irclog2html for #asterisk on 20050903

00:03.47*** join/#asterisk popvoxdave (i=user@dave2.toad.net)
00:08.20*** part/#asterisk Malthus (n=dumbdumb@port0005-adb-adsl.cwjamaica.com)
00:08.59*** join/#asterisk JamesDotCom (i=jamesdot@sweep.bur.st)
00:09.17*** join/#asterisk Beccara (n=Beccara@60-234-136-147.bitstream.orcon.net.nz)
00:09.49*** join/#asterisk CoolAcid (n=jason@216.99.98.39)
00:11.00*** join/#asterisk DougRoyer (i=doug@70-67.69-92-cpe.cableone.net)
00:12.32*** join/#asterisk pauldy (n=pauldy@c-67-162-191-108.hsd1.tx.comcast.net)
00:13.55*** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
00:14.39redder86Does Asterisk always require a 60-second refresh?  I'm using libiax and no matter what I tx in the registration request for a refresh I always get a response indicating a 60-second refresh.
00:15.49*** join/#asterisk bjohnson (n=bjohnson@i216-58-58-178.cybersurf.com)
00:18.24*** join/#asterisk yogurt2ungue (n=yogurt2u@44-170-114-200.fibertel.com.ar)
00:21.50*** join/#asterisk oe1 (n=oej@63.77.68.196)
00:26.31*** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net)
00:30.39*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
00:31.03*** join/#asterisk wolfson` (i=hehe@kdh-res-4.beachlink.com)
00:35.22*** join/#asterisk CoolAcid (n=jason@216.99.98.39)
00:36.50*** part/#asterisk cj (n=cjcollie@pdpc/supporter/monthly-bronze/CJ)
00:37.54*** join/#asterisk NirS (n=Nir@84.94.52.50.cable.012.net.il)
00:38.43*** join/#asterisk da_monumental_1 (n=da_monum@rrcs-24-172-102-198.midsouth.biz.rr.com)
00:42.07*** join/#asterisk zippp (n=zip@68-118-126-104.dhcp.sprn.tx.charter.com)
00:51.41Rav1974not a lot of cativity today :)
00:52.09fugitivoit's friday
00:52.36Rav1974:) I don't have a life
00:53.19Rav1974when I installed a x100p after a TE110p aterisk dies
00:57.04*** join/#asterisk used_ (n=used@c-24-22-125-179.hsd1.or.comcast.net)
01:03.28*** join/#asterisk DrukenHME (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com)
01:03.28shido6?
01:03.54Qwell!
01:03.59MikeJ[Laptop]*
01:05.23Rav1974when I installed a x100p after a TE110p aterisk dies, any suggestions?
01:05.44Qwelldies how?  and don't repeat
01:06.01redder86anyone here that can assist with using libiax2 ?
01:06.17Qwellredder86: if you ask a question about it
01:06.19*** join/#asterisk CoolAcid (n=jason@216.99.98.39)
01:06.19Rav1974:) It says there is something wrong with tor or zap
01:06.29QwellRav1974: that help none
01:07.11*** join/#asterisk NirS (n=Nir@84.94.52.50.cable.012.net.il)
01:07.12Qwellhelps*
01:07.21Rav1974Qwell: I know, but i'm a newbie, I am rebooting the server, give me a minute
01:07.27Qwellrebooting?!
01:07.42QwellI hope you just compiled a new kernel or something
01:07.57Rav1974Qwell:I'm from windows world, if it doesn't work reboot
01:08.01Qwell...
01:08.05Rav1974if it still doesn't work kick it
01:08.16Rav1974if it STILL doesn't work, then it won't ever work
01:08.24Rav1974chuck it
01:08.26*** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net)
01:08.36QwellRav1974: let me send you my mailing address
01:08.37Rav1974I think I know what is wrong, my zaptel or zapata config files
01:08.56QwellI'll even pay for shipping
01:08.58Rav1974Qwell: do you want some kicked hardware
01:09.11Rav1974I have many pieces hope you have glue
01:10.44redder86Qwell: I'm having trouble using the return value of iax_get_event.
01:11.25redder86Qwell: for example, after getting the REGACK frame I want to reset an internal refresh value (so that I can refresh the registration in time).
01:11.35*** join/#asterisk prav33n|home (n=praveen@59.144.38.160)
01:11.41redder86Qwell: how do I use iax_get_event?
01:11.46Qwellgot me
01:11.47prav33n|homeHello
01:11.49Rav1974this is to be continued... Qwell: thanks for helping.  I have some kids to play with
01:12.06*** join/#asterisk used_ (n=used@c-24-22-125-179.hsd1.or.comcast.net)
01:12.09Rav1974i deeply appreciate all the helpful people who keep coming back to help us newbies
01:12.16redder86Qwell: you don't use iax_get_event, then?  What do you use instead?
01:12.23Rav1974anyone in NYC, I'd buy you a beer
01:12.27Qwellredder86: I never said I did
01:12.33prav33n|homeI have ordered Linksys WRT54GP2-NA to use with astreisk from VeriLAN.com
01:12.48prav33n|homeAssuming that this model is not locked to anyone
01:12.59prav33n|homeCan I use it fruitfully?
01:13.22prav33n|homeI need some help here, please.
01:13.53redder86Qwell: ah, so why'd you want me to ask the question if you don't know about libiax2?
01:15.06*** join/#asterisk onhold (n=maxgluck@200.109.166.83)
01:16.13Qwellredder86: so that you wouldn't have to keep repeating yourself
01:16.31Qwell"Does anybody know how to fix this error?"
01:16.35QwellWHAT error?  ya know?
01:16.46redder86Qwell: I specified that the error was with using libiax2
01:16.48Qwellif somebody sees it, they might answer why
01:17.02onholdHi, I need some advice please. I've configured musiconhold.conf and the calls are placed on hold, I have mp3 files, without tag in the /var/lib/asterisk/mohmp3 directory but no music is heard at all. Does asterisk installs mpg123 or I have to do it?
01:17.08prav33n|homeAnyone here have some idea on Linksys WRT54GP2-NA
01:17.20Qwellredder86: see, what onhold said was an example of a good question
01:17.41Qwellonhold: from the asterisk source dir, do `make mpg123`.  It'll download and compile it for you
01:17.52sivanatzanger: ping
01:17.55*** join/#asterisk NirS (n=Nir@84.94.52.50.cable.012.net.il)
01:18.05onholdthanks Qwell, let me try that... =)
01:18.09redder86Qwell: I asked whether anyone or not was listening that knew about libiax2 enough to answer some questions about its usage.  The answer to that question was apparently no.
01:18.10Qwellactually, it might just download it, then you probably have to `cd mpg123 && make && make install`
01:19.21onholdyes, it created the directory mpg123-0.59r
01:19.35Qwellonhold: go in there, and `make && make install`
01:21.15*** join/#asterisk ManxPower (i=eric@1Cust721.an4.dfw28.da.uu.net)
01:23.09*** join/#asterisk pauldy (n=pauldy@c-67-162-191-108.hsd1.tx.comcast.net)
01:24.17*** part/#asterisk prav33n|home (n=praveen@59.144.38.160)
01:26.30onholdQwell, nothing... same happens. May it have something to do with the format not being configured? I'm entering moh classes show, I see the classes but only slin format. I really have searched over and over but can't find where to add mp3 format, if that is what's missing.
01:27.42onholdQwell: moh files show doesn't display anything, but the files are there...
01:29.08*** join/#asterisk huslage (n=used@c-24-22-125-179.hsd1.or.comcast.net)
01:30.46*** join/#asterisk shido6 (n=shido6@d57-87-253.home.cgocable.net)
01:31.39*** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
01:34.05*** join/#asterisk hubbaba (n=me@71-12-171-64.dhcp.gsvl.ga.charter.com)
01:36.32*** join/#asterisk Nir_S (n=Nir@84.94.52.50.cable.012.net.il)
01:37.35Qwellonhold: doesn't show any for me either
01:42.19*** join/#asterisk visik7_ (n=mierda@host70-140.pool8258.interbusiness.it)
01:45.21L|NUXhow can i update my * server on DUNDi graph ?
01:45.47*** join/#asterisk drbrown (n=chatzill@63.238.118.90)
01:46.40*** join/#asterisk cianhughes (n=cian@cian.ws)
01:49.15*** join/#asterisk loud (n=ariel@cypher.punk.net)
01:50.27*** join/#asterisk cianhughes (n=cian@cian.ws)
01:50.38*** join/#asterisk musictomyears (n=maxgluck@200.109.166.83)
01:51.48musictomyearsQwell: thank you so much, there was just one thing left, the missing link!
01:51.56Qwellwhats that?
01:52.19musictomyearsln -s /usr/local/bin/mpg123 /usr/bin/mpg123
01:52.29Qwellhmm
01:52.37Qwell/usr/local/bin/ not in your path or something?
01:53.01*** part/#asterisk hubbaba (n=me@71-12-171-64.dhcp.gsvl.ga.charter.com)
01:54.23musictomyearsapparently not by default, it's a new system, but on the wiki it is commented
01:55.02musictomyearshttp://www.voip-info.org/tiki-index.php?page=Asterisk+config+musiconhold.conf
01:55.44*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
01:58.15musictomyearsThanks, have a great weekend, and a good night...
01:58.19*** join/#asterisk cianhughes (n=cian@cian.ws)
01:58.45*** join/#asterisk iq (n=iq@209-180-97-92.omah.qwest.net)
02:00.37*** join/#asterisk oej (n=oej@63.77.68.196)
02:01.21Qwellhow is the polycom 301 for a cheap phone?
02:01.30DaPrivateerits nice Qwell
02:01.46DaPrivateeronly problem is it doesnt re-register sometimes after a loss of network
02:01.57DaPrivateeryou need to reboot the phone, and it takes about 4 minutes to load
02:02.14QwellWhat else is fairly cheap?
02:02.20Qwell(besides GS)
02:02.44DaPrivateer*shrug*
02:03.09DaPrivateerif you are gonna go with Polycom i would recommend going for the 501 though. i believe its only a differnce of 30 dollars but there are a bunch more features
02:03.27Kattybeep.
02:04.52filebeep beep
02:05.39MikeJ[Laptop]tickling?  what is that all about?
02:05.45Kattyfile: ...
02:06.11*** join/#asterisk cianhughes (n=cian@cian.ws)
02:07.01MikeJ[Laptop]does anybody in here have an audiocodes UA?
02:07.06filenope
02:07.46MikeJ[Laptop]file, do you have the ability to send dtmf notify via ser?
02:08.29fileif you set it up to route that method, yeah
02:08.32fileSER is just a proxy though
02:08.45*** join/#asterisk JamesDotCom (i=jamesdot@sweep.bur.st)
02:08.48MikeJ[Laptop]I need to be able to convert to it
02:08.54MikeJ[Laptop]I don't have anythign that can send it
02:09.07filethen no.
02:09.17MikeJ[Laptop]hmmm do any of the sip test stacks support notify?
02:09.20MikeJ[Laptop]sipp or somthing
02:09.26filedoubt it
02:09.30MikeJ[Laptop]dtmf notify specifically
02:09.47MikeJ[Laptop]damn, I just want to test this one damn bug and figure it out..
02:09.57visik7_does anyone can give me an extract of dialplan for personalized (I mean selectable extension for park and other ) supervised call transfer ?
02:14.00*** join/#asterisk cianhughes (n=cian@cian.ws)
02:15.21*** join/#asterisk Assid (n=assid@203.115.64.60)
02:15.22Assidheya
02:16.30Ariel_OK I am almost giving up on a system. Argh I hate to do this.... Why are there so many different distro's out there? Why?
02:16.47MikeJ[Laptop]Ariel_, windows? :D
02:17.03Assidhrmm.. tell me abt it.. my xwindows stop working.. and i frogot what made it do that
02:17.07Ariel_In my view this is just as bad as windows.
02:17.13Assidand that too just in time since i got my KVM switch
02:17.58Ariel_I have a SuSe setup that just does not want to run php4's correctly.  not allowing me to get it to work with a mySQL database....
02:19.13iqHi, anyone know how to reset SPA-3000 ATA to factory default?
02:19.31*** join/#asterisk cianhughes (n=cian@cian.ws)
02:20.03Assidhow do i come to know which version of CVS head i am using?
02:20.21Ariel_show version
02:20.33Assidjust says CVS HEAD
02:20.59QwellThen its cvs head
02:21.08Qwellcvs isn't really versioned
02:21.16MikeJ[Laptop]show version should show a date\time as well
02:21.32QwellMikeJ[Laptop]: thats compile date/time, right?
02:21.36Qwellshow version files
02:21.44*** join/#asterisk izo (n=izo@193.202.114.43)
02:21.54QwellThats as close as you'll get to the actual "version"
02:22.00MikeJ[Laptop]ummm.. I don't recall now
02:22.07MikeJ[Laptop]I do beleive so
02:22.18MikeJ[Laptop]ahhh.. gotta love CVS... bleh
02:22.22AssidAsterisk CVS-HEAD built by root@voip on a i686 running Linux on 2005-07-29 07:17:46 UTC
02:22.29Assidthink its time to upgrade?
02:22.38Qwelldaily
02:22.50*** join/#asterisk lters (n=lters@mrtcdsl-034.mis.net)
02:22.56Assidhehe.. nah.. actually that server is production
02:23.00QwellI have a cron running every 45 minutes that downloads and compiles latest
02:23.15Assidi could do it on my test bed box
02:23.18pr0mumm.. overkill.  ;-)
02:23.27QwellI was kidding :p
02:23.30MikeJ[Laptop]Qwell, that;s kinda dangerous with a project on cvs...
02:23.31MikeJ[Laptop]heh
02:23.31MikeJ[Laptop]ok
02:23.45QwellI'd do it if it was svn though :D
02:23.45Assiddo i use the same cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds
02:23.49MikeJ[Laptop]bout to say, without atomic commits, you'd be getting half commited stuff all the time
02:23.53QwellAssid: still got the old source?
02:23.57Assidyep
02:23.57QwellAssid: if so, just cvs up
02:24.22Assiddunno :(
02:25.11*** join/#asterisk cianhughes (n=cian@cian.ws)
02:25.59Assidcant find a package called cvsup in debian
02:26.31QwellAssid: cvs up
02:26.34Qwellnot cvsup
02:27.10Assiddo i have to set cvsroot etc ?
02:27.23Qwellif you're in the dir, it should just know
02:28.31Assidathlonxp:/usr/src# cvs up asterisk-sup
02:28.32Assidcvs update: cannot open CVS/Entries for reading: No such file or directory
02:28.32Assidcvs [update aborted]: no repository
02:28.43*** join/#asterisk cianhughes (n=cian@cian.ws)
02:29.42MikeJ[Laptop]asterisk-sup?
02:30.18KattyHmmhesays: are you out drinking yet?
02:30.34Assidyeah.. thats my file i created
02:31.04MikeJ[Laptop]Assid, huh?
02:32.13Assidguess i will just re-download the whole thing
02:33.15Assidhrmm dont i need zapata as well.. for x100p ?
02:33.21*** join/#asterisk apardo (n=apardo@23.Red-83-44-179.pooles.rima-tde.net)
02:33.29Qwellwhy not just type what I said?
02:33.32*** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net)
02:33.38Qwellcd asterisk && cvs up
02:34.03*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
02:34.33BhaalAnyone else here using FWD via iax2 and their asterisk server is constantly reconnecting to FWD?
02:34.35*** join/#asterisk cianhughes (n=cian@cian.ws)
02:35.00Assidyeah that seems to work
02:36.01Assiddont i need zapata fro x100p ?
02:37.38*** join/#asterisk JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net)
02:39.09Kattygod bless yoga pants.
02:39.19file[laptop]Katty: that's hot
02:39.27*** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net)
02:39.30Kattyfile[laptop]: they're so /soft/
02:39.37Kattyfile[laptop]: i love outfits that feel like pjs.
02:41.01MikeJ[Laptop]pants?
02:41.24MikeJ[Laptop]Assid, yes, x100p is zap
02:42.18Assidhrmm ..
02:42.19Assidmake: *** No rule to make target `doc/ztcfg.sgml', needed by `doc/ztcfg.8.gz'.  Stop.
02:42.33*** join/#asterisk cianhughes (n=cian@cian.ws)
02:42.42KattyMikeJ[Laptop]: pants for yoga.
02:43.12Netgeeks_theus the term 'yoga pants'
02:43.27KattyMikeJ[Laptop]: http://www.campus-classics.com/images/Product/img1-15371.jpg <- not me, but she does have yoga pants.
02:43.50*** join/#asterisk ericm_06 (n=ericm_06@c-24-9-147-88.hsd1.co.comcast.net)
02:44.01*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
02:44.06Kattythose must have been some scary pants.
02:44.10KattyMikeJ[Laptop]: did you get url?
02:44.42Netgeeks_he's too busy staring at the url
02:44.53Kattyi wouldn't doubt it.
02:44.56MikeJ[Laptop]what url?
02:44.56Kattyshe's hot
02:45.03Katty21:51 < Katty> MikeJ[Laptop]:
02:45.03Katty<PROTECTED>
02:45.03Katty<PROTECTED>
02:45.23MikeJ[Laptop]no.. I got bounced off..
02:45.42Netgeeks_she's 'get you in jail' young too
02:45.51MikeJ[Laptop]what makes those "yoga" pants
02:46.05KattyMikeJ[Laptop]: that's what the tag said.
02:46.07QwellMikeJ[Laptop]: corporate branding
02:47.04Netgeeks_actually it's probably just like martial art's pants, the design of the crotch area allows for much greater leg motion without stress or binding
02:48.05*** join/#asterisk cianhughes (n=cian@cian.ws)
02:50.05Assidumm.. what am i missing? i am still getting this error
02:50.05Assidmake: *** No rule to make target `doc/ztcfg.sgml', needed by `doc/ztcfg.8.gz'.  Stop.
02:52.16*** part/#asterisk ericm_06 (n=ericm_06@c-24-9-147-88.hsd1.co.comcast.net)
02:52.19fugitivoasterisk compiled in 1.49 min on a dual opteron
02:52.22Netgeeks_amazingly quiet here all of the sudden
02:52.47Netgeeks_fugitivo:  yeah,  ran time make on a X2 4800+ the other day came in at 1m 12s
02:53.14fugitivothat's better
02:53.23Assidit compules for you?
02:53.27Assidhow come im getting errors?
02:53.28Netgeeks_and runs
02:53.36fugitivoAssid: i use gentoo
02:53.37Assidyour using head?
02:53.42*** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net)
02:53.42Netgeeks_*shrug* my guess would be dependancies
02:54.08Netgeeks_yes, I compiled 2005-08-28 head with 8 different patches from mantis
02:54.13fugitivogentoo does all for you!
02:54.14fugitivo;)
02:54.31*** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn)
02:54.37Netgeeks_and I compiled it on redhat FC4 x86_64 with a 'install everything' dist
02:55.17Netgeeks_did you build zaptel first?
02:55.57Assidim tyring to
02:56.02Assidit keeps giving me a stupid error
02:56.08Netgeeks_you are trying to build zaptel?
02:56.20Ariel_Assid, your doc directory is not there.
02:56.26Qwellcvs up -d
02:56.38Ariel_did up rm -fr zaptel directory before you download the new build?
02:56.41Assidfound it.. docbook-utils wasnt installed
02:56.49Assidstill doesnt help
02:57.19Assidthere we go.. -d did the trick
02:57.49*** join/#asterisk cianhughes (n=cian@cian.ws)
03:01.42*** join/#asterisk cianhughes (n=cian@cian.ws)
03:02.27Netgeeks_I was going to drop a couple 4 port cards into the X2 and loop them and see how it would run 192 channels of SIP-ZAP-SIP, but alas the motherboard locked solid after post with the cards installed
03:02.45BhaalIs anyone else here experiencing problems with FWD's iax2 gateway?
03:02.47Netgeeks_Never could get past the post screen
03:05.26visik7_is there a better call parking around ?
03:06.04*** join/#asterisk opus_ (n=opus@dahphish.org)
03:06.28opus_Hello
03:06.40opus_is anyone experiencing Realtime database problems with CVS HEAD?
03:06.51opus_http://pastebin.ca/22016
03:07.01opus_I'm getting "Unkown database 'asterisk'
03:07.09opus_however it succesffuly connects to mysql
03:07.10opus_(?)
03:07.21zipppis there an asterisk db?
03:07.25fugitivodo you have a database called "asterisk" ?
03:07.32zipppmysql -u root
03:07.33opus_i have it called 'asterisk_realtime' instead
03:07.46fugitivothen, change the name or change the config
03:07.52zipppthat is the problem, it is looking for asterisk
03:07.55Assidwats realtime database?
03:07.59opus_well, i'm more interested in why its not working
03:08.04Netgeeks_~wiki
03:08.15Netgeeks_gah
03:08.18Netgeeks_damn jbot
03:08.27zipppopus_, because there is no `asterisk` db, only a `asterisk_realtime`
03:08.38Netgeeks_There is a good description of Asterisk realtime on www.voip-info.org, Assid
03:08.45fugitivoopus_: because it's look for a database called "asterisk" and you don't have one
03:08.45opus_realtime database = database driven asterisk backend, for all .conf files queues/voicemail/sip/iax-channels
03:09.19opus_zippp - but none of my configs have 'asterisk' anywhere only asterisk_realtime. :(
03:09.25fugitivojust, read the messages :)
03:09.31*** join/#asterisk cianhughes (n=cian@cian.ws)
03:09.39Assidyep,,, reading it
03:10.17zipppopus_, `grep -r asterisk /etc`
03:10.23zipppread each line and make sure
03:10.36zipppif you don't want to check _everything_ just grep in /etc/asterisk
03:10.51*** join/#asterisk JamesDotCom (i=jamesdot@sweep.bur.st)
03:11.02opus_zippp that was the first thing I did
03:11.09Kattyhmm.
03:11.12opus_http://pastebin.ca/22017 is my res_mysql.conf
03:11.51zippptry this
03:11.55zipppmysql -u asterisk -p
03:13.06*** join/#asterisk cianhughes (n=cian@cian.ws)
03:13.10Assidhrmm nice.. everything sql based
03:13.19Assideasier to make reporting/front end for this
03:13.43opus_echo select \@\"hello world\"\;| mysql -uasterisk -pasterisk asterisk_realtime
03:13.46opus_echo $? = 0
03:13.49Netgeeks_yeah, but careful, realtime isn't fully ready for primetime
03:13.55Nuggetneither is mysql.  :)
03:14.05Assidwas referring to pgsql ;)
03:14.07Netgeeks_same could be said for asterisk too
03:14.09Nuggetyay  :)
03:14.21opus_same could be said about anything
03:14.32opus_nobody's willing to help
03:14.38opus_then it will stay that way:)
03:14.40Netgeeks_yeah, I find it pretty impressive that yahoo's entire backend is... mysql
03:14.49Nuggetit's friday night.  I'm drunk.  ask again during business hours.
03:14.58Assidreally? i thought only yahoo finance
03:15.27Netgeeks_nah, apparently in a recent interview thier head architect said the whole backend was mysql
03:15.38opus_yeah right.
03:15.42Assidhrmm
03:15.47opus_:)
03:16.00opus_and AT&T runs asterisk realtime
03:16.11Assidisnt pgsql supposed to be faster for more complex queries?
03:16.13Netgeeks_*shrug* if all you are doing is tons and tons of reads....  mysql would be fine
03:16.34Qwellgoogle uses mysql
03:16.44opus_insert into table1 select * from table2, swap 1/2 around and it will eventually corrupt your data. i love mysql
03:16.54NuggetI can't think of any reason to choose mysql over pgsql in any circumstance, unless you have software that only supports mysql.
03:17.28opus_nugget thats exactly why i am using mysql. the least buggiest in my opinion with asterisk realtime from my experience..
03:17.35Netgeeks_I think nugget hit the nail on the head.... I wonder what yahoo's choices where when they started down the road with mysql
03:17.54Assidsave money?
03:17.57Qwelldoes pgsql really scale?
03:18.03QwellAssid: pgsql is truly free, afaik
03:19.01*** join/#asterisk cianhughes (n=cian@cian.ws)
03:19.21*** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net)
03:20.54*** join/#asterisk SplasPood (n=jwb@207-237-244-166.c3-0.nyw-ubr2.nyr-nyw.ny.cable.rcn.com)
03:22.34h3xpostgres rules
03:22.52h3xyou know, informix is based on postgres
03:23.09h3xalthough postgres is way better now
03:23.24Nuggetpgsql is freer than mysql.
03:25.01*** join/#asterisk [2]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net)
03:26.53*** join/#asterisk cianhughes (n=cian@cian.ws)
03:28.29*** join/#asterisk ytk (n=aaa@82.211.129.231)
03:28.48fugitivopostgresql is a real database
03:28.49*** join/#asterisk zip_ (n=zip@68-118-126-104.dhcp.sprn.tx.charter.com)
03:29.08Ariel_fugitivo, yes it is so is MYSQL
03:29.15fugitivono way
03:29.21fugitivomysql is not a real database
03:30.18Nuggethttp://www.livejournal.com/users/dirigibles/75781.html  <-- unrelated to databases.
03:30.29fugitivodatabases sux!
03:31.46Ariel_fugitivo, I am just messing with you. I think all databases have there use and need
03:31.55fugitivoi agree
03:32.27*** join/#asterisk cianhughes (n=cian@cian.ws)
03:33.02fugitivoms sql is a great database
03:33.16fugitivoin terms of functionality
03:34.33*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
03:34.50fugitivoNugget: burgertime is great
03:35.12*** join/#asterisk The_Ball (n=alex@static-227.35.240.220.dsl.comindico.com.au)
03:35.49*** join/#asterisk Gronker__ (n=Gronker2@adsl-217-229-114.ags.bellsouth.net)
03:36.00*** part/#asterisk Gronker__ (n=Gronker2@adsl-217-229-114.ags.bellsouth.net)
03:36.15*** join/#asterisk drbrown (n=chatzill@63.238.118.90)
03:37.56*** join/#asterisk spackle (n=spackle@209.234.83.19)
03:39.33Nuggetyeah, mssql is solid.
03:39.49*** join/#asterisk Grubs (n=Miranda@c220-239-96-230.eburwd5.vic.optusnet.com.au)
03:40.08spackledid I get here in the middle of a holy war?
03:40.36Nuggetnah
03:42.22GrubsQ.  For connecting 2 phones to asterisk - is it better to use a TDM400P with 2 x FXS ports and do it directly or use a Linksys PAP2 (half the price).  Any pros or cons on either approach?
03:42.29*** join/#asterisk cianhughes (n=cian@cian.ws)
03:43.37fugitivoGrubs: i think i'll cheaper 2 ip phones instead of a tdm400 with 2 fxs ports
03:43.47h3xjesus h christ superstar
03:43.49spackleGrubs - I'd take the sipura.
03:44.00h3xvoip-info could make a fortune selling clickthrus
03:44.15h3xim getting hammered with leads...
03:44.15fugitivoGrubs: i use the pap2-na, it works fine
03:44.22GrubsIP phones are expensive here.  more than $100 each.   PAP2 is $90.
03:44.25h3xscrew google
03:44.35fugitivoGrubs: how much is a tdm400 with 2fxs there?
03:44.38Nugget$100 is expensive?
03:44.46fugitivoGrubs: around 350?
03:44.50Grubsno
03:45.15fugitivoNugget: in some countries, it is :)
03:45.20Grubsany way you look at ...if you have the phone already a PAP2 is cheaper
03:45.34fugitivoGrubs: a pap2 is always cheaper
03:45.36NuggetI also suggest s sipura over a tdm400p.  Zaptel can be a pain in the ass
03:45.54NuggetIP phones are still the better bet, though, unless you need wireless.
03:46.04fugitivoGrubs: i was saying, that 2 ip phones, are going to be cheaper than a tdm400 with 2 fxs
03:46.04*** join/#asterisk cianhughes (n=cian@cian.ws)
03:46.33GrubsI need the tdm400p because X100p's are mostly crap in AU because of the different line impedance...  so I can get the TDM with 1FXO only and use a PAP2 - or I can get the TDM with 1FXO and 2 FXS.
03:46.35fugitivoGrubs: if you want, go with the pap2-na, it works ok
03:47.19fugitivoGrubs: if you don't want to spend too much money, go with the pap2
03:47.42fugitivoif money is not a concern, buy ip phones
03:48.13Grubssounds like there is no down side to the PAP2 then - thanks fugitivo.   IP phones are great - but I love being cordless so I'll re-use my DECT phones.
03:48.15spacklebut not grandstream ;-)
03:48.22fugitivoand not atcom
03:48.27spackleyeah
03:49.15*** join/#asterisk TheCops (n=mdb@206-248-136-187.dsl.teksavvy.com)
03:49.20fugitivoGrubs: i'm using pap2-na with a siemens gigaset, it works fine, all the functions you'll need
03:49.52fugitivoGrubs: remember to get the NA
03:49.53GrubsSipura SPA-841  is the cheapest IP phone here:  $160 ... but there are not a lot of cordless IP phones around so PAP2 it is then.
03:49.58GrubsNA?
03:50.05fugitivoGrubs: the pap2-na is not locked
03:50.17fugitivothe pap2 is locked for providers
03:50.24TheCopsI have a callwaiting option on my PSTN line, where can I start with asterisk for manage this feature ?
03:50.29Grubsoh - mine is not locked.. bought independed from a vendor
03:50.34fugitivook
03:51.27*** join/#asterisk bmg505 (n=leon@rndf-146-34-115.telkomadsl.co.za)
03:51.34spackleTheCops: what is your interface to the POTS?  Zap?
03:51.39Grubsthanks for your help.  Seems the PAP2 will work straight up....  my experience with X100P's here burned me a bit on the zaptel anyhow :)
03:51.46TheCopsspackle, yup, FXO interface
03:52.06fugitivoGrubs: but, x100p is fxo, not fxs
03:52.15spackleThen check your settings in zaptel.conf.  There are some available for call waiting.
03:52.38Grubsyes...    I was thinking of the TDM400P with FXS modules using zaptel.
03:53.26*** join/#asterisk onhold (n=maxgluck@200.109.166.83)
03:53.58*** join/#asterisk cianhughes (n=cian@cian.ws)
03:54.22*** join/#asterisk sycofly (n=syco@sycofly.com)
03:54.38GrubsSomeone needs to make some DECT IP phones.
03:54.53spacklewhat is DECT phone?
03:55.32TheCopsspackle, and what is doing the callwaiting options in the zaptel, how can I use it with a snom ?
03:55.34Grubsdigital cordless
03:55.57fugitivoGrubs: there are wireless phones
03:56.21GrubsWireless is great until the good wife turns on the microwave :)
03:56.45spackleTheCops: whether to pass the call waiting signal through to the phone while in use I believe.
03:56.51GrubsPAP2 + DECT is secure and perfect sound... so will do for now.
03:57.19fugitivosiemens have one that looks like a celphone
03:57.21GrubsDECT  IP PHone!:  http://www.gizmodo.com/gadgets/voip/index.php
03:57.26spackletheCops: is there a flash-hook button on your phone?
03:57.40TheCopsspackle, no, its a snom 320, a lot of button, but not flash
03:58.00Qwellip phones don't really have "call waiting"
03:58.05QwellThey have multiple line appearances
03:58.17TheCopsQwell, I can choose many line
03:58.32fugitivoGrubs: that's nice, it's a wireless phone and ata
03:58.38TheCopsbut when I have the special beep, I can't take the line
03:59.05QwellTheCops: there is no button to activate a line?
03:59.31*** join/#asterisk cianhughes (n=cian@cian.ws)
03:59.34TheCopsyeah but it dont take the other call on the call waiting line
04:00.17TheCopshrmm
04:00.26TheCopsI guess it have a way to make Flash with a cmd from asterisk
04:00.31Qwellyou should just have to press the button for the line
04:00.45Qwellit'll put the other line on hold, and switch the active line to that one
04:00.48fugitivoTheCops: what phone?
04:00.54TheCopssnom 320
04:01.00fugitivois that the atcom?
04:01.04fugitivoyea, atcom 320
04:01.08fugitivoi have the same issue
04:01.11fugitivoflash doesn't work
04:01.14TheCopsatcom ?
04:01.16TheCopssnom!
04:01.20fugitivoatcom
04:01.21fugitivo!
04:01.32h3xsnom rules!!!!
04:01.52fugitivohmm, no, it's not the atcom
04:02.08TheCopsTheres a way to make a extension command when youre in line on a zap channel
04:02.14*** join/#asterisk nick125 (n=nick125@unaffiliated/nick125)
04:03.04nick125hey, anyone know where i can get some basic configs (not the example ones, they are too large and hard to work with), with maybe one extension and etc?
04:03.13spackleapparently some phones have a link button too , http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Flash#comments
04:04.07spacklenick125, I think John Todd or someone like that has some good config file samples posted publicly.
04:05.02*** join/#asterisk cianhughes (n=cian@cian.ws)
04:05.09nick125ok, the hard part: some example configs like that, except for asterisk 1.2
04:06.49opus_Grubs how well does that work?
04:07.15spackleand do you have link to the specs on it?
04:09.01*** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se)
04:09.26opus_233mhz is way to slow for asterisk
04:09.33opus_it would be great for an ATA however
04:09.36opus_:)
04:10.06spackleopus: not much slower than soekris, is the same as my wrt54g with Asterisk running on it.
04:10.08*** join/#asterisk {tasker} (i=sdfsddad@modemcable064.169-203-24.mc.videotron.ca)
04:10.11Grubsopus_ - seems to work well enough for my single line.  Currently using iLBC which should max out the CPU but I am thinking that all the encoding is being done on my soft-phone client box and asterisk is just passing it through.  A bit slow to boot and reconfigure though.  Using Debian.
04:10.35opus_spackle how well does the wrt54g work
04:10.37*** join/#asterisk cianhughes (n=cian@cian.ws)
04:10.49Grubstoo slow?  Its been working great for a few weeks now *shrug*
04:10.54opus_Grubs hmmm.
04:11.04opus_The wrt54g with asterisk, I ment.
04:11.16spacklecan make two calls through it, no transcoding.  does a basic job.
04:11.25opus_why are you using iLBC? doesn't that codec suck?
04:12.23opus_spackle - when you say no transcoding, do you mean canreinvite=yes or something to that effect?
04:12.28GrubsI'll do a little test and connect to asterisk using aLaw and then see if asterisk can talk to my VSP using iLBC :)
04:12.34{tasker}asterisk HEAD, 1.2.x and > 1.0.5 still don't pass DTMF to the terminating side
04:12.37*** join/#asterisk Darwin35 (n=kvirc@ip70-179-214-245.dl.dl.cox.net)
04:12.51{tasker}RFC2833
04:12.53opus_tasker -- You have that problem to
04:12.58{tasker}everyone does
04:13.09{tasker}there is even a bug opened and no progress since Aug 24
04:13.15Qwelldoesn't pass dtmf under what circumstances?
04:13.29{tasker}1- dial into Asterisk
04:13.35*** join/#asterisk musictomyears (n=maxgluck@200.109.166.83)
04:13.36spackleopus: no, I pass the calls through the wrtg54g to nufone or another set.  not using canreinvite=yes
04:13.38{tasker}2- transfer call out
04:13.40opus_tasker - i've was told by bkw that its because the media gateway has fucked RFC2833 when I showed him the rtp dump
04:13.56{tasker}3- punch some tones
04:14.04{tasker}terminating side receives no digits
04:14.07{tasker}rfc2833
04:14.10{tasker}H.323 and SIP
04:14.24opus_i have the problem as well, with calling in. asterisk will sporatically not register the DTMF digits for one call
04:14.24spackleso you have to use a cantain crunch whistle with CVS head.
04:14.25{tasker}it works the other way
04:14.38{tasker}oh
04:14.38{tasker}no
04:14.42opus_or the next, then like 10 hours later will work. its the only thing stopping me from rolling asterisk out production wise
04:14.42{tasker}Asterisk receives the tones
04:14.58{tasker}if I dial into an Asterisk IVR app, it works fine
04:15.04opus_I have it so asterisk receives RTP packets, but doesn't register them as DTMF
04:15.07{tasker}if the app redials out, DTMF does not pass
04:15.31*** join/#asterisk stepcut (n=user@ip68-107-21-88.sd.sd.cox.net)
04:15.38{tasker}so, I dial into asterisk and the app transfers my call out to, say, my home voicemail or a PBX, I cannot dial any digits
04:15.51{tasker}however
04:16.00{tasker}i tested it by calling myself
04:16.03{tasker}on another phone
04:16.09opus_tasker howabout updating the bug with the rtp debug of the call in question
04:16.14*** join/#asterisk cianhughes (n=cian@cian.ws)
04:16.17Grubscan people vote on the open bugs to elevate their priority?
04:16.21{tasker}it's already there
04:16.27{tasker}here's the thing
04:16.28opus_url pls
04:16.32{tasker}it's 100% reproducible
04:16.34opus_i think i know which one but not sure
04:16.57{tasker}http://bugs.digium.com/view.php?id=4989
04:17.04{tasker}but check this
04:17.07{tasker}i call myself
04:17.10{tasker}through asterisk
04:17.17{tasker}phone 1 ----> asterisk -----> phone 2
04:17.23{tasker}phone 2 can dial DTMF back to phone 1
04:17.29{tasker}phone 1 cannot send DTMF to phone 2
04:17.33{tasker}you hear nothing on phone 2
04:17.35opus_tasker is this only for h323?
04:17.36{tasker}rfc2833
04:17.40{tasker}no
04:17.41{tasker}SIP as well
04:17.53opus_OK i skipped it because it didn't say SIP big enough
04:18.08{tasker}it's an RTP issue
04:18.28{tasker}RTP debug receives the rfc2833 tone but does not send it to the other leg
04:18.30{tasker}it eats it
04:18.44{tasker}this started happening some time after 1.0.1
04:19.03{tasker}if I compile an older version, it works without a problem
04:19.34*** join/#asterisk dr123 (n=asdf@pcp09065496pcs.northw01.in.comcast.net)
04:19.37{tasker}i've tested it with several different types of endpoints, including Cisco AS53xx gateways
04:19.40{tasker}no joy
04:20.01{tasker}if i switch to inband & G711, it's working fine
04:20.17{tasker}because it's plain sound :)
04:20.23{tasker}but rfc2833 will not pass through
04:20.28{tasker}one way
04:21.23{tasker}i've tried to track the problem down but I cannot find the logic that eats the tone
04:21.42dr123Is there a way to dial Via URL on Cisco 7960 via IAX2 url and have it pass it to asterisk for the call so it would not be a SIP to SIP call I still want it to go through the server
04:21.47*** join/#asterisk cianhughes (n=cian@cian.ws)
04:22.07opus_tasker i added a comment saying it effects sip as well
04:22.13{tasker}ok
04:22.22brc_~seen j4m3s
04:22.26jbotj4m3s <n=debbie@digium.com> was last seen on IRC in channel #asterisk, 1d 7h 31m 26s ago, saying: 'brc_ kevin told me to talk to you about jira'.
04:22.26{tasker}do you have asterisk setup?
04:22.29opus_i bookmarked it
04:22.35opus_tasker yeah
04:22.38nick125am i crazy for wanting to build a asterisk solution using mysql database (i want to link asterisk to the customer database, so, when a customer calls up, it can automatically lookup the info for the customer and such)?
04:22.48opus_i will bug everyone to get that one looked at I think I am having the same problem
04:22.59{tasker}it's most likely the same issue
04:23.02opus_nick125 yup
04:23.07{tasker}do you have an external IP endpoint you can dial out to?
04:23.14opus_tasker I also have an issue is RTP and NAT
04:23.16nick125i mean, doesnt asterisk support mysql or something like that?
04:23.26Qwellnick125: That actually won't be too difficult to do
04:23.32opus_tasker yeah, but i can't touch it
04:23.35Qwellnick125: just a simple AGI, or a C app
04:23.42{tasker}opus: but you can dial out to it, right?
04:23.47Qwellapp_customerlookup.c, or some such
04:23.49{tasker}is it connected to a PSTN circuit?
04:23.58nick125hrm...c...
04:24.00Qwell(mysql is very easy to query up from C)
04:24.03opus_tasker no I don't touch analog at all
04:24.09nick125Qwell: any docs on AGI?
04:24.17*** join/#asterisk X-Rob (n=rob@dsl-202-173-151-24.qld.westnet.com.au)
04:24.21{tasker}opus: what's the endpoint connected to?
04:24.22Qwellnick125: If you're going to use this with much volume, I wouldn't suggest agi
04:24.33Qwellagi scales poorly, afaik
04:24.47opus_Qwell well, the problem with Queue how can you run a Macro before or when an agent picks up. the only idea i had was to loop the SIP channel back through asterisk with another channel that loops back to the real agent and runs a macro in the dial plan that way..
04:25.02*** join/#asterisk pauldy (n=pauldy@c-67-162-191-108.hsd1.co.comcast.net)
04:25.04{tasker}opus: what I was going to suggest is to call yourself through asterisk and dial some numbers on both phones.
04:25.08opus_Macro(./send_user_desktop_popup.sh)
04:25.22opus_tasker - that almost never works for me.
04:25.26*** join/#asterisk cianhughes (n=cian@cian.ws)
04:25.26opus_tasker - or site-to-site
04:25.38{tasker}opus: dialing or DTMF?
04:25.50opus_DTMF
04:26.03opus_i can't even use my own company's voice tree 9 times out of 10 right now
04:26.14{tasker}might be the same issue
04:26.23{tasker}I have 100% failure
04:26.24nick125well, the idea here is that i would build a app that would access a database. then, when a customer calls in and an agent picks up, asterisk would drop a record in the db, and the app would look at entry up by agent number, and then the app can find the customer number and look it up in the customer db.
04:26.32opus_tasker I had problems dialing 1-800-CALL-ATT and entering digits
04:26.39opus_tasker - and fixed it with modifying rtp.c
04:26.44{tasker}how?
04:26.46{tasker}where?
04:26.51opus_but Sprint/Verizion/T-mobile voicemail box don't like it
04:27.22{tasker}where did you modify in rtp.c ?
04:27.31opus_search for the comment  '800ms' and change the value on the next line  from '100ms'(?) or whatever to above 2000 in rtp.c
04:28.02{tasker}ok
04:28.18opus_my co workers also go it working by hitting the dtmf pad extremely fast as well
04:28.18{tasker}checking now
04:28.50opus_tasker -- i think that area of the code is what is fucked. there was a patch against that area somewhere between 1.0.5-1.0.7
04:29.56stepcutis MoH working properly under FreeBSD these days (i am rather out of date...)
04:30.04{tasker}opus: that's when the problem started
04:30.07{tasker}> 1.0.5
04:30.09stepcut(I don't have any zaptel hardware...)
04:30.57opus_are you guys listening to the lousiana scanner shoutcast links, they're pretty hard to hear
04:31.03{tasker}this is what I found
04:31.04{tasker}/* Make duration 800 (100ms) */
04:31.04{tasker}rtpheader[3] |= htonl((800));
04:31.07opus_yeah
04:31.09PakiPenguinanyone here uses linksys pap?
04:31.10opus_change 800 to 5000
04:31.12{tasker}ok
04:31.20opus_which doesn't make sense, but worked for me.
04:31.34opus_"hold down the fuckign dtmf key for 5 seconds you buggy piece of shit"
04:31.52*** join/#asterisk cianhughes (n=cian@cian.ws)
04:33.51{tasker}didn't work
04:34.44{tasker}I called myself and still couldn't hear DTMF tones in one direction
04:34.49*** part/#asterisk spackle (n=spackle@209.234.83.19)
04:34.52{tasker}:(
04:35.30opus_hmmm
04:35.40opus_did you 'reload' or 'stop now'
04:35.48{tasker}I had to recompile, right? :)
04:35.57opus_yea
04:36.02{tasker}lol
04:36.14opus_i dunno, i tried to hot swap modules before
04:36.15{tasker}so reload or stop now are implied
04:36.34{tasker}ok
04:36.57opus_but that area code is where its fucked, in my fuzzy logic brain here
04:37.18opus_whoever patched it didn't even bother updating the comment
04:37.26opus_wait nevermind
04:37.29opus_i'm dum
04:37.43{tasker}lol
04:37.47{tasker}i'm dumber
04:38.01{tasker}because I didn't think of looking at that piece of code
04:38.03*** join/#asterisk Moc_ (n=mochouin@229-198-0-72-ppp.3menatwork.com)
04:38.44dr123has anyone tried 1.2 b1 is that out?
04:38.55opus_what day was 1.0.5 released, i'll look at cvs diff
04:39.12*** join/#asterisk cianhughes (n=cian@cian.ws)
04:39.14{tasker}i don't remember off hand
04:39.20{tasker}dr123: yes, I have it running now
04:39.27dr123anything new and interesting?
04:39.45{tasker}new, more reliable H.323 using up-to-date OH323 stack
04:39.58{tasker}some other new functionality
04:40.00opus_<                       rtpheader[3] |= htonl((240));
04:40.17dr123I see
04:40.19{tasker}that's what it used to be?
04:40.42dr123Is it possible w/ asterisk to use an IVR that is voice activated
04:41.04*** join/#asterisk DrRighteous (n=DrRighte@ool-457843d1.dyn.optonline.net)
04:41.25{tasker}dr123: you mean have VAD in asterisk?
04:41.35{tasker}dr123: no, it's not implemented
04:41.51Qwellno, he means speach recognition
04:41.55Qwell(I think?)
04:41.58Qwellspeech
04:42.10PakiPenguinanyone got this?
04:42.12{tasker}ah, ok
04:42.13PakiPenguin<PROTECTED>
04:42.24Qwelldr123: You can use sphinx, but it isn't great
04:42.30dr123oh so that is called VAD i didnt realize it had its own acronymn
04:42.30opus_<t3t:#interdictor-scanner2> ...sitting in front of the brigade. eta 3 sep. original (unt) capt. (unt) are in iraq? over.
04:42.38dr123is that a feature that will be worked on
04:42.38{tasker}Palk: yes, you need to kill that
04:42.43*** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com)
04:42.52*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
04:43.17{tasker}dr123: at the rate they're introducing bugs into asterisk, you'd better hope they don't
04:43.23{tasker}DTMF is broken
04:43.24dr123hahahahaha
04:43.27{tasker}the most basic feature
04:43.31dr123wow
04:43.50dr123I am running the current stable on debian 3.0r1 and it is working great!
04:43.53dr123no problems at all
04:44.00{tasker}Palk: in frame.c, that frame-dropping business on G729 needs to be disabled
04:44.03dr123but I am not really using a lot of the fancy features
04:44.25*** join/#asterisk shido6 (n=shido6@d57-87-253.home.cgocable.net)
04:44.30dr123Ok how about this: Has anyone heard a tentitive release date for the Cisco 7970 SIP image firmware?
04:44.50{tasker}dr123: try dialing into asterisk and transfer your call out to an external voicemail or dialing your cell-phone, then hit DTMF tones
04:45.13JDLSpeedyhow do you setup a private caller? like someone calls me, if its a unknown under or what ever, it ask for its name, it would then contact me saying hey <person name> on the line, press 1 to accept the call, press 2 or hang up to deny the call
04:45.40JDLSpeedyunknown number*
04:46.02{tasker}JDL: check caller ID. if blank, jump to a voice prompt to record the user's name, dial out to you and play a prompt with an option
04:46.22*** join/#asterisk drbrown (n=chatzill@63.238.118.90)
04:46.43JDLSpeedyok {tasker}, I'll try that
04:46.49JDLSpeedythx
04:46.53{tasker}np
04:47.01{tasker}it's a very simple script
04:47.03*** join/#asterisk cianhughes (n=cian@cian.ws)
04:47.12*** join/#asterisk header (n=maxgluck@200.109.166.83)
04:47.13opus_tasker. I don't think this will work, but I just check out asterisk 1.0.5's rtp.c into CVS HEAD and modified it to compile.
04:47.16opus_want to try it?
04:47.25{tasker}sure
04:47.39JDLSpeedyyou think its simple, LOL, im setting it up for the first time today
04:47.51opus_or wait i'll just use pastebin
04:48.20{tasker}ok
04:48.34{tasker}JDL: you'll get the hang of it easily enough :)
04:49.04headerwhat could be wrong if I receive only the header when calling from a 7940? If I call TO the 7940, the call gets through, but not the way around, I only see a header with 0 lines in asterisk...?
04:49.10*** join/#asterisk santiago (n=santiago@63.245.86.163)
04:49.16JDLSpeedyI think asterisk is soooooooo cool
04:49.41{tasker}JDL: yeah, but it's riddled with bugs that don't get fixed because the developers don't use the bugged features
04:49.44{tasker}for example
04:49.56{tasker}dialing through asterisk to an external number that requires DTMF!!
04:50.10{tasker}caller -----> asterisk -----> company PBX
04:50.15Qwell{tasker}: May I rephrase your statement?
04:50.18{tasker}no joy sending DTMF tones
04:50.23{tasker}Qwell: please do
04:50.26{tasker}lol
04:50.32QwellIt's riddled with bugs that don't get fixed because the people who use the bugged features don't develop.
04:50.40{tasker}not true
04:50.51opus_http://pastebin.ca/22025
04:51.02{tasker}i've been busting my head on trying to find who fucked up the DTMF passthrough for a month now
04:51.03opus_also run
04:51.07opus_cvs update -D  "01/23/2005" include/asterisk/channel_pvt.h
04:51.09GrubsGiven there is a bugs database it does seem odd that some bugs dont get fixed
04:51.22{tasker}Grubs: agreed
04:51.30{tasker}opus: ok
04:51.43opus_i don't think it will work, but it compiled
04:51.43opus_hehe
04:51.46*** join/#asterisk Jabroni (n=Hercules@red-corp-200.76.249.142.telnor.net)
04:52.18opus_i just commented out the non-compiling parts. something i learned when porting ircII to dynix back in 1993
04:52.33{tasker}Qwell: nobody knows asterisk better than the original developer and nobody has a better idea of where DTMF broke than the one who actually broke it
04:52.34*** join/#asterisk cianhughes (n=cian@cian.ws)
04:52.47{tasker}opus: hehe
04:52.51{tasker}opus: i'll give it a try
04:52.54{tasker}opus: hang on
04:53.06JDLSpeedy{tasker}: so your trying to have it set up so if you press a option, it would forward to a outside phone?
04:53.21{tasker}JDL: no
04:53.30*** join/#asterisk shido6 (n=curtis@d57-87-253.home.cgocable.net)
04:53.49{tasker}JDL: if you use asterisk as a passthrough switch, where you dial in and it transfers your call to an outside line, you cannot send DTMF to that outside line
04:54.01Jabroniguys i got a lil' problem.. maybe someone can iluminate a bit my head... im using a A@H setup, created a IVR, and one of the options is to try to dial a number within the pbx, and link it up with another Zap Channel, but the problem is that channels never hang up :s
04:54.26JDLSpeedyDTMF?
04:54.26Jabroniexten => _.,1,Goto(from-internal,${EXTEN},1)
04:54.29{tasker}caller ----> asterisk -----> some PBX somewhere
04:54.31Qwell~dtmf
04:54.32jbotDTMF: Dual Tone Multi-Frequency. The technical term describing Touch Tone dialing. Basically the combining of two tones, one low frequency and one high frequency.
04:54.47opus_jabroni don't use h,hangup , make sure you actually have a hangup after you dial. don't use Dial(asdfas||r)
04:54.49{tasker}opus: no joy
04:55.02{tasker}opus: same problem
04:55.16opus_well fuck
04:55.22{tasker}lol
04:55.36{tasker}it may have been broken somewhere related to catching tones
04:55.42{tasker}such as the '*' to hangup a call
04:56.03{tasker}you can disconnect an active call by hitting * during a call transfer
04:56.29{tasker}there were issues and they patched something at some point to be able to distinguish who pressed *
04:56.42opus_haha
04:56.51opus_yeah, i could just dial up a customer and hit *
04:57.18{tasker}shortly after that, DTMF stopped passing through from originating to terminating end
04:57.18opus_tasker fuck
04:57.21{tasker}:(
04:58.10*** join/#asterisk cianhughes (n=cian@cian.ws)
04:58.24opus_asterisk needs a unit tester
04:58.35opus_that also does RTP
04:58.36{tasker}i have a whole carrier network at my disposal
04:59.30opus_asterisk isn't disaster proof yet
04:59.47{tasker}it's fairly solid
04:59.54{tasker}we use it for prepaid card platforms
05:00.02{tasker}but that DTMF issue is a real ass-biter
05:00.07opus_then why do i have so many problems. i guess its just my fucked up config
05:00.13{tasker}no, no
05:00.18{tasker}depends on the application
05:00.23QwellYou know, you could also pay to have a bug fixed.  That jumps the priority pretty high
05:00.31{tasker}yeah
05:00.33{tasker}sure
05:00.40{tasker}people are buying asterisk licenses
05:00.47{tasker}i can't imagine they haven't the bug
05:01.01{tasker}because a fix would make its way into the tree
05:01.04opus_if i had the money i would
05:01.12{tasker}most people are using asterisk as a PBX
05:01.16{tasker}not a passthrough switch
05:01.20Qwellasterisk is a pbx
05:01.30{tasker}asterisk is an IVR
05:01.32{tasker}not a PBX
05:01.40{tasker}it can be made to behave as a PBX
05:01.52{tasker}but it's nothing more than an engine that interprets instructions
05:01.56QwellAsterisk? - the Open Source PBX!
05:01.58Juggieasterisk is a pbx
05:02.05{tasker}yes, yes, that's the name they gave it
05:02.09{tasker}but technically it is not
05:02.10Qwellbecause thats what it is
05:02.21QwellI'm not even going to argue that point.
05:02.21{tasker}without extensions.conf instructing a PBX app, it does nothing
05:02.26opus_what would be a better name description for it
05:02.35{tasker}Asterisk IVR
05:02.39Qwell{tasker}: it wouldn't be an IVR without extensions.conf
05:02.41Juggiei dont know of any ivr's that register peers, convert between sip/h323/iax/zaptel
05:02.46{tasker}not true
05:02.48Juggieyes
05:02.54Juggieno extensions.conf and its nothing
05:02.56{tasker}an IVR implies it can do whatever
05:03.03{tasker}a PBX is specific
05:03.12{tasker}an IVR can be made to do anything interactive
05:03.14QwellThis is a stupid argument
05:03.15Juggieits a pbx on drugs
05:03.19{tasker}lol
05:03.24Juggieits a pbx + ivr
05:03.25{tasker}im not arguing
05:03.26Juggienot a ivr
05:03.33{tasker}do you know what pbx means?
05:03.35Qwella pbx is an ivr
05:03.38Qwellan ivr is not a pbx
05:03.43{tasker}public branch eXchange
05:03.45*** join/#asterisk cianhughes (n=cian@cian.ws)
05:03.55Juggietasker, that term is null these days
05:04.02{tasker}an IVR is a blank system that accepts programmed instructions ;)
05:04.03Juggiesince you can also have private branch eXchanges
05:04.04Qwellso, me picking up a phone and talking to you is "Interactive Voice Recording"?
05:04.09Netgeeks_Interactive Voice Response doesn't  describe asterisk any more than Private Branch Exchange does
05:04.22{tasker}i've been developing interactive voice response systems since Dialogic first started in the early 90's
05:04.23{tasker}lol
05:04.23Qwellis it recording, or response?
05:04.26Qwell~ivr
05:04.32Juggie{tasker}, me too...
05:04.35{tasker}response
05:04.48Juggiei started writing dialogic in like 99
05:04.50{tasker}an IVR is a very generic term
05:04.56{tasker}I started Dialogic in 93
05:05.08Juggielike Qwell said, a pbx is an ivr but an ivr isnt a pbx
05:05.08{tasker}writing Dialogic apps, i mean
05:05.13{tasker}nope
05:05.16{tasker}the other way around
05:05.24Juggieno thats right
05:05.24{tasker}an IVR can be made to behave as a PBX
05:05.26opus_er}> writing Dialogic apps, i mean
05:05.32Juggieno it cannot
05:05.37Juggiean ivr is not a switch
05:05.46{tasker}a PBX will not become a prepaid app, or an informational retrieval service, etc
05:05.51{tasker}no, but it can switch calls
05:05.53opus_[H.323 0004989]: was just updated tasker, the guy saids he patched h323 and fixed it and will soon past a patch.
05:06.03{tasker}opus: KEWL!
05:06.07Netgeeks_asterisk is currently more a softswitch than anything else
05:06.07{tasker}opus: thanks for the update
05:06.08Juggie{tasker}, have you looked at pbx's in the last 5 years?
05:06.17Juggieno one makes hardware based switches anymore
05:06.20{tasker}Netgeeks: that's a better description
05:06.29{tasker}Netgeeks: but a softswitch is not necessarily interactive :)
05:06.47Juggietasker, when did you last put your hands on a mitel, cisco, nortel pbx
05:06.51Juggiethey are all softswitches
05:06.54{tasker}every day
05:06.56{tasker}unfortunately
05:06.57Juggiewell me too
05:07.10Juggieeverything is soft these days... asterisk is a pbx like any other
05:07.12{tasker}we have Cisco AS5xxx series all over our network around the world
05:07.13Netgeeks_but softswitch is a much wider description, much so that I think it applies more so than the other two
05:07.18Juggiehell, the original cisco crap ran windows
05:07.23Juggieuntil they ported it to linux
05:07.25{tasker}just the callmanager
05:07.34Juggieyah, the original call manager ran windows
05:07.36Juggienow its linux
05:07.43Juggiemitel released a softpbx running windows
05:07.50{tasker}blech
05:07.50opus_do you mean they don't use any DSP's
05:07.51opus_?
05:08.08Juggiethey do, i know mitel has some dsp's on their cards
05:08.12Juggienot sure about the cisco call manager gear
05:08.13opus_i recently installed a shoretel switch and it was all FPGAs/DSPs
05:08.30Juggieopus, was it an ip switch?
05:08.36opus_yeah
05:08.38Juggieor digital phones/analog
05:08.51Juggiewhat i see around here is all in like 2U racks
05:08.56opus_of course. it uses the polycom IP-500 rebadged as the "shorepoint 100"
05:09.10Juggieit has dedicated hardware, but alot of it is software as well
05:09.20*** join/#asterisk cianhughes (n=cian@cian.ws)
05:09.22opus_windows 2000 box to do its voicemail
05:09.34opus_however you can reboot that box and no lines will go down
05:09.48{tasker}kind of like Excel
05:09.55{tasker}the call manager runs off a separate box
05:09.59{tasker}be it under windows or whatever
05:10.14{tasker}*sigh*
05:10.15opus_when you say 'call manager' what do you mean
05:10.26{tasker}call setup / teardown
05:10.29{tasker}switching
05:10.29opus_shoretel world = call manager is desktop CAD software
05:10.32{tasker}prompts
05:10.32{tasker}etc
05:10.50Grubsdudes - you could have had that bug fixed by now if your were not bickering over semantics ;)
05:10.51opus_cisco world = call manager is IP-PBX?
05:11.02Juggieyes callmanager = ip pbx
05:11.09{tasker}opus: nooooooo
05:11.10{tasker}lol
05:11.21{tasker}ok, ok, have it your way
05:11.47opus_shoretel is nice, has computer desktop helpers etc
05:11.53{tasker}in Excel, you can create custom apps and send instructions to the switch via TCP/IP
05:11.59opus_.asp admin interface
05:12.04{tasker}they're hex code commands
05:12.18opus_haha.
05:12.26{tasker}Cisco uses TCL that it directly interprets for custom apps
05:12.34{tasker}frankly, that's a butt ugly language
05:12.41{tasker}but, hey, their choice
05:12.41Juggietcl is gross
05:12.42{tasker}lol
05:12.49Juggie* can use any scripting or programming language
05:12.52Juggieanything you choose
05:13.04Juggiei personally use php :P
05:13.06{tasker}yes, but the mechanics is ugly
05:13.08{tasker}example
05:13.15opus_when I look at the local yellow pages under 'phone equipment' everyone is advertising IP-based phones, what the heck are the adverting
05:13.19{tasker}Dial() locks your app until it's complete
05:13.25{tasker}but
05:13.35{tasker}I tried throwing Dial() into a thread so my app can continue
05:13.45opus_Like 3com VOIP
05:13.48{tasker}and I would check DIALSTATUS to get the line status
05:13.54shido6thats what the manager is for
05:13.55{tasker}and it blocks there :)
05:13.55{tasker}lol
05:14.01{tasker}until Dial is done
05:14.07opus_its not really voice over an IP stack. I call it VoCAT5
05:14.09Juggiethen feel free to fix it :P
05:14.16{tasker}I tried
05:14.28{tasker}unfortunately, the whole pbx_xxxx interface is sequential
05:14.32Juggiewhy would you want to move before a dial was done
05:14.36{tasker}it needs to much of a rewrite
05:14.43shido6use the manager.
05:14.44Juggiewhat would be the scenario where that was required
05:14.51opus_what technology does Nortel use for VOIP?
05:14.56{tasker}if you needed your app to perform other tasks during an active call
05:15.05{tasker}i.e. account updates, etc
05:15.08shido6then use the manager, taskbar
05:15.16shido6tasker
05:15.28{tasker}there's something new for me
05:15.32{tasker}how?
05:15.53*** join/#asterisk twisted[home] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted)
05:15.53*** mode/#asterisk [+o twisted[home]] by ChanServ
05:15.54{tasker}wouldn't I have to retrieve the channel name and open a new STDIN?
05:16.01twisted[home]wheee
05:16.03twisted[home]there are 3 of me
05:16.05{tasker}doesn't that become resource intensive?
05:16.10Juggietasker, there are ways you could do that, however i'm not sure what you would want to do that you coudnt do before or after a call
05:16.23{tasker}furthermore, there's an open bug about having too many manager sessions open
05:16.32twisted[home]hmm
05:16.35twisted[home]bkw_, you around?
05:16.38opus_tasker use the manager proxy
05:16.58{tasker}sequentially process requests?
05:17.02{tasker}that would work
05:17.04opus_how many manager api connections until it crashes?
05:17.15{tasker}don't remember
05:17.21{tasker}but it wasn't many
05:17.25Juggietasker, one solution that comes to mind, use php, do spawn your thread to do what you want, then when your dial completes, signal the thread that its time to move on.
05:17.26opus_hmm. that could be aproblem for me shit thanks for letting me know
05:17.33shido6using Asterisk::Manager
05:17.38shido6perl
05:17.44{tasker}kewl
05:18.00{tasker}so I can throw Dial() into a thread and use Manager to retrieve channel status?
05:18.04shido6use strict;
05:18.08shido6use CGI'
05:18.15shido6use Asterisk::Manager;
05:18.35shido6sendcommand
05:18.49{tasker}ok
05:19.01{tasker}i'll try it
05:19.14Juggietasker, do the dial in the main thread, and use a worker thread to do what you want
05:19.20Juggiedo it in the reverse of how your thinking
05:19.27*** join/#asterisk cianhughes (n=cian@cian.ws)
05:19.32{tasker}Juggie: that's also a good idea
05:19.42{tasker}Juggie: then have a flag to indicate call completion
05:19.47{tasker}awesome
05:19.49{tasker}thanks
05:19.52Juggieyes, signal your thread
05:20.01Juggieto let it know if the call is setup or not, etc.
05:20.03{tasker}my brain is too fried from all the projects and the lack of vacation
05:20.12{tasker}I haven't gone on vacation since 2002
05:20.15{tasker}I need a break
05:20.16Juggiehah :P
05:20.16{tasker}*sigh*
05:20.20Juggiewelcome to my world
05:20.23{tasker}lol
05:20.24Juggiemy only vacation is visiting family
05:20.29Juggienot really a vacation
05:20.33{tasker}no kidding
05:20.39{tasker}visiting family != vacation
05:20.40Juggiewhat language are you going to attempt this in?
05:20.44{tasker}Perl
05:20.48Juggiek
05:20.59Juggiei was gonna say with php you'll need a custom compile most likely to enable thread control
05:20.59{tasker}it's a whole lot faster to prototype
05:21.14{tasker}i thought of PHP but the threading is the issue
05:21.21Juggiehere is also a res_php
05:21.23Juggiefyi
05:21.32Juggiewhich means no using stdin/out io etc
05:21.33{tasker}kewl
05:21.41Juggieno using a socket for chatter
05:21.49Juggieand a res_perl
05:21.51{tasker}i looked at res_perl but it looks unstable
05:21.51Juggiefor that matter
05:21.56Juggieit should be ok
05:22.00Juggiecheck www.pbxfreeware.org
05:22.12Juggiebug bkw_ if there are problems with res_perl
05:22.24{tasker}k
05:22.31{tasker}bk = burger king
05:22.34{tasker}i'm hungry
05:22.44*** join/#asterisk knoppix (n=knoppix@cm52.theta17.maxonline.com.sg)
05:22.47{tasker}i think i'm gonna bug my wife and take her out for a midnight snack
05:22.54Juggiehah
05:22.57Juggieburger king
05:22.58Juggieeugh
05:22.58{tasker}lol
05:23.00Juggiegross
05:23.01{tasker}no, no
05:23.03*** join/#asterisk cianhughes (n=cian@cian.ws)
05:23.04{tasker}the burger idea
05:23.07Juggiemcdonalds is better then buger king
05:23.09{tasker}Carl's Junior is my fav place
05:23.17{tasker}but there aren't any here
05:23.26Juggierotten rons
05:23.34{tasker}BK used to be good when they flame broiled their patties
05:23.37{tasker}now they boil them
05:23.43{tasker}in that stupid drawer
05:23.53{tasker}who the hell boils their burgers?
05:23.59{tasker}and still claims flame broiled
05:24.08{tasker}BK does
05:24.26{tasker}I haven't seen a grill mark on their burgers in nearly half a decade
05:24.28Juggiehttp://www.pbxfreeware.org/res_perl.tgz
05:24.34Juggiethats the latest res perl
05:24.34{tasker}thanks, J
05:24.35Juggiecheck it out
05:24.53{tasker}so am I
05:24.56Juggiei think i'll get a ham samwhich
05:25.00Juggieblackforest ham
05:25.00Juggieumm
05:25.08{tasker}yum
05:25.11{tasker}that sounds good, too
05:25.14Juggieyah
05:25.21Juggiei have this spicey mustard
05:25.23Juggiethats really good
05:25.36{tasker}man, i'm coming to your place
05:25.46{tasker}i have nothing in the fridge
05:25.53{tasker}we eat out too much
05:26.10{tasker}and never fill that fridge
05:26.29Juggiehah
05:26.36Juggiei filled it because i am trying to save money
05:26.45{tasker}that's the right idea
05:26.48{tasker}and save your health
05:26.57{tasker}have you seen Supersize Me?
05:26.59{tasker}lol
05:27.00Juggieno
05:27.00GrubsCan asterisk make any use of dual CPU's?
05:27.06Juggiebut i did have rotten rons today
05:27.13{tasker}how is that?
05:27.13Juggiegrubs, yes
05:27.20{tasker}good burgers?
05:27.21Juggierotten rons = mcdonalds
05:27.24{tasker}lol
05:27.26JuggieRotten Ronnies
05:27.28Juggieget it :P
05:27.36{tasker}LOL
05:27.41Juggiejust like
05:27.42JuggieKFC
05:27.44Juggie=
05:27.47Grubsthinking of a nice slow dual rackmount server for *.  Dual 500 MHz ample?
05:27.48JuggieThe dirty bird
05:27.52{tasker}lol
05:28.06shido6grubs
05:28.11{tasker}Grubs: as long as you don't do a lot of transcoding
05:28.19Juggiegrubs, depends on call volume and codecs
05:28.25Juggieif you are using like 1 t1 card or less
05:28.29Juggieyou should be fine
05:28.29{tasker}Grubs: on our all software IVR setups, we even prerecord prompts in G729
05:28.40Grubsnice idea
05:28.51GrubsI prefer iLBC :)
05:28.55{tasker}im able to get 1800 simultaneous passthrough calls on a dual Xeon
05:29.01Juggienot many hard phones support ilbc
05:29.10Juggieif any
05:29.13Juggiei've never seen one
05:29.17{tasker}speex is a nice codec. I wish it would gain mainstream support
05:29.25Grubsonly 1-3 simultaneous calls here though.
05:29.32Juggiethen you are best kind
05:29.46{tasker}ITU unfortunately has too much invested in making royalties
05:29.50opus_i perfer ultrawideband :)
05:30.20{tasker}sounds like a heavy man's underwear
05:30.21GrubsI am sure I saw an ATA or two with iLBC support.
05:30.27Juggiehas anyone been paying attention to the blog from neworleans
05:30.38Juggiethe NOC thats running in new orleans and has been running the entire time
05:30.39shido6which blog ?
05:30.43*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
05:30.45{tasker}wow
05:30.45Qwelldirectnic
05:30.55shido6url?
05:31.00*** join/#asterisk cianhughes (n=cian@cian.ws)
05:31.08opus_i'm cohosting there
05:31.16Qwellhttp://www.livejournal.com/users/interdictor/
05:31.18Juggiei bet your servers up ;P
05:31.18opus_i wonder how much :)_
05:31.31Juggiei must say i am truly impressed
05:31.34Juggiethe city is underwater
05:31.39opus_are you listening to the scanner?
05:31.39Juggieand these guys are on irc chatting
05:31.41Juggieyes
05:31.52opus_theres live transcripts on #interdictor-scanner1 and scanner2
05:32.00Qwellopus_: freenode?
05:32.03opus_yeah
05:32.08*** join/#asterisk shidan (n=shidan@CPE0013107d30c4-CM001371871af0.cpe.net.cable.rogers.com)
05:32.13opus_<nop_:#interdictor-scanner2> [better than "only fix wing aircraft allowed inbound" if you ask me]
05:32.22*** join/#asterisk tabo23 (n=acura0@S010600119503044b.vc.shawcable.net)
05:33.02Juggiethey are on a quest to get deisel now
05:33.05Juggieand get another oc3 up
05:33.09Juggiethey are down to one peer
05:36.30*** join/#asterisk cianhughes (n=cian@cian.ws)
05:43.35{tasker}OMG!
05:43.41{tasker}I fixed the DTMF problem
05:43.47{tasker}lol
05:43.51{tasker}how stupidly trivial
05:44.23{tasker}opus: you there?
05:46.07hugo-v6ok asterisk-phones rock the hood.
05:46.53{tasker}if anyone is interested in the fix to the DTMF passthrough problem, go here:
05:46.54{tasker}http://pastebin.com/353386
05:49.37*** join/#asterisk Rowters (n=SilverDr@dsl-201-129-89-96.prod-infinitum.com.mx)
05:51.08{tasker}'nite all
05:51.14{tasker}c y'all next time
05:51.16*** part/#asterisk {tasker} (i=sdfsddad@modemcable064.169-203-24.mc.videotron.ca)
05:52.05*** part/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
05:55.19*** join/#asterisk azrishahril (n=azrishah@60.50.193.179)
05:59.08hugo-v6did isay asterisk-phones? well im still sleepy, meant elmeg-phones rock da hood.
05:59.36hugo-v6and btw gd morning *yawn*
06:00.03opus_hmmm
06:00.13opus_instead of rebuilding all analog lines in NO, maybe they could use VOIP
06:00.15opus_:)
06:00.47hugo-v6opus_: i wont use analog in the house anymore. i deplay voip-phones :>
06:00.55hugo-v6roxx.
06:01.21hugo-v6s/. i/. i will/
06:02.00*** join/#asterisk coppice (n=chatzill@167.203.17.210.dyn.pacific.net.hk)
06:02.33hugo-v6is there a way to save the "regexten" entry in a phone-section in sip.conf?
06:04.17*** join/#asterisk [hC] (n=hardcore@c-24-127-192-210.hsd1.fl.comcast.net)
06:04.56hugo-v6well found something, but i cant call i.e. "myuser" as in example on voip-info *think* at at least not from an isdn-phone.
06:06.18*** join/#asterisk Dybdahl (n=Lars@oelsted.dybdahl.dk)
06:08.34*** join/#asterisk Rowters (n=SilverDr@dsl-201-129-89-96.prod-infinitum.com.mx)
06:10.29*** join/#asterisk [hC] (n=hardcore@c-24-127-192-210.hsd1.fl.comcast.net)
06:14.08konradsHello
06:14.13gordonjcpmorning
06:14.18konradsI am having problems with SIP->H323 cals
06:14.25konradsNOTICE[9301] src/chan_h323.c: Don't know how to deal with mode 0
06:14.26konradsx40 (slin)
06:14.39konradsit attempts some sort of native bridge between stip and h323
06:14.48konradsi guess it tries to send same rtp payload
06:14.56konradshow do I force re-encoding?
06:20.35*** join/#asterisk gdsm (n=gdsm@e1-1.ns500-1.ts.milt.as9105.net)
06:23.04*** join/#asterisk r0d3nt (i=anonymou@tinfoilhat.net)
06:29.47*** join/#asterisk used (n=used@c-24-22-125-179.hsd1.or.comcast.net)
06:34.11azrishahrilkonrads:
06:35.15*** join/#asterisk drbrown (n=chatzill@63.238.118.90)
06:36.54hugo-v6puh. now going to get a box of tobacco. l8r
06:37.53X-Rob'box'?
06:37.57X-Robyou 'mericans are wierd.
06:38.39glm2kX-Rob: tobacco does come in boxes
06:38.51glm2ker, do
06:43.04*** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net)
06:45.58hugo-v6X-Rob: .de ;) not .us. well i roll from hand. so u can buy a bag or a box ;)
06:51.29X-Robwhat size is a 'box' of tobacco?
06:51.38X-Robwhen you say 'box' I would assume 'carton'
06:51.44X-Robeg, 10kg or so
06:53.57*** join/#asterisk santiago (n=santiago@63.245.86.163)
06:53.57hugo-v6x-rob: look after the ranslation ;)
06:54.02hugo-v6in my case it means 120g
06:54.05hugo-v6grains
06:54.48hugo-v6grains? am i stupid?
06:55.08hugo-v6120 gram which are about 4 ounces
06:55.35hugo-v6so no on a walk with my dog. l8r.
06:58.35*** join/#asterisk Romik (n=romik_@1.fix.netvision.net.il)
06:59.27*** join/#asterisk Romik (n=romik_@1.fix.netvision.net.il)
07:01.07X-RobI must say that this song I've been listening to (on repeat) all day is damn good.
07:01.24PakiPenguinX-Rob, which one?
07:01.29X-Rob'Cobrastyle' by Teddybears Sthlm
07:01.35X-RobWas just typing the same
07:01.36X-Robname
07:01.49X-RobDefinately 'Sthlm'
07:02.09X-RobIf you've seen the heineken advertisment where they're all breakdancing, it's the background music to that
07:02.20X-Rob'pop and lock' I've seen it described as
07:02.48X-Roband if you haven't seen that
07:02.51X-RobI'll upload it somewhere 8)
07:03.34X-RobIf you have limewire running, it's worth getting 8)
07:17.08*** join/#asterisk coppice (n=chatzill@167.203.17.210.dyn.pacific.net.hk)
07:23.14*** join/#asterisk Romik (n=romik_@1.fix.netvision.net.il)
07:24.13*** join/#asterisk Romik (n=romik_@1.fix.netvision.net.il)
07:27.36*** join/#asterisk audela (n=audela@82-33-115-145.cable.ubr08.azte.blueyonder.co.uk)
07:34.18*** join/#asterisk criptos (n=criptos@201.135.121.4)
07:34.53criptoshello everyone! :)
07:34.55*** join/#asterisk file[lap1op] (n=jcolp@mctnnbsa30w-156034035106.nb.aliant.net)
07:38.35*** join/#asterisk pbxbart__ (n=pbxbart@p54B00652.dip0.t-ipconnect.de)
07:38.40*** part/#asterisk pbxbart__ (n=pbxbart@p54B00652.dip0.t-ipconnect.de)
07:39.09*** join/#asterisk Pkunk (n=Pkunkage@mbbs.munnabhai.info)
07:39.48PakiPenguinPkunk: awesome domain
07:41.03X-RobWhat's the relevance of munnabhai.info ?
07:41.37PakiPenguinmunnabhai mbbs is a movie :)
07:41.39PakiPenguinhaha
07:41.50*** join/#asterisk samborambo (n=samboram@219-89-7-115.dialup.xtra.co.nz)
07:41.51glm2kanother bollywood production?
07:42.00PakiPenguinyeah
07:42.04X-Robaaah
07:42.16PakiPenguinanyone here has seen proof the movie?
07:42.18samborambohello all :)
07:42.20glm2khmm, i should try to find it then, might be fun
07:43.35samborambocould someone please help me with a problem I'm having with Amp?
07:43.48X-RobI was just going through all the old PC's here, and I found a P4 2.4ghz with a 40g HDD and 256M
07:43.53X-RobIT seems to work too
07:44.15X-Rob*woo*
07:44.19glm2knice
07:45.00PakiPenguinwhat
07:45.03PakiPenguin:o
07:45.06PakiPenguinX-Rob, sendo it to me :p
07:45.08PakiPenguinhaha
07:45.12PakiPenguinyou dont need it anyways
07:45.16*** part/#asterisk criptos (n=criptos@201.135.121.4)
07:45.43samboramboFrom what I understand, Apache2 needs to run as user "asterisk", however, asterisk is not the only thing running on that server, and so those other services need to be www-data
07:46.29*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
07:46.37samboramboanyone got any ideas?
07:47.26ByteX-Rob: that's an "old" pc?
07:50.32X-RobByte - it was sitting in the 'old' pile, and it had a rusted back panel thingo (around the parrallel port, keyboard, etc, the little metal clip in)
07:50.49X-RobI'm going through 'em all, stripping good bits out of 'em, and binning the rest.
07:51.08X-Robsamborambo - Are you talking about AMP?
07:53.46gordonjcpX-Rob: cool
07:53.54X-RobIt's got two passes of memtest. Woo. I think it's time for the old Piii/500 gateway to be upgraded!
07:54.05zedkatufsamborambo: afaik, AMP uses apache1.3 rather than apache2....is there any way you could get the other services to run under apache2 at the same time?
07:54.08gordonjcpsamborambo: can't it be in the www-data group?
07:54.53X-RobAh, he _is_ talking about AMP (reads the top of the screen)
07:55.39X-Robzedkatuf - there's no _real_ requirement for 1.3
07:55.53*** join/#asterisk GaryH (n=ghawkins@gromit.garysoft.co.uk)
07:56.02X-RobWe've just committed some patches to make it work with php5 on FC4, and I think FC4 uses apache 2.0, doesn't it?
07:58.29zedkatufX-Rob: I agree..my suggestion was just a workaround...not convinced it's the best way to do it though!
07:58.44X-RobIt can be worked around.
07:58.56X-RobWhen we release AMP2 I'll have to figure out a way to make the installation a lot easier
07:59.04X-RobI really don't like having to run apache as asterisk
08:03.00*** join/#asterisk MikeJ__ (n=ircatjer@adsl-68-74-8-29.dsl.sfldmi.ameritech.net)
08:04.06*** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se)
08:05.06samborambosorry X-Rob.....was away looking for more answers on the web. Yeah, I'm talking about AMP
08:05.24*** join/#asterisk GaryH (n=ghawkins@gromit.garysoft.co.uk)
08:07.03samborambogordonjcp: you think run apache as www-data group and asterisk user?
08:07.16*** join/#asterisk wasim_ (n=wasim@pdpc/supporter/active/wasim)
08:08.20samboramboI thought there might be a workaround by running a virtual server in apache just for amp
08:08.39samborambothough, I'm no apache expert
08:08.47GaryHHi, has anyone tested chan_capi with asterisk 1.2.0beta1 yet?
08:11.23*** join/#asterisk VoIPMasta (n=John@dsl-200-95-42-39.prod-infinitum.com.mx)
08:11.43VoIPMastaHi, I'm having some problems with a dialplan... is someone available and willing to help?
08:11.54samboramboX-Rob: is there any documentation on this?
08:12.30*** join/#asterisk zoo (i=nobody@ip-172-16.travedsl.de)
08:17.09VoIPMastaWhen I try to forward a call it just rings once and then is cut off
08:17.49*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
08:17.52*** join/#asterisk mastix (n=mastix@ip-85-160-13-68.eurotel.cz)
08:18.09mastixhi
08:18.14VoIPMastaif anyone can help, the logs and confs are in http://pastebin.com/353419
08:18.39*** join/#asterisk atmel (n=vlad@ip68-4-101-199.oc.oc.cox.net)
08:21.04mastixhi does anyone know where can i find some help for configuration of h.323
08:21.33mastixwould be great if some manual exists
08:22.31VoIPMastamastix: pm me, I might be able to help you
08:23.58mastixsorry what does it mean "pm me"<
08:24.01mastix?
08:24.10VoIPMastasend a query/private message/msg/etc
08:34.25VoIPMastaanyone here has experience forwarding calls from a DID provider to a SIP extension?
08:36.25*** join/#asterisk Tili (n=Tili@202-133-67-168-dialup.sat.net.pk)
08:37.46PakiPenguinhey Tili
08:37.58PakiPenguinVoIPMasta, i've tried it and it works
08:38.00Tilihi PakiPenguin
08:38.03PakiPenguinwhat do you exactly want to do?
08:38.06VoIPMastagreat :)
08:38.16PakiPenguini mean DID -> SIP :)
08:38.21VoIPMastaohh
08:38.26VoIPMastacan you help me a little bit here?
08:38.30PakiPenguinyeah sure why not
08:38.35VoIPMastaI thought that you were talking about your h323 issue
08:38.39PakiPenguinwhat are youe xactly trying to do
08:38.43PakiPenguinhaha i am still compiling pwlib
08:38.57VoIPMastaI just purchased a DID from a DID provider, they deliver the DID to a regular softphone
08:39.12PakiPenguinokay
08:39.15VoIPMastaso I added a register => line in my sip.conf, so that my asterisk box acts as a client
08:39.20PakiPenguinfine
08:39.35VoIPMastamy asterisk server registers with the provider's gateway
08:39.42PakiPenguinso when you call your did , what happens?
08:39.52VoIPMastanow, when I dial the DID from my cellphone, it just rings once and then hangs up
08:40.03VoIPMastabut my cellphone keeps displaying the call as active
08:40.27VoIPMastahttp://pastebin.com/353419 <-- that will give you an idea of what's going on
08:42.06*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
08:43.01*** join/#asterisk glm2k_ (n=GLM@rrcs-24-199-11-41.west.biz.rr.com)
08:46.17*** join/#asterisk drbrown (n=chatzill@63.238.118.90)
08:49.34zedkatufofftopic Q: If I want to flush all iptables rules, I simply do iptables -F ..is this correct?
08:50.26zedkatuf(my problem atm is that I can't access my asterisk box..ie no pinging possible...I'm wondering if it's as a result as of installing shorewall on my * box :P)
08:51.01zedkatuf(so I've stopped shorewall & done iptables -F but still no access..the phyiscal eth connectors are ok)
08:51.25*** join/#asterisk nemisus (n=nemisus@203-217-87-135.dyn.iinet.net.au)
08:52.51*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
08:58.03*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
09:01.05*** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net)
09:01.05*** join/#asterisk Netgeeks (n=Chris@68-185-24-2.static.mdfd.or.charter.com)
09:01.05*** join/#asterisk Meaty-Wrk (n=cp_simbu@office.abi.ca)
09:01.06*** join/#asterisk IronHelixz (n=irc@ool-45785cfe.dyn.optonline.net)
09:01.06*** join/#asterisk TheSexyKen (n=ksandell@c-24-5-129-114.hsd1.ca.comcast.net)
09:04.00[Airwolf]Is it also possible to somehow script my sip configuration ? So that I don't have to make the same entry 30 times ?
09:04.27Byte?
09:07.53[Airwolf]I have a few sip configurations now, but they are all the same. So I just want one configurations and then just specify the number and password
09:08.41*** join/#asterisk Romik (n=romik_@1.fix.netvision.net.il)
09:14.21newl[Airwolf]: you can use a macro.
09:18.49*** join/#asterisk apardo (n=apardo@23.Red-83-44-179.pooles.rima-tde.net)
09:23.12[Airwolf]newl, just the same as I use in my extensions.conf ?
09:23.21[Airwolf]I didn't knew that was possible, tnx
09:32.47newl[Airwolf]: yep.  I use a macro for dial actions, makes things much cleaner with larger dial plans.
09:36.01*** join/#asterisk RoyK (n=roy@cm-80.111.22.187.chello.no)
09:37.02Assidokay .. i have this funny issue.. my format in cdr_custom.conf but.. same thing in cdr.conf doesnt get updated (in the database)
09:37.20*** join/#asterisk Alcee (n=Alcee@tre93-3-82-237-221-213.fbx.proxad.net)
09:42.57*** join/#asterisk razu (n=razu@217-159-242-106-dsl.est.estpak.ee)
09:58.13PakiPenguinhmm
10:04.05*** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-68-74-8-29.dsl.sfldmi.ameritech.net)
10:04.15*** join/#asterisk meppl (n=mephisto@p54AAFD76.dip.t-dialin.net)
10:06.25Assidokay i wanna change the format of cdr_pgsql
10:06.34Assiddo i just add a format => line ?
10:06.45Assidor does it go into another conf file?
10:08.44*** join/#asterisk r0m (n=SysOp@bl8-28-113.dsl.telepac.pt)
10:08.49r0mgood morning!
10:13.04*** part/#asterisk zoo (i=nobody@ip-172-16.travedsl.de)
10:21.43*** join/#asterisk phoe (n=phoe@82.171.44.25)
10:22.38phoegood afternoon
10:32.25coppicegood evening
10:33.34*** join/#asterisk hat (n=hat@cm157.epsilon173.maxonline.com.sg)
10:35.10hathi, i am new to asterisk but we need to deploy a not too small system support web/sms callback for 5000 users. Who can suggest the hardware configuration to me?
10:35.28*** join/#asterisk JamesDot1om (i=jamesdot@sweep.bur.st)
10:36.21*** join/#asterisk drbrown (n=chatzill@63.238.118.90)
10:41.19*** join/#asterisk sneak (n=sneak@64.220.234.21.ptr.us.xo.net)
10:47.50*** join/#asterisk sigmounte (n=sigmount@www.sighq.net)
10:50.09*** join/#asterisk ksh_ (n=ksh@pcp08979908pcs.huntsv01.al.comcast.net)
10:52.55*** join/#asterisk bjohnson (n=bjohnson@i216-58-14-59.cybersurf.com)
10:54.23jeffikanybody know how to use meet-me?
10:54.53wasimjeffik: how hard can it be?
10:55.22jeffikwasim: i have never done it and have been reading and can not figure it out
10:55.35phoewasim: hard if you dont know what meet-me is :)
10:56.03jeffikwasim: as i understand it's a conference bridge users can dial in to
10:56.09wasimjeffik: it is indeed
10:57.02jeffikok, so may i ask you how to set up a room? for example my extension is 1500 as i understand i should be a blre to dial 81500 and create a conferene room
10:57.49wasimin meetme.conf put conf => 1500
10:57.59wasimin extensions.conf put exten => 1500,1,Meetme(1500)
10:58.01wasimdial 1500
10:58.56wasimdon't forge to reload, do a show application meetme for more options, and read the wiki
10:59.27wasimoh, AND you need a timing interface like a zapcard  or ztdummy
11:00.06jeffikok i'll read more thanks
11:00.38phoeAsterisk looks very nice, but it's all very complicated to me :P
11:00.48phoegonna start using is this monday
11:00.54wasimwhy wait?
11:01.18phoelol, tbh I did a little bit at my home pc
11:01.26phoeit's for work experience, and that starts monday
11:01.51phoenever done anything before with VoIP or Linux. it's all very new to me :P
11:02.00phoeonly managed to install it so far hehe
11:02.45newlheh wicked, Meetme worked.  That was simple.  I've been wondering the same for ages (knew what it was and where to look for config, but never did anything about it until now).
11:04.36*** join/#asterisk zedkatuf (n=audela@82-33-115-145.cable.ubr08.azte.blueyonder.co.uk)
11:08.41Alcee19h30 ?
11:08.54Alceesorry .... bad window
11:11.56visik7_is there a way to customize the call parking commands? like change the # or the behaviour of the app ?
11:20.15wasimvisik7_: features.conf
11:23.38*** join/#asterisk _Pkunk (n=Pkunkage@mbbs.munnabhai.info)
11:37.07*** join/#asterisk Pkunk778 (n=Pkunkage@mbbs.munnabhai.info)
11:40.24*** join/#asterisk queuetue (n=queuetue@h69-21-252-54.69-21.unk.tds.net)
11:42.26queuetueI have a broadvoice account inbound to Asterisk.  BV gives me a second number to dial into that same account - is there any way to makse * act differently when that call comes in on the alternate number?
11:42.37konradsqueuetue: yeah
11:42.59konradsqueuetue: if the broadvoice signals on what number it is incoming (that a sip/voip sutff?)
11:43.08konradsyou can do a [broadvoice-in]
11:43.14konradsand exten => number1,rules...
11:43.24konradsexten => number2,rules...
11:43.44queuetuekonrads: It is sip.  I'm not sure what "signals" means, though.
11:43.55konradsqueuetue: in your case it is fine :)
11:44.12konradsqueuetue: just do as I said, write two extensions
11:44.34konradsexten => SipNr1,Dial(CAPI/contr1/b123)
11:44.44konradsexten => SipNr21,PlayMpr
11:44.46konradsmp3
11:44.51queuetueI'm looking at the reports that amportal provides and the source on both numbers seems identical.
11:45.08queuetueDoes that not matter?
11:46.37*** join/#asterisk meppl (n=mephisto@p54AAEDEB.dip.t-dialin.net)
11:46.50*** part/#asterisk lters (n=lters@mrtcdsl-034.mis.net)
11:47.03konradsqueuetue: when broadvoice gives you a call, it gives you a SIP INVITE in form extenssion@yourhost.whatever.net
11:47.14konradss/ss/s/
11:47.40konradsi would exptect, that your two numbers would have different extensions
11:47.41queuetueAnd how do I "see" that invite?  sip debug?
11:47.51konradsqueuetue: asterisk console should show it
11:47.58konradsbut sip debug will definetl;ey so
11:49.15*** join/#asterisk pauldy (n=pauldy@c-24-1-152-131.hsd1.tx.comcast.net)
11:50.09queuetueBoth show SIP/Phone1-random  phone2 does not show.
11:50.48konradscan you paste those two strings?
11:50.55konradsYou should look for a To: string iirc
11:53.22queuetueFROM_DID ?
11:53.51konradslook for INVITE envelope
11:53.57konradswith sip debug turned on
11:58.17*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
12:01.05*** join/#asterisk qweee (n=vikramhe@61.246.231.50)
12:01.35queuetueOn with BV support...
12:01.49visik7_wasim is it configurable only in 1.2.x branch or olso in 1.0.x
12:01.52visik7_?
12:02.05queuetuedoes anyone know an incoming-only company that I could deal with instead of BV?
12:02.25queuetueI'm paying for outgoing BV service s I don't need..
12:02.29*** join/#asterisk loick (n=loick@APuteaux-151-1-63-126.w82-120.abo.wanadoo.fr)
12:02.45*** join/#asterisk podzap (n=podzap@roswell.pp.saunalahti.fi)
12:02.49podzaphi
12:03.10podzapi am having a really hard time getting * setup. i need some help.
12:03.36podzapi tried asteriskathome and it seems like an overcomplicated, buggy mess.
12:04.04podzapi just need to get a bare-bones * setup, please, who can help by sharing some config files, etc?
12:05.48queuetuepodzap: A@H is remarkably simple, and probably th very best way to get a bare bones setup...  What specifically is causing a problem ( A@H or otherwise?)
12:06.53podzapqueuetue: A@H , I can't get it to connect to FWD, although I followed the example in the A@H handbook verbatim.
12:07.03*** join/#asterisk apardo (n=apardo@23.Red-83-44-179.pooles.rima-tde.net)
12:07.32podzapqueuetue: i have a sipura-2002. i configured the other line with a second fwd account, and could connect directly, no problem. but not with A@H.
12:08.05queuetuepodzap: that seems pretty strange - did debugging shed any light as to why?
12:08.09podzapqueuetue: i looked at the config files, A@H leaves include lines commented out, but those commented out files are the ones which are edited via the GUI
12:09.16podzapqueuetue: it would immediately drop the IAX2 connection after establishing it
12:09.50queuetueDid it actually establish and drp, or did it not successfully establish?
12:09.54podzapthere are something like 50 configuration files
12:10.05podzapit established and dropped
12:10.51podzapmy brain can not wrap itself around 50 configuration files. i want to get an extremely simple 3-4 file setup working, then expand from there.
12:11.01queuetueYes, you are not supposed to be digging in A@H' config files.  If you have t hand-tweak then install asterisk and do it yourself... A@H is not meant to be tweaked - definitely not by a newbie...
12:11.14podzapwell, it does not work
12:11.15drrayor by anyone
12:11.44podzapnot according to the examples in the handbook
12:12.22queuetueIdf you're doing config changes by hand, then i'm not surprised it it not working.  I guess install asterisk , and then try one of the many explanations of how to get pulver up and going on asterisk.  Come on back with specific problems...
12:12.44podzapi didn't try to change anything until after it did not work.
12:13.06queuetueOk.  My advice doesn't change. :)
12:13.31podzapi can put the original config files back, and show you, it still does not work.
12:15.07queuetueGive em the url for the FWD setup on A@H.
12:15.35podzapjust a second
12:19.02podzapthe box is really slow right at the moment, gotta wait til it calms down
12:19.30*** join/#asterisk zotz (n=zotz@24.231.36.100)
12:20.09queuetuepodzap: give me (in here or pmsg) your pulver number so I can test the setup.
12:22.21podzappm
12:24.04*** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn)
12:29.40*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-241-108.buckeyecom.net)
12:35.15*** join/#asterisk djspeck (n=djspeck@ewersbach.net)
12:35.32djspeckHELLO everybody
12:36.17djspeckI have a problem: when i run asterisk as root, everything is fine, but when i try to run as asterisk i don't hear anything...
12:36.23djspeckasterisk is in group audio
12:36.58*** join/#asterisk simprix (n=simprix@24-247-14-62.dhcp.aldl.mi.charter.com)
12:37.19simprixWhat is a good speed box for a 25 user system with a pri
12:37.45konradsa new PC?
12:37.59simprixyea
12:38.43*** join/#asterisk zedkatuf (n=audela@82-33-115-145.cable.ubr08.azte.blueyonder.co.uk)
12:40.44konradsi meant a new PC is enough
12:41.13konradsbasically, CPU power is required when you translate from codec to codec
12:41.21konradswhich is sometimes the VoIP case
12:41.45konradsotherwise I/O is the limiting factor
12:42.00konradsif you have active ISDN card, a recent server class stuff should work
12:42.12konrads1CPU P4 2.4Ghz will be more than sufficient
12:43.04visik7someone should split essential config files from auxiliary ones
12:43.43*** join/#asterisk asteriskn00b (i=user65@adsl-68-91-7-225.dsl.tulsok.swbell.net)
12:44.10asteriskn00bmorning all
12:44.41cochimaybe OT. but is there some tool which emulates CAPI by accessing TAPI? got some app which just supports showing callerid using CAPI :|
12:46.17*** join/#asterisk drbrown (n=chatzill@63.238.118.90)
12:46.39*** join/#asterisk Rhinos (n=sev@ram94-3-82-225-10-84.fbx.proxad.net)
12:49.07asteriskn00bget anyones opinions on the best hosted pbx solution out there?
12:56.17djspeckhello everybody.. i have a problem with starting asterisk:
12:56.27djspeckwhen i start it as root everything works fine
12:56.42djspeckwhen i start asterisk -U asterisk everything works fine
12:56.55djspeckbut when i try to start it from init.d
12:57.10djspecki don't hear anything
12:59.49*** join/#asterisk MikeJ__ (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
13:04.45*** join/#asterisk w0w0_ (n=apardo@23.Red-83-44-179.pooles.rima-tde.net)
13:05.28*** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de)
13:06.53cochimh is there any way to dial using another channel if the first group's full? like CAPI ports full, switch to SIP for dialout?
13:08.35wasimcochi: see which priority it goes to if the CAPI ports are full, and put your SIP on that, should be +1
13:09.43cochiwell but isn't dial supposed to go to next prio if other party doesnt answer, too?
13:10.22wasimyou can do a chanisavail as well
13:10.33cochioh wasnt aware of that command. thanks :)
13:13.45Ariel_cochi, you can use a macro to create your dial checks before you dial out.  You can also count how many calls are on a channel with setgroup. Take a look on the wiki.
13:13.50Ariel_~docs
13:13.52jbotsomebody said docs was 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
13:14.09cochialready doing this via macro. just had overlooked the ChanIsAvail command :)
13:14.09Kattybeep.
13:14.14cochithanks for your help :)
13:14.28Ariel_Katty, morning...
13:14.43KattyAriel_: (=
13:15.04Ariel_Katty, I was under the impression you had a 3 day weekend and were going on a trip?
13:15.16*** join/#asterisk _cleric_ (n=dacleric@p54829007.dip0.t-ipconnect.de)
13:15.29KattyAriel_: just to mamma's.
13:15.46Ariel_ahh
13:26.16*** part/#asterisk Grubs (n=Miranda@c220-239-96-230.eburwd5.vic.optusnet.com.au)
13:27.49*** join/#asterisk Nukemizer (n=Nuke@160.7.249.15)
13:27.52podzaphi, where can i find a simple howto: asterisk, FWD, with sipura-2000?
13:32.20*** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim)
13:34.27Nugget~docs
13:34.28jbotfrom memory, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
13:36.38phoewhat does wiki mean?
13:38.24Ariel_phoe, the wiki is like a live reference doc's that if you have access too you can edit and post your info there as well.
13:38.38Nuggethttp://justfuckinggoogleit.com/?q=what+is+wiki
13:39.05*** join/#asterisk ChrisN (i=[v838MAk@166.84.1.5)
13:39.40NivexAh yes.  Google is the answer!  I don't care what the question is! :)
13:39.49*** join/#asterisk Rav1974 (n=r@ool-18bd2d63.dyn.optonline.net)
13:39.57NuggetIt's certainly the answer to that question.
13:39.59Ariel_Nugget, would it not have been easyer to just do this:
13:40.05Ariel_~google what is wiki
13:40.37Nuggetthat perhaps would not have made the point quite as well.  especially in light of your lack of success just now.  :)
13:41.02Ariel_looks like jbot is not going out to the google today
13:41.11Ariel_~weather KTMB
13:41.15PakiPenguinjbot's been drinking
13:41.25Ariel_well it's kinda working
13:41.59Rav1974google is so great, don't know what I did without it
13:42.45Ariel_Rav1974, have you yahoo.... they post they have more but they lie
13:43.19Ariel_before google.... I used to use altavista allot back then.
13:43.26Nuggetbefore google I altavista'd.  before altavista I yahoo'd.
13:43.31gordonjcpaltavista was good
13:43.48Nuggetbefore yahoo, everyone had a big list of cool links on their home page.  :)
13:43.51Rav1974yesterday, 2 printers started spewing out this foul smell, changed drum & toner but it was still smelling. Called brother for any known issues, they had none.  Looked on google & then found the answer!!!
13:44.22Ariel_Rav1974, what you found you needed to change the ozone filter
13:44.25Rav1974it was the paper.  It was the most aweful smell.  Like vomit.  The receiptionist threw up twice.
13:44.50Rav1974Ariel_: we just used a different brand of paper and it was fine
13:45.01*** join/#asterisk Alcee (n=Alcee@tre93-3-82-237-221-213.fbx.proxad.net)
13:45.22Rav1974Thank god for google
13:45.35Rav1974if they made it a paid service, I'd subscribe
13:45.50Ariel_don't evey think of that
13:46.07Rav1974:)
13:46.12Ariel_argh  paid service what will we do... Oh it's already paid service it has ads
13:46.49Rav1974true
13:47.24Ariel_in fact on there site the first 2 or 3 are paid service links... kinda selling out in my view
13:48.16Rav1974at least it works, searching with other companies doesn't work as well
13:48.28Ariel_ok I have an error with a php page on Suse. Anyone here good with php and suse?
13:48.59Rav1974all the tech dudes are sleeping
13:49.55Ariel_well I think this is more of something missing in suse's apt installer.
13:50.32*** join/#asterisk DrJolo (n=chatzill@host-ip58-113.crowley.pl)
13:50.39*** join/#asterisk bjohnson_ (n=bjohnson@i216-58-59-46.cybersurf.com)
13:51.38Rav1974phoe: those tech guys are going to be grogy so you better summon some women to wake them up
13:52.07Ariel_Rav1974, sometimes it's not a matter of women but beer
13:52.37phoemy skill is too low to summon women :(
13:52.59Ariel_beer then
13:53.30phoewhen was the first release of Asterisk?
13:53.40Rav1974:)
13:53.53Ariel_last year version 1 around the first astercom
13:54.02phoeastercom?
13:54.07Rav1974zapatatelephony.org was the starting point
13:55.07Ariel_astricon
13:55.09Ariel_sorry
13:55.25phoelol, nm. I dunno anything about it ;)
13:55.48mog_homeasterisk is aprox 5 years old
13:55.48Ariel_Rav1974, yes it was I was using .5 version a little over 3 years ago. It was very different back then. No wiki yet
13:55.53mog_homein the current form 4 years old
13:56.05mog_homea non threaded version was made for lss back in the day
13:56.19mog_homemark rewrote his code for a threaded version 10 -12 months later
13:56.25mog_homeand is the same base we use now
13:56.32Ariel_mog_home, great info
13:56.41*** join/#asterisk Corydon76-home (i=orange@pdpc/supporter/sustaining/Corydon76-home)
13:56.54Ariel_perks hummmmm
14:08.04*** join/#asterisk header (n=maxgluck@200.109.166.83)
14:10.35*** join/#asterisk natted (n=maxgluck@200.109.166.83)
14:12.38nattedGood morning everyone... I'm having some problems with NAT, for example, an X-Lite receiving calls that were dialed to an ATA-186, both behind the same address... The Asterisk Server is outside the NAT, any suggestions please?
14:15.10*** join/#asterisk clyrrad (n=ddd@CPE000c4120d329-CM0011aea484a4.cpe.net.cable.rogers.com)
14:18.29clyrraddoes anyone know if you can pattern match using gotoif something like exten => s,1,GotoIf($["${ARG1}" = "_XXXX"]?16:2)
14:19.24Ariel_yes but that matches everything
14:19.32Ariel_take the _ off it
14:22.55clyrradexten => s,1,GotoIf($["${ARG1}" = "2XXX"]?16:2)
14:23.01clyrraddidnt seem to make much difference
14:23.25clyrradthat "should" matching all 4 digit extensions starting with 2
14:23.51Ariel_wait
14:24.45PakiPenguinanyone here uses linksys pap?
14:24.49*** join/#asterisk DrJolo (n=chatzill@host-ip58-113.crowley.pl)
14:25.41ZawPakiPenguin: i have one, but it sucks and goes down all the time. i'm guessing it's just my particular one going bad, but i dunno.
14:27.42PakiPenguinZaw, mine is acting strange , sometimes it calls , sometimes it just gives me a fast busy
14:28.27Zawyep
14:28.33Zawsometimes it doesn't have a dial tone
14:28.38Zawthat's what mine does too
14:29.50Ariel_clyrrad, exten => s,1,GotoIf($[${ARG1} = "2XXX"]?16:2)
14:30.32Ariel_PakiPenguin, is it behind a nat?
14:31.28clyrradAriel_ that yields the same results, * skips right over it... It does not seem to evaluate the pattern expression...
14:31.52Ariel_your getting this from another macro
14:32.07clyrrad?
14:32.25clyrradI am in a macro where I run that line you just pasted... if thats your question ;)
14:32.48Ariel_what is your dial string that goes to this macro
14:33.28clyrradthe only thing that goes to this macro is an extension number
14:33.42clyrrada 4 digit extension thats held in ARG1
14:34.26PakiPenguinAriel_: yeah
14:34.28PakiPenguinit is
14:34.37PakiPenguinand i cant seem to find a place to add a stun server
14:35.30Ariel_PakiPenguin, your firewall/nat is not keeping the port open need to set it to register less then every minute. This might be the issue
14:35.42Ariel_qaulify=yes might help with this
14:36.12*** join/#asterisk drbrown (n=chatzill@63.238.118.90)
14:36.18clyrradAriel_ is it because I am in a macro that gets it all screwed up?
14:36.42PakiPenguinhmmjs testing
14:36.43Ariel_clyrrad, it's suppose to work in a macro
14:36.59clyrradhrm.... where are we going wrong then? lol
14:38.09PakiPenguini think thats the issue!
14:38.14PakiPenguinthanks Ariel_
14:38.22Ariel_PakiPenguin, np
14:39.09*** join/#asterisk |cleric| (n=dacleric@p5482A5D6.dip0.t-ipconnect.de)
14:41.26Ariel_clyrrad, put a exten => s,1,NoOp(${ARG1}  exten => s,2,GotoIf($["${ARG1}" = "2XXX"]?16:3)
14:41.38Ariel_just to see what you have for ARG1
14:41.51Ariel_clyrrad, put a exten => s,1,NoOp(${ARG1})  exten => s,2,GotoIf($["${ARG1}" = "2XXX"]?16:3)
14:41.59Ariel_sorry forgot the )
14:42.42clyrradlol thats ok i cought that :)
14:43.24clyrradarg1 = 2000
14:44.15konradsi thikn that asterisk extensions are the most scary part
14:44.21konradsthe rest either works or not
14:44.27konradslike my capi stuff :(
14:44.38konradsor h323->sip bridging
14:47.20clyrradAriel_ it works like this exten => s,2,GotoIf($["${ARG1}" = "2000"]?16:3)
14:47.38clyrradonly problem is I would need to do that for EACH extension.... that would be a real pain
14:50.18PakiPenguinkonrads: capi always works for me
14:53.51*** join/#asterisk fataia`falamu (i=cLeRiSY@CPE-61-9-197-32.nsw.bigpond.net.au)
14:54.37*** join/#asterisk kwoot (n=kvirc@i2rs-son.xs4all.nl)
14:54.55konradsPakiPenguin: did you use misdn with capi?
14:55.53kwoothello. Is there anybody willing to help me out configuring zaphfc to an BRI? I am really stuck now.
14:56.11*** join/#asterisk juice (n=juice@mo-69-69-116-98.sta.sprint-hsd.net)
14:57.32*** join/#asterisk patrick_ (n=patrick@pc-205-21.scpe.quickclic.net)
14:58.54clyrradAriel GotoIf(Condition?label1:label2) is from the manual, perhaps we cant do pater matching, and can only use a LABEL or Static value?
14:59.18kwootwhen starting * it  does say it sees 1 bri interface with 2 channels but when I want to call it says no channel avail. Why?
15:02.28*** join/#asterisk rbd (n=robbyd@24-181-194-064.dhcp.hckr.nc.charter.com)
15:07.30*** join/#asterisk Nukemizer (n=Nuke@160.7.249.15)
15:07.40kwoothello? anybody?
15:07.52*** join/#asterisk pr0m (n=pr0methe@24-75-196-70.chvlva.adelphia.net)
15:08.20Wonkahttp://data.4chan.org/b/src/1125712295576.jpg
15:09.03hugo-v6Wonka: lol
15:09.14phoehi kwoot
15:09.17Wonkanot nearly pc, but funny :)
15:09.37*** join/#asterisk Byte (i=byte@2001:4bd0:1000:0:202:44ff:fe47:d3ee)
15:11.29kwootHy phoe: okay. I see somebody is awake :-) very good. Can you help me with the question I stated earlier?
15:12.11phoeno sorry :), I don't know anything about that stuff
15:12.31kwootthats too bad. thanks at least for reading.
15:13.05*** join/#asterisk Byte (i=byte@proxima.arlott.org.uk)
15:13.06*** part/#asterisk ChrisN (i=[v838MAk@166.84.1.5)
15:17.56*** join/#asterisk file[laptop] (n=jcolp@156.34.35.106)
15:20.19[Airwolf]I'm wondering. I'm using macro's in my dailplan. But is it possible to use a macro in de sip configuration. Because administration of sip.conf will be more difficult when I have more phones.
15:20.27[Airwolf]Now I have something like this:
15:20.28[Airwolf]restrictcid
15:20.31[Airwolf]aarg
15:20.34[Airwolf]this:
15:20.37[Airwolf][498]
15:20.37[Airwolf];Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
15:20.37[Airwolf];Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
15:20.37[Airwolf]type=friend
15:20.37[Airwolf]regexten=498                 ; When they register, create extension 1234
15:20.37[Airwolf]username=joost
15:20.39[Airwolf]callerid="Joost" <498>
15:20.42[Airwolf]host=dynamic
15:20.43[Airwolf];nat=yes                       ; X-Lite is behind a NAT router
15:20.45[Airwolf];canreinvite=no                ; Typically set to NO if behind NAT
15:20.49[Airwolf]disallow=all
15:20.51[Airwolf]allow=gsm                     ; GSM consumes far less bandwidth than ulaw
15:20.53[Airwolf]allow=ulaw
15:20.55[Airwolf]allow=alaw
15:20.57[Airwolf]mailbox=498@default
15:20.59[Airwolf]context=putgraaf
15:21.01[Airwolf]And I would like to have just one configuration and then just one line for every user to configure it
15:21.06[Airwolf]Like I have in my dial plan
15:21.53*** part/#asterisk greekman (n=alex@host254.209.113.199.conversent.net)
15:22.16newlafaik, that isn't possible.
15:22.54drumkillaactually, yes you can
15:23.07drumkillain cvs head, there is a way to do inheritance in configuration ...
15:23.36newlfrom AEL to ASL? lol
15:23.52drumkillaIf you read doc/README.configuration in the source directory, it will tell you about how it works
15:24.48drumkillait's not the same as a macro, but ... it cuts down on repeated config
15:25.10drumkilla~thwack file[laptop]
15:25.13jbotACTION smacks file[laptop] on the forehead with a toaster
15:25.17drumkilla!
15:25.25newlflee!
15:25.57file[laptop]drumkilla: what'cha doing?
15:26.22*** join/#asterisk hat (n=hat@cm157.epsilon173.maxonline.com.sg)
15:26.54clyrraddoes anyone know how to pattern match with GoToIF?  Seems * does not.... exten => s,2,GotoIf($["{ARG1}"= "2XXX"]?16:3)
15:27.20[Airwolf]drumkilla, thanks
15:27.36*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-241-108.buckeyecom.net)
15:27.48[Airwolf]But I'm not using the cvs-head
15:28.24[Airwolf]But isn't it possible to use variables in de sip.conf. That would atleast make it a littlebit easier.
15:28.39*** join/#asterisk Samoied (n=Samoied@201.3.210.150)
15:28.49*** join/#asterisk chet (n=chet@cpe-065-190-056-004.triad.res.rr.com)
15:29.01drumkillano, it's not.
15:29.05hathi, we need to setup a asterisk web callback system. The buget is about USD$6,000(including all hardware). What configurations of these hardware ?
15:29.06clyrradAirwolf.... I was not able to get variables to work in SIP.conf or IAX.conf, only in EXTENSIONS.conf
15:29.10drumkillafile[laptop]: about to get my party on :D
15:29.20*** join/#asterisk used (n=used@c-24-22-125-179.hsd1.or.comcast.net)
15:29.35[Airwolf]clyrrad, ok then I just have to copy & paste
15:29.35[Airwolf]:P
15:29.51clyrradyup, thats what I had to do.... kinda sucks but at least it works
15:29.55file[laptop]drumkilla: excellent
15:30.23drumkillafile[laptop]: first football game is today, so this town is going crazy
15:30.39file[laptop]not as crazy as here, the rolling stones play tonight
15:30.45file[laptop]so it's a wee bit insane
15:31.36kwootcan someone tell me how to debug the dial application to find out why * doesn´t use a channel?
15:32.10kwooto, sorry. Addon: please. :-)
15:32.27file[laptop]kwoot: that made no sense to me
15:32.34*** join/#asterisk Laureano (n=tomas@OL155-33.fibertel.com.ar)
15:33.05kwootwell, i seem to have a working zaphfc, but I still can not dial out. Dunno why.
15:35.14hatwhich digium card i should get to support maximum numbers of concurrent calls and less CPU usage?
15:35.20kwooteh, hello?
15:35.37kwoot<hat> an expensive one? :-)
15:35.40konradskwoot: error messages?
15:35.44file[laptop]my world doesn't revolve around IRC you know
15:36.03[Airwolf]file[laptop], is there any other world then ?
15:36.10hatmy budget is about 6000 USD$. Of course, it must be stable
15:36.19file[laptop][Airwolf]: indeed, the real world
15:36.26kwootkonrads:  unable to create channel of type zap
15:36.32chethat, here are some products- http://www.digium.com/index.php?menu=product_category&category=hardware
15:36.33hatkwoot, including a server
15:36.35[Airwolf]file[laptop], aahhh, I have to check it out one time ;)
15:36.45hatchet, thanks & let me see
15:38.18*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal)
15:38.29file[laptop]hahahahahahaha
15:38.29*** part/#asterisk Laureano (n=tomas@OL155-33.fibertel.com.ar)
15:38.35file[laptop]my parents went to PEI
15:38.51chetwhich linux distro do you guys recommend for *? cent, debian, etc..
15:39.02chetor freebsd?
15:39.18[Airwolf]My personal favorites are debian and gentoo
15:39.30[Airwolf]But that's different for everybody
15:39.33chetyeah
15:39.39Nuggetasterisk is the same no matter what unix you use.
15:39.44chetjust wondering what some people have had success with
15:39.51Nuggetzaptel is a total pain in the ass unless you use linux.
15:39.51file[laptop]PARTY!
15:39.57chetwell, hasnt there been some issues with freebsd?
15:40.04Nuggetonly with zaptel
15:40.07chetok
15:40.27hatchet, what is 3.3 volt PCI and 5.0 volt PCI ? DOes it mean i need to choose my server first? and what is the difference between echo cancellation card and non echo-cancellation-card?
15:40.29NuggetI run three asterisk servers.  One in Linux, one in FreeBSD, and one in OSX.
15:40.42chetyes hat, based on motherboard specifications
15:41.03chethat- you may want to find asterisk consultants on voip-info.org and ask for recommendations
15:41.24chetNugget, for recreation or production?
15:41.44Nuggetboth
15:42.35hatchet, i hope i can get help from irc :)
15:42.43NuggetMy home machine is linux because I wanted a zaptel card.  My "production" server is freebsd because I didn't need zaptel.  My powerbook is osx because osx rocks.
15:42.53file[laptop]Nugget: yes, yes it does
15:43.20chetvery nice
15:43.22NuggetI run asterisk on the powerbook because I've had much better luck running x-lite connecting to a local asterisk install and then talking iax to my "real" server.
15:43.37Nuggetsip over hotel ethernet with its nat hell is often impossible
15:44.23chetyeah, i read they were looking at making or changing sip for that reason? and other business reasons
15:44.25*** join/#asterisk DrJolo (n=chatzill@host-ip58-113.crowley.pl)
15:44.33Nugget"they"?
15:44.47chetwell, some people were proposing rfc additions
15:44.57cheti think i read it in networkworld
15:45.00Netgeekshat: just check the manual on the motherboard your computer will use, it will tell you whether or not your PCI slots are 3.3 or 5.0 volts
15:45.26file[laptop]SIP has changed so much, it's not funny
15:45.46Netgeekshat: as for echo cancelling versus lack of echo cancelling in hardware.  Once the hardware and drivers are stable, the chard with echo-cancelling will perform better both in CPU utilization as well as in echo cancelling quality
15:45.49*** join/#asterisk DrJolo (n=chatzill@host-ip58-113.crowley.pl)
15:45.54file[laptop]"Let's add this, let's change this - oh one method of putting a call on hold? nooo - let's add another!!!"
15:46.08chetfile[laptop]- has that changed interoperability? or has everyone pretty much kept up?
15:46.16Netgeekshat: right now, if I understand it correctly people are having stability issues with the cards that have the on-board canceller
15:46.23file[laptop]chet: for the most part stuff works, as implementation of basic things are fine
15:46.26hatthanks Netgeeks
15:46.28file[laptop]and most stuff retains backwards compatibility
15:46.32chetcool
15:46.35*** join/#asterisk drbrown (n=chatzill@63.238.118.90)
15:46.52file[laptop]there's not many ways you can screw up a basic SIP call
15:46.52NetgeeksI'm writing an extension to the rfc to allow SIP to flush my toilet
15:47.08file[laptop]Netgeeks: INVITE to the toilet, then send BYE :P
15:47.13chethttp://www.networkworld.com/techinsider/2005/061305ti-sip-b.html
15:47.19drumkillawhat does the SDP look like?
15:47.19chetarticle on sip-b
15:47.41Netgeeksfile: not so simple, due to the whole water conservation thing, we have to check the status before the flush
15:47.42hatNetgeeks, do you mean current echo-cancellation-card is not stable?
15:47.58Netgeeksif the bowl is fresh, no need to flush...
15:48.02file[laptop]Netgeeks: ah so you get a status code back from the toilet
15:48.18chetif=yellow then=mellow
15:48.21*** join/#asterisk bmg505 (n=leon@rndf-146-11-247.telkomadsl.co.za)
15:49.19NetgeeksBut the toilet hardware is incapable of determining bowl status (it only knows full of water or empty) so I was thinking about attaching a web cam and using some code to determine cleanliness via the some visual recognition system.... ;)
15:49.31chetnice
15:49.50Netgeekshat: the mix of hardware/drivers... I'm not sure, I haven't played with the echo can cards yet
15:50.27NetgeeksAs for the current software echo-can in asterisk, it doesn't crash asterisk, so you could call it stable.
15:50.48hatthanks. Netgeeks.
15:50.52NetgeeksHeya drumkilla: did you by chance get my email with updates on some work I'm looking to get done?
15:51.17file[laptop]Netgeeks: drumkilla is very very silly
15:51.53Netgeeksand that makes him different from the rest of us, how?  ;)
15:52.05Netgeeksbesides timecop of course, he's just mean
15:52.06file[laptop]we're only very silly
15:53.26Netgeekswell, for any asterisk devs looking for a little income on the side, I've got some small patches I need done in the next 10 days....
15:53.44*** join/#asterisk bkw__ (n=brian@adsl-69-155-21-122.dsl.tulsok.swbell.net)
15:53.44*** mode/#asterisk [+o bkw__] by ChanServ
15:53.48file[laptop]Netgeeks: what is it?
15:53.59file[laptop]which reminds me, I need to hunt down Leif and submit this to the bug tracker
15:54.02Netgeekshrm, bunch of small stuff.... let see
15:54.19Netgeeksadd setvar= to zaptel, function just like sip and iax setvar
15:54.30*** join/#asterisk huslage (n=used@c-24-22-125-179.hsd1.or.comcast.net)
15:54.31*** join/#asterisk nain (n=nain@137.101.144.131)
15:54.40nainHI
15:54.43file[laptop]a likely story.
15:55.24file[laptop]hey Mr. DJ, in case you forgot
15:55.36file[laptop]I came to get down so you better make it hot, cause I can't jump around when I hear...
15:55.37Netgeeksadd a global system_id, which is a field in all realtime db's, so that when a system loads config info from realtime you can have one database for numerous systems and identify the entries either as general (for all) or for a specific system_id
15:57.07Netgeeksand a few more little patches that I can provide in email if you are interested.  However I'm on a deadline... need them by 9/12 for testing, and if issues are found, fixed by 9/19
15:57.10nainSome one would like to tell me that how to apply patch rtcp_patch_20050817B.patch.txt to asterisk avaialble at http://bugs.digium.com/view.php?id=2863 ?
15:58.40Netgeekswell, I'm going back to sleep....
15:59.20file[laptop]sleep is overrated
15:59.32Netgeeksas is growing old!
15:59.33*** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net)
15:59.56Netgeeksunfortunately they are directly related, older = need more sleep
16:00.16file[laptop]not... OLD!
16:00.49Netgeekswas up til 3 am beating my head against some html/python/sql stuff and had to get up 7 am to help the wife set up her farmer's market stand
16:00.54Netgeeksit's nap time!
16:01.13Netgeeksbut only after watching yesterday's Battlestar Galactica episode
16:02.05Netgeeksif you are interested in doing any of those patches, or want info on the others, let me know file...  compensation is included....  chris at netgeeks dot net
16:03.49*** join/#asterisk Nukemizer (n=Nuke@160.7.249.15)
16:04.44*** join/#asterisk YoYo (i=YoYo@pool-151-199-13-222.roa.east.verizon.net)
16:05.40nainfile[laptop]: Hi
16:06.16file[laptop]...hello
16:07.05naini would like to patch asterisk 1.0.9 against this patch rtcp_patch_20050817B.patch.txt, at http://bugs.digium.com/view.php?id=2863. But i don't know how to apply patch to asterisk ? can u help plz .....
16:07.29file[laptop]download the patch to your asterisk directory
16:07.36file[laptop]and type patch -p0 < filename
16:08.02nainOK, Only needed to patch or i have to recompile asterisk again after patching ?
16:08.15file[laptop]you have to recompile asterisk
16:08.24file[laptop]or else your asterisk won't be patched
16:09.18nainok, it should be like this? patch -p0 < /path-to-path/patch rtcp_patch_20050817B.patch.txt or i have to define the file name which is going to be patch ?
16:09.29file[laptop]nope.
16:09.34file[laptop]just do like I said.
16:09.41nainOk
16:09.58file[laptop]probably won't apply totally, so have fun!
16:10.01DannyFmorning folks
16:10.04DannyFlo file
16:10.08file[laptop]hi hi
16:10.42Ariel_nain, that patch is for CVS Head you said you have stable 1.0.9
16:11.03nainyes i have 1.0.9
16:11.05file[laptop]ah yes, I didn't even see that - yeah you need CVS head...
16:11.08nainso i can't apply it ?
16:11.20file[laptop]and that patch is for August 17th, SOOO it may not apply cleanly to today's CVS
16:11.26file[laptop]nain: correct, it won't work on 1.0.9
16:11.36nainhmmmmm,
16:11.57nainActually i am getting this error and calls going to drop.  what does this error means "Unable to write fd 60 (32, Broken Pipe)" this cause to hangup calls
16:12.04Ariel_there have been lots of changes to the chan_sip
16:12.30*** join/#asterisk darkskiez (n=darkskie@host86-132-169-66.range86-132.btcentralplus.com)
16:12.51file[laptop]nain: it means something very wrong happened
16:13.12file[laptop]because I know what that error means, but I've never seen anybody have it happen, 'nor do I know what would cause it
16:13.25nainfile[laptop]: yes this is very serious error and i m in very trouble i didn't find any help regarding this error.
16:13.41nainI don't konw what's wrong with my setup
16:14.31nainfile[laptop]: would you explain error plz ? what does it mean...
16:14.49file[laptop]it means the file descriptor where it was trying to write to returned an error
16:15.29file[laptop]what are you using for a distro? or dare I ask... OS
16:15.43nainso it is the operating system error causing dropping of calls ?
16:15.45nainsure
16:15.52nainI am using Fedora core 3
16:15.58*** join/#asterisk natted (n=maxgluck@200.109.166.83)
16:16.00file[laptop]that should work fine
16:16.17file[laptop]I think my devel box is FC3, I forget...
16:16.19nainI have asterisk 1.0.9 with asterisk-oh323.0.6.6 running
16:16.46file[laptop]try 1.2.0 beta1
16:16.53nattedhi, does insecure = very really exist? because I can only find explanations for invite and port on the wiki
16:16.53DannyF&%¤%#
16:17.06file[laptop]natted: yes, it's both invite and port
16:17.10DannyFis everyone on sixtel on vacation?
16:17.17*** join/#asterisk Corndawg_ (i=whoisit@c-66-176-249-51.hsd1.fl.comcast.net)
16:17.19file[laptop]DannyF: always.
16:17.21naini tried it but failed to compile so i continue to asterisk 1.0.9 and waiting for stable release of 1.0.2
16:17.29nattedhmmm, ok.
16:17.47DannyFheh tried one of their DID's and it permanently busy... aint even reaching me....
16:18.02DannyFah well, guess i'll us someone else
16:18.03nainsorry asterisk 1.2 not 1.0.2
16:18.05DannyFuse*
16:18.36file[laptop]nain: then the logical course is to figure out why 1.2.0 didn't compile
16:18.50*** join/#asterisk rob314 (n=rob314@cpe-65-185-169-238.neo.res.rr.com)
16:19.18*** join/#asterisk jpayne (n=jpayne@baconhouse.sackheads.org)
16:19.24file[laptop]I wonder when my food will be here
16:20.19*** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com)
16:21.15nainfile[laptop]: i didn't try hard to compile 1.2.0.  Actually asterisk 1.0.9 and previous stable release also tested but all have same error with my setup.
16:21.50file[laptop]oh food is here
16:22.04Ariel_nain, what is the error your getting and what is your system?
16:22.18file[laptop]Ariel_: file descriptor go kaboom
16:22.35*** join/#asterisk RoyK (n=roy@100.80-203-27.nextgentel.com)
16:22.36Ariel_nain, I see it's fedora Core 3
16:22.49nainAriel_: OS: FC3, asterisk1.0.9, asterisk -oh323-0.6.6 and getting "Unable to write fd 60 (32, Broken Pipe error)"
16:23.14*** join/#asterisk werkenbass (i=werkenba@pc-14-79-86-200.cm.vtr.net)
16:23.15RoyK~seen kpfleming
16:23.17jbotkpfleming <~kpfleming@207.111.174.1> was last seen on IRC in channel #asterisk, 156d 10h 27m 4s ago, saying: 'no, there is no specific plan at this time'.
16:23.21nainthe value of fd xx is random some time it is 43, 60, 5x etc
16:23.26werkenbasshey motherfuckers :D
16:24.33werkenbass:(
16:24.40*** part/#asterisk werkenbass (i=werkenba@pc-14-79-86-200.cm.vtr.net)
16:25.06nainAriel_: Any idea on error ?
16:25.32Ariel_nain, what hard ware do you have in the system?
16:27.43nainAriel_: I am not using any digium or other card, SYstem works as h323 to SIP GW, I took call in h323 and forward to sip provider
16:27.55*** part/#asterisk rob314[laptop] (n=rob314@cpe-65-185-169-238.neo.res.rr.com)
16:28.29nainAriel_: and regarding system specification, it's intel Piv machine
16:30.54*** join/#asterisk BurnedOutGeek (n=mark@66-191-172-049.dhcp.gnvl.sc.charter.com)
16:31.45Ariel_nain, well that should work fine. P4 turn off Hyper threading and what MOH program are you using? mpg123.59r
16:32.05*** join/#asterisk kamathln (i=foobar@220.226.14.205)
16:32.41nainAriel_: MOH program? well i m not using voicemail feature so no mpg123xxxx usage
16:33.12Ariel_nain, is it installed?
16:33.40nainAriel_: How can i check i did default compilation of asterisk ?
16:33.53*** join/#asterisk speck (n=root@ewersbach.net)
16:34.46Ariel_at the prompt #mpg123 will give you info if there is one installed FC3 comes with there own mpg321 which is not compatible with asterisk
16:35.18speckhi everybody i have a problem with dialing in asterisk, when i place a call to sip, it alerts, but only 20 seconds, then there is a message: I IND: TIMEOUT and it hangs up. CAN SO HELP ME??
16:35.56speckI didn't define a timeout in DIAL command
16:35.59Ariel_speck, it alerts like rings the device
16:36.07speckjes..
16:36.07nainAriel_: server*CLI> mpg123
16:36.08nainNo such command 'mpg123' (type 'help' for help)
16:36.18Ariel_ok
16:36.36specknain: try to install mpg123 but not this from debian package
16:37.01clyrradAriel_ did you have any other thought on the pattern matching?
16:37.09Ariel_speck, what are your dial string
16:37.13speckmom
16:37.38nainAriel_: how does mpg123 relates to File descriptor error ? and should i install it and from where if not from debian ?
16:37.40Ariel_clyrrad, not really.
16:37.58clyrradAriel_ maybe it cant be done?  Have you ever done it?
16:38.02Ariel_nain it relates allot to broken pipe
16:38.12speckAriel_: Dial(SIP/${EXTEN}@sip.1und1.de,,m)
16:38.32Ariel_clyrrad, I was looking for a mcro I had with it setup like that. But could not find it.
16:38.43speckwhen i try ...de,1000,m) its the same problem
16:38.47nainAriel_: strange, so from where should i installed it and which one is compatible with asterisk ?
16:38.52Ariel_speck why m
16:38.54clyrradAriel_ so then you had it working?  Means there is a way to do it.....
16:39.04speckwant to have music while dialing
16:39.39Ariel_speck, do you have canreinvite=no in the sip.conf
16:39.47speckyes
16:40.52speckAriel_: it's not only with SIP, also with mISDN
16:41.07SkramXhello all.
16:42.22speckI wonder why it is I IND: TIMEOUT, that normaly means it receives timeout from ISDN Bus
16:44.42speckAnd i have the error message all the time: dev_del_timer: timer(8153f14) not running
16:45.29Ariel_speck, don't know the mISDN device you setup but I know most systems do need a timing device
16:45.45Ariel_do you have the timing set correctly for the device?
16:45.46*** join/#asterisk azrishahril (n=azrishah@60.50.193.179)
16:46.02speckAriel_ where can i set the timing??
16:46.18speckand what is dev_del_timer??
16:47.12DaminWoot..
16:48.28file[laptop]Damin: YOU!
16:49.59JDLSpeedyanyone use teliax
16:50.00JDLSpeedy?
16:50.45RoyKteliax??
16:50.56JDLSpeedyvoip service
16:51.26RoyKah
16:51.30RoyKulaw voip
16:51.37JDLSpeedyallows u to make phone calls
16:51.39JDLSpeedyya
16:51.44RoyKOH
16:51.50RoyKYOU MUST BE KIDDING??????
16:51.59RoyKa voip service allowing THAT?????
16:53.16bkw__haha
16:53.19bkw__smartass
16:53.31JDLSpeedyIm just trying to figure out if they send u mail to ur home address
16:53.52speckmISDN: I IND :TIMEOUT when dialing afer 10 secs of no answer.
16:53.57speckHELP!
16:54.46RoyKwe should have separate channels for mISDN and h.323
16:56.19*** join/#asterisk pr0m (n=pr0methe@24-75-196-70.chvlva.adelphia.net)
16:57.40*** join/#asterisk Dogzz (n=bob@66.148.168.234.nw.nuvox.net)
16:57.49*** join/#asterisk doolph (i=doolph@201.226.146.178)
17:00.00*** join/#asterisk knoppix (n=knoppix@cm52.theta17.maxonline.com.sg)
17:03.18*** join/#asterisk bsd3 (n=bsd@203.134.192.202)
17:04.02*** join/#asterisk pauldy (n=pauldy@c-67-162-255-229.hsd1.tx.comcast.net)
17:07.56*** join/#asterisk jake1932 (n=jake1932@pool-70-16-130-29.phil.east.verizon.net)
17:08.03*** join/#asterisk netsurfer (n=bbjunkie@i-83-67-48-18.freedom2surf.net)
17:08.06netsurferhi ppl
17:08.34netsurferhas anyone here got any docs on app_groupcount ? it's missing on the wiki
17:08.50netsurfer~docs
17:08.51jbot[docs] 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
17:09.06jake1932is there a way to ge the "user" variable from sip.conf inside the dialplan?
17:10.03Ariel_Netgeeks, http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+SetGroup
17:10.10Ariel_it's part of setgroup
17:11.03jake1932i'm trying to get the username automatically for voicemail - but the callerid i want to use for the pstn
17:11.07*** join/#asterisk f_meehan (n=fmeehan@whoami8.cedval.org)
17:11.16netsurferAriel_ - I guess u meant netsurfer ;)
17:11.26netsurferthx.. i'd seen that page.. but wanted more info
17:11.28Ariel_netsurfer, yes
17:11.37Ariel_netsurfer, like?
17:12.07Ariel_jake1932, username automaticly to voicemail?
17:12.42jake1932right - i.e. i press the voicemail key and do a Voicemail Main with the username
17:12.47netsurferAriel_ - i've overlooked the main section of that page heh.. think it covers my questions.. i'll try a reconfig now and see how it goes
17:12.57jake1932but only value i could find was the callerid
17:13.29jake1932callerid != username
17:13.55Ariel_callerID="Name One"<200>
17:14.24jake1932my callerid is 215XXXXXXX
17:14.32Ariel_then you have ${CALLERIDNUM} and ${CALLERIDNAME}
17:14.49jake1932right - but I'm looking for the username (not callerid)
17:15.01jake1932userid is 2000
17:16.32jake1932looks like hack to do it: http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+channels - but wanted to know if there's any variable
17:18.21Qwelljake1932: sip?
17:18.26jake1932yep
17:18.26Qwellmailbox=
17:18.56jake1932i have that - but that seems to be only for MWI
17:19.55Qwelldunno, I hit my voicemail button, and it takes me to my voicemail box
17:19.58Ariel_well looks like your going to have to use the cut unless your username is the same as the exten number.
17:20.33jake1932Ariel_: yep - just wanted to make sure
17:20.36Qwellnowhere in my cisco config does it have my voicemail box, and in extensions.conf, all I have is Voicemailmain
17:21.00jake1932Qwell: but, your callerid is probably your userid?
17:21.05QwellI think, anyhow
17:21.10Qwellno
17:21.22Qwellhmm
17:21.44Qwellthere should be a var
17:22.22jake1932i call it - VoicemailMain(s${CALLERIDNUM})
17:22.27*** part/#asterisk bsd3 (n=bsd@203.134.192.202)
17:23.15Ariel_jake1932, that goes in using your callerIDnumber and by passes the password.
17:23.50jake1932Ariel_: right - which would work if my callerid was a valid mailbox
17:24.19podzaphi, is anybody here running * behind NAT, connected to FWD?
17:24.51*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
17:24.53Ariel_podzap, yes
17:25.11QwellWhy isn't the value of mailbox= in a variable?
17:25.18Qwellseems like that would be incredibly useful
17:25.22jake1932hehe
17:26.25jake1932Qwell: i wish I knew C better - I'd do it
17:26.59jake1932proabaly not that difficult
17:27.04podzapAriel_: do you use a sipura?
17:27.16Ariel_I have do.
17:27.38*** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com)
17:27.49podzapAriel_: i am having a hard time getting connected with * to fwd, behind nat, using iax. it works from sipura to fwd directly, though.
17:27.52fugitivomorning
17:28.02podzapAriel_: could you possibly share some configuration with me?
17:28.45Ariel_podzap, did you go into the fwd web and asked for your account to be iax?
17:28.53*** join/#asterisk mrverbose (i=YQ0flwgF@i53874332.versanet.de)
17:28.53podzapAriel_: yes
17:28.54mrverbosehi
17:29.01Ariel_the samples form there web site works fine.
17:29.08fugitivohello
17:29.12Ariel_fugitivo, buena tarde
17:29.21fugitivobuenas tardes
17:29.22podzapAriel_: i can not get it to work
17:29.32mrverboseis it possible to make asterisk gateway for voip telephones into ordinary isdn phone net?
17:29.36fugitivoyou're right, it's 2:30pm here
17:29.48podzapAriel_: could you share some config files with me?
17:29.56fugitivomrverbose: yes, with the correct hardware
17:29.57Ariel_fugitivo, it's 1:30 pm here
17:30.19mrverbosefugitivo: i have 2 hfcs-usb isdn cards and a voip phone
17:30.51fugitivopodzap: i think there's a problem with fwd
17:30.59nainAriel_: Hi, I installed mpg123 but still problem is same
17:31.12fugitivopodzap: mine isn't registering, some time ago it was working
17:31.27fugitivoAriel_: fwd is working for you?
17:31.28podzapfugitivo: i can get it registered, but the call won't work
17:31.43fugitivohmm, i need to check my account status then
17:31.45podzapfugitivo: it always either gives all circuits busy
17:31.57podzapfrom sipura to fwd directly, it works fine
17:32.16*** join/#asterisk santiago (n=santiago@63.245.86.163)
17:32.18podzapwhy don't more people post working examples of * setups?
17:32.32Qwellpodzap: there are hundreds of examples out there
17:32.45fugitivopodzap: the example that fwd gives you works fine
17:33.05podzapQwell: there is not one complete example for  sipura -> * -> NAT -> FWD
17:33.09Ariel_podzap, http://pastebin.ca/22046
17:33.09Qwellpodzap: if you're able to call fwd from your sipura, without going through asterisk, your account is setup for SIP
17:33.19Ariel_it was yesterday I have not tried it today.
17:33.31podzapQwell: fwd accounts can be setup for multiple protocols
17:33.43Qwellyou sure about that?
17:33.49nainAriel_: Hi, i m back after testing mpg123 with asterisk but still i m getting the same fd error ?
17:33.53podzapQwell: i read it from their website
17:33.58DaPrivateeri am setup for IAX on FWD and I usually can login with X-Lite
17:34.07Qwellmultiple protocols simultaneously?
17:34.07fugitivopodzap: check, that your iax account is not the same as your sip account
17:34.30DaPrivateerbut as far as i can tell FWD is still down
17:34.44podzapQwell: of course not simultaneously
17:35.10*** join/#asterisk drbrown (n=chatzill@63.238.118.90)
17:35.11podzapfugitivo: i signed up for one iax account and one sip only account
17:36.28podzapAriel_: can you give me your entire /etc/asterisk directory, minus your passwords, of course...
17:36.38podzapi am tired of this not working.
17:36.39Ariel_podzap, no
17:36.49fugitivopodzap: that's being lazy
17:37.02podzaplazy, i've been fucking with this thing for 4 days now
17:37.22Ariel_podzap, I have too many custom settings I would have to take at least 2 hours to get it to where you can make sence of it.
17:37.34fugitivopodzap: http://www.freeworlddialup.com/content/view/full/1501/
17:37.37Qwellpodzap: how about you post YOUR configs?
17:37.54podzapQwell: well, OK, it's A@H, so there is about 50 config files.
17:37.58Qwell...
17:38.03*** join/#asterisk audela (n=audela@82-33-115-145.cable.ubr08.azte.blueyonder.co.uk)
17:38.07Qwelltheres your problem
17:38.08podzapQwell: but I'd be glad to do it, if you would care to analyze them.
17:38.09fugitivonice
17:38.10fugitivolol
17:38.25podzapI am trying to use A@H, what's wrong with that?
17:38.36podzapis it a piece of shit, or what?
17:38.55Qwellare you using the GUIs to make the changes to the configs?
17:38.59fugitivono
17:38.59Ariel_podzap, no it's not it works fine on it
17:39.04podzapI am using the GUI
17:39.07Ariel_I have it working on a@H as well
17:39.10fugitivoit's just difficult to help somebody that uses aah
17:39.29Ariel_fugitivo, no it's not.
17:39.34podzapfugitivo: OK, I could use the normal * if I had a set of config files to start with then.
17:39.43*** join/#asterisk zoo (i=nobody@ip-172-16.travedsl.de)
17:39.46Qwell* comes with sample configs
17:40.00Ariel_podzap, on the web site for asterisk@home there is a doc that has how to setup fwd via iax setting for amp
17:40.13podzapI followed the asteriskathome handbook to the letter to setup with fwd, and it DOES NOT WORK
17:40.17podzap:-)
17:40.46Ariel_podzap, the iax server of there is an experiment and they don't always have it working correctly
17:41.03podzapAriel_: is yours working now, yes or no?
17:41.36Ariel_podzap, I have had it registered with them for over 8 months it could be that new settings are not up there yet for yours.
17:42.03podzapAriel_: I am willing to bet that this is a * configuration problem.
17:42.10fugitivomine is not working
17:43.07podzapi get the idea that people are trying to make a lot of money with * "consulting"
17:43.09Ariel_podzap, what is your number? let me call it to see what I get
17:43.19Ariel_podzap, really
17:43.45podzapAriel_: you mean the one that is direct sipura -> fwd, or the * number?
17:44.11Ariel_podzap, the one to asterisk. sipura is an different number?
17:44.34fugitivoservice numbers are not answering for me
17:44.37sivanais it safe to use the lastest SpanDSP version?
17:44.47sivanaie: pre20
17:44.51podzapi have two lines on the sipura, one to * and one directly to fwd. it takes me a minute to enable the * again.
17:44.55Ariel_sivana, I have pre18
17:45.12Ariel_podzap, never mind it's not working for other then 800 numbers for mine.
17:45.26Ariel_They must be having issue's with the iax server
17:45.34fugitivoAriel_: can you receive calls?
17:45.52Ariel_don't know 65342
17:46.50fugitivonothing?
17:46.59Ariel_I guess there down
17:47.03fugitivoyep
17:47.16Ariel_podzap, it's not your settings it's FWD
17:47.37podzapAriel_: do you know somebody else i could test with, then?
17:48.14Ariel_for iax free account or for sip?
17:48.34podzapeither one i could get working; I am behind NAT.
17:48.38*** join/#asterisk huslage (n=used@c-24-22-125-179.hsd1.or.comcast.net)
17:48.38fugitivopodzap: stanaphone gives you a free new york number, but it's sip
17:48.39Ariel_podzap, you can get a free sip account from stanaphone and sipphone
17:49.18podzapAriel_: well, my one of my sipura lines does sip to fwd if i enable the nat settings and outbound proxy.
17:49.35podzapcan you enable nat and outbound proxy with * and fwd?
17:49.40doolphwhat's the latest zaptel version
17:49.50Ariel_sip behind nat you need to put in the sip.conf externip=XXX.XXX.XXX.XXX localnet=XXX.XXX.XXX.XXX/255.255.255.0
17:50.23podzapi can't edit sip.conf with asteriskathome, or I will go to hell and die
17:50.31Ariel_doolph, I am using the one from cvs head for zaptel and then putting in the echo cansule from KB1 works great with stable
17:50.34podzapaccording to some in this channel...
17:50.55Ariel_podzap, yes you can and it will not die
17:51.01doolphzaptel.c:4083: error: dereferencing pointer to incomplete type
17:51.36Ariel_A@H only changes with the gui sip_additional.conf
17:51.55podzapcan I just remove all those A@H files, and just put some normal * files there and will it still work?
17:52.03doolphwhy my debian doesnt want to compile zaptel
17:52.04podzapi am tired of this gui thing
17:52.11Ariel_podzap, yes
17:52.13doolphit's because I dont have zaptel hardware?
17:52.26podzapAriel_: where can i get a working set of files?
17:52.30Ariel_did you comment out the #ztdummy
17:52.53*** join/#asterisk omarc55 (n=omarc55@200.63.200.121)
17:52.54Ariel_podzap, just copy the ones out of there and then go to /usr/src/asterisk and do make samples
17:53.21podzapAriel_: OK
17:53.35doolphnop
17:53.48doolphno
17:53.49doolphoh yeah
17:53.53podzapthat A@H shit is waaaaay to complicated for me. i want like 4 simple files, that's it.
17:54.36Ariel_wow a gui is too complicated hummm sounds like either a dos person or a lunix command prompt guy to me.
17:54.44omarc55hi all, does anybody know how to use iptables? I have 2 internet connections and I want to pass IAX2 calls through one connection and data through the rest, does anybody know how to do this?
17:54.51podzapAriel_: yes, that's me.
17:55.34Ariel_iax2 is udp port 4569
17:55.45fugitivoomarc55: there're a lot of examples on the net
17:56.11omarc55yeah, I know, I've tried a few but can't get it to work, and I am using udp port 4569
17:56.32podzapAriel_: that A@H, if you look at e.g. iax.conf, the include iax_additional.conf is commented out, but that is what the GUI writes to. How the hell does * use it, when it's commented out?
17:56.36fugitivoiptables -A interface0_in -p udp -m udp --dport 4569 -j ACCEPT
17:56.51fugitivoiptables -A interface0_in -p udp -m udp --dport 5036 -j ACCEPT
17:56.56Ariel_the # is not a commenting it out
17:57.05podzapAHA
17:57.13podzapwhat is it, then?
17:57.24Ariel_#include =>/external/file
17:57.31Ariel_the ; will comment it out
17:57.33podzaplike a C include line
17:57.47omarc55fugitivo: isn't that for port forwarding? I am trying to use 1 internet connection just for IAX2
17:58.03fugitivoomarc55: no, it's not
17:58.27fugitivoomarc55: replace interface0_in for whatever your rule is called
17:58.50podzapAriel_: ok, i did `make samples'. I got 41 new files into /etc/asterisk. How many of those can I remove? What is the bare minimum which I need to keep?
17:59.10Ariel_~docs
17:59.11jbotwell, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
17:59.53omarc55fugitivo: not sure what my rule is called, I have the whole bunch of rules I am trying to get to work, I can put it in a paste bin...
18:00.14Ariel_podzap, don't know what you want to work with. But if you don't want to use mgcp.conf then noload them in the modules.conf file
18:00.31Ariel_podzap, there are more then 4 files you will need to use.
18:01.03podzapAriel_: I only want to send and recieve calls to fwd
18:01.09podzapfrom one phone
18:01.44omarc55http://www.ninethsense.com/praveen/pastebin/index.php?p=245
18:01.55podzapi want the smallest configuration which I can get.
18:02.16*** join/#asterisk pauldy (n=pauldy@c-67-162-255-229.hsd1.tx.comcast.net)
18:02.42Ariel_podzap, then why not just use the sipura then
18:03.39podzapAriel_: starting to sound like a good idea
18:04.29podzapi mean how in the hell is a new person supposed to learn how to setup a software which has 41 configuration files?
18:04.47zip_rtfm
18:04.49zip_like the rest
18:04.51Ariel_podzap, by using it and reading
18:05.18*** join/#asterisk Roldyx (n=Roldyx3@201.255.105.250)
18:05.20Roldyxhi
18:05.26Ariel_funny thing when I started with asterisk over 3 years ago there was not wiki. or at least it was just starting. It was allot harder then.
18:05.37Roldyxbut no rule 't' in context 'default'
18:05.55Roldyx""but no rule 't' in context 'default'"" <-- what means this?
18:06.13fugitivoRoldyx:  you didn't set a t rule in your context default
18:06.18Ariel_Roldyx, it means there is no time out rule in your default context
18:06.35Roldyxfugitivo, !!
18:06.55Roldyxfugitivo, en español
18:07.00[Airwolf]podzap, start slowly. First make one thing work, then another and so one. After a few things, look at how it can be done better and then move on.
18:07.18Hmmhesayswham bam another successful asterisk install
18:07.26fugitivoRoldyx: exten => t,1,Hangup() for example
18:07.34Hmmhesaysand i've paid for my drinking tonight
18:07.37Hmmhesaysand gas to the lakes
18:07.49[Airwolf]Hmmhesays, did that last week
18:07.52[Airwolf]:P
18:07.57fugitivoRoldyx: es la regla que buscara asterisk cuando se produzca un timeout
18:08.30*** join/#asterisk tainted_ (n=identd@adsl-71-129-42-166.dsl.irvnca.pacbell.net)
18:08.33Roldyxfugitivo, esa regla la debo poner en extensions?
18:08.42fugitivoRoldyx: si
18:08.46tainted_anyone use 'screen' to keep asterisk console up
18:09.06Ariel_tainted_, why do you need to do that use safe_asterisk
18:09.07faa_tainted_ whaT?
18:09.09Roldyxfugitivo, una sola linea es?
18:09.18tainted_what is safe_asterisk
18:09.22fugitivoRoldyx: podes utilizar la cantidad que quieras
18:09.28tainted_is it in 1.0.7?
18:09.30tainted_i'm using .7
18:09.36fugitivoRoldyx: t,1 t,2 t,3 etc
18:09.37Roldyxfugitivo, pero si pongo esa linea ya lo soluciono?
18:09.41Ariel_tainted_, ahh stop asterisk then start it with safe_asterisk then connect to it via asterisk -r
18:09.55fugitivoRoldyx: si, pero usa una linea que sirva a tu proposito
18:10.23Roldyxfugitivo, el tema es que aso le aparece a mi amigo cuando yo lo llamo
18:10.33fugitivoRoldyx: tiene un ivr tu amigo?
18:10.35Roldyxfugitivo, a mi no me pasa
18:10.52Roldyxfugitivo, solo instalamos el asterisk
18:11.22*** join/#asterisk roulduke_ (i=kw374vri@p508D0ADF.dip0.t-ipconnect.de)
18:11.32fugitivoRoldyx: eso no afecta en nada el funcionamiento, pero es mejor si lo tiene
18:11.35tainted_Ariel_ oh wow
18:11.49Roldyxfugitivo, por que podria dar timeout?
18:11.55*** join/#asterisk Legend` (n=legend@24.244.142.133)
18:12.13Ariel_tainted_, what OS do you have?
18:12.22fugitivoRoldyx: por ejemplo si tiene un IVR, y el que hace el llamado no ingreso ningun DTMF, el asterisk salta al timeout para saber que tiene que hacer
18:12.50*** part/#asterisk zoo (i=nobody@ip-172-16.travedsl.de)
18:13.24omarc55fugitivo: were you able to make sense of what I sent?
18:13.32Roldyxfugitivo, IVR?
18:13.42Roldyxme desconciertas
18:13.48fugitivoRoldyx: interactive voice response, un menu interactivo
18:14.06tainted_Ariel_ linux
18:14.15Roldyxfugitivo, y como le digo que lo saque?
18:14.16fugitivoRoldyx: cuando llamas y la operadora te dice, presione 1 para soporte, 2 para tal cosa, etfc
18:14.22Roldyxaha
18:14.25fugitivoRoldyx: eso es un IVR
18:14.26Roldyxentendi de primera
18:14.31Ariel_tainted_, yes which distro ... OS
18:14.42Roldyxfugitivo, como puedo sacarlo?
18:14.45fugitivoRoldyx: entonces si el que llama, no ingresa ningun numero en un tiempo, asterisk va a saltar a la regla de timeout
18:14.47Roldyxsaco el modulo?
18:14.54fugitivosacar que?
18:15.02Roldyxpara que no conteste
18:15.19Roldyxo sea para que atienda directamente el asterisk
18:15.19fugitivoel ivr?
18:15.22Roldyxsip
18:15.29tainted_Ariel_ fedora
18:15.44Roldyxfugitivo, mi amigo no quiere usar IVR
18:15.47fugitivobusca exten => s,1   eso es lo que hace atender al asterisk reemplazalo por lo que quieras
18:16.02Roldyxmira
18:16.17Ariel_tainted_, then you can go into the /usr/src/asterisk and do ... make config that way when you reboot it will load it automaticlly for you.
18:16.35fugitivoRoldyx: podes poner exten => s,1,Dial(SIP/1234,30,t) para que cuando el asterisk atienda, llame al interno 1234
18:16.48doolph.........
18:16.52Roldyxfugitivo, excelente!
18:16.57Roldyxfugitivo, ya entendi
18:17.49Roldyx:D
18:17.54Roldyxgracias again
18:17.57Ariel_fugitivo, your a better man for that exchange then I.
18:18.04fugitivo:)
18:18.12Roldyxfugitivo, por que usan SIP en lugar de IAX?
18:18.24tainted_Ariel_ does it require sendmail for NOTIFY=email@address.com to work?
18:18.35Ariel_tainted_, yes
18:19.04fugitivoRoldyx: depende de lo que utilices, si utilizas telefonos IAX, entonces seria Dial(IAX2/1234,40,t)
18:19.33Roldyxfugitivo, el tema es que nosotros no usamos telefonos
18:19.40Roldyxfugitivo, NO TENEMOS PLACAS
18:19.48Roldyxsorry por las mayusculas
18:19.50tainted_Ariel_ so how do u get asterisk to run on startup?
18:20.12Qwelltainted_: there are init scripts in the contrib/ dir
18:20.16Ariel_did you read what I posted to you earler?
18:20.36tainted_yea make config
18:20.38Ariel_tainted_, just go to the /usr/src/asterisk directory then make config
18:20.39fugitivoRoldyx: utilizan softphones?
18:20.44Roldyxnop
18:20.49Roldyxfugitivo, CLI
18:21.06fugitivook, entonces reemplaza el comando por el que desees usar
18:21.11fugitivoyo te di un ejemplo
18:21.12Roldyxfugitivo, no encuentro un softphone como la gente
18:21.16Ariel_tainted_, if you have RH/CentOS/Fedora it will make the script for you that on boot up will load it all for.
18:21.29fugitivoRoldyx: para windows o linux?
18:21.34Roldyxlinux
18:21.47Ariel_oh boy using asterisk as a soft phone....
18:21.53fugitivoRoldyx: xlite
18:22.34Roldyxfugitivo, qt o gtk?
18:22.43Ariel_strange they were dialing via iax to each others box... neat..
18:22.44tainted_Ariel_ what script would that be?
18:22.51fugitivoRoldyx: no lo utilice, no es opensource, pero si gratuito
18:22.54tainted_Ariel_ the filename.. maybe i already have it
18:22.58fugitivoRoldyx: creo que es gtk
18:23.28Ariel_tainted_, even if you already have it. It will not hurt just in case.
18:23.55tainted_found it.. /etc/rc.d/init.d/asterisk
18:24.11Ariel_yep
18:27.11tainted_mmm.. i have 15 asterisk processes now
18:27.16tainted_after i rebooted
18:27.16tainted_lol
18:28.24razuanyone have experience with linksys PAP2 echo problem ?
18:28.35fugitivorazu: i don't have echo problem with the pap2
18:29.24razuhmm
18:29.35fugitivorazu: how do you know that it's a pap2 problem?
18:29.50razucause everybody with pap2 is wining :P
18:30.02Qwellwining?
18:30.03bkw__tainted_, thats 15 threads boi
18:30.12shido6erf?
18:30.20shido6my pap2 sounds better than my 7960
18:30.29fugitivomine works great
18:30.39doolphwhere did you get that pap2
18:30.47doolpheverywhere is solt out
18:30.56shido6I have 9k in stock
18:30.57razustrange ...
18:31.02fugitivoshido6: how much?
18:31.03doolphreally
18:31.07doolphI want 2
18:31.10shido6$70
18:31.15doolphbah
18:31.17doolphtoo expensive
18:31.24fugitivorazu: how are you using the pap2? zap calls? or 100% sip calls?
18:31.26Qwellshido6: Do I want to know why you have 9k pap2's?
18:31.27PakiPenguinshido6, i have a pap2 , how do i put in a stun in there?
18:31.29fugitivoshido6: expensive
18:31.29shido6then order from teh sold out
18:31.38doolph2100 for $64.95
18:31.56fugitivoshido6: how much for 100 ?
18:31.59Qwell$70 for a pap2-na is pretty damn good
18:32.01razui'm testing here right now and pap2 doesn't have an echo really ... but the other number where i called gets an echo :S
18:32.02Ariel_doolph, get the 2002 instead
18:32.11razufugitivo : 100% sip calls
18:32.26fugitivorazu: what codec?
18:32.26doolphAriel_ what's the difference 2000 with 2002
18:32.28shido6go to the admin interface, PakiPenguin
18:32.46doolph2002 is $5 more expensive
18:32.54PakiPenguinokay , am in admin interface
18:32.55Ariel_doolph, the 2002 has a updated dps and can do 2 g729 calls. the 2100 has a wan port that sucks
18:33.08doolphohhh
18:33.10razufugitivo : G711a
18:33.12doolphbut i dont care
18:33.15doolphi will use g723
18:33.16doolphheh
18:33.23fugitivorazu: is your bandwitdh ok?
18:33.23Ariel_doolph, same thing
18:33.29razufugitivo : yep
18:33.40Ariel_2002 will do two not one. the 2000 only can do one
18:33.47doolphthat's why
18:33.55doolph5 more expensive
18:34.22fugitivorazu: try a softphone and check if there's echo
18:34.36razuok
18:35.27fugitivoanybody downloaded the free linspire? :)
18:37.35mog_homedebian...?
18:37.53Qwellfugitivo: I try to only use real Linux
18:39.38fugitivoi was thinking on my mother for linspire
18:39.53mog_homeubuntu
18:41.27Ariel_I have not seen linspire but for desktops I like ubuntu for gnome or Mepis for KDE setup.
18:41.39*** join/#asterisk Nukemizer (n=Nuke@160.7.249.15)
18:42.12doolphi hate X
18:42.16doolphI prefere windows
18:42.17doolphheh
18:42.19[Airwolf]ubuntu rules for desktop
18:42.35fugitivowhat's different about ubuntu?
18:42.52tainted_it's spelled differently from debian
18:42.59[Airwolf]gehe
18:43.28Ariel_actually Mepis and ubuntu are both based on Debian
18:43.45fugitivoi can make my gentoo a great desktop too
18:43.55Ariel_fugitivo, then do that
18:43.57Roldyxfugitivo,
18:44.23fugitivobut linspire seems to be nice for people that doesn't use linux
18:44.24Ariel_fugitivo, you made a new friend.
18:44.44tainted_why are there 15 asterisk threads
18:45.32bkw__different things require threads
18:45.54doolphi am getting error compiling asterisk 1.2 duh
18:46.11*** join/#asterisk drbrown (n=chatzill@63.238.118.90)
18:46.22tainted_etc/init.d/asterisk generates 15 threads
18:46.27tainted_just wanted to make sure that is kosher
18:46.33bkw__yes
18:46.41bkw__thats normal when your ps output shows each thread
18:46.48bkw__load the box up with calls
18:46.52bkw__you'll see even more
18:47.08bkw__I only see one thread on mine
18:47.09bkw__:P
18:47.15robin73hi, there
18:47.15*** join/#asterisk mhnoyes (n=mhnoyes@user-38lc0dh.dialup.mindspring.com)
18:47.16doolphyeah me too
18:47.20doolphlol
18:47.22tainted_why does it only show one when i 'asterisk -vvvvgc' from command prompt
18:47.23bkw__because we use NPTL threads
18:47.37bkw__tainted_, it should still show them all
18:47.50tainted_only one!
18:47.55tainted_weird
18:47.58doolphbkw_
18:48.16robin73i'am working on a web admin interface for asterisk 1.2b. is there already good opensource web admin UI ?
18:48.34bkw__robin73, If I had 10 bucks for everytime I have heard someone say that
18:48.36bkw__I would be RICH
18:48.45doolphheh
18:48.46Ariel_robin73, I like amp
18:48.55tainted_lol
18:48.57*** join/#asterisk iswm (i=iswm@unaffiliated/iswm)
18:49.01doolphpbx_dundi.c:31:18: zlib.h: No such file or directory
18:49.07robin73yes amp is good, but not complete
18:49.14doolphasterisk:/usr/src/asterisk-1.2.0-beta1# locate zlib.h
18:49.14bkw__doolph, install zlib-devel
18:49.15tainted_i've got some pretty sweet billing written in .NET
18:49.16robin73i'am using asterisk@home
18:49.23bkw__you needs zlib headers boi
18:49.24doolph/usr/include/linux/zlib.h
18:49.29bkw__not the same
18:49.35doolphisnt that one?
18:49.38bkw__no
18:49.38tainted_getting asterisk to talk to MS SQL was a bitch
18:49.41bkw__thats in the kerenl boi
18:49.48bkw__tainted_, no its not.. its really easy
18:49.53robin73any other suggestion for a good existing WEN UI ?
18:50.08bkw__tainted_, but I was also the first person to bring ODBC to asterisk :P
18:50.10tainted_bkw_ through AGI i mean
18:50.17bkw__oh thats simple too
18:50.27bkw__I can do that in three lines of perl
18:50.28doolphthnx bkw_
18:50.31tainted_i've resorted to freetds & DBD::Sybase
18:50.46*** join/#asterisk Nukemizer (n=Nuke@160.7.249.15)
18:50.47bkw__yes but you must use freetds 0.62 I think
18:51.03bkw__or is it 0.63 but they made chnages it no longer works correctly
18:51.05tainted_freetds-0.63/
18:51.12bkw__any higher version will not work
18:51.14tainted_that's what im using
18:51.17doolphbkw_ there's anyway to see a calltimer in a call
18:51.18*** join/#asterisk santiago (n=santiago@63.245.86.163)
18:51.41tainted_bkw_ where did u stick $SYBASE env var
18:51.41bkw__I want realtime cdr events every X seconds
18:51.43PakiPenguintainted_, is your .net app opensource?
18:51.58doolphbkw_ does it exist?
18:52.01bkw__tainted_, its all wrapped in this perl module we wrote
18:52.14file[laptop]it's beautiful
18:52.16tainted_see.. not even close to three lines
18:52.19tainted_heh
18:52.30bkw__tainted_, yes in my perl scripts I do use SQLSERVER;
18:52.34tainted_i got pissed at mysql and kicked that bitch to the curb
18:52.49bkw__my $db = init_isp();
18:52.55tainted_lol
18:52.58doolphi dont need mpg123 for asterisk 1.2 eh
18:53.00tainted_cheater
18:53.03bkw__$db->do("blah");
18:53.08bkw__done
18:53.10bkw__NEXT!!!
18:53.11bkw__haha
18:53.21bkw__or
18:53.24tainted_sigh
18:53.32bkw__my $sql = "some sql query";
18:53.34doolphbkw_ how you can see the calltime in a call
18:53.36bkw__foreach my $obj (@{$db->sql2allhash($sql)}) { do something }
18:53.36tainted_i tore my hair out getting that shit to work
18:53.43mog_homebkw_
18:53.44faa_do anyone use cisco call manager? or cisco call manager express?
18:53.47mog_homei will pay
18:53.48mog_homeyou
18:53.53fugitivofaa_: #cisco
18:53.54mog_hometo go to mongolia
18:53.54bkw__for?
18:53.55mog_homefind a monk
18:53.58mog_homeand learn
18:54.01mog_homePATIENCE
18:54.02mog_home^_^
18:54.05tainted_lol
18:54.08bkw__why?
18:54.12mog_homeno reason
18:54.12PakiPenguinlol
18:54.12file[laptop]patience is overrated
18:54.14mog_home^_^
18:54.19bkw__don't make me bitch smack you
18:54.24tainted_patience is for the weak
18:54.32tainted_we need more progress
18:54.33mog_homepatience is for the strong
18:54.40mog_homelike tolerance
18:54.48mog_homethe weak can be intolerant and impatient
18:55.42tainted_patience is waiting for spilled water to evaporate rather than wiping it up
18:55.49mog_homelol
18:55.51mog_homeno
18:56.00mog_homepatience is when someone spills toxic waste
18:56.05Nivexno I believe that is called laziness
18:56.06Qwelltainted_: no, thats just lazy :p
18:56.08mog_homeand you call the guy to get it lcened
18:56.13mog_homecleaned*
18:56.21mog_homeand you wait and make sure no one steps in it
18:56.37Qwellpatience is waiting for the cable guy to come "sometime between 9 and 4"
18:56.43Qwelland then punching him in the face when he gets there
18:57.04mog_homelol
18:57.05tainted_punching is barbaric
18:57.11Qwellokay, stabbing
18:57.21tainted_beating him down with tire iron is more civilized
18:57.24sivanabkw__: it seems the sangoma cards can't handle data very well
18:57.39bkw__sivana, what do you mean?
18:57.41bkw__the TDM boards?
18:57.47sivanaI went back to the TE405 and I was able to fax 2 pages
18:57.53sivanawithout hangup
18:57.55sivanaya
18:58.00bkw__its your clocking
18:58.10mog_homedigium is awesome ^_^
18:58.11bkw__we have a sangoma board for testing works great
18:58.15sivanawell... the clocking is better with the 405 then
18:58.21mog_homebut its all about clock
18:58.28bkw__sivana, chances are you had the clock set wrong
18:58.33mog_homeif clock is bad there will be no faxing
18:58.35bkw__its not trivial to set that
18:58.40tainted_shit
18:58.41sivanabkw__: you're faxing with the sangoma?
18:58.43bkw__yes
18:58.46tainted_forgot to register for astricon
18:58.51doolphNow I am getting compiling zaptel
18:58.53mog_homeme thinks digium card is better
18:58.56tainted_bkw_ what are your rates for writing customer stuff for asterisk
18:58.57sivanaeven the dial up pool was all fubared with the sangoma
18:58.59bkw__we aren't even using zaptel with ours
18:59.00mog_homebut they are very similar
18:59.01tainted_custom
18:59.11bkw__its not even setup on asterisk
18:59.12bkw__:P
18:59.15sivanabkw__: fax machine -> FXS -> * -> PRI   ?
18:59.27doolphwhat do i need to install zaptel
18:59.33mog_homeme needs money for making asterisk code
18:59.35bkw__sivana, you were using a TDM board?
18:59.43bkw__TDM400P?
18:59.46sivanaFXS = CAC Channel Bank
18:59.52sivanawith the A104
19:00.02bkw__i bet you the clock was wrong
19:00.03*** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com)
19:00.08mog_homedoubt it
19:00.14bkw__no I bet it was
19:00.16mog_homeas he didnt change it when he got card
19:00.19mog_homeif he did
19:00.23mog_homethen he should have known
19:00.26sivanapossibly... span 1 was getting clock from PRI and the other spans were getting it from 1
19:00.26bkw__no in the wanpipe driver you ninny
19:00.28bkw__it was wrong
19:00.42bkw__like I said that is an easy one to get wrong in there
19:00.49*** join/#asterisk pr0m (n=pr0methe@24-75-196-70.chvlva.adelphia.net)
19:00.54sivanabkw__: I have the span 2 set the clock reference to 1
19:01.12bkw__chances are you didn't need it like that
19:01.14mog_hometaking timing
19:01.21sivanaI dunno... didn't friggin work right though
19:01.42mog_homewell thats awesome
19:01.45sivanamy customers did notice increase in echo since going back to TE405 though
19:02.02mog_homewhats your echocan set at sivana
19:02.07sivanaKB1 now
19:02.15mog_homein zapata
19:02.17tainted_bkw_ so u dumped mysql as well?
19:02.20mog_homeechocan=?
19:02.22sivanayes
19:02.26mog_homebump it
19:02.26sivanaechocan = yes
19:02.27mog_homeup
19:02.29mog_home128
19:02.30Hmmhesaysanyone know how to set no callerid for voipjet?
19:02.32mog_homeor 256
19:02.44mog_homeor get one of those new sexy 406s ^_^
19:02.54sivanaI'll try next week... I just changed the type to KB1, so maybe that'll be better
19:02.55mog_homebut bumping value up will fix it i bet
19:03.36sivanaI need to wait until Monday now for the business volume to pick back up:)
19:04.19sivanabkw__: it certainly sounds like timing, like after 6 mins, things would go haywire
19:04.36bkw__sivana, ya
19:04.41sivanabkw__: but I don't know enought that shit... I'm programmer not an engineer dammit :)
19:04.58doolphdo I need kernel header to install zaptel?
19:05.04JerJeryes
19:05.17mog_homeor complete kernel source ^_^
19:05.36sivanaI haven't tried faxing > 2 pages yet though, so maybe it's not 100% solved
19:05.39*** join/#asterisk tessier (n=treed@wsip-68-224-172-77.sd.sd.cox.net)
19:06.18sivanabut with the sangoma card, I was lucky to get 1 page out before comm error
19:06.28doolphmmm
19:06.33doolphthat might be the problem
19:06.33doolphthen
19:07.13sivanacould have just been configuration issue though with the sangoma... I sent them an email about it and got no reply
19:07.17doolphmmm same errors
19:07.28tainted_anyone going to astricon?
19:07.45DaminYeah.. I'm supposed to be doing a talk on something, but I'm not sure what yet..
19:07.57doolph/usr/include/linux/stat.h:68: error: storage size of `atime' isn't known
19:08.00doolph/usr/include/linux/stat.h:70: error: storage size of `ctime' isn't known
19:08.02mog_homeand speaking i hope
19:08.03tainted_Damin are you serious lol
19:08.09doolph/usr/include/linux/stat.h:69: error: storage size of `mtime' isn't known
19:08.15sivanaanyone here from Manx yet?
19:08.16doolphany idea
19:08.24sivanas/here/hear
19:08.26tainted_the speaker list doesn't seem too strong right now
19:08.39sivana~seen manxpower
19:08.43jbotmanxpower <i=eric@1Cust2920.an7.dfw28.da.uu.net> was last seen on IRC in channel #asterisk, 1d 10h 19m 50s ago, saying: 'Um, actually responsetimeout might be more what you are looking for.'.
19:08.43mog_homeyeah he needs to update it
19:08.47bkw__sivana, whats your email?
19:09.01tainted_mog_home who is organizing
19:09.15sivanabkw__: richard@aspworld.com
19:09.27*** join/#asterisk L|NUX (n=linux@202.5.145.14)
19:09.33mog_homeole and sokol
19:10.02Damintainted_: Yeah.. I've got a couple of different topics into Olle and Steve, but I probably won't know which one they'll pick until I get there.. :)
19:10.04Ariel_mog_home, an 5ess ATT switch type is a normal T1 for e&m  not a pri system is it?
19:10.19mog_homeyeah
19:10.23tainted_Allison Smith: the voice of asterisk is a speaker!?!?!?!
19:10.27mog_homeyeah
19:10.31mog_homeshe comes to all of them
19:10.50tainted_Damin which topics?
19:12.36Damintainted_: Either intro to asterisk tutorial or Deploying Asterisk for the SOHO market..
19:13.02Damintainted_: Or I could just get up there and spew profanity like Zoa did in Atlanta.. ;)
19:13.17mog_homei think i am gonna be part of a q and a session and hopefully talk about jabber and asterisk
19:13.28tainted_Damin what was he swearing about
19:15.02bkw__Damin, the soho one is great I think
19:15.19bkw__mog_home, jabber and asterisk.. thats pretty funny
19:15.42bkw__mog_home, what does res_jabber do now?
19:15.43file[laptop]\
19:15.48file[laptop]very funny
19:16.10mog_homeres_jabber does a lot
19:16.27Damintainted_: Don't know.. I couldn't understand a whole lot of what he said other than "F**K" alot. ;)
19:16.44mog_homeill release it when it makes me smile
19:16.48mog_homeand doesnt crash my box
19:17.02file[laptop]that would be a good thing
19:17.06file[laptop]crashes are usually frowned upon
19:17.09tainted_astricon 2004 was pretty strong
19:17.17Daminfile[laptop]: Unless you are a crash test dummy..
19:17.21mog_homeheh
19:17.43Damintainted_: I'm looking forward to Anaheim..
19:17.45file[laptop]nope, I'm not a crash test dummy
19:17.49doolphanyone can help me out with zaptel compiling
19:17.49tainted_i'm going just to see the faces behind the characters
19:17.50tainted_hahaha
19:17.51bkw__mog_home, I could accomplish the same thing via AGI using Net::Jabber
19:17.57doolphI am using debian
19:18.02tainted_bkw_ well put
19:18.05mog_homeyou could crash it?
19:18.11bkw__no make it not crash
19:18.16mog_homeand /me haates agi
19:18.16bkw__via agi and net::jabber
19:18.21bkw__agi has its place
19:18.23bkw__even I hate it
19:18.25mog_homewell it doesnt seem to go
19:18.26bkw__but it does have its place
19:18.33file[laptop]omg this movie is evil
19:18.37file[laptop]if she dies I swear I will kill someone
19:18.41tainted_it'd be sweet to stdout < NET::jabber
19:18.42tainted_haha
19:18.42*** part/#asterisk santiago (n=santiago@63.245.86.163)
19:18.44bkw__file what are you watching?
19:18.52file[laptop]City of Angels
19:18.56file[laptop]on TBS
19:19.08bkw__everyone stay away from file
19:19.14file[laptop]she's dead.
19:19.18mog_homeagi is not the place for something this important...
19:19.25mog_homeand this big i think
19:19.31mog_homeor as big as i hope for it to be
19:19.31glm2kfile[laptop]: i was just about to tell you that >:)
19:19.31bkw__mog_home, yes it is.. its all in how you impliment it
19:19.37tainted_mog javver is mission critical?
19:19.37file[laptop]bah
19:19.38bkw__you can even do it via the manager
19:19.40doolphhello?
19:19.46mog_homelol
19:19.47bkw__creating a custom manager event from the dialplan.. with an external jabber proxy
19:19.50mog_homeyeah you could
19:19.55mog_homeits how the other people are doing it
19:19.59bkw__thats more safe than trying to stick it inside of asterisk
19:20.11bkw__just because you can doesn't mean you should
19:20.14bkw__unless you do it correctly
19:20.23tainted_grasshoppa
19:20.24mog_homewell youll see
19:20.26mog_homeone day
19:20.41mog_homeunless you go aeferion on us bkw
19:21.03bkw__and i'm shocked you can even fucking spell that
19:21.05bkw__I never can
19:21.31*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
19:21.35mog_homewhich is why no one ever gets to the site...
19:21.45bkw__haha
19:21.49bkw__I just said that today also
19:21.57mog_homethey could use a better name
19:22.43file[laptop]I'm angry now
19:22.49file[laptop]what a mean mean ending
19:23.08bkw__file thats why I haven't watched that movie since
19:23.47wasimor nextgen
19:25.24doolph/usr/include/linux/time.h:198: error: dereferencing pointer to incomplete type
19:26.14file[laptop]I am the very model of a modern major general
19:26.18doolphis that kernel problem or what
19:27.36bkw__doolph, you seem to be having all kinds of problems
19:27.42bkw__what distro are you using?
19:28.38file[laptop]dammit this movie has made me miserable now
19:30.33Daminfile[laptop]: That movie really sucks...
19:31.03doolphdebian
19:31.21doolphLinux asterisk 2.4.27-2-686 #1 Mon May 16 17:03:22 JST 2005 i686 GNU/Linux
19:31.36bkw__oh lord help ya
19:31.38bkw__go get a real distro
19:31.40bkw__:P
19:32.11zedkatufdoolph: is that a pure debian distro?
19:32.15doolphyes
19:32.45doolphI got asterisk compile resolved already
19:32.48doolphnow the problem is zaptel
19:33.03doolphi think that it's kernel heador or default kernel problem i dunno
19:33.19doolphusr/include/asm/mpspec.h:8: error: `MAX_MP_BUSSES' undeclared here (not in a function)
19:33.25*** join/#asterisk tzafrir_laptop (n=tzafrir@CBL217-132-129-167.bb.netvision.net.il)
19:35.31*** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net)
19:35.38*** join/#asterisk dudes (n=dudes@12-215-34-84.client.mchsi.com)
19:36.37*** join/#asterisk Borgon (i=Borgon@70-100-53-251.dsl1.tbr.ga.frontiernet.net)
19:36.57BorgonAny of you use a vps/vds to run asterisk?
19:37.09*** join/#asterisk SexyKen (n=ksandell@c-24-5-129-114.hsd1.ca.comcast.net)
19:38.28Hmmhesaysyeah that brian guy on the mailing list is kind of a jerk
19:39.01file[laptop]Hmmhesays: just a little
19:39.30Hmmhesayshey file howsit
19:39.35file[laptop]peachy, you?
19:39.48Hmmhesayst'ain't bad, just finished an install, I think i'm going to take my motorcycle down to the lakes now
19:39.55file[laptop]that's crazy talk
19:39.59file[laptop]you should be a geek and stay on IRC
19:40.02Hmmhesayshaha
19:40.13file[laptop]you know you wannnna
19:40.14HmmhesaysI am a geek, maybe not so much as you
19:40.18Hmmhesaysfanta?
19:40.24file[laptop]yes!
19:42.13*** join/#asterisk Dybdahl (n=Lars@oelsted.dybdahl.dk)
19:42.46bkw__Hmmhesays, you evil bastard
19:42.58blitzragebkw__: you know js right?
19:43.00file[laptop]bkw__: he's only slightly evil
19:43.06file[laptop]bkw__: RUN!
19:43.09bkw__blitzrage, yes kinda why?
19:43.12bkw__its not that hard
19:43.20blitzragebkw__: I know
19:43.34blitzragebkw__: I'm just curious if you've ever run into problems running JS from within an included PHP file?
19:43.46blitzragei.e. index.php includes something.php, which contains JS
19:44.03bkw__as long as its wrapped in a <script language=javascript></script>
19:44.14bkw__firefox won't honor it if its  just <script></script>
19:44.37blitzrageyep... it is... :(  same script in the index.php works, in the included php I get "document.test.event is null or not an object"
19:44.43blitzragebkw__: its IE
19:44.43Ariel_mog_home, can I ask you some questions about connecting a te410p to an att 5ess switch
19:44.47blitzragehttp://pastebin.com/353819
19:44.49mog_homeshoot
19:44.52mog_homei am a little busy
19:44.56mog_homebut ill get back to ou
19:45.14bkw__blitzrage, do you have javascript turned off?
19:45.30bkw__check the parsed output
19:45.36bkw__make sure it isn't missing anything
19:45.37blitzragebkw__: nope - like I said, it works in the index.php page, just not the included page
19:45.40bkw__does it work if you save it static?
19:45.43blitzragebkw__: I know the script works :)
19:45.48bkw__thats odd
19:45.57Ariel_mog_home, they said to use em_w for the setup. But when they send there call I dont' get the digits asterisk just says no s exten found
19:46.06blitzragebkw__: cp/pst to index.php, works. cp/pst to the included page, error
19:47.19BorgonAny of you use a vps/vds to run asterisk?
19:47.32*** join/#asterisk Dybdahl (n=Lars@oelsted.dybdahl.dk)
19:51.16bkw__bbl
19:51.31mog_homeyou using featd?
19:52.18Ariel_mog_home, no. Here are my settings: http://pastebin.ca/22060
19:53.02*** join/#asterisk Nix (n=Nix@81.213.125.220)
19:53.41*** join/#asterisk r0m (n=SysOp@bl8-28-113.dsl.telepac.pt)
19:56.02*** join/#asterisk rjx (n=rj@ppp-71-138-89-44.dsl.irvnca.pacbell.net)
19:56.14*** join/#asterisk file (n=jcolp@mctnnbsa30w-156034035106.nb.aliant.net)
19:56.19Borgon128RAM decent enough to run a personal asterisk server with 1 call at a time?
19:56.37fugitivoyes
19:56.47rjxhey all, asterisk@home newblet here ;) wondering if someone can help me troubleshoot an issue w/ an incoming trunk using broadvoice
19:56.48Nixwithout X yes...
19:57.42Ariel_rjx, you might get more help from #amportal
19:57.48rjxok
19:57.49rjxthanx
19:58.01rjxhrm, opless and emptier :P
19:58.32*** join/#asterisk tatum- (n=nospam@s142-59-92-194.ab.hsia.telus.net)
19:58.36rjxanyhow if anyone wants to take a look, summary is basically I dont seem to be registering, all the while i've tried a few register lines as well as settings changes
19:58.37rjxhttp://pastebin.com/353832
19:58.44*** part/#asterisk tatum- (n=nospam@s142-59-92-194.ab.hsia.telus.net)
19:58.53rjxget that over and over, as it keeps trying to re-register, but nothing is ever received
19:59.53chetare you supposed to be registering with your internal ip addressing?
20:00.04chetor useing your external ip?
20:00.35rjxi have externip set in sip.conf
20:00.39chetsince you are most likely using nat
20:00.43rjxthat i am
20:00.46chetits using 192.168
20:00.53chetwhich is internal
20:00.55rjxright
20:01.00bkw__h323 ROCKS!!!
20:01.15rjxmaybe it didnt take my externip settings
20:01.20doolph?
20:01.22cheti dont think it did
20:01.31rjxok, thx, checking on that
20:01.48cheti assume you have to reg via the external ip so broadvoice can reach you
20:02.26*** join/#asterisk tzafrir_ (n=tzafrir@CBL217-132-115-194.bb.netvision.net.il)
20:02.36rjxright
20:02.49rjxwouldnt that also imply a port forward would be needed on router?
20:03.01rjxin all howto's ive read, which imply being behind a nat, i havent seen it though
20:05.01rjxdo i also need a nat=yes ?
20:05.18chetim not really sure, maybe someone else can help, sorry
20:05.43rjxit's ok, thx anyhow
20:05.49rjxsame error though
20:05.50chetim new to * but that looked balatantly wrong
20:05.50rjxhttp://pastebin.com/353841
20:05.54rjxyeah
20:06.13rjxwell those arent necessarily *errors* but im not getting any response
20:06.24rjxso i dont know what a proper response looks like
20:08.18*** join/#asterisk Junbug (i=Junbug@AC8A59D4.ipt.aol.com)
20:09.24Junbugis it poss for an external script php etc... to reload asterisk?
20:10.37Ariel_Abort (6) on Primary D-channel of span 1
20:10.37Ariel_Sep  3 13:10:17 NOTICE[2398]: chan_zap.c:7977 pri_dchannel: PRI got event: HDLC
20:10.49bkw__its a frame slip
20:10.52Ariel_argh my settings are off for the pri
20:10.57Ariel_frame slip
20:10.59chetrjx- using asterisk@home?
20:11.11Ariel_bkw_, so how do I fix that
20:12.51*** join/#asterisk r0d3nt (i=anonymou@tinfoilhat.net)
20:13.03Junbugcan i use an external script php etc... to reload asterisk?
20:13.58bkw__Ariel_, you can't really
20:14.08QwellJunbug: asterisk -rx "reload"
20:14.42Junbugok thx
20:15.03rjxyes chet
20:16.05chetdoes your config look like this one at the top?
20:16.07chethttp://lists.digium.com/pipermail/asterisk-users/2004-November/074809.html
20:16.15chetnat=yes
20:16.19JunbugQwell: quick q   all aol members are banned in EFNET #linuxhelp?
20:16.23Ariel_I need to get these pri's working.
20:16.31chetcause i see this"#
20:16.32chetRetransmitting #3 (no NAT):"
20:17.23QwellJunbug: yep
20:18.17rjxyeah chet, i saw that and thought it was weird too
20:18.23rjxeven when i put nat=yes it still says no nat
20:18.26*** join/#asterisk netsurfer (n=bbjunkie@i-83-67-48-18.freedom2surf.net)
20:18.34chetdid you reload after making changes?
20:18.41chetor restart service
20:19.14rjxyes
20:19.22rjxim noticing that when i do things via AMP
20:19.27rjxthey dont save even if i hit apply etc
20:19.36rjxso im just now edit'ing configs by hand
20:19.39rjxsaving, and reloading
20:19.44rjxnow it says NAT
20:19.45chetyou can always edit it manually and restart then
20:20.02rjxthat's what im doing, still not registering tho hrm
20:20.10rjxbut does say NAT now though
20:20.17chetpaste your config at pastebin
20:20.35rjxyeah, 1 sec
20:20.37rjxgot a new error
20:21.22netsurferhey Ariel_ - that page answered all the questions.. got it workin now :) thanx
20:21.46rjxhttp://pastebin.com/353859
20:22.32chetpaste your config
20:23.40rjxhttp://pastebin.com/353862
20:23.54rjxi removed the passes myself, in the config, they are in there where you see x's
20:24.27*** join/#asterisk SexyKen (n=ksandell@c-24-5-129-114.hsd1.ca.comcast.net)
20:26.17Ariel_bkw_, have configured any te410p? I need to fix this slip and they gave me what there switch is set too
20:26.58chetrjx, how did you change your ip before?
20:27.07netsurferif I use #include filename.conf in extensions.conf can I duplicate contexts ?
20:27.15rjxin sip.conf, under general, i put externip =
20:27.41Corydon76-homenetsurfer: the include simply inserts the text of that file
20:27.54Corydon76-homeYes, you can include the same file multiple times
20:28.14Corydon76-homebut no, you can't duplicate contexts
20:28.38*** join/#asterisk pbxbart__ (n=pbxbart@p54B00652.dip0.t-ipconnect.de)
20:28.39netsurferCorydon76-home - so if I have [default] in main file and in an #include 'd file it is actually just like one big [default] context ?
20:28.57Corydon76-homeDepends on which file
20:29.30Corydon76-homeThe best way to do that, though, is simply not to have ANY context definition in an included file
20:29.56netsurferCorydon76-home - then where do the extensions end up?
20:29.58Corydon76-homeJust extensions, then include those extensions into whatever contexts you want them
20:30.15netsurferah hmmm
20:30.39netsurfersounds like it could get messier than what the file already is lol
20:30.43Corydon76-homeHowever, it's probably better to create special contexts and "include" them instead of "#include"ing them
20:31.16netsurferthe idea was to keep internal extensions in a seperate file, to ease the admin a bit
20:31.23Corydon76-homei.e. [internal] include=office [incoming] include=office
20:31.54Corydon76-homeSure, you can do that
20:32.12Corydon76-homeJust don't duplicate contexts
20:32.15chetrjx, can you paste your sip.conf? im not really sure why its not working
20:32.23sivanausing perl, is it difficult to parse a .conf file?
20:32.28Corydon76-home[internal] #include "extensions/internal.conf"
20:32.49Corydon76-homesivana: you don't know perl, do you?
20:32.55sivananope
20:33.07Corydon76-homesivana: then yes, it's going to be difficult for you
20:33.08sivanaI have a perl script.. that I paid someone to write, but I need some mods done
20:33.17rjxchet, sure, 1 sec
20:33.46chetrjx, here is a link with working config, but its not broadvoice, but assuming you have that right it should work- http://www.voip-info.org/wiki-Asterisk+FWD+NAT+Config+Example
20:33.55Corydon76-homeThe file isn't that difficult to parse, no matter what language you use (unless you use a language that doesn't know how to open files)
20:35.38*** join/#asterisk pauldy (n=pauldy@c-67-162-255-229.hsd1.tx.comcast.net)
20:35.40sivanaok
20:35.55rjxok chet
20:35.57rjxhttp://pastebin.com/353871
20:36.15*** join/#asterisk drbrown (n=chatzill@63.238.118.90)
20:36.43*** join/#asterisk nemisus (n=nemisus@203-217-87-135.dyn.iinet.net.au)
20:37.31rjxchet, im wondering about an auth_password using broadvoice
20:37.42rjxim using the password that I am logging in to their web system with
20:37.51rjxive put in a helpdesk ticket
20:38.02rjxbecause under my device settings on the web, it doesnt have an auth pass listed
20:38.42*** join/#asterisk used (n=used@c-24-22-125-179.hsd1.or.comcast.net)
20:40.56*** join/#asterisk RoyK (n=roy@80.232.64.251)
20:42.22*** join/#asterisk queuetue (n=queuetue@h69-21-252-54.69-21.unk.tds.net)
20:43.58*** join/#asterisk Nox (n=secret@pool-71-243-231-201.tampfl.fios.verizon.net)
20:45.08*** join/#asterisk ericm_06 (n=ericm_06@c-24-9-147-88.hsd1.co.comcast.net)
20:45.15*** join/#asterisk n0ise (i=n0ise@d38-16-2.commercial1.cgocable.net)
20:45.57*** part/#asterisk n0ise (i=n0ise@d38-16-2.commercial1.cgocable.net)
20:49.15*** join/#asterisk Phoebers (n=phoenix_@82-168-146-4-bbxl.xdsl.tiscali.nl)
20:49.27Phoebersgood evening :)
20:49.54L|NUXevening
20:50.19*** join/#asterisk Borgon (i=Borgon@70-100-53-251.dsl1.tbr.ga.frontiernet.net)
20:52.41PhoebersI have a question :P, how do I load config files?
20:52.52L|NUXreload sip
20:52.57L|NUX:)
20:53.09*** join/#asterisk jeffik (n=Jeff@toronto-HSE-ppp3985137.sympatico.ca)
20:53.37L|NUXsip reload
20:53.41L|NUXextension reload
20:53.45L|NUXblah blah
20:53.55Phoebersconf files are auto used?
20:54.35Phoeberserm read
20:54.35Phoeberswotever u call it
20:54.35*** join/#asterisk tzafrir_ (n=tzafrir@CBL217-132-115-193.bb.netvision.net.il)
20:56.40rjxim on hold w/ broadvoice
20:56.44rjxwhich seems like forever
20:56.52rjxand their onhold music makes me wanna punch a baby
20:58.35chetyeah, but once its working you will be happy
20:58.55rjxyeah some1 just came on and hung up on me
20:58.59rjxgaaaahdamn
20:59.05*** join/#asterisk tzafrir__ (n=tzafrir@CBL217-132-72-158.bb.netvision.net.il)
20:59.19chethaha
20:59.30*** join/#asterisk Borgon (i=Borgon@70-100-53-251.dsl1.tbr.ga.frontiernet.net)
20:59.35chetyou would think they would post a working * config since it must be so common
20:59.35rjxi have an email in, maybe i'll just put this project to rest till i get a reply
20:59.41rjxoh they have one
20:59.51rjxi think i need a diff't password
21:00.02rjxive tried like 10 diff configs, all similar
21:00.46ericm_06rjx: remember, the web portal password is diff. then the auth. password
21:00.57*** join/#asterisk ManxPower (i=eric@1Cust2442.an1.dfw28.da.uu.net)
21:01.11rjxah ha! thanks
21:01.14rjxyou just confirmed my thoughts
21:01.28*** join/#asterisk huslage (n=huslage@c-24-22-125-179.hsd1.or.comcast.net)
21:01.28rjxi need to call support and have them change my device and get that pass
21:01.33rjxthat'll probably fix my prob
21:01.43rjxwell at least it's a step in the right direction
21:01.59ericm_06well, good luck
21:02.09rjxtheir onhold music is the worst i think ive ever heard
21:02.22rjx*shuts down *@home box till he gets an email reply
21:10.19*** join/#asterisk razu (n=razu@213-35-173-39-dsl.prn.estpak.ee)
21:11.45*** join/#asterisk tzafrir_laptop (n=tzafrir@CBL217-132-231-46.bb.netvision.net.il)
21:14.20*** join/#asterisk BVEBVE (n=bvebve@virgo.minbas.cu)
21:15.23*** part/#asterisk ericm_06 (n=ericm_06@c-24-9-147-88.hsd1.co.comcast.net)
21:21.22ManxPowerYAY!  My main server is back up!
21:22.01QwellManxPower: congrats
21:23.00*** join/#asterisk ipso (n=ipso@d207-81-249-35.bchsia.telus.net)
21:24.06ManxPowerAll our AT&T T-1s are down, but the frame relay network is up
21:24.50*** join/#asterisk BVEBVE (n=bvebve@virgo.minbas.cu)
21:26.08ManxPowerAnd the spam is flowing
21:26.33QwellManxPower: where are your servers?
21:26.48ManxPowerQwell, Covington LA.
21:26.55ManxPower30 miles north of downtown New Orleans
21:26.57Qwellpretty bad there?
21:27.27ManxPowerQwell, no.  power is on, many trees down.  One of the offices (Mandeville LA) has a tree thru part of it's roof
21:27.44Qwellahh
21:27.49ManxPowerI'm still in TX.  But the MIS manager is at corporate office in Covington
21:27.51Ariel_ManxPower, have you seen this:Sep  3 14:04:05 NOTICE[13585]: chan_zap.c:7977 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1
21:28.00ManxPowerAriel_, Yup.
21:28.02Ariel_do you know of a quick fix?
21:28.17QwellManxPower: where you staying at?
21:28.22ManxPowerAriel_, Um, HDLC abort errors can be as hard to fix as echo.
21:28.31Ariel_ahh
21:28.33Ariel_it works
21:28.35ManxPowerQwell, 7 miles outside of Atlanta TX (30 miles from Texarkana)
21:28.43QwellManxPower: no, I mean, hotel or something?
21:29.19*** part/#asterisk BVEBVE (n=bvebve@virgo.minbas.cu)
21:29.39ManxPowerStaying with the BF's family
21:30.00ManxPowerthree of us, three cats, and a dog all evacuated.
21:30.03Qwellahh
21:30.28*** join/#asterisk tzafrir_ (n=tzafrir@DSL217-132-221-74.bb.netvision.net.il)
21:31.58Ariel_ManxPower, glad your safe which is important
21:32.16Ariel_ManxPower, so where should I start to trouble shoot and change settings
21:32.22ManxPower90% of the city where I live(d) is reported to be totally destroyed.
21:32.33ManxPowerAriel_, What distro?
21:32.48Ariel_fedora 4 using head
21:33.15*** join/#asterisk Nix (n=Nix@81.213.125.220)
21:33.24ManxPowerAriel_, look in /etc/sysconfig/harddisks and enable all the options, then reboot.
21:33.44Ariel_ok will do
21:34.07*** join/#asterisk ipso (n=ipso@d207-81-249-35.bchsia.telus.net)
21:34.38ManxPowerAriel_, HDLC abort means "got corrupted data from the card, usually from missed or delayed interrupts"
21:35.10Ariel_great will start doing checks tonight when they are less active
21:35.17Phoebersw00t, I made my first connectivity with asterisk!!! hooray!!!
21:36.03*** join/#asterisk tzafrir__ (n=tzafrir@CBL217-132-231-4.bb.netvision.net.il)
21:36.35ManxPowerAriel_, the occasional HDLC abort error is not usually a problem
21:36.57Ariel_yes your right but they don't want to see them.
21:37.12ManxPowerAriel_, other common causes are IRQ shareing, evil built in ethernet controllers, and evil ide controllers
21:37.48*** join/#asterisk kg (n=kg@chello062179062077.chello.pl)
21:37.51Ariel_wow that is great info.  I am writing it down. Tonight I am going to start to play with those settings.
21:38.06fugitivoyou forgot evil sata controllers
21:38.35ManxPowerfugitivo, Yeah, evil SATA controllers and evil RAID controllers
21:38.54ManxPowerfugitivo, were you the one that got rid of HDLC errors by switching away from SATA?
21:39.27fugitivono, but i experienced problems with sata, usb, and evil hardware on an evil motherboard
21:40.12ManxPowerAh
21:40.20MikeJ__EVIL?
21:41.11*** join/#asterisk ericm_06_ (n=ericm_06@c-24-9-147-88.hsd1.co.comcast.net)
21:41.23MikeJ__hello ericm_06_
21:41.30MikeJ__welcome to the party
21:41.42ericm_06_hello
21:47.38*** join/#asterisk xai (n=pasta@cpe-70-112-17-10.austin.res.rr.com)
21:48.16xaimanx: where are you?
21:48.49xaiManxPower: where you at now?
21:49.07ManxPowerxai, texas
21:49.23xaiManxPower: i'm gonna dialog you..
21:49.52ManxPowerok
21:54.14DarthClueManxPower: have you seen the pictures of the waveland area?
21:54.41ManxPowerDarthClue, I'm trying to avoid those pictures
21:54.52MikeJ__noaa has some arial pictures
21:55.01DarthClueManx: just to see if your place is still standing ... http://ngs.woc.noaa.gov/katrina/KATRINA0000.HTM
21:55.13stepcutargh, sipphone.com's new user sign seems to be busted -- I never get the email I am supposed to get :-(
21:55.22ManxPowerDarthClue, I'm trying to avoid those pictures
21:55.37ManxPowerBut thank you for trying to insist I look at depressing pictures.
21:55.38DarthClueManxPower: understood.
21:56.12DarthClueManxPower: last i knew you weren't sure if it was still standing or not.
21:57.35ManxPowerI don't want pics, I want to see in person
21:59.16*** join/#asterisk iCEBrkr_ (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com)
21:59.38MikeJ__ManxPower, do you have any word yet?
22:00.21ManxPowerMikeJ__, no
22:00.38MikeJ__The last person I knew in the area got out of the french quarter yesterday and is on her way back to MI
22:00.39ManxPowerMy domain and servers should be back to full operation this evening, however.
22:01.40MikeJ__ManxPower, where in waveland is your place?
22:01.41ManxPowermy landlines should be forwarded soon as well
22:01.52ManxPowerMikeJ__, Old Spanish trail and McLauren
22:07.22*** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no)
22:08.50*** join/#asterisk f_meehan (n=fmeehan@whoami8.cedval.org)
22:11.08*** join/#asterisk Borgon (i=Borgon@70-100-53-251.dsl1.tbr.ga.frontiernet.net)
22:11.18MikeJ__ManxPower, I found an arial.. looks like the storm surge did not make it that far up
22:11.30Borgonwhats the command to patch asterisk with a diff? "patch -p0 < blah.diff ?
22:14.39ManxPowerMikeJ__, *nod*
22:16.23blitzrageBorgon: aye
22:16.40DarthClueManxPower: Intersection of OST and McL, which direction from there?
22:16.49blitzrageMcL?
22:17.18DarthClueOld Spanish Trail and McLauren
22:17.42Borgonblitzrage: thanks
22:19.16ManxPowerDarthClue, at the corner
22:19.24ManxPowerSE corner
22:19.30Byteis there a way to put arguments onto multiple lines?
22:20.13DarthClueright on the corner, with driveways on both OST and McL?  looks like it is still there.
22:20.42ManxPowerDarthClue, yup
22:21.15DarthCluelooks like a few houses in that area got hit, but yours looks like it is still there.
22:23.51*** join/#asterisk huslage_ (n=huslage@c-24-22-125-179.hsd1.or.comcast.net)
22:25.14Bytehttp://redrum.arlott.org/tmp/macro-call.txt <-- does that macro look ok? it seems to work but I don't know how many arguments I can add before it'd stop working right
22:27.01*** join/#asterisk clive- (n=pirch@rrba-146-117-44.telkomadsl.co.za)
22:29.48Phoebersgood night
22:30.21*** join/#asterisk nemisus (n=nemisus@203-217-87-135.dyn.iinet.net.au)
22:32.22clive-does anyone know what ast_verbose  would do in channel.c ?
22:34.57*** join/#asterisk ManxPwr (n=eric@1Cust4089.an1.dfw28.da.uu.net)
22:35.36clive-Manx, hi  do you know what ast_verbose  would do in channel.c ?
22:37.35*** join/#asterisk Borgon (i=Borgon@70-100-53-251.dsl1.tbr.ga.frontiernet.net)
22:37.46bkw__Byte, wtf kind of macro is that
22:37.47BorgonAfter i apply an asterisk patch do i have to recompile and build from srcs again?
22:38.16bkw__you can make that smaller
22:38.25Bytehow?
22:38.38bkw__all those args
22:38.40bkw__how nasty
22:38.44bkw__you should just pass it a number
22:38.49bkw__and let the macro do all the dirty work
22:38.51bkw__too many args
22:40.21Borgonbkw__: after applying RPID patches, is a recompile from /usr/src/asterisk required?
22:40.28*** join/#asterisk mhnoyes_ (n=mhnoyes@user-2ivflke.dialup.mindspring.com)
22:40.28QwellBorgon: yes
22:40.30bkw__Borgon, yes
22:40.32Byteso there's a simplier way to express: exten => _01.,1,Macro(call,e164,${UKCID},44${EXTEN:1},sip,${UKCID},${EXTEN}@sipgate2) ?
22:40.35Borgonok
22:40.47bkw__Byte, yes you seem to have some duplication of code there
22:41.01bkw__the same thing is passed twice
22:41.05bkw__in two differnt args
22:41.06Bytewhere?
22:41.15bkw__${UKCID}
22:41.17Bytethe callerid could change depending on what is used to make the call
22:41.26bkw__and ${EXTEN} should be passed once
22:41.26Byteso that's a bad example of why I'm passing the caller id
22:41.48bkw__you can manage the differences in the macro
22:41.58Byteif I pass ${EXTEN} once, how am I supposed to tell it to remove 1 digit from the start and add 44?
22:42.07Qwellsame way
22:42.14bkw__you do that in your macro ${ARG1:1}
22:42.20bkw__i'm just saying thats an ugly macro :P
22:42.28Bytebut the macro doesn't know what to do :P
22:42.31tzangerbkw__: your asterlinke mail goes into no detail.  what specific issues are you seeing with chan_iax2?
22:42.50bkw__tzanger, outbound audio to the pstn will be clippy and choped to hell
22:42.59bkw__i'm kinda shocked you haven't heard that
22:43.18tzangerbkw__: I haven't
22:43.27tzangerand every single call we take and make is to PRI
22:43.41bkw__hrm
22:43.41tzangerbkw__: is this only under significant load?
22:43.46bkw__not really
22:43.48bkw__20 calls
22:44.01bkw__you'll have 10-15 seconds of audio just disappear at times
22:44.18bkw__and as josh and i found out
22:44.19Qwellbkw__: Why haven't you fixed it yet? :p
22:44.23bkw__it either happens really badly for you
22:44.25bkw__or not at all
22:44.28tzangerbkw__: hmm
22:44.29bkw__and when it doesn't happen WTF
22:44.33tzangerI have been seeing iax2 jb sync issues
22:44.44bkw__like if a call is up
22:44.47tzangerbut only on some iax2 connectiosn
22:44.54bkw__and say a few registers hit the box
22:45.04bkw__no audio for that instant when its doing the register
22:45.06*** join/#asterisk anonymouz666 (n=lynx@allende.redetaho.com.br)
22:45.09bkw__when it has 20 or so calls on the box
22:45.29tzangerbkw__: that's really weird
22:45.35bkw__it is
22:45.39bkw__I had to switch to sip
22:45.44bkw__so people I called could actually hear me
22:45.44tzangeris this something that just started happenning?  I.e. if you roll back 3 months does it still occur?
22:45.51tzangerbkw__: yeah I saw that in the email
22:45.52bkw__I can't roll back
22:46.07bkw__we already use things that are in the current CVS that didn't exist 3 months ago
22:46.18*** join/#asterisk drbrown (n=chatzill@63.238.118.90)
22:46.19bkw__its an iax2 issue
22:46.19tzangerbkw__: just for testing, I mean
22:46.23bkw__because it doesn't happen on SIP
22:46.30tzangerbkw__: I understand that
22:46.51tzangerwhat I'm looking for is WHEN it started happennign on IAX2, to try and pinpoint what the change to chan_iax2.c was that does it
22:46.56bkw__tzanger, well like I said some people it happens to.. some it doesn't and we can't nail down exactly what causes it.. I have checked timestamps
22:46.59tzangerthere have been lots of changes to all of the channel drivers over the past little whiel
22:47.00bkw__I have checked jitter buffer on and off
22:47.18tzangerbkw__: yeah bnut it does happen to you so you're in the best position to try and nail down when it started to happen
22:47.18bkw__its really strange really
22:47.23*** join/#asterisk tzafrir_laptop (n=tzafrir@CBL217-132-115-152.bb.netvision.net.il)
22:47.28bkw__I have done all I can to try
22:47.30Qwellbkw__: really?
22:47.33bkw__its the strangest thing
22:47.39bkw__nothing makes sense
22:47.43*** join/#asterisk CaNaBiS (i=canabis@ip68-111-51-215.br.no.cox.net)
22:47.46bkw__i'll try again next week to get more time to check it out
22:48.03tzangerbkw__: yeah that's the shitty part... it usually happens to people with no additional time :-S
22:48.13CaNaBiShey, I am in Baton Rouge, making calls has been difficult. Are there any providers like voicepulse or broadvoice that have a free trial period?
22:48.17Qwelltzanger: Murpheys Law
22:48.18bkw__I think tuesday I'll have time
22:48.36tzangerhas anyone heard from ManxPower since the hurricane?
22:48.37bkw__CaNaBiS, who do you need to callo?
22:48.40bkw__tzanger, yes
22:48.42tzangerhe was in NO IIRC
22:48.43Qwelltzanger: yeah, he's in TX with family
22:48.46bkw__ManxPower, is in here
22:48.48tzangerQwell: oh good
22:48.49CaNaBiSbkw_, family in TN
22:49.05bkw__CaNaBiS, ok you just need to make a few calls?
22:49.10CaNaBiSnawp
22:49.18bkw__nawp?
22:49.20tzangerI mean I wasn't sure what I was gonna do if I couldn't tease him about STABLE.  :-)
22:49.21CaNaBiSnope
22:49.25bkw__how many calls?
22:49.51CaNaBiSI can make calls, theyre just scattered
22:50.21CaNaBiSand when I go to TN I need to be able to call back here which is the hardest
22:50.59*** join/#asterisk santiago (n=santiago@63.245.86.163)
22:51.06CaNaBiSI thought that I had remembered seeing a provider that had a trial period
22:53.02*** join/#asterisk shido6 (n=curtis@d57-87-253.home.cgocable.net)
22:54.27*** join/#asterisk Moc_ (n=Moc___@229-198-0-72-ppp.3menatwork.com)
22:55.40*** join/#asterisk tessier (n=treed@wsip-68-224-172-77.sd.sd.cox.net)
23:01.13*** part/#asterisk santiago (n=santiago@63.245.86.163)
23:01.38*** join/#asterisk tessier (n=treed@wsip-68-224-172-77.sd.sd.cox.net)
23:07.10dougheckaanyone have one of these zyxel phones?
23:08.11*** join/#asterisk comadreja (n=comadrej@pdpc/supporter/active/comadreja)
23:08.31comadrejahello, I'm getting a connection refused on an inbound call, why could that be ?
23:08.52*** join/#asterisk borgasman (n=aa@bl5-39-66.dsl.telepac.pt)
23:08.54borgasmanEvery geeks dream clock muahahaha »» http://systemsecure.org/ssforum/viewtopic.php?t=162
23:11.03comadrejathat is a "Rejected connect attempt from..."
23:11.06*** join/#asterisk CaNaBiS_ (i=canabis@ip68-111-51-215.br.no.cox.net)
23:13.32*** join/#asterisk ManxPower (i=eric@1Cust5281.an8.dfw28.da.uu.net)
23:13.59*** part/#asterisk borgasman (n=aa@bl5-39-66.dsl.telepac.pt)
23:15.45*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
23:16.01*** join/#asterisk smcmahon (n=admin@digitaldatabits.net)
23:19.25*** join/#asterisk PBXtech (i=nik@228.sub-70-218-21.myvzw.com)
23:19.58PBXtechdoes anyone make a pre-installed vicidialer config?
23:20.20tzangernot I no
23:20.53tzangerha
23:21.00tzangerwha the hell is hour 13-15?
23:21.27doughecka'\?
23:21.28tzangerI have a binary clock at home
23:24.08*** join/#asterisk liberie (n=liberie@unaffiliated/liberie)
23:24.18smcmahonHour 13 - 15? That would be 1PM - 3PM ...
23:24.22*** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com)
23:25.07smcmahonI need a binary clock that would be cool, I learned binary at the age of 3 yrs old lol
23:27.26tzangersmcmahon: so how doe sit work...3pm then go to 12 midnight, or does it skip those three hours or?
23:28.01smcmahonHow does what work? tzanger...
23:28.09*** part/#asterisk clive- (n=pirch@rrba-146-117-44.telkomadsl.co.za)
23:29.00*** join/#asterisk smcmahon (n=admin@digitaldatabits.net)
23:29.25ManxPower~docs
23:29.27jbotfrom memory, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
23:29.28ManxPower~mailinglist
23:29.29jbotwell, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
23:29.42Qwellsmcmahon: thinkgeek.com, they have two clocks, and a watch or two
23:30.58*** join/#asterisk tzafrir_laptop (n=tzafrir@DSL217-132-221-100.bb.netvision.net.il)
23:31.17MikeJ__waving?
23:31.52tzangerManxPower: hey!  Glad ot see you're at least out of harm's way...  how are things?
23:32.05dougheckaManxPower: hows it hangin?
23:32.24ManxPowertzanger, slightly better.  main customer's corporate HQ is back up.
23:32.31ManxPowermost of the frame relay network is working.
23:32.42ManxPowerPower is on.  The AT&T T-1s are down, however.
23:33.00dougheckawhere is it located?
23:33.02tzangerManxPower: at least you'll have some reprieve from the real estate people :-)
23:33.03ManxPowerdoughecka, I'm alive
23:33.08ManxPowerdoughecka, Covington LA
23:33.12dougheckaah
23:33.26ManxPowerthe Mandeville LA office has a tree thru one part of the roof, but no damage to the NOC there.
23:33.30dougheckabtw anyone use zyxel phones?
23:33.40dougheckaManxPower: and thats whats important
23:34.08ManxPowerI need to write up a donations page for us.
23:34.20smcmahonThe LED CLock is pretty neat.. Totally understand that one
23:34.24ManxPowerI'm in decent shape, but the two others that evacuated with me are not in good shape moneywise.
23:34.55ManxPowerthey both work for the nola tourist office
23:34.58Damin"She touched my pepe, Steve."
23:35.00*** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net)
23:35.11smcmahonLOL
23:36.35DaminOh.. sorry about that..
23:36.40*** join/#asterisk clona (n=clona@c213-158-235-5.sdsl.no)
23:38.08tzangerwhich led clock
23:38.23tzangerDamin: hahahaha  I forgot about that movie
23:38.27tzanger"hey teeve"
23:38.33comadrejaI'm getting a rejected connect attempt from.... why could that be ?
23:39.02comadrejaI'm quite noob, sorry :/
23:46.30*** join/#asterisk fugitivo (n=ajf@201.255.104.41)
23:53.00*** join/#asterisk nemisus (n=nemisus@203-217-87-135.dyn.iinet.net.au)
23:56.48Damintzangre: Yeah.. it showed up on my PVR the other day..
23:57.01Damintzanger: So I watched it...
23:57.15tzangerDamin: :-)
23:58.32Damintzanger: I asle watched the Pilot episode of the 1987 Buck Rogers series w/ Twiki and Dr. Theopolis
23:58.48JDLSpeedycan someone help me add a privacy screener to asterisk?
23:59.24tzangerJDLSpeedy: what is a privacy screener
23:59.31JDLSpeedyI just found out how to record a name and play it back to the person
23:59.56DaminJDLSpeedy: Get a GrandStream BT100 and use Ringotne #2 on it. ;) "You have a call from 216-619-2000"

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