebin.ca/578905 if you can take a look...
12:57.20pj_the asterisk is in a vmware
12:58.06*** join/#asterisk friedrich| (n=friedric@e177242102.adsl.alicedsl.de)
12:59.00[TK]D-Fenderpj_: Seems to look fine, Where's the failure in there?
12:59.29*** join/#asterisk D-side (n=brian@pool-71-251-32-32.nwrknj.east.verizon.net)
12:59.29pj_No sound at all in the phone
12:59.43pj_And see the tcpdump ? Nothing comes from the * server (1.0.1.3) to the phone
12:59.56JT1.0.1.3? interesting ip
12:59.59pj_though I should see all the "Enter your password" data packets
13:00.01[TK]D-Fenderpj_: pastebin your sip.conf
13:01.51D-sidehi there boys. I've got a silly question concerning ATAs, routers with a built-in ATA, and QOS. This isn't specifically asterisk related, so does anyone know of a more general purpose voip channel or web forum? really i'm trying to figure out if an ATA behind a router w/ built-in ATA will take advantage of the router's QOS.
13:02.28*** join/#asterisk vgster (n=vgster@host81-157-72-207.range81-157.btcentralplus.com)
13:02.39pj_[TK]D-Fender: updated the pastebin
13:02.42*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
13:02.51[TK]D-Fenderpj_: Dpesn't work that way...
13:02.58[TK]D-Fenderpj_: takes a new link
13:03.05pj_http://pastebin.ca/578943
13:03.17pj_(damn it)
13:05.42[TK]D-Fenderpj_: ok, lan looks local, SIP seems fine, configs seem fine... not sure where to go from here, but that may help someone else in assisting you.
13:06.25mockerDoes a softphone in the VMWare instance work?
13:06.47*** join/#asterisk _DAW (n=chatzill@adsl-222-41-108.msy.bellsouth.net)
13:07.21pj_[TK]D-Fender: :(
13:07.46pj_thanx anyway... Glad to have had a "double check"... it seems quite odd for * not to produce any packet at all
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13:27.41cy303yo
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13:38.39mrdigitalcan someone help me select multiple fields in a sql statement
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13:41.28anonymouz666join mysql channel
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13:46.20Nuggetwhat if it's not mysql?  :)
13:46.48*** join/#asterisk oej (n=olle@136.240.13.217.in-addr.dgcsystems.net)
13:47.23[TK]D-FenderNugget: Go back to playing with your .... telnet ;)
13:47.38sergeecy303: g'day :)
13:47.47Nuggetdenied
13:48.09sergeemrdigital: select a,b,c fom mytable;
13:48.10[TK]D-FenderNugget: Only when you're idle I take it?
13:48.57Nuggetonly once a day, across all the channels I'm in
13:49.18Nuggetso someone's said "telnet" in some other channel in the past 24 hours apparently
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13:51.34penguinFunktelnet
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13:53.03rsd99when i make a call in to my asterisk server, i seem to be getting some static.
13:53.28JTmore details on the setup please
13:53.47cy303sergee: ;)
13:54.27cy303Anyone here have any experience with the RAMI rubygem?
13:56.10rsd99just using sip for everything.  4 extensions, 2 for phones, 1 for VM, and 1 for auto attendant.  16mbps down and 1mbps up for internet
13:56.37*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
13:57.44*** part/#asterisk D-side (n=brian@pool-71-251-32-32.nwrknj.east.verizon.net)
13:58.29[TK]D-Fenderrsd99: that says notihng about the EQUIPMENT that can be causing the static.  We want to know what HARDWARE you are using.
13:58.42[TK]D-Fenderrsd99: What TDM cards, what phones exactly.
14:01.06rsd99it's all SIP.  i have a sip proxy from a provider, and two 7960 series phones
14:01.16jeremy_gwhats the regex from extracting the tag out of To: <sip:sipp@192.168.0.2>;tag=26068SIPpTag011
14:01.23jeremy_ganyone?
14:03.00[TK]D-Fenderrsd99: Do you get static directly between your phones or between a phone & *'s voicemail for instance?
14:03.30rsd99i am calling in from the outside from my cell phone.  i didn't check it internally this morning.
14:04.32rsd99now it seems to be gone
14:04.38*** join/#asterisk waptaxi (n=waptaxi@45.151-224-87.telenet.ru)
14:04.40*** join/#asterisk Infested (n=infested@24.148.112.10)
14:04.57rsd99and also, i have asterisk running on a Dual 867mhz g4 Mac with OS X 10.3.9
14:05.03*** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198)
14:05.40Dr-Linuxdue to power fluctuation my two FXO 4 port cards burned out
14:05.54cy303Dr-Linux: that blows
14:05.55Dr-LinuxStrom_M:
14:06.18Dr-Linuxcy303: is there any varrenty from digium? :S
14:06.26Dr-Linuxi baught them 4 months ago
14:06.36Qwell[]Dr-Linux: call and talk to them...
14:06.39mrdigitalyup
14:06.41cy303Dr-Linux: Not sure.. I'd be pissed heh
14:06.45Qwell[]only they can tell you for sure what's covered
14:07.52*** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com)
14:07.56Dr-Linuxdamn, i'm already using a couple of more FXO's in US servers, also 4 T1's card
14:08.06Dr-Linuxbut thoso cards were in Pakistan server here
14:08.09Nuggethttp://macnugget.org/projects/asterisk/page15
14:08.27Dr-Linuxnot sure how can i bug again, difficult to ship from US again :S
14:09.40Dr-LinuxQwell[]: we hve' about 200 servers in the same data center here with different hardwares, my bad is only these cards got burned :S
14:09.54Qwell[]You need to get a nice UPS on them...
14:10.17Dr-LinuxQwell[]: we are UPS company and that's APC
14:10.29rsd99are those digium cards mac compatible
14:10.42Dr-Linuxdon't know
14:10.43*** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca)
14:10.43Dr-Linuxnever tried
14:10.48Qwell[]rsd99: It's just PCI.  If you're running Linux, it should work - in theory.
14:10.59Qwell[]don't quote me on that though :)
14:11.00*** join/#asterisk Cherebrum (n=jgarland@pdpc/supporter/base/Cherebrum)
14:11.03CherebrumJun 20 10:09:38 WARNING[5109]: channel.c:807 channel_find_locked: Avoided initial deadlock for '0x81da9d8', 10 retries!
14:11.03CherebrumJun 20 10:09:39 ERROR[5213]: chan_sip.c:11602 sipsock_read: We could NOT get the channel lock for SIP/84071-0815ecc8!
14:11.06CherebrumJun 20 10:09:39 ERROR[5213]: chan_sip.c:11603 sipsock_read: SIP MESSAGE JUST IGNORED: ACK
14:11.08CherebrumJun 20 10:09:39 ERROR[5213]: chan_sip.c:11604 sipsock_read: BAD! BAD! BAD!
14:11.22anonymouz666bad bad bad
14:11.31anonymouz666ast msgs are funny
14:11.32CherebrumIt seems Asterisk developers don't know how to write multithreaded applications.
14:11.46Nuggetheh
14:11.48Qwell[]Cherebrum: yes, that's clearly the problem
14:12.05anonymouz666Ooh format changed
14:12.11rob0But the error messages are entertaining!
14:12.41*** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com)
14:12.47CherebrumI'd hate to see what happens if I had more than 1 call at a time.
14:13.34*** join/#asterisk murr4y (i=murray@valhall.samfunnet.no)
14:13.59tzangerQwell[]: heh
14:14.20murr4yoooh damn
14:14.28murr4yyea this *has* to be the official channel
14:14.31murr4yhi everyone :)
14:14.39*** part/#asterisk Cherebrum (n=jgarland@pdpc/supporter/base/Cherebrum)
14:14.43rsd99can someone PM me a simple dialout dialplam.  i racked my brain last night, and for somereason my brain wasn't working and couldn't figure it out for some reason.
14:15.06rsd99basically when i pick up the phone, if i want to make an outbound call to an actual phone number, i have to dial a 1
14:16.24rob0Asterisk ... it's not your father's Shift+8
14:17.14Dr-Linuxanybody know what time Digium support comes in to their office? :S
14:17.41mrdigitalyes
14:18.03mrdigitalMon. - Fri., 7 am - 7 pm CST (GMT -6)
14:18.25mrdigital256.428.6000
14:18.44Qwell[]it's currently after 9am here
14:18.47sulanrsd99: in the context your phone enters when dialing a number, add an extension _1X. with priority 1 that calls the Dial application
14:20.25*** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net)
14:20.35sulanrsd99: let appdata for the Dial application include the interface you want your outgoing call to take, for example SIP/peername/${EXTEN:1}
14:20.48rob0Some of them probably are not up yet. :)
14:21.09*** join/#asterisk mattchis (n=mchisenh@216.54.143.246)
14:21.15sulanrsd99: the outgoing call will now be called using 'peername' and the number will be the extension with the first digit removed
14:22.12mattchisdoes anyone know what "Internal RTCP NTP clock skew detected" in asterisk means?
14:22.48*** part/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net)
14:25.08*** join/#asterisk anthm_mobile (n=anthm@CPE-72-131-113-50.wi.res.rr.com)
14:28.19Mercestesmattchis:  It means there is a time issue, most likely.  Reset your ntp and give it a go again.
14:28.25Mercestesmattchis:  may want to pull a nice restart too.
14:28.41Mercesteson asterisk, because it sounds like the RTP timestamps are off from your server timestamps..probably inbound tho
14:28.44Mercestesso nevermind about the restart
14:29.27mattchisso all I need to do is resync the aginst my time server
14:29.29mattchis?
14:30.03Mercestesmattchis:  correct.
14:30.07*** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse)
14:30.15Mercesteswhich as I recall is pool.us.ntp.org
14:30.45*** join/#asterisk af_ (n=getsmart@81-174-45-5.dynamic.ngi.it)
14:30.56*** join/#asterisk holiday_42 (n=no@spike.wcta.net)
14:31.09mattchiscool thanks always for you help
14:31.20MercestesNp.  :)
14:31.24errrIf I have multiple interfaces in my pbx is it possible for asterisk to bind its self to only 1 of those interfaces, or will it listen on all of them?
14:32.13*** join/#asterisk E-bola (n=bola@cpe-76-179-4-233.maine.res.rr.com)
14:32.39E-bolaDo anybody use snom phones with asterisk? Im wondering if the snom addon module can show the status of other sip phones
14:32.59errryes, it supports blf
14:33.05_DAWerrr: that depends on the protocol.  It is specified in iax.conf sip.conf etc....
14:33.07errrwe have a few 320's
14:33.11mattchisMercestes: It appears that my time is in sync with the time server.  Any ideas.
14:33.25E-bolaIm referring to the snom 360 Expansion Module
14:33.25errr_DAW: ahhh, ok, thanks Ill look into that
14:33.29E-bolathat u attach ont he side
14:33.35E-bolatypically for secrearies etc.
14:33.46*** join/#asterisk blackbyte01 (n=blackbyt@89.119.146.121)
14:33.57errrE-bola: we dont have any, but the phones do support blf, so I dont see why the module wouldnt
14:33.58E-bolaerrr: was that meant for me?
14:33.58mrdigitaldoes anyone know sql coding in asterisk?
14:34.06E-bolaok, whats blf?
14:34.12blackbyte01Sorry, I need some help...
14:34.20errrE-bola: blf is what youre looking for, I dont recall what it stands for
14:34.41Qwell[]~blf
14:34.41jbotwell, blf is Busy Lamp Field, aka little lights next to speed dials that light up when the person is on the phone and blink when that line is ringing.  hint extensions are static mapped to SIP or other channels.
14:34.46Mercestesmattchis:  type date.
14:34.55E-bolaahh i gotta read up on that
14:35.12*** join/#asterisk jwh (i=jwh@knickertron.com)
14:35.16E-bolathanks
14:36.05E-bolado anybody know where i can find out how to "map" a line to a button on either a snom or linksys phone?
14:36.20*** join/#asterisk TJ` (i=ch220207@bnc.crazybnc.com)
14:36.25jwhE-bola: i've been trying to get the snoms to do that for months
14:36.31jwhE-bola: at least on the 190 it's not possible
14:36.40TJ`what a smallworld it is
14:36.45walhalaE-bola: with 320 it's possible
14:36.53jwhTJ`: dirty freepbx user
14:36.53jwh:p
14:36.54E-bolai got a 320
14:36.58TJ`lol
14:36.58TJ`no
14:37.00TJ`:P
14:37.04TJ`trix
14:37.15jwhsame thing, negates the need to know what you're doing :P
14:37.19[TK]D-FenderTJ`: Trixbox = FreePBX
14:37.22walhalaE-bola: so which version a of * do you use ?
14:37.25*** join/#asterisk af_ (n=getsmart@81-174-45-5.dynamic.ngi.it)
14:37.32TJ`[TK]D-Fender did u know grass was green too?
14:37.48E-bolawalhala: 1.4
14:37.55*** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga)
14:38.06[TK]D-FenderTJ`: Indeed.  and do you know why the sky is blue?  Because if it were green we wouldn't know when to stop mowing!
14:38.16walhalaE-bola: i'm running 1.2 and work but you have to patch your chan_sip.c
14:38.28TJ`the sky is actually black
14:38.31pj_Blue is green in japan.
14:38.32blackbyte01I am developing a php script to manage my voip system based on asterisk... I need commands (or functions) to log every call done by an account...
14:38.47blackbyte01can someone halp me?
14:38.51blackbyte01*e
14:38.56Qwell[]~cdr
14:38.56jbotwell, cdr is Call Detail Record, a log of what happens to the call at each step through its traversal of the PBX, details like from, to, time, duration, number dialled etc, useful for billing also - it could also be Compact Disc Recordable, see cdrw
14:39.07walhalablackbyte01: use the cdr in sql
14:39.28CelticSoulHi, If I use PHP AGI to dial an extension (for example 33333), what will happen when the user at 33333 pick up the phone?
14:39.38TJ`....
14:39.41TJ`he'll say hello
14:39.43blackbyte01what's that?
14:39.56walhalablackbyte01: cdr is Call Detail Record
14:40.12walhalayou can store it into a database such as MySQL
14:40.26CelticSoulI meant What happen in PHPAGI and Asterisk?
14:40.27pj_[TK]D-Fender: can the fact that I have no sound card explains why I get no sound in voicemail ?
14:40.30blackbyte01thanks!
14:40.33blackbyte01i will!
14:40.47walhalablackbyte01: take a look on voip-info at realtime
14:40.47pj_(I remember it had some impact "back in the days")
14:40.49[TK]D-Fenderpj_: No.
14:40.55pj_:(
14:41.02pj_I wish you'd say yes :/
14:41.12pj_At least I'd know
14:41.13Qwell[][TK]D-Fender: it isn't too late
14:41.24jkiffYou wish to be lied to?
14:41.27jkiffInteresting.
14:41.49Mercestespj_:  Could it be one way audio?
14:42.02pj_It could
14:42.15pj_My box doesn't want to tell me if it hears me or not though
14:42.18pj_No nat
14:42.24pj_but no packet leaving asterisk box either
14:42.27walhalasome sccp or skinny users here ?
14:42.48pj_The server is in a vmware box... I have replicated the same exact conf on a physical box and it works fine
14:43.09Mercestespj_:  Maybe it is vmware then.
14:43.26*** join/#asterisk xacatecas (n=jkroon@c1-218-13.tbnb.isadsl.co.za)
14:43.31pj_doesn't explain much...
14:44.27walhalaobody use a cisco 79XX with asterisk ?
14:45.37xacatecashi folk, i'm busy trying to get asterisk up and running on a gentoo installation.  I've now managed to finally compile version 1.2.17 (there was an issue with the h323 module it seems) and have it installed.  however, i'm seeing a version of 1.4.x as well, I've been trying to figure out what the difference between 1.2.x and 1.4.x is, so far with no succes - anybody mind explaining?
14:45.44holiday_42pj_: if conf is same, perhaps something else.. errant firewall rules maybe?
14:46.22pj_no firewall
14:46.25pj_:/
14:46.29CelticSoulsomeone please
14:46.58pj_and packets do not appear on a local tcpdump on the * server
14:46.59holiday_42pj_: is the file size of the vm zero?
14:47.17pj_you mean space left ?
14:47.30pj_there's two whole gigs free
14:47.59holiday_42nope... i mean if you look at... uh /var/spool/asterisk/voicemail (or whatever) and look at the voice mail files
14:48.12*** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net)
14:49.21*** join/#asterisk inv_arp[work] (n=junya@c-71-229-122-61.hsd1.fl.comcast.net)
14:52.12*** join/#asterisk ^^ARcANgEL^^ (n=arcangel@189.129.88.8)
14:52.32^^ARcANgEL^^hello
14:53.01Mercesteshello
14:53.12pj_Ohh
14:53.19jwh:o
14:53.21pj_holiday_42: thing is, it doesn't even play "password"
14:53.34pj_well it does play, except there's no sound coming out
14:53.46*** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net)
14:53.50Qwell[]yeah, that's one-way audio for sure
14:53.55^^ARcANgEL^^somebody podria to help me, is that i am obstructed installs asterisk but not to form it
14:54.11^^ARcANgEL^^???
14:54.14Qwell[]what?
14:54.30_DAWwow
14:54.34MercestesDid you ./configure?
14:54.36Qwell[]babelfish FTW
14:54.53Qwell[]Mercestes: how the heck did that make any sense to you?
14:55.01pj_Qwell[]: and any reason why it should not even send the RTP packets ?
14:55.05^^ARcANgEL^^that if somebody podria to help with the installation of the servant asterisk me
14:55.12MercestesQwell[]:  google translate, gibberish to english.
14:55.23pj_I mean, I can understand one way if it sent packets to the wrong adress, wrong port, or had a NAT or firewall
14:55.31Qwell[]Mercestes: god bless google
14:55.39Mercestesgod bless it!
14:55.47pj_But packets are not even trying to leave the server... LAZY PACKETS !! /me starts whiping the packets
14:56.07Mercestes^^ARcANgEL^^, what wiki, howto, or instructions are you looking at?
14:56.26CelticSoulHi, If I use PHP AGI to dial an extension (for example 33333), what will happen to PHPAGI and Asterisk when the user at 33333 pick up the phone?
14:57.00[TK]D-Fender^^ARcANgEL^^: Pick another language because nobody will understand your english.
14:57.03MercestesCelticSoul, Hang there until dial completes.
14:57.25Mercestes[TK]D-Fender, LOL.
14:57.27Qwell[]^^ARcANgEL^^: what is your native language?
14:57.44Mercestes^^ARcANgEL^^, what is your prime directive?
14:58.06[TK]D-FenderMercestes: "Take us to your leader!"
14:58.30pj_All the rtp paketz are belong to us
14:58.44holiday_42pj_:how do you know that no packets leave?  seems like that and "well it does play, except there's no sound coming out" seem contradictory?
14:58.45^^ARcANgEL^^spanish but a little english
14:58.55Qwell[]^^ARcANgEL^^: very little english.  Please speak spanish.
14:59.05Mercestesque es podria?
14:59.28^^ARcANgEL^^es que como vi que hablaban ingles por eso
15:00.02pj_holiday_42: it does "-- Playing password"
15:00.19pj_And no packet leave because I don't see any on any interface with tcpdump
15:00.24Qwell[]pj_: turn on rtp debug, and see where it's trying to send the packets
15:00.31pj_it's not trying
15:00.35pj_(tried rtp debug)
15:00.49mihinomenestI should be able to do something like, "dtmfmode=inband,rfc2833" in sip.conf, right?
15:01.04Qwell[]mihinomenest: I don't believe so
15:01.11pj_it only gets the packets, never try to give them back
15:01.13pj_evil *
15:01.13Mercestespj_:  =/   so there isn't even an RTP stream attempted?
15:01.18mihinomenestI say then, "grumble."
15:01.21pj_doesn't seem so
15:01.29Qwell[]mihinomenest: is there a reason you need to be sending both?
15:01.30Mercestesmihinomenest, dtmfmode=auto
15:01.55pj_if you mind to take a look : http://pastebin.ca/578979
15:02.22*** part/#asterisk ^^ARcANgEL^^ (n=arcangel@189.129.88.8)
15:02.49pj_oh, wait, I got one Sent RTP packet
15:02.53*** join/#asterisk dcm_ (n=dcm@207.59.3.77)
15:03.42*** join/#asterisk rgsteele (n=rgsteele@nat-pool.agora-net.com)
15:03.58mihinomenestQwell[]: well, no.
15:04.09Mercesteshow come your voicemail log is like a dozen lines, and my voicemail log is like....2 lines?
15:05.18rgsteeleHey folks.  I've got several analog lines from the telco going into a TDM400P in my asterisk box.  The problem I've got is horrible echo on the internal side - clients calling in hear no echo.  But, internally if our mouths get too close to the phones, or we speak too loudly, we get a really bad echo on our end.  I've tried futzing with echocancel, echotraining, rx and tx gain, and still...
15:05.20rgsteele...can't resolve it.  Any suggestions?
15:05.52mihinomenestthe problem is, the voip provider that we resell to our customers demands that we use linksys PAP2s set to automatically download their configs via TFTP.
15:05.56rgsteeleAsterisk 1.2.13 and zaptel  1.2.11
15:05.59Mercestesrgsteele, handset or speakerphone?
15:06.06rgsteeleMercestes: Handset.
15:06.12Mercesteshrm.
15:06.15Mercesteswhat type of phone?
15:06.26pj_Mercestes: no idea... trixbox 2.2 here
15:06.40rgsteeleSeveral different types.  One's a Sipura SPA-841, the others are a variety of Cisco's.
15:07.03mihinomenestwell, their default config specifies DTMF via SIP INFO, and apparently, if you call from one of their SIP "circuits" to another, the DTMF is never changed from SIP to whatever the other side needs.
15:07.17Mercestespj_:  oh, that makes it easy then.
15:07.19Mercestes~trixbox
15:07.20jbotTrixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org
15:07.25mihinomenestso, my PBX is looking for inband and ignoring my customer's key presses.
15:07.38pj_Well it's not trixbox related at all
15:07.44*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)
15:07.48mihinomenestof course, the provider's Voicemail server is ignoring all DTMF, so it's really their problem.
15:07.49[TK]D-FenderMercestes: I'd have done that, but he is decidedly NOT a chump even though using Trxbox....
15:07.50Mercestesdoesn't matter.  It's a mess to troubleshoot
15:07.57jwhset the correct mode on asterisk and voip phones?
15:08.10Mercestespj_, honestly tho, I think vmware is doing something silly with yoru network interface.
15:08.14jwhlike, i use info
15:08.15Cyber-Doggdoes anyone have any good references on how to setup asterisk with MSSQL?
15:08.18jwhbut my providers use inband etc
15:08.25Cyber-DoggI've seen some on PostgreSQL and MySQL
15:08.25Mercestespj_, I will *bet* it has something to do with natting between your virtual interface and your *real* interface.
15:08.28[TK]D-Fenderpj_: I suspect the same (VMWare)
15:08.30jwhjust setting the right mode in sip.conf sorts out any interactivity problems
15:08.36Cyber-Doggshould ODBC work the same with MSSQL?
15:08.38pj_Mercestes: eating the RTP packets before they come out from the linux server and only one way ?
15:08.40holiday_42pj_: *shrugs* vm works here on vmware, but moh very choppy
15:08.58Mercestespj_:  Or in, really, as I'm not 100% sure that even works
15:09.02pj_I have an ESX server, I want to give it a shot
15:09.10Mercestesgo for it.
15:09.19mihinomenestjwh: if I set "rfc2833" in my sip.conf, no one works, not even incoming calls from PSTN works.
15:09.21pj_And there is no natting
15:09.25mihinomenestI have to have it set to inband.
15:09.42Mercestesrgsteele, Well, thsi is the standard stuff but, give fxotune a try, and check yoru interrupts
15:09.44jwhmihinomenest: inbound you will need to match your provider
15:09.44holiday_42meh, it was discussed on the list, virtualization seems to mess up the timing pretty
15:09.45*** join/#asterisk svenna_ (n=svenna@p548D2E1A.dip0.t-ipconnect.de)
15:09.45mihinomenestand, when I set it to inband, voip customers from the same provider can't navigate my IVR.
15:09.51jwhoutbound can be set differently
15:10.00pj_holiday_42: timing as in clock time ?
15:10.11mihinomenestoutbound doesn't matter.
15:10.22jwhok, well what do you providers use?
15:10.24holiday_42pj_: that too
15:10.29jwhyou need to have the same setting as they do
15:10.30Mercestesmihinomenest, try dtmfmode-auto
15:10.36pj_and you're basically saying that the sound should be choppy
15:10.39pj_it's not, I got none
15:10.44jwhhowever, if you pass IVR onto something else, then that may cause issues
15:10.45pj_I wish I had choppy
15:10.53holiday_42:)
15:10.55pj_then I could say it's because of vmware
15:11.08pj_But the packets not being emitted by asterisk is another problem
15:11.17holiday_42agreed
15:11.18pj_which has nothing to do with trixbox, or vmware
15:11.32mihinomenestjwh: and for inbound, I do.  all of my stations are set to rfc2833 because it seems like that works best.
15:11.45pj_or maybe, but then give me a "reasonable cause" beside "it's black magic, don't go there"
15:12.08jwhmihinomenest: hm
15:12.38holiday_42pj_: are you using asterisknow pre built vmware image?
15:13.06pj_nope
15:13.29pj_I had to install from scratch because the prebuilt image is not tailored for ESX servers
15:13.36mihinomenestjwh: apparently, we've got the 2nd highest guy at the voip provider working on it.
15:13.36holiday_42oic
15:13.49jwhmihinomenest: hehe
15:14.16rgsteeleMercestes: Ah, I found a solution that seems to work - I set the txgain to -4.5
15:14.35rgsteeleMercestes: I noticed only when I raised my voice or spoke really closely to the handset that I got awful feedback.
15:14.48rgsteeleSo, I turned that down, and I could still hear myself well on the cell phone I called in from.
15:14.53holiday_42pj_: can you easily migrate the virtual machine to vmware server or workstation?
15:15.28*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)
15:15.37*** join/#asterisk HockeyInJune (i=HockeyIn@pool-70-18-14-219.nycmny.east.verizon.net)
15:15.51*** join/#asterisk Splat (n=splat@home.heehawhills.com)
15:16.50Mercestesrgsteele:  Hrm, as long as it works.  :)
15:17.04pj_it's "doable" yes
15:17.17rgsteeleMercestes: Yeah.  To hell with all that "in theory" stuff ;)
15:17.28Mercestesrofl
15:17.30pj_Oh wait, I got _one_ RTP packet :/
15:18.02Mercestespj_:  then quit complaining!  :P
15:18.29*** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com)
15:18.36pj_:)
15:19.13*** join/#asterisk falz (n=falz@proxy.supranet.net)
15:20.29*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
15:20.40falzmorning. Is tehre any hacky way to background an agi() call? I'm converting Read() to an agi script that uses swift from bourne. (I can't use Swift() as I can't upgrade to 1.4.x yet)
15:21.00falzso, any way to read text like Background() or WaitExten() does, but while an AGI is running that's playing voice
15:21.23ManxPowerfalz: you cannot background an AGI.
15:21.40*** join/#asterisk gisasi (n=chatzill@ip-240-130.sn2.eutelia.it)
15:21.53ManxPowerYour agi can put it'self into the background, just like any other daemon, but it would usually be bad idea.
15:21.59falzis there any command that will background itsself and wait for input PRIOR to the agi, or would the agi simply always take the foreground?
15:22.10falzyea, that sounds like a lot of orhpaned processes
15:22.17falzand just messier than I'd like
15:22.18ManxPowerfalz: ALL things in the dialplan take place in the forground.
15:22.39ManxPowereven Background() runs in the corefround
15:22.43ManxPowerforeground, that is.
15:22.44falzBackground() is the type of functionality I'm looking for obviously
15:22.57ManxPowerfalz: so why don't you use it?
15:22.58falzbut without playing back a pre-recorded .gsm
15:23.08JerJereeek - If anyone is flying united today, stay home!
15:23.11ManxPowerthen use Read or Waitexten
15:23.22falzthe stuff Im playing back isn't prerecorded.
15:23.30JerJerFox just alerted that United has canceled all flights
15:23.34falzit's sounding like I'll have to prerecord things.
15:23.49falzor upgrade to 1.4 if Swift() has any background type of options
15:23.52ManxPowerfalz: Make up your mind.  Do you want to play audio or not?
15:24.04falzof course I do. but I'd like the audio to be interrupted by user input
15:24.18ManxPowerbest of luck with that.
15:24.33ManxPowerThe way I solved the issue many years ago is still, in my opinion, the best way.
15:24.46falzand that is _______?
15:24.53ManxPowerUnless your text-to-speech audio file rendering takes more than 2 seconds.
15:25.16ManxPowerfalz: generate the audio file, use background, delete the audio file.
15:25.33Qwell[]JerJer: what, why?
15:25.34falzdo you have some delayed backgrounded job that delets it?
15:26.10MercestesFalz:  Just use System() at that point.
15:26.11falzor are you using like agi_uniqueid, passing it to a new variable, and after the Background() calling a new script to rm?
15:26.14JerJerQwell[]:  major 'system' problem - more detail soon... they just did a fox news alert
15:26.21Qwell[]huh
15:26.23JerJeri'm presuming computer problems
15:26.28Qwell[]good thing my wife is flying delta :P
15:26.32[TK]D-FenderQwell[]: terrists!
15:26.44MercestesWell, today is 6-20.
15:26.57Qwell[]Mercestes: that makes perfect sense
15:27.00anonymouz666JerJer!
15:27.00Qwell[]except that it doesn't
15:27.01MercestesGood day to fly a plane into something since it has such a strong, numerical meaning
15:27.14ManxPowerfalz: you can put the file in a temp directory, then have a cron job that deltes the files.  Yo can also have exten => h do the deleteions
15:27.16Mercestesqwell[]:  Exactly!
15:27.30MercestesOr you can delete it right after you background it
15:27.43Mercestesanything involving rm -dvfr should suffice.
15:27.45falzI'll experiment with that fun stuff.
15:27.48Qwell[]I made a script that used festival...  I did some slick md5 hashing stuff on it
15:27.57Qwell[]so if the same text was ever repeated, it wouldn't re-generate it
15:28.02falzknow offhand if the built in Swift() stuff can be backgrounded, or same issues there?
15:28.22ManxPowerfalz: no apps do that
15:28.26falzok.
15:28.27[TK]D-FenderQwell[]: Oh sorry... I should have said Democratic Atheist Lesbian ACLU spokespeople en-route to an Anti-war rally! ;)
15:28.30MercestesManxPower, don't do that..:(  It makes my files hang out and refuse to be deleted.
15:28.50ManxPowerMercestes: exactly, you short sighted pbx admin
15:28.55falzwell, no matter what there will have to be some cron job anyway, I wouldnt trust anything deleting. find /tmp/dir -name "*.wav" -mtime +1
15:28.57Mercestes:(
15:29.54MercestesI'm feeling violated again...am I in the right channel?  This is ASTERISK isn't it?  *looks around*  oh..good.
15:30.01ManxPowerMercestes: Good.  Maybe next time you'll think about the implications of doing something wrong.
15:30.29falzManxPower: when you're creating your .wav files, are you just using agi_uniqueid to name them? That var is new to me as of this morning, but it seems unique enough :)
15:30.39ManxPowerMercestes: Do you also add "r" to all your Dial lines and use exten => _. all over the place too?
15:31.01ManxPowerfalz: It was 4 years ago.  I don't remember.
15:31.09falzoh. so not using it anymore?
15:31.25Qwell[]falz: when I was doing my festival script, I just named them the md5 of the text
15:31.38falzhmm yea, an md5 shouldn't take long
15:31.44Qwell[]it doesn't
15:32.05ManxPowerfalz: You entered in your Zip Code (USA ONLY) and the app would connect to weather.com, screen scrape the required text, sent it to Cepstral (older verison), then played the file, then deleted the file.
15:32.09Qwell[]even with long text, it took less than half a second to search through existing files and if needed, generate the file
15:33.11falzhmm I'll look at some of the old festival stuff online and see if there's some prewritten stuff I can.. "borrow"
15:33.41falzseems like a chicken/egg thing (or I'm dumb) in knowing the md5sum prior to looking for it
15:33.55ManxPowerWhat was the previous name "swift"?
15:33.55Qwell[]falz: You always md5 the text
15:34.00falzahhh
15:34.38falzhence, the dumb disclaimer.
15:34.40ManxPowerif I know that I can prolly fine my ancient script somewhere.
15:34.46falzthat's a good starting point.
15:35.27*** join/#asterisk galeras (n=root@201.244.240.173)
15:35.57MercestesManxPower, only when it's your mom calling. :P
15:37.55[TK]D-FenderManxPower: Retreat!!! He used the "Ur Mom" defense, you have no chance to survive!  Make your time!
15:38.12Mercestesbwahahaha
15:38.47Mercestesyay!  i'm blonde!
15:38.51ManxPower[TK]D-Fender: he drops 5 hit points for using "Ur"
15:40.20[TK]D-Fenderwhee!
15:45.23Mercesteswhee.
15:52.55*** join/#asterisk ghenry (n=ghenry@212.159.59.85)
15:55.07*** join/#asterisk FreezeS (n=bla@82.208.157.125)
15:55.11FreezeShey guys
15:55.21FreezeSI've got some core dumps from asterisk when it crashed
15:55.26FreezeShow do I analize them ?
15:56.26pj_Ok, update to the captivating story of my lazy asterisk.... it seems to block on playing sound... the single RTP packet I get must be from vm-password, but then if I enter the password asterisk stay blocked on vm-youhave and doesn't go further
15:56.54pj_Any idea why * would block on playing a sound file ? (permission, file size, checked)
15:58.17[TK]D-Fenderpj_: any chance of a VLAN issue?
15:59.05pj_Well I would have seen them leaving the server at least... but now I'm really sure about it
15:59.19pj_because it freezes at "-- Playing 'vm-youhave' (language 'en')"
16:00.18pj_where it should go on and go "-- Playing 'digits/3'" instead
16:01.12pj_so it looks more like asterisk block reading the sound file
16:02.30*** join/#asterisk Waverly360 (n=Waverly3@231.sub-70-222-77.myvzw.com)
16:03.15Qwell[]Nugget: ping
16:03.31Waverly360Hey guys, I'm doing some research on "tie lines" and what I've found just states that they are just PRI's connecting one PBX to another..is that correct?
16:03.59Mercestespj_:  Permissions issue?
16:04.37pj_nope, no change, and chmoded 775 just to make sure
16:04.38[TK]D-FenderWaverly360: that is 1 kind.
16:04.43pj_good user, good group
16:04.44pj_:(
16:05.10[TK]D-FenderWaverly360: You could also be dealing with Frame relay gear, DS0 E&M boxes, TONS of stuff
16:05.14*** join/#asterisk Yozhik (n=Yozhik@72.171.70.169)
16:05.30jwhhm
16:05.32YozhikHello
16:05.38jwhany major carriers/operators in here?
16:05.40YozhikI am new using this chat
16:05.45jwh(who use asterisk in production)
16:06.06Waverly360[TK]D-Fender: Yeah, I'm trying to figure out what I have.  We're looking to connect an asterisk box to another PBX via a tie line.  I don't know exactly what they're using though.  I have the model number of the card in the PBX though.
16:06.17Yozhikjoined in to ask questions regarding using asterisk on MPLS
16:06.25jwhYozhik: shoot
16:06.29[TK]D-FenderWaverly360: How about boring old analog?
16:06.34Yozhikanybody kow anything about it?
16:06.51jwhYozhik: what do you want to know?
16:06.53YozhikWell, I have already setup asterisk servers
16:06.57Yozhik(mostly AAH)
16:07.05Yozhikan dnow I want to set it up in a company
16:07.05Waverly360[TK]D-Fender: the tie lines are already in place.  I've tried analog before with another pbx and we had a lot of problems with disconnect supervision
16:07.07NuggetQwell[]: pong
16:07.19Yozhikand they are about to implement MPLS to connect their remote offices
16:07.25Qwell[]Nugget: flightaware needs a way to call people when a flight lands/takes off :p
16:07.28Yozhikmy question is: is there a difference?
16:07.28*** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar)
16:07.38Nuggetyeah, I've thought about that, but it's low on the list.
16:07.38jwhYozhik: shouldn't matter
16:07.42[TK]D-FenderWaverly360: well a "tie line" is not a MEANS, its a PURPOSE
16:07.44Yozhiknot at all?
16:07.47jwhYozhik: as long as the providers network isn't congested
16:07.55Yozhikperfect J
16:08.08Yozhikthanks
16:08.13Waverly360[TK]D-Fender: Oh..so I need to figure out what they're using for tie lines
16:08.15[TK]D-FenderWaverly360: Just like people who think they bought an RRSP ;)
16:08.34*** part/#asterisk Yozhik (n=Yozhik@72.171.70.169)
16:08.56Waverly360[TK]D-Fender: hah hah
16:08.56[TK]D-FenderWaverly360: I caught a perch on my fishing "line"... perhaps you could use that! ;)
16:08.56*** join/#asterisk Yozhik (n=Yozhik@72.171.70.169)
16:09.17Waverly360[TK]D-Fender:Ok ok, so my knowledge of terminology sucks..I get it :P
16:10.07YozhikOne more question
16:10.19festr__hello, any idea, why in 1.4.4 does not work Pickup (show channels: SIP/festrntb-b68257f 117@macro-dial_ext_1 Ring    Dial(SIP/festrntb|60|jtT)
16:10.29festr__Pickup(117) should work is it?
16:10.30Qwell[]Nugget: and the search thing is totally not showing me this flight :p
16:10.44YozhikIs there a big difference in the programming and configuring of asterisk with business edition?
16:10.50filefestr__: specify the context.
16:10.58festr__file: i've tried that
16:11.00festr__file: no success
16:11.17festr__file: if you mean Pickup(117@macro-dial_ext_1)
16:11.19jwhYozhik: not sure, only used the normal version, but i'd assume its the same with more features
16:11.25filethat's not the context
16:11.26Waverly360[TK]D-Fender: Well, from what I can gather, it looks like their using a pri out of the back of a cisco 2600 into their current pbx to a rdtu1 card
16:11.32YozhikI am interested in implementing in the company but feel a little insecure to do it myself and so far I read at Digiums
16:11.33filewhere did you call Macro from?
16:11.39FreezeSmy asterisk crashes a lot here: 0xb6dac307 in set_pointer () from /usr/lib/asterisk/modules/format_mp3.so
16:11.44Yozhikbusiness edition is quiet easy
16:11.59Waverly360[TK]D-Fender: I'm guessing that I can plug that into a pri card on an asterisk box, and configure it to work the same.
16:12.00FreezeSand I get this interface.c:215 decodeMP3: Junk at the beginning of frame 49443303
16:12.04festr__file: so thats the context? i'll try thank you!
16:12.05neverbluegood softphone?
16:12.09neverblueekiga?
16:12.11festr__file: but in 1.2 it works
16:12.12[TK]D-FenderWaverly360: Entirely likely
16:12.16FreezeSwhat program should I use to encode proper mp3 files for asterisk ?
16:12.33Qwell[]Nugget: yeah, no info for DAL1185 today :p
16:12.34[TK]D-FenderFreezeS: LAME
16:12.47FreezeSthat's what I've been using :(
16:12.49Waverly360[TK]D-Fender: guess I won't know for sure unless I test it out...
16:12.51denonI agree, MP3s are lame ..
16:12.53denonuse ulaw
16:12.53FreezeSwhat is the proper format ?
16:12.53denon:)
16:13.03FreezeShmm, actually, very good idea
16:13.08YozhikJWH: have you set up an asterisk server with AAH?
16:13.16Waverly360[TK]D-Fender: Ok, I feel a bit better about this now...maybe one day you won't make fun of me when I ask these n00b questions :)
16:13.17jwhYozhik: no, I just use vanilla asterisk
16:13.18FreezeSwhat windows program can I use for that ?
16:13.19[TK]D-Fenderdenon: Amazing how I say so much with so few words ;)
16:13.28[TK]D-FenderFreezeS: **LAME**
16:13.29Yozhikforgive my ignorence
16:13.36Yozhikbut  how is it?
16:13.38denonhehe
16:13.41Yozhikis there a link to download it?
16:13.42FreezeS[TK]D-Fender: to convert into ulaw
16:13.55jwhYozhik: I meant, standard asterisk distribution from www.asterisk.org
16:14.10YozhikI see
16:14.33YozhikI will dowload it. Is there any handbook or manual which might guide me?
16:14.45Yozhikto set uo the vanilla edition?
16:14.52jwhvoip-info.org is a fairly good resource
16:15.02jwhbut the config examples that come with it are good
16:15.08[TK]D-FenderYozhik: ...
16:15.09[TK]D-Fender~book
16:15.10jbotmethinks book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
16:15.14YozhikYes
16:15.19YozhikGot it
16:15.24Yozhikto be honest
16:15.26[TK]D-FenderYozhik: Get reading then
16:15.34jwhhm
16:15.36Yozhikmy deficiency my be that I have never studied linux
16:15.43jwhis that the same one that was published a few yars ago?
16:15.44Yozhikan dhave been trying to learn on my own
16:15.45jwhyears*
16:15.57[TK]D-FenderYozhik: So NOW you're deciding to be honest?  ;)
16:16.02YozhikThe book is from year sago
16:16.13[TK]D-FenderYozhik: The book is still fine.
16:16.14`SeanGuys [TK]D-Fender can i have this work like this
16:16.15`Seanexten => s,1,Answer()
16:16.15`Seanexten => s,n,Playback(rh-british)
16:16.15`Seanexten => s,n,Dial(Zap/1,45)
16:16.21[TK]D-FenderYozhik: I suck at linux and do just fine.
16:16.22`Seaninstead oof Dail(Zap/1)
16:16.25`Seancan i pout a phone number?
16:16.29`Seanto forward it to
16:16.32`Seanlike my cell number?
16:16.40Yozhikinteresting, I feel intimidated to try the original version because
16:16.45YozhikI did not study linux
16:16.56[TK]D-Fender`Sean: Assuming Zap/1 is an FXO portm sure
16:17.01Yozhikand I have done fine with AAH
16:17.21Yozhikwhich forced me to program some stuff inlinux with commands
16:17.23[TK]D-FenderYozhik: Well don't.  You need to learn a FEW basics
16:17.41YozhikAny suggestiong for new bie?
16:17.52Yozhikwebsite, book, something?
16:17.55MercestesYozhik, vanilla asterisk
16:18.15festr__file: the problem is, that i've context for every extension. when calling from exten to exten it will call from that context macro(dial exten..) so i cannot predict context :(
16:18.25Yozhikso fa I understand vanilla asterisk is the standard version for download.
16:18.29Yozhikis that correct?
16:18.44Mercestesfestr__, That would be your problem then.  I have no clue what the question was, but that's definately the problem.
16:18.54jwhvanilla just means plain, as in, the original asterisk
16:19.18festr__Mercestes: :)
16:19.32[TK]D-FenderYozhik: Sounds like what you're looking for.
16:19.32`Sean[TK]D-Fender anyway to forward without having FXO?
16:19.39festr__so i've to backport working app_directedpickup from 1.2 to 1.4 then
16:19.48`Seanlike is it possible i can forward it out to my celly via asterlink
16:19.56[TK]D-Fender`Sean: Ask yourself this "how is * going to send the call to me?"
16:20.09*** join/#asterisk Trevor_b (n=tbenson@69.12.220.201)
16:20.49Yozhikany manual or procedure reference for vanilla version?
16:20.50*** join/#asterisk brea (n=brea@c-71-195-248-169.hsd1.ut.comcast.net)
16:22.26YozhikCan I download it from asterisk.org?  version Asterisk 1.4.5 ?
16:22.30[TK]D-FenderYozhik: the BOOK, and for download & instal : www.asterisk.org
16:23.00Yozhikok, perfect
16:23.04Yozhikgot to work
16:23.09joe[TK]D-Fender: what's the largest call center you know of using * ? ie number of agents/calls?
16:23.11YozhikI appreciate your help guys
16:23.14joe[TK]D-Fender: oh and hi btw :)
16:23.15Yozhikthank you
16:23.22Yozhikgood luck
16:23.32*** part/#asterisk Yozhik (n=Yozhik@72.171.70.169)
16:23.36[TK]D-Fenderjoe : not sure really
16:25.53*** join/#asterisk dlynes_laptop (n=dlynes@d154-20-9-152.bchsia.telus.net)
16:25.53*** join/#asterisk VJFROMGT (n=vjfromgt@static-68-161-227-229.ny325.east.verizon.net)
16:28.06Mercestesjoe:  Remind me waht acd and icd is again..
16:29.12joeMercestes: http://www.voip-info.org/wiki-ICD http://www.voip-info.org/wiki/view/ACD
16:30.18Mercestesjoe:  I didn't use ACD/ICD, I used queues and ring strategys
16:30.26Mercestesbut, I beat 100
16:31.15[TK]D-FenderMercestes: By ACD I'm sure he means app_queue , ICD is 3rd party
16:31.40joe[TK]D-Fender: thanks :)
16:31.53jwhanyone here who uses asterisk in a production/carrier environment, and knows what sort of simaultaneous calls you can get from asterisk, whether that was clustered etc?
16:32.29Qwell[]Somewhere between 1 and an infinite number of calls, depending on hardware, bandwidth, and many other factors
16:33.05*** join/#asterisk joebob777as7 (n=corn13re@67-42-57-190.eugn.qwest.net)
16:34.36joebob777as7how do i record a call? someone told me *1 but when i press it in the person on the other end hears me and nothing seems to happen... can someone point me in the right direction?
16:35.12*** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net)
16:35.20jwhjoebob777as7: Monitor()
16:35.43*** join/#asterisk uwe (n=uwe@dogbert.palnet.com)
16:35.52*** join/#asterisk waptaxi (n=waptaxi@45.151-224-87.telenet.ru)
16:35.56joebob777as7i don't want to record all calls only on demand...
16:36.23jwhmmm
16:36.53[TK]D-Fenderjoebob777as7: Go lookup "features.conf" on the WIKI
16:37.05joebob777as7ok thanks
16:37.10[TK]D-Fenderjoebob777as7: and "show application dial"
16:39.01joebob777as7[TK]D-Fender, where is the wiki?
16:39.32[TK]D-Fender~wikis
16:39.33jbotrumour has it, wikis is http://www.voip-info.org
16:39.34*** join/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)
16:39.40Dandrehello,
16:41.03Dandrewhen using the manager interface (AGI) what is the difference between update and append in updateconfig command? Update seem
16:41.23Dandreseems to append if the variable does not exists
16:45.09*** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net)
16:46.39*** join/#asterisk KDan (n=KDan@87-194-122-30.bethere.co.uk)
16:47.28*** join/#asterisk Alan_Hicks (n=alan@208.62.162.112)
16:47.41*** join/#asterisk atumanov (n=atumanov@192.219.104.10)
16:47.45Alan_HicksHowdy folks.  This is slightly off-topic, but I figured this would be the best place to ask my question.
16:48.50*** join/#asterisk zpertee (n=root@oh-69-34-21-229.sta.embarqhsd.net)
16:48.54atumanovI heard there's a moddable router which self-sufficiently could be used to run as an asterisk gateway? does anybody here know what that is?
16:48.57Alan_HicksI have a customer that wants to update her phone system, and I've considered an Asterisk solution.  Currently she has 24 PSTN lines, and would be better served with a PRI.  Some other companies are quoting her on proprietary PBX systems, and they know they can make theirs work with a PRI (which would save my client money monthly).
16:49.44zperteedoes anyone have or know of any example expect scripts that are used with asterisk
16:50.08Alan_HicksI can purchase the equipment needed to test PSTN lines, and perhaps even a PRI, but I have no actual PRI lines coming into our business, or any other business that I know of.  Is there anyway to setup a test box and test a PRI without commiting to an expensive lease from the phone company?
16:50.33atumanovis it linksys wrt54G?
16:50.36rsd99i know they make isdn simulators  would that help
16:51.13Alan_Hicksrsd99: I'm not certain. :^)  I'm realy a newb to asterisk (but I have read the book!) and I know very little about digital voice lines.
16:51.33*** join/#asterisk Slingky (n=Slingky@modemcable199.182-200-24.mc.videotron.ca)
16:51.55[TK]D-FenderAlan_Hicks: Whats to test?  It just works...
16:52.07Slingkycould someone tell me if linksys spa-2102 is best adapter for connecting analog phones and fax to asterisk ?
16:52.34Alan_Hicks[TK]D-Fender: I don't want to sell my client a solution that I haven't personally implimented and can garauntee will work.
16:52.34[TK]D-FenderSlingky: You should keep analog fax machines on a line of their own thats as far away from * as possible.
16:52.55falzhmm does Background() only playback .gsm's (not .wav)?
16:53.07[TK]D-FenderAlan_Hicks: We've done it, and continue to do so.  You're chicken&egging yourself.
16:53.11Alan_HicksIn other words, I want to know something inside-out before I give it to a client.  My reputation isn't worth chancing on anything.
16:53.25[TK]D-FenderAlan_Hicks: Lack of experience with * and telco gear is preventing you from DOING so.
16:53.44*** part/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)
16:53.46[TK]D-FenderAlan_Hicks: then sub-contract parts of it
16:53.49Alan_Hicks[TK]D-Fender: I wouldn't be as skittish if it were my business.
16:53.59Alan_HicksThat's an idea...
16:54.05*** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose)
16:54.33*** join/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net)
16:54.52pj_Hear hear, I have found the root of the evil in my * box, and would like your advice on how to fix it... It seems that unloading the zaptel modules makes the sound come back !
16:55.18Alan_HicksI just don't want to get in the situation where I may be struggling to make something work, particularly with this client.  The previous company she used sold her this horrible "solution" which they worked on for a year and never made work.
16:55.33Alan_HicksNaturally, she's skittish of anything "new" and untested.
16:55.55tzafrir_laptoppj_, what is your timing source? ztdummy or a card?
16:56.11pj_Hmmm... that could definitely be it
16:56.27pj_Right now a "card", however it's a redfone box, and line is not plugged in yet
16:56.36pj_so it's a ztd_eth card
16:57.19Alan_HicksCan anyone link me to some particularly good documentation on PRIs?
16:57.31tzafrir_laptoppj_, what does zttest show? is the timing source working ok?
16:57.43Alan_HicksThe O'Reilly book didn't have a lot to say on that subject.
16:58.05bkruseit will :]
16:59.40pj_( tzafrir_laptop checking ...)
16:59.47*** join/#asterisk marcan (i=1337@117.Red-88-5-77.staticIP.rima-tde.net)
16:59.57pj_Anyway it sounds good.. it means I probably will have my problem solved when I plug in the line
17:00.04*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
17:01.21galerasPlease be cool and give me your points to choose the right phone: Aastra, Grandstream, Linksys, Polycom, ...
17:01.28*** join/#asterisk paolob (n=donpaolo@196.3.84.214)
17:01.34*** part/#asterisk paolob (n=donpaolo@196.3.84.214)
17:01.36pj_it seems to block and freeze on "measuring accuracy..."
17:01.51Nuitari~phones
17:01.52jbotphones is probably http://bani.anime.net/phones/.  While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows:  Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else.  Do not consider Grandstream phones.  Ever.
17:02.10[TK]D-Fendergaleras: Polycom > All
17:02.29[TK]D-Fender~gs
17:02.29jbotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
17:02.31[TK]D-Fender^^^^^^^^^^^^^^^
17:02.58pj_we bought 50 of them and 1/4 were defective
17:03.07pj_I mean, _more_ defective than the other ones
17:04.01joebob777as7[TK]D-Fender, so do i need to create an extension to make Monitor() work i'm confused
17:04.16*** join/#asterisk dennisharrison (n=dennisha@68-114-124-171.dhcp.slid.la.charter.com)
17:04.22dennisharrisonman, I hate asking for help
17:04.26dennisharrisonbut I am pulling my hair out ;p
17:04.42EmleyMoorI appear to have a strange problem - whenever a certain number is dialed over my POTS line by Asterisk, I seem to be put through to a fax machine... is there any good way I can test what the FXO port is actually dialing?
17:04.43dennisharrisonhave an older box that lost power last night
17:05.10galerasEmlseMoor: set verbose 10
17:05.13dennisharrisonwhen it came back up it spews
17:05.14dennisharrisonJun 20 12:03:16 WARNING[12663] loader.c: Loading module chan_zap.so failed!
17:05.21dennisharrisonso ...
17:05.44Mercestesdennisharrison, check your permissions.
17:05.48galerasBtw, Thanks, seems Polycom is the right path
17:06.09dennisharrisonMercestes, :)  I hope you are right!
17:06.13[TK]D-Fenderjoebob777as7: Go read it all again.  you're looking for Dynamic features, and the "wW" optiosn in app_dial
17:06.23MercestesI'll bet /dev/zap is owned by root:root
17:06.42Mercestesor root:dialout more likely
17:07.05dennisharrisonyeah modprobe tells me it can't locate chan_zap
17:07.08dennisharrisonlemme check
17:07.09tzafrir_laptopadduser asterisk dialout
17:07.15Mercestesyou don't modprobe chan_zap
17:07.22tzafrir_laptop# adds asterisk to the group dialout on debian
17:07.22Mercestesyou modprobe zaptel  and whatever driver yoru card uses
17:07.26dennisharrisonok
17:07.28dennisharrisonzaptel then
17:07.29dennisharrisonyes
17:07.37MercestesDid zaptel load?
17:07.41dennisharrisonyes
17:07.45Mercestesk.
17:07.48tzafrir_laptopyour module's driver pull zaptel
17:07.51Mercestesnow load up your zaptel stuff.
17:08.03dennisharrisonok it is loaded
17:08.08Mercestesnow try to load asterisk
17:08.19tzafrir_laptopztcfg ?
17:08.23Mercestesdunno if it's /etc/init.d/zaptel start or wahtever it is...
17:08.33Mercestesztcfg == zaptel stuff btw.
17:08.36Mercestes>.>  I hope
17:08.44dennisharrisonZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25)
17:09.21dennisharrisonbad card?
17:09.54MercestesGoogle the error
17:10.27joebob777as7reading the wiki is great fun! stupid information database that is poorly outlined!
17:10.32*** part/#asterisk joebob777as7 (n=corn13re@67-42-57-190.eugn.qwest.net)
17:11.16dennisharrisonapparently this is because I just compiled 1.4
17:11.21dennisharrisonand I should be using 1.2 ?
17:11.22Mercestes...wow, chatting somethign derogatory then ditching the channel...
17:11.27rsd99anyone know a good site for ivr prompts?
17:11.33Mercestesthat's kinda like kicking a retarded kid in the nutz and then running for it.
17:11.43dennisharrisonMercestes, lol, he did say he had fun though
17:11.49dennisharrison... he has that going for him ;p
17:11.59Mercestesit is fun!  I was advocating it.  :D
17:12.11dennisharrisonhaha
17:12.38Mercesteshttp://www.syednetworks.com/asterisk-zaptelconf-configuration-with-x100p-fxo-and-tdm04b-cards/feed/
17:13.09Mercesteshttp://www.google.com/search?hl=en&q=ZT_CHANCONFIG+Inappropriate+ioctl+for+device
17:13.22MercestesI have to head to lunch soon but those results aren't looking real promising as a common error
17:13.56dennisharrisonyeah
17:13.58dennisharrisondoesn't seem so
17:14.04dennisharrisonI am thinking the card is borked
17:14.09dennisharrisoncheapo cards anyhow
17:14.11dennisharrisonI hate fxo
17:14.16dennisharrisonso backwards
17:14.20[TK]D-Fenderdennisharrison: Did you upgrade your Zaptel at the same time?
17:14.27[TK]D-Fender^^^^^
17:14.29dennisharrisonhey [TK]D-Fender ! ;p
17:14.36dennisharrisonyes I just updated zaptel to 1.4
17:14.44Mercestesoh...
17:14.48Mercestesthen it is a common error...
17:14.49NuggetQwell[]: DAL1185 has filed their flight plan.
17:14.54dennisharrisoneh?
17:15.02MercestesI thought by "old box" you meant...it just lost power one day then came back online with this new error
17:15.04dennisharrison1.4.3
17:15.06Nuggethttp://flightaware.com/live/flight/DAL1185
17:15.15[TK]D-Fenderdennisharrison: Any chance you updated your kernel as well?  That'd nuke the Zaptel .ko's
17:15.20dennisharrisonyes I did
17:15.21Mercestesyea, my google search is riddled with zap 1.4 erors on that.
17:15.23dennisharrisonI updated the kernel
17:15.26dennisharrisonwell ok
17:15.30dennisharrisonthe kernel was updated via yum
17:15.32dennisharrisonby someone
17:15.33[TK]D-Fenderdennisharrison: recompile & install Zaptel
17:15.34dennisharrisonnot sure who yet
17:15.36dennisharrisonI did
17:15.41dennisharrison1.4.3
17:15.42Mercestesdo a make clean
17:15.44dennisharrisonyep
17:15.47Mercestesthen a make && make install
17:15.47dennisharrisondid that first
17:15.54[TK]D-Fenderdennisharrison: re-modprobe zaptel & the drive for your card(s)
17:15.59dennisharrisoneven had to reset the spinlock typo for centos
17:16.14dennisharrison[TK]D-Fender, gotcha
17:16.38rsd99anyone know a good site for ivr prompts?
17:16.42[TK]D-Fenderdennisharrison: Then pastebin "cat /proc/interrupts", "ztcfg -vvvv"
17:17.06[TK]D-Fenderrsd99: http://www.theivrvoice.com/
17:17.16Jinglesway I understand it, you'll want to be sitting in front of the asterisk box when you 'cat /proc/interrupts'
17:17.23Jinglesand not be ssh'd in.
17:17.28Dr-Linuxtoday all issues :(
17:17.33[TK]D-FenderJingles: No difference
17:17.40Dr-Linuxnow cisco saying >>   Protocol Application Invalid
17:17.41[TK]D-FenderDr-Linux: www.drphil.com
17:17.45EmleyMoorgaleras: And that will help me how? Does it give more information than what asterisk told it to dial?
17:17.48*** join/#asterisk prashant_jois (n=prashant@mail.consolidated.ab.ca)
17:17.57MercestesJingles:  Close.  You want to be sitting at the asterisk box for "tcpdump -i eth0"  :)
17:18.32MercestesDr-Linux, It sounds like the version in your config file does not point to the P0S file you have availabe in TFTP.   or vice verse.
17:18.45MercestesDr-Linux, Or you are trying to rollback/forward to a version the system isn't liking
17:19.04dennisharrison[TK]D-Fender, http://pastebin.ca/579399
17:19.07prashant_joisI recently upgraded to asterisk 1.4.5 and now I'm seeing the message "Internal RTCP NTP clock skew detected" come up every so often, what does this mean?
17:19.08MercestesDr-Linux, regardless, it's your P0S file...
17:19.16EmleyMoorHmmm... it turns out it's MY fa machine that is for some reason answering
17:19.18EmleyMoorfax
17:19.33Dr-LinuxMercestes: well, i don't wanna do anything, this phone was just working fine from last 3 months, and now they reported me this issue
17:19.39Dr-Linuxand i can't do anything
17:19.41[TK]D-Fenderdennisharrison: show me the same w/ the modprobe calls, and add "dmesg" to it.
17:19.43*** join/#asterisk Toerkeium (i=oo@201.216.206.221)
17:19.55Dr-Linuxnot even i can see what TFTP server it is requesting for
17:20.04*** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com)
17:20.05Dr-LinuxMercestes: but i can ping this phone
17:20.11Dr-Linuxfrom local LAN
17:20.18rsd99i am just looking for some free ones.  this is a project i am doing at home for my own experience
17:20.19dennisharrisonmy manual modprobes or the barfs or both?
17:20.19[TK]D-FenderEmleyMoor: Let me guess, got an FXS port of a TDM400P with the fax attached, right?
17:20.39[TK]D-Fenderrsd99: Go record your own then.
17:20.46MercestesDr-Linux, That error doesn't mean the phone is offline, quite the contrary, it means, "I am online, I got my config server, but the firmware I am told I am supposed to have is different than the one that is available."
17:20.54[TK]D-Fenderdennisharrison: BOTH :)
17:21.07EmleyMoorNo, wrong. Fax is in parallel with FXO
17:21.10[TK]D-Fenderdennisharrison: more > less !
17:21.20dennisharrisonlol
17:21.21[TK]D-FenderEmleyMoor: That blows.
17:21.40Dr-LinuxMercestes: my all cisco phone is working since long with same version 7.4
17:21.54Dr-LinuxMercestes: btw, what's your suggestion
17:21.55Dr-Linux?
17:22.00*** join/#asterisk denon (n=denon@tooth.decay.org)
17:22.00*** mode/#asterisk [+o denon] by ChanServ
17:22.26MercestesDr-Linux, Check the .cnf for that phone and make sure it ponits to the correct firmware
17:22.34dennisharrisonmodprobe didn't error this time
17:22.42dennisharrisonhttp://pastebin.ca/579412
17:22.55Dr-LinuxMercestes: how can i check? since it's not requesting to TFTP server
17:23.14MercestesIt is not?  how are they configured?
17:23.21MercestesCisco pretty much demands a TFTP server.
17:24.33Dr-LinuxMercestes: yeah, but it's not requesting on my machine, but how  i can i check on what IP this phone is requesting for TFTP?
17:24.57[TK]D-Fenderdennisharrison: YUCK.
17:25.08dennisharrisonyeah
17:25.10[TK]D-Fenderdennisharrison: kill your zaptel source folder, and redo from scratch.
17:25.17dennisharrison[TK]D-Fender, ok
17:25.24dennisharrisonshould I go back to 1.2?
17:25.27dennisharrisonwhere it was working?
17:25.30dennisharrisonor should that not matter?
17:25.31rsd99dr linux, under settings, select network configuration
17:25.34MercestesDr-Linux, Oh, somewhere around settings, and then network settings and you browse on down to TFTP something.  TFTProuter and "manual TFTProuter" or something like that.
17:26.04Dr-LinuxMercestes: i know all that, but in this situation, i doesn't let me to press any button
17:26.15dennisharrisonthis trixbox box has been a bane to me
17:26.25dennisharrisonnothing else gives me problems
17:26.40Mercestesdennisharrison, That would be my expectation of trixbox
17:27.04dennisharrisonMercestes, yeah ... customer got hyped about all the widgetness, and it was one of my first fxo setups using asterisk
17:27.13dennisharrisonused to use avaya exclusively
17:27.39dennisharrisonnow I am using just straight asterisk on debian, but would rather not redo this box
17:27.53Mercestesfollow fender's suggestion
17:28.01Mercestesbut...I now undrestand why you have a cryptic error that doesn't show up often.
17:28.10dennisharrisonlol
17:28.13*** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com)
17:28.46dennisharrisonhttp://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.18.tar.gz sound good?
17:28.55dennisharrisonor should I grab trunk or some other tag/branch?
17:30.11[TK]D-Fenderdennisharrison: wait... you upgraded to 1.4 on trixbox?!
17:30.22dennisharrisonyeah
17:30.26dennisharrisonI haven't touched trixbox in ....
17:30.30dennisharrisonalmost a year
17:30.39[TK]D-Fenderdennisharrison: FreePBX doesn't support 1.4, and god only knows what else will blow up...
17:30.39dennisharrisonsomeone told me they used 1.4 now
17:30.44dennisharrisonyeah
17:30.45dennisharrisonnice :)
17:30.51dennisharrisongoing to 1.2 now
17:30.52*** join/#asterisk DaveCanoe (n=Dave@H6.C30.B96.tor.eicat.ca)
17:30.54[TK]D-Fenderdennisharrison: Get your ass off that POS!
17:30.58dennisharrisonI know :(
17:31.02dennisharrisonI know I should just redo it
17:31.19dennisharrisonjust ... god I am busy man
17:31.19[TK]D-Fenderdennisharrison: To know and not to do is not to know!
17:31.23dennisharrisonhey have you seen openmoko.org
17:32.02*** join/#asterisk matsk (i=matsk@h110n2fls32o882.telia.com)
17:32.02dennisharrisonwe have been writing python bindings for all the functions and a few other apps for it
17:32.06dennisharrisonshould be releasing soon
17:33.14dennisharrisonok
17:33.30dennisharrisonjust compiled
17:33.36dennisharrisonthe only error I saw was
17:33.36dennisharrison/usr/src/zaptel/ztd-eth.c:189: warning: initialization from incompatible pointer type
17:34.02dennisharrisonthis is 1.2.18
17:35.28Mercesteszap 1.2.18 or asterisk 1.2.18?
17:35.41dennisharrisonzap
17:36.35MercestesI wasn't aware there was a zap 1.2.18...hrm.
17:37.03dennisharrisonlol
17:37.09dennisharrisonwell now the address for the error has changed
17:37.11dennisharrisonfrom 25 to 6
17:39.48[TK]D-Fenderdennisharrison: Don't think of this as "tragic system failure', but rather "personal growth motivator and upgrade opportunity" ! :)
17:40.14dennisharrison[TK]D-Fender, lol ... dude, this blows my day ;p
17:40.27dennisharrisonoh well eh?
17:40.34[TK]D-Fenderdennisharrison: and you didn't even have to hit red-light district!
17:40.49dennisharrison[TK]D-Fender, although .... I might be looking for it tonight ;p
17:42.17*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
17:42.18codeyis it possible to run 2 IAX2 sessions on one port?
17:42.32codeymy asterisk acts like its not possible..
17:43.32bkruseLOL
17:43.34Corydon76-workSessions, yes.  Servers, no.
17:43.56bkruseoh you mean asterisk and another iax service? or 2 asterisks
17:44.01codeyso i can't trunk to my isdn-pbx and to my gsm-pbx?
17:44.01bkruseno :[
17:44.19codeyi've got pbx0, pbx1 and pbx2
17:44.24codeypbx0 is the isdn-"relay"
17:44.31codeypbx1 connects to pbx0 via iax2
17:44.34codeyand to pbx2 via iax2
17:44.36codeyon the same port
17:44.38bkruseyes
17:44.41bkrusethats fine
17:44.48codeyseems like its not
17:44.50bkruseyou can make more than 1 iax call correct?
17:44.52Corydon76-workWell, you can, if that port is simply an asterisk service which "proxies" for the other two services
17:45.04bkruseSo your saying you can call one, or call the other, not both?
17:45.09bkruseI find that hard to believe :[
17:45.19codeyi can call out over the isdn-pbx
17:45.23codeybut i cant call the gsm-pbx
17:45.43codeyso i would need a fourth box, that proxies the connections of pbx0 to pbx1 and pbx2?
17:45.43bkrusethen its a problem between that box and gsm, not because your using two at the same time
17:45.53Corydon76-workWe're talking 3 different machines on the same network?
17:46.05codeyone box is on a dynamic ip
17:46.11codeythe other 2 are on the same net
17:46.26Corydon76-workThen the box on the dynamic IP needs to register to the others
17:46.32EmleyMoorI've fixed the problem - certain numbers were waking up the fax machine
17:47.14codeythe dynamic one rejects me with:  No authority found
17:47.25Corydon76-workThat's a different problem, then
17:47.33Corydon76-workThat's authentication
17:47.35EmleyMoorI wish I could detect dring but that won't be possible here with 1.2
17:48.03bkruseyep
17:48.11codeyCorydon76-work: okay.. i'm trying Dial(IAX2/pbx2/1223)
17:48.13bkrusenot because it cannot do 2 on the same port :P
17:48.14codey*123 even
17:48.36[TK]D-FenderEmleyMoor: You could get an of-the-shelf DRING line selector and put it in front.
17:48.41Corydon76-workcodey: probably your secret isn't set up correctly
17:48.54EmleyMoor[TK]D-Fender: I could
17:48.56Corydon76-workor it doesn't match
17:49.04*** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk)
17:49.23EmleyMoorWhat I do for the moment is use a fax machine that only responds to one of them and delay asterisk answering until after the fax machine has had a chance
17:49.46codeyCorydon76-work: it is. i've copy&pasted it from the iax.conf on pbx2
17:50.02Corydon76-work~pb
17:50.03jbotfrom memory, pb is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org
17:50.17Corydon76-workShow me
17:51.10*** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net)
17:51.28magic_hatanyone got experience installing/using * with Xubuntu?
17:51.37rob0I have a piddly little system here: X101P to PSTN, Sipura SPA-2000 to analog phone. I'm hearing echo when calling out to PSTN. Any ideas what to check first? No fancy settings so far, defaults.
17:52.01rob0(That is: using the Sipura to bridge to Zap.)
17:52.39Corydon76-workmagic_hat: we strongly recommend that you do NOT run X on your Asterisk server
17:53.05magic_hatcorydon76: why's that?
17:53.17Corydon76-workBecause X eats interrupts
17:53.29*** join/#asterisk jpeeler (n=thepeel@5-124.generic.clemson.edu)
17:53.35magic_hatah.
17:53.37magic_hatcrap!
17:53.45bkruseuse X if you want to set it up
17:53.50bkrusebut def not during usage or production :[
17:53.55bkruseit doesnt play nice
17:53.57Corydon76-workIt's fine for development machines, just not for production
17:53.59*** join/#asterisk mihinomenest (i=pTBB@cerebus.clandestineresearch.com)
17:54.05rob0Hmmm, maybe that's my problem too.
17:54.17magic_hatyeah, i'm looking for a production setup
17:54.19Corydon76-workEven on my development machine, I occasionally get jittery audio because I run X
17:54.34bkrusefigures
17:54.37rob0Because it seems okay when calling in on SIP to the Sipura-connected phone.
17:55.15magic_hatwhat's easy to setup, runs well with *? CentOS?
17:55.17Corydon76-workmagic_hat: if you're accustomed to Xubuntu, I'd suggest using Ubuntu with the GUI
17:55.31Corydon76-worki.e. install Ubuntu as a "LAMP Server"
17:55.50Corydon76-workerr, withOUT the gui
17:56.07rob0Any GNU/Linux should be fine. The choice should be made according to what you as an admin need.
17:56.31magic_hatCorydon: I'm not accustomed to Xubuntu... just saw an opportunity to take an xubuntu box off someone's hands.
17:56.41codeyCorydon76-work: http://slexy.org/paste/3172
17:57.01awkwell xubuntu has nothing to do with the console
17:57.10awkit doesn't use the same way to update packages, you have a stupid ui pakage manager so knowing apt isn't needed.
17:57.13magic_hatBasically what I need is something easy to set up that runs * well. I'm using it solely as as an asterisk server.
17:57.44awkmagic_hat well i have a couple debian boxes a couple ubuntu boxes some fedora boxes some gentoo boxes and they all run lovely
17:57.48awkI believe there is alot of documentation for ubuntu though
17:57.51*** join/#asterisk [hC] (n=hardcore@66.119.167.162)
17:58.01awkaswell as fedora
17:58.05bkruseha, its like the biggest room in this server
17:58.09Corydon76-workcodey: uh, which one is the dynamic server?
17:58.22codeypbx2
17:58.29the_5th_wheelis there some way to use a modem as an incoming line as one can do with isdn?
17:58.41Corydon76-workcodey: why register from pbx1 to pbx0, then?
17:58.43codeyi actually had to set all hosts to host=dynamic because i wasn't able to register to pbx0 anymore
17:58.48awkthe_5th_wheel no...
17:58.57awkan isdn modem with a hfc chipset yes
17:59.02codeyCorydon76-work: pbx1 is the box that does all my sip-stuff
17:59.11codeyCorydon76-work: routes calls to either gsm or isdn
17:59.24Corydon76-workcodey: you should have 2 register statements, both on pbx2... pbx0 and pbx1 should know each other by static IP, not by registration
17:59.26codeyCorydon76-work: so i actually need to hook up to pbx0 to do the isdn stuff.
17:59.43codeyCorydon76-work: they did - til i added the pbx2 user
17:59.51awkthe_5th_wheel but i wouldn't recomend using isdn modems, ive had some bad eperience with a couple boxes. some worked allright, but iyou get what you pay for.
17:59.55Corydon76-workcodey: not according to what you have here
18:02.01the_5th_wheelwhat would the cheapest way be to interface a analouge line to a server then? just for experimenting and stuff?
18:02.04codeyokay, just changed it ... it wasn't able to get called a few mins ago
18:02.07codeyso i had to change it.
18:02.15codeyhow pbx0 is static on pbx1
18:02.17codey*now
18:02.52codeybut i'm still not able to call pbx2
18:03.25codeyomfg
18:03.27codeyfound the problem
18:03.29Corydon76-workcodey: get rid of your friend definitions... spell each one out as user and peer
18:03.40codeythe iax name wasn't the same on pbx2
18:03.43*** join/#asterisk austin_j (n=chatzill@austin-j.its.dist.maricopa.edu)
18:03.49awkthe_5th_wheel you get analouge cards, but i wouldn't sugest them either, ive had problems with them too.
18:04.02awkthe_5th_wheel so your experiemnce wouldn't be the best.
18:04.09austin_jAnyone use asteribank from xorcom?
18:04.35EmleyMoorThis is going to sound really stupid - I can't get MusicOnHold to work
18:05.55the_5th_wheelawk, well, what would i then use? i cant get a digital line(as theanalouge line comes  at next to nothing)(this is for a not for profit org)
18:06.40awkthe_5th_wheel well you can get analouge cards just make sure you buy a decent card, do your homework.. you need something with good echo cancelation..
18:07.21awkthe_5th_wheel the last thing you want, is calls dropping, some calls not getting answered. etc etc
18:07.44awki've had numerous problems, but once i started spending money all the problems went away :)
18:08.00the_5th_wheelawk: the problem is i cant spend much money
18:08.14awkthe_5th_wheel what is your budget?
18:11.17the_5th_wheeleverything i want to do is just to make my life easier. my budget is a couple of under rands(which translates into maybe a 100 US$)
18:11.29*** join/#asterisk raining (n=raining@135.197.233.220.exetel.com.au)
18:11.50codeysame problem again
18:11.52codeywhat the fuck
18:12.02[TK]D-Fenderthe_5th_wheel: how many lines?
18:12.09raininggreetings all
18:12.37the_5th_wheel[TK]D-Fender: two. one one a premicell one pots
18:12.51EmleyMoorHow do I see what formats are actually supported for native MoH?
18:13.03awkthe_5th_wheel iesh, i'm not the right person to ask then. try ask somebody about cheap analouge cards
18:13.16[TK]D-Fenderthe_5th_wheel: ....
18:13.23[TK]D-Fender~ygwypf
18:13.24jboti heard ygwypf is You Get What You Pay For.  If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there.
18:13.28raininghow can i set up my asterisk to accept direct calls to say sip:123@mydomain.com?
18:13.49dennisharrison[TK]D-Fender, alrighty man, thanks a bunch again :)  I am headed out to go futz with it
18:15.33*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
18:16.04*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
18:18.56raininghello?
18:22.31*** join/#asterisk mxmasster (n=Max@129.47.12.101)
18:23.06uwehello, any idea what can possibly make the music on hold play only once ?
18:23.12*** join/#asterisk rogerius (n=mama@201.29.18.64)
18:23.15rogeriushello
18:24.38raininghi
18:24.51raininghow do you alllow unauthenticated incoming calls via SIP?
18:25.04rogeriushow do i configure asterisk to pass a call to openser? when i type the extension of a sip number, i want to call this sip registered @ openser, in the same machine
18:25.08rainingbasically i want anyone to be able to call sip:500@mydomain.com
18:25.17rogeriuscalling from pstn
18:25.48rainingset up your dialplan?
18:26.15awkrogerius wouldn't you do it the other way around?
18:26.45rogeriusawk what do you mean?
18:27.22awkwell why would you wat asterisk to do the registration and not the ser?
18:28.01awkbrb
18:28.51EmleyMoorI got native MoH working but it sounded awful!
18:29.11codey<PROTECTED>
18:29.12codeywhee
18:29.13codeynew error.
18:29.38codeyand that extension is there for sure.
18:29.38VJFROMGTI am getting a SIP/2.0 401 Unauthorized but user/pass are correct
18:29.53*** join/#asterisk fbffff (n=fbffff@dsl092-129-089.chi1.dsl.speakeasy.net)
18:30.02VJFROMGTextension is there for sure
18:30.15VJFROMGTif i config it with xlite it works,, with pap2 i get 401
18:30.16awkback
18:30.45awkcodey: and is the context correct?
18:30.47*** join/#asterisk kombi (n=kombi@213.160.14.18)
18:30.52VJFROMGTpap2 reads Registration State:Can't connect to login server
18:31.09*** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il)
18:33.11holiday_42VJFROMGT: I have pap2v2, had to turn up the verbosity in the asterisk CLI... I found with my pap2 the secret was munged... had to change it to some other value, save settings, change it back, save again.
18:33.47VJFROMGTwhat sort of value works for u?
18:33.54holiday_42VJFROMGT: oh, sound like it's not getting to the * box at all in your cae.
18:34.13VJFROMGTbut * is giving a 401 as soon as i turn on pap
18:35.05codeyawk: yes
18:37.43*** join/#asterisk GaryH (n=GaryH@wallace.garysoft.co.uk)
18:38.07[TK]D-Fendercodey: "No such context/extension" <- this is NOT lying.  your configs are wrong keep reading until you find it or your eyes bleed.
18:38.57holiday_42VJFROMGT, i'll check mine.. sec.
18:39.06*** join/#asterisk tsurko (n=tsurko@77.70.24.142)
18:39.16codeytsurko: it IS there
18:39.17codeypbx0:/etc/asterisk# grep $(cat iax.conf | grep context | sed -e 's,^context=,,') extensions.conf
18:39.20codey[fsincoming]
18:39.38*** join/#asterisk GaryH (n=GaryH@wallace.garysoft.co.uk)
18:39.42codeyxten => _XX,1,Answer()
18:39.42codeyexten => _XX,2,Dial(SIP/${EXTEN},60)
18:39.42codeyexten => _XX,3,Congestion
18:39.45codey+e
18:40.09VJFROMGTif i want to see pasword, what should i set verbose to?
18:40.32holiday_42I dunno... i just used 9 nines
18:40.40*** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net)
18:41.29codey[TK]D-Fender: i don't even see a call coming in on the other pbx
18:42.30[TK]D-Fendercodey: pastebin the CLI output of a call that fails with SIP debug & verbose 10.  And include the full configs related to that server (sip & extensions)
18:42.44*** join/#asterisk marcan (i=1337@117.Red-88-5-77.staticIP.rima-tde.net)
18:43.02codey[TK]D-Fender: it's not sip
18:43.10[TK]D-Fendercodey: IAX, whatever
18:43.22[TK]D-Fendercodey: Everything related to the call attempt
18:43.28[TK]D-Fender\~pb
18:43.30[TK]D-Fender~pb
18:43.31jboti guess pb is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org
18:43.32[TK]D-Fender^^^^^^^^^^^^^^^
18:45.14Alan_HicksWould anyone recommend the O'Reilly book "T1: A Survival Guide" to some one interested in learning more (a lot more) about T1s, or is there another book you prefer for that?
18:45.33*** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net)
18:45.39Jinglesmost anything from ORA is worth owning as reference material.
18:46.46Alan_HicksYeah, I love their books, but was wondering if there might be something better out there that I don't know about.
18:47.56[TK]D-FenderAlan_Hicks: may be more of a DATA book than a VOICE book....
18:48.29holiday_42VJFROMGT, i leave the SIP stuff alone, just change line 1 (and line 2 if you want) just specify username/password/proxy server & registrar address
18:48.44Alan_HicksThe more I read online, the more I realize how much I don't know, and the more my curiosity grows.
18:48.55*** join/#asterisk perf3kt (i=perf3kt@iupui-vpn-32-59.noc.iupui.edu)
18:50.40[TK]D-FenderAlan_Hicks: A lot of this stuff you only really learn when you start getting your hands dirty, and *'s side of T1 is a lot simpler than the DATA side
18:59.34*** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net)
19:02.31*** join/#asterisk ramindia_ (n=ramindia@202.63.96.9)
19:03.21*** join/#asterisk shtoom (n=godson@59.93.115.56)
19:04.07ramindia_any one recomend me tools to diagnosis voice break problems in network ?
19:11.54*** join/#asterisk enmaca (n=enmaca@189.157.117.149)
19:13.14*** join/#asterisk mrdigital (n=mrdigita@207-172-229-100.c3-0.tlg-ubr2.atw-tlg.pa.cable.rcn.com)
19:13.43mrdigitalwhats the exten => command to dial ring group 600
19:14.00*** join/#asterisk linuxsouth (n=chatzill@72.242.225.99)
19:15.46*** part/#asterisk Alan_Hicks (n=alan@208.62.162.112)
19:17.29*** join/#asterisk sysreq (n=sysreq@210.145-ppp.3menatwork.com)
19:17.44[TK]D-Fendermrdigital: "show application dial" and there is no inherent thing called "group 600".
19:18.14ramindia_[TK]D-Fender: any suggestion on diagnosis voice breaks
19:19.23*** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net)
19:19.58[TK]D-Fenderramindia_: check your bandwith , latency & jitter
19:21.14*** part/#asterisk linuxsouth (n=chatzill@72.242.225.99)
19:22.09*** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net)
19:22.09*** mode/#asterisk [+o mog] by ChanServ
19:23.12*** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga)
19:23.36*** join/#asterisk eatmypiano (n=eatmypia@host81-155-21-242.range81-155.btcentralplus.com)
19:30.37ramindia_[TK]D-Fender: i dont see any problem in terms of bandwidth ,latency
19:30.45ramindia_how can i check the jitter any tools
19:32.32*** join/#asterisk linsouth (n=allen@72.242.225.99)
19:32.44*** part/#asterisk shtoom (n=godson@59.93.115.56)
19:33.55[TK]D-Fenderramindia_: Don't know.
19:34.30_DAWramindia_:  ethereal does some basic stuff.  Get me a tcpdump and I'll run it through wineyeq for you.
19:37.07*** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net)
19:42.21errrwhen using the directory from an ivr, where does it look to see if it can find someone to match whats being searched for?
19:45.00russellbDoes anyone else get *really* annoyed with the coiled cables attached to phone handsets?
19:45.09rob0:)
19:45.10[TK]D-Fendererrr: "show application directory"
19:45.10russellberrr: voicemail.comf
19:45.27errrgreat guys thanks
19:46.32errris there anyway I can make it look on some other server?
19:46.45errrwe have a central voicemail server
19:47.02rob0I used to work in a multi-floor office building, and we'd take the handset cords to the stairwell to dangle / untangle them.
19:47.03russellbyou can store voicemail config in a database that is available across servers ...
19:47.04[TK]D-Fendererrr: Sure, mount its volume into the voicemail folder.
19:47.15russellbrob0: nice :)
19:48.15[TK]D-Fenderrob0: Shows your users were shcmucks who'd allow long term mangling to occur.  I had the same phoone with my Norstar 8x24 for at least 7 years and it was IMPECCABLE
19:49.21*** join/#asterisk SirThomas (n=tomc@mail.kendeco.com)
19:49.46rob0That, and low-bid (state gov't) equipment :)
19:52.30*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
19:56.42errrhey I have some patton sn4112's has anyone ever set any of these up?
20:05.15*** join/#asterisk philippel (n=p_lindhe@c-24-17-254-189.hsd1.mn.comcast.net)
20:08.33*** part/#asterisk dcm_ (n=dcm@207.59.3.77)
20:09.05*** join/#asterisk pifiu (n=someone@216.5.79.1)
20:13.57*** join/#asterisk rsd99 (n=rsd095@h-67-103-23-130.phlapafg.covad.net)
20:14.55*** join/#asterisk sponger (n=sponger@66-193-153-10.static.twtelecom.net)
20:15.56breaWhat will happen if all of my identifiers don't get cleared?
20:16.04breaWill it take up memory?
20:18.54*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
20:18.58philippelran into an interesting challenge I'm wondering if I'm missing anything a way to excape it, the following line of dialplan (which I auto-generate):
20:19.01philippelexten => 220,n,Set(__ALERT_INFO=${IF($["x${ALERT_INFO}"="x"]?http://127.0.0.1/Bellcore-dr3:${ALERT_INFO}})
20:19.43philippelthe : in http: is interpreted as the end of the 'true' result
20:20.36philippelI've tried excaping it with \, putting the whole thing in quotes and even setting a variable one statement ahead and using the variable instead of the string in the clause, all with the same results - any thoughts, different way of escaping it?
20:21.24philippel(vs. a different way of writing the dialplan which I can otherwise do but this seems to have interesting consequences for the general use of the IF() function
20:21.54spongerthat has always seemed to me one of the sucky things about the asterisk dialplan
20:22.07Mercestesanyone installed asterisk on Debian?
20:22.18Mercestess/debian/ubuntu/
20:22.32spongerphilippel check out adhearsion (am i allowed to say that in here?_
20:22.45spongerits lets you write the dialplan in ruby
20:22.57philippelsponger I'm not interested in doing that
20:23.22rsd99i just got asterisk up and running last night.  and wow......it'll take me a while to learn
20:23.27*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)
20:24.00philippelI'm interested to know if there is a way in the IF clause to handle a : in the results escaped some how or if I have to change all of my dialplan generation for such cases
20:24.08Mercestes[TK]D-Fender, what do you install * on?
20:24.32*** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br)
20:24.52[TK]D-FenderMercestes: Generally on Inten X86 based computers
20:24.57[TK]D-FenderIntel*
20:25.19Mercestes[TK]D-Fender, Sorry, what distro?
20:25.19[TK]D-FenderMercestes: Usually CentOS, if not, Slackware
20:25.56Mercestesoh...
20:26.51spongerhas anyone in here got ztdummy running on a VE using openvz?
20:27.47tzafrir_laptopMercestes, I got it installed on Ubuntu (not much more than that)
20:28.30Mercestestzafrir_laptop, Yea, I get the compile errors on it, I founda howto that might (should) work.
20:28.43tzafrir_laptopWhat compile errors?
20:29.20tzafrir_laptoppackages names generally rather similar to debian
20:29.52pifiuhey fender
20:29.53pifiuwasup
20:30.51*** join/#asterisk diclophis-work (n=jbardin@65.203.37.58)
20:30.54diclophis-workhello all
20:31.05diclophis-workhow would i get all manager events/actions logged in the console?
20:31.20*** join/#asterisk kombi (n=kombi@213.160.14.18)
20:31.21[TK]D-Fenderdiclophis-work: Start coding
20:31.22diclophis-worki thought there was a manager.conf setting for it, but i cant seem to find it
20:31.29[TK]D-Fenderpifiu: y0
20:32.02kombiwold someone know the command line argument to start wireshark and listen on eth0? The manpage is somewhat hefty there..
20:32.29kombiof the top of his/her head I mean..
20:32.45diclophis-work[TK]D-Fender: are you saying that the manager interface doesnt have any facilities to log to the console?
20:32.47*** join/#asterisk robin_z (n=robin@rapid2.gotadsl.co.uk)
20:32.50kombis/wold/would/
20:32.56robin_zmeep?
20:32.56[TK]D-Fenderdiclophis-work: Nothing I'm aware of.
20:33.02diclophis-workarg
20:34.02*** join/#asterisk nasls_lsa (n=chatzill@athedsl-366565.home.otenet.gr)
20:34.23diclophis-workcould have sworn i saw a setting for it
20:34.26kombior is there a command line packet sniffer somewhere?
20:34.32*** join/#asterisk zotz (n=zotz@24.244.163.157)
20:34.51diclophis-workkombi: isnt that what wireshark is?
20:34.52[TK]D-Fenderkombi: tetherial
20:34.59diclophis-workthe only other thing i know of is etheral
20:35.03[TK]D-Fenderkombi: I think its called tshark now
20:36.02kombithanks!
20:37.26*** part/#asterisk falz (n=falz@proxy.supranet.net)
20:37.31mockerAre toll free numbers pointed directly to a PRI or are they pointed to a DID at the LEC?
20:37.46mockerI always though it was done at the LEc.
20:37.50*** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br)
20:38.21Jinglestethereal is the command line sniffer, wireshark is the gui version.
20:38.25Jinglesgood times. I use it often.
20:38.44anonymouz666good times bad times you know I had my share
20:40.35*** part/#asterisk ramindia_ (n=ramindia@202.63.96.9)
20:40.44*** join/#asterisk joshaidan (n=brianj@thunderbay-voip-4.vianet.ca)
20:41.59*** join/#asterisk skymeyer (n=skymeyer@bxlsrvit03.itconnect.be)
20:43.03joshaidanDoes asterisk give you the option to erase your temporary greeting?
20:43.20Qwell[]joshaidan: same way you record it
20:43.42[hC]you know, early versions did not give you a choice
20:43.45[hC]you had to remove it from the fs
20:44.11joshaidanah... thanks :)
20:44.14[hC]chances are if you are running anything above 1.2.5(?) you just go in to the same place you did to record it
20:44.17[hC]and theres an option to delete it
20:47.27*** join/#asterisk skymeyer (n=skymeyer@bxlsrvit03.itconnect.be)
20:47.46*** join/#asterisk tsurko (n=tsurko@77.70.24.142)
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20:52.34*** join/#asterisk diclophis-work (n=jbardin@65.203.37.58)
20:54.41mrdigitalhey voicemeup?
20:57.28perf3ktI get a 407 proxy authenication required in my debug when trygin to recieve a call in
20:57.38*** join/#asterisk yangvnc (i=yang@84-16-238-88.internetserviceteam.com)
20:57.38*** part/#asterisk rsd99 (n=rsd095@h-67-103-23-130.phlapafg.covad.net)
20:57.39kombipeople don't hear me when they call, everything else is fine. Where must I investigate?
20:57.48yangvncHello, I am looking for VOIP provider which can give FREE (european) numbers accessible over PSTN, can anyone suggest me anything?
21:00.57*** join/#asterisk nasls_lsa (n=chatzill@athedsl-366565.home.otenet.gr)
21:01.07nasls_lsakl-anill ?!?!?
21:01.10*** part/#asterisk nasls_lsa (n=chatzill@athedsl-366565.home.otenet.gr)
21:01.11kombiJingles: would you know a nice way to invoke it, debugging by one way audio?
21:01.11*** join/#asterisk skymeyer (n=skymeyer@bxlsrvit03.itconnect.be)
21:02.10*** join/#asterisk mitcheloc (n=mitchelo@001-794-703.area1.spcsdns.net)
21:04.26kombiwhich ports are used for RTP?
21:04.54austin_jAnyone here have experience with McCloud USA for Voice/Data connectivity?
21:05.11Jingleskombi: invoking the command line packet sniffer?
21:05.26austin_jMercestes: Asterisk works on Ubuntu.  My preference is CentOS though.
21:05.28*** join/#asterisk skymeyer (n=skymeyer@bxlsrvit03.itconnect.be)
21:06.11Mercestesaustin_j, Your awesome
21:06.21*** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com)
21:07.00kombiJingles: just did like tshark -i eth0 and noticed way more RTP packets with two way audio..
21:07.18kombifirewall is open for 10000:20000 though..
21:07.32Jinglesright - when you're engaged in a 'phone conversation', there's loads of RTP traffic - carries the voice payload.
21:07.56austin_jAnyone plopped freepbx on top of an existing asterisk installation before?
21:08.37*** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net)
21:08.46kombiJingles: my inbound calls have no incoming audio and - as I now found out - hardly any RTP packets, I wonder why since iptables is wide open
21:09.05tzafrirapt-get install freepbx ...
21:09.36bkruseapt-get remove self
21:09.39austin_j:)
21:10.08kombiJingles: I wonder where I should investigate next
21:10.18austin_jI'm just wondering what'll happen when freepbx encounters the hand written sip.conf/etc
21:11.09austin_jI've only ever installed freepbx on a system with nothing on it; that's why I ask
21:11.25*** join/#asterisk CunningPike_ (n=CunningP@204.239.12.183)
21:11.56tzafriraustin_j, with sip.conf the requirements are not that strict. However you're taking your chances regarding when an upgrade of freepbx will run over your sip.conf
21:12.01kombion a box with its own static ip, absolutely no need for nat right?
21:12.16austin_jtzafrir: I suspected that ;)
21:12.36austin_jkombi: maybe for the phones connecting to it
21:13.35kombiaustin_j: I meant in [general] where the sip provider is registered, no nat=yes necessary
21:14.39austin_jkombi: if both ends are not natting then nat=yes is not needed.  It doesn't hurt as far as I can tell.. I always set nat=yes.
21:14.40dijungalanyone here with experience in getting Cisco (H.323) to pass calls into Astrerisk 1.4.5
21:14.42dijungal?
21:15.28austin_jkombi: nat=yes tells asterisk to put extra stuff in the headers
21:17.13mrdigitalwhy am i getting # not in service?
21:17.17*** join/#asterisk skymeyer (n=skymeyer@bxlsrvit03.itconnect.be)
21:18.19kombiJingles: in tshark, how do you filter udp?
21:18.20austin_jmrdigital: misconfigured?
21:18.48perf3kttz: you knwo of the 407 authenication error, that would prevent incoming calls
21:18.53holiday_42it could be true (for example the sip user not registered) or misconfiured dialplan
21:19.40*** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl)
21:20.15tzafrirperf3kt, there are quite a few potential reasons
21:20.22holiday_42peek at asterisk cli, with verbose should give a hint
21:20.32tzafririncorrect password?
21:21.02*** join/#asterisk joetester (n=joeteste@216.191.34.13)
21:21.05tzafriryou should see users and passwords with 'sip show peers' (sorry, I know you're not a newb)
21:22.51Jingleskombi: not udp (in the filter line)
21:23.34*** join/#asterisk ramindia_ (n=ramindia@202.63.96.9)
21:23.34kombithanks!
21:24.34kombi(what do you mean by "in the filter line" though?, can you give an example?
21:25.02perf3ktholiday: were you talking to me
21:25.21kombitshark -i eth0 -f udp looks good
21:25.23Jinglesthere's a spot in the top toolbar area of wireshark.
21:25.31Jinglesoh - not using the gui?
21:25.38kombioh, no gui here (never had one ever..;)
21:25.46*** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com)
21:25.56kombiat least not on a nix
21:26.25*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
21:26.29Jingleswell, you can do just a flat capture (tethereal -i eth1 -w capturefile)
21:26.42Jinglesthen open the capturefile in wireshark on a machine with a gui.
21:27.04JinglesI can't stand fiddling with packet caps without a gui tool.
21:27.26kombithey nicely run by you there..;)
21:27.54kombiI can see loads of RTP zooming by on the ok connection and way less on the one with one way audio
21:28.06kombijust what does that tell me?
21:28.15Jinglesok. so, usually that means there's a firewall blocking your rtp traffic.
21:28.36bkruseor being filtered, and slowly.
21:28.50kombi$IPTABLES -A FORWARD $UDP -d $IP_afm/32 --destination-port 10000:20000 -j ACCEPT though..
21:29.20kombislowly.. hmm
21:30.54kombinay, couldn't, the working line shows hundreds of packets coming in through it, it's the inbound call that does not let audio travel outside..
21:31.08kombiweired..
21:33.42bkrusei could see with maybe hundreds of calls, but not 1.
21:35.47*** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net)
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21:37.30*** part/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net)
21:37.32kombikbruse: I wonder where all those RTP packets go..
21:40.02*** join/#asterisk rantsh (n=chatzill@201.210.16.238)
21:40.19rantshHi guys
21:40.43kombihi rantsh
21:41.03*** join/#asterisk Trevor_b (n=tbenson@69.12.220.201)
21:41.04rantshI came back just to tell you you guys rule! :) I got to record calls trouble-free with * 1.4.5
21:42.31rantshand I believe I understand * better than ever before (remember, I'm a complete n00b hehehe)
21:43.37*** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com)
21:45.19Qwell[]rantsh: that's always a good thing
21:46.08rantshyup
21:46.14*** part/#asterisk jm|laptop (n=jm|home@zen.jamiem.com)
21:46.34*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
21:54.44rantshanyway, on my new assigment now (actually it was my first assigment but my boss was mercifull and let me study with call recording)
21:55.32rantshI need to setup codec translations, I've been searching all over voip-info.org, but haven't got any useful information on where to start looking
21:56.10rantshanyone knows where I can start at least searching or reading on this subject?
21:56.28*** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net)
21:57.07*** join/#asterisk dijungal (n=chatzill@209.59.110.5)
21:57.40dijungalanyone with Cisco <-(H.323)-> Asterisk experience?
21:58.16dijungali'm trying to send voip calls from a cisco router to asterisk using ther H.323 protocol
21:58.25[TK]D-Fenderrantsh, * translates automatically if both sides of a bridged call can't agree on a common codec.
21:58.35dijungali see the router actually sending packets but asterisk does not pickup
21:59.21[TK]D-Fenderdijungal, Have you set up an H.323 entry for it to use & a dialplan context to land in for processing?
22:00.05dijungali've setup the h.323 entry in h323.conf (i'm using chan_h323)
22:00.17dijungali have not setup any dial plans as yet
22:00.41dijungali am expecting asterisk to answer the call and then drop it.. cause it does not know how to route it
22:01.10dijungalbut it does not even acknowledge the call
22:01.35[TK]D-Fenderdijungal, if * doesn't have an appropriate dialplan to match, you won't see ANYTHING unless you're in chennel debug mode.
22:01.50[TK]D-Fenderdijungal, Even picking up and hanging up is all DIALPLAN.
22:01.58dijungalhow do i go into channel debugging mode ?
22:02.29[TK]D-Fenderdijungal, First just set the context in your H323 entry and make a catch-all exten for it to land on.
22:02.37dijungalok... i will create a simple dialplan and see if it works.. Answer, wait(2), then hangup :)
22:02.48[TK]D-Fenderdijungal, that should tell you if your auth is correct without too much work.
22:02.56dijungalok
22:02.57dijungalwill do
22:03.15[TK]D-Fenderdijungal, for this test, I'd suggest "_X." but only for the test period.
22:03.46dijungalirie
22:04.22*** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net)
22:07.39rantshI know it translates codecs automatifcally,BUT for some reason (I don't really know or understand well enough) My company needs to translate calls from g729 to g711 on certain calls
22:10.40*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
22:14.05*** join/#asterisk johann8384 (n=johann83@gateway.myogre.com)
22:15.38[TK]D-Fenderrantsh, it will only happen as needed
22:16.19rantshso there's no way I can do this translation at will?
22:19.30[TK]D-Fenderrantsh, you'd have to have each side disagree on codec.
22:20.36[TK]D-Fenderrantsh, if you say that one side can only speak G729 and the other G711 then all calls from their respective sides will be in that codec.  if they end up being bridged together , * will automatically translate them
22:21.53rantsh(clueless)
22:22.59*** join/#asterisk Fulk (n=jon@cpc3-stap7-0-0-cust848.nott.cable.ntl.com)
22:26.30*** join/#asterisk mxmasster (n=Max@129.47.12.101)
22:26.32mxmassterhi all
22:27.11mxmassteron a g.711 codec average bandwidth used is ~80 kbps
22:27.23mxmassterso if both call legs are on  your system that's ~160 kbps
22:27.25lesouvageDoes anybody has good experience with integrating Gtalk in an Asterisk box, does it worth a try
22:27.50mxmassterso 1000 calls would be 160 Mbit of sustained data traffic?
22:28.09mxmassteris this an accurate number?
22:30.25[TK]D-Fendermxmasster, there are only 3 kinds of people..... those that get mat, and those that don't.
22:30.28[TK]D-Fendermath*
22:32.21Mercestes[TK]D-Fender, You spend most of your life in IRC....and you can't type.....
22:32.24Mercestes*tsk tsk
22:32.39kombijeez, i've tried everything now, just cant find why there is one way audio on inbound calls..
22:33.41kombiis it certain that you can't cause this behaviour just my messed up configs?
22:34.01Jinglesbest way to test that theory - put both phones on the same network segment.
22:34.33kombigood one Jingles!
22:34.38bkw__mxmasster, you can get 1200 calls on a 100mbit @ 40ms on ulaw
22:34.42bkw__but you don't have much wiggle room
22:34.58*** join/#asterisk ToyMan (n=Stuart@cpe-68-175-3-247.hvc.res.rr.com)
22:34.58kombilet me find another soft phone..
22:35.32*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-47-145.red.bezeqint.net)
22:38.00mxmassterhow does that math work
22:38.02mxmasster?
22:39.10Fulkping
22:39.26FulkI'm using a digium TDM card and I have to configure my * box to reboot daily to stop it locking up
22:39.29JTmxmasster: it works fine
22:39.30Fulksome kind of memory leak
22:39.33JTmxmasster: how does yours?
22:39.38mxmassterg.711 with 40ms and rtcp and 1200 channels shows 90.72 Mbit per second
22:39.43Fulkanyone else experienced that?
22:39.49JTmxmasster: you don't add up both directions
22:40.24mxmassterJT i'm not talking about both directions between UA and Server - i am talking about UA - Server - Gateway
22:40.33mxmassterleg 1 UA - server
22:40.38mxmassterleg 2 Server - Gateway
22:40.40JT<PROTECTED>
22:40.49*** join/#asterisk ber111 (i=brad@neu.cow.org)
22:40.52mxmassterright that was my question
22:41.02JTand you were adding up both directions
22:41.05JTyou don't do that...
22:41.06tzafrir_laptopFulk, memory leak? why won't you sample it?
22:41.16ber111if you execute a script via DeadAGI, how do you check if hte hcannel is hung up in that script so you can clean it up
22:41.40mxmassterJT you just agreed with me, and then you just told me that it is not the case
22:41.48Fulktzafrir, munin shows the memory usage grow and grow throughout the week
22:41.50tzafrir_laptopadd an hourly cron of ps aux or whatever
22:42.08JTmxmasster: i disagreed both times, you do NOT add both directions in bandwidth calculations
22:42.13JTthey're seperate calculations
22:42.18tzafrir_laptopFulk, usage of what?
22:42.26rantshbtw, going back to call recording, I just noticed that it doesn't work on uLaw calls, any clue on why, it works perfectly with the rest of my codecs
22:42.28Fulklemme check
22:42.30tzafrir_laptopthe total system memory? swap space?
22:42.49mxmassterJT okay so how do I properly make the calculation then
22:42.55Fulkbeen a while, the cron job "solved" the problem
22:42.57mxmassterIf I have one user make a g.711 call to a vendor
22:43.02mxmassterthrough our server
22:43.07Fulkbut I thought I'd take a punt in here - can't stand things not being "perfect"
22:43.14JTthat's approximately 85kbit/s
22:43.20mxmassterwhat is my total bandwidth usage
22:43.38JT85kbit/s each way, maybe a fraction under
22:43.48JTusing sip
22:44.23*** join/#asterisk Strom_M (n=strom@60-241-200-26.static.tpgi.com.au)
22:47.43Fulkhmm, appears to be cache - that would preclude an application leak
22:48.04Fulkperhaps the kernel doesn't play well with the PowerEdge server
23:02.11*** join/#asterisk CrashHD (n=timf@70.96.98.65)
23:02.26CrashHDhello
23:03.09CrashHDhow can I capture the sip response code once returned?
23:03.10jwhHi
23:03.24*** join/#asterisk SwK_ (n=SwK@user-69-73-37-99.knology.net)
23:03.25CrashHDI have a server sending a 603 when I've hit my call limit
23:03.44CrashHDwhich should then roll to the next carrier
23:04.00CrashHDbut because I have busy handling setup s-BUSY sends busy()
23:04.42CrashHDsip_header() looked good but I saw in the sip debug that the 603 code wasn't followin gany of the fields but rather was at the top of the message
23:05.00CrashHDany thoughts?
23:05.53*** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar)
23:07.14jwhhmm, not sure tbh
23:07.32Jinglesdon't you use 'IsChanAvail' to roll over?
23:08.08CrashHDI'm dialing multiple providers
23:08.12CrashHDnot end devices
23:08.18_DAWa
23:08.26CrashHDwell trying at least
23:09.00*** join/#asterisk perf3ktion (n=perf3kt@adsl-68-73-150-167.dsl.ipltin.ameritech.net)
23:10.10*** join/#asterisk xpander4 (n=gaston@adsl-074-169-108-059.sip.bct.bellsouth.net)
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23:12.14*** join/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net)
23:12.38Cybertoyhi .. I'm trying to register a new device but get "Device does not match ACL" error... anyone know what that means?
23:12.46*** join/#asterisk paolob (n=donpaolo@196.3.84.214)
23:12.51CybertoyI see that errror on the asterisk CLI ...
23:15.08*** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com)
23:15.32*** join/#asterisk CoffeeIV (i=rgr@rrcs-71-42-183-82.sw.biz.rr.com)
23:16.41*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
23:16.49CoffeeIVI have a voicetronix openpci4 card, and I am trying to use it with asterisk 1.2.  Voicetronix has drivers version 2.*, 3.*, and 4.0 -- is there a particular version of the voicetronix drivers that works with asterisk 1.2, or should they all work ?
23:18.29bcnlhas anyone here used ParkAndAnnounce
23:18.31bcnlmuch?
23:18.52bcnlI would like to dial multiple extensions to notify, it seems to crap out at the &
23:21.28CrashHDso nobody knows how I can parse the sip return code?
23:21.35CrashHDand use it in my dial plan?
23:22.01snuffy22there isn't a way i know of after dial to get the exact code no
23:22.22snuffy22generally if its a 500 error.. its 'congestion' returned in ${DIALSTATUS}
23:22.51CrashHDwell 603 is being returned
23:23.03CrashHDbut I don't want it to think busy signal
23:23.08*** join/#asterisk forsaken_ (i=forsaken@201.64.24.247)
23:23.12*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
23:24.09snuffy22well only thing i can think of is ask in asterisk-dev or on the bugs.digium page about it see if you get shut down so to speak
23:25.14JerJermoo
23:25.26_VoiceMeUp_COMjerjer
23:25.29_VoiceMeUp_COMcan i ask ou something ?
23:25.34JerJercan you?
23:26.11_VoiceMeUp_COM$tU
23:26.23_VoiceMeUp_COMin ser.. i need to regex if starts with "6666" then strip first 4
23:26.28_VoiceMeUp_COMyou good on that ?
23:27.59snuffy22mm.. i'd thought that wouldn't be too hard but i've got noidea on ser regex
23:38.51dijungalTKD-Fender: ok so it's still not working after i've add the dialplan
23:39.01dijungalasterisk just not picking up the call from the cisco
23:39.51[TK]D-Fenderdijungal, try "h323 debug" and see if you cn see the call coming in.  also pastebin your h323 & extensions configs
23:39.52[TK]D-Fender~pb
23:39.52jbotwell, pb is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org
23:52.20*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
23:52.57dijungalTKD-Fender: http://pastebin.ca/580122
23:58.19[TK]D-Fenderextension.conf ---------------- ?
23:58.42dijungaloooh yeh that's the extensions.conf config
23:58.45[TK]D-Fenderjust a typo on PASTING I hope, and I'm missing the CLI output & debug...
23:58.59dijungali got nothing on the CLI
23:59.17dijungali'm telling u, asterisk does not budge
23:59.27[TK]D-Fenderdijungal, go enable debug on H323.
23:59.34dijungaldid that
23:59.36dijungalsame thing
23:59.55[TK]D-Fendermaybe your Cisco in improperly configured then
23:59.56dijungali did "h.232 debug"

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