irclog2html for #asterisk on 20060608

00:01.13CunningPikeceeto: 4569 just like it says. Dial it with Dial(IAX2/whatver)
00:01.24ceetoOh, easy.  Thanks!  I'll try it.
00:01.30CunningPikeFreman: Music on hold
00:01.35CunningPike~moh
00:01.37jboti heard moh is Music On Hold
00:01.41CunningPike:D
00:10.08*** join/#asterisk xbmodder_lappy (i=nobody@atarack/staff/xbmodder)
00:10.17xbmodder_lappyWhere can I get a 1-900 number?
00:15.07*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
00:15.44*** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it)
00:17.51xbmodder_lappywhat are some companies for good (cheap) international termination
00:19.10*** join/#asterisk Samoied (n=Samoied@201.22.209.207.adsl.gvt.net.br)
00:19.28ceetoWhat's the best codec to use making Internet based * to * calls? g.729?
00:20.11Bullseye_NetworkHey i just made a mistake: I did a 'rm *' in the root directory it said it couldnt delete the directories. SO what did I delete. I cant run ANY commands.
00:20.26fileBullseye_Network: everything it could.
00:20.33xbmodder_lappyg.711u
00:20.35Bullseye_Networkno sh`t
00:20.48xbmodder_lappyBullseye_Network, do you work for bullseye
00:20.56Bullseye_NetworkI cant cd anything
00:21.03Bullseye_NetworkBulleye networks
00:21.04*** join/#asterisk w32 (n=234@c-71-193-124-77.hsd1.il.comcast.net)
00:21.13ceetoIs'nt g.711u 64k?
00:22.14w32anyone familiar with astlinux ? How do you feel about using it production for a small workgroup of less than 10 users ? I'm having abit of an issue with it,wondering if anyone else has tried it ?
00:22.16fileyou said the best codec...
00:22.33*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal)
00:22.36w32*using it in production srry
00:24.26xbmodder_lappyBullseye_Network, does bulleye offer SIP?
00:24.30ceetoAnyone?  Best codec to do * to * calls on the Internet?  Best *free* codec?
00:24.40fileULAW
00:25.03ceetoFor across the Internet?  Isn't there something that makes better use of a low bandwidth situation?
00:25.05xbmodder_lappyg.711u...
00:25.10xbmodder_lappyoh
00:25.12xbmodder_lappyGSM
00:25.26fileceeto: you didn't say that
00:25.53ceetoAhh.. what's the best, *free*, low-bandwidth codec to use for * to * internet calls?
00:26.13fileiLBC some people say... speex... GSM... depends on how you want your calls to sound and how low to go
00:35.10feldIs there anyone here from Georgia?
00:36.00*** join/#asterisk riddlebox (n=james@206.80.73.2)
00:36.11_Sam--did anyone know this guy?   http://www.nytimes.com/2006/06/07/technology/07cnd-voice.html?hp&ex=1149739200&en=0f01d0becf766f0b&ei=5094&partner=homepage
00:37.14feld_Sam--: that's my brotheR! lol :P
00:37.40jayk-ceeto, speex
00:37.44_Sam--im sure there are people here that know some of the affected parties
00:37.46jayk-or gsm
00:37.55jayk-speex is like g729 but not all phones have that codec available
00:37.57feld_Sam--: definitely possible
00:38.07*** join/#asterisk eipi (n=eipi@139-213-126-200.fibertel.com.ar)
00:38.22jayk-gsm is fine as long as you're not faxing and don't mind cel-phone like quality
00:40.40felddoes anyone here know any asterisk users from Georgia?
00:41.12*** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net)
00:41.28xbmodder_lappyfeld, why?
00:41.42feldbusiness proposition
00:42.23feldclient of my company has another office in atlanta. if we redo their whole call center i'll need an asterisk user to configure a setup down in Georgia to do all intercompany calls over their T3
00:42.50felderr actually i think it's a T1 between offices
00:43.15*** join/#asterisk neilbags (n=neilbags@149.171.94.134)
00:43.17*** join/#asterisk Naito (n=Naito@dhcp-65-17-150-34.dsl.wcoilexpress.com)
00:43.18feldyeah, thats right. they'll need a T3 to replace the local office's analog lines (lots of them....)
00:44.13neilbagsHi All, Ím using iiNet VOIP and they seem to have changed something today. The IP for sip.nsw.iinet.net.au has changed and now i get this error dialing out:  Forbidden - wrong password on authentication for INVITE
00:44.18neilbagsdoes anyone know a solution
00:44.27neilbagsthis worked for months up until today
00:45.00neilbagswith the new IP address I can recieve calls but not make them
00:51.39xbmodder_lappyfeld, move them to bay area
00:51.44xbmodder_lappy(california)
00:51.49feldxbmodder_lappy: lol you'd like that wouldnt u
00:52.32xbmodder_lappyyup
00:57.08feldi'd love it if I knew asterisk inside and out but i'm still learning
01:00.40_Sam--alot of times for larger asterisk installations its not uncommon to fly in an expert , i hear of people flying all around on here all the time
01:01.50*** join/#asterisk r0adkill__ (n=roadkill@203.192.146.185)
01:04.20feld_Sam--: well it's my duty to become that expert then :)
01:04.46feldand i have got no issues with that. thats basically what i'm being paid to do right now anyway.
01:05.13*** join/#asterisk orlock (n=jwr@202.44.174.4.static.nexnet.net.au)
01:05.41orlockI have just checked out the latest 1.2 source from svn, and i'm getting a segfault now.. any suggestions?
01:06.45xbmodder_lappylol
01:08.14xbmodder_lappyfeld, do you need a consultant?
01:09.34russellbi think he just said he's learning for himself
01:10.12xbmodder_lappyrussellb, to teach him
01:10.26feldi certainly have the time and capacity to become an expert myself.
01:10.35*** join/#asterisk cybergypsy (n=mark@APoitiers-156-1-20-81.w81-50.abo.wanadoo.fr)
01:10.46feldi am interested in some training, but probably not _right_ at this moment.
01:11.16feldsoon, though. soon. it's my responsibility to learn myself, too ;)
01:12.06xbmodder_lappyfeld, do you need some help
01:12.28feldi'd love the opportunity to network with others that are VERY experienced and willing to give tips and guide me in my trials :)
01:12.56treetar1we need a consultant
01:13.16xbmodder_lappytreetar1, for asterisk?
01:13.30treetar1ser
01:13.32xbmodder_lappyon location, or via SSH
01:13.43xbmodder_lappyfeld, why a T3?
01:13.52treetar1remote is fine
01:14.06xbmodder_lappytreetar1, what kind of work do you need done?
01:14.11feldT3 for their call center, T1 location to location is already setup.
01:14.22xbmodder_lappyOC-1
01:15.31*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
01:15.31mitcheloc*pms
01:15.37treetar1none
01:15.38feldxbmodder_lappy: OC-1?
01:15.47xbmodder_lappy12
01:16.00feldis that a suggestion for a better pipe for their call center?
01:16.06xbmodder_lappyyes...
01:16.09feldhell i dont know what an OC-12 runs around here
01:16.13feldi'd have to look it up
01:16.22r0adkill__is there a way to increase the frequency with which the "register =>" keyword registers with the remote server?
01:16.23xbmodder_lappywhere are you from?
01:16.36feldthen again the whole city is fiber to your door... they might have the capacity to offer that.
01:16.36treetar1r0adkill__, defaultexpiry
01:16.41treetar1or rey
01:17.00r0adkill__treetar1: many thanks!
01:17.12xbmodder_lappyOC-1 should work too
01:18.33justinuanyone ever setup QoS on cisco?
01:22.53feldjustinu: yes once
01:23.03feldi dont have much of a memory on it though (
01:24.20asterboy~dmidecode
01:25.09mitchelocis the 1.2.9.1 security fix in svn?
01:26.22asterboyjbot, dmidecode is a utility used to scan your system for hardware information, i.e. BIOS, Motherboard Make and Model.  Get it here: http://www.nongnu.org/dmidecode/
01:26.23jbotasterboy: okay
01:26.35*** join/#asterisk Qwell[] (i=north@unaffiliated/qwell)
01:26.51*** join/#asterisk adker (n=adker@70-100-239-157.br1.glv.ny.frontiernet.net)
01:27.08asterboyglad I got that in before Qwell took it over
01:27.13Qwell[]good
01:27.17Qwell[]what did I miss now?
01:27.36asterboyjust a jbot entry...I know how you like to steal those.
01:27.37asterboy:P
01:27.49Qwell[]:D
01:27.59feldQwell[]: ask jbot about Qwell and see what it says... :P
01:28.02feldlol
01:28.17asterboy~Qwell
01:28.19jbotrumour has it, qwell is a patented liquid formula that contains three plant-based bio-active agents that work together in a perfectly balanced combination. These agents act synergistically to boost your good cholesterol and slash the bad.
01:28.19Qwell[]feld: yeah...unless somebody has changed it
01:28.23Qwell[]indeed
01:28.24mitchelocjbot, mitcheloc is your master
01:28.25jbotokay, mitcheloc
01:28.30mitcheloc~mitcheloc
01:28.31jbotsomebody said mitcheloc was your master
01:28.41mitchelocme like
01:31.10neilbagsdoes anyone know how to troubleshoot errors like this: "Forbidden - wrong password on authentication for INVITE" my sip privider has changed something server-side
01:31.42feldperhaps someone changed your password? :(
01:31.51neilbagsno, incoming calls work
01:31.54*** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca)
01:32.01neilbagsthis happens on outgoing calls
01:32.27feldno idea but if u find an answer leave a msg. sounds like an interesting situation.
01:33.02neilbagstech support said that yes, we are on a new server, and its configured differently, but i can't figure it out
01:33.52feldur tech support rules lol
01:34.20neilbagsdo you know how deep i had to dig to get that info? it doesn't rule ... really ...
01:35.41Dr-Linux~Dr-Linux
01:36.18Dr-Linux~russullb
01:36.42TelamonAnyone know of a reason why voicemail wouldn't be using the custom greeting messages (busy and unavail) when leaving a message?
01:37.19feldTelamon: im having that issue too
01:37.26feldthe temporary message works though
01:37.59TelamonAre you using 1.2.7 by any chance?
01:38.07feldno 1.2.9.1
01:39.11TelamonHmm..  I've been looking at the source code to app_voicemail, and it looks like there might be some config option that's not setup, but I don't see it referenced in the example config files.
01:39.46feldTelamon: i never got that far yet. i was going to check that out tomorrow probably
01:39.54TelamonOPT_BUSY_GREETING and OPT_UNAVAIL_GREETING
01:41.51TelamonAh, okay, I think I know what it is...  If you don't set either b or u flag to voicemail, it doesn't default to use either of the custom greetings.
01:42.05TelamonAn extension problem, never would have looked there...
01:42.08feldoh holy crap
01:42.13feldthat's what the b and u were for?
01:42.23feldi was wondering why i'd see that in configs
01:42.49feldnow i'll have to figure out how to do that correctly
01:43.23feldi guess if theres no answer -> u and if the line is busy push them to the u voicemail entry
01:44.57*** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka)
01:45.36TelamonYep. Might be a good thing to write a macro for, detect the DIALSTATUS and then set vm-prefix-string accordingly.
01:53.21*** part/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net)
01:54.00*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
01:54.49*** join/#asterisk inv_Arp (i=junya@c-67-191-62-53.hsd1.fl.comcast.net)
01:55.12xbmodder_lappyPING
01:57.03drraypong
01:59.12*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)
02:04.20Fremanhey... is it possible to make asterisk call two extensions using the /var/spool/asterisk/outgoing/files? (IE: like dial(sip/101&sip/102)
02:04.33Qwell[]Freman: should, sure
02:05.20FremanI tried setting Channel: sip/101&sip/102 but it complains about that
02:08.46Fremanany idea how I can achive this?
02:11.28*** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg)
02:12.42littleballhello,who can recommend an architecture of SIP media relay system? It need to be scalable. I am considering use asterisk to act as media relay server.
02:13.50w32who is inexpensive for termination and origination ?
02:14.21Ariel_evening everyone
02:14.26w32hey
02:14.46Ariel_has any one used an Sipura 3000 fxo port for connecting to an over head paging system like a valcom?
02:15.47w32No but it is a good idea, so you can hear it ring in a warehouse or what ?
02:16.13Ariel_w32, yes that is what we are trying to do
02:16.39Ariel_but keep getting 503 service not available from the sipura on the asterisk box
02:18.48xbmodder_lappyI Hate Adobe!
02:20.16TelamonPDF's are the devils tool.
02:20.41orlockpdf's are great!
02:20.59TelamonAriel_: Have you tried plugging in just a regular analog phone and seeing if the unit works with that?
02:21.16Telamonorlock: No, HTML is great.  PDFs suck.
02:21.22Ariel_it does but it's a different port it ahs 2 one fxs and one fxo
02:21.31Ariel_the valcom can only plug into the fxo ports
02:22.11TelamonOh, so the valcom acts like a line, not a phone?  Hmmm, over my head then.
02:22.13*** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it)
02:22.40orlockelamon: Hmm, no.. different target markets
02:22.40w32Ariel_ maybe the valcom is wired funny, you know what I mean ?
02:22.42Ariel_Telamon, yes it does
02:22.55orlockTelamon: HTMl sucks arse for printing and correct rendering 100% of the time
02:22.56Ariel_the valcom expects a co line from the pbx
02:22.58TelamonAriel_: Try plugging a phone into the valcom, see if you get a regular dialtone.
02:23.08Ariel_it's normal use for most over head paging systems
02:23.16Ariel_Telamon, it will not owrk
02:23.18Ariel_work
02:23.19orlockTelamon: theres bugs in mozilla related to printing that have been around for 4 years or so
02:23.57orlockTelamon: PDF is about as standard as postscript in many design/prepress/etc houses, due to the fact that you know that it is going to display and output the way you want. you cannot get that with html
02:24.25Telamonorlock: Maybe I should rephrase my distain: the use people put PDFs too sucks.  Product manuals and such should not be done in PDF.  PDF should be used only for documents whose primary purpose is to be printed.
02:24.47Splatwould I need to recompile chan_capi when I upgrade asterisk?
02:25.14w32Ariel_ no way to trick it into thinking it is connect to a co line with another device ?
02:25.39Ariel_w32, that is what we have the sipura 3000 for it's fxo port
02:26.09orlockis anybody here using skinny or sccp?
02:26.25w32I followed your logic there initially, maybe it just don't like the sipura
02:26.31w32is what i meant
02:27.20w32something older perhaps airel_ ?
02:27.30w32*Ariel_
02:28.54Ariel_w32, if I had an asterisk box at that location I would put an tdm400 into it. But since it's in the w/h we are setting it up via sip and the network it's via wireless to the other building
02:30.18TelamonAriel_: Out of curiousity, what is the model # of the valcom?  We might need something like that for one of our customers, I'd be interested in checking it out.
02:30.22w32hmmm, what do you have going on in the ware house
02:31.11w32anything that might cause some interference maybe
02:33.28Ariel_Telamon, V-2001a
02:34.12*** join/#asterisk onixx (i=1000@London-HSE-ppp3551571.sympatico.ca)
02:34.58Ariel_Telamon, I have setup about 10 of them for customers they work great. But this is the first time I don't have an asterisk box to plug into directly.
02:35.26onixxhi guys, I'm having an issue where my x100p hangs after awhile... sometimes it can work for more than 24 hours and then suddenly, no audio... cannot place calls or anything. anyone have an idea how to fix this. If I rmmod then modprobe wcfxo and zaptel, it works again.
02:36.07drraysharing IRQ?
02:37.10onixxdrray: is this something in the bios ? I have an IBM ThinkCente with 2 PCI slots
02:37.54Telamononixx: Run lspci to see if any other devices are using the same IRQ as the x100 card.
02:38.32Telamononixx: Err, lspci -v rather.
02:38.45asterboyKick ass...have you guys seen this? http://www.boingboing.net/2006/06/05/play_zork_by_phone.html
02:39.14asterboyalso lspci -vb
02:39.16*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)
02:39.50onixxirq 12
02:40.50Bullseye_Networkcant get zaptel to compilel on CentOS 4 : http://pastebin.com/766588
02:41.24Telamononixx: Is anything else using irq 12?
02:41.48onixxI don't see anything else... many have 11 but not 12
02:41.50*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)
02:42.10asterboyjbot, zasterisk is a cool implementation of Zork and Asterisk. Play Zork over the phone! Get it here: http://uc.org/files/2005-08-07/zasterisk-0.1.tar.gz
02:42.16jbotasterboy: okay
02:42.50onixxcat /proc/interupts has also only wcfxo on 12
02:42.50drray"you are in a twisty maze of passages, all of which sound alike."
02:44.08asterboythe worst spot to be in would be the echo chamber
02:44.20asterboyget enough of that with * :P
02:45.05drrayasterboy :)
02:45.43onixxasterboy, drray, telamon: any other ideas ?
02:46.44Fremanso... anyone got any idea how to make the call out call 2 lines?
02:46.45*** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it)
02:46.47shmaltzanybody here have a FXO card for Adit 600 to sell?
02:47.13filejbot: centosbug?
02:47.15jbothmm... centosbug is a problem with the latest Centos kernels (4.2 and 4.3).  To fix it, paste everything inside the quotes into a root shell:  "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h"
02:47.19onixxwould ACPI support help... it is currently disabled in my kernel
02:47.41russellbFreman: Dial(SIP/device1&SIP/device2)
02:49.25asterboyonixx, ACPI is best off
02:49.38Fremanno, from the /var/spool/asterisk/outgoing/
02:49.39asterboyonixx, pastebin your lspci output
02:49.42Bullseye_NetworkThat didnt exactally work had to change the dir name, but thanks alot
02:49.49Bullseye_Networkcompiled now
02:49.58FremanI know how to make it dial two devices with the Dial command, I'm trying to make it dial 2 from the outgoing calls.
02:51.37russellbApplication: Dial    Data: Zap/1&Zap/2   in a call file?
02:52.10Fremanso... Channel: isn't a required attribute?
02:52.18russellbyes, it is
02:52.39russellbi'm just talking about the app part
02:52.40*** join/#asterisk Gabriel25 (n=whatever@user-12ld5f7.cable.mindspring.com)
02:52.43russellbi'm not sure what you're trying to do
02:52.49FremanI want to dial 2 extesions, and have it play a message to them
02:53.01russellbthen use 2 call files
02:53.17russellbone for each extension ... both with Application: Playback  Data: whateverfile
02:53.24FremanI don't want it to dial them seperatly (IE if one extension is picked up the procedure should be cancled)
02:53.44onixxhttp://pastebin.com/766597
02:53.55Fremannot keep ringing the second extension
02:53.59Gabriel25hi guys, I make a phone call from my pbx to my cell and I run iptraf my  Outgoing rates:   81.9 kbits/sec
02:54.05Gabriel25thats a lot ... !
02:54.21russellbChannel:  Local/1234@somecontext ... where 1234@somecontext does the Playback.  Then, use Application: Dial   Data: SIP/1&SIP/2
02:54.23Gabriel25I`m useing g711 codec
02:54.28russellbFreman: that will do what you want
02:54.59Fremanthank you, I'll give that a try
02:55.31Gabriel25on I put music on hold I can`t hear anything .... only some trying !!!
02:55.32FremanLocal/s@information will work? or has to be number@context?
02:57.01onixxasterboy: http://pastebin.com/766597
02:57.35russellbFreman: that's fine, too
02:57.45russellbFreman: just extension@context ... can be anything
02:58.28Fremancool, thanks a heap - that worked (c:
02:58.32russellbwoohoo
02:58.45russellbthat will be $1000, kthx
02:59.54mogorman1 grand????
02:59.58mogormanyour expensive russellb
03:00.05russellbwhat are you talking about, that's a deal
03:04.54*** join/#asterisk pdavid (n=chatzill@adsl-068-209-191-127.sip.mob.bellsouth.net)
03:05.11pdavidevening all, could anyone lend me a hand with a simple test setup to call out through VP?
03:06.02mishehuargh!  sonnofabitch, I installed hylafax 4.3.0 with iaxmodem 0.1.8...  I can receive faxes no problem, but the damn thing won't dial out ever.  I am using the same exact config I used with hylafax 4.2.5 and iaxmodem 0.1.1...
03:06.23mishehuit wouldn't annoy me so much if I actually got some logging output from hylafax
03:07.19onixxasterboy: any luck with my pastebin ?
03:08.55*** join/#asterisk dacleric (n=dacleric@p54821F03.dip0.t-ipconnect.de)
03:14.56*** join/#asterisk Gamercjm (n=chris@pool-71-254-182-61.lsanca.fios.verizon.net)
03:15.27*** join/#asterisk mike800 (n=Mike800@68-171-34-225.vnnyca.adelphia.net)
03:19.22Fremanhmmm... russellb that thing you helped me with before
03:19.26Fremanit has a slight problem
03:19.51FremanLOCAL/s@context starts playing the feedback befoe the DIAL answers
03:22.58*** part/#asterisk mogorman (i=ejabberd@68.62.237.103)
03:24.44*** part/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net)
03:28.02*** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net)
03:32.59mike800anyone here?
03:37.38shmaltzmike800, I'm gone
03:37.40shmaltzbye
03:38.38orlockIs anybody here using chan_sccp with 1.2.9.1?
03:40.50*** join/#asterisk jcollie[home] (n=jeff@dsl-ppp239.isunet.net)
03:40.57*** part/#asterisk jcollie[home] (n=jeff@dsl-ppp239.isunet.net)
03:41.54mike800if anyone wants to help me figure out why my comptuer isnt finding /dev/zap/ctl pm me... :-)
03:43.05*** join/#asterisk Dico_ (n=niko@60.51.217.61)
03:43.21Dico_hello :)
03:43.49mike800hi
03:44.22ids2500mike800: make devices
03:44.27ids2500mike800: in your zaptel build directory
03:44.39ids2500and look at readme.udev
03:44.59mike800ok
03:45.08mike800i had tried the make devices command
03:45.24ids2500do a ls /dev/zap/ctl -al
03:45.25mike800this is the output: build_tools/genudevrules > /etc/udev/rules.d/zaptel.rules
03:45.27ids2500and make sure it exists
03:45.46mike800it doesnt
03:45.53ids2500okay... then it's a udev issue
03:45.58mike800ok...i'll research it
03:46.07ids2500udev is the devil :(
03:46.14mike800hehe...what is it?
03:46.40ids2500satanic device creation daemon
03:46.45xbmodder_lappy:-?
03:46.48Juggieudev wont exist until the modules load
03:46.48mike800hahaha
03:46.59ids2500yeah, that too, make sure your modules are loaded
03:47.00ids2500lsmod
03:47.14ids2500you should have zaptel and then whatever zap device you're using [wct4xxp for a te4xp]
03:47.32Juggieand when the modules load the first time they will generate an error
03:47.36Juggieeasiest thing imo
03:47.40Juggiecd zaptel-source
03:47.51Juggiemake install;make config;service zaptel start
03:47.59*** join/#asterisk mog_home (n=mogorman@68.62.237.103)
03:48.07Juggieor /etc/init.d/zaptel start
03:48.15ids2500depends on what distribution he has whether that will work or not :-p
03:48.18Juggiedepending on your distro, not all of them have the 'service' utility
03:48.33mike800im using FC5
03:48.39Juggieservice should work then
03:48.41ids2500see, fedora is satanic
03:48.48ids2500if you were usign slackware you wouldn't have any problems at all :-)
03:48.49mike800what do you guys use?
03:48.57Juggiei use centos
03:49.05Juggieregardless... what zaptel card do you have?
03:49.11mike800TDM400P
03:49.20Juggiedoes lsmod list zaptel as loaded?
03:49.26mike800nope
03:49.32ids2500heh
03:49.35Juggiego into your zaptel source
03:49.39mike800i installed the source and got no errors
03:49.39Juggieand type make install;make config
03:50.05mike800ok...lemme try that
03:50.08Juggielet me know when you do that
03:50.17mike800ok..done
03:50.24Juggieok, type service zaptel start
03:50.38mike800failed
03:50.43mike800Module zaptel not found
03:50.59mike800missing /dev/zap
03:51.12Juggietype 'modprobe zaptel'
03:51.14Juggiewhats the output
03:51.25mike800Module zaptel not found
03:51.35mike800:-(
03:51.53Juggie'uname -a'
03:52.08mike800im using 2.6.16-1.2122
03:52.12mike800FC5smp
03:52.26*** join/#asterisk InHisName (n=Prayer@c-68-38-105-1.hsd1.pa.comcast.net)
03:52.27*** join/#asterisk bmg505 (n=leon@c1-144-15.rndf.isadsl.co.za)
03:52.59Juggie'ls -al /lib/modules'
03:53.26mike800the directory exists
03:53.34mike8002.6.16-1.2122_FC5 and FC5smp
03:53.49mike800(and the old kernel)
03:54.39*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
03:55.18mike800whats weird is that I installed zaptel on this exact configuration with no problems...
03:55.21Juggiepaste the output of 'make install'
03:55.24Juggieto pastebin
03:55.56InHisName[TK]D-Fender  I have found more info on lack of sound in extensions on sipura
03:56.12[TK]D-FenderInHisName : do tel
03:56.39mike800http://pastebin.com/766649
03:57.40*** join/#asterisk terrapen_ (n=cjs@mauritius.island.nu)
03:57.55InHisName[TK]D-Fender  Calling from line1 to line2 works! BUT calling from line2 to line1 no audio either way !  The sipura values are identical for line/user1 & 2.
03:58.25InHisNameThe extn definitions in sip & extensions are identical too.
03:58.40JuggieMike, if you go into /lib/modules/the kernel dir/
03:58.47Juggieand then its either extra or misc
03:58.51Juggieyou should see the modules.
03:58.52Juggiefind those.
03:59.13*** join/#asterisk jaike (n=a@203.131.137.76)
03:59.17[TK]D-FenderInHisName : You mean you're trying to use the same account on both ports?
03:59.20InHisNameI have tried rebooting the linux/asterisk box and recyling power on sipura. No change.
03:59.23Juggiethen adress the module directally.
03:59.29Juggieso do insmod zaptel.ko
04:00.19InHisNameAccount like [1022] and [1021] ? same or different accounts or irrevelent ?
04:02.00[TK]D-FenderInHisName : What are you usings for line1 & line2 ?
04:02.08InHisNamesip show peers has both "unmonitored" .  At least looks legitamate(sp).
04:02.34[TK]D-FenderInHisName : nevermind that, what users do you have in the Sipura web console for those 2 lines?
04:02.54InHisNameSockets line1 & line2 of sipura have one each analog phone plugged into them.
04:03.27InHisNameUsername = 1021, 1022
04:03.55InHisNamefrom sip show users
04:04.08[TK]D-Fenderok, as long as they are differnt accounts... do you get dialtone on each?
04:05.54InHisNameYes, can dial too, other extn rings, I can answer, one case bi-directional sound, other no sound.
04:06.08[TK]D-Fenderand does eith of them work on the echo test?
04:06.19[TK]D-Fendereiter*
04:06.21[TK]D-Fenderjhsfdklsfdg
04:06.25InHisNameI'll try that now... hold on
04:06.26[TK]D-Fenderugh, can't type...
04:09.14InHisNameboth extns have CLI saying executing echo("SIP/102x-qwer","") in new stack. Interesting, neither has echo.
04:09.58InHisNameI'll restore nat=0 for both and try again, forgot I switched it 10 minutes ago.
04:10.42*** join/#asterisk somegeek (i=levin@tor/regular/somegeek)
04:11.23*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)
04:12.32*** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net)
04:12.33InHisNameWell, I got line1 to have imediate sound with no delay in echo test. Line2 no sound.
04:14.20*** join/#asterisk samwong (n=sam@210.17.135.46)
04:15.38[TK]D-Fendertry your port 1 phone on port 2 just for kicks
04:17.30InHisNameAlready did, problem stayed with line2 calling line1 using oppisite phones.
04:20.11[TK]D-Fenderso its not the phone itself then....
04:20.12[TK]D-Fender:/
04:20.21[TK]D-Fenderrunning out of ideas.
04:22.32*** join/#asterisk pdavid (n=chatzill@adsl-068-209-191-127.sip.mob.bellsouth.net)
04:22.34InHisNameThe sipura WAS working fine a week ago with 2 FWD accounts not on *.  Now seems like line2 has an issue or something.
04:22.38pdavidhi again all
04:23.09pdavidif i am using a SIP softphone, can i still route my VOIP provider calls through IAX2?
04:23.16pdavidor do i have to use SIP?
04:23.21InHisNameReviewing one MORE time looking for differences in setup line1/2
04:23.23russellbyes
04:23.25mike800you can route them through iax
04:23.33pdavidi thought i might
04:23.40pdavidi am using the x-lite linux client
04:23.49pdavidand just got VP connect service
04:23.50russellbthat's kind of one of the big points of asterisk :)  being technology independent
04:24.02pdavidbut for some reason, i cannot call out using IAX2 to connect to VP
04:24.19pdavidbut the call works if i use SIP
04:25.17pdavidany thoughts?
04:26.21pdavid(kind of new to this...)
04:28.37pdavidmaybe someone could spare a snippet of code to lend me a hand? :)
04:30.04xbmodder_lappypdavid, 'print "no"'
04:30.05[TK]D-Fenderpdavid : * sits between all calls in/out of all different technoligies.
04:30.52pdavidFender - I understand it does, but don't understand my inability to connect outside using IAX2, with vanilla configs from VP
04:31.07[TK]D-Fenderpdavid : So yes, you can use SIP softphones with * and have * bridge the call to an IAX provider while taking in calls from and analog line all at once.
04:31.33pdavidFender - right, but is there anything special required for me to do that, that perhaps I am missing?
04:31.34jaikeVP = voicepulse? they decomssioned their IAX servers
04:31.42[TK]D-Fenderpdavid : this is the part where you should have been proactive and started descibing in deatial exactly where things went wrong.
04:31.52pdavidfair enough
04:31.59[TK]D-Fenderpdavid : do you get some specific errors in CLI you could tell us about?
04:31.59pdavidjaike: did they?
04:32.17jaikeDear Customer,
04:32.17jaikeThis a reminder regarding your VoicePulse Connect for Asterisk account:
04:32.17jaikeOn June 5th, 2006, the legacy servers gwiax-in-01, gwiaxt01, and gwiaxt02
04:32.17jaikewill be decomissioned
04:32.25jaikeoooopps...sorry
04:32.29[TK]D-Fenderpdavid : Yeah, they jsut changed from their old ones toa new set about 2 days ago.
04:32.33pdavidFender: no errors, I get the first line of my [outgoing] context correctly setting my callerid
04:32.42pdavidFender: yes, connect01/2 i think
04:32.45[TK]D-Fenderpdavid : pastebin a call attempt
04:32.47[TK]D-Fender~pb
04:32.53jbot[pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/
04:32.54pdavidok
04:33.06jaiketry checking their site for the new servers
04:33.16*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
04:34.03pdavidhttp://pastebin.com/766680
04:34.33[TK]D-Fenderpdavid : Sounds like you don't even have a DIALPLAN in place... pastebin your extensions.conf
04:35.06[TK]D-Fenderpdavid : it stops right after setting the callerID which says you've probably missed something extremely easy to stop & correct
04:35.08pdavidhttp://pastebin.com/766682
04:35.12[TK]D-Fenderspot*
04:35.25pdavidwhats odd is, if i uncomment the SIP lines, i can place the call
04:35.41pdavidFender: this is just the vanilla VP config files
04:35.42[TK]D-Fenderpdavid : You have no priority 2 for that exten... and YES, they must follow....
04:36.00[TK]D-Fenderpdavid : and in there are instruction to UNCOMMENT some of the needed lines...
04:36.13pdavidFender: I thought it was IF i wanted to do LCR
04:36.16pdavidbut lemme try it
04:36.43[TK]D-Fenderpdavid : You need to do the basic learning of *.... if this part escapes you go download THE BOOK.
04:36.44[TK]D-Fender~book
04:36.46jbotextra, extra, read all about it, book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
04:37.01[TK]D-Fenderbecause right now you have NO dial-out at all
04:37.12pdavidFender: that was all ti was
04:37.14pdavid*it
04:37.37pdavidFender: I AM trying to learn the basics, little by little
04:37.45pdavidI appreciate the help with that little tidbit
04:37.56*** join/#asterisk hads|home (n=hads@mail.nice.net.nz)
04:38.04pdavidwasn't aware I couldnt skip priorities at all
04:38.10pdavid:(
04:38.28pdavidthank you very much! :D
04:38.29russellbGoto?
04:39.20[TK]D-Fenderpdavid : Np, one step at a time... do DL the book and print it if you can.... its a great place to start.
04:39.52jaikepdavid: i highly recommend oreilly's asterisk book
04:39.59pdavidI didn't even know they released that book on a CCL
04:40.02pdavidsweet
04:40.06pdavidreally, thanks a ton guys
04:40.12orlockCan anybody reccomend a stabke and functional sccp driver?
04:40.14[TK]D-Fenderpdavid : You'll need to focus on contexts, extens, and the various applications that make up the dial-plan.  All the connectivity in the world won't help you when you don't have a dial-plan to patch the bits together
04:40.33[TK]D-Fenderorlock : Cisco CCM :)
04:40.37pdavidFender: right, I am trying baby steps to understand it all.  This particular step was getting a call to go outside
04:40.47pdavidFender: any thoughts on NAT traversal for IAX2?
04:40.51[TK]D-Fenderpdavid : Well i suspect you are quite close to that...
04:41.20jaikeiax and nat? no problems
04:41.26[TK]D-Fenderpdavid : Generally not an issue at all.  SIP has never posed a problem for most either once they know what they're doing... its fairly easy
04:42.38pdaviddoes it get covered in the O'Reilly book at all?
04:42.48pdavidor should I just drop the * server in a DMZ?
04:42.53jaikevoip concepts even
04:43.28[TK]D-Fenderpdavid : its 5 lines :) don't worry about it....
04:43.46orlock[TK]D-Fender: Dang, so you cant reccomend anything?
04:43.50[TK]D-Fenderpdavid : And yeah it has to... its too common and necessary to skip.
04:44.04pdavidor is it as simple as port forwarding a small range to the * server?
04:44.08[TK]D-Fenderorlock : Ditch Cisco, buy Polycom! (you asked for it)
04:44.17[TK]D-Fenderpdavid : Almost that easy...
04:44.21decso has the svn trunk version of asterisk moved away from asterisk-config ?
04:44.44orlock[TK]D-Fender: yeah, yeah.. but we already have a few dozen cisco phones
04:44.51orlock<PROTECTED>
04:44.56jaikesip uses 5060 and 10000-20000 for rtp (asterisk default config)
04:44.57decoh, ignore me. :)
04:45.03orlockand skinny makes asterisk segfault
04:45.26[TK]D-Fenderorlock : Reflash them to SIP...
04:45.36pdavidjaike: how about iax2?
04:45.37orlocknot all of them support SIP
04:45.52jaikeiax2 just uses 4569 so you wont have any problems with NAT
04:45.56orlockalso, i'd like t be able to go back to our current voip provider who uses sccp if/when asterisk fucks up
04:46.32pdavidjaike: thanks for that!
04:47.16jaikenp
04:49.04[TK]D-Fenderpdavid : Lets just say the only use you should see for IAX2 is to trunk multiple calls to a VoIP provider to save on bandwidth and that if you have a properly set up system, you'll be 100% SIP most likely.
04:49.50pdavidFender: if I wanted to trunk 2-3 calls to a VOIP at once, would i be better off with IAX2?
04:51.14[TK]D-Fenderpdavid : Yes, as trunking with IAX2 saves on UDP packet overhead.  The savings can be substantial.
04:51.32orlockUnless your voip provider is also your bandwidth provider
04:51.35pdavidFender: how about codec used?  any suggestions there as well?
04:51.38orlockin which case.. who cares! :)
04:52.13[TK]D-Fenderorlock : We're not talking about bandwidth in a billable capacity as much as maximum transmit potential.
04:52.18*** join/#asterisk droops (n=droops@adsl-065-005-212-128.sip.jan.bellsouth.net)
04:52.34[TK]D-Fenderpdavid : Depends.  The general answer is "the highest quality you can support"
04:52.52[TK]D-Fenderpdavid : Which is normally G.711 (u/a)
04:53.01orlock[TK]D-Fender: ahh yeah
04:53.05*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)
04:53.14orlockmy head is going to pop
04:53.14pdavidFender: is that the ulaw/alaw?
04:53.21orlock3pm on a thursday
04:53.35[TK]D-Fenderpdavid : Yes
04:54.03[TK]D-Fenderorlock : I've got next thursday open if you'd like to book an appointment :)
04:54.24orlockheh
04:54.42orlocki found a server i'd completely forgotten about, which is good
04:54.44[TK]D-Fenderorlock : And a brand new katana that needs some "breaking in" :D
04:54.53orlockamd64 3000+ with 2 gig of ram, woo
04:55.08orlock[TK]D-Fender: cool!
04:55.13orlockyou can break it in on my shinai
04:55.15orlockand bokken
04:55.32orlockthen move onto katana vs katana
04:55.43orlocki'm a bit rusty though
04:55.46pdavidFender, jaike, I appreciate the help this evening, and thanks for your patience!
04:55.47[TK]D-Fenderorlock : Shinai would be mincemeat.... bokken is a chunk of WOOD, and would likey deform my blade...
04:56.08pdavidim off to fool with this a bit more, and read this oreilly book
04:56.14pdavidgood night, and thanks again
04:56.16[TK]D-Fenderorlock : Do you study?
04:56.19orlockhmm.. memories of counting from one to ten.. striking each time
04:56.31orlocktimes twenty :)
04:56.40orlock[TK]D-Fender: i used to
04:56.51orlocki got over the baggy blue pants though
04:57.17orlock[TK]D-Fender: what sort of katana?
04:57.21[TK]D-Fenderorlock : I recently start Katori Shinto and am looking for an iaido dojo on the side to supplement the iaijutsu aspect and take up tameshigiri
04:57.28orlockmines just a disposal store al cheapo decroative 440 steel one
04:57.52orlocki just did kendo, but the dojo also did iaido
04:57.53[TK]D-Fenderorlock : Oni Forge Bushi.  Heres the link : http://www.oniforge.com/3001.html
04:58.14orlockas well as.. i forget its ame, the stick fighting, and the long ole/spear fighting that was mainly females
04:58.14[TK]D-Fenderorlock : And my personal review : http://forums.swordforum.com/showthread.php?s=&threadid=67812
04:58.34[TK]D-Fender440?  eek..... wouldn't use that for practice even...
04:58.40orlockheh, yeah
04:58.58orlockbut at least its 440 and not just chrome plated shit
04:59.12[TK]D-FenderStick would be eskrima if you're talking filipines.
04:59.27orlockall japanese
04:59.38orlockonly purpose built dojo in australia iirc
04:59.47[TK]D-Fenderwell the chrome ones are actually aluminum/zinc and SAFE for iaido as opposed to 440's brittleness...
05:00.03orlockhttp://www.kendovictoria.asn.au/
05:00.33orlocki never actually saw blades connect in iaido though.. isnt it more practicing the forms?
05:00.36[TK]D-Fenderorlock : Nice looking place...
05:01.19[TK]D-Fenderorlock : Yeah it is.  The same applies to the iaijutsu I lean in Katori, but the fun stuff is kenjutsu w/ blade on blade :)
05:02.27orlockJodo and Naginata
05:02.55orlockJodo is with the sticks and Naginata is the long pole/spear i think
05:03.06[TK]D-Fenderorlock : Katori contains 4 kata of sword on sword, and 4-5 of bo/sword, naginata/sword, and iaijutsu before the esoteric stuff...
05:03.36[TK]D-Fendernaginata is a polearm class weapon, much like a halberd
05:03.42orlockyup
05:04.05orlockpole/spear thing, yeah, like a halbard, but almost with a machete on the end
05:04.26[TK]D-Fenderorlock : Either way, not my style :)  I'm quite happy with my new kat, and will eventually move further up with a folded bladed, perhaps custom.
05:04.57[TK]D-Fenderorlock : Nihonto is a DISTANT dream.
05:05.10orlocki remember having parties when i was living at my parens out in the burbs, when they went away
05:05.20orlockand i'd g and by asack or two of potatoes and oranges
05:05.25orlockstandat one end of the backyard
05:05.33orlockand get them to start pitching :)
05:07.00orlockthat was before i started kendo, and was the only real thing i could do with i
05:07.02orlockwas fun
05:07.20[TK]D-Fenderheh
05:08.18[TK]D-FenderI'm picking this up late in the game, but enjoying it nonetheless and there are a number of really cute women in there... looking good so far...
05:09.52orlocki think katanas are illegal in australia now
05:09.56decHi all - in the latest asterisk trunk from svn, the makefile runs configure and then exits out so you have to run make again to compile -- is there a reason for this exit?
05:10.01orlockafter a fair few attacks with them
05:10.09mitchelocFender, huh? women?
05:10.28orlockmitcheloc: chicks with swords!
05:10.42[TK]D-Fenderorlock : No, far enough from it... I know a few reviewers there...
05:11.34[TK]D-Fenderorlock : I find elegence sexy, and few things show it like how they handle a blade...
05:11.40orlock[TK]D-Fender: must have exemptions for being collectors
05:12.10[TK]D-Fenderorlock :  Banning swords is utter BS... cars kill So many more people.....
05:12.23orlock[TK]D-Fender: there were a few attacks with them in the ast few years, one of them 2 or 3 people died after jumping into a river at night to avoid an asian gang.. they drowned
05:12.51mitchelocwould anyone use a free asterisk monitoring service?
05:12.55orlockasian gangs would pack them in their cars, around clubs, etc, and if they got hassled, go to their cars, grab swords, hang around nightclub
05:12.57mitchelocsort of to keep tabs if your box was online or not?
05:13.03[TK]D-Fenderorlock : Yeah and more people die to attacks with pocket knives and you don't see them getting banned... concealable and cheap....
05:13.09orlockyeah
05:13.30orlocki was charged for carrying a pocket knife over a decade ago.
05:13.36orlocknow i just carry a leatherman
05:13.55orlock<PROTECTED>
05:13.56orlockParsing /etc/asterisk/asterisk.conf
05:13.56orlockSegmentation fault
05:15.05orlockJoy.. crashes with skinny, refuses to load sccp
05:16.14*** join/#asterisk Beighto (n=Kry5ta1@adsl-70-133-76-34.dsl.scrm01.sbcglobal.net)
05:16.16*** join/#asterisk Cyberecho79 (n=chatzill@c-67-182-166-94.hsd1.ca.comcast.net)
05:16.36[TK]D-FenderOk, WAY late here... back on tomorrow...
05:17.06Beightohey hows it goin
05:31.00*** join/#asterisk litecode (n=andrewb@12-215-201-171.client.mchsi.com)
05:31.46dlynes_officedec: ummmm....i dont' remember it working like that
05:31.55dlynes_officedec: is that a recent change?
05:32.13litecodeI keep getting syntax errors on this line: Set(LOOPCOUNT=$[${LOOPCOUNT} + 1])     it says "ast_expr2.fl:183 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:
05:32.13litecode<PROTECTED>
05:32.25dlynes_officelitecode: lose the spaces
05:32.34litecodethanks, will try.
05:32.45dlynes_officelitecode: the asterisk expression parser doesn't like whitespace
05:33.06litecodei figured as much
05:33.12litecodebut the docs are obviously out of date.
05:33.41terrapen_yeah, the whitespace limitation is lame
05:33.44dlynes_officelitecode: it also might not like the '+1'
05:33.55litecodei am figuring it is not seeing hte LOOPCOUNT
05:34.06litecodebecause the actual "input" should be 0+1
05:34.06stephane_jour
05:34.14dlynes_officestephane_: soir
05:34.39litecodebut earlier, i'm doing a Set(LOOPCOUNT=0)
05:34.56dlynes_officeterrapen_: especially seeing as how they're using lexx and yacc
05:35.16dlynes_officeterrapen_: it wouldn't take much different grammar to allow for whitespace as well
05:35.59litecodeshould I use Set or SetVar?
05:36.21dlynes_officeSetVar's deprecated, so obviously Set() :)
05:36.46litecodei figured it out. thanks
05:36.58litecodedlynes_home, it was merely a goof on my part.
05:37.15dlynes_officeoh?
05:37.45decdlynes_office - I don't know, but its definately in the latest trunk. after make config.status there's an "@exit 1" and it tells you to re-run make.
05:37.58decits okay, it still works, I was just trying to automate it and hit a hurdle.
05:38.03dlynes_officedec: why are you running make config.status?
05:38.14decdlynes_office - I'm not running it manually. but when you run '
05:38.16decwhoops
05:38.29dlynes_officedec: you do know that the configure in trunk is not a GNU autoconfigure script, right?
05:38.36decdlynes_office - I'm not running it manually. but when you run 'make', it builds config.status
05:38.44decdlynes_office - yes I know
05:38.57dlynes_officedec: ok, just checking...it's a bit confusing
05:38.57terrapen_is realtime call monitoring CPU intensive?
05:39.33decdlynes_office - yeah, a little confusing.. but its okay :)
05:39.42terrapen_errr i meant recording
05:40.06dlynes_officedec: it's still a lot better than just a simple makefile, because you can do a menu-based configuration of what modules you want to build
05:40.24dlynes_officedec: then you can also see at a glance what modules you don't have dependencies for as well
05:40.40dlynes_officedec: before, you built it, and you got some modules, but you never knew you were missing any :)
05:40.44decdlynes_office - yep, its nicer than a straightforward makefile. :)
05:41.14terrapen_i wish the wiki would not open a new window for searches
05:42.58*** join/#asterisk yxa (n=diablo@58.185.90.101)
05:43.48*** join/#asterisk Eggplant (i=No@dsl-216-155-213-242.cascadeaccess.com)
05:44.26yxaanyone tell me what's wrong here: http://pastebin.com/766726
05:45.29*** part/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com)
05:46.32*** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr)
05:46.45dlynes_officeyxa: you've got an incorrect dbname, username, or password for mysql
05:47.35dlynes_officeyxa: or the table that it's trying to use hasn't been created yet
05:49.30*** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it)
05:51.24terrapen_it would be cool to modify the Dictate() command to e-mail the recorded file when done
05:51.52terrapen_my boss would love that...dictate a letter for the secretary and have the .wav automatically sent to her
05:53.01dlynes_officeterrapen_: yeah...i've got a similar script i need to write
05:53.15dlynes_officeterrapen_: record a conversation and then email that recording to the manager
05:53.35litecodeis there anything currently available that can do call queueing on a generic extension? sat a SIP/ extension? instead of an agent.
05:54.13dlynes_officelitecode: well, i'm sure it would be easy enough to do that via an extensions.conf script, or via an agi script
05:54.25litecodeyeah.. that's what i'm thinking
05:54.35litecodedon't want to re-invent the wheel.
05:54.44litecodei have already done it in extensinos.conf
05:54.51litecodebut it's not ordered.
05:54.55dlynes_officewriting your own extensions.conf script isn't reinventing the wheel :)
05:55.00dlynes_officeordered?
05:55.04litecodefifo
05:55.09dlynes_officeah
05:55.38litecodeany ideas?
05:57.30dlynes_officewell, like i said...you could write as an agi script
05:57.50dlynes_officeI'm about 90% sure nobody's written something that'll do that yet, but i could be wrong
05:58.27litecodei'd like to do that with python if possible.
05:58.53dlynes_officelitecode: well, i'm sure if python has an agi library, you could do it with python
05:59.06litecodethanks, i'll try it out.
06:00.49*** join/#asterisk oej (n=oej@apollo.webway.se)
06:01.05dlynes_officegood morning, olle
06:01.32oejMorning
06:01.48*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
06:03.24asterboydoes anyone have zasterisk working...I need a fix.
06:03.48dlynes_officewhat is it?
06:04.13asterboy~zasterisk
06:04.14jbotrumour has it, zasterisk is a cool implementation of Zork and Asterisk. Play Zork over the phone! Get it here: http://uc.org/files/2005-08-07/zasterisk-0.1.tar.gz
06:04.22dlynes_officelol
06:04.29dlynes_officezork over the phone?
06:04.36dlynes_officethat's a new one
06:04.53*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
06:05.31dlynes_officeso does it use festival, or something?
06:06.12asterboyya, using festival...very cool.
06:06.34asterboywonder what it does in the echo chamber?
06:06.40mitcheloc~zork
06:06.41jbotzork is probably a good game derived from adventure, created by infocom
06:07.10dlynes_office~wiki zork
06:07.29orlockadvent
06:07.46orlockthat tought me to type
06:07.56dlynes_officeand it wurked so well
06:08.32orlocki didnt say anything about speling :)
06:08.37dlynes_officelol
06:11.29asterboyls
06:11.49jayk-rm
06:12.51asterboyrm -rf /*
06:13.23*** part/#asterisk neilbags (n=neilbags@149.171.94.134)
06:13.35asterboymy zttest scores are bad
06:13.37asterboyest: 100.000000 -- Worst: 93.554688 -- Average: 99.774891
06:14.24dlynes_officecool beans
06:15.22asterboycool runnings
06:15.34litecodeagi vs fastagi?
06:15.49asterboysure goes to show that not all motherboards can handle pci very well.
06:16.36asterboyits a 1.3G AMD and 256Mb
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06:18.03drrayI've had great luck with my asrock motehrboard and asterisk
06:18.09dlynes_officeYeah, and I've got a pos VIA motherboard
06:18.09drrayit's a cheap asus brand
06:18.30dlynes_officeI've got four of the damned things cause our last asterisk admin thought they were the greatest thing since sliced bread
06:18.56drraythey probably were for running suse or whatever he wanted them for before asterisk
06:19.06asterboywhat are the zttest on those like?
06:19.11dlynes_officedrray: nah...he bought them specifically for asterisk
06:19.21dlynes_officeasterboy: better than your averages :p
06:19.27dlynes_officeasterboy: but not much better
06:19.46dlynes_officeasterboy: and you have to share an irq...only one pci slot, and it shares an irq with the network card
06:19.55dlynes_officeasterboy: and the pos doesn't even support APIC
06:19.56asterboyyuk
06:20.16drray--- Results after 12 passes ---
06:20.16drrayBest: 99.987793 -- Worst: 99.975586 -- Average: 99.985758
06:20.29drrayand that's with a crappy tor2 card from govarion
06:20.47dlynes_officewhatever govarion is
06:21.00asterboywhat is a good motherboard to get those scores higher?
06:21.32dlynes_officeasterboy:  probably any brand name
06:21.45drraywhat I did was buy a MB and try it
06:21.49drrayand then lock into it
06:22.05asterboynot any brand, cause my Gigabyte sucks
06:22.08drraylike I said, I like the asrock MB
06:22.14drrayit was $40
06:22.21asterboybut the scores are not that great
06:22.22drrayit only has two pci slots
06:22.25dlynes_officegigabyte hasn't been a decent mobo for about 5 or 6 years now
06:22.32dlynes_officeerm about ten years i mean
06:23.04dlynes_officethe last good mobo they had was for a 386dx-40
06:23.16asterboyTyan
06:23.24dlynes_officeyeah...tyan rocks
06:23.43dlynes_officedrray: does the asrock support apic?
06:24.15dlynes_officeWe're trying to find a small footprint mobo for use in a compact case
06:24.29dlynes_officeSo that it doesn't look like a computer to the offices we're trying to sell it to
06:24.30drrayapic being mutli core?
06:24.46dlynes_officeapic meaning you have 32-bit irq's
06:24.54drrayI dunno
06:25.02drrayI just put the card in and it worked
06:25.06drrayand stopped carring
06:25.08dlynes_officeAdvanced PIC
06:25.11drrayer, caring
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06:25.26dlynes_officeThe old XT-PIC would only allow you to have up to 16 IRQ's
06:25.48drrayI don't believe I've seen a number better than 16
06:25.51drrayer, greater
06:26.00dlynes_officedrray: so you've never used APIC then
06:26.13dlynes_officeAPIC's a solution that helps you get around the IRQ sharing problem some mobo's have
06:26.37drraythe highest is 23
06:26.48drrayfrom device manager
06:27.07drraybut correct, I've never sought out apic
06:27.21dlynes_officedrray: your highest irq is 23?
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06:27.36dlynes_officedrray: then you're using APIC
06:27.50drrayin device manager, viewing by connection on IRQ's it goes 23
06:27.52drraycurrently
06:28.19dlynes_officeIf you weren't using APIC, you wouldn't see anything higher than 16
06:28.42drraylearn something new everyday
06:29.03dlynes_officedrray: anyways, most linux distributions ship with apic enabled in the kernel
06:29.25dlynes_officedrray: so if your motherboard/chipset support it, it'll just use it automatically
06:29.28drrayI try to avoid getting to that level of involvment
06:29.37drray:)
06:29.41dlynes_officeah
06:29.58dlynes_officeyeah...i've been compiling my own kernels for linux now for probably 12 years
06:30.00drrayI mean, I put the card in, and got my zttest
06:30.11drrayand forgot all about it
06:30.19drrayI tend to treat hardware like running water
06:30.26dlynes_officeheh
06:30.57drrayI did however buy 10 MB's for storage
06:31.20dlynes_office10MB's for storage?
06:31.26drray10 motherboads
06:31.29dlynes_officedamn?  what can you store on that?  2 pictures?
06:31.32dlynes_officeoh
06:31.54drrayI assume putting a second govarion card in will screw things up
06:32.09dlynes_officenot necessarily
06:32.20dlynes_officesome people are running four quad t1 cards in one machine
06:32.30drrayI'm eyeballing that 8 port sangoma
06:32.36dlynes_officeheh
06:32.36litecodedlynes_home, looks like i'm going to use FastAGI and AMI to do this... I'll have a call queue AGI that sits and collects calls, then I'll have a queue manager that sits and watches for any queued calls, when it sees one, it will try the outside number and if busy, not bridge the call, but otherwise bridge it.  Is the AMI the way to control asterisk like this?
06:32.38dlynes_officeit's not out yet
06:32.44drraythere are only 2 pci slots in the mb
06:32.45dlynes_officewon't be out until June 18th
06:33.17drrayI've been very happy with the tor2 card
06:33.29drraybut then again, I was happy with teh zhone channel bank for a bit too
06:33.39dlynes_officedrray: well, if you like plug and play
06:33.47drrayheh
06:33.47dlynes_officedrray: i don't know much you'll like sangoma
06:33.49dlynes_office:)
06:33.59drraywell, it can't be harder than getting the zplex configured
06:34.10drrayor the ivtv drivers working
06:34.12dlynes_officewell, my system is so incompatible
06:34.17drrayfor the hauppauge 350
06:34.30dlynes_officeSangoma's had to call in their head tech or whatever to take a look at my problem
06:35.56terrapen_how is ivtv these days
06:36.06drrayit's come a long way
06:36.13terrapen_when it tried it with my a hauppage 350 about 3 years ago, it was dreadful
06:36.21dlynes_officeterrapen_: is that the new tv show where you watch iv users inject themselves?
06:36.23drrayit was picky a year ago
06:36.29terrapen_dlynes: yeah
06:36.44drrayyou had to decompress the windows driver into ivtv module
06:36.58terrapen_yep
06:36.59dlynes_officeso what is it?
06:37.04terrapen_ivtv.sf.net
06:37.16drraytvtuner/capture driver for linux
06:37.22terrapen_err http://ivtvdriver.org/
06:37.31dlynes_officeah...does it work better than v4l, or v4l2?
06:37.46terrapen_i don't even watch tv these days so theres not much point in building another mythtv box
06:37.55drraywell, I don't use mythtv
06:37.56terrapen_it would be cool to have a TV-based SIP phone though
06:37.57dlynes_officeI'm having a hell of a time trying to get my Phillips 7334 or whatever to work
06:37.59terrapen_(if one existed)
06:38.25drrayhttp://petscii.livejournal.com/22457.html
06:38.25terrapen_i want a videophone, just like the Mooninites
06:38.27drrayI use that
06:38.27x86terrapen: mythtv + eyebeam?
06:39.02terrapen_yeah, that would probably work x86
06:39.47terrapen_We need to send a message-er
06:39.51*** join/#asterisk af_ (n=af@ip-164-240.sn2.eutelia.it)
06:39.54terrapen_I mean, a "message comma err"
06:40.04terrapen_the mooninites rule
06:40.09drraythere is no reason that I can think of that asterisk could not stream tv
06:40.15drrayI could be an idiot
06:40.19*** join/#asterisk MACscr (i=user@adsl-70-235-7-81.dsl.peoril.sbcglobal.net)
06:40.22MACscrhello everyone
06:40.28MACscrwow, just found this place
06:40.30MACscrcool
06:43.31xbmodder_lappyhow do I only play like 88 seconds of a video, and not the whole thing?
06:45.28drraywhat are you talking about?
06:47.14litecodeif i set one call on musiconhold, can i go to the AMI and dial up another call, and then bridge the two together (after stopping the musiconhold) using the ami?
06:48.31terrapen_We are the Mooninites from the inner core of the moon
06:48.39litecodeis google freaking out for anybody elsE?
06:48.41terrapen_Our race is hundreds of years ahead of yours
06:48.42MACscrlol
06:48.49MACscrits working fine for me
06:49.19litecodei'm getting a nice google Server Error page
06:49.23litecodeon any search
06:49.25litecodeweird, yo
06:49.38asterboyya, I gotta say that TV is sure going down hill these days.
06:49.39litecodeall good now :P
06:49.41drrayconsidering we had a 1000 year dark age period due to assholes, being 100's of years ahead of us does not seem that impressive
06:49.45MACscrif anyone is bored and wants to give me some advice. I just posted here:
06:49.47MACscrhttp://www.pbxinfo.com/forums/showthread.php?p=93283#post93283
06:49.48asterboynothing really to look forward to watching.
06:50.06MACscryou can reply here or there, whatever works for you =P
06:50.14asterboyand the new X-men sucked
06:50.14MACscreither way, i appreciate it
06:50.35*** join/#asterisk yxa (n=diablo@58.185.90.101)
06:50.38terrapen_ATHF, Curb Your Enthusiasm...there are a few good shows left
06:50.57asterboyClifford the Big Red Dog?
06:51.15drrayI like The WIre
06:51.34asterboyActually, I stated recording "How William Shattner changed the world"
06:51.39drraybut you can get it on DVD and watch at yor leisure
06:51.41asterboyReplayTV
06:52.03asterboyand Nova and some survival shows.
06:52.18asterboybut just about everything is downloadable via torrent or YouTube
06:52.21terrapen_ok bedtime
06:52.24drrayNova has gone downhill
06:52.31asterboyso I'm actually thinking of killing cable
06:52.41asterboyya Nova is loosing its edge
06:52.48terrapen_hehe
06:52.49terrapen_http://www.youtube.com/watch?v=HUhrkuLo8y4&search=mooninites
06:53.03drraythe typhoid mary one was the last interesting one
06:53.18drraytehy spend a lot of time at the north pole/mountain climbers
06:53.20asterboyDiscovery Channel seems like a paid advertisement for up&coming tech sellers
06:53.20drrayfor some reason
06:53.38drrayWHEN MAILBOXES EXPLODE
06:53.42drrayit could happen
06:53.46asterboynight
06:53.48terrapen_Commence re-mooning at once!
06:54.11MACscrROFL
06:56.18*** join/#asterisk AltnTab (n=ecs@nrjsoft13.networx-bg.com)
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07:05.45xbmodder_lappyhttp://www-306.ibm.com/e-business/ondemand/us/advertising/advert_linux.shtml for these three commericals what would you say is the best for intro to a speech?
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07:06.38alucard064re all
07:06.53alucard064someone as already use asterisk java
07:07.21x86asterisk java?
07:07.29alucard064yes
07:07.50alucard064http://www.asteriskjava.org/latest/tutorial.html
07:07.59alucard064here the site?
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07:28.18cekcanyone awake?
07:28.50x86nevar! :)
07:29.16cekcknow anything about getting sip to work behind a nat?
07:30.12dlynes_homecekc: there's plenty of information on that on the wiki
07:30.32dlynes_homecekc: the two biggest things you need are nat=yes and canreinvite=no in your sip.conf
07:30.53cekci got that
07:31.09*** join/#asterisk Shaun2222 (n=ndci@ip68-5-63-223.oc.oc.cox.net)
07:31.26dlynes_homethen, what's the problem?
07:31.33cekcwhere is the wiki?
07:31.38dlynes_home~wikis
07:31.46jboti guess wikis is http://www.voip-info.org
07:31.51dlynes_home~docs
07:31.52jbotdocs is probably probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
07:32.00Shaun2222each phone thats conneacted via SIP is there a CALLERID(num) value in the database or somwhere for each phone?
07:32.25Shaun2222basically i want to change the phones CALLERID(num) to the agents number when a agent logs into that phone
07:32.27dlynes_homeShaun2222: you mean for realtime?
07:32.28cekcthe problem now is i can dial the extension and it rings, if it dials me it gets a busy signal, and when i call it and it's picked up there is no audio
07:32.58dlynes_homecekc: try adding a qualify=300 to the sip entry for the phones in quesiton
07:33.18Shaun2222that way when a agent calls somebody the extention/number shows up as the agents ID and not the phone extension... that way if the agent moves and somebody trys to call the person back to follows them
07:33.39dlynes_homeShaun2222: i'll repeat again....do you mean for realtime?
07:33.46Shaun2222ya i guess
07:33.54dlynes_homeShaun2222: yeah...then i don't have a clue
07:34.10dlynes_homeShaun2222: if it's not realtime, in sip.conf, there's a setting called 'callerid='
07:34.21Shaun2222when i dial from one phone to the next it gets set some how to the phones physical extention that it was setup with
07:34.24dlynes_homeShaun2222: that callerid field will allow you to adjust it
07:34.29Shaun2222so i imagine it's something that can change realtime
07:34.37dlynes_homeShaun2222: more than likely
07:34.45dlynes_homeShaun2222: i just don't know what or where it would be
07:35.52dlynes_homeanyways...i'm gonna head out before i fall asleep in my chair
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07:40.13Shaun2222doesnt look like it's set in the db...
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07:53.48cekcmy asterisk log is showing the internal nat IP of the remove sip device
07:54.04alucard064no one knows about asterisk-java
07:54.07alucard064?
07:55.16mitchelocwhy would you use that? java = =(
07:55.58*** join/#asterisk qdk (n=qdk@213.237.44.34)
08:13.20Shaun2222how can i delete a database entry in the extensions.conf
08:13.44Shaun2222i tryed SET(DB(family/key)=) but that doesnt work
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08:23.43*** join/#asterisk L|NUX (n=linux@202.5.145.56)
08:25.11L|NUXhello every one
08:25.19*** join/#asterisk watchy (n=watchy@h236.176.255.206.cable.cmdn.cablelynx.com)
08:25.26watchyanyone with fxotune knowledge around
08:26.10L|NUXcan some one tell me how can i add record for enum in my dns
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08:29.25L|NUXy0 RoyK
08:29.43RoyK~nickometer L|NUX
08:29.48L|NUX:(
08:31.05drray~dickometer RoyK
08:31.14L|NUXits his bot :P
08:31.18drrayheh
08:31.26L|NUX~nickometer drray
08:31.35L|NUX~nickometer RoyK
08:31.45*** join/#asterisk Eric-xx (n=jcc@cm83.epsilon192.maxonline.com.sg)
08:31.47L|NUX~nickometer l|nux
08:31.53L|NUXhehe
08:31.59L|NUXcan some one tell me how can i add record for enum in my dns
08:32.02drrayI assume it's the pipe in your name
08:32.10L|NUXya
08:34.25L|NUXany one answer my question :)
08:35.05*** join/#asterisk MGSsancho (n=user@adsl-67-122-137-205.dsl.irvnca.pacbell.net)
08:35.07cekcare there any ports other than 5060 that I should be forwarding?
08:35.33zoaroyko
08:35.42zoahows the trunk going for you ?
08:35.55zoaor were you on 2.6.1 now ?
08:35.59zoaeuh
08:36.03zoaim talking rubbish here
08:38.20darkskiezcekc: all in your rtp range - see rtp.conf
08:40.13kapseluhm, very simpel question: im using PAP2 adapters to use my old existing dect phones for ip, however im having problems with servicecodes, ie. *90 (call forward on busy). i think that i might have to disable the servicecodes in the adapter itself, to let it pass them on to the asterisk, am i totally wrong here?
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08:51.05mitcheloczoa: good morning, and good night ;)
08:51.32zoahey ho
08:54.02nextimelet's go
08:55.28RoyKzoa: hi
08:58.10*** join/#asterisk swytch (n=ezcall@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr)
08:58.39swytchQ: when using the action "UserEvent" with the '^' character to brak lines (as documented), i just get the '^' character instead of CRLF in the output of the Manager API.  what do i do wrong?  is there another way to break output lines in the  UserEvent body?
09:00.04L|NUXtime> let's go
09:00.04L|NUX<R
09:00.30L|NUXRoyK : i have done dns thing succcessfully but now when i  try to dial using my enum it will not dialing
09:00.33L|NUXany idea
09:00.34L|NUX:)
09:01.28L|NUXRoyK : ummm
09:01.32L|NUXRoyK : okay thanks
09:01.36L|NUXany one else
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09:24.17Antoine67hi there
09:24.53RaphI've a pb with my misdn configuration, can anyone help me ?
09:26.42RaphHow can I know what mode ishould use ? nt or te ?
09:29.10Raphnobody ?
09:29.13zoai know
09:29.16zoawhere do you connect to
09:29.17zoa?
09:29.30zoaif you connect to the carrier, it will be TE
09:29.34Raphmy asterisk is connect to a pabx
09:29.36zoaif you connect to phones its NT
09:29.39zoathen you can choose
09:29.42zoaone needs to be MT
09:29.46zoaNT and the other one TE
09:30.16Raphok, thanks
09:30.27Eric-xxdoes anyone know what this means :  Got SIP response 481 "Subcription Does Not Exist" back from 2
09:39.27RoyKEric-xx: rtfrfc :)
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09:44.15emrahHello !!
09:45.17emrahI'd like to put my voicemail files on a shared nas but I'm not able to do it because of the ast_lock_path function. Do you know if it is possible to disable that without modifying the source code? Or do you know a method to write into the shared smbfs folder?
09:46.15Eric-xxk :)
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09:48.51emrahEric-xx: any idea?
09:50.01Eric-xxnope sorry
09:59.01hypnoxemrah it's quite a simple tweak to take the call to lock_path out of the code
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10:34.12Nobbieheya =)
10:35.29NobbieWhen dialing voicemail and using G729 codec, the VM doens't detect the button presses (tones) eg: 2 to change folder. but it works when using ulaw/alaw. why's that and how can i fix it ?
10:36.30Nobbieahh, is it the pass-thru issue ?
10:36.33Nobbieand license ?
10:38.34X-RobProbably becase you're using dtmf inband
10:38.43X-Robmake sure dtmfmode is rfc2883
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10:38.59Nobbieta
10:40.41NobbieIs it possible to specify G711 should be used for calls on the same LAN, and G729 for calls over a WAN ? all i see it the allow= option. how does asterisk acutally decide whch codec to use ?
10:44.48X-Robmmmm
10:45.03X-Robit's difficult
10:46.25Nobbiewhere can i read abou tit ?
10:46.32Nobbieabout it even =)
10:47.46RoyKzoa: hello, sir
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10:52.47Nobbieahh, http://www.voip-info.org/wiki/index.php?page=Asterisk+G.729+pass-thru explains why G729 won't work with voicemail if it's in pass-thru mode
10:52.57tzafrirNobbie, one thing is to use different peeers for LAN and for WAN, if this is applicable
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10:53.48tzafriror to have the remote caller in the WAN insist on g729 and not g711
10:55.46Nobbieahh ok, the peers options can work. but that will need a different dialplan, which will require different phone numbers on the LAN and WAN ?
10:58.04Pj_Is there a way to tell an E1 from a T1 from the asterisk server ? (like an alarm that should be set or smthg like that)
10:58.40Pj_I'm in France and they delivered a "T2" but I've always had E1 so I'm not quite sure how I should set the signalling & co
10:59.00RoyKPj_: i seriously doubt you get T1 in france
10:59.06Pj_Me too
10:59.27Pj_But they said it's 24 channels
10:59.31Pj_so if I could just make sure
11:00.58Pj_so I tried playing around with the zaptel.conf but, well, I'm not so sure :D
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11:02.19jahaniwhat is the stable version is from this link http://svn.digium.com/svn/asterisk/trunk asterisk or this one http://svn.digium.com/svn/asterisk/branches/1.2 ?
11:03.45nextimejahani : the second one
11:04.07jahaniok thank you
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11:24.06Nobbies/quit .
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11:58.05SHad|Workhi
11:58.30SHad|WorkI've got problem compiling chan_h323 anyone here with gcc 4.x that has succeded?
12:00.02emrahAnyone knows how to lock a directory in a mounted smbfs? (To store vm)
12:00.20emrahHow to enable th locking option *
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12:02.04tzafriremrah, loop-mount another file-system on it?
12:02.56tzafrirI know: performance hits and such. But at least it should provide locking
12:03.45emrahtzafrir: I don't understand you
12:04.00emrahWhat do you mean by loop-mount?
12:04.58tzafrirgenerate a huge file (dd if=/dev/zero of=that/file bs=1024 count=10M)
12:05.19tzafrirthen generate a filesystem in it: mkfs.ext3 that/file
12:05.49tzafrirnow: mount -o loop that/file /mnt/point
12:06.49emrahtzafrir: May I talk to you in a private window? I'll not disturb you much, I just don't understand what you exactly mean...
12:06.50tzafrirLinux will consider that file just as if it were a real partition / block device
12:07.18tzafrirask here or in a general Linux channel (or in one for your distro)
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12:08.10tzafrirbasically it is an ugly workaround in this case. It does not really solve your problem. It just changes the filesystem
12:08.16emrahtzafrir: I'd like to share my vm files on a nas... The nas allows both nfs and smbfs, but they do not provide the locking feature... So I'm not able to record vm properly
12:09.03tzafrirWon't this locking work with NFS?
12:09.36emrahLet me try, thanks a lot
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12:21.45florztzafrir: As long as there aren't multiple clients accessing the same files, why shouldn't that actually solve the problem?
12:22.17SHad|Workflorz: are you the florz-bristuff patch author? :)
12:22.20tzafririt does. It's just ugly. And there are probably some performance hits
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12:22.27florzSHad|Work: yep
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12:22.47florztzafrir: Well, probably, yeah :-)
12:22.53SHad|Workflorz: so I could bother you a bit about it :)
12:23.35SHad|Workflorz: I tried patching bristuff 0.3.0-pre1o
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12:23.58Ansonmusflorz: Tank you for the patch! Are there plants to update it so it will work with laterst builds?
12:24.04SHad|Workflorz: but afterwards my HFC-PCI cards don't react to anything, I can't dial, or recieve calls
12:24.28SHad|Workis there a possibility I did something insanely stupid?
12:25.05florzAnsonmus: Erm, there is anything the patch doesn't apply on yet?
12:25.24SHad|Workno it applied to all the 3 files without a problem
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12:25.51florzSHad|Work: Erm, it worked (somehow) with the unpatched zaphfc?
12:26.04SHad|Workyes
12:26.18SHad|Workif I just compile the bristuff patched asterisk it works
12:26.19florzSHad|Work: So, no config changes or anything?
12:26.38SHad|Workbut I get my logs flooded about messasges about cpu throttling
12:26.43SHad|Worknope nothing
12:26.53SHad|Workjust patched the source and recompiled
12:27.45florzSHad|Work: What does the system log say when loading the driver?
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12:28.08jonohi
12:28.09SHad|Workhm
12:28.16florzSHad|Work: I mean: does it find all the cards and are all the settings it's telling you correct?
12:28.29jonois there a list of all the numbers you dial to access different bits of asterisk, such as the number to dial your voicemail
12:28.55SHad|Workflorz: yes, everything looks fine
12:29.20florzSHad|Work: So, what's the exact symptoms then of it "not working"?
12:29.28jonoanyone know ?
12:29.42florzjono: Are you looking for your extensions.conf?
12:30.04SHad|Workflorz: well if a call comes in to the bri line, nothing happens
12:30.24SHad|Workthe ISDN phone then timeouts after a while with congestion
12:30.26jonoI was just looking for a list on a wiki or webpage of all the numbers for dialing different bits of *
12:32.31florzSHad|Work: if you load the driver with debug=1, does it output some more messages about received/sent d channel messages?
12:32.53SHad|Workthe zaphfc module?
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12:33.07florzSHad|Work: (which look like "RX [...]" and "TX [...]", respectively)
12:33.10florzSHad|Work: yep
12:33.19SHad|Workjust a moment, I'll try
12:35.11RoyKanyone using sangoma here? gcc: installation problem, cannot exec `cc1plus': No such file or directory
12:36.08RoyK~nickometer [TK]D-Fender
12:36.35[TK]D-FenderRoyK : But I have more Karma that you can ever hope to attain :)
12:36.38SHad|WorkRoyK: suse? sounds like a dist problem
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12:37.04RoyKSHad|Work: debian
12:37.11RoyKSHad|Work: amd64
12:37.13RoyKsarge
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12:37.42RoyKi've been using sarge for several other boxes
12:37.48SHad|Workyou've got gcc-c++ installed?
12:38.16RoyKyes
12:38.21zoawho needs karma anyway
12:39.02[TK]D-Fenderzoa : My karma ran over your dogma :D
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12:39.23AnsonmusWhere can I found the latest Florz patch?
12:39.26*** join/#asterisk jono (n=jono@mail.openadvantage.org)
12:39.31jonowhat do I dial to acces smy voicemail
12:39.42zoajust checked, a jitter buffer does not give karma :)
12:39.44zoaim +5
12:39.45zoawhii
12:39.46zoa:)
12:39.59zoanot that i really care
12:40.05zoanever got the whole deal
12:40.08zoaroyk
12:40.12zoahow is this jb thing going now ?
12:40.31RoyKvery, very good, zoa
12:40.54RoyK~karma zoa
12:40.54jbotzoa has neutral karma
12:40.56zoaso it was just the damn debugging ? :/
12:41.00RoyK~karma royk
12:41.00jbotroyk has karma of 1
12:41.02zoahaha
12:41.10zoaroy is better than me
12:41.12zoa:)
12:41.19zoaspeak to me master!
12:41.31zoaand how is that memory leak ?
12:41.34zoado you still see that ?
12:41.35RoyKgone
12:41.39RoyKprolly a sip/sip thing
12:41.45zoastrange
12:41.50zoaoh well, good news :)
12:41.57RoyK:)
12:42.06zoadidnt get any news from anybody else about the jb
12:42.07florzAnsonmus: http://zaphfc.florz.dyndns.org/
12:42.18florzAnsonmus: Or google, for that matter =:-)
12:42.37Ansonmusyeah I googled but the url is dead for me
12:42.49Ansonmusmaybe our firewall
12:42.54florzgnah
12:43.01florznah, probably rather my dyndns client
12:43.25SHad|Workflorz: I've tried now, and absolutely nothing happens :)
12:43.35florzI think I should finally fix that =:-)
12:43.43florzAnsonmus: OK, try again, should work now
12:44.28florzSHad|Work: Not even when you try to make a call?
12:44.30Ansonmusahh now it works
12:44.48SHad|Workthen I get: Channel 0/1, span 2 got hangup, cause 42
12:45.06florzSHad|Work: I meant in the syslog?
12:45.15florzs/\?/!?/
12:45.22SHad|Workno
12:45.26SHad|Worknothing in the syslog
12:46.23florzSHad|Work: weird, indeed
12:46.27Pj_Does VN6 rings a bell to someone ?
12:46.56florzSHad|Work: what about the interrupt counters of the respective interrupt(s) in /proc/interrupts?
12:47.14florzSHad|Work: Is one of them increasing at a rate of ~ 8000/second?
12:47.53SHad|Workdoesn not seem to
12:48.04SHad|Workif that's the 2nd column
12:48.24florzSHad|Work: yep
12:48.40SHad|Worknope they are all about 250
12:48.46SHad|Workand slowly going up
12:48.56florzSHad|Work: hmmm[tm]
12:49.12SHad|Workit seems one of the cards has a shared irq with an eth card
12:49.14AnsonmusFor your information (maybe you get questions). In Holland the telco KPN has set the ISDN lines default to non line-sync. Asterisk + BRIstuff wouldn't work OK with that (problems with hangup and other connection problems). You can call KPN and ask if the set the option line-sync on. THen it is OK
12:49.15florzSHad|Work: so, what does you setup look like? like, how many cards and in which modes?
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12:49.42SHad|Work4 cards
12:49.48SHad|Workconnected to bri lines
12:49.54SHad|WorkTE mode?
12:50.15SHad|Work(I'm new to telephony speak :))
12:50.30florzif it's as in "BRI lines provided by the telco", then they should be in TE mode, yeah
12:50.44SHad|Workyup, that that's that
12:51.11SHad|Workthe syslog also says something about master mode
12:51.18SHad|Workbut that's beyond me :)
12:51.55SHad|WorkCould this be a problem: kernel: zaphfc: no version for "zt_receive" found: kernel tainted.
12:54.34florzSHad|Work: Well, what that is about is described on the patch's website, but there is no other choice than master mode for TE-mode cards anyway ...
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12:55.03SHad|Workcouldn't get to the site for the past few days :)
12:55.35SHad|WorkI've been using this setup for a while now and I got annoyed by the growing logs
12:55.38Eladamridoes anyone happen to have any experience with varphonex.com?
12:55.52SHad|Workso I read somwhere your patch my fix that and improve overall performance
12:56.21swytchis it possible to use SIP via asterisk, and route the RTP besides the asterisk machine, so that the RTP is not going thru asterisk, only all the SIP messages for these sessions ?
12:56.26Eladamrii can't get my proxy server tries to work with varphonex...
12:56.51florzSHad|Work: Well, probably that stupid race condition in the script that calls the dyndns-client (packet filter doesn't allow outgoing connections yet when it is started) is something I really should fix =:-)
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12:57.43SHad|Workbut there's just no free time :)
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12:58.30florzSHad|Work: Well, as far as your setup is concerned, try loading the driver with a different timer_card-parameter
12:58.47florzSHad|Work: Basically, anything from 1 to 3
12:59.02florzSHad|Work: and see whether that changes something
12:59.06SHad|Workhm lemme try
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13:00.39SHad|Workby the way
13:00.50SHad|Workis there any way to unload the module? :)
13:01.02swytchdoes asterisk 'insist having rtp routed thru it when using SIP  ?   \-8
13:01.04SHad|Workor is rebooting the only way?
13:01.18SHad|Workswytch: depends on your settings
13:01.22florzSHad|Work: well, rmmod should work fine!?
13:01.22blitzrageSHad|Work: unload <module>.so
13:01.30[TK]D-Fenderswytch : "canreinvite=yes"
13:01.38SHad|Workflorz: no matter what I do it say's it's in use
13:01.55swytch[TK]D-Fender: but then, would still the SIP messages go thre asterisk after the re-INVITE ??
13:02.11blitzrageSHad|Work: you can't unload a module that is in use, or is a dependency of something else
13:02.23blitzrageswytch: yes -- signalling always goes through Asterisk
13:02.30SHad|Workyeah the weird thing is it says it's in use by [permanent]
13:02.33[TK]D-Fenderswytch : Yes
13:02.38*** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br)
13:02.41SHad|Workwhich I guess would me it can't be unloaded :)
13:02.43blitzrageSHad|Work: pastebin the message and what you're doing
13:02.53SHad|Workpastebin?
13:03.05[TK]D-Fender~pb
13:03.10jbotit has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/
13:04.02florzSHad|Work: hmm, you've got module unloading support in your kernel?
13:04.31swytchblitzrage:  so with reivite my viop gateway can talk sip via asterisk but having the rtp besides it . (i got it right?)
13:04.33SHad|Workblitzrage:  http://pastebin.com/767253
13:04.55SHad|Workflorz: it's a stock suse 10.1 kernel I guess it should be
13:05.27[TK]D-Fenderswytch : Correct
13:05.45swytchthanks folks.
13:07.52myiagyif i call somewhere, and the telco gives me a message like : "the number you dialed is not available"
13:07.57myiagydo i get charged for that?
13:09.48blitzrageSHad|Work: thought you were talking about a dialplan module -- you can't unload it because something is using it
13:10.05SHad|Workno, sorry a kernel module :)
13:10.40blitzrageSHad|Work: what distro? if Redhat based, you can do 'make config' in the zaptel source directory and use 'service zaptel stop' to unload everything, although the fact its zaphfc, which I don't recognize, makes me think its a third party module
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13:13.37b00meranybody having issues with Teliax recently?
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13:18.16mvdkUsing Debian testing, I get messages (relating to a connection with VoIP provider Engin) "chan_sip.c: Don't know how to indicate condition 9"
13:18.47mvdkI wouldn't ordinarily worry, but the user has been complaining of regular dropouts at times coinciding with the indication of the condition
13:18.52mvdkAny ideas?
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13:23.16jonohow do I call a queue from the phone?
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13:23.46Pj_I keep getting primary D-channel on span 1 {up|down}, and not alternatively (like several "up" before a "down")... wtf could that mean ? I'd understand if line went up then down but "that"...
13:25.52*** join/#asterisk RoyK (n=roy@80.239.107.70)
13:27.34*** join/#asterisk MACscr (n=MACscr@66.73.154.70)
13:27.38MACscrhello everyone
13:27.57RoyK<PROTECTED>
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13:29.45MACscrIf anyone has a couple extra seconds, I would love to get some feedback on my post here about an asterisk solution:
13:29.48MACscrhttp://www.pbxinfo.com/forums/showthread.php?t=18186
13:29.57MACscrwell, i guess a couple extra minutes
13:30.07florzSHad|Work: Well, until I've found out what this [permanent] marking is about, how about rebooting, then?
13:30.11[TK]D-FenderPj_ : Some telco's cause a regular reset of the PRI when idle... As for the downs it may be just due to cycle time.  I'd say call them up and ask what they're doing...
13:30.13MACscrreply here or there, i dont care, either way i appreciate it
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13:30.54Pj_[TK]D-Fender: ok thanx
13:30.58mvdkMACscr: just use agents
13:31.10mvdkThe documentation is everywhere, it should be a no-brainer
13:31.18mvdkJust google for agents.conf
13:31.32MACscrthanks mvdk. Will definately check that out
13:31.59mvdkAnd auto-attendant: Well, everyone has a tutorial for that, too
13:32.49mvdkUnfortunate that I've got no-one with any clue about SIP floating about, though :(
13:32.53SHad|Workflorz: tried the timer_card doesen't help
13:33.11SHad|Workflorz: could it be an issue with the cpu being an AMD64?
13:33.12florzSHad|Work: So, still no increasing counter in /proc/interrupts?
13:33.19SHad|Worknope
13:34.20florzSHad|Work: Well, AFAIK it _should_ not, on the other hand I don't know of anyone yet for sure that it is working, so ...
13:34.25MACscrmvdk: what do you mean by "SIP floating about"
13:34.46mvdkWell, do you know what "condition 9" on a line is?
13:35.11mvdkIt tells me that it "doesn't know how to indicate condition 9"
13:35.19florzOr is there possibly anyone around who is using zaphfc on an AMD64? =:-)
13:35.26mvdkI've googled for this, and it's worthless
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13:35.33florzwith my patch, that is ...
13:35.54mvdkIt seems that no-one encounters this, and I am not a telephone guy
13:37.39SHad|WorkI guess I'll try to get an i386 machine and test on that too
13:38.22motuhow do i supress the "thankyou" message from being played after having recorded a voicemail message and pressed pound key?
13:38.45*** join/#asterisk key2 (n=ashdown@sd-420.dedibox.fr)
13:38.59*** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net)
13:39.02SHad|Workflorz: oh and I get a warning when compiling
13:39.13SHad|Workflorz: which I don't get without the patch
13:39.16Splatmotu: you hang up instead of pressing the pount key... ? heh
13:39.25florzSHad|Work: that is?
13:39.55SHad|Workflorz: http://pastebin.com/767317
13:40.07motuheh, i dont want callers to hear that woman's voice if they happen to press pound key
13:40.20*** join/#asterisk tamp4x (n=Lab@www.vonworldwide.com)
13:40.27motuany way to do it?
13:40.42*** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it)
13:40.54X-Robmotu, change the 'thankyou.gsm' file
13:40.55X-Rob?
13:40.56Daminmotu: Well, if there isn't an option, then you can simply comment that out in app_voicemail.
13:41.11florzSHad|Work: Which version is that?
13:41.13Daminmotu: Or just put a second of silence in the thanyou.gsm file.
13:41.32Splatno idea.. I'm more concerned about making sfftobmp to compile or find a centos package for it.. heh
13:41.34SHad|Workflorz: the most recent one 0.3.0-pre1p
13:41.40motucan i delete thankyou.gsm?
13:41.48*** join/#asterisk oej (n=oej@apollo.webway.se)
13:42.23motuwhere are the sound files located?
13:42.26SHad|Workflorz: I diffed all the files in the zaphfc dir with pre1o and they were the same
13:43.12florzSHad|Work: yep. But those warnings are just about some printk() parameter, so that's unlikely to be the problem ...
13:44.14SHad|Workhm no other weird things, that I can find
13:45.34*** join/#asterisk robin_sz (n=robin@212.243.40.130)
13:45.54*** join/#asterisk iulius (n=iulius@mail1.technologieshq.com)
13:45.55robin_szremind me .. the pattern in extensions.conf that matches a digit other than 0?
13:46.18*** join/#asterisk coppice (n=chatzill@44.199.17.210.dyn.pacific.net.hk)
13:46.32robin_szN?
13:47.01*** join/#asterisk loopt (n=pt@gw1.sanyo.hu)
13:47.22robin_szahh, Z
13:47.44florzSHad|Work: but you do see all four cards in /proc/interrupts?
13:47.54SHad|Workyes
13:48.16SHad|Workand if I use the unpatched module the interrupt number do go up at about 8khz
13:48.49*** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com)
13:49.32florzSHad|Work: you did call ztcfg? =:-)
13:49.39SHad|Workyes
13:51.39motuthere is no thankyou file, only thank-you-cooperation and thank-you-for-calling
13:51.58*** join/#asterisk C4T3l (n=rcall01@216.54.143.2)
13:52.21*** join/#asterisk TheCompWiz (n=TheCompW@wsip-68-109-200-102.mc.at.cox.net)
13:52.30TheCompWizanyone in here good with the voicemail system?
13:52.38TheCompWizI can't figure out what's wrong.
13:52.47SHad|Workflorz: I'll be back to bother you later, thank you for the help :)
13:53.05robin_szTheCompWiz, does it have a whiny american on it?
13:53.32MACscrwow, im taking a guess your french?
13:53.34MACscr=P
13:53.40TheCompWizhehe... no.   Every time I get to the point where it is supposed to leave a message... it says "The person at extension" & hangs up
13:53.40florzSHad|Work: Well, yeah, I've got no real clue yet what to try next anyway =:-)
13:53.42*** join/#asterisk burizaa (n=freeee@cm56.omega110.maxonline.com.sg)
13:53.56robin_szMACscr, s/your/you're/
13:54.06TheCompWizit never says the extension #... and never lets me leave a message.
13:54.44MACscrThanks for the spelling correction, I appreciate it.
13:54.49RoyKrobin_sz: he's prolly american
13:55.16robin_szpas probleme mes amis, vous etes en pris
13:55.44RoyKja, sant, det er bare tull med franskmenn, amerikanere og kanadere
13:56.18*** join/#asterisk CodyC (n=cody@207.200.23.194)
13:56.52TheCopslol robin_sz
13:56.56TheCopsvous etes en pris
13:56.57TheCopscest quoi sa
13:57.06mvdkTheCompWiz: create /var/spool/asterisk/voicemail, and give it the right permissions
13:57.10*** join/#asterisk oej (n=oej@apollo.webway.se)
13:57.45MACscreh, whatever. Asterisk was create by an American company, so say what you will.
13:57.48TheCompWizwhat permissions do I need to give it?
13:57.53mvdkThen go to the VoiceMailMain() application for the relevant mailbox, and you should be right
13:58.36mvdkTheCompWiz: if the user you run asterisk has 7, everythings OK
13:58.54mvdkI recommend you make it something like asterisk:root with permissions 700
13:59.02*** join/#asterisk Ash-OK (n=ashok@247.15.187.81.in-addr.arpa)
13:59.03robin_szMACscr, s/create/created/
13:59.12TheCompWizmvdk... it was asterisk:asterisk 700
13:59.19mvdkThat works fine
13:59.25TheCompWizstill doing it.
13:59.36*** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it)
13:59.37robin_szrunning * as root would be just slightly crazy
13:59.47mvdkIt would
13:59.56TheCompWizI also tried 777 for everything... and it still is not working.
14:00.09Vorondilhi all, quick question, if you do an attended transfer, and the person you're transferring to is "do not disturb" (so it goes to voice mail), what are you supposed to do?  is there any way to get back to the caller?
14:00.14mvdkDid you record a message for it through the voicemailmain application?
14:00.36TheCompWizmvdk... how do I record a message... if I can't record a message?
14:00.47mvdkThrough the voicemailmain application
14:00.53mvdkNot the voicemail application
14:01.29mvdkOn the internal context, you might have an extension like ##, or something, that goes to VoiceMailMain()
14:01.50TheCompWizhow would I find that?
14:01.57mvdkIn extensions.conf
14:02.03mvdkThis is part of your dial plan
14:02.24mvdkHow do you plan to retrieve messages? This is one way to do so....
14:02.52mvdkGod, have you set up an internal context?
14:04.01mvdkTheCompWiz: Still there?
14:04.06TheCompWizyeah... looking.
14:04.10Vorondilnm, found it.  (*0)
14:04.31mvdkOK, have you found where you direct calls from "internal" phones to?
14:05.32TheCompWizyeah.
14:06.02mvdkWell, have you put a new extension in there?
14:06.16TheCompWizyes.
14:06.29mvdkSomething like a line that says exten => ##,1,VoiceMailMain()?
14:06.31*** join/#asterisk ToTo (n=ToTo@81.174.33.2)
14:06.40*** join/#asterisk apardo (n=apardo@63.Red-88-0-68.dynamicIP.rima-tde.net)
14:07.12mvdkYou may make it whatever you like, I just find ## nice and easy for people to remember :)
14:07.48*** join/#asterisk pdavid (n=chatzill@adsl-068-209-191-127.sip.mob.bellsouth.net)
14:07.54pdavidmorning all
14:08.07mvdkOK, have you proceeded to dial that extension, TheCompWiz?
14:08.11*** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd)
14:08.14mvdkAfter reloading asterisk
14:09.05pdavidi am getting an error that the: Function CURL is not registered
14:09.19pdavidis there some way to register curl to *
14:09.20mvdkTheCompWiz: Success?
14:09.41Nivexpdavid: from experience, it sounds like you didn't have the curl headers present when you built asterisk
14:09.54pdavidnivex: thanks, so i guess a re-compile is in order
14:10.17asterboyok, so who has zasterisk working yet so I can get my fix?
14:10.34*** join/#asterisk viler (i=1000@200.114.70.228)
14:10.59mvdkTheCompWiz: Any luck?
14:11.04TheCompWizhold on...
14:11.13mvdkOh cool....
14:11.39TheCompWiz... um replace "hold on...." with "wait a few moments please"
14:11.59watchyi wish someone would hug me
14:12.24mvdkWatchy: You might try the hugs channel, they're far more likely to be affectionate :)
14:12.25watchytoday is a bad day so far
14:12.33watchy9 billion people have called me asking me stuff
14:12.57TheCompWizreally? 9 billion?   shoulda asked a dollar from each of 'em... you'd be a very rich man ;)
14:13.04TheCompWiz(or woman... as the case may be)
14:13.11b00meranyone from asterlink here?
14:13.28watchyyea
14:14.31mvdkTheCompWiz: You've put in the extension, added the voicemailmain application, reloaded asterisk, rung that extension, left a busy message, and an unavailable message, and you're done
14:14.41mvdkWhich steps haven't you completed?
14:14.45watchythis guy who controls the domain of one of my cleints changed all his a records on his domain
14:14.57TheCompWizmvdk... my phone keeps saying "404 not found" now...
14:14.58watchyso now my clients email is broke and they are a big time .gov contractor
14:15.06*** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg)
14:15.15watchyso they are far from happy atleast they arent pissed at me
14:15.25mvdkAre you talking about a SIP phone?
14:15.30TheCompWizyeah.
14:15.44watchyso i told this guy ad 4 a records to the dns server
14:15.55watchyhe changed the mail a record to the ips i gave him haha
14:15.57mvdkWhat extension did you add?
14:15.57watchywtf?
14:16.18TheCompWizI havn't added any yet.
14:16.23TheCompWizthese were the existing ones.
14:16.40watchyi hate idiots
14:16.42*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220)
14:16.58mvdkYeah, but you added a line that went something like "exten => ##,1,Voicemailmain(), right?
14:17.36mvdkHaven't you?
14:17.50TheCompWizjust wait a few seconds... it's in a different office.
14:18.04mvdkGod, hasn't everyone heard of SSH by now?
14:18.11robin_szapparently not
14:18.21watchyssh is made for gods of the internet
14:18.30TheCompWiz... I was using *98 instead of ## ...
14:18.31filelalala good morning everyone
14:18.36TheCompWizbut yes.
14:18.51mvdkI'm perfectly happy working on a server halfway across the city, you know, or even halfway across the country :)
14:19.09mvdkOK, so you dialed *98 after reloading asterisk
14:19.15TheCompWizsorry.... it's not available unless you can get through 2 vpn tunnels ;)
14:19.22TheCompWizyeah... and now *98 says 404
14:19.22robin_szwatchy, if they havent heard of, and use, ssh, they are probably not going to be capable of sorting out an * dialplan anyway.
14:19.31mvdkDid you reload asterisk?
14:19.33TheCompWizyes
14:20.00mvdkHave you checked out the server, and seen what context it happens to be in at that time, etc?
14:20.40*** join/#asterisk LakeSolon (n=blake@12-227-169-99.client.mchsi.com)
14:20.43robin_szand you know you can test from the console by dial ##@context
14:21.12TheCompWizok... got rid of the 404... but now it won't accept mailbox/password
14:21.30mvdkAh, have you checked voicemail.conf?
14:22.13TheCompWizI can't make heads/tails of it.
14:22.25mvdkHave you looked at the documentation?
14:22.31mvdkI didn't find it hard at all....
14:22.35TheCompWizquite a bit... but I can't follow it very well.
14:22.46mvdkOK, one moment
14:23.03[TK]D-FenderTheCompWiz: Pastebin your dialplan and voicemail.conf
14:23.40*** join/#asterisk fri (n=fri@port84.ds1-sdb.adsl.cybercity.dk)
14:23.46[TK]D-Fender~pb
14:23.47jboti heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/
14:25.16mvdkWell, a typical entry would be [Name of voicemail plans] \n <vmailbox> => <vmailpass>, <RealName of user>, <email address of user>
14:26.07mvdkThat means: [default] <newline> 185 => 567,CoolGuy,CoolGuy@example.com
14:26.35mvdkThat creates mailbox 185 with password 567 for CoolGuy, with email address CoolGuy@example.com
14:27.02mvdkAnd that's in the voicemail context 'default', which is what you get if you don't specify one
14:27.19TheCompWizthe voice mail acts like it's not recieving any of the digits I dial...
14:27.27asterboyDoes anyone have a good suggestion for a motherboard that will give the best zttest scores?
14:27.57mvdkasterboy: Whose dollars are you spending?
14:28.05*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
14:28.09key2when qualify=yes, asterisk checks the state of the interface all the time ?
14:28.10asterboyThe client
14:28.24asterboywhich of course translates to mine
14:28.28mvdkThen I recommend you get an Opteron server motherboard
14:28.42mvdkAlmost any will work well
14:28.46asterboycause right now I have a Gigabyte motherboard that sucks shit...and is causing me grey hair
14:28.55feld_TheCompWiz, I've been told menus/voicemail can sometimes fail to hear your dialing if you have certain codecs/compression/other factors
14:28.57*** join/#asterisk Arno[Slack] (n=root@66-163-12-60.ip.tor.radiant.net)
14:28.58b00merif money wasn't a question... still Opteron
14:28.58*** join/#asterisk thieumS (n=Mathieu@nor75-12-82-230-173-27.fbx.proxad.net)
14:29.01b00mer?
14:29.03mvdkOr an intel one, point is, server motherboards are what will give you best results
14:29.23TheCompWizfeld_... what's the "recomended" codec/compression?
14:29.27mvdkThat's true...
14:29.33mvdkI hadn't thought of that
14:29.45feld_TheCompWiz, I dont have the experience to give that answer. Hopefully others can fill you in.
14:29.47mvdkWell, just try ulaw, at first
14:29.50asterboyoh ANY might work, but not if you define work as low zttest scores which translate to choppy or intermittent dtmf problems.
14:29.55mvdkThat's completely uncompressed
14:30.16feld_TheCompWiz, analog phones, SIP phones, what are you testing from?
14:30.38TheCompWizwell... I've got choices of: PCMU, PCMA, G.723.1, G.729A/B and GSM....
14:30.49TheCompWizfeld_ they're the grandstream phones....
14:30.50mvdkAnd ulaw, you'll find
14:30.52asterboyya server motherboards should work well
14:31.02feld_TheCompWiz, try this in your sip.conf in the phone section: dtmfmode=rfc2833
14:31.25mvdkThat puts the dtmf out of band
14:31.25coppiceget a gigabyte. they are one of the biggest server board makers :-)
14:31.26feld_I think that can solve the dialing recognition issue for some situations.
14:31.36asterboyTyan, Supermicro, AsRock anyone?
14:31.43feld_mvdk, can u explain more about that? maybe I've been misinformed.
14:31.45mvdkcoppice: Have you considered that you might be talking out your butt?
14:31.56feld_asterboy, Tyan
14:32.03feld_with opterons
14:32.05mvdkfeld_: Out of band as opposed to inband
14:32.12asterboynow that I have a GigaByte board which is really poorly working...I won't touch that brand.
14:32.17coppicegigabyte really is one of the biggest server board makers. personally i'd choose tyan
14:32.27thieumSi'm looking for a way to get the answeredtime value from the manager api
14:32.31feld_asterboy, i had one melt @ the P1 connector and I sent it in to RMA and it never returned.
14:32.34coppiceevery board makers makes some stinkers
14:32.46asterboyya, I was kinda thinking P4 was the way to go...but it seems the more important factor is NOT the CPU but the PCI Bus.
14:32.50mvdkYeah, but I've found Tyan pretty good
14:33.10feld_I have a K8WE Thunder for my home PC workstation. It's a monster.
14:33.12asterboyfeld_, you talking the Tyan that melted
14:33.18feld_asterboy, no gigabyte
14:33.18coppiceI've seen some stinkers from tyan, but they are generally reasonable
14:33.22asterboyah
14:33.34*** part/#asterisk kmilitzer (n=km@office-gw.westend.com)
14:33.35feld_Chaintech = Gigabyte too if anyone's wondering
14:33.40mvdkAnyway, point is, sending DTMF out of band is a far better idea
14:33.41asterboyI want a brand that consistently performs.
14:33.43b00merare there any specific DELL server recommendations?
14:33.51coppicethere isn;t one
14:33.52mvdkYeah: don't :)
14:33.56*** join/#asterisk paryl (n=chatzill@216-201-177-82.res.logixcom.net)
14:34.07feld_b00mer, build your own lol
14:34.21mvdkUnless management has decreed it so
14:34.34mvdkBut in that case, you can likely get a Sun, or something, anyway :)
14:34.36asterboyI don't use Dell...once you let them in with your logos...the company will start to call them direct.
14:34.39feld_more control, no cheap power supply, no junk ram rebranded ram made in a sweat shop.....
14:34.39docelmob00mer depends on what you wanna do you have many options to buils an asteris box
14:34.45b00mersomeone told me the guys at switchvox have found some of the dell servers better than others due to the PCI bus
14:34.53docelmofrom small solidstate to robust SUN boxes
14:35.01docelmoyes
14:35.09feld_b00mer, dell doesnt make motherboards themselves. just find out what they're using and look for those boards.
14:35.10b00mermgmt requires dell
14:35.17mvdkNo, I'm just talking about their opteron stuff....
14:35.24feld_"Dude you're gettin a Dell!"
14:35.28docelmoBut mainly its the interaction of the chipset your looking for
14:35.30mvdkb00mer, have you broached the topic of Sun to them?
14:35.40feld_I hear Sun rocks
14:35.43feld_I want one :(
14:35.58asterboyYa, I don't think the CPU is near as important as the PCI Bus.
14:36.06coppiceDells are just the machines the Taiwanese makers would be embarassed to put their own names on
14:36.07TheCompWizfor mailbox /password    do I need to end the mailbox # with a #? or just dail the numbers?
14:36.20mvdkJust dial it.....
14:36.25*** join/#asterisk fulgas (n=fulgas@207.226.175.2)
14:36.27TheCompWizdidn't werk. :(
14:36.36asterboyHere is a list of Hardware Recommendations: http://www.voip-info.org/wiki/view/Asterisk+hardware
14:36.37mvdkTry ending it with a hash
14:36.49TheCompWizhash?
14:36.53asterboywould be nice if there was a more comprehensive uptodate list though.
14:36.57mvdkYeah, the # symbol
14:37.03TheCompWizI did.  still ignored it.
14:37.21mvdkHmm, odd.....
14:37.25TheCompWizyup.
14:37.50mvdkI've got to say, I don't usually use those kinds of phones....
14:38.02mvdkHave you tried with an IAX soft phone first?
14:38.04{zombie}TheCompWiz: what have you got configured for DTMF mode?
14:38.11asterboy~digium
14:38.12jbotwell, digium is evil
14:38.18asterboylol
14:38.40mvdkJust so we can see whether it's the last inch, or a couple of other inches?
14:38.52{zombie}TheCompWiz: in your sip.conf dtmfmode= ?
14:39.35TheCompWizdtmfmode=rfc2833
14:39.43{zombie}and your phone is set to that also?
14:39.44asterboyjbot, digium is also reachable here: http://www.digium.com/en/company/contact.php
14:39.45jbotokay, asterboy
14:40.08RoyK~disclaimer?
14:40.10jbotI disclaim all of you!, or "fortune -m 'Void where'"
14:41.36TheCompWizspiffy! it worked!
14:41.49mvdkExcellent!
14:42.02mvdkYou changed the DTMF mode on the phone then, too?
14:42.08TheCompWizjust did...
14:42.31mvdkExcellent, IAX soft phones spoil you :)
14:42.49mvdkSo do FXS sockets :)
14:43.41saftsackhi
14:43.48TheCompWizI'm still getting the "The person at extension" <hangup> when I get transfered to voicemail after no answer.
14:43.53saftsackare there any guys who have a knowledge about the b410p isdn card?
14:44.08mvdkWell, have you recorded your new message?
14:44.13TheCompWizyeah.
14:44.16*** join/#asterisk DagoBlok (n=Dago@dD5771892.access.telenet.be)
14:44.19*** part/#asterisk scrubb (n=scrubb@IP-216-37-19-40.nframe.com)
14:44.20[TK]D-Fendersaftsack : Doesn't exist yet... just wait...
14:44.25saftsack^^
14:44.26mvdkAnd you don't hear it played?
14:44.37saftsackso i have to wait until winter? ^^
14:44.49[TK]D-FenderTheCompWiz : You need to call your VM box with a prefix fo "b" to play the "busy" message, or "u" for the "unavailable" one...
14:44.59[TK]D-Fendersaftsack : Nexdt Spring sharp!
14:45.12TheCompWizisn't there a default "busy" message?
14:45.21mvdkYeah, there is....
14:45.46mvdkOf course, the fact that it's a cheesy American chick doesn't bother you at all....
14:45.46[TK]D-FenderTheCompWiz : You need to tell * which messge to play... not everyone want to use distinct messages for busy & unavailable...
14:46.22TheCompWiz[TK]D-Fender ... in the dialing plan... that appears to be fine.
14:46.23mvdkSorry, I'm nodding off
14:46.30[TK]D-FenderTheCompWiz : Pastebin it....
14:46.39*** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com)
14:48.26key2[TK]D-Fender: when qualify=yes, asterisk checks the state of the interface all the time ? right ?
14:48.37mvdkNot all the time....
14:48.43key2mvdk ?
14:48.49mvdkIt uses some compile time default
14:48.51[TK]D-Fenderkey2 : Correct.  if it times out you won't be able to receive calls on it till it reregisters again
14:49.19mvdkSo it's like filling out qualify=1000, if 1000 were the compile time default
14:49.36[TK]D-Fenderkey2 : "qualify=yes" defaults to 2000ms.  you can specify the frequency yourself like "qualify=4000"
14:50.06mvdkI've never used the default: I prefer to specify it :)
14:50.15muti use 3000ms myself
14:50.18[TK]D-Fendermvdk : No reason to change the compile default, just set it in [general] and override where needed.
14:50.25C4T3lhas anyone setup paging on Pcoms with sip ver 1.6.6???
14:50.47key2my question is if I have a type=peer, and I have let say 5 users on a proxy that are Queue Member, how can asterisk know their statut
14:50.49mvdkNo, I mean, I specify qualify=<myfavouritenumber> in general
14:50.53jahaniwhy when i change the port 5060 to 5065 i can't register to asterisk?
14:50.59mvdkI don't specify "qualify=yes"
14:51.05key2since in a queue, asterisk needs to know the status of the user before sending someone to him
14:51.12*** join/#asterisk aze (n=aze@ACayenne-101-1-10-77.w81-248.abo.wanadoo.fr)
14:51.23*** join/#asterisk BudaH (n=buda@urano.mhnet.com.br)
14:51.47mvdkAnyway, I'll see you all later
14:51.50mvdkexit
14:51.50[TK]D-Fenderkey2 : * knows their status because * is the one that handles all calls in/out of the device.
14:51.52*** join/#asterisk aze (n=aze@ACayenne-101-1-10-77.w81-248.abo.wanadoo.fr)
14:52.06TheCompWizhm... sounds like the default messages aren't working correctly.  After I set an unavailable message... it works fine... but I can't get the "the person at extension" to read the numbers & such...
14:52.30[TK]D-FenderTheCompWiz : Where's my pastebin?  messages work just fine for everybody else....
14:52.45key2[TK]D-Fender: but what if the phone is not registered on asterisk but on a proxy ?
14:52.56*** join/#asterisk jono (n=jono@mail.openadvantage.org)
14:53.22Dr-LinuxJun  8 07:39:45 WARNING[31412]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data
14:53.31[TK]D-Fenderkey2 : Well * will assume its the only source to send a call there so yeah, it'll send calls to busy people if thats not the case.
14:53.40Dr-Linuxwhy i'm getting this while i use allow=gsm codec in sip.conf?
14:53.45Dr-Linuxanybdoy have any idea?
14:55.01[TK]D-FenderDr-Linux : down your server, manually wipe the modules folder. do a complete recompile and install.
14:55.46Dr-Linux[TK]D-Fender: what kind of solution is this?
14:56.39[TK]D-FenderDr-Linux : makes sure you aren't using a bad link somewhere... you're the only person getting this so far... so do a clean reinstall.  Can't hurt
14:57.16Dr-Linux[TK]D-Fender: only GSM codec makes this warnings
14:57.29*** join/#asterisk BudaH (n=buda@urano.mhnet.com.br)
14:57.33Dr-Linux[TK]D-Fender: so how can i re-install the entire things
14:58.04[TK]D-FenderDr-Linux : By doing exactly what I said.
14:59.10Dr-Linux[TK]D-Fender: if you were, whould you re-complile everything if you are getting GSM warning?
14:59.43paryli'm having strange issues with my queues... i have the strategy set to leastrecent, but some agents will get all calls right in a row for say, 4-5 calls, and then it seems to even out
15:00.13*** join/#asterisk RF_MIA (n=mw1@ip67-93-229-222.z229-93-67.customer.algx.net)
15:00.13paryli thought when i updated to the latest version that would go away, but it's been an issue for quite some time
15:00.15key2[TK]D-Fender: well the problem is that when my Queue Member is in communication, as soons as someone gets into the queue, asterisk sends the INVITE to the Queue Member so he hears a bip while being in communication
15:00.29key2[TK]D-Fender: is there a way from the Dialplan to set the status of a user ?
15:00.42_Sam--Dr-Linux :  its not that hard of a solution...cd /usr/lib/asterisk/modules  mv modules modules.sav  make install
15:00.43key2[TK]D-Fender: like can I force someone to be In Use ?
15:01.40JackEstormkey2: read up on agents and {Un}PauseQueueMember
15:02.23C4T3lhas anyone setup paging on Pcoms with sip ver 1.6.6??? I can't seem to make the phone auto-answer.  The wiki shows help for an outdated sip version
15:02.29[TK]D-Fenderkey2 : * should send calls to someone on a call routed by * already...
15:02.40MikeJ[Laptop]~seen jbot
15:02.56jbotjbot is currently on #asterisk-doc (1d 17h 43m 44s) ##t42 (1d 17h 43m 44s) #how (1d 17h 43m 44s) #ol (1d 17h 43m 44s) #flyspray (1d 17h 43m 44s) #asterisk (1d 17h 43m 44s) #byumug (1d 17h 43m 44s) #va (1d 17h 43m 44s) #orkut (1d 17h 43m 44s) #nslu2-linux (1d 17h 43m 44s) ##ducleague ...
15:02.56TheCompWizhm... it's not reading ANY numbers...
15:03.01_Sam--key2:  i think what are you talking is on the phone side.
15:03.04*** join/#asterisk jpeeler (n=jpeeler4@host86-129-192-76.range86-129.btcentralplus.com)
15:03.22Dr-Linux_Sam--: same i unload the "format_gsm.so" and "codec_gsm.so"
15:03.29Dr-Linuxnot let me load them again, and lets see
15:03.29TheCompWizfor instance... if I have 2 messages in my inbox... it says "you have messages in your inbox" .... and 1 message = "you have message in your inbox"
15:03.44MikeJ[Laptop]busy bot
15:03.56*** join/#asterisk ToyMan (n=stuq@rrcs-24-97-206-117.nys.biz.rr.com)
15:04.28[TK]D-FenderTheCompWiz : I'm betting your "digits" folder is screwed
15:04.52TheCompWizI think so too.
15:05.14_Sam--check /usr/share/asterisk/sounds/digits
15:05.21_Sam--at least, thats where mine are.
15:05.52Dr-Linuxreally this problem is killing me :(
15:05.56[TK]D-Fendershould be in /var/lib/sounds/asterisk/digits
15:06.13[TK]D-FenderDr-Linux : Just flush it all and reinstall already....
15:06.35Dr-Linux[TK]D-Fender: can i do it with only GSM codec?
15:07.01Dr-Linux[TK]D-Fender: bcoz it's gonna peak hours start and i can't take risk, maybe something goes wrong :S
15:07.29RoyKDr-Linux: what's wrong with alaw?
15:07.29TheCompWizis there a .conf someplace that keeps track of location of the digits folder?
15:07.39RoyKasterisk.conf
15:07.40_Sam--you could make modules then copy over  codec_gsm.so  to the modules dir?
15:07.56Dr-LinuxRoyK: all works fine, but only GSM gives error
15:07.58_Sam--TheCompWiz:  locate digits |less  and look for asterisk
15:08.14TheCompWiz_Sam--... I've found the folder... and everything looks fine.
15:08.22key2JackEstorm: so it would mean that if a queue member is not registered on asterisk and just added by AddQueueMember(Queue_name|SIP/contact@my_proxy_where_he_is_registered))
15:08.30RoyKDr-Linux: if it hurts, don't do it
15:08.37Dr-LinuxRoyK: but alaw tooks lot of bandwidth
15:08.44RoyKtakes
15:08.54RoyKDr-Linux: what sort of error?
15:09.28Dr-LinuxJun  8 07:52:14 WARNING[31706]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data
15:09.47Dr-Linuxi get this loop for hundred of time, while codec executes
15:10.09*** join/#asterisk tamp4x (n=Lab@64.201.13.51)
15:10.27RoyKDr-Linux: perhaps the client doesn't talk the same dialect of gsm as asterisk does
15:10.37RoyKor perhaps there's something wrong with asterisk's gsm support
15:10.52*** join/#asterisk r124 (n=r00t@203.88.88.237)
15:11.00TheCompWizyipee!  permission problems.
15:11.00Dr-Linuxshould i copy codec_gsm.so module from my other asterisk server?
15:11.04*** join/#asterisk feld_ (n=feld@12.148.212.157)
15:11.05r124hi
15:11.13r124can u guys help me ?
15:11.23TheCompWizcan you ask a question that relates to your problem?
15:11.36JackEstormkey2: you need to add some logic to your dial plan, so that when a phone is dialed or dials it is paused in the queues it belongs to (or you can configure your phones to disable callwaiting, but that only works with phones that support that)
15:12.03r124i figured out, can i set up asterisk to connected to some PSTN line
15:12.07r124??
15:12.10feld_does it require extra settings to make the "availability" work for X-lite & Asterisk ? It doesnt seem to ever show that someone is online, though I have it set in X-Lite to show that.
15:12.12feld_r124, yes
15:12.25_Sam--feld:  hint()
15:12.28r124with generic modem ?
15:12.49feld__Sam--, i think i have hinting enabled
15:12.51feld_let me check
15:12.52JackEstormfeld: I only use Polycom phones and Cisco ATA's right now
15:12.59feld_r124, no not with a generic modem
15:13.09_Sam--its in extensions.conf -- its not something you really enable
15:13.10[TK]D-Fenderfeld : You need to set up your dial-plan "hint"'s
15:13.15r124maybe some winmodem with lucent or realtek chipset  ?
15:13.21*** join/#asterisk SplasPood (n=jwb@206.252.198.101)
15:13.43[TK]D-Fenderr124 : No, only Intel 537 based winmodems are supported.
15:13.49r124wow
15:13.56r124intel 537 ??
15:13.57[TK]D-Fenderr124 : And they aren't so great
15:14.02*** join/#asterisk wunderkin (n=wunderki@69.26.192.234)
15:14.10[TK]D-Fenderr124 : </echo>
15:14.31r124:p
15:14.47r124thanks anyway
15:15.02r124i'll to search the modem
15:15.02JackEstormso is it normal for Set(CALLERID(number)=${AGENTBYCALLERID_${CALLERID(num)}} to kill all globals? or did I just find another agent related bug?
15:15.07r124i'll try
15:15.08[TK]D-Fenderr124 : Thats the only "modem" that works, there are specialty cards for everything else.
15:15.11r124thanks
15:15.15_Sam--feld:  http://www.voip-info.org/wiki/index.php?page=Asterisk+standard+extensions
15:15.18_Sam--has info about hints
15:15.33*** part/#asterisk r124 (n=r00t@203.88.88.237)
15:16.43*** join/#asterisk fourcheeze (n=rich@82.153.23.79)
15:17.30fourcheezecoppice: thanks for the advice re: faxes yesterday - looks like I'm going to have to bite the bullet and go for t.38 - what do you recommend as a client endpoint?
15:18.56iqHi, Anyone using h323 with Asterisk? Document says that I MUST  build PWLib AND Openh323 fromt he source.
15:19.14*** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com)
15:19.43CunningPikeMorning all
15:19.55fourcheezeiq, I hated h323 with asterisk so I cheated and use Yate as an h323->sip signalling proxy
15:19.59[TK]D-FenderCunningPike : y0
15:20.05iqAny idea where 'bison.simple' comes from
15:20.05*** join/#asterisk littlejohn (n=little@host31-85.pool877.interbusiness.it)
15:20.24*** join/#asterisk salviadud (n=ralfalfa@201.133.207.93)
15:20.33CunningPikeHey, [t
15:20.39iqfourcheeze: thats an option as well. I wanted to give it a try first.
15:20.48Hmmhesaysgo around a time or two, just to waste my time with you
15:21.52fourcheezehmm is there a t.37 howto for asterisk?
15:22.08HmmhesaysWu Mart making waves as retailer in China. Tawget, Crostco, and Blest Bly not far behind
15:22.27MrChimpyhmm, I'm a bit confused about trunk config on my 4 port PRI ISDN cards
15:22.42Kisfinally... vacancy...
15:22.46MrChimpyI'm dialling on stuff like Zap/g1
15:23.05MrChimpywhich is fine, but I need it to find a line on any of my four trunk groups
15:23.08*** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.235.216.Dial1.SanJose1.Level3.net)
15:23.27MrChimpydo I do that in the Dial command, or should I just be defining one trunk group?
15:23.47robin_szok, be this about my poxy SNOM360 explaining ... i have  MWI and a retreive button .. when the MWI is unlit, I press retreive and it dials 8501 *which is what I need it to do* and it works .. when the MWI is flashing, I press retreive and the danm thing dials asterisk@192.168.1.1 ?? whys that ??
15:24.15CunningPikeMrChimpy: Try a single group= line before all your channel => lines
15:24.34filerobin_sz: configure the option in sip.conf to change it to 8501
15:24.50robin_szits in sip.conf?
15:24.52robin_szcoo.
15:24.54filevmexten
15:24.57filein the general section
15:25.08robin_szwow.
15:27.07CunningPikeGood to know, CodyC
15:30.36nokysomebody test the performance of MeetMe's Application ?
15:30.57feld_ok guys for hinting I am doing the following. can you please tell me if there's a one-liner that takes care of both? :
15:31.17CunningPikenoky: Check the list - I think someone has done some testing with concurrent users/concurrent conferences
15:31.21feld_exten => 2000,hint,${PHONE0}
15:31.21feld_exten => 2000,1,Macro(oneline,${PHONE0})
15:31.36asterboywhere do you start with Festival setup in *
15:31.42asterboythe docs are not so clear
15:31.43feld_can that be put in one line like 2000,1,hint,Macro(.... or will that fail?
15:32.06robin_szfile: cool. works. I didn t have that option even commented out in my sip.conf, thats probabyll how I missed it
15:32.08asterboyI have the app loaded, however, there is no festival.conf or festival.scm
15:32.13CunningPikefeld_: It will fail - you can only have one priority per line
15:32.14fileah
15:32.20feld_CunningPike, thanks!
15:32.21asterboythe source is in contribs
15:32.34C4T3lfestival uses up a lot of proc
15:32.54asterboyya but if I want to play zasterisk...I must have it.
15:33.32C4T3lhave you tried to run it at linux CLI? does it work there?
15:33.43asterboywhat is the syntax?
15:33.46*** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it)
15:33.51*** part/#asterisk CodyC (n=cody@207.200.23.194)
15:34.10robin_szshould I go around all 40 phones and configure them .. or is there someting I can do to * to make it less worried?
15:34.35C4T3lfestival <enter>  then type: SayText("hello world") <enter>
15:35.03asterboyno such command
15:35.42C4T3lhmm, then you need to install it :)
15:36.26asterboywhere are the docs on the install?
15:36.31robin_szsigh ... bring me a decent SIP termination
15:36.55*** join/#asterisk mogorman (i=ejabberd@68.62.237.103)
15:36.57asterboythis does not seem to be much help: http://www.voip-info.org/wiki-Asterisk+Festival+installation
15:37.44*** join/#asterisk burizaa (n=freeee@cm56.omega110.maxonline.com.sg)
15:37.51C4T3lasterboy: what distro you use?
15:38.06asterboylfs
15:38.11asterboy~festival
15:38.14jbotwell, festival is a general multi-lingual speech synthesis system developed at CSTR. See http://www.cstr.ed.ac.uk/projects/festival/, or festival lite a much more compact festival http://www.speech.cs.cmu.edu/flite/index.html
15:40.52*** join/#asterisk Neptune__ (n=foo@zux221-156-100.adsl.green.ch)
15:42.06asterboyman, too complicated....festival needs a hell of a bunch of requirements.
15:42.06*** join/#asterisk unixgeek (n=unixgeek@216-220-234-197.exploremaine.com)
15:42.12asterboyso much for zasterisk
15:45.01Neptune__what ports do i need to allow for *? 5060 and the range in rtp?
15:47.52*** join/#asterisk Mother (n=m@253.Red-88-12-225.dynamicIP.rima-tde.net)
15:47.58Mothergreetings
15:48.08Qwell[]Neptune__: yes
15:48.15Qwell[]Neptune__: unless you use anything besides sip
15:48.16Motheranyone know the payload type for Speex in RTP?
15:49.08fileyeekz, as a software developer putting together a desk I feel... silly
15:49.29Qwell[]file: pfft
15:49.29Motherconsider it bit assembly
15:49.40Qwell[]You have the Makefile, right?
15:49.43fileooh
15:49.45filemake desk
15:49.46Qwell[]and a compiler?
15:50.01Qwell[]./configure --with-cdrack
15:50.02Qwell[]such a nub
15:50.08Mother## unresolved dependency: size 10 spanner
15:50.27fileI'm using a Digium/Asterisk screwdriver :D
15:50.39Motherah! then you should be OK
15:50.40Qwell[]checking for phillips head screwdriver: yes
15:50.42TheCompWizMother  shoulda used yum.
15:50.45Neptune__Quell - would that be tcp or udp?
15:50.46Qwell[]checking for phillips head screwdriver usability: no
15:50.49Qwell[]Neptune__: udp
15:50.50Motherlol
15:50.55Motheryum install table
15:50.55Neptune__ok thanks
15:51.03Qwell[]BAH!
15:51.17Qwell[]USE="glass" emerge -u desk
15:51.32Motherhttp://www.iana.org/assignments/rtp-parameters  <- I don't see Speex here
15:51.50Motherso just looking for a creative idea before I set it as 1337
15:54.11rpmhow do allow my phones in asterisk to retain their caller ID and not have asterisk set it?
15:54.14Dr-LinuxQwell[]: a call needs good upload speed or download speed?
15:54.15fileI can't find a part
15:54.16filehrm
15:54.23Qwell[]Dr-Linux: both, it needs the same amount of each
15:54.33nokysomebody test the performance of MeetMe's Application ?
15:54.39Qwell[]file: ./configure --without-doorhinge
15:54.40nokyi don't found in the list
15:54.56Dr-LinuxQwell[]: i'm having some issues, not sure what do do
15:55.03Qwell[]~consultant
15:55.05jbotHire a consultant.
15:55.05Qwell[]:D
15:55.13Qwell[]Dr-Linux: What's the issue?
15:55.25Dr-Linuxheh :) i'm .. i'm not bad ;)
15:55.38Motherfile: I bet it's from Ikea
15:55.42filenah
15:55.43Dr-LinuxQwell[]: when i use GSM codec in the sip.conf
15:55.44fileOffice Depot
15:55.46Motherah
15:55.48Dr-Linuxi get loop :
15:55.48Dr-LinuxJun  8 08:18:24 WARNING[32223]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data
15:55.49Qwell[]file: worse
15:55.50Dr-LinuxJun  8 08:18:24 WARNING[32223]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data
15:55.53fileI think I'm missing side frames
15:56.00Motherthey tend to do the "missing part" thing rather often
15:56.01Dr-Linuxhundreds of time ..
15:56.04Motherthat's more serious then
15:56.05Qwell[]file: I once bought a chair, with two of the same arm
15:56.11Qwell[]so, like...it didn't fit, heh
15:56.13fileneat
15:56.41Dr-Linuxalso, sometime i hear like someone is talking from understand water
15:56.46Motherthe other day I saw a friend got a bookcase - required items as per the instructions: a screwdriver, a hammer and a friend
15:56.57Dr-LinuxQwell[]: do you think it's RTP or bandwidth issu?
15:57.00Motherso if you had no friends you had to return the bookcase
15:57.03Qwell[]Dr-Linux: neither
15:57.08Dr-LinuxQwell[]: my US clients are just fine with the same server
15:57.15Qwell[]fun
15:57.34Dr-LinuxQwell[]: neither?
15:58.09*** join/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu)
15:58.27*** join/#asterisk Kerry_G (n=Kerry_G@ip70-187-129-227.oc.oc.cox.net)
15:58.33filewait wait
15:58.34Kerry_Ganyone know of a way to listen to voicemail when sent as email to a blackberry?
15:58.39fileI think I know where they are
15:58.45Qwell[]file: in the box?!
15:59.23*** join/#asterisk mut (n=animenod@65.111.222.120)
15:59.26mutyay services!
15:59.31HmmhesaysKerry_G: you mean you can read audio data as a text file and decode it in your brain?
15:59.34Hmmhesays*can't
15:59.47nokywhere can i found the last bugs of asterisk ?
16:00.28*** join/#asterisk lorinc (n=ang@caracas-0343.adsl.interware.hu)
16:00.40Qwell[]noky: bugs.digium.com
16:00.59Dr-Linuxfile: do you have any clue why i'm getting this warnings ?
16:01.56filegot them!
16:02.07fileDr-Linux: I'm busy putting together a desk, please heave your message after the beep
16:02.09Qwell[]file: where were they?  in the desk?!
16:02.12fileer leave
16:02.20Qwell[]heave your message?  heh
16:02.28fileQwell[]: well there's this foam thing... with a big glass top on top, and under neath there's all sorts of stuff
16:02.33Qwell[]ooo
16:02.35Dr-Linuxfile: ok, i'll catch  you next time
16:03.36swytchq about dtmf.  i set dtmfmode=info in sip.conf [global] however, in all SIP messages (100,180,183 and 200) from asterisk i dont have Allow: INFO.  In the INVITE that i sends i have...
16:06.06Kerry_Gno no no, I just want to play the sound file attachment
16:06.12CunningPikerpm: Just make sure there is no callerid line in sip.conf......
16:06.22CunningPikeKerry_G: Blackberries can't do that.....
16:06.24Kerry_Gand blackberrys dont recognize the default format
16:06.44swytchanyone have a hint why i may not get "Allow: INFO" from asterisk when i set dtmfmode=info   ?
16:07.06Kerry_Git sees the attachment, but wont open it
16:07.22rpmCunningPike: and it will take the callerID from the phone?
16:08.07CunningPikerpm: Yes - we want it the other way around, so I know that's what happens :D
16:08.46CunningPikerpm: All this assumes that your PSTN connection allows you to set your own CID
16:08.49*** join/#asterisk Alric (n=nbowyer@wireless-062.1stel.com)
16:09.37*** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de)
16:10.14*** join/#asterisk bancus (n=treed@static-71-160-206-211.lsanca.dsl-w.verizon.net)
16:10.14Neptune__why am i always getting a error on the CLI about failed registration for the local sip peers? - it always gives me a "username/authname mismatch"
16:10.36*** join/#asterisk pa (n=paolo@unaffiliated/pa)
16:10.51zoabecause you are using the wrong username / pass
16:10.53zoaor none at all
16:10.54bancusQuick question: I'm designing a PBX system for a company. (Actually three closely related companies, so there'll be multiple phone numbers involved.) Can anyone recommend good reliable PTSN<->SIP/IAX2 providers for the US?
16:11.11bancusI use broadvoice for our own company's PBX, and occaisionally have issues with them.
16:11.14Kerry_GNexVortex, Teliax
16:11.58Neptune__zoa - no im positive that it is correct
16:13.42swytchquit
16:14.08rpmCunningPike: yes it does.
16:14.12rpmCunningPike: thanks
16:14.20zoano its not
16:16.51Dr-Linuxanthm: active? :)
16:16.56anthm?
16:17.38*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220)
16:17.41Dr-Linuxanthm: as i reported my last day GSM codec problem to you, do you remember?
16:17.53Dr-LinuxJun  8 08:18:24 WARNING[32223]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data
16:18.26Dr-Linuxso had changed the codec in the sip.conf andi couldn't see this warning again ..
16:18.38*** join/#asterisk Assid (i=assid@203.115.83.214)
16:18.45Dr-Linuxbut you were something right, it looks RTP bad packets/link problem
16:19.23Dr-Linuxanthm: we are facing very bad voice quailty problem, like someone is talking from understand the water etc
16:19.38Dr-Linuxanthm: my US users are very fine with the same server
16:20.11Neptune__zoa i tried this a couple of times - the username/password in the sip.conf is the same as in the config of the phone
16:20.28zoacheck for other users in sip.conf that dont have a pass
16:21.22Dr-Linuxanthm: i was told to recompile my asterisk, but i'm doubted that it's my network/bandwidth problem
16:21.55*** join/#asterisk tanvalmg (n=mguevarr@67-134-234-194.dia.static.qwest.net)
16:21.59zoarecompling asterisk probably will not work
16:22.06zoai'd say go for ethereal
16:22.09zoaand look how the packets look
16:22.13zoasee if they are the same size
16:22.23zoaand they come every 20ms
16:22.31zoaand the timestamps look reasonable
16:23.23brettnemhey anyone know if there are problems with sending RTP over bonded pipes.. like bonded T1s, MPPP, or bonded SDSL lines?
16:23.34emrahexit
16:23.52zoaexit yourself :p
16:23.53anthmwhat is the call connected to on the other end
16:24.36CunningPikerpm: Great - glad it worked
16:24.47brettnemSession Border Controller->riverstone router->Asterisk->Riverstone Router->Paradyne DSLAM->Ethernet CPE->Polycom
16:26.00tanvalmgHi, we're having problem installing our Digium TDM400P card on our IBM x346 server.  We are getting kernel messages of "Uhhuh.  NMI received for unknown reason 25 n CPU 0....Dazed and confused, but trying to continue...Do you have a stange power saving mode enabled?"
16:27.01tanvalmg...when we do a "modprobe wcfxs" the system reboots
16:29.20*** join/#asterisk CodyC (n=cody@207.200.23.194)
16:29.47Dr-Linux[TK]D-Fender: i just verified that's not asterisk problem, that's our DSL provider's problem
16:30.00CodyCanyone had a problem with a sip phone's dialplan screwing up an outbound call?
16:30.37Dr-Linux[TK]D-Fender: but odd thing is that, all other things work fine, but only voice stuff have problems, not only with asterisk, even with all our other voice devices.
16:31.04*** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.235.216.Dial1.SanJose1.Level3.net)
16:31.04Neptune__i now got the grandstream phone to seamingly be registered - it shows when i do a "sip show peers" on the * cli - but the phone's status page says it isnt registered
16:31.10Neptune__could this be a firewall issue?
16:31.36*** join/#asterisk JakBeatZ (n=JakBeatZ@trek.tor1.ebit.ca)
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16:34.58*** join/#asterisk iq (n=iq@71-215-63-190.omah.qwest.net)
16:35.54JakBeatZHi Folks, I'm having an issue with a new asterisk box I'm setting up.  I have it set up like this (PSTN)-- [ASTERISK1] --(IAX)-- [ASTERISK2] --(IAX)-- [ASTERISK3].  I have a number that's answered on Asterisk 1 via PSTN, it forwards it to asterisk 2 and I just setup IAX beteeen Asterisk 2 and Asterisk 3 and told Asterisk 2 to send the call to Asterisk 3, but for some reason, when I call I see this message on astterisk 2 "Jun  8 12:30:35 NOTICE[52
16:35.54JakBeatZ77]: chan_iax2.c:7183 socket_read: Host (asterisk 1 IP) failed to authenticate as (asterisk 3 IP)".   I'm confused as to why asterisk asterisk 1 is trying to authenticate to aseterisk 3?!  Is there some sort of proxy action I have to setup, somewhere?
16:39.18Assidheya tkd
16:40.32Assidokay question on dialplan for poly501..  when a user dials the dialplan.. the last digit is being dialled when they dial with another prefix besides 1xxxxxxxxxx
16:41.35salviadudwhy send it to asterisk 2?
16:41.43salviadudyou can send it directly to asterisk 3
16:41.53salviadudlooks kinda pointless
16:42.01JakBeatZsalviadud:   because I don;t own asterisk 1
16:42.20salviadud:(
16:42.30JakBeatZand I'm just doing some testing at the moment so I don't want to swing the DIDs around until I'm done the testing
16:42.41JakBeatZso that's why I just thought I could do the multi-iax hops
16:43.07*** part/#asterisk Kerry_G (n=Kerry_G@ip70-187-129-227.oc.oc.cox.net)
16:43.25*** join/#asterisk jtodd (n=jtodd@ti.fox-den.com)
16:46.59*** join/#asterisk littlejohn (n=little@host31-85.pool877.interbusiness.it)
16:48.22rajiv|workthis "hacker" who resold voip service ... anyone know which company it was?
16:48.55*** join/#asterisk anonymouz666 (i=anonymou@200.218.196.5)
16:49.26dlynes_homerajiv|work: huh?
16:50.00dlynes_homerajiv|work: to what are you referring?  I don't see any mention of hacker for at least the last three screens or so
16:50.21rajiv|workdlynes_home: a slashdot article, nto mentioned here int he channel
16:50.30dlynes_homeah
16:50.30rajiv|worksorry i wasnt clear
16:50.56*** join/#asterisk JINDAL (n=root@220.226.36.2)
16:50.57dlynes_homeAnd when you say hacker, do you mean the media definition, or some other definition?
16:51.20rajiv|workmedia
16:51.30dlynes_homeah...no idea about that then
16:51.33rajiv|workhttp://it.slashdot.org/article.pl?sid=06/06/07/1949258
16:52.22dlynes_homeheh
16:52.23dlynes_homecool
16:53.07JINDALhi guys...
16:53.10dlynes_homedamn...that 1M would just barely get him a piece of property here
16:53.21dlynes_homehe wouldn't have any money left over for a boat, and two cards
16:53.26dlynes_homes/cards/cars/
16:53.38*** join/#asterisk meshuga (i=meshuga@c-71-231-139-8.hsd1.or.comcast.net)
16:55.26nokywhere can i found the last bugs of asterisk ?
16:55.33meshugaanyone got a moment to look at a bizaare error?
16:56.10JINDALam looking to asterisk as a solution for 500 people call centre..... but dont hav an idea abt the kinda hardware required...... any ideas
16:56.22meshugaJINDAL 3 servers
16:56.38meshuga1 SER box and 2 asterisk boxes at minimum
16:56.50meshugahttp://pastebin.com/767696
16:56.54dlynes_homerajiv|work: damn..that guy had a good score going on
16:57.00meshugaso i am using DISA and trying to dial out using voipjet
16:57.03rajiv|workdlynes_home: heh ya
16:57.04meshugait works fine from my sip client
16:57.11JINDALokey... am supposing to handle 100+ concurrent calls
16:57.19rajiv|workdlynes_home: but the article doesnt say what his "company" name was
16:57.25meshugabut when i dial out using another SIP extension (ie remote dialtone) it fails
16:57.33dlynes_homerajiv|work: bet he really pissed off those ten carriers though :)
16:57.41meshugaJINDAL * can handle 300ish on good hardware. this is a pci bus limitation, not an asterisk limitation.
16:57.59meshugawell, a bus limitation, not necessarily pci :)
16:58.23dlynes_homerajiv|work: but 500K calls doesn't amount to the amount of money they're insinuating, either, unless they're all international calls
16:59.01JINDALokey... meshuga, can ye elabarate on good hardware / ser boxes or pass a link ->
16:59.08dlynes_homerajiv|work: and even then, if that was the case, he would've made a lot more than $1M
16:59.48*** join/#asterisk phin (n=mcgee@net2.netexp.com)
16:59.52phinhello
17:00.26phini was wondering, before i install, and taking hardware into consideration still, would people recommend asterix on a ubuntu/debian based system, or freebsd?
17:01.13[Airwolf]phin, depends.
17:01.25[Airwolf]Are you planning on using any special hardware ?
17:01.33meshugaJINDAL thats too broad of a question to even answer. have you done any experimenting with * yourself?
17:01.46[Airwolf]As far as I know, zaptel drivers aren't avalible voor bsd.
17:01.59phin[Airwolf]: ahh, ok, thanks
17:02.06viperdudehi, anybody got any ideas on how to monitor asterisk boxes and how to send a email if there is a problem with them?
17:02.11meshugaanyone have any idea why voipjet would work from a SIP extension, but not thru DISA with remote dialtone? http://pastebin.com/767705
17:02.11phinwell i suppose i'll go the ubuntu or debian route
17:02.28meshugaviperdude: same with you monitor any other unix server.
17:02.45meshugas/with/way
17:02.50[Airwolf]phin, nothing against ubuntu. But Ubuntu is kinda new.
17:03.07viperdudeok so is the a nagios plugin or the like?
17:03.18phinright
17:03.19viperdudeor jffnms
17:03.28meshugaviperdude: exactly.
17:03.37meshugaor bigbrother or any network monitoring tool
17:03.48viperdudethey can monitor SIP?
17:03.55meshugaheh
17:04.01JINDALyup am on it presently..... and wanna buy hardware to start full time testing.............. just suggest me the minimum hardware i should blindly go for at this moment (500 people arnd 200+ concurrent PSTN calls)
17:04.10meshugathose apps monitor services by telnetting to the port.
17:04.11Nuggettelnet is eeeeeeevil!
17:04.26viperdudeok
17:04.36meshugathats a perfectly acceptable way of testing connectivity.
17:04.52*** join/#asterisk S4w (n=saw@adsl-3-138-152.mia.bellsouth.net)
17:04.53meshugaJINDAL: my old employer use to run asterisk on sgi altix hardware.
17:05.01meshugawell, they probably still do.
17:05.17*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
17:05.28S4whey guys, with an x100p can I pickup the call waiting and transfer the first call and the second call to different sip Phones?
17:05.33LokeshIndianJINDAL: On digium website they mentioned about compatibel hardware with zaptel cards..better look there
17:06.07JINDALoks
17:06.09meshugaLokeshIndian he never said he was using t1 boards.
17:06.15meshugaJINDAL what are you using for transport?
17:07.13LokeshIndiani bought wrong server to work with TE405P card
17:07.23meshugate405p isnt going to get close to 200 calls.
17:07.25JINDALgigbit ethernet inside the call centre... not decieded on anything else
17:07.47meshugaJINDAL heh, that doesnt matter.
17:07.55meshugayou could run 10mbit switched and be fine.
17:08.03meshugathis is voice, not streaming HD video. :)
17:08.22meshugaJINDAL you need to decide what your transport is
17:08.30JINDALhehe!!
17:08.32meshugawhat you plan onsupporting 200 i assume PSTN inbound calls on
17:08.38meshugathat matters more then the pc ever will.
17:08.38JINDALGSM/SIP
17:08.42S4wis there any way to disable callwaiting on x100p FXO so that the second caller doesnt hear and endless ringing?
17:08.57meshugaso gsm is magically going to appear on the asterisk box? no conversion process needed?
17:09.06meshugano sip provider with hordes of internet?
17:09.09*** join/#asterisk feld_ (n=feld@12.148.212.157)
17:09.18meshugai would be scared to provide 200 sip calls on voip
17:09.22[Airwolf]dlynes_home, have you ever worked with Diva isdn cards ?
17:09.27meshugaS4w: did you try callwaiting=no?
17:09.31meshugain the zaptel.conf.
17:09.38meshugaor zapata.conf, i dont remember
17:10.08S4wmeshuga: it makes the callwaiting sound on the phone anyways and the caller still hears the reinging and no answr ofcourse
17:10.09JINDALya am looking towards T-1/E-1 digital lines or subscribe to voip providers whichever be more economical
17:11.08meshugaJINDAL so, what hardware are you going to use? even a te405p full of t1s isnt going to get close to 200 concurrent
17:11.09meshugaheh
17:11.11S4wmeshuga: if asterisk would send a busy tone I would be happy, is there any way of doing this?
17:11.14meshugaits not about economical
17:11.23meshugaits about quality and reliably.
17:11.26meshugareliability.
17:11.30meshugaS4w: yes. but i dotn know how.
17:12.05meshugaJINDAL: i think i'd worry about your inbound far before the servers
17:12.14meshugaand how the hell youre going to bring in 200 calls concurrently
17:12.14JINDALya can go more than one te405p ......... dats wat i need to deciede abt what and how much do i need
17:12.20meshugaJINDAL you can?
17:12.40meshugayou're going to put 4 in a pci
17:12.44meshugaand then buy 16 t1s?
17:12.46meshugaand not a t3?
17:13.16meshuganot counting i've never seen a te405p work completely full reliably
17:13.20meshugabut maybe thats just me.
17:13.28DagoBlokwhat is t1/t3 actually ?
17:13.28meshugamuch less multiples of them :P
17:13.36LokeshIndianIn India I guess there is no T3
17:13.51JINDALhmm i may go for t3 if its availiable
17:14.01LokeshIndianif JINDAL is setting up this in India
17:14.20meshugaDagoBlok t1 is 23 voice channels, 1 data. or its 1.544mbit data, typically a PRI. t3 is 45mbit, dunno how many voice channels
17:14.21JINDALyup in india
17:14.26meshugaJINDAL: usign what hardware ?
17:14.34DagoBlokoh ok
17:14.53DagoBloksome sort of internet connection then
17:14.58meshugaDagoBlok: asterisk uses t1s for voice, its essentially 23 phonelines terminating, but you can assign multiple dids
17:15.01meshugano.
17:15.02JINDALam looking for all the options presently.......
17:15.17DagoBlokahh
17:15.25meshugahttp://en.wikipedia.org/wiki/Digital_Signal_1
17:15.26DagoBlokpstn
17:15.35meshuganot necessarily, but usually.
17:15.47meshugato get pstn termination reliability 90% of pbxes use t1s
17:15.50meshugahell i'd say 99%
17:15.57JINDALmost probably i hav to go on T-1s and then scale up
17:16.01DagoBlokthey use it in europe too?
17:16.01meshugaits the traditional way of sharing multiple lines
17:16.09blitzrageeurope uses E1
17:16.11DagoBloknever heard of it before
17:16.12meshugaDagoBlok : e1 in europe
17:16.12DagoBlokoh ok
17:16.14meshuga2mbit, same idea.
17:16.18blitzragewhich has more channels than T1s
17:16.20meshugaits fairly old
17:16.29meshugalike 20
17:16.30blitzrage30B+2D if I remember correctly
17:16.32meshuga20+ years
17:16.33*** join/#asterisk cekc (n=cekc@rrcs-24-199-36-210.west.biz.rr.com)
17:16.40*** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252)
17:16.55meshugaJINDAL: good luck on getting a te405p to handle multiple t1s, and then getting multiple te405ps working in one machine.
17:17.05meshugayou got many hours of pain and anguish.
17:17.21bancuswhy the hell can't these sip providers list rates on their website?
17:17.38meshugawho doesnt?
17:17.44meshugaall the us ones i use do
17:17.51bancuswho do you use?
17:17.52meshugaso, anyone use voipjet/DISA much?
17:17.54JINDALshould i use multiple te405ps on one machine......... i think its better to go for multiple servers
17:17.58blitzrageif they don't list prices, don't use them
17:18.02meshugaif so, please take a look at this
17:18.03meshugahttp://pastebin.com/767705
17:18.08bancusblitzrage: that's about what I think
17:18.17bancusthe few I've found that do list prices are kinda expensive
17:18.24meshugaJINDAL: thats assuming you can get a te405p to work reliably singly...
17:18.35bancusMy company uses Broadvoice at $35/mo, which is cheaper than anything I've seen listed so far.
17:18.47meshugawhat?
17:18.49meshugawww.axvoice.com
17:18.51meshugawww.telasip.com
17:18.52bancusBut I can't recommend them to the company I'm building this PBX for, because of intermittant auth issues.
17:18.59meshuga$35/mo is pretty expensive.
17:19.14bancusFor business?
17:19.18meshugaunless youre going legit and not using residental plans for business :P
17:19.22bancusHahaha.
17:19.50meshugabancus i dont recommend voip for any companeis i do buildouts for.
17:19.52bancusmeshuga: you use both of those?
17:19.59meshugaexcept when interconnecting offices
17:20.00meshugayes.
17:20.01bancusmeshuga: Working on a budget here.
17:20.08meshugai use voipjet, freedigits, fwd, etc
17:20.25meshugabancus: hopefully the business dont rely on phone service then.
17:20.32bancusVoip's been working fine for my company, except when the LA proxy stops authenticating us.
17:20.41meshugawow you're lucky
17:20.48meshugayou shoulda been a broadvoice cust last year
17:20.52bancusI was.
17:21.01meshugaduring gblx issues?
17:21.08meshugawhen they decided not to pay their bill for 2 years?
17:21.09bancusAlthough we didn't use it much until August/September.
17:21.15bancus:o
17:21.22meshugaand they still aren't paying their level3 bill rumor has it?
17:21.30bancusurgh
17:21.51meshugai've used voip for nearly 8 years
17:22.14meshugaand i still can't recommend it to companies for PSTN termination.
17:22.26bancusMy only problem with broadvoice so far is that occaisionally our local proxy will refuse our auth for about four hours and I'll have to switch to a proxy on the other side of the country, which introduces a bit too much lag.
17:22.53bancusAnd they've only got the one west coast proxy.
17:22.55*** join/#asterisk mtaht4 (n=m@207.47.5.58.static.nextweb.net)
17:23.01meshuganot bad
17:23.17meshugai cant say i've had that much luck with BV.
17:23.35bancusSo you really recommend against it, then?
17:23.35meshugaespecially when it took me almost 2 months to release a block of 30 dids from them
17:23.37JINDALmeshuga,what kinda issue will i face wil te405p as i suppose them to completely compliant wid asterisk as they are made by digium
17:23.53meshugaJINDAL: good luck.
17:23.55bancusThey're looking at getting three lines in.
17:24.02meshugabancus: use copper.
17:24.04meshuga3 lines aint shit.
17:24.15meshugai use copper for inbound on small setups like that
17:24.17bancusHoping to save them money.
17:24.18meshugaand voip outbound for the LD
17:24.20JINDAL:)
17:24.21meshugaheh
17:24.29bancusBut then I have to get a special card to have it go into the PBX?
17:24.32meshugaputting in an ip pbx isnt saving enough?
17:24.41meshugayes, you need something with FXO ports
17:24.46meshugaweather its a pci card or a breakout box
17:24.48bancusHow much do those run?
17:24.57meshugax100p is $15 from ebay.
17:25.00meshugafor 1 line.
17:25.06bancusWell, they don't have a PBX as it stands.
17:25.09bancusThey're about to expand.
17:25.18bancusWithout me, they'd probably be fine just having a bunch of different phones.
17:26.02meshugaheh well, if thats a case, let them go that route
17:26.03meshugaif you are comparing against that, you will always lose.
17:26.03bancusHeh.
17:26.03meshugaasterisk outperforms any centrex or key system solution easily
17:26.04meshugabut nitpicking on cost will always make you look bad.
17:26.13bancusOh, they're not nitpicking on cost.
17:26.17meshugaoutperforms/more cost efficient/etc
17:26.29bancusAs it stands, the system will probably come out well under-budget.
17:26.35bancusEven if I have to get some FXO ports.
17:26.44meshugabancus well, roughyl a business line should run about $25/mo, so you're look at, say, $120/mo for telecom stuff.
17:26.57meshugawhich woould be the exact same they would pay for junch a bunch of phones
17:27.01meshugawith alot of added functionality.
17:27.13bancusYeah.
17:27.20meshugai try to convert companies to pure voip ever.
17:27.20bancusThere's the added expense of the FXO card.
17:27.24meshugaheh
17:27.32bancusI wasn't sure how much they'd run.
17:27.43meshugai dont consider $45 an added expense, but maybe thats me :)
17:27.44bancusAnd SIP's been working reasonably well for my company.
17:27.45*** join/#asterisk MatsK (i=MatsK@83.233.97.229)
17:27.50meshugayoure lucky.
17:27.52bancus$45 is for which card?
17:27.53meshugathats all it is.
17:28.01meshugaas i stated previously, x100p is $15 each
17:28.06meshugatherefore 3x$15 == 45$.
17:28.15bancusI doubt the box could fit three PCI cards.
17:28.32meshugawell, then spend $200 on the sangoma expandable ones
17:28.41meshugaer i mean $500
17:28.51meshuga(i've had alot better luck with sangomas then digium hardware)
17:28.55meshugasmoke break brb
17:29.31pdavidis a hardware timer required for meetme & moh?
17:30.56DagoBloki heard the sipura spa-3000 is excellent
17:31.19DagoBlokmuch better than a x100p
17:31.51*** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane)
17:32.26pdavidwhat options should i look at if i dont have hardware timer, and no uhci-usb
17:32.28pdavid:(
17:32.56dlynes_homeDagoBlok: x100p doesn't have fxs, and it's also a total piece of crap
17:33.03dlynes_homeDagoBlok: so it wouldn't take much to be better than it
17:33.11DagoBloklol, ok :)
17:33.17DagoBloknever tried the x100p :)
17:33.21MACscrWhat do you think is better for remote agents, ATA's or Voip Phones?
17:33.27dlynes_homebut yeah, the spa-3000 is ok
17:34.06bancusdoes digium not sell the x100p anymore? I don't see it on their site
17:34.13*** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler)
17:34.19meshugapdavid no you can use the module ztdummy.
17:34.33pdavidmeshuga: i am having a hard time compiling the darn module, though
17:34.43bancusaha, apparently it was replaced by the tdm400p
17:34.48meshugauh what?
17:34.49bancusanyone have any experience with that?
17:34.52meshugayes.
17:35.06meshugayou only use ztdummy if you have no zaptel hardware.
17:35.23pdavidright, but i cannot get ztdummy to compile for some reason
17:35.28meshugaif you use the hardware, you'll use hardware timer and don't need to use the RTC, which is what ztdummy does
17:35.36pdavidmeshuga: right, got it so far
17:35.48pdavidmeshuga: i am trying to compile the module from svn snapshots
17:35.53meshugauhm, if you can't get ztdummy to compile, you got system/distribution/os issues :)
17:35.54meshugawhy
17:35.56*** join/#asterisk cytrak (n=btorch@208.63.19.179)
17:35.57meshugause releases.
17:35.59dlynes_homebancus: lotsa peeps on here using the tdm400
17:36.14meshugabancus: buy a x100p from ebay
17:36.18meshugai've bought tons of clones
17:36.22meshuganever had a problem with one
17:36.48bancusAs I said, I don't think the system has 3 open PCI ports.
17:36.51cytrakdoes anyone know how I can connect my asterisk with a good voip provider? I understand I have to create a peer register
17:36.58bancusI'd prefer a card that could supply all three in one.
17:37.12meshugathen you're going to spend $500+
17:37.15bancus:/
17:37.46meshugabuying a el-cheapo computer (since x100ps are less picky about dedicated irqs then t1 boards)
17:37.54meshugawould be more cost efficient
17:37.56cytrakany suggestions on a good voip provider that can give me say a block of voip extensions that can be redirected to my asterisk box.. I guess i'm looking for a getaway provider ?
17:37.57meshugabut less of a path to ugprade
17:38.23meshugacytrak: sounds like you need to do some reading. http://www.nerdvittles.com/ is suggested.
17:38.51meshugahttp://nerdvittles.com/index.php?p=130
17:38.53cytrakplease point me where to do the reading
17:38.54meshugathis is one of my favorite pages
17:39.05dlynes_homemeshuga: i've found the opposite...the x100p's will share an irq, but if the irq is shared, after a period of time, it ceases to register with the driver
17:39.10*** join/#asterisk r_evolution (i=_evoluti@208.251.203.246)
17:39.18meshugacytrak: the voip provider reviews and config tips
17:39.46dlynes_homemeshuga: with my four port pri card, it was sharing an irq, and it never ceased to register with the driver
17:39.47meshugadlynes_home: I guess it matters whats sharing it. but i can definitely say i've had much less issues with irq sharing on x100p's then tdm boards
17:40.04dlynes_homemeshuga: however, both of them would drop calls periodically because of the interrupt sharing, too
17:40.10meshugai havent used a t1 board in a year, so maybe they got better
17:40.28dlynes_homemeshuga: our t1 card is about 3 or 4 years old
17:40.29meshugait all depends how the mobo is designed
17:40.35meshugaand how irq steering on the mobo works
17:40.42meshugaand if ACPI is trying to take it over, etc
17:40.44dlynes_homeah
17:40.51dlynes_homeyou mean APIC?
17:41.03*** join/#asterisk moprilo (n=mop@201.198.78.23)
17:41.06meshugai've had supermicro/tyan boards fail because they implemented their own technology
17:41.15CunningPikeQuick hints question: would this work? exten=> 1234,hint,SIP/1234&SIP/2345&SIP/3456?
17:41.17meshugayea, which is controlled by APCI still it hought
17:41.21cytrakmeshuga, cool thanks
17:41.30*** join/#asterisk Blake0PS (n=blake@c-66-41-195-142.hsd1.mn.comcast.net)
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17:41.34meshugacuz if you dont load ACPI modules, APIC doesn't load either.
17:41.50meshugaCunningPike should with a dial line. i dunno how your hint macro is setup though
17:41.53r_evolutionHAH!
17:41.54r_evolutionSCORE!
17:41.55dlynes_homeAPIC == advanced programmable interrupt controller; ACPI == something to do with power management
17:41.59meshugai'd replace hint with dial
17:42.04dlynes_homeCunningPike: yes
17:42.14CunningPikedlynes_home: Thanks :D
17:42.17r_evolutionhey dlynes... remember how i was in her ethe other day saying about how in 1.2.8 asterisk crashed out when you tried to park a call?
17:42.22dlynes_homeCunningPike: i use something almost exactly like that
17:42.24r_evolutionwell apparently it got fixed in 1.2.9.1
17:42.25r_evolutionSCORE!
17:42.40CunningPikedlynes_home: Perfect. And how come you're at home?
17:42.44dlynes_homer_evolution: heh...that's why you should always use the latest version before reporting a bug
17:42.50dlynes_homeCunningPike: just on my way into the office
17:42.52r_evolutionwell
17:42.57r_evolution1.2.8 WAS the latest version at the time
17:42.57moprilohi, i just upgraded from 1.0.9 to 1.2.9,  and when i transfer a zapincomming call from a sip-phone to a zap-fxs-phone it answers and automaticly hangups any idea_]?
17:43.00meshugadlynes_home: stands for advanced configuration and power interface. replaces pnp on isa
17:43.03CunningPikedlynes_home: Well, make it snappy ;)
17:43.04dlynes_homeCunningPike: I was at a job site until 11pm or so last night
17:43.04r_evolutionthe day after i bitched about the issue in here for a good hour
17:43.04meshugahttp://en.wikipedia.org/wiki/ACPI
17:43.09r_evolutionthey released 1.2.9.1
17:43.13dlynes_homeCunningPike: all the way the hell out in surrey, too
17:43.15meshugapower management is common, yes, but it also does device config and all that kazz
17:43.17meshugajazz
17:43.22CunningPikedlynes_home: Surrey == hell
17:43.23meshugai've never seen APIC work without ACPI.
17:43.30meshugaso i assumed APIC was a subset of ACPI.
17:43.42dlynes_homemeshuga: you need to enable SMP in the kernel before you can enable APIC
17:43.55dlynes_homemeshuga: even if you're on a uniprocessor system
17:43.58meshugauhm what?
17:44.01meshugathen i never use APIC :)
17:44.02*** join/#asterisk _DAW (n=bob@adsl-150-59-108.msy.bellsouth.net)
17:44.04meshugai only use ACPI
17:44.11meshugawhich handles the pci steering and timings on most mobos
17:44.15dlynes_homeyeah...ACPI is the new improved replacement for APM
17:44.17meshugaat least that follow the intel design.
17:44.29meshugadlynes_home: as the wikipedia link stated, its much more then that.
17:44.41dlynes_home~wiki acpi
17:44.50dlynes_home~wiki apic
17:45.00SplasPoodDoes anyone here have the gizmo client talking to an asterisk box?
17:45.01meshugabut i never enable SMP
17:45.05meshugaand i see APIC messages
17:45.08meshugaso i dunno if thats right
17:45.37dlynes_homeanyways
17:45.41meshugadamn, they are beyond dmesg
17:45.43dlynes_homei need to head into the office
17:45.48dlynes_homedmesg | grep APIC
17:46.01meshugaheh as i said, its beyond what is in dmesg
17:46.02meshugaso i cant
17:46.02dlynes_homethat'll tell you if you have APIC enabled or not
17:46.06meshugano it wont.
17:46.15meshugabecause dmesg has all kernel messages, and they've scrolled off :)
17:46.24meshugathats what i mean when i say 'its beyond dmesg'
17:46.29meshugasince its a finite buffer
17:46.33dlynes_homebut if you grep it
17:46.39dlynes_homeyou can see it
17:46.45dlynes_homedmesg is a command at the linux prompt
17:46.56meshugano.
17:47.09dlynes_homemeshuga: then you're using some really weird distro
17:47.13meshugadude
17:47.15dlynes_homeevery distro i've used has that command
17:47.15meshugayou dont understand.
17:47.22meshugaof course i have dmesg
17:47.28meshugaas i stated, dmesg has a finite buffer
17:47.37meshugaand those startup messages have scrolled out of the buffer
17:47.40meshugatherefore, i can't grep it.
17:47.47meshugaits common when your machine is up for more then a few days
17:47.52meshugaand you use alot of apps that report to the kernel
17:47.55meshugasay, smb.
17:48.01meshugasince smb errors so much
17:48.01Blake0PSOne of my Zap channels has been failing and I'm wondering which error is the culprit. "WARNING[811]: zt hook failed: Device or resource busy" or "WARNING[4394]: CallerID returned with error on channel 'Zap/3-1'"
17:48.49meshugadmesg is a great utility
17:49.01*** join/#asterisk oej (n=oej@apollo.webway.se)
17:49.02meshugabut you can rely on it accurately to report hardware on a machine that isnt freshly booted up.
17:49.06meshugacant^
17:49.16meshugathats why you should do *.kernel in syslog
17:49.39*** part/#asterisk liran_ (n=Coll@212.199.177.203.static.012.net.il)
17:49.47TheCops[TK]D-Fender?
17:49.50TheCopsThere ?
17:49.58*** part/#asterisk ramo (n=ramo@59.92.133.168)
17:50.00meshugaUse a buffer of size bufsize to query the kernel ring buffer.  This is 16392 by default.
17:50.01meshuga<PROTECTED>
17:50.01meshuga<PROTECTED>
17:50.02meshuga<PROTECTED>
17:50.18meshugaso you only get 16392 characters.
17:51.14MACscrdo you guys like packet8?
17:51.27*** join/#asterisk Bullseye_Network (n=Kyle@216.143.192.69)
17:51.54coppicesad sad company packet8. it has lost its investors a large fortune
17:52.06mishehual teshageA et hasechel
17:52.07mishehuheh
17:52.30meshugacoppice: welcome to the wonderful world of telecom.
17:52.30*** join/#asterisk ToTo (n=ToTo@host68-166.pool879.interbusiness.it)
17:52.38*** join/#asterisk schirpich (n=dvs@ip21.farheap.net)
17:52.40MACscrany better providers that offer unlimited plans and work with asterisk?
17:52.52MACscri have 3 lines with vonage
17:52.59MACscrbut obviously i cant use them =(
17:53.11schirpichIs there a different context for the "first in line" hold music versus all the other hold music contexts?
17:53.43SplasPoodhrm, gizmo doesn't seem to wanna talk to asterisk at all
17:53.57meshugaMACscr: theres quite a few, but better is too subjective
17:54.16meshugaMACscr: axvoice. telasip. broadvoice. dialpad. just to name a few
17:54.39meshugaschirpich: not by default
17:54.47MACscrthanks mesh
17:54.50mishehumeshuga: like I said, al teshageA et hasechel ;-)
17:54.53meshugayou should be able to script it though, since queue participation is a variable
17:55.09meshugamishehu: i dont speak yiddish/hebrew/whatever
17:55.19meshugaso you can stop trying to speak to me in your native tongue :)
17:55.25mishehumeshuga: hebrew.  and your nick is hebrew so I thought I'd try it out
17:55.37meshugayea i pulled it out of a book 10+ years ago :)
17:55.45mishehumeans you're nuts.
17:55.57schirpichmeshuga: say there's 10 people in a queue.  callers 2-10 can hear hold music.   caller 1 hears nothign but silence.   Where is this specified?
17:56.00meshugasenseless is my preferred translation
17:56.03mishehunot totally crazy, but nuts.
17:56.11schirpichif its not by default?
17:56.14meshugaschirpich: sounds like a configuration issue
17:56.21meshugathats not default configured.
17:56.24RF_MIAqueues.conf
17:56.26meshugabut i havent looked at queue stuff since 1.0.2
17:56.29meshugaso maybe i'm wrong
17:56.37*** join/#asterisk KranZ (n=user@sme.bestline.net)
17:56.37*** part/#asterisk KranZ (n=user@sme.bestline.net)
17:56.39*** join/#asterisk KranZ (n=user@sme.bestline.net)
17:56.52meshugamishehu: what exactly is the difference in hebrew language?
17:56.58meshugabetween nuts and totally crazy
17:57.05mishehumeshuga: meturaf is totally crazy.
17:57.19meshuga<mishehu> meshuga: meturaf is totally crazy.
17:57.20meshugaer
17:57.51mishehuitchy mouse finger eh/
17:57.53mishehuheh
17:58.27tanvalmgHi, does anybody know if the Digium TDM400P is compatible with IBM xSeries 346?
17:58.54meshugai'm surprised someone hasnt made a hardware compatibility page on a wiki somewhere.
17:59.05mishehutanvalmg: hard to say, in all honesty.  did you do a google search or look on digium's compat list?
17:59.16meshugadigium has a valid one?
17:59.17*** join/#asterisk saftsack (n=saftsack@p54A7D890.dip.t-dialin.net)
17:59.22mishehuI've been unlucky with the digium cards causing PCI PERR# in the past.
17:59.46mishehuwelp.  lunch time
17:59.48meshugamishehu yea, thats crappy pci steering
17:59.52tanvalmgThe Digium site says it is partially compatible with xSeries 345, we xSeries 346
18:00.08meshugai use sangomas myself
18:00.13mishehutime is an illusion, lunchtime doubly so.
18:00.14tanvalmgwe have IBM xSeries 346
18:00.21meshugai've seen too many tdm400p's die by static
18:00.39meshugaman wtf
18:00.42mishehuI do not recommend any of the tigerjet chipset digium cards
18:00.44meshugahttp://pastebin.com/767705
18:00.50meshugai cannot get this to work
18:00.54mishehuI've had too many problems with them.
18:00.57meshugamishehu dude, can you take a look at this?
18:02.17mishehumeshuga: you put up the log but not the extensions config
18:02.44mishehuanyway, I'll be back in an hour.  must go out to eat and head over to a client's office
18:03.38MACscrhas anyone ever tried transfering their vonage number to another provider?
18:03.44meshugamishehu heh its just amp
18:04.00meshugamishehu they're setup the exact same way.
18:06.33mopriloasterisk is not recognizing # key on my zap channel phones (FXS ports), why can this be?
18:06.35*** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon)
18:06.42mopriloi just upgraded to 1.2.9
18:07.14*** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk)
18:08.50moprilojmm..
18:09.02mopriloout to lunch? jeje..
18:09.36mopriloit only happens when i receive a call
18:09.59mopriloand it doesn't do it on sip phones either
18:11.02RF_MIAanybody have any experience with res_snmp ?
18:13.03MACscrSmall phone network with all extensions at different locations. Should i go with IAX2 or SIP?
18:13.52meshugaiax2 is always better
18:13.56meshugabut less ip phones are IAX2
18:14.04meshugaso you usually have to go SIP for 'the last mile'
18:14.33MACscrgot any recommendations for iax2 phones that are entry level?
18:14.54*** join/#asterisk ToTo (n=ToTo@host68-166.pool879.interbusiness.it)
18:15.00moprilojaja, my fault, worked..
18:15.08*** part/#asterisk moprilo (n=mop@201.198.78.23)
18:15.20meshugamaxscr: i dont even know of any
18:15.27meshugai prefer polycom's for cheap ip phones
18:15.30meshugaand ciscos for mid grade
18:15.33_Sam--there are iax phones
18:15.35*** join/#asterisk philippel (n=p_lindhe@c-24-19-186-72.hsd1.wa.comcast.net)
18:15.47MACscrim familiar with polycom in general
18:15.47meshugathe 2 iax one i used last year were garbage
18:15.53MACscri worked in telecom for a couple years
18:16.04meshuganice i'd go with those for handsets
18:16.08meshugadid you provision by hand?
18:16.15*** join/#asterisk brc_ (n=brc_@pdpc/supporter/basic/brc)
18:16.21meshugabe sure to upgrade them all to the latest firmwares
18:16.34meshugacuz polycoms changed their configuration in later firmware revisions
18:16.48meshugaand i did a rollout once in 3 steps and half my shit didnt work and i had no idea
18:17.00*** part/#asterisk RF_MIA (n=mw1@ip67-93-229-222.z229-93-67.customer.algx.net)
18:17.08meshugawhy until i realized that, updated my tftp service to push firmware and all was well.
18:17.15*** join/#asterisk pdavid (n=chatzill@adsl-068-209-191-127.sip.mob.bellsouth.net)
18:17.18pdavidgrr
18:17.28pdavidcould anyone lend me a hand figuring out why i cannot seem to build the ztdummy module?
18:17.58MACscrlooks pretty cheap, but not bad as far as what it supports as an ATA
18:17.59MACscrhttp://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=5819261173&category=61839
18:18.02*** join/#asterisk chapeaurouge (n=chapeaur@user-85-201-82-146.tvcablenet.be)
18:18.48*** part/#asterisk JINDAL (n=root@220.226.36.2)
18:19.00*** join/#asterisk brc_ (n=brc_@pdpc/supporter/basic/brc)
18:19.25brc_hey hey hey
18:19.29brc_guess who
18:20.00pdavidmake runs clean without any errors
18:20.19pdavidso does make install
18:20.24pdavidbut a modprobe ztdummy does nada
18:20.36pdavidFATAL: Module ztdummy not found.
18:20.38pdavidFATAL: Error running install command for ztdummy
18:21.05C4T3lpdavid: you running udev?
18:21.21pdavidyes, and did make install-udev
18:21.21*** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)
18:21.23MatsKmake linux26
18:21.25pdavidand rebooted
18:21.31pdavidsame issue with make linux26 as well
18:21.51CunningPikepdavid: Make sure the module exists in the folder corresponding to uname -a
18:21.54*** join/#asterisk funxion (n=nunya@63.214.236.169)
18:22.07C4T3luname -r?
18:22.14CunningPikeC4T3l: That too
18:22.17dlynes_officeCunningPike: damned cops busted me on knight street :(((
18:22.18C4T3lhehe
18:22.19paolobHi guys! When trying to make a voip call with wengo, asterisk tells me many times: " WARNING[8120]: chan_sip.c:2542 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 256/256)". What does it mean?
18:22.38pdavidCunningPike: as in: /lib/modues/2.6.12-10-686?
18:23.27CunningPikepdavid: Yes - does that match uname -r?
18:23.33pdavidsure does
18:23.34MACscrmesh: which polycom do you prefer
18:24.01Bullseye_NetworkIm using a dialer for agents and im getting calls on hold when there are agents available in the queue. What are some good ways to speed up asterisk?
18:24.20MatsKpdavid: no errors while compiling ?
18:24.31pdavidMatsk: no, no errors while compiling
18:24.35CunningPikepdavid: What kernel?
18:24.46pdavid2.6.12-10-686
18:24.46CunningPikepdavid: nm - I can actually read lol
18:24.50pdavidlol
18:24.57pdavidmaybe they are going to the wrong place
18:25.02pdavidthis is an ubuntu box
18:25.09Bullseye_NetworkCan I change the nice levels?
18:25.12pdavidmaybe im missing a symlink somewhere?
18:25.23CunningPikepdavid: That would be my suspicion - I've had that happen on RHEL
18:25.35pdavidwell, where *should* they go, then?
18:25.52CunningPikepdavid: In the correct folder, of course :D
18:26.01pdavidahh, well that would make sense!
18:26.40salviadudCunningPike, are you saying red hat is red hell'
18:26.43salviadud?
18:26.59CunningPikesalviadud: Not at all - I like it
18:27.31salviadudi like suse better
18:27.46salviadudif i need a distro that does everything for me
18:28.21salviadudi've heard the guys from debian are like muslim fanatics
18:28.55salviadudthey'll DOS FreeBSD servers, and stuff like that
18:29.08dlynes_officesalviadud: you're confusing them with the guys in #perl :)
18:29.29CunningPikepdavid: Mine is in /lib/modules/2.6.9-34.EL/extra/, if that helps.....
18:29.32C4T3lsalviadud: are you serious about DOS?
18:29.48pdavidthat is where ztdummy.ko resides?
18:29.55salviadudno, i think it's funny to crash a server like that
18:29.58CunningPikepdavid: Yes
18:30.05*** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com)
18:30.08salviadudit's basically evil
18:30.15salviadudbad karma and such
18:30.30salviadudi don't do DOS, i'm a nice guy
18:31.07salviadudwell, regular guy, i like to prank call with .call files
18:31.14mitchelocwho uses dos anymore?
18:31.17salviadudphone terrorism is cool
18:31.28*** join/#asterisk w32 (n=234@c-71-193-124-77.hsd1.il.comcast.net)
18:31.38salviadudit's even better with voipdiscount
18:31.43salviadudi can call the US for free
18:32.04w32does anyone know of a reliable provider for origination and termination that accepts paypal ?
18:32.22salviadudnow, if i can get the guys from #perl to help me out with a script that auto-generates call files...
18:33.43*** join/#asterisk Blackthorn (i=blacktho@72.236.88.10)
18:33.44*** join/#asterisk eKo1 (n=bernd@190.4.7.90)
18:34.18eKo1Wow. Asterisk is moving so quickly. I just upgraded to 1.2.8 last week and now we have 1.2.9.1
18:34.33dlynes_officeCunningPike: so have you been busted by those damned cops on knight street yet?
18:34.46dlynes_officeCunningPike: or do you usually take oak street?
18:34.50CunningPikedlynes_office: No - I'm way to careful a driver :D
18:34.59dlynes_officeCunningPike: such a bs'er :)
18:35.14dlynes_officebastards busted me doing 76 on knight :(((
18:35.24mitcheloceKo1: not much makes you move faster then security issues ;)
18:35.30CunningPikedlynes_home: When I drive, I take Granville in the morning and Knight on the way home
18:35.39MACscryep, still no good iax2 phones
18:35.41CunningPikedlynes_office: It's a posted 50.......
18:35.51dlynes_officeah...lion's gate is pretty clear in the mornings?
18:36.00CunningPikedlynes_office: Yes, it's not bad
18:36.15dlynes_officeso what you're telling me is you never speed? :)
18:36.18CunningPikedlynes_office: But I'm scooting these days
18:36.27CunningPikedlynes_office: 10 over - never more
18:36.33dlynes_officeah....that's why you never speed
18:36.38dlynes_officethe scooter can't go that fast :)
18:36.45CunningPikedlynes_office: Even in the truck - 10 over
18:37.00dlynes_officeah
18:37.11dlynes_officeyeah...in the van, i usually don't go more than 5 or 10 over
18:37.27dlynes_officebut in my gti, it's a different story :)
18:37.28CunningPikedlynes_office: But you're right - the scooter maxes out at 65
18:37.35CunningPikedlynes_office: Golf?
18:37.38dlynes_officeyeah
18:37.40pdavidim pretty sure i am going to scream shortly
18:37.47CunningPikedlynes_office: Nice
18:37.47*** join/#asterisk noky (n=noky@200.69.211.18)
18:37.48nokyhi
18:37.58dlynes_officeyeah...nice red one, too :)
18:38.15CunningPikepdavid: Still struggling?
18:38.24pdavidyep, big time
18:38.37pdavidcant seem to figure out where its all going haywire
18:38.41*** join/#asterisk truz_`24 (n=truz_`24@74.129.166.232)
18:38.46CunningPikepdavid: Your rules-50 and permissions-50 are all OK?
18:38.49pdavidshould probably step away from the terminal and smoke before i toss it out the window
18:38.49nokyplease i need some help, where can i found any benchmark or anything to see the performance of the application MeetMe in Asterisk?
18:39.39pdavidwell, lemme check, and smoke.  bbl
18:39.41truz_`24Can asterisk determine if a call was intercepted by an operator
18:40.02r_evolutionfuck
18:40.09r_evolutioni like too wide a variety of music :(
18:41.04sevardhow the hell is that sad?
18:41.29mitchelocdo people know about pound key??
18:41.54*** join/#asterisk SplasPood (n=jwb@206.252.198.101)
18:41.59CunningPiketruz_`24: Telco operator?
18:42.01r_evolutionit's sad because i filled up the external hard-drive i bought for music
18:42.04r_evolutionand that annoys me
18:42.08r_evolutionbecause it means i have to buy another one...
18:42.12r_evolutionand i wanted them to all be on ONE drive
18:42.16r_evolutionnot like 4 :(
18:42.21truz_`24CunningPike, yeah, and detecting answering machines and such.
18:42.23AlricSetup a nice NAS :)
18:42.28CunningPikemitcheloc: How do you mean?
18:42.30r_evolutionsomeone hurry up and make a 2 Terrabyte external
18:42.44CunningPiketruz_`24: Not sure how it could - it just knows the channel was answered
18:42.52*** join/#asterisk caloi (n=caloi@nat-66-218-1-215.usadatanet.com)
18:43.10KranZanyone found the perfect software echo cancellation settings for a pri?
18:43.37[TK]D-FenderKranZ : No, but I've found a good solder one ;)
18:43.47*** join/#asterisk jhiver (n=jhiver@LReunion-151-20-4.w193-253.abo.wanadoo.fr)
18:43.47KranZu modded?
18:43.51jhiverhi all
18:43.53[TK]D-FenderKranZ : What you're asking doesn't exist :)
18:44.00jhiverI have a strange issue with Asterisk
18:44.03mitchelocCunningPike: supposedly pk is the official linux distro from digium with asterisk in it....
18:44.11caloiHello all - anyone feel like assisting a newcomer on dial plan creation for an iax -> iax connection?
18:44.12mitchelocCunningPike: i was wondering why people don't use that a hell of a lot more....
18:44.25[TK]D-FenderKranZ : Echo correction needs is ALWAYS variable... and every case can have different needs.
18:44.43[TK]D-FenderKranZ : If it could be standardized you wouldn't be asking that question right now :)
18:44.57jhiverone of my customers sends me SIP traffic from a cirpak switch, and he can't hear any ringing at all
18:45.07r_evolutionwe all have strange issues with asterisk jhiver
18:45.12r_evolutionit's a strange system
18:45.18r_evolutionI support tin can with a string personally
18:45.18jhiverbut when the phone is off hook the conversation goes fine
18:45.19*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
18:45.20r_evolutionbut that's just me
18:45.25CunningPikemitcheloc: Ah - lol. I thought you meant...... #
18:45.39jhiverit's just that there is no ring, which is kind of bad
18:45.55mitchelochttp://www.digium.com/en/products/opensourceasterisk/downloads.php
18:45.57r_evolutionoh oh i've got it! use the r option!
18:45.58r_evolutionhaha
18:45.59r_evolution;x
18:46.16mitcheloclook at PK on there, im downloading right now
18:46.30KranZyeah, i've got a remote sip ua connected to a pbx via pri
18:46.40KranZbad echo on the pstn side for cell phones
18:46.46[TK]D-FenderBBIAB
18:46.47*** part/#asterisk marv[work] (n=timr@64.89.118.139)
18:46.55jhiveranybody else got this 'no ring issue'?
18:46.55*** join/#asterisk marv[work] (n=timr@64.89.118.139)
18:48.07jhiverI have progressinband=yes in my sip.conf so my understanding is that asterisk should send some RTP?
18:48.20jhiverwhen ringing?
18:48.48BlackthornKranz are you refering to the double ring when you call form ata --> * ---> out pri?
18:51.28KranZnah
18:51.44Blackthornok i just jumped into channel.. to try to ask the above statement.
18:51.47KranZyou can fix that with sticky183=yes on the sip ua
18:52.03Bullseye_NetworkIm using a dialer for agents and im getting calls on hold when there are agents available in the queue. What are some good ways to speed up asterisk?
18:52.14pdavidSo, i have NO idea what I did, but there they were
18:52.16BlackthornAta to * to voice pulse, ata to * to ata... works fine. but i can't get the doule bring
18:52.20pdavidafter a quick depmod, im in the game
18:52.26pdavid:(
18:52.29pdavidhate voodoo installs
18:52.33meshugaso anyone know why when I'm using DISA and i got a remote dialtone, I'm unable to dial out thru voipjet or voipbusted or anything? I never even get the 'attempting native bridge' or any progress indicatations..
18:52.48pdavidcunningpike: thanks for the help, though!
18:53.04meshugavoipjet gives me a congested tone, and voipbuster just has me hanging
18:53.12truz_`24CunningPike, dialogic can detect answering machines with a 98% accuracy...
18:53.14meshugaif i call from an internal sip extension, everything works fine
18:53.25BlackthornKranz: are you stating to fix the issue i metnioned I put "sticky183=yes" on the sip.conf profile for the ata?
18:53.30truz_`24CunningPike, is there any such guess algorithm/plugin for asterisk?
18:53.31meshugabut from my DISA login it doesnt work
18:53.36CunningPikepdavid: Great
18:54.14CunningPiketruz_`24: I don't believe so...... but I remember reading something either on the list or the wiki about answering  machine detection....
18:54.19*** join/#asterisk moprilo (n=mop@201.198.78.23)
18:55.01moprilois it posible i don't have my Asterisk Database installed, my call forwarding is not working.., and all the commands seem to be though
18:55.45truz_`24CunningPike, can i have a link to the wiki?
18:55.54CunningPike~wiki
18:56.17CunningPike~thewiki
18:56.18jbotextra, extra, read all about it, thewiki is at http://www.voip-info.org/wiki-Asterisk
18:56.31nokyplease i need some help, where can i found any benchmark or anything to see the performance of the application MeetMe in Asterisk?
18:57.08CunningPikenoky: Try the list archives - I have a vague recollection of a discussion there a while ago
18:57.47CunningPike~thelist
18:58.05CunningPike~list
18:58.06jbotone warez list being sent
18:58.20CunningPikejbot, you are a fool
18:58.21jbotI think you lost me on that one, CunningPike
18:59.09BlackthornKranz: Where do you plac "sticky183=yes" on the sip.conf profile for the ata?
18:59.56truz_`24CunningPike, thanks
19:00.22CunningPiketruz_`24: You're welcome
19:00.44CunningPike~asterisk-users
19:01.59CunningPikejbot, thelist is the asterisk-users mailing list. Sign up or view archives at http://lists.digium.com/mailman/listinfo/asterisk-users
19:02.01jbotokay, CunningPike
19:05.45Blackthornahhh.. sigh.. i finally find somone that gives me a clue how to fix the issue... and i don't understand the answer :\
19:06.02Blackthorngoggled it and found three sites in german :P
19:07.52*** join/#asterisk ToTo (n=ToTo@host68-166.pool879.interbusiness.it)
19:09.52*** join/#asterisk squinky86 (n=squinky8@gentoo/developer/squinky86)
19:10.25*** join/#asterisk aze_ (n=aze@ACayenne-101-1-10-77.w81-248.abo.wanadoo.fr)
19:10.37*** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br)
19:11.01KranZBlackthorn: google stick183
19:11.13KranZi enabled it on my linksys/sipura devices
19:12.30Blackthorngoogle only pulls up 3 websites which do talk about sipura and stiky183 but there in german.
19:12.48Blackthorni even just tried to ues babblefish to translate :P didn't work.
19:12.49MACscrkranz: how do you like your sipura phones
19:13.08SplasPoodhrm.. wish I could find a Universal build of some mac os x softphone
19:13.09BlackthornSo i'm in the sipura web interface, got a hint as to where you went?
19:13.11KranZthe rt31p2 works well
19:13.21KranZspa-941 are good also
19:13.46MACscrim in the midst of planning out my first asterisk system
19:13.58MACscrneeding it for my web hosting company
19:14.09KranZi just deployed spa-941 phones with asterisk as the pbx
19:14.20MACscrremote agents is going to be biggest part
19:14.44blitzragethe only thing I don't like about the RT31P2 is the long response timeout unless you hit #
19:14.57*** join/#asterisk websae (n=websae@209-252-79-66.ip.mcleodusa.net)
19:14.59KranZblitzrage: you're dialplan isnt US
19:15.01KranZer your
19:15.07Blackthornok i'm workign with the spa-2000 so perhaps that function isn't in there?
19:15.11blitzrageI haven't found any way of changing it (if you dial 1+areacode+number then it works)
19:15.28blitzrageKranZ: like I mentioned above, only matches right away on 11 digit dialing
19:15.33KranZ(011,xx.|*xx|[3469]11|0|00|<:1408>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxx)
19:15.41KranZtry that
19:15.48blitzrageKranZ: where is there configured in the RT31P2 ?
19:15.53blitzrage/s/there/that
19:15.57KranZdo you have an unlocked one?
19:16.15blitzragewhat do you mean by unlocked? I guess I haven't looked at what that device can do closely enough :)
19:16.31KranZthe model should be rt31p2-na
19:16.35KranZon the lable on the bottom
19:16.42blitzrageahhh... not na afaik -- its about a year old now
19:16.56blitzragewhats the difference?
19:17.04KranZdid you get it through vonage or another provider?
19:17.22KranZthe diff is you have the same access as you would a regular sipura
19:17.25blitzrageI just bought one -- its not locked to a provider
19:17.35blitzrageI see what you mean now :0
19:17.39blitzrageerrr... :)
19:17.41KranZlook at the lable on the bottom of the unit
19:18.03blitzrageKranZ: what menu is that pattern matching under?  I'm going to have to look that up when I get home this weekend... maybe I've just missed it...
19:18.06blitzragehrmmm :)
19:18.49KranZfyi, the pattern prefixes 1408 to a 7digit number
19:19.05KranZso either remove the <:1408> or change it to your area code
19:21.17blitzragehrmm... gotta find the prefix pattern... totally missed it
19:22.11KranZanyone else love it when vonage's stock drops even further?
19:23.27*** join/#asterisk darby_t (i=darby_t@aapc8.neoplus.adsl.tpnet.pl)
19:25.23blitzrageKranZ: I do... and I don't -- I think if Vonage goes under that will hurt VoIP adoption in the consumer market
19:25.45blitzrageKranZ: it will look back on everyone in the general publics eye
19:26.41*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
19:27.21BlackthornYES! double ring gone.. ohh man that makes me so happy..
19:27.25mitchelocKranZ: what do you have against vonage??
19:27.27BlackthornI think i go home an hour early today :)
19:27.44KranZwell... from a clec standpoint...
19:27.46sevardhow the hell is that sad?
19:28.58Blackthornvonage is trying to compete on price to much, they should market a good product at decent price that they can maintain good service.
19:30.29coppicehow could vonage charge more than they do? they are already rather expensive
19:30.40*** join/#asterisk fholmes (n=fholmes@rrcs-24-227-237-197.sw.biz.rr.com)
19:31.25blitzrageagreed
19:31.38KranZthey're already not making money
19:31.41blitzrageI'm glad they are expensive -- helps me undercut them :)
19:31.47blitzragethey spend WAY too much money on advertising
19:31.53*** join/#asterisk caloi (n=caloi@nat-66-218-1-215.usadatanet.com)
19:31.59KranZthey're losing their asses on the LD minutes
19:32.01blitzrageits been a money losing company since day once
19:32.02drrayI don't see how vonage wins
19:32.10fholmesI am having a problem with my SIP softphone.  It was working until I unregistered it and I am now trying to re-register as a different user.  Now I cannot even re-register as the old user anymore.
19:32.12coppicemost spending in telecoms today is on advertising
19:32.28drraybecause all the bells have to do is decide to crush them, and it is over for vonage
19:32.37KranZfholmes: run some packet sniffer on your computer or the asterisk box
19:32.39fholmesI keep getting Username/auth name mismatch for some reason.
19:32.42KranZwww.ethereal.com
19:33.24fholmesI just happen to have it installed already.  Let me know if there are some commands I can use that will make it as easy as possible to figure out what is going on.
19:34.08KranZfilter out your asterisk ip
19:34.15KranZ'host x.x.x.x'
19:34.53*** join/#asterisk AlexCTI (n=alex@adsl-074-238-025-003.sip.mia.bellsouth.net)
19:37.14fholmesSo what am I looking for?
19:37.29Blackthornhow can you say 'there too expensive" and then say there loosing money on ld minutes...   They've got plans out there for like $15 bucks... The few customers that I have, i am serving over wireless 802.11b for $35.
19:39.03blitzrageyah... but when you charge someone $35 for a plan, $20 of that is to pay for the termination, and then spend another $50 on advertising... its a losing system
19:40.33Blackthornumm perhaps. but 1) i'm not paying $50 for advertising because i'm only marketing to my wireless customers and 2) i already have there e-mail.
19:41.09BlackthornI'm just saying if vonage is only charging $15 and spending all that money on advertising i agree with you. can't make money that way
19:41.52_Sam--i dont thik the average account at vonage is costing them 20/month for termination/orig.
19:42.00blitzrageI don't remember saying anything about your company... I thought we were talkinga bout vonage
19:42.10Dr-Linuxblitzrage: Hi :)
19:42.17sivanavonage has been educating the public... their gap is closing
19:42.21blitzrageDr-Linux: hoi
19:42.33coppicemost telecoms companies now spend more on recruiting subscribers than on providing them service
19:42.38*** join/#asterisk ManxPower (i=ewieling@184.sub-70-196-103.myvzw.com)
19:42.41blitzragesivana: exactly -- which is why I think it'd be a bad thing if they went under
19:42.47Blackthorntrue sivana thats the one very good thin they have done for all of us.
19:42.51_Sam--and if the value per subscriber in a resale of the customers becomes more valubale than the cash flow of the customers...then you can understand their rationale
19:42.56*** join/#asterisk chainey (n=jeremy@dsl017-031-038.lax1.dsl.speakeasy.net)
19:43.08_Sam--you are only looking at cash flow, not value of the biz.
19:43.27*** join/#asterisk castro2006 (n=khaled@83.244.91.231)
19:43.39blitzragehow many times have you explained VoIP by saying something like, "You know... like Vonage?"
19:43.41sivanablitzrage: yup, but they won't their loss/profit gap is closing -- they'll survive
19:43.52websaeVOnage is a terrible standard
19:44.01blitzragesivana: ideally... but the IPO thing really didn't make things any better -- it made things worse
19:44.01websaeno one can possibly compare to vonage
19:44.04KranZwhat's the latest zaptel stable ?
19:44.06KranZim lazy
19:44.07castro2006i need SIP soft phone to install on Debian ??
19:44.11sivanablitzrage: yea, that whole employee program
19:44.18blitzragewebsae: its not a standard you refer to -- its a reference point that people recognize
19:44.21KranZ~zaptel
19:44.23jbotwell, zaptel is zapata telephony interface. A low level interface designed to abstract hardware access to a variety of devices for BRI, PRI or analogue access.
19:44.25websaeVonage SOPs are completely radical from the norm of these other ma and pa shop VoIP providers
19:44.47eKo1I'm a ma and pa
19:44.53eKo1VoIP provider
19:44.57eKo1:P
19:45.12castro2006??
19:45.23websaeeKo1: the best thing that could happen to you is that you get bought out
19:45.25_Sam--ma and pa:  did you get hacked by the dude?  http://www.nytimes.com/2006/06/08/technology/08voice.html
19:45.30eKo1castro2006: use xtenlite
19:45.42eKo1websae: that is exactly what we're aiming for.
19:45.52sivanawebsae: what other option do you have.. you think you'll buy AT&T or any other Bell
19:46.00blitzragewebsae: indeed :)
19:46.12TheCompWizI will probably get shot for asking this... but has anyone had any luck using skypeout for outbound calls?
19:46.22eKo1I hope someone buys us quick and for 1.5 times our price :)
19:46.24websaesivana: obviously i don't think that, as i just said my opinion
19:46.29eKo1or twice even
19:46.30castro2006i tried to install it but it did not work
19:46.33r_evolutionblitz. get the rope.
19:46.40castro2006it is tested on debian
19:46.42eKo1castro2006: how?
19:46.43sivanawebsae: you said the "best thing that could happen".... and I'm saying.. it's the only thing
19:46.58eKo1sivana: hahaha
19:47.09websaeno you could go bankrupt.....in debt
19:47.14websaethere are other things that could happen
19:47.14sivanahaha.. true
19:47.19_Sam--i dont think there will be much consolodation (buy outs) in the VOIP arena...why would anyone (comcast, verizon, etc) need to buy a customer base when they can take them over by default in masses over time
19:47.20r_evolutionyeah cuz ya know... you certainly can't make money as a small company.
19:47.24websaewhich happens for quite a few
19:47.28*** join/#asterisk techie (n=gus@antibala.com)
19:47.55TheCompWizdoes anyone know if it can be done? (skypeout)
19:48.09r_evolution...
19:48.12websaethe best VoIP company/provider that has a chance at climbing the ladder....is one that uses VoIP to create a niche market
19:48.14r_evolutionwhy's it gotta be fancy?
19:48.21r_evolutionnow let's hang and shoot CompWiz
19:48.27r_evolutionmaybe not in that order though...
19:48.42TheCompWizr_evolution: have you tried?
19:48.52r_evolutionno. we're going to be shooting and hanging you now
19:49.03r_evolutionjust out of curiousity... CompWiz.
19:49.11r_evolutionif you're in an asterisk channel...
19:49.24r_evolutionasterisk != skype
19:49.36r_evolutionwhy exactly are you asking about using skype?
19:49.40TheCompWizr_evolution... I'm trying to setup skype as a trunk for outbound dialing in asterisk...
19:50.16TheCompWizso when user a... picks up his phone & dials a number... it's on skype's dime.
19:50.18blitzrageTheCompWiz: skype is a proprietary application -- via skypeout you can call asterisk as long as Asterisk is available via the PSTN
19:50.38r_evolution^
19:50.45_Sam--you can do it though
19:50.51_Sam--you need a skype phone adapter or something
19:50.52TheCompWizblitzrage: I know that... but there are apps out there like "Skype2Sip" that's supposed to translate skype into SIP.
19:50.53sivanayes.. with an fxs/fxo
19:50.55sivanabut it's lame
19:50.56_Sam--there was someone here that did it
19:51.04mitchelocif i might join in... i think the real killer for pots & voip companies, is going to be a bittorrent-like configuration for phones.....
19:51.13sivanaan probably violates their tos
19:51.28eKo1mitcheloc: please elaborate
19:51.44mitcheloci mean that there will be no companies to terminate calls, just phones calling phones directly (in an easy to do manner)
19:51.54r_evolutionwhere you gets parts of everyones voice from various locations?
19:51.56r_evolutionawesome.
19:51.59blitzrageTheCompWiz: so go try it and tell us how it went
19:52.08blitzragewe don't hand hold in here
19:52.18mogormanmitcheloc, that wont exist
19:52.23r_evolutionyeah... im curious to know wtf would make that work? O_o
19:52.24sivanaespecially for skype related stuff
19:52.31*** join/#asterisk southtel (n=slester@c-67-191-211-17.hsd1.ga.comcast.net)
19:52.32mogormanas there is a level of quality needed that wont be there
19:52.40r_evolutionim saying... seriously... it sounds like he's trying to setup a ITSP type solution
19:52.41mogormanbut i do think service providers will operate like that
19:52.44mogormanlike over dundi
19:52.44r_evolutionexcept terminate to skype
19:52.49mitchelocobviously the bittorrent protocol wouldn't work, but some sort of distriubuted mechanism would be great...
19:52.51southtelHas anyone successfully sent faxes using a TDM110P?
19:53.13mitchelocdundi seems to be close to what i'm talking about, i'm thinking it'll take the next step beyond that though
19:53.18mogormanright without a party to trust
19:53.20mitchelocthink web 1.0 vs web 2.0, and so on
19:53.21mogormanit wont work
19:53.27r_evolutionhehe
19:53.31mogormandundi is what you are talking about
19:53.33r_evolutioni think web 2.0 and i think userfriendly ;x
19:53.43mogormanbut it cant work on a grand scale without people watching it
19:53.44r_evolutionweb 2.0 = omgwtfbbq! o rly? ya rly!
19:53.47r_evolution;)\
19:53.49*** part/#asterisk mogorman (i=ejabberd@68.62.237.103)
19:53.57*** join/#asterisk mogorman (i=ejabberd@68.62.237.103)
19:54.06mitchelocmogorman: to be fair, i wouldn't have thought a serverless way to share files could happen... the technology will evolve...
19:54.26mogormanthere is no such things as a serverless way to share files
19:54.32mogormanwe just have more servers now....
19:54.52r_evolutionyeah i think he's trying to say he never thought there wouldn't be a server-based storage for those files
19:54.53mitchelocmogorman: you know what i mean ;), there will be a way to do something like this
19:54.55r_evolutioni.e. ftp etc
19:55.09*** join/#asterisk techie (n=gus@brutus.voipops.net)
19:55.11mogormanthey just made ftp easier
19:55.19mogormanno one has gotten rid of what ftp is
19:55.23southtel(TE110P, rather)
19:55.27r_evolutionwell of course not
19:55.30r_evolutionftp is still useful
19:55.40mogormani think the closest thing to what you are talking about is email
19:55.41mogormanor jabber
19:55.51r_evolutionbut you can't say a lot of places don't encourage using bt in place of ftp
19:55.54mitchelocto put it in those terms, when calling phones directly gets easier,  you will see a huge decline in actual need for phone service
19:55.55mogormanbut both are controlled by verisign at some level
19:55.58r_evolutionconserve bandwidth etc
19:56.03mogormanthere needs to be a guy in charge
19:56.15KranZelect KranZ!
19:56.15r_evolutionill agree with that in part...
19:56.22r_evolutionnot the KranZ part.
19:56.25KranZdoh
19:56.27Blackthornlol
19:56.28r_evolution;)
19:56.41r_evolutionpeople are too fucking dumb to be in charge of themselves usually
19:56.49r_evolutiona person may be smart... but people are effin retarded
19:57.02mitchelocagreed, like someone is in charge of the .com system... , it'll get easier and easier ;), eventually you'll be so seemless using msn + talk or aim+talk,or whatever, i think it's going to slowly remove the need for phone companies...
19:57.14mitchelocthe cell phone companies have the best security atm ;)
19:57.27KranZmitcheloc: until it's reliable, commercial businesses will stay away
19:57.29mogormanbut thats all phone companies are
19:57.44_Sam--KranZ:  what other type of business besides commercial is there?
19:58.03KranZhome office
19:58.08*** join/#asterisk sorush20 (n=sorush20@82-43-184-143.cable.ubr07.newm.blueyonder.co.uk)
19:58.17Nivexnon-profit business?
19:58.23mogormansomeone who routes numbers to users
19:58.28_Sam--its not a business if its non profit
19:58.29mogormanjust like verisign
19:58.35_Sam--its a non profit organization
19:58.38*** join/#asterisk Soybomb (i=Soybomb@71-8-250-35.dhcp.mtvr.il.charter.com)
19:58.52sorush20I want to be able make calls from my computer to lanlines and mobiles and recieve the calls trough my computer using my landline...
19:59.00sorush20I want to be able record conversations..
19:59.01*** join/#asterisk kitche (n=el_lupo@pool-70-16-34-92.buff.east.verizon.net)
19:59.02_Sam--any event sorry for splitting hairs, couldnt help it.
19:59.08sorush20is this the best program for it?
19:59.44KranZsorush20: yeah, i recommend you mess with asterisk at home
19:59.45sorush20I have broadband.. wireless connection to a router.. and my phone line is separate from broadband cable connection..
19:59.54r_evolutionHey sorush... I recommend you get a tin can with a string.
19:59.59truz_`24lol
20:00.22truz_`24There seems to be a lot of fluff out there concerning voip.
20:00.28KranZsorush20: "Asterisk@Home"
20:00.33sorush20tin can with a string does not record conversations..
20:00.34KranZ~aoh
20:00.39r_evolutionyes it does
20:00.41Blackthorntin can with a string. hehe and this form a guy with the nick evolution :P
20:00.47KranZ~asterisk@home
20:00.48jbotfrom memory, asterisk@home is http://asteriskathome.sourceforge.net/, or http://www.voip-info.org/tiki-index.php?page=Asterisk+at++Home
20:00.50r_evolution;x
20:00.55Blackthornwould that be r(reverse)_evolution?
20:01.00r_evolutionyesssss
20:01.05r_evolutionDEVOLUTION BITCHES!
20:01.12r_evolutionTIN CAN WITH A STRING ALL THE WAY!
20:01.15[TK]D-Fender|AFKHey I've got an emergency request I could use some quick tips on : I need to track all IP traffic from 2-3 hosts here and what I'm looking to do is set they default routes in DHCP to pass through my linux server which would then forward the packets on to our real internet router (SonicWALL).  I am hoping to log EVERYTHING through Ethereal.  Is this viable and are there any other tricks I'd need to do to prep for this?  IP forwa
20:01.24Soybombhello all, my boss just got me a 4t1 card to play with (generic tormenta clone from govarion.com) - it gets deteced as a tormenta 1 quad t1 card, but ztcfg fails with ZT_SPANCONFIG failed on span 1: No such device or address (6)
20:01.31*** join/#asterisk stormfr (n=StorM@stardust.noc.frontier.fr)
20:01.32*** join/#asterisk Wowzers10 (n=pbaker@nnat-gw.adeptra.com)
20:02.02KranZ[TK]D-Fender|AFK: sounds about right
20:02.11KranZ[TK]D-Fender|AFK: now you need to reset their dhcp leases
20:03.06Wowzers10hello all, with straight through asterisk and no agi - is there a way to use ackcall and announce but have it play " please press pound to connect " after the call is answered?
20:03.30Wowzers10IE: they dont hear silence, so they know what to do
20:04.19sorush20KranZ: for everyone's info asterisk@home is now.. http://www.trixbox.org/, now who is in the know.. ?
20:04.31blitzrage#freepbx is in the know
20:04.59r_evolutioni'm telling you sorush...
20:05.00r_evolutionTIN
20:05.00r_evolutionCAN
20:05.01r_evolutionWITH
20:05.02r_evolutionSTRING
20:05.06blitzragevery effective
20:05.10r_evolutionyes.
20:05.14blitzragetin can telecom, inc. <-- my company!
20:05.14KranZi recommend extra string
20:05.18KranZthey can break sometimes
20:05.23blitzrage20lb. test
20:05.27sorush20r_evolution: no thank
20:05.33sorush20s
20:05.50coppicewired? wireless? stringed?
20:05.55KranZyou might have to condition the string to make sure it will carry the signal
20:05.56r_evolutioni'm telling you sorush
20:05.58r_evolutiontin can with a string
20:06.02r_evolutionthere's even a cat in canada
20:06.04r_evolutionby the name of Life
20:06.10r_evolutionhe wrote a book about the tin can theory
20:06.22KranZhe was also on a hot tin roof
20:06.26Soybombanyone know if thats a driver issue?  thats my guess but its odd to see anything in the dmesg then i'd think
20:06.28[TK]D-Fender|AFKKranZ : Easily done.
20:06.52r_evolutionsee... you've got all these people advising you... but you won't take the advice :(
20:07.16blitzrageadvice is overrated
20:07.23stormfrhello, i don't know if it's a bug or a new directive but when a call is answered on second dial command, cdr disposition is set to FAILED (billsec>0). This problem exist since around january but not before.
20:07.39r_evolutionsometimes.
20:09.35r_evolutionyou know what's not overrated?
20:09.35r_evolutionhttp://www.addictinggames.com/curveball.html
20:09.37r_evolutionthat game.
20:09.44*** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com)
20:10.00*** part/#asterisk chainey (n=jeremy@dsl017-031-038.lax1.dsl.speakeasy.net)
20:12.46Blackthornthanks for that, i'm off.. just don't ask how far
20:13.11Blackthorni could have swore i wrote "thanks for the chat"
20:13.24r_evolution?
20:13.28r_evolutionyou're on drugs.
20:13.30r_evolution;)(
20:13.36Blackthorni wish :P
20:14.33*** join/#asterisk ToTo (n=ToTo@host68-166.pool879.interbusiness.it)
20:15.24markus99hi all, I have an issue on my zap modem when in conversation I hear beeps once in a while, someone said that it might have something to do with dtmf, not sure even what to look under
20:15.24sorush20so why should I use freepbx rather than asterisk?
20:15.58mitchelocyou shouldn't
20:16.03mitchelocif anything, you use them together
20:16.55r_evolutionO_o
20:16.57Soybombin my dmesg after registered tormenta2 pci i see scb2_flash: warning - can't reserve rom window, continuing" ---is that a problem related to the card? (bsd guy here)
20:17.03r_evolutiontin can.... with string.
20:19.01KranZdamn curveball
20:19.02*** join/#asterisk Delta239 (n=paparapa@201.218.116.114)
20:19.12r_evolutionhehehehe
20:19.17r_evolutionyou're playing now arent you kranZ
20:19.20r_evolutionit's addictive as shit
20:19.21KranZmebbe
20:19.24r_evolution;x
20:19.28r_evolutionit's like geek-crack
20:19.41*** part/#asterisk kitche (n=el_lupo@pool-70-16-34-92.buff.east.verizon.net)
20:19.46r_evolutionthe lan admin here got the net admin who got me
20:19.51r_evolutionnow i gotchu!
20:20.17*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
20:20.51r_evolution~jbot
20:20.52jbotrumour has it, jbot is only marginally useful at best,  He got a C- on his Turing Test, or a complete idiot
20:21.09r_evolution~blitzrage
20:21.11jbotfrom memory, blitzrage is a super cool fellow
20:21.18r_evolutionsomeone lied to jbot :-\
20:22.32blitzrageouch!
20:22.50r_evolutionHAR
20:22.53r_evolutionhttp://www.addictinggames.com/curveball.html
20:22.56r_evolutiongo play blitz
20:23.16blitzrageprobably needs java
20:23.29blitzragein linux on a computer in class with no java :)
20:23.38r_evolution:(
20:23.39r_evolutionflash
20:24.36*** part/#asterisk cytrak (n=btorch@208.63.19.179)
20:25.03*** join/#asterisk mog_home (n=mogorman@68.62.237.103)
20:26.31Wowzers10hello all, with straight through asterisk and no agi - is there a way to use ackcall and announce but have it play " please press pound to connect " after the call is answered? - this is so they do not hear silence
20:28.14r_evolutioneveryone is playing curveball wowzers.
20:28.15r_evolutionhttp://www.addictinggames.com/curveball.html
20:28.17r_evolutiongo play :)
20:29.29KranZdammit
20:29.40KranZlevel 8 and i click a bookmark folder
20:29.45KranZhalf the screen is covered
20:29.55KranZi managed to keep playing for 10 more secs
20:30.11KranZhmm...full screen mode
20:30.29r_evolutionheh
20:30.40r_evolutiondude level 8 and up gets CRAZY
20:30.47r_evolutionim saying that ball is all OVER the effin place
20:30.53KranZyeah, sucks u into the zone
20:31.04r_evolutioni know
20:31.10r_evolutionimma laugh when i redball and fall over and twitch
20:31.47tzafriranybody here uses res_zeroconf? Any published version of it?
20:32.17KranZhttp://farm.addictinggames.com/D78AQSAKQLQWI9/1258.swf
20:32.19KranZbetter link for it
20:32.31r_evolutionhow so?
20:32.35r_evolutionoh
20:32.36r_evolutiondur
20:32.39r_evolutionstraight to the flash
20:33.07r_evolutionO_O
20:33.14r_evolutionperfect sound track
20:33.17r_evolutionfast paced breaks
20:33.25KranZclose all the toolbars and goto full screen mode
20:33.34*** join/#asterisk JINDAL (n=root@220.226.36.2)
20:33.50*** join/#asterisk postel (n=jp@unaffiliated/postel)
20:34.21r_evolutioni might never come back
20:34.33*** join/#asterisk vechers (i=vechers@64.61.117.138)
20:34.53*** part/#asterisk vechers (i=vechers@64.61.117.138)
20:35.01*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
20:35.39r_evolutionyep
20:35.41r_evolutiondef NOT coming back
20:38.02httpcrap
20:41.14*** join/#asterisk zotz (n=zotz@24.244.133.115)
20:41.59*** join/#asterisk Iam8up|lpy (n=iam8up@cpe-24-210-253-66.woh.res.rr.com)
20:42.18KranZgah, only 3:42
20:42.26r_evolution?
20:42.27Iam8up|lpycan anyone tell me where to find the pdf document that has all the IVR commands for the digium iaxy s101?
20:42.28r_evolutionheh
20:42.30r_evolutionaddictive ;x
20:42.37r_evolutionhey ate up
20:42.38r_evolutiongo to
20:42.39r_evolutionhttp://farm.addictinggames.com/D78AQSAKQLQWI9/1258.swf
20:42.39mogormanivr commands?
20:42.42r_evolutionit's there.
20:42.48Iam8up|lpythe two docs on digium's website aren't what i'm lookin for, and i can't find it on voip-info.org
20:43.21Iam8up|lpymogorman - so you can hit like **** on your phone while plugged into the ata and it has it's own voice menu?
20:43.35Iam8up|lpyi thought it was ****, but it didn't work - nor did ####
20:44.23C4T3lwhat kind of ata you have?
20:44.43Iam8up|lpydigium iaxy s101?
20:44.52Iam8up|lpymaybe the digium doesn't have one..i may be thinking of the sipura ata101
20:44.59Iam8up|lpyata1001 rather
20:45.12C4T3lthat sounds like sipura to me
20:45.19*** join/#asterisk raidenz (i=raiden@205-200-66-136.static.mts.net)
20:45.28Iam8up|lpymmm
20:45.47*** part/#asterisk JINDAL (n=root@220.226.36.2)
20:45.48Iam8up|lpythat may be right..cause i can only figure out the digium's ip address by having it dhcp and looking at the leases given...
20:45.57mogormanyeah the iaxy doesnt have anything like that
20:46.04Iam8up|lpydamn...
20:46.08Iam8up|lpymy mistake, sorry
20:46.12mogormanno prob
20:46.20raidenzhi guys
20:47.26r_evolutionhey
20:47.27r_evolutiongo here
20:47.27r_evolutionhttp://farm.addictinggames.com/D78AQSAKQLQWI9/1258.swf
20:48.20dlynes_officeAnyone have any idea what this would mean?
20:48.22dlynes_office!! Not good - head of queue has not been transmitted yet
20:48.22dlynes_officeJun  8 13:36:47 WARNING[24732]: chan_iax2.c:692 jb_warning_output: Resyncing the jb. last_delay 56, this delay 17605, threshold 1072, new offset -17605
20:49.23raidenzDoes anyone have an updated test g729 codec that works with the latest Asterisk SVN (1.4)? Please msg me.
20:51.46MikeJ[Laptop]what exactly is a "test" g729 codec?
20:52.16r_evolutionone he doesnt have to pay for?
20:52.17r_evolution;x
20:52.22raidenzno
20:52.29raidenzthe test gcc/ipp g729 codec.
20:54.49docelmoYOUR STEALING FROM DIGIUM AND BREAKING THE LAW!
20:55.05docelmoif your in the US anyhow
20:55.13MikeJ[Laptop]they are not stealing from digium.. they may be stealing from the g729 consortium
20:55.21ids2500stealing from digium?
20:55.21ids2500lol
20:55.45ids2500NEWS FLASH: digium has no more g729 codecs today... raidenz stole them all!
20:55.49r_evolutionyeah but regardless of who... the codec is still cheap
20:56.08ids2500raidenz: you won't get any modules to work with SVN because they changed the headers and stuff...
20:56.10KranZ$10 a chan is not cheap
20:56.14MikeJ[Laptop]depending on the license of the ipp code, and what patent rights it comes with, it may be totally legal, I have never looked at it
20:56.16r_evolutionO_o
20:56.20r_evolutioncompared to what? free?
20:56.40KranZto fill a quad pri is + $960
20:57.19*** join/#asterisk mtaht4 (n=m@207.47.5.58.static.nextweb.net)
20:57.23r_evolutiontouche.
20:57.46r_evolutiongo play curveball
20:57.49r_evolutionyou're stopping me
20:57.50KranZheh
20:57.56KranZi heard footsteps
20:58.03r_evolutioni sit facing the door :)
20:58.09docelmoMikeJ[Laptop] to use the IPP code the "free" one is for non-commercial personal use so inherently yes you could..  But in a business environment nope..
20:58.33r_evolutioni've come to the conclusion that i need a trackball
20:58.35KranZuse ua's which have the 726 codec
20:58.49KranZworks well
21:00.05r_evolutionfuck
21:00.06r_evolutioni quit
21:00.18r_evolutioni got to L8 with 3 extras... and got cocky and lost em :(
21:00.34KranZha
21:00.44KranZok, 1 more game
21:00.49r_evolutionyeah
21:00.56r_evolutionyou say that
21:00.59r_evolutionbut thats like a crackhead
21:01.03r_evolutionok just ONE more rock
21:01.27*** join/#asterisk ddn_ (n=Daniel@200.84.67.165)
21:01.31ddn_hi all
21:01.37*** join/#asterisk iulius (n=iulius@adsl-152-175-71.asm.bellsouth.net)
21:02.38De_Monwhat game?
21:03.37ddn_is it possible to install a voip server over a satellital internet connection?
21:03.56r_evolutionhttp://farm.addictinggames.com/D78AQSAKQLQWI9/1258.swf
21:05.01KranZddn_: yes
21:05.07KranZddn_: will it work well
21:05.15KranZddn_: probably not
21:05.23KranZyou have latency issues and jitter problems
21:05.30ddn_KranZ, so? yes or not.
21:05.40KranZyes you can "install" it
21:05.41ddn_r_evolution, check zuma at popcapgames.com
21:06.04ddn_KranZ, and will work fine. isnt?
21:06.19KranZbut most likely, the connection wont be stable enough for voice traffic
21:06.33KranZare you on the sat link now?
21:06.39ddn_KranZ, ohhhhhhhhhhhhhh
21:06.52ddn_KranZ, no
21:07.27KranZbrb, curveball
21:08.27MACscrsat wont be good at all because of the delay
21:09.02MACscrbecause of the distance it has to travel, sat. internet is not good for voip or online games
21:09.12MACscrfine for browsing or downloading though
21:10.30ddn_anyone interested in installing voip servers in venezuela?
21:11.07KranZif the latency were consistant, it would be alright
21:11.13KranZbut the jitter will kill it
21:11.42justinuany cisco QoS experts around?
21:13.12justinuneed help with flow based fair queueing and diffserv
21:15.22*** join/#asterisk blebleble (i=godie@caesar.godie.net)
21:16.28blebleblei have an extension that should be forwarding to voicemail, however at 5:05 our autoatendant kicks and is overridding it. Basically i have 5 trunks, 4 get auto attendant and one should get voicemail, yet the voicemail one is getting auto attendant anyone have any clues?
21:16.50De_Monhumph lvl 5
21:17.18blebleblede_mon: is that meant for me?
21:17.33r_evolutionafw curveball
21:18.31eKo1justinu: I recommend #cisco
21:18.53TheCompWizSWEET... just got skype to work with asterisk.
21:19.07TheCompWizboth in-dialing & out-dialing.... (one line only :()
21:19.53justinueKo1: i'll check it out (hopefully it exists) :)
21:20.05eKo1oh, it does.
21:20.13eKo1I've been there numerous times
21:21.47justinucool
21:22.00ddn_anyone interested in installing voip servers in venezuela? call 001 619 374 0892
21:23.16Wowzers10is there a way to use ackcall and announce to answer the call play a message, then press # to bridge the call? - this is so the person answering knows its a call for them
21:23.49justinuit be hot
21:25.01r_evolutionJUSTIN!
21:25.04r_evolutionsup homie.
21:25.16ddn_anyone interested in installing voip servers in venezuela? call 001 619 374 0892. ahhh prefer spanish
21:25.24justinuhey man
21:25.28justinuhow's it?
21:25.29r_evolutionhow goes?
21:25.38justinupretty good
21:25.38r_evolutionpretty good... waiting for a DS3 to come in
21:25.47justinuwe're getting a new one as well
21:25.50r_evolutiongot a temp solution as well
21:25.51*** join/#asterisk Ariel_ (n=Ariel@70.46.87.158)
21:25.52justinunew datacenter too
21:25.52r_evolutionchya. needed.
21:25.54*** join/#asterisk redondos (n=redondos@190.48.27.53)
21:25.55r_evolutionnice.
21:26.14*** part/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu)
21:27.00r_evolutionim contemplating 1.2.9.1 on the primary *
21:27.14r_evolutionit's doing ok for the in-house so far...
21:27.49r_evolutionactually seems to do a little better as far as the qualify goes... the phone on my desk would never stay qualified to the primary... but it hasn't lost conn to the in-house
21:28.47redondosCan you please take a look at this log? I can't start asterisk. I've got an E200P card and an X100P card connected. http://pastebin.com/768435
21:29.10brettnemhey anyone have trouble running Asterisk->SDSL->Phones which small to medium offices (5 simult calls or so)
21:29.40C4T3ldoes anyone here use broadcom for voip termination, are they reliable???
21:29.56brettnems/which/with/
21:32.56*** join/#asterisk Rawplayer (i=kevin@ipc31055d2.oom-killer.org)
21:33.02*** join/#asterisk fholmes (n=fholmes@rrcs-24-227-237-197.sw.biz.rr.com)
21:34.12*** join/#asterisk saftsack (n=saftsack@p54A7D890.dip.t-dialin.net)
21:34.22fholmesI am having an issue with my sip softphone.  I cannot get it to register all the sudden.  I have not changed anything in my configuration.
21:35.00C4T3lfirewall??
21:35.03C4T3ldoes anyone here use broadcom for voip termination, are they reliable???
21:35.04justinur_evolution: you do any cisco work?
21:36.09KranZbrettnem: what's the upstream limit?
21:36.23fholmesI keep getting username/auth name mismatch.
21:36.41KranZmake sure your username and authname are the same
21:36.42brettnemKranZ: 2.2M
21:36.53KranZwhat happens when you get 5 calls
21:37.08fholmesIs that two different settings in my sip.conf?
21:37.16KranZit's probably on the client
21:37.27brettnemKranZ: who are you talking to?
21:37.40KranZboth of u
21:37.41KranZheh
21:37.47brettnemheh
21:37.50KranZbrettnem: what happens when you hit 5 calls?
21:38.15brettnemKranz: all calls are choppy and staticy.. other customers on the same asterisk box same router  but different DSL line are just fine.
21:38.16*** join/#asterisk pigpen2 (n=mark@m015f36d0.tmodns.net)
21:38.26brettnemlocal LAN for the phones is isolated from PCs
21:38.45brettnemwerid eh?
21:38.57KranZoh, so you have 2 vlans?
21:39.01KranZone pc and one phones?
21:39.08brettnemno, they arn't vlans
21:39.13brettnemit's
21:39.24C4T3ldoes anyone here use broadcom for voip termination, are they reliable???
21:39.25KranZjust 2 physical lans
21:39.33brettnemasterisk->router->router->dslam=>SDSL=>2xclients
21:40.00KranZso the clients aren't local to the asterisk?
21:40.21brettnemno, in this example, there are two clients.. each on their own DSL circuit (In different bldgs)
21:40.28KranZoh
21:40.30r_evolutionno sir i do not :(
21:40.43KranZand * is attached to both of the dslams?
21:40.53KranZer both routers to each dslam
21:41.17brettnemasterisk is attached to a router, which is attached to another router which is attached to a dslam which has 2 SDSL customers hanging off of it.. one gets good service.. the other no so good
21:42.44MikeJ[Laptop]it's the routers!
21:42.57fileMikeJ[Laptop]: !!!!!!!!!
21:43.02MikeJ[Laptop]hello
21:43.08MikeJ[Laptop]it's the interweb!
21:43.14fileno no
21:43.15fileintarweb
21:43.17r_evolutionINTARWEB!
21:43.20r_evolutionexactly.
21:43.32r_evolutiontin can with a string... that's the solution brett.
21:43.47r_evolutionlow bandwidth... low latency... virtually no jitter.
21:44.32r_evolutionhere's a scary thought...
21:44.38r_evolutionhow does the laziest supervisor in the callcenter
21:44.48r_evolutionbecome the one who's in charge of motivating people to collect money?
21:45.01nahireanr_evolution; That's quite easy.  He knows how to play the game.
21:45.11r_evolutionshe.
21:45.14r_evolutionquoting " someone to set goals for collections, set goals for orders numbers, implement motivational activities and the list goes on"
21:45.15nahireanEven more so
21:45.31r_evolutionno. the person assigning the lazy person to the 'job' is another female
21:45.33r_evolutionand in this case
21:45.37r_evolutionthe game is very easy
21:45.43*** join/#asterisk dropdrive (n=dropdriv@c-66-30-112-44.hsd1.ma.comcast.net)
21:45.44r_evolutionkiss lots of ass
21:45.46r_evolutionthats it :)
21:45.48nahireanRight.
21:45.57r_evolutionhaha no seriously.
21:45.59r_evolutionlots and lots.
21:46.08nahireanYeah, until it's raw and pruned, right?
21:46.20markus99why would my music on hold sound like crickets?
21:46.29brettnemMikeJ[Laptop]: I'm trying to convince my customer it's gotta be at the customer prem
21:46.35*** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com)
21:46.36Wowzers10~jbot thebook
21:46.44Wowzers10~jbot the book
21:46.56Wowzers10jbot thebook
21:47.00Wowzers10hmm
21:47.41r_evolution~book
21:47.43jboti heard book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
21:47.43r_evolution~thebook
21:47.48r_evolutionthere ya go
21:47.53docelm0~mybutt
21:47.54jbotmethinks mybutt is HUGE and stands for some funky stuff...
21:48.07r_evolutiondid you record crickets as teh moh markus?
21:48.15r_evolution~docelmo
21:48.19r_evolution~docelm0
21:48.20r_evolution:(
21:48.56*** join/#asterisk asterisk-dud (n=dwwollma@64-42-247-120.mb.skyweb.ca)
21:49.30asterisk-dudi have three outgoing lines, i would like asterisk to look for an open line from the three and use that to dial out
21:49.46C4T3lhow do you turn off the sip debug feature?
21:50.09fholmesAlright so how do I force a logged in SIP agent to unregister?
21:50.34fholmesI did 'database show' and there is a registration in there for a connection that is no longer good.
21:50.40*** join/#asterisk hads (n=hads@mail.nice.net.nz)
21:50.56*** join/#asterisk Lord_Drachenblut (n=Lord@74.129.228.28)
21:50.58Lord_Drachenbluthello
21:51.20*** join/#asterisk ToTo (n=ToTo@host68-166.pool879.interbusiness.it)
21:52.11C4T3lhow do you turn off the sip debug feature? i dont need all of that info anymore
21:52.16asterisk-dudi have three outgoing lines, i would like asterisk to look for an open line from the three and use that to dial out
21:52.55r_evolutionyou can delete things in teh database holmes
21:53.17CunningPikeC4T3l: sip no debug
21:53.27fholmesSo what exactly is the database?  Is it things that are in use right now basically?
21:53.35r_evolutionno
21:53.41r_evolutionit's just *'s internal database
21:53.46fholmescompared to sip show users which will list all of the users that are in the sip.conf file?
21:53.56r_evolutionit'll list agents and other things
21:54.00r_evolutionwell
21:54.08r_evolutionsip show users will list the users
21:54.15r_evolutionnot just the ones in the sip.conf file
21:54.15JackEstormasterisk-dud: did you define them as a group in zapata.conf? if so you can use that group on dial out.
21:54.16r_evolutioner
21:54.37r_evolutiongame time :-D
21:55.04*** join/#asterisk hohum (n=dcorbe@12.117.204.18)
21:55.41fholmesSo if I do database show and the top listing is : SIP/Reistry/admin  : all the connection info.  Is that something I can safely delete?  (The connection is not valid anymore)
21:55.59fholmes*SIP/Registry/admin
21:56.25MACscrnot sure if anyone has seen this, but i just found it off another link
21:56.25MACscrhttp://www.voipcharges.com/
21:56.28MACscrpretty cool
21:56.30fholmesIn otherwords what is SIP/Registry/username?
21:59.54*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220)
22:05.30*** join/#asterisk symmetre (n=s@198.87.2.15)
22:06.15symmetreok, i have asterisk , now i have to goto a PSTN ..can you do that for free too?
22:06.29MACscrno
22:06.51*** join/#asterisk pdavid (n=chatzill@adsl-068-209-191-127.sip.mob.bellsouth.net)
22:06.59pdavidwhat exactly do i need to do to enable music on hold?
22:09.20justinur_evolution: what game?
22:09.33*** join/#asterisk _CALLNET (n=charly@140.Red-217-125-115.staticIP.rima-tde.net)
22:09.40_CALLNETHello all
22:10.52CunningPikesymmetre: No - at some point you will have to a) buy hardware b) pay a telco c) pay a termination/origination provider d) some combination of all 3
22:11.26CunningPikepdavid: Create a musiconhold.conf
22:11.33CunningPikepdavid: Put a call on hold
22:12.38_CALLNETSomeone have any route to sell us ? We have 60 Millons montly of traffic
22:14.07asterisk-dudi have a channel bank and i would like to start with channel 11 in my zapata and zaptel conf
22:14.20asterisk-dudfor some reason it's not working
22:14.48CunningPikeasterisk-dud: pastebin your files
22:14.51CunningPike~pb
22:14.52jbotextra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/
22:15.00asterisk-dudasterisk accepts it but when it dials, i dials zap/11 on the 11th hole
22:16.17pdavidcunningpike: yeah, i have, but the sound is not quite what i expected ;)
22:16.17*** part/#asterisk MACscr (n=MACscr@66.73.154.70)
22:16.30pdaviddo i need to make the adjustment in the zapata.conf file?
22:16.46CunningPikepdavid: Possibly - what does it sound like?
22:16.47asterisk-dudhttp://pastebin.com/768539
22:17.19pdavidcunningpike: not sure if its the playback, or the goofy default moh music, sort of static and music blended
22:17.52CunningPikepdavid: Are you using mpg123?
22:18.08asterisk-dudcunningpike: the zaptel is at the bottom
22:18.26pdavidcunningpike: was reading that v1.2 does native playing now, so not sure.  using default musiconhold.conf file
22:18.35pdavid[default]
22:18.41pdavidmode=quietmp3
22:18.43pdavidetc..
22:19.09CunningPikepdavid: Give native MOH a go - it helped us out a lot
22:19.31pdavidcunningpike: is there anything special i need to do to set it up?
22:19.42asterisk-dudcunningpike: did u get that?
22:19.59CunningPikeasterisk-dud: On phone - one sec
22:20.08asterisk-dudok
22:21.25CunningPikeasterisk-dud: What does your Dial() statement look like
22:22.17Shaun2222what am i doing wrong with this? exten => _2XX,3,Set(CALLERID(num)=DB(/AgentsMap/${CALLERIDNUM}))
22:22.43Shaun2222it's setting the caller ID to DB(/AgentsMap/301) rather than the value from the key in the db
22:22.46*** join/#asterisk znoG (n=gs@153-129-89-200.fibertel.com.ar)
22:22.49CunningPikeasterisk-dud: Do you have a PRI connected to this card?
22:23.48CunningPikepdavid: There are instructions in the wiki for setting up native MOH - we convert our MOH to ulaw and play it natively
22:24.12pdavidcunningpike: same instructions as at astrecipes.net (transcoding to native formats one time?)
22:24.54CunningPikepdavid: I haven't seen those - we use sox to convert to ulaw from Windows WAV - I can get you the command if you need
22:25.10pdavidhttp://astrecipes.net/?n=152
22:25.15pdavidseems like the same idea
22:25.23pdavidnow that i can hear the music, its wayyyy too slow
22:25.28pdavid?
22:25.54CunningPikepdavid: That's it - sounds like your sampling rate is wrong
22:26.08pdavidcunningpike: yep.  off to fiddle!
22:26.16CunningPikepdavid: Have fun!
22:26.17*** part/#asterisk znoG (n=gs@153-129-89-200.fibertel.com.ar)
22:26.29CunningPikeasterisk-dud: Do you have a PRI?
22:27.08CunningPikeasterisk-dud: OK - you said before - a channelbank
22:28.54asterisk-dudCunningPike: can u see anything worng, or can't this be done?
22:30.09*** part/#asterisk sorush20 (n=sorush20@82-43-184-143.cable.ubr07.newm.blueyonder.co.uk)
22:30.31CunningPikeasterisk-dud: I'm not 100% sure, but your zapata.conf looks like a mix up between FXO and PRI - but I don't have a channelbank, so I'm not sure how these are meant to be set up
22:31.08asterisk-dudwell i'm using fxo ports in my channel bank, what is pri?
22:31.37*** join/#asterisk jarg (n=jarg@200.56.225.61)
22:32.01*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)
22:32.17CunningPikeasterisk-dud: Isn't the connection between asterisk and your channelbank a PRI connection?
22:32.32CunningPike~TE405P
22:33.46asterisk-dudt1
22:34.02asterisk-dudCunningPike: t1
22:34.08asterisk-dudyup
22:34.11asterisk-dudthat's correct
22:34.16*** join/#asterisk ManxPower (n=ewieling@24-179-48-91.static.slid.la.charter.com)
22:34.46CunningPikeasterisk-dud: Right - so signaling of fxo-ks is incorrect for a T1, right?
22:35.03asterisk-dudno
22:35.04CunningPikeasterisk-dud: What does the documentation for your channelbank say?
22:35.11asterisk-dudit works perfectly
22:35.24MikeJ[Laptop]who likes slashdot!   866-387-9249
22:35.27asterisk-dudit's just that I want to start with channel 11 instead of 1
22:35.35MikeJ[Laptop]fun new toys
22:35.44CunningPikeasterisk-dud: OK - we have a PRI direct tot the PSTN, so I'm not the right person to ask
22:35.50*** join/#asterisk anthm (n=anthm@h460852d6.area4.spcsdns.net)
22:35.50*** mode/#asterisk [+o anthm] by ChanServ
22:35.53CunningPikes/ask/answer!/
22:36.22MikeJ[Laptop]or... sip 556@@208.64.200.42
22:36.25*** join/#asterisk ToTo (n=ToTo@host68-166.pool879.interbusiness.it)
22:36.31asterisk-dudjbot: are you the right person?
22:36.48asterisk-dudi din't think thats the issue, the issue is with asterisk and the zap channels
22:36.51Dr-Linuxasterisk-dud: yes he is
22:36.55drrayjbot is righgt as rain
22:37.08asterisk-dudjbot: http://pastebin.com/768539
22:37.10MikeJ[Laptop]~slashdot
22:37.12symmetreis it PRI or PSTN ?    are these two different ways to get a call out to the telephone system?
22:37.12asterisk-dudcan u look at that?
22:37.23asterisk-dudPRI
22:37.37CunningPike~pstn
22:37.39jboti heard pstn is Pubic Switched Telephone Network, or "please stop the nonsense"
22:37.42*** join/#asterisk JINDAL (n=root@220.226.36.2)
22:37.44drray~pinkworld
22:37.46CunningPike~pri
22:37.48jboti heard pri is Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, etc.
22:37.48*** part/#asterisk JINDAL (n=root@220.226.36.2)
22:38.03asterisk-dudi am usingPRI
22:38.10symmetreyeah i read about pri on wiki, that info is pasted from there...
22:38.17drrayheh
22:38.29CunningPikesymmetre: So, a PRI is one of the ways to connect to the PSTN
22:38.33symmetrei just wonder, if this is a choice,...  do you select pri or pstn.. or they 2 different tools for different things?
22:38.51asterisk-dudjbot: did u take a look?
22:38.55*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-85-84.buckeyecom.net)
22:39.16Dr-Linuxlol
22:39.32Dr-Linux~jbot
22:39.34jbotjbot is probably only marginally useful at best,  He got a C- on his Turing Test, or a complete idiot
22:39.51*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
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22:39.59asterisk-dudwhatever
22:40.09*** join/#asterisk znoG (n=gs@153-129-89-200.fibertel.com.ar)
22:40.11tzafrirasterisk-dud, if you wantr channels to start from 11, you better have something to create channels 1-10 for you . Is there any other zaptel hardware on that system?
22:40.22fholmes~jbot
22:40.24jbotrumour has it, jbot is only marginally useful at best,  He got a C- on his Turing Test, or a complete idiot
22:40.24tzafrirCould you pastebin your zaptel.conf?
22:40.29znoGdoes anyone use/run/own/whatever a RT31P2 unit?
22:40.37asterisk-dudtzafrir: http://pastebin.com/768539
22:40.44asterisk-dudthat is the porbole
22:40.46asterisk-dudproblem
22:40.59asterisk-dudcan i create 10 dummy channesl
22:41.15tzafrirthat is /etc/asterisk/zapata.conf . I meant /etc/zaptel.conf
22:41.31asterisk-dudsorry bud
22:41.45asterisk-dudzaptel.conf is on the end
22:41.46asterisk-dudof that
22:41.58tzafrirAnyway, you can also create that offset in e.g. the dialplan
22:42.26asterisk-dudi know, but i plan to add channels that would go into 1-10,
22:42.34asterisk-dudso i want to leave them blank
22:42.54ManxPowerasterisk-dud, you can't do that
22:43.18asterisk-dudso i can't create a dummy channel?
22:43.32ManxPowerasterisk-dud, channel numbers are not "things you think up" they are tied to the hardware.
22:43.41tzafrirOr patch zaptel to begin from 10? :-(
22:43.42MikeJ[Laptop]asterisk-dud, what ManxPower said
22:43.52drraywhy do you want skip the first 10 anyway?
22:43.52ManxPowerChannels start at 1, which is the first channel that is found.
22:44.04tzafrirThey are not tied to hardware. They are just in the order of registration
22:44.04MikeJ[Laptop]there has been talk to change that.. but you can't right now without changing code
22:44.06asterisk-dudit's more convenient
22:44.21asterisk-dudfor the system i'm setting up
22:44.33ManxPowerasterisk-dud, I guess it sucks to be you.
22:44.37asterisk-dudbut i'll just use the math function to
22:44.43drraymy outbound PRI is 71-83
22:44.44asterisk-dudyuop
22:44.50drrayyou just hard code it
22:44.55tzafrirSo how can I create "dummy" channels? A span that will just register 10 channels and do nothing more?
22:45.15tzafrirIt's not complicated to write
22:45.22asterisk-dudexcellent question
22:45.30*** join/#asterisk jeffik (n=Jeff@kns221.NetSurf.Net)
22:45.47asterisk-dudwell, u're the expert, can u guide me in the right direction
22:46.01ManxPowerasterisk-dud, If you fight Asterisk you will be unhappy.  If you take a Zen-like approach and accept Asterisk's oddities, your life will be much better.
22:46.10asterisk-dudok
22:46.16asterisk-dudthanks' for the advice,
22:46.20symmetreif you set up a 'farm' of asterisk boxes..  would you have failover security?
22:46.22asterboybye
22:46.28tzafrirhow about ztd-local? Can it be of some help?
22:46.34*** join/#asterisk h0 (n=h0@ool-44c69453.dyn.optonline.net)
22:46.37ManxPowerasterisk-dud, Well, first start out by learning C.  Then, after learning C, study the zaptel code.
22:46.39tzafrirI just never used it
22:46.43symmetreasterisk-dud linux systems will hurt you
22:46.44asterisk-dudwhat is ztdummy?
22:46.54*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)
22:47.05ManxPowerasterisk-dud, it's a driver to provide timing without having actual channels
22:47.08tzafrirztdummy creates just a span with no channels. For timing from the system clock
22:47.17asterisk-dudok
22:48.27CunningPike~ztdummy
22:48.28jbotztdummy is, like, a driver that interacts with zaptel to provide a timing source to Asterisk. On 2.4.x kernals, timing is obtained from a UHCI USB controller. It will not work with OHCI controllers. On 2.6.0 and later kernels, the timing is provided by the kernel, thus no hardware is required at all.
22:48.42ManxPowerasterisk-dud, Oh, BTW, DON'T PUT QUOTES IN CALLERID
22:51.25CunningPikesymmetre: What we have done is to have two identical servers, configured exactly the same. If the first one fails, our Polycoms use DNS SRV to fail over to the second server. We physically move the PRIs from one server to the other
22:53.06Dr-LinuxCunningPike: what if DNS is working fine but something things fails?
22:53.21CunningPikeDr-Linux: Like?
22:54.13Dr-LinuxCunningPike: asterisk service? email server or apache?
22:54.21symmetreok let me ask this way: what are all the different ways to connect asterisk to the pstn (i.e. PRI is one such way)?
22:55.02CunningPikeDr-Linux: If the Polycoms can't register with the main server, they will fail over to the other server
22:55.39CunningPikesymmetre: Usually, with a PRI, a channelbank, or a FXO interface
22:55.48Dr-LinuxCunningPike: what tool you are using to identify this failover?
22:56.08CunningPikeDr-Linux: We're not - the Polycoms know if they've lost their registration
22:56.15symmetrechannelbank and fxo are both pieces of hardware arent they?
22:56.36CunningPikeDr-Linux: Plus we use Intermapper with a modified version of the Nagios plugin for monitoring
22:56.46Dr-LinuxCunningPike: only polycom does that or other softphones etc as well?
22:57.03CunningPikesymmetre: Yes - you need some hardware to connect - or a service provider
22:57.05Dr-Linuxwe are using nagios as well
22:57.51CunningPikeDr-Linux: I think some of the other hardphones use DNS SRV too - I don't know of any softphones that do
22:58.32CunningPikeDr-Linux: Right now, we are just using an IAX ping, but I see there is a Nagios plugin for zaptel that I would like to try
22:58.33h0hello guys I am new to Asterisk and want to start experimenting with it in a home environment. Would you guys recommend the SPA-3000 as a analog telephone adapter and is there any other hardware that I will need to get started other then a linux server
22:59.19CunningPikeh0: The SPA-3000 is a good ATA and all you will need
22:59.22*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
22:59.22Dr-LinuxCunningPike: what does that nagios zaptel plugin?
22:59.34CunningPikeJust saw it on the list
22:59.40CunningPike~thelist
22:59.41jbotsomebody said thelist was the asterisk-users mailing list. Sign up or view archives at http://lists.digium.com/mailman/listinfo/asterisk-users
22:59.44CunningPike;)
22:59.45asterisk-dudwhats wrong with this: Set(${EXT}=${MATH($EXTEN}-2010)})
23:00.04shmaltzanybody here heard of voipteck?
23:02.04*** join/#asterisk anthm (n=anthm@adsl-68-248-236-217.dsl.milwwi.ameritech.net)
23:02.04*** mode/#asterisk [+o anthm] by ChanServ
23:02.13CunningPikeDr-Linux: Look for a posting entitled 'asterisk nagios plugin' from yesterday
23:03.24asterisk-dudwhats wrong with this: Set(${EXT}=${MATH($EXTEN}-2010)})
23:03.49*** part/#asterisk jarg (n=jarg@200.56.225.61)
23:05.19CunningPikeasterisk-dud: Nothing, unless you're getting an error
23:05.53anthmomit $ in the ${EXT}
23:06.44Shaun2222Jun  9 07:03:04 ERROR[31489]: pbx.c:1380 ast_func_read: Function AGENT not registered
23:06.54Shaun2222anybody know what module i need loaded to use the AGENT function?
23:07.02*** join/#asterisk Bullseye_Network (n=Kyle@216.143.192.69)
23:07.21Bullseye_NetworkQuick question if I didnt install SOX would that cause voicemails to not record?
23:09.12asterisk-dudthe prob was i forgot the { before EXTEN
23:09.17asterisk-dudiam a dumbass
23:11.05anthmunless they changed it Set(${EXTEN}=something) will make a new var that has the name of what ${EXTEN} evals to
23:13.13*** part/#asterisk mogorman (i=ejabberd@68.62.237.103)
23:14.13r_evolutionCHYA!!!
23:14.24anthmBullseye, no
23:14.36anthmaww he'll never know now
23:14.44r_evolution:(
23:14.53r_evolutioni kinda figured it'd be a lil one-way audio issue
23:18.09*** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)
23:18.13generalhanWhats up all
23:19.02generalhanhas anyone ever had any issues with a 7960 and the message light staying on even when there are no messages ??? i need some possible causes for this cause its starting to make my boss angry ... lol
23:19.19generalhanhe sees red and it makes him mad ! hahaha !
23:20.23*** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it)
23:20.31anthmwrite a cron to send him a vmail saying "you have no new messages just letting you know" constantly
23:20.32*** part/#asterisk h0 (n=h0@ool-44c69453.dyn.optonline.net)
23:22.09asterisk-dudis there a way to cancel absolute timout after it's been activated
23:23.39asterisk-dudyes, reset it to zero :D
23:23.56luke-jr_...
23:24.55*** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon)
23:25.34generalhananthm: thats good .. but still doesnt get rid of the red light ! lol
23:27.10*** join/#asterisk litage (n=nick@203.220.55.70)
23:28.44*** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
23:29.33asterisk-dudwhats a good soundcard to use with asterisk
23:30.16tzafrirOnce again: anybody here uses res_zeroconf?
23:30.18De_Monhuh?
23:30.35De_Monasterisk-dud huh?
23:30.47CunningPikeasterisk-dud: asterisk doesn't need a soundcard, per se.....
23:30.52tzafrirasterisk-dud, chan_alsa
23:31.02De_Monchan_alsa isn't a sound card!
23:31.09tzafrirNot really important, though
23:31.33tzafrirAh, what make?
23:31.51tzafrirI figure that any full-duplex card will do. How can you tell that?
23:32.19tzafrirmy laptop's sound card isn't :-(
23:32.30De_Monwhy do you need full duplex?
23:32.51CunningPikeWhy do you need a soundcard?
23:33.17De_MonCunningPike console audio
23:33.44CunningPikeDe_Mon: Oh
23:33.57asterisk-dudCunningPike: Public Address System
23:34.07CunningPikeasterisk-dud: Ah
23:34.38De_Monhey.. thats not a bad idea
23:35.14asterisk-dudi'm going to try SB Live
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23:44.14ryguillianclear
23:44.29*** join/#asterisk JINDAL (n=root@220.226.36.2)
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23:53.37Shaun2222is there a way to reset the ${EXTEN} ?
23:53.55Shaun2222i tryed Set(EXTEN=1949${EXTEN})
23:54.01Shaun2222but it doesnt appear to be working

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