irclog2html for #asterisk on 20051214

00:00.05_Vileharlequin, look at generating "call files", in /var/spool/asterisk/outgoing
00:00.15harlequin516dialout commandline control?
00:00.21cripito:))
00:00.43harlequin516Ah right!!!Thanks _Vile, I remember reading this...
00:01.04_VileHarle - http://www.voip-info.org/wiki-Asterisk+tips+Wake-Up+Call+PHP
00:01.05_Vilemay help
00:01.28tzangerharlequin516: you did not read my message
00:04.56Dr-Linuxtzanger: i faced a problem today, when i call reciever hears me very low, but i can hear him fine, what things need to be chacked?
00:05.00*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal)
00:05.35*** join/#asterisk newl (n=newlook@203-59-61-62.dyn.iinet.net.au)
00:05.38tzangersounds like your txgain is set too low
00:06.29*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)
00:06.33_Viletzang, ltns -- ever played with routing on diversion headers?
00:07.25_Vileis bkw still around?
00:07.54tzanger_Vile: nope
00:08.28*** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net)
00:08.32harryvvseen Ariel
00:08.37_Vileseen bkw-
00:09.58Corydon-wbkw decided he'd be better off by himself.  His choice.
00:10.14mog_homeheh
00:10.56_Vilehis time of the month already?
00:11.52Corydon-wMore like, time of the year.  It's been several months
00:11.58mog_homeow
00:12.09mog_homeno he is in cough opbx if you want him
00:12.10_Vilehm, i've been gone for 4-5.. maybe 6
00:12.29_Vileinteresting
00:12.31mog_homethat sounds about right then
00:12.53Corydon-wHe basically renamed a whole bunch of APIs... and left it at that
00:12.55_Vilebranch?
00:13.06harlequin516tzanger:  Did not get a response from you, looked like you tried to respond but had a mixup
00:13.14mog_homehey they have autoconf....
00:13.30_Vilethat sucks, he's a good developer
00:13.35Corydon-wWow, they can autoconf...
00:13.45mog_homeheh whatever we have make menuconfig ^_^
00:13.51Corydon-wBoy, that's a good reason to step back 4 months in development time
00:14.01_Viledid he branch?
00:14.13Corydon-wYep
00:14.20Corydon-wFork, actually
00:14.30Corydon-wbut it might as well be a branch
00:15.00mog_homei would have disagreed but when they played catch up when we went 1.2 made me think again
00:15.02_Vileouch
00:15.49_Vilewell shit, who's going to answer my questions now? ;)
00:15.55mog_homeask
00:15.58mog_homeand we might answer
00:16.04_Viletrying to route on diversion sip headers
00:16.11_Vilesip diversion headers
00:16.16_Vilenot RDNIS
00:16.19rob0I'll answer but it will be wrong :\
00:16.27rob042?
00:16.31file[laptop]42!
00:16.31_Vilewell, RDNIS, but the "reason" part
00:16.45_Vilereason=no-answer
00:16.51_Vileanswer to life and everything?!?!
00:17.05Corydon-wDon't forget the universe
00:17.08_Vileor reason=user-busy
00:17.10_Vileoh yeah
00:17.21_Vile"Diversion: <sip:1231230001@123.12.0.123>;reason=no-answer" -- example header
00:17.39_VileI can get the 1231230001
00:19.28Dr-Linux<tzanger> sounds like your txgain is set too low
00:19.40Dr-Linuxtzanger: but the same everything was working fine
00:19.51Dr-Linuxtxgain = 0.0
00:19.59Dr-Linuxand thats in zapata.conf
00:20.09Corydon-w_Vile: have you filed a bug report?
00:20.13Dr-Linuxi m not saying problem with zap channels
00:23.51Dr-Linuxtxgain is related only with zap channels/ or its matter if we are using only softphones?
00:26.54*** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net)
00:27.32_VileCory, of course not
00:27.44_VileI don't know if it's a bug or not
00:27.50jimmy_deanPBHmmhesays, getting this error when trying to compile asterisk-1.2.1 on debian stable: "configure: error: termcap support not found"
00:28.04_Vileseems like a feature request, if it isn't available? trying to figure out if that reason code is routable before I file anything
00:28.04jimmy_deanPBthe only package with "termcap" in it I installed
00:28.36jimmy_deanPBs/Hmmhesays/hmm
00:28.54_VileScenario is: Sipura -> MetaSwitch -> Sipura (no answer) -> MetaSwitch (routes on NA) -> Asterisk
00:28.57_VileAsterisk picks up
00:29.15_Vilebut I need to figure out how to determine busy or no answer
00:29.42_Vileif it's not there, I could probably add the code and submit a patch
00:31.26Corydon-w_Vile: if you file a feature request on the bugtracker in the next 5 minutes, I'll upload a patch.
00:31.39_VileOK
00:40.40Corydon-w10 minutes later, still no feature request...
00:40.42litageif userA is connected to *boxA and makes a call to userB who is connected to *boxB, and everyone's using h323, which * box would need an h323 gatekeeper?
00:41.09Corydon-wNeither
00:41.54Corydon-wA gatekeeper is never really needed... it's a convenience when you don't know which gateway you need to connect to
00:42.05*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
00:42.21Corydon-wbut you can go gateway to gateway just fine and never encounter a gatekeepr
00:42.48_VileCory, I hadn't logged into Mantis in 6 months -- took me a couple of minutes to grab my pw
00:42.49_Vilesorry
00:42.50_Vilehttp://bugs.digium.com/view.php?id=5995
00:42.54litageCorydon-w: ah i see. but you can only go gw-gw IFF you know which gw you want to connect to?
00:43.28Corydon-wCorrect
00:43.49litageCorydon-w: awesome, that *really* clears up that question  =)
00:43.50Corydon-wA gatekeeper is essentially a directory service
00:44.35litageCorydon-w: so a gk needs to know about all available gateways. do gateways register with 1+ gatekeepers?
00:45.03Corydon-wThey can
00:45.51Corydon-wTypical installs, though, you have a single gatekeeper, and you hardcode that gatekeeper to send all calls starting with e.g. 1615 to a certain gateway, 1423 to another gateway, etc.
00:45.51robl^and the gatekeep has to know about the dungeon master and have the keys to the bridges.  :)\
00:47.07_Vileu rock
00:47.14_Vilethx
00:47.16litageCorydon-w: is gnugk the most "popular"/widely used/"best" gk?
00:48.03Corydon-wYou'd have to ask someone who actually gasp enjoys using h323
00:48.20litageCorydon-w: hahah. why does everyone i talk with hate h323?
00:48.32Corydon-wbecause it's an overly complex protocol
00:48.36}btorch{how can I convert a time like 10:15:00 to UTC format ? I guess if I enter 10:15:00 to the SAY TIME command that won't work
00:48.39}btorch{right ?
00:49.16Corydon-wIt implements every single condition that could possibly occur on a PRI, even though most of those conditions will never occur on an IP connection...
00:49.30litage}btorch{: how about an agi that runs the 'date' command?
00:49.49litagehrmm
00:50.15}btorch{no you are talking about shell dat cmd right
00:50.17Corydon-wUnfortunately, the solution for H.323 was also designed by committee, and suffers from some of the same maladies as H.323:  SIP
00:50.31litagei'm stuck with h323 for my protocol, and g729 for my codec, for 98% of my comms
00:50.39Corydon-wHowever, SIP is still a whole lot better than H.323
00:50.40}btorch{now you are talking about shell date cmd, right?
00:50.43*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)
00:50.53litage}btorch{: yes
00:51.37}btorch{litage: You see I get a pre- determined time from a database field
00:52.04}btorch{can the date command conver the time value I get into UTC ?
00:52.15}btorch{checking man page
00:52.17litage}btorch{: i can't remember off-hand. ``man date''
00:52.34rob0date -u
00:52.34litage}btorch{: you can convert times and timezones and whatnot in perl, too
00:53.44Corydon-wdate is essentially just a command line frontend to strftime, anyway
00:55.12}btorch{tried -u with date looks the same as date
00:55.24}btorch{tried date -u -d 10:15
00:56.35}btorch{isn't it wiered though that SAY time only takes the time in seconds ?
00:56.38Corydon-wTry +
00:57.40litage}btorch{: as Corydon-w said, don't forget the mandatory "+" prefix on the date string
00:58.39rob0Wed Dec 14 10:15:00 UTC 2005
00:59.18}btorch{Ithat's what I got
00:59.34rob0what did you want/expect?
00:59.42*** join/#asterisk adker (n=adker@67-51-239-152.dsl1.glv.ny.frontiernet.net)
01:00.26rob0try date --help to see the ways you can format it
01:00.27}btorch{well here ... according to "SAY TIME time escaped_digits " time is the number of seconds elapsed since 00:00:00 on january 1,1970
01:00.30Dr-Linuxtxgain is related only with zap channels/ or its matter if we are using only softphones?
01:00.44rob0date +%s (in GNU date only)
01:01.28}btorch{just saw that on the man page
01:02.07}btorch{cool thank you all
01:02.15}btorch{got test it now
01:04.25*** join/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net)
01:05.48*** part/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net)
01:05.51jimmy_deanPBDr-Linux, only zap channels
01:06.54*** join/#asterisk ncconquer (n=info@203.113.172.79)
01:06.59Dr-Linuxjimmy_deanPB: then what could be the problem, when i call i can hear fine, but reciever hears me very low?
01:07.03ncconquerhello
01:07.12ncconqueri've problem w asterisk
01:07.32ncconquerwhen i try: genzaptelconf -s -d
01:07.40ncconquer<PROTECTED>
01:07.54ncconquerHint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters.
01:08.15litagevoip-info.org (voip-info.org/wiki/index.php?page=Asterisk+sip+type) says that a peer sends calls, and a user receives calls. does that mean that a peer can't receive calls, and that a user can't send calls?
01:08.32*** join/#asterisk g0mb0 (n=test@external.micom.mng.net)
01:08.58*** join/#asterisk riddlebox (n=james@24-217-15-91.dhcp.stls.mo.charter.com)
01:09.21ncconquerplz help
01:09.36litagencconquer: please don't forget what i told you yesterday
01:09.52ncconquer?
01:09.56ncconquerhow litage
01:10.40litage[12:25]   litage ncconquer: continually asking for help will only annoy people
01:11.10robl^can someone help me annoy people?
01:11.16litagencconquer: after asking a proper question (which you didn't do; you only said you have a "problem"), you wait for a response. you don't say "please help"
01:11.41ncconquerok
01:12.10litagegreat  :)
01:12.32ncconquerdo u know how about my error?
01:13.08litagencconquer: i don't. however, all you gave us was an error. we need more information than that to help you. what did you do? what did you expect to happen? what happened? etc..
01:13.45*** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net)
01:14.04ncconquercan u login my system?
01:14.12ncconqueru can know how about
01:14.50*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
01:14.57jimmy_deanPBDr-Linux, you are using all SIP based things, no analog phone lines?
01:16.03Dr-Linuxjimmy_deanPB: yes
01:16.38Dr-Linuxjimmy_deanPB: it was workinf fine before :S
01:16.55jimmy_deanPBDr-Linux, it could be your SIP phone then...bad quality or something...I'm not sure how to adjust the volume though for SIP connections
01:16.56Dr-Linuxi'm not sure if my net speed were slow this time or what:S
01:17.15Dr-Linuxjimmy_deanPB: its X-Lite
01:17.22asterboyWhat is the cheapest *unlimited* local and long distance number going?
01:17.32*** join/#asterisk bweschke (n=bweschke@24-75-229-26.atlaga.adelphia.net)
01:17.44asterboyVonage has $40/month...anyone cheaper?
01:17.58De_Monbroadvoice is only $10 or something
01:18.20asterboy$10 is nice, is that all the features and long distance?
01:18.32De_Monyeah
01:18.55De_Monsupport isn't that great, but if it works who cares!
01:18.56asterboyno, that $10 is in state only.
01:19.08asterboy$20 for world
01:19.14De_Monnation wide = longdistance
01:19.34De_Monoverseas != long distance
01:20.18asterboybroadvoice is only $10 for in state, not nation wide
01:20.26asterboystill dam cheap
01:20.51De_MonI'm trying to setup some x-lite sip softphones, followed the wikis and howtoos, yet I'm getting Registration from '<sip blalblabla>' failed for 'myip'
01:20.59asterboywhy would you get DID?
01:21.31asterboyguess if you have low volume on appearances it would work out ok.
01:21.49De_Monrunning 1.2
01:21.52De_Mon.0
01:22.14asterboyotherwise you might as well pay only $10 and talk all you want and have inbound an outbound capabilities.
01:22.29asterboyThis landscape is changing fast.
01:23.51asterboyno doubt we have already reached 'singularity'
01:24.17*** join/#asterisk apardo (n=apardo@29.Red-81-39-85.dynamicIP.rima-tde.net)
01:24.52asterboyhttp://en.wikipedia.org/wiki/Technological_singularity
01:24.55De_Montheres a Bring Your Own Device rate of $6
01:26.46asterboynice.
01:27.12asterboyfor $6 DID is almost dead.
01:27.37*** join/#asterisk rezEdit (n=rezEdit@zapdos.omnigroup.com)
01:30.45riddleboxhello, I am trying to get a sip phone working with asterisk but I get the registration fails due to a username/authname mismatch?
01:32.03rezEditriddlebox: what kind of phone?
01:32.13riddleboxxten xlite
01:32.25rezEditI am using that with Asterisk as well
01:32.48riddleboxhow did you set it up in sip.conf?
01:33.43rezEdit[700]
01:33.43rezEdittype=friend
01:33.43rezEditnat=yes
01:33.43rezEditincominglimit=3
01:33.43rezEditusername=700
01:33.43rezEditsecret=1111
01:33.45rezEdithost=dynamic
01:34.12rezEditThen Authorization User and Username are both 700 in X-Lite
01:35.37rezEditX-Lite seems to be pretty unstable for me, unfortunately.  Crashes a lot.
01:35.51harryvvnever has for me. always stable.
01:36.07riddleboxI just want to get something running lol
01:36.59harryvvgo buy a cheap ip phone
01:37.16rezEditX-Lite will sometimes crash on launch, several times over, then be fine.  Other times, after several crashes it automatically re-creates it's preferences
01:37.31rezEditI just use it for testing, have Polycom 501's for real use
01:38.06*** join/#asterisk los415 (i=los415@los.race.com)
01:38.32rezEditI am having an issue with ChanIsAvail(), which doesn't seem to be working when I use it to test a SIP extension for availability.  Anyone else having issues?  For me it always returns like the channel is free when in fact, it is not.
01:39.02rezEditI am using it like: exten => 100,n,ChanIsAvail(SIP/224|sj)
01:40.14rezEditand it always goes to n+1, never to n+101, even when the phone is on another call
01:41.21harryvvrez, only complaint i dont like about the polycoms is no lit display.
01:41.54rezEditharryvv: yeah for sure.  I work on the system late at night sometimes, with very little light around and it sucks
01:41.58harryvvI dont know if thats a optional feature of was never considered when thay were designed.
01:42.14MstlyHrmlsI just turn a light on...
01:42.17rezEditI don't think it's available at all.
01:42.26rezEditthe 4000 conf phone has a backlight
01:42.27MstlyHrmlsyou could get an IP 4000s :-)
01:42.34harryvvI have a bright halogen light on behind me and the display is still on the dark side.
01:42.55MstlyHrmlsI've just got a desk lamp, and that works fine
01:43.10harryvvip4000?
01:43.44MstlyHrmlsSoundStation IP 4000
01:43.55MstlyHrmlsconference phone version of the SoundPoints
01:44.28rezEdityeah the 4000 has the backlight, but the screen is also very tiny
01:44.37los415anyone ever setup gr303 in asterisk ?
01:45.03MstlyHrmlsrezEdit: yeah, and you have to lean over it to see anything
01:45.16*** join/#asterisk mog_work (n=mogorman@gateway.digium.com)
01:45.40rezEditMstlyHrmls: yup.  thankfully we don't use it all that often
01:48.38litagevoip-info.org/wiki/index.php?page=Asterisk+sip+type says that a peer sends calls, and a user receives calls. does that mean that a peer can't receive calls, and that a user can't send calls?
01:49.06*** join/#asterisk klictel (n=klictel@modemcable185.108-200-24.mc.videotron.ca)
01:49.35*** join/#asterisk bch1 (n=bch@CPE-24-26-173-250.mn.res.rr.com)
01:50.34bch1anyone have trouble entering extensions on a cell phone and having Asterisk process it right?
01:51.02bch1say I enter extension 1234, sometimes it goes to extension 1233
01:51.13bch1or 1123
01:51.15*** join/#asterisk milkyflava (n=milkyfla@240-7-237-24.gci.net)
01:51.19milkyflavahello
01:51.46bch1no matter how long or short I hold down each key
01:52.25bch1is there some kind of workaround for that?
01:52.49milkyflavashould echocancel equal yes or some value i.e. 16,32,64,128?
01:53.53milkyflavasame thing with echo training, should it just be "yes" or 400 or 800
01:55.46litagemilkyflava: i've no idea, but why not try each value and see what happens?
01:56.18riddleboxrezEdit: this is the error I get:  chan_sip.c:10817 handle_request_register: Registration from '523 <sip:523@192.168.0.101>' failed for '192.168.0.3' - Username/auth name mismatch
01:56.48milkyflavalitage: I was asking because I have heard people say use yes and others say use 800, in the case of echotraining
01:57.21*** join/#asterisk xachen (i=justin@magnum.thisgeek.com)
01:57.25xachenHey all :)
02:00.36_Vileriddle, check the username/password on the box
02:00.52_Vilethen, make sure "use auth id" is off
02:01.19riddlebox_Vile:in sip.conf? or what I am trying to use xten-xlite softphone
02:01.23_Vileor auth name.. not sure what box you have
02:01.25_Vilehm
02:01.28jimmy_deanPBmilkyflava, having any of them = yes just uses a default number...that's the only difference that I'm aware of
02:01.28_Vileon the xten
02:01.42_Vilemake sure "use auth name" is off
02:02.04milkyflavajimmy_deanPB: Thanks, thats what I was thinking it did but wasn't sure
02:02.19jimmy_deanPByeah, I was just dealing with those config options today a lot :)
02:02.22jimmy_deanPBso they're fresh in my mind
02:02.25_Vilemake sure you have an entry for username
02:02.29milkyflavaI will be tonight
02:02.30_Vilemake sure your password is right
02:02.31_Vilewill work
02:02.43riddlebox_vile:I do not see where to turn use auth name?
02:02.49_Vileblah sec
02:02.55milkyflavaI read on the * list that Sangoma has a little utility to show how many taps are needed
02:03.24jimmy_deanPBdidn't know that, did google turn up anything about that for you?
02:03.27milkyflavaSo I want to fire that up and make a call and fine tune my echocancel and echotraining
02:03.44jimmy_deanPBmilkyflava, try out fxotune too
02:03.57milkyflavaNo, it was posted on the list today under the subject of X100p
02:04.00milkyflavaI am going to
02:04.04jimmy_deanPBok
02:04.08_Vilei'm wrong
02:04.12_Vileon the xlite
02:04.17jimmy_deanPBmilkyflava, what version of asterisk by the way?
02:04.21_Vileyou need "Authorization user" and username to be the same
02:04.27_Vileas what is in sip.conf
02:04.29milkyflavanow if I use fxotune I still need to setthe echocancel, echotraining but not the tx/rx settings, correct?
02:04.32milkyflava1.2
02:04.41_Vileand password to be the same as the secret= you use in sip.conf
02:04.45jimmy_deanPBok, have you tried things out yet to see if there is echo?
02:04.54milkyflavathere is echo
02:04.54_VileEnabled must be switched to YES - otherwise nothing will work even if your user registration is correct. Display Name may be whatever you want to be displayed. Username and Authorization User must be as in sip.conf and in our example it is ivan. Password has to be same as secret which is set in sip.conf for the user.
02:04.54_VileDomains and Proxy have to be the IP addresses of your asterisk server. Note that here my asterisk IP is 10.3.3.25 but in your case it might be different.
02:04.59_Vileoops...
02:05.00jimmy_deanPBI was told that just using * v1.2 makes a huge difference by default
02:05.19jimmy_deanPBmilkyflava, with or without echocancel, echotraining, etc?
02:05.19milkyflavaprobably because it uses a different echo can than 1.1
02:05.21_Vilehttp://www.asteriskguru.com/tutorials/xlite_softphone.html
02:05.33milkyflavaI have them both set to yes right now
02:05.38jimmy_deanPBok, hmm
02:05.59jimmy_deanPBthat makes me a little nervous, I'm hoping an upgrade to version 1.2.1 for me takes care of my echo problems
02:06.25milkyflavaYou might try using the MG2 echo can or maybe it is MG1 I never can remember
02:06.35jimmy_deanPBI am using MG2 now
02:06.38milkyflavahow bad is your echo?
02:06.41jimmy_deanPBI already changed to that
02:06.49milkyflavaI used MG2 but reverted back to KB1
02:06.58jimmy_deanPBmilkyflava, users of the system I setup described the phone system as "unusable" because of the echo
02:07.17milkyflavahave you adjusted the tx/rx gains?
02:07.24jimmy_deanPByep, everything
02:07.29jimmy_deanPBspent all day today working on that
02:07.39milkyflavadid you use fxotune and make sure it starts before asterisk on a reboot
02:07.43jimmy_deanPByep
02:08.02milkyflavadid you reset your rx/tx gains back to 0.0 after using fxotune
02:08.07*** join/#asterisk ldnblk (n=Just@optbom1.uk.access.vodafone.net)
02:08.25jimmy_deanPBmilkyflava, what do you mean by that exactly?
02:08.45milkyflavaWhen you use fxotune you shouldn't adjust your rx/tx gains
02:08.50milkyflavause either one or the other
02:09.21milkyflavafxotune takes the place of adjusting the rx/tx gains
02:09.22jimmy_deanPBoh right, yeah I did that
02:09.24jimmy_deanPBright
02:09.25milkyflavaok
02:09.35milkyflavaso you had echo on both
02:09.51milkyflavaafter fxotune and when you tried to adjust rx/tx gains
02:10.03milkyflavaand how bad is the echo you are hearing
02:10.13jimmy_deanPByeah, best I could get it to was no echo to begin with, then 3-4 seconds later the echo was back to full strength
02:10.29jimmy_deanPBmilkyflava, the echo is as loud as the voice going into the handset microphone
02:10.30milkyflavawow
02:10.47milkyflavawhat are you using for a phone
02:10.50jimmy_deanPByou can't hear yourself think when using it
02:10.54milkyflavai bet
02:11.16*** join/#asterisk TheCops (i=nobody@got.securebinary.com)
02:11.16jimmy_deanPBmilkyflava, cordless phone, low-end SIP phone, and high-end PolyCom SIP 501 all have the same echo
02:11.51jimmy_deanPBso the cordless phone is a direct bridge
02:12.03jimmy_deanPBon the TDM400P card
02:12.12jimmy_deanPBand yet even it echos just the same as the SIP phones
02:12.27*** join/#asterisk apardo (n=apardo@29.Red-81-39-85.dynamicIP.rima-tde.net)
02:12.41milkyflavahow far down did you adjust your tx andrx when you were trying to adjust them
02:12.45litageusing the sample configs, when i connect to extension 500 and disconnect, i get a busy tone for a second. why might that be?
02:13.08jimmy_deanPBmilkyflava, I got far enough down to where you couldn't hear the incoming at all nor hear the outgoing
02:13.19milkyflavalol and still echo?
02:13.23jimmy_deanPBno echo then
02:13.26jimmy_deanPBnothing then
02:13.31jimmy_deanPBno echo, but no voice either :)
02:13.35TheCopsI guess that PRI dont have any echo on that type of technology ?
02:14.09milkyflavathats because there are two hybrids removed out of the equation with pri lines
02:14.21milkyflavait is 4 wire to 4wire from you to the CO
02:14.30jimmy_deanPBnice
02:14.36jimmy_deanPBbut costs a lot more :)
02:14.39milkyflavaI have been thinking of doing the same thing
02:14.43milkyflavayep
02:14.54jimmy_deanPBdoing what?
02:15.03milkyflavawhat are your settings in zapata.conf
02:15.11milkyflavabuying a pri line instead of analog
02:15.25_Vilemost recurring problem I've heard w/ * is echo....
02:15.52milkyflavayep, plan and test before you implement _Vile
02:15.58jimmy_deanPBmilkyflava, how much are they if you know?
02:16.11_Viletested, planned, implemented..
02:16.14milkyflavaI am not sure but I have heard 400-2000 per month
02:16.15jimmy_deanPBecho is a hard thing with phone systems in general
02:16.19_Vileecho only happens ocassionally on our PBX
02:16.20jimmy_deanPBdang!
02:16.27jimmy_deanPBnot cheap
02:16.30drrayI've hardly got any echo
02:16.38drrayI've been blessed, I guess
02:16.44milkyflavaecho happens at the beginning of the call and then once in awhile during the conversation
02:16.46_Vilespeakerphone is an issue on 7940s
02:16.52_Vilew/ echo
02:17.06jimmy_deanPBthe funny thing is, this is our second * PBX...the last one was a way underpowered one running an old Asterisk@Home version with the same TDM400P card
02:17.22jimmy_deanPBno echo, but lots of chirps and clicks due to a weak processor
02:17.44milkyflavathats caused by shared IRQs with the TDM card and some other device
02:17.58litagecan users/peers/friends specified in h323.conf register with the asterisk server?
02:18.17jimmy_deanPBperhaps, but really I know it's partially due to it only being a Celeron 350 Mhz with 192 MB of RAM
02:18.26milkyflavalol
02:18.26jimmy_deanPBway under the minimal requirements
02:18.34jimmy_deanPByet it was useable
02:18.42jimmy_deanPBmost of the time
02:19.01jimmy_deanPBit was running, IAX, SIP and zaptel
02:19.10jimmy_deanPBI'm amazed it did as well as it did
02:19.17milkyflavahow many users
02:19.39jimmy_deanPB1-5
02:19.52rob0I had to give up my PII 400 because of the lack of PCI 2.1 or whatever. Didn't work with the TDM card.
02:20.04jimmy_deanPBmog_work, how many users for you?
02:20.15mog_workits just my house
02:20.21mog_worki think i have had 4 calls bridged on it
02:20.22jimmy_deanPBok, yeah :)
02:20.32mog_worki also use for apache
02:20.34mog_workjabber
02:20.40mog_workand samba file server
02:20.44mog_workits pretty busy box
02:20.47jimmy_deanPBwow, no kidding
02:20.48mog_workavg load is 1.4 or so
02:20.51jimmy_deanPBhehe
02:20.58jimmy_deanPBlinux?
02:21.01mog_workyeah
02:21.05jimmy_deanPBwhich distro?
02:21.06mog_workits a cobalt raq 2
02:21.09mog_workgot it for 50 bucks
02:21.11milkyflavaah
02:21.12mog_workdebian
02:21.13jimmy_deanPBnice
02:21.21mog_workits also firewall and ssh front end
02:21.24jimmy_deanPBdebian, the savior of the world :)
02:21.26mog_workdamn spiffy little box
02:21.31mog_workno dma on hd though
02:21.47mog_workso its not much fun to work on
02:22.42mog_worki got 128mb more ram for it today though so all is good
02:22.49jimmy_deanPBgood :)
02:22.56jimmy_deanPBit'll thank you
02:23.11milkyflavait also does the laundry and the dishes and can change out the main bearings in a 78 Ford?
02:23.39mog_workhmm /me thinks about hooking it up to dish washer and laundry over its serial port
02:23.47milkyflavalol
02:23.50milkyflavapost the pics
02:23.55mog_workwill do
02:23.56jimmy_deanPBharryvv, I do that sorta thing for a living :)
02:24.01mog_workits a pretty little box for what it is
02:24.05jimmy_deanPBoops
02:24.08jimmy_deanPBdang tab completion
02:24.16jimmy_deanPBha is what I meant
02:24.55mog_workyeah
02:25.41milkyflavajimmy_deanPB: have you made sure you tdm is on its own IRQ?
02:25.46*** join/#asterisk Nukemizer (n=Nuke@67.137.28.165)
02:25.58jimmy_deanPBmilkyflava, I haven't no, how do I make sure of that?
02:25.59litagethe * CLI has commands such as ``iax show registry'' and ``iax show peers''. why is there no ``h323 show registry'' or ``h323 show peers''?
02:26.02mog_workyou do linux home integration jimmy_deanPB?
02:26.11milkyflavacat /proc/interrupts should tell you
02:26.27jimmy_deanPBmog_work, no, device that communicate over RS232 and other busses
02:26.41jimmy_deanPBmilkyflava, ok, I'll try that
02:26.55jimmy_deanPBthere's a chance it could be shared...it's in a new dell dimension 3000 desktop box
02:26.58mog_workahh is there a washer i can get that has serial?
02:27.08milkyflavajimmy_deanPB: Like my little IR reciever for my remote controller for MythTV?
02:27.23jimmy_deanPByou could easily put your own serial transceiver on it
02:27.40mog_workreally?
02:27.41jimmy_deanPByou just need a max232 chip and the proper power
02:27.51milkyflavathe transceiver will both recieve and send, correct?
02:27.52jimmy_deanPBand a few caps and resistors
02:27.55jimmy_deanPByep
02:28.07mog_workwant to make me one?
02:28.09jimmy_deanPBruns at +12volts usually
02:28.10milkyflavalol
02:28.24jimmy_deanPBmog_work, google for "max232 projects" or something like that
02:28.31jimmy_deanPBshould come up with stuff
02:28.34mog_workokies
02:28.42mog_worki really need a washer /drier on my lan
02:28.47milkyflavalol
02:28.50jimmy_deanPBI haven't actually done that from scratch yet, but all the products we make at work use that chip
02:28.58jimmy_deanPBhaha
02:29.02milkyflavatry Pluto home automation
02:29.12jimmy_deanPBbut you can't get the clothes into it automatically
02:29.14jimmy_deanPBso what's the point?
02:29.23mog_workheh
02:29.30mog_worknah but i would want it to run at 3 in morning
02:29.36jimmy_deanPBthat you could do
02:29.41mog_workso i dont have to worry about no hot water in shower
02:29.46jimmy_deanPBmonitor water temp, level, ph level, etc
02:30.22jimmy_deanPBand if you press '3' on your * server, it'll start a new load :)
02:30.40jimmy_deanPBfrom your phone I mean
02:30.59milkyflavaget bluetooth proximity
02:31.10jimmy_deanPBnow you're talking!
02:31.22milkyflavaThats in Asterisk
02:31.23mog_workman that would be cool
02:31.34jimmy_deanPBwhoa, what does that do in *?
02:31.51milkyflavaIf it detects your bluetooth phone it rings the office
02:32.07milkyflavaif it doesnt then it rings your cell phone
02:32.23mog_worki dont have any bluetooth devices
02:32.42milkyflavayour cell doesnt have it or do you have one
02:32.54mog_worki dont have a cellphone
02:32.57mog_worki hate the damn things
02:33.02milkyflavaword
02:33.14mog_workmy favorite parts of the day are when i am completely out of reach
02:33.27milkyflavamine are when the batteries in the cell die
02:33.32mog_workheh
02:34.00Nukemizeris it possible to configure a PRI for 12 B channels ? If so, would I need to leave my D channel as 24 ?  thank you for any help
02:34.09*** join/#asterisk ahattar (n=kjsd@ool-435292d6.dyn.optonline.net)
02:34.12jimmy_deanPBmilkyflava, that's cool!
02:34.21mog_workyes and yes
02:34.26mog_workwell not ness. 24
02:34.31mog_workbut most telcos run d on 24
02:34.36mog_workeven if you dont have full span
02:34.41jimmy_deanPBso what, you hook up a bluetooth proximity device to your local desktop or laptop computer and that communicates with asterisk?
02:35.03Nukemizermog_work, thank you  . That helps
02:35.18milkyflavaI'm not sure, let me see if I can find the article for you.
02:35.24jimmy_deanPBcool
02:35.25jimmy_deanPBthanks
02:35.54ChujiIs walmart.com working for anyone?
02:36.03milkyflavahttp://mundy.org/blog/index.php?p=78
02:36.08milkyflavaPOW!
02:36.13jimmy_deanPBhehe
02:37.13milkyflavaThats a good site
02:37.18*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
02:37.25milkyflavaI'm going to eat and then battle the dragon of echo
02:37.30milkyflavayou all have a good night
02:38.27*** join/#asterisk froguz (n=froguz@244-142-222-201.adsl.terra.cl)
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02:40.26froguzwhy doesn't zaptel work if i install it by a bash script called by /etc/rc.local? (missing /dev/zap! on startup)
02:40.36litagedo commands such as ``h323 show registry'' or ``h323 show peers'' only appear in the asterisk CLI if you're using an h323 gate*keeper*?
02:40.49froguzdo i need tu add sudo to the install script commands?
02:41.10litagefroguz: by default, the init script will be run as root
02:41.13jimmy_deanPBfroguz, no, don't add sudo...you don't want to type your password everytime you run that
02:41.53froguzwhy doesn't works then?
02:43.01froguzbut it works if i install it by hand
02:43.26litagefroguz: "doesn't work" doesn't tell us anything about your problem
02:44.06froguzi think the problem could be the user, just root is permitted in my udev.permissions files
02:45.16froguzlitage, on linux startup, when zaptel is loading (waiting zaptel to come online) it says missing /dev/zap!
02:45.50litagefroguz: are you using udev?
02:46.28froguzbut i install it typing the same commands as root i have no problem, and it load lots of modules like ztdummy on startup
02:46.30froguzyep
02:46.32froguzudev
02:46.48litagefroguz: maybe udev hasn't created /dev/zap yet
02:46.49froguzCentos
02:47.10litagefroguz: i know nothing about zaptel though
02:48.02jimmy_deanPBfroguz, are you sure you did "modprobe zaptel" "modprobe wcfxs" modprobe "wctdm" if you are using a TDM card?
02:49.13*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
02:49.13*** topic/#asterisk is Asterisk 1.2.1 has been released! -//- http://www.asterisk.org
02:49.37froguzit supose zaptel do that for me, because i did a make config after make install
02:51.47froguzany clue?
02:52.10jimmy_deanPBI recommend doing a manual modprobe for them
02:52.48froguzi can't, because it complains about the missing /dev/zap device
02:53.20Qwellwhat commands do you run that makes it work?
02:54.04froguzdoing exactly the same zaptel install process, but writing directly
02:54.33froguzthe problem is just running the script
02:54.37Qwellwhat happens if after you reboot, you run /etc/init.d/zaptel start?
02:54.41froguzwich is called by rc.local
02:55.18froguzi haven't tried that
02:56.05froguzi would have to do that tomorrow
02:56.10marcus2why doesnt atacomm send any email confirmations of orders? grr
02:56.40froguzso, you think is not about the script user
02:56.47litagehow does chan_h323 compare to ooh323c?
02:58.34NukemizerI could use some zaptel assitance as well..  zaptel start will not work for me, but if I manually load wcte11xp I can get the module to load.  when running asterisk -vvvvcg 12 of my PRI B channels will register and then un register .. then "broken pipe" I do not know where to log for further trouble shooting
02:59.07QwellNukemizer: Did you edit the config file for the zaptel init script, to tell it you need wctc11xp?
02:59.48Qwellor, that must be just mandrake
02:59.49NukemizerQwell, box had been working for a month until i had to reboot tonight
02:59.58Nukemizerahh yes Mandrake it is
03:00.28NukemizerI take it Zaptell and Mandrake are a bad mix ?
03:00.30Qwellyeah, then either edit /etc/sysconfig/zaptel or /etc/init.d/zaptel and change the modules
03:00.37QwellMODULES=
03:01.12Qwellso, in your case
03:01.24QwellMODULES="zaptel wcte11xp"
03:01.51Nuggetzaptel and mandrake are fine.  zaptel and failing to read the documentation is apparently a real problem, though.
03:02.06Qwells/documentation/init script/
03:04.34NukemizerWell, one could say that. But I have had Digium support login to box prevoiusly and unsuccesfully to get modules to load properly.. I was just wishing that something  might have changed for me :( Thank you both I will read more
03:05.27NukemizerQwell, I am trying your suggestion. thank you for the help
03:07.22*** join/#asterisk toddf (n=toddf@ns0.fries.net)
03:08.28Flautoqwell, i got the problem fix
03:08.33Flautoed
03:09.20Flautoi got a script to solve the modprobe problem
03:09.53QwellFlauto: what was the problem?
03:10.16Flautoi had to modprobe everytime i start my computer
03:10.22QwellI know what the problem was
03:10.24Qwellwhat was the cause?
03:10.45Flautothere was not starting script
03:10.46*** join/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net)
03:10.53Flautoi friend of mine send me one
03:11.02Flautothen, that is it
03:13.11*** join/#asterisk franx (n=garra@215-78-246-201.adsl.terra.cl)
03:13.37trixterthe pens that digium gives out are nice, uniball
03:13.49file[laptop]yeah
03:13.53file[laptop]I used my pen all up :(
03:13.58file[laptop]I'll grab more next time
03:14.00trixterI got 30 of em today
03:14.23Qwell30?
03:14.26trixterfor the sacramento asterisk users group along with mouse pads, some marketing materials and screwdrivers
03:14.40trixtergive aways for the meet
03:14.44Qwellnot a day goes by where I don't use my digium screwdriver, heh
03:14.49Qwellor...at least...
03:14.56Qwellnot a day goes by where I don't think about stabbing somebody with it
03:15.05trixterwell it is reversable
03:15.12file[laptop]I need another mousepad
03:15.24file[laptop]I stacked up on them last time but they got packed :(
03:15.25NukemizerFlauto, I would be interested.. just got back from the server room and no luck thus far.. would you be willing to share script ?
03:15.26trixterI just prefer uniball pens so I was suprised they werent hte standard cheap bic ones
03:15.39file[laptop]but yeah, most excellent pens
03:22.28*** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net)
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03:24.45rezEditI am having an issue with ChanIsAvail(), which doesn't seem to be working when I use it to test a SIP extension for availability.  Anyone else having issues?  For me it always returns like the channel is free when in fact, it is not.
03:24.54rezEditI am using it like: exten => 100,n,ChanIsAvail(SIP/224|sj)
03:25.01rezEditand it always goes to n+1, never to n+101, even when the phone is on another call
03:25.14rezEditanyone else using that app?
03:25.21QwellrezEdit: svn trunk?
03:25.27rezEdit1.2.0
03:25.32QwellThe functionality of chanisavail and many others changed...
03:25.42Qwellthey don't jump to n+101 anymore by default
03:25.47Qwellshow application chanisavail
03:25.54rezEdityeah that's why the j is there
03:25.59Qwellright
03:26.40Qwellperhaps it doesn't work :P  Have you checked the returned variables?
03:27.02rezEdityeah
03:27.05Qwelland?
03:27.11rezEditone sec
03:27.22QwellI'd NoOp all three
03:27.55rezEditExecuting ChanIsAvail("SIP/227-26ef", "SIP/224|sj") in new stack
03:27.55rezEdit<PROTECTED>
03:28.17QwellSo what happens when you dial out with SIP/227?
03:28.21Qwellerm, 224
03:28.42litagehow does chan_h323 compare to ooh323c?
03:28.46rezEditto test this, I am calling 224 from 700
03:29.00rezEditthe calling an extension from 227 that tests 224 (which IS on a call to 700)
03:29.51rezEditI checked bugs.digium.com as well, doesn't seem to be anything there on it
03:30.40rezEditthat NoOp line is NoOp(${AVAILCHAN},${AVAILORIGCHAN},${AVAILSTATUS})
03:31.16rezEditahhh: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanIsAvail
03:31.27rezEditin the comments there is a list of states
03:32.06rezEdit<PROTECTED>
03:34.00distortionlitage: http://www.voip-info.org/wiki-Asterisk+H323+channels
03:34.12distortionthere's a comparison section on that page
03:34.37QwellrezEdit: yeah, it checks for <= 1
03:34.55QwellSo, if the device is in an unknown state...
03:35.13rezEdityeah.  unfortunately, that doesn't help me understand why it's unknown
03:35.42rezEdit* has all the info it needs to know that it's in use.
03:36.03QwellIs the device registered?
03:36.08rezEdityeah
03:36.22litagedistortion: yeah i've read that, but it's not much of a comparison
03:36.29QwellrezEdit: Is call limit set?
03:36.41rezEditQwell: Yeah, to 3
03:36.57Qwellpeer?
03:37.07rezEditQwell: I needed to fdo that to get 'sip show inuse all' to work
03:37.16rezEditQwell: friends
03:37.21Qwellso, yes
03:38.29asterboyAnyone using the Cisco 7920 wireless IP Phone
03:38.31asterboy???
03:39.18*** join/#asterisk mtupper (n=mtupper@pc-14-172-104-200.cm.vtr.net)
03:39.25distortionlast i checked it was not upgradable to sip, and the skinny support in * is a little light
03:41.17Qwelldistortion: it's a little skinnt
03:41.19Qwellskinny*
03:41.26Qwelland no, actually, sccp rocks my world
03:42.38denonSCCP on Asterisk rocks your world?
03:42.43Qwellit does
03:42.49denonis it really that cool yet?
03:42.50mtupperplease help on completing a call from * to PSTN through a SPA-3000!!!   I have googled it all over the place, tried dozens of configurations and I just can't get the PSTN line to dial-out.  I HAVE been able to complete calls from the Phone jack in the SPA-3000 to an x-ten softphone via SIP...  but the PSTN line is not having any of it!!!
03:42.52file[laptop]Qwell: that was bad
03:42.53Qwelldenon: for sure
03:42.55*** join/#asterisk znoG (n=gs@33-138-114-200.fibertel.com.ar)
03:42.58denonbetter than SIP?
03:42.59file[laptop]Qwell: therefore, I sentence you to... DEATH BY MUFFINS!
03:43.02Qwelldenon: yes
03:43.03Qwell:P
03:43.07denondetails, details
03:43.08QwellI don't like SIP though...
03:43.11denonoh well ..
03:43.15denonyou're one of those guys...
03:43.18QwellI converted my 7960 to sccp at home, and I've got a few at home
03:43.20Qwellat work...
03:43.31denonso you run OSX too, right? just because you hate microsoft. .
03:43.33Qwellnah, it works really well though.
03:43.50Qwelldenon: I wrote some realtime support for it...it's hot
03:44.01denonim not much of a realtime guy
03:44.03QwellI use the chan_sccp from berlios.de
03:44.04denonI like config files
03:44.15denonso what fun stuff does it do, that I cant do with sip?
03:44.18denon(mostly 7960s here)
03:44.34Qwelldevice messaging is cool
03:44.48Qwellcan they use hints with SIP?
03:44.53QwellYou can with sccp...
03:45.59Qwelland I think you can only do XML pushing with sccp...that's real nice
03:46.39rezEdithmmm.  'sip show registry' doesn't show anything....
03:46.41Qwellfile[laptop]: ha!  tcp only.  You're stuck on udp. ;]
03:46.46rezEdit'sip show peers' does though
03:46.49file[laptop]bah
03:46.53file[laptop]UDP over TCP
03:46.54file[laptop]HA!
03:46.56file[laptop]:P
03:46.57Qwelleh?
03:47.01Vcostuck on "teh udp"
03:47.58*** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net)
03:48.17Qwelloh, and...duh...the obvious
03:48.22Qwellyou can't use the addons with sip
03:49.00*** part/#asterisk mtupper (n=mtupper@pc-14-172-104-200.cm.vtr.net)
03:49.03*** join/#asterisk _daver_ (n=daver@ns1.tmok.com)
03:49.07Qwelland immediate dialplan checking...that
03:49.12Qwell's cool.  Just like zap
03:49.20Qwellor the iaxy
03:49.26rezEditQwell: regarding your earlier question about whether my devices were registered... shouldn't they show up in the list with the command 'sip show registry'?
03:49.34litagewhat's the difference between the 'host' and 'ip' settings for a friend/peer/user in sip.conf or iax.conf?
03:49.39QwellrezEdit: dunno, I don't use sip :p
03:49.48rezEditQwell: heh ok.
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03:53.39*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
03:53.52distortionIf i wanted to make sure that ulimit -c unlimited and ulimit -n 65535
03:53.58distortionerr
03:54.31distortionIf i wanted to make sure that ulimit -c unlimited and ulimit -n 65535 was added to the initscript for redhat, would i just add it to the /etc/init.d/asterisk file?
03:55.20*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
03:59.09*** join/#asterisk L|NUX (n=linux@202.5.145.58)
03:59.50SkramXHiya.
04:01.48litagedistortion: what's ulimit?
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04:08.14L|NUXlitage : ulimit is the command :)
04:08.15*** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn)
04:08.23L|NUXlitage : man ulimit
04:08.30Vcothe opposite of melimit
04:08.56L|NUXheh
04:09.10L|NUXulimit is the bash builtin command
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04:10.24*** mode/#asterisk [+o twisted] by ChanServ
04:10.34franxhi, what softphone for linux do you recommend?
04:10.55L|NUXfranx : sjphone
04:11.12L|NUXfranx : gnomemeeting or xten xlite :)
04:11.58_Vilego xten, less questions for this chan
04:12.18litageL|NUX: ah. don't have ulimit on my machine
04:13.09*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) [NETSPLIT VICTIM]
04:13.09*** join/#asterisk newl (n=newlook@203-59-61-62.dyn.iinet.net.au) [NETSPLIT VICTIM]
04:14.07franxL|NUX: thanks a lot
04:14.10*** join/#asterisk Tili (i=Tili@202-133-67-94-dialup.sat.net.pk)
04:14.50L|NUXfranx : np
04:15.03L|NUXlitage : humm
04:15.13L|NUXlitage : which distro you are using ?
04:15.28*** join/#asterisk CoaxD (i=coax@shell1.cornernet.com)
04:19.46franxis it possible to test asterisk without having to buy extra hardware?
04:20.24Qwellfranx: sure, got speakers and a mic?
04:24.16rob0litage: ulimit is a bash builtin
04:24.57rob0IOW, if you have bash, you have ulimit
04:25.23trixterfranx: yes if you have all the hardware you need
04:25.46trixterastlinux.org even has a vmware image for the free vmware player which lets you test without reformatting, they have a liveCD if you would rather that, and there are a few other lvie CDs out there
04:26.29trixterif you only want to play with call mnagers, do voip, etc you dont need hardware other than a system, perhaps the one you are using to get on irc perhaps another, if you want to interface with the pstn directly you may need hardware to do that if you dont already have it
04:27.23trixterso depending on what you mean by 'test asterisk out' the answer could go either way ...
04:31.38tessierLooks like quite a few of these things trade hands on ebay
04:31.47tessierI always see a bunch on there.
04:31.57tessierIt seems to have driven the price down a bit too.
04:32.31rob0tessier: how much are you wanting?
04:35.19tessierrob0: It's on ebay. No reserve. So...whatever I can get! :)
04:36.09rob0what do you think they will go for? Seems like most of this stuff costs more than I want to pay ... /dev/cheap0 :)
04:36.33tessier$200 or so probably
04:39.03rob05841378897 ?
04:39.15rob0haha the item number not the bid :)
04:39.27tessier5842276412 and 5842276731
04:39.34tessiercrap...I screwed up the title on the listing for my ata
04:42.50trixtercan you skype:5842276731@ebay.com ?  or do they not yet have that integration? :P
04:43.16*** join/#asterisk inv_arp (n=junya@c-66-176-211-109.hsd1.fl.comcast.net)
04:47.11shido6$10
04:47.12shido6?
04:49.33*** join/#asterisk ComputerWarm (n=workingg@66.244.235.210)
04:49.57ComputerWarmHello all question music on hold anyway i can get it to play music from a online radio station/
04:50.26asterboyyes
04:50.47asterboybut you will go to jail if you do.
04:50.49ComputerWarmasterboy sorry i should have finished that off. while they are in Queue?
04:51.01asterboy:P
04:51.12ComputerWarmnot with this radio station
04:51.15asterboylol
04:51.28asterboyI just say that cause its funny in here.
04:51.45SkramXComputerWarm: somepeople have weird senses of humor..
04:51.45ComputerWarmoh ok... anyway is there anyway to get it to play in queue?
04:51.54ComputerWarmSkramX i see that lol
04:52.17Sedoroxtessier: power supply with the ata?
04:52.42asterboysomeone sell a X100m!
04:53.00tessierSedorox: Yep
04:53.03loudmy cat peed my 7960.
04:53.09SkramXloud: lol
04:53.13Sedoroxloud: hopefully it wasn't a male cat...
04:53.19loudno, female
04:53.23loudfucking hate her now.
04:53.27Sedoroxah.. so it shouldn't smell for long :p
04:53.30Sedoroxif you cleaned it
04:53.47Sedoroxlol
04:53.49tessierMy male cat peed on my nice leather jacket
04:53.53tessierHad to have it dry cleaned
04:53.55Sedorox:/
04:54.04Sedoroxmy cat took a dump on my bed one time...
04:54.06tessierThen after it aired out for a while it was good as new.
04:54.14SedoroxI deserved it tho.. forgot to clean her box for a while....
04:54.21asterboyeasy fix...with a 22 cal. gum
04:54.26Sedoroxlol
04:54.37asterboysticks to the cat.= :P
04:54.58Sedoroxpersonally I still prefer cats over dogs...
04:55.34*** join/#asterisk Ayano (n=Ayano_@adsl-70-245-190-90.dsl.spfdmo.swbell.net)
04:55.37*** join/#asterisk SwK (n=SwK@12-219-151-128.client.mchsi.com)
04:56.16AyanoWhat is the web interface port for a polycomm 301?
04:59.12*** join/#asterisk toddf (n=toddf@ns0.fries.net)
05:00.26*** join/#asterisk EriSan (n=erisan@81-174-42-85.f5.ngi.it)
05:01.49SwKayano it should be port 80... but use the XML from a ftp or a tftp server instead you'll be a lot happier
05:03.40twistedSwK, fsck off :P
05:05.22litagei just installed ooh323c, and it said it installed several libraries into /usr/lib . how do you get asterisk to use ooh323c though?
05:05.30litage^--- http://rafb.net/paste/results/uZuIoU20.html
05:11.37*** join/#asterisk amir (n=amir@gentoo/developer/amir)
05:11.41*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
05:11.43SwKtwisted stfu
05:11.44SwK:P
05:11.55twistedwell, that only took 8 minutes
05:12.01SwKheh
05:12.12twistedSwK, i have most of my voice back
05:12.12*** join/#asterisk trixter (n=trixter@65.172.209.246)
05:12.14SwKhad to go find some beer
05:12.29twistedi'm guessing not talking hardly at all today helped
05:12.32SwKyeah
05:12.43SwKtalking yesterday probably didnt help much
05:12.47twistedno, probably not
05:13.09twistedbut i can talk now, as long as I don't try to get loud or high pitched, then it cuts out
05:13.32SwKtwisted add me to you msg list
05:13.36twistedlol
05:14.02SwKI don't accepting private messages or dialogs at this time. Please talk with me in the public channel. { Automessage }
05:14.04SwKheh
05:14.18twistedk
05:14.38twistedtook me a minute to remember the command
05:14.56Nuggetso clearly you should keep messaging so you're sure to notice when they do start accepting them.  :)
05:15.16twistedlol
05:17.08bsdfreakherh
05:19.41*** join/#asterisk inv_arp (n=junya@c-66-176-211-109.hsd1.fl.comcast.net)
05:21.12*** join/#asterisk Gamercjm (n=gamercjm@pool-70-104-145-143.lsanca.fios.verizon.net)
05:21.35GamercjmIs there a way to record calls?
05:21.46Nuggetyes.
05:22.16*** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)
05:22.23alephcomGreetings Everyone.
05:22.37sbingnergreetings earthlings
05:22.47Gamercjmi tried to use the *1 but would not work, just says not found..
05:22.49inv_arpGamercjm: monitor
05:22.53Nuggethttp://justfuckinggoogleit.com/?q=asterisk+record+calls
05:23.37sbingnerheh
05:24.17inv_arpwow domain actually exists
05:24.38tessierWish I had thought of that.
05:25.03Nuggetme too
05:25.20sbingnerthis is retarded, I can burn a DVD at 16x in 6 mins, but then it only can verify it at about 7x which takes like 12 mins
05:25.37SwKhah
05:25.43SwKthats lame
05:25.51bsdfreakyeh
05:26.21sbingneralthough it's been pretty reliable on this media I may just give up verifications heh
05:32.29*** join/#asterisk Assid (n=assid@203.115.64.62)
05:32.31Assidheya
05:32.52Assidanyone found any issues updating to 1.2.1 from an old CVS head?
05:33.07QwellAssid: modules still existing that shouldn't
05:33.22Qwellrm /usr/lib/asterisk/modules/*
05:33.27Qwellthen make install
05:33.40Assidahh.. so i gotta shut it down for a bit then
05:34.08Assidi thought i could do an update by just reloading
05:34.25*** join/#asterisk santiago (n=santiago@208.195.215.160)
05:34.34Assiddo i unload the zaptel ?
05:38.47*** join/#asterisk trixter (n=trixter@65.172.209.246)
05:39.46*** join/#asterisk enmaca (n=enmaca@dsl-201-129-118-166.prod-infinitum.com.mx)
05:41.33enmacahi all
05:44.59alephcomhi
05:45.34Rowteranyone has a Dual AMD® Opteron™ CPU with asterisk? I wonder if works ok as in Intel processors.
05:46.01mog_homeintels seem to work a little better
05:46.03mog_homebut not much
05:46.18MikeJ[Laptop]you got home in 2 min?
05:46.27mog_homelol
05:46.28mog_homeno
05:46.33MikeJ[Laptop]heh
05:46.35Rowtermog_home, you test it out?
05:46.41mog_homemy work machines switch doesnt always work
05:46.43litageafter compiling ooh323c, how do you generate chan_ooh323.so?
05:46.52mog_homewe have a few of everything in the lab rowter
05:47.01MikeJ[Laptop]hehe
05:47.08Rowtermog_home, excelent.. so its a safe choice?
05:47.15mog_homeyeah its fine
05:47.25mog_homeasterisk is not very cpu specific
05:47.30mog_homei run it on my mips box at home
05:47.42enmacahi have a sunfire v20z working for two weeks ago
05:47.44MikeJ[Laptop]yes... but it is anti windows!
05:47.47enmacaits working fine
05:48.07mog_homenice
05:48.28Rowtermog_home, thanks a lot
05:48.45mog_homeno problemo
05:51.37enmacaanyone have worked whit a sonus psx, gsx gateway?
05:51.47enmacaand *?
05:54.58Assidumm.. can someone paste me a copy of their musiconhold
05:55.10Assidit first said my copy is incompatible..
05:55.22Flauto<PROTECTED>
05:55.28mog_homeyou have your /usr/src/asterisk/configs/musiconhold.conf.sample
05:55.29Flautowhat is this message about
05:55.36Assidso i deleted my copy.. and ran a make install on asterisk again.. now it says Dec 14 00:53:58 WARNING[12318] res_musiconhold.c: No music on hold classes configured, disabling music on hold.
05:56.02Flautowould anyone tell me?
05:57.26*** join/#asterisk kusznir (n=kusznir@pool-70-110-34-187.sea.dsl-w.verizon.net)
05:57.36mog_homeit means flauto dnandnana peer is gone
05:57.51mog_homeaka it has lost contact with it for a set amount of time
05:58.11shido6http://cgi.ebay.com/Cisco-7960-IP-Phone-with-AC-Adapter_W0QQitemZ5841191851QQcategoryZ51204QQrdZ1QQcmdZViewItem
05:58.23Flautodo you mind if i send you pravite messages? mog
05:58.24shido6sonus!
05:58.26shido6LORD
05:58.31shido6reminds me of Global Crossing
05:58.34mog_homenope
05:58.46Flautothanks
05:58.59Qwellshido6: eww, sip
05:59.15*** join/#asterisk Umaro (n=umaro@68.142.142.105)
06:00.01Umarohey guys.. I have a tormenta 2 PCI card, and i'm told it should have a pci id of d00d.. but mine has a pci id of 1000. it's a 4 port T1/E1 card.. does anyone have experience with this?
06:02.13*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
06:06.35*** join/#asterisk s0lid (n=usahnem@58.69.102.22)
06:08.40sbingnerd00d.
06:09.00_Vilewha
06:09.12sbingnerheh
06:09.29Assidhrmm.. what do i do if the moh sound is too less
06:10.40_Vileturn your volume up a little?
06:10.58Assiderr.. its max..
06:11.11_Vilethere's a setting in *, sec
06:12.54_Vilehow are users connecting to your moh?
06:13.16_Vilerather, what kind of cards do you have in you * box?
06:13.20Assidwhenever they are put on hold.. OR.. i have a extension where i play it from
06:13.24Assidno cards..
06:13.29Assidall over ip
06:13.30_Vileall SIP?
06:13.38_Vileok
06:13.41_Vilesec
06:13.58Assidsip for phones.. iax for providers
06:14.14_Vilewhat version of * are you running?
06:14.44Assid1.2.1
06:15.11Assidshouldnt mpg123 do something?
06:15.36MikeJ[Laptop]who the hell uses mpg123 for moh
06:15.42MikeJ[Laptop]use native
06:15.51Assidnative huh?
06:16.49_Vileyou don't need mpg123
06:17.50Assidhow do i make it change? just override the settings inside the default context?
06:20.28AssidDec 14 01:19:58 WARNING[12580]: file.c:508 ast_openstream_full: File /var/lib/asterisk/personelmoh/londonl_auto does not exist in any format
06:20.38Assidits a .mp3 file..
06:21.07mog_homeman compiling all of asterisk on my box
06:21.10mog_hometakes foreva
06:21.36Qwellmog_home: distcc
06:21.57mog_homeyeah maybe
06:21.59mog_homenext time
06:22.07QwellYou'll never do it :p
06:22.25mog_homeheh yeah
06:22.32mog_homebut im not gonna compile it again on this box
06:22.36Qwellheh
06:22.38mog_homei just need to get gsm working for it
06:24.08Assidwow.. this is weird
06:24.23Assidmy show translation shows almost 1 ms more for most of my codecs
06:24.30Assidwhat happened
06:24.30QwellAssid: than?
06:24.39Assidthe last time.. using CVS head
06:24.40MikeJ[Laptop]Assid, use format_mp3, or even better, convert the file to raw or gsm or somthing better than mp3
06:24.51*** join/#asterisk edwar64896 (n=edwar648@220-244-62-6.static.tpgi.com.au)
06:25.36AssidMikeJ[Laptop]: just curious.. how does it help?
06:26.28drumkillaAssid: he's saying Asterisk doesn't know how to read an mp3 file by default
06:26.35drumkillaunless you install format_mp3 from asterisk-addons
06:26.46drumkillaand the other option is to just convert your mp3's into another format that asterisk can read
06:27.10Assidjust curious.. whats so bad abt mpg123?
06:27.34drumkillawell, that's another topic, but ...
06:27.36_Vilehm, slow
06:27.39drumkillait's just an external dependency
06:27.44_Vileand we can debate that for an hour
06:27.48drumkillawe have to execute an external process
06:27.55drumkillaand we depend on a specific version of it ...
06:28.24drumkillait's much more efficient to use files of a native format
06:28.28drumkillaand let asterisk read them directly
06:28.52_Vile.   /|\
06:29.10konfuzedive probably asked before but is gsm a non commercial license encumbered codec as in open source
06:29.16Assidokay .. so whats suggested? raw?
06:29.28_Vilegsm
06:29.53Assidgsm? hrmm.. is the transcoding overhead worth it?
06:30.01_Vileheh
06:30.02konfuzedgsm is lightest vis a vis high compression and cross telecom network compatible
06:30.05konfuzedis that right
06:30.28konfuzeds/vis a vis/via/
06:30.35*** part/#asterisk s0lid (n=usahnem@58.69.102.22)
06:30.41_Vilehttp://www.marko.net/asterisk/archives/0212/0350.html
06:30.43drumkillawell actually there isn't a best
06:30.50_Vilego there
06:30.54_Vilefollow directions.
06:31.00drumkillathe *best* is the same format as the other leg of the call
06:31.16drumkillathe *best* is to have it available in every format, and let asterisk pick the best one for you :)
06:31.25_Vilestop confusing him
06:31.28_Vile;)
06:31.33konfuzedok that works for mwe
06:31.37*** part/#asterisk Ayano (n=Ayano_@adsl-70-245-190-90.dsl.spfdmo.swbell.net)
06:31.42konfuzed;^)
06:31.45drumkillaconfusing?
06:31.51konfuzednot for me ;^)
06:32.39_Viles/confusing/konfusing
06:32.41Assidin that case.. ulaw would be the best for me.. coz thats what my incoming and outgoing stream is in.. EXCEPT for my personal SIP account, sinec i am remote
06:33.10_VileAssid, follow sox directions.
06:33.21konfuzedid like to do all IAX/gsm
06:33.33*** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com)
06:33.34konfuzedbut hey thats just me
06:34.39Assidoh crap.. upgrading broke my voicemail config
06:34.40AssidDec 14 01:34:21 WARNING[19236]: app_voicemail.c:2384 leave_voicemail: No entry in voicemail config file for '201'
06:34.56Assidim guessing i gotta use u201@context ?
06:34.57_Vileadd an entry for 201
06:35.06Assidthere already is.. in the ila context
06:35.09_Vilein the proper context.
06:35.26_Vileperhaps you must do an @context.
06:35.28_Vileyes.
06:36.06_Vilebeer break bbiam
06:38.20*** join/#asterisk edwar64896 (n=edwar648@220-244-62-6.static.tpgi.com.au)
06:38.26*** join/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com)
06:38.38*** part/#asterisk edwar64896 (n=edwar648@220-244-62-6.static.tpgi.com.au)
06:38.48Assidoay now to make it into files
06:39.09*** join/#asterisk edwar64896 (n=edwar648@220-244-62-6.static.tpgi.com.au)
06:41.10_Vileread sox, google: sox site:voip-info.org
06:41.17_Vileor
06:41.18_Vilesec
06:41.37*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
06:44.12Assidyeah i already have it
06:44.18Assidsox blah.mp3 blah.ul
06:44.47*** join/#asterisk Ayano (n=Ayano_@adsl-70-245-190-90.dsl.spfdmo.swbell.net)
06:44.52Qwellthey need to be mono 8hz
06:44.55Qwell8khz...
06:45.03_Vile;exten => metaswitch,1,Answer
06:45.03_Vile;exten => metaswitch,2,Playback(/tmp/asterisk-recording)
06:45.03_Vile;exten => metaswitch,3,Wait(2)
06:45.04_Vile;exten => metaswitch,4,Record(/tmp/asterisk-recording:gsm|2|20)
06:45.04_Vile;exten => metaswitch,5,Wait(2)
06:45.04_Vile;exten => metaswitch,6,Playback(/tmp/asterisk-recording)
06:45.05_Vile;exten => metaswitch,7,Wait(2)
06:45.07_Vile;exten => metaswitch,8,Goto(metaswitch,4)
06:45.30_Vilegood
06:45.52_Viledont include the extension on a Playback
06:46.43_Viledont include the ;'s either
06:46.52_Vilecomments
06:47.18AyanoI just got a used polycom 300, How do I configure the phone itself.  I have asterisk configured for it, now what?
06:47.26_Vileenjoy
06:47.38Assidhrmm.. still gotta use sox
06:48.30_VileAssid, no, only for conversion
06:48.31Assidyeah
06:48.36Assid-v 1.5 -c 2
06:48.39Assidshould do the trick
06:48.50_Vileyou got it then
06:48.57_VileAyano
06:49.21_VileWhat's the problem, 300<->*?
06:50.10AyanoI don't know where to put the asterisk ip, and user info.  I have used the cisco 7940 a lot, but this doesn't have a place to put it
06:50.12*** join/#asterisk edwar64896 (n=edwar648@220-244-62-6.static.tpgi.com.au)
06:50.19*** join/#asterisk L|NUX (n=linux@202.5.145.58)
06:50.34*** part/#asterisk edwar64896 (n=edwar648@220-244-62-6.static.tpgi.com.au)
06:50.36_Vilesip.conf
06:50.42_Vilein /etc/asterisk
06:51.05_Vilesec
06:51.14AyanoI have sip.confi configured.  How do i configure the phone itself?
06:51.48_Vilecan't help ya there
06:52.02_Viletell the phone to look at *
06:52.27Ayanoyea, that's what I can't find is where to put that
06:52.48_Vileexplain what 'options' you have
06:53.04_Vileshould allow you to put in an auth name
06:53.14_Vileand display name
06:53.14Ayanonone that have anything to do with a auth name or pass
06:53.15_Vileand a password
06:53.25_Vilethen your phone is probably locked
06:53.35_Viletry logging in as admin
06:53.46Ayanolol tried that
06:53.52_Vilelocked.
06:54.01_Vilewhere did you get your phone from?
06:54.28_Vileif you say eBay I will kick you in the shin
06:54.41mog_homeebay ^_^
06:55.07_Vilehide all you want
06:55.08_Vile;)
06:55.15Ayanono, it was from some dealer I think.  I don't know, its a friend of mine.
06:55.21_Vileok
06:55.34_Vilecan you SEE the display name?
06:55.34Ayanohave you used a polycom 300?
06:55.43Ayanono, it doesn't get that far
06:55.44*** join/#asterisk edwar64896 (n=edwar648@220-244-62-6.static.tpgi.com.au)
06:55.46_Vileno, never.
06:55.54_VileI know the functions.
06:56.07AyanoI didn't think so, your describing cisco stuff. lol
06:56.07_VileI use 7940's
06:56.10Assidman.. it sounds bad
06:56.28_Vileand 7960's
06:56.59_Vilebut, I suspect that you're locked out of any good config options -- mog please confirm?
06:57.37*** part/#asterisk edwar64896 (n=edwar648@220-244-62-6.static.tpgi.com.au)
06:58.25_VileAssid, what sounds bad?
06:59.21*** join/#asterisk edwar64896 (n=edwar648@220-244-62-6.static.tpgi.com.au)
06:59.24Assidthe moh
06:59.51_Vileexplain...
07:00.01Assidhrmm.. cracks
07:00.09Assidi think i will leave it on mp3 for now
07:01.03_Vilecracks?
07:01.39*** part/#asterisk EriSan (n=erisan@81-174-42-85.f5.ngi.it)
07:01.43_Vileexplain better
07:02.38_Vileis it consistent?
07:02.56_Vileor random
07:03.21_Vilebbiam, smoke break
07:03.27Assidconsistent
07:03.31_Vileok
07:03.39Assidmy bass just goes boom boom.. every few seconds
07:03.41_VileSIP?
07:03.55Assidim gonna try this locally on my asterisk box here.... to see what the issue is
07:03.59Assidyeah.. over sip
07:04.02_Vileok
07:04.04_Viletry local
07:04.08_Vilei'll bbiam
07:04.12Qwellusing the right kind of mp3?
07:04.21Qwell8khz mono
07:04.34_VileI tell everyone to use the right format
07:04.39_Vilenoone listens to me, I dunno
07:04.55Qwell_Vile: because you aren't saying it in 8khz mono
07:04.59Qwellduh
07:05.02_Vileok
07:05.05_VileAssid
07:05.13_Viletry 8khz Mono
07:05.24Qwellwith no id3
07:05.27_Vilesee what happens lemme know in 5 min
07:05.30_Vileyeah
07:05.31Qwellseems to help for whatever reason
07:05.33_Vileand with no id3
07:06.24mog_homeyay i have gsm compiled
07:06.24mog_homeunoptimized
07:06.24mog_homebut compiled all the same
07:06.24_Vileoptimize that
07:06.29*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
07:06.32mog_homemeh i dont need it
07:06.37mog_homeonly a channel of audio at any point
07:06.57_Vileim going to smoke
07:07.12_VileI need one, assid is busy
07:07.52_Vileinteresting, I allow 1024 appearances to my * server...
07:08.01Assidyeah let me finish this
07:08.04Assidgot a client one sec
07:08.07_Vileand I think 70 simultaneous
07:08.24_VileI should probably reconfigure the appearance list
07:08.26_Vilehm
07:08.27_Vileblah
07:08.28_Vilebbiam
07:09.54litagedo h323 users/friends register with a gateway or gatekeeper?
07:09.59Assidback
07:10.24Assid_Vile: i sox'd it from mp3 to ulaw
07:10.33Assidsox -c 1 blah.mp3 blah.ul
07:10.41Assidi also removed the mp3 files
07:11.00Assidso it only plays the ulaw ones.. but all i get is a FART outta the bass
07:11.19*** join/#asterisk bartpbx (n=bartpbx@p54B0411C.dip0.t-ipconnect.de)
07:11.30Assidhrmm.. i think i better go head for a shower
07:12.12litageAssid: after being farted at, a shower's a good idea
07:12.40Assidyep
07:12.52Ayanohow do you unlock a polycom ip 300?
07:13.00Assidunlock?
07:13.15Ayanoto get to the configuration
07:13.28Assiduse the username and password Polycom/456
07:13.32QwellAssid: You need to make it 8khz too :p
07:13.41Assidhrmm.. gotta find out how
07:15.05Qwell-r I think
07:15.11Qwell-r8000
07:16.09bartpbxrealtime voicemail does not seam to work with 1.2.1 anyone has the same problem?
07:16.40AssidQwell: will reload res_musiconhold.so   work aftr that?
07:18.04*** join/#asterisk lorinc (n=ang@caracas-2904.adsl.interware.hu)
07:19.09*** join/#asterisk koperniqs (n=koperniq@129.187.15.143)
07:19.11koperniqshi
07:22.26*** join/#asterisk |cleric| (n=dacleric@p5482A960.dip0.t-ipconnect.de)
07:22.27Assidbah.. i will do it locally.. remote is tough
07:24.54*** join/#asterisk trixter (n=trixter@65.172.209.246)
07:24.54*** part/#asterisk Ayano (n=Ayano_@adsl-70-245-190-90.dsl.spfdmo.swbell.net)
07:26.00Qwellbartpbx: yeah, hold on
07:26.45Umarodoes anyone here have a tormenta 2 PCI card? quad-T1?
07:26.54_VileQwell, stop what you are doing.
07:27.10bartpbxIt seams to we working in current branch 1.2 but not in the released 1.2.1
07:27.17Qwellbartpbx: http://bugs.digium.com/view.php?id=5960
07:27.25_VileQwell.
07:27.26Qwellbartpbx: yeah, it was fixed in the branch
07:27.54_Vileu done?
07:27.54bartpbxQwell, thanks. so I'll use the branch
07:28.04Qwell_Vile: sure
07:28.09_Vilebeer break man. you work too hard.
07:28.15Qwellbreak?  please
07:28.20_Vileshup
07:28.22_Viledrink
07:28.57Qwellwho says I'm not?
07:29.08_Vileyou type too fast
07:29.27_Vilebbiam
07:29.33_Vilesmoke
07:29.37Umarosome people practice typing fast with one hand while busy with the other, you know :P
07:29.55_Vilehaha, I can type with three hands
07:29.57edwar64896sox input.wav -r 8000 -c 1 output.ul
07:30.07_Vilebbiam
07:30.47kuku5anyone have echo problems on a pri ?
07:30.56bartpbxis there an easy way to find changes from 1.2.1  to the current branch?
07:31.19shido6im not touching your keyboard _Vile
07:31.51Qwellbartpbx: diff
07:32.20bartpbxok
07:32.35Umaroso, guys? no one here has a tormenta 2 PCI card? it looks like this card should have a pci id of d00d, but mine has an id of 1000
07:32.55mog_homenope umaro
07:32.57Qwelld00d, that's a 1337 card
07:33.06mog_homebut just change the subvender id in the driver
07:33.10mog_homeblamo it will probably work
07:33.14Umaroand tor2.ko won't load it
07:33.19mog_homeyeah it wouldnt
07:33.26Umaromog_home: tried that, couldn't find the id in the code yet
07:33.50mog_homelook for PCI_ANY_ID
07:33.51Qwellztd-eth.c
07:33.56mog_homein tor2.c
07:33.57Qwell0xd00d?
07:34.07Umaroztd-eth???
07:34.16Qwelldunno
07:34.19mog_homewhy qwell would it be there
07:34.27Qwellgot me.  That's what grep said
07:34.37Qwellztd-eth.c:#define ETH_P_ZTDETH  0xd00d
07:34.47mog_homeyikes i dont have zaptel
07:34.55mog_homeoh well i cant find it for you at the moment umaro
07:34.59QwellThat's the only place d00d is defined...will try -i
07:35.30Qwelltor2.mod.c?  tor2-hw.h?
07:35.40Qwelltor2-hw.h:      { PCI_VENDOR_ID_PLX, 0xD00D, PCI_ANY_ID, PCI_ANY_ID, 0, 0, (unsigned long)"Tormenta 2 Quad T1/PRI or E1/PRA" }, /* Tormenta 2 */
07:35.40Qwelltor2.mod.c:MODULE_ALIAS("pci:v000010B5d0000D00Dsv*sd*bc*sc*i*");
07:35.40Umarooic, tor2-hw.h
07:36.04Qwelltor2ee.c:#define EEPROM_MAGIC 0xD00dF00d
07:36.13QwellWho wants to bet they were incredibly hungry at the time?
07:36.48Umarowoot woot
07:37.14Umaronow let's hope the te210p works in this other box
07:37.23Umaroit should, supposedly it's just an alias to wct4xxp
07:37.58mog_homethey use a "unified" driver
07:40.36Umaroyay, te210P works too
07:40.38Umarotoday is a good day
07:40.45mog_homealways
07:40.52litagedo h323 users/friends register with a gateway or gatekeeper?
07:42.00_Vileqwell, why do you get the easy questions
07:42.26Qwellumm
07:42.35Qwellvolume?
07:42.38_Vilelit, gw
07:42.52_Vileh.323 hmm
07:42.56litage_Vile: are you sure?
07:43.05_Vilelit, no
07:43.16QwellIs there a difference between the two?
07:43.29litage_Vile: gnomemeeting has username and password options in its gatekeeper tab, but not its gateway tab
07:43.35litageQwell: yessum
07:44.02_Vilelit, use the same, connect ton the same ip
07:44.17litage_Vile: ?
07:44.24*** part/#asterisk edwar64896 (n=edwar648@220-244-62-6.static.tpgi.com.au)
07:44.34_Vileignore gnomeet's wording
07:44.35*** join/#asterisk edwar64896 (n=edwar648@220-244-62-6.static.tpgi.com.au)
07:44.41litagek
07:45.12*** part/#asterisk edwar64896 (n=edwar648@220-244-62-6.static.tpgi.com.au)
07:45.18*** join/#asterisk edwar64896 (n=edwar648@220-244-62-6.static.tpgi.com.au)
07:45.40litage_Vile: with chan_h323 loaded, * doesn't have commands like 'h323 show peers'
07:45.43_Vilenow, back to important stuff..
07:46.01_Vileuse sip
07:46.03Umaroh323.. need to play with that soon.. dreading that
07:46.29litage_Vile: i'm forced to use h323
07:46.32_Viledont use h323 unless you feel like getting a beating
07:46.44_Vileok
07:46.56_VileI defer to others  then
07:47.00litagedoh
07:47.05_Vileim a sip man
07:47.40_Vileany possibility for sip?
07:47.44litagenone
07:47.45_Vileactually
07:47.53_Vileexplain your hardware to me
07:48.35litage_Vile: not much at the moment. just a couple of asterisk boxes
07:48.53litage_Vile: but my carrier only supports h323
07:49.12_Vilelit, explain a sample route to me
07:49.16_Vilevia h323
07:49.20Umaroif I don't actually have a T1 hooked up to my digium card, but I want it to be used as a timer, do I need to put anything in zaptel.conf and zapata.conf, or do I just need to have the module loaded?
07:50.03_VileI'd the differ this question already
07:50.16litage_Vile: ?
07:50.24_Vilebut noone's awake?
07:50.57litagebob's softphone connects to *box1. joe's softphone connects to *box2. bob calls joe, so the call goes from bob to *box1 to *box2 to joe
07:51.00mog_homeset with a 0
07:51.03litage_Vile: is that what you meant?
07:51.07_VileIf noones awake, gimme errors, I can work erroes
07:51.11_Vileerrors
07:51.13mog_homeint zapata.conf
07:51.17mog_homeer zaptel.conf
07:51.24Umaromog_home: how do you mean?
07:51.28_Vileyes that's what I meant
07:51.31_Vilesec
07:51.35_Vilemog, you got this?
07:51.37Umaromog_home: set a span?
07:51.50mog_homeX,X,X,esf,b8zs first number is span, second number is timing source, third is build out
07:52.06steffhi all
07:52.06mog_homea 0 there meens card provides timing for the board
07:52.19mog_home1 means take timing from far end and replicate it on board
07:52.26mog_homeall that hardware can only have one source
07:52.30mog_homeer each card
07:52.38mog_homeso you can only have one 1
07:52.41mog_homebut many 0s
07:52.44Umaroright
07:52.45Umarocool
07:52.48mog_homeand on that note
07:52.50mog_homei sleep
07:52.56kuku5would any of those cause echo ?
07:52.58mog_homepeace we outa here
07:52.59_Vilemog
07:53.00*** part/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
07:53.05_Vileshiet
07:53.47_Vilelitage
07:54.56litage_Vile
07:54.59_Vilethat is what I meant
07:55.06_Vilethough...
07:55.24_Vileexplain the problem from boxX to me.
07:55.50_Vilepretend I'm bob.
07:56.33litage_Vile: the problem is that i can't get bob to register with *box1 using h323
08:02.00_VileI have to defer the question to someone who knows h323
08:02.27_Vile<PROTECTED>
08:02.37litagethanks for trying to help  :)
08:03.22_Vilegimme logs and I can maybe do more
08:03.45_Vileyw
08:06.13Qwellbed
08:07.33_VileQwell >>>> I can't answer litage's question
08:08.14_VileQwell <<<< going to bed, too.
08:08.40_Vilelitage, go to bugs.digium.com.
08:08.47*** join/#asterisk konfuzed (n=KonfuzeD@H135.C72.B0.tor.eicat.ca)
08:09.02*** join/#asterisk BadaBing (n=BadBug@c-24-61-4-191.hsd1.ma.comcast.net)
08:09.20BadaBinghello everyone
08:09.32*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
08:09.32_VileIf it is not found there, then submit a 'feature request'.
08:09.58BadaBing<PROTECTED>
08:10.08_VileDigium will take care of you from there.
08:10.14BadaBingsection 3.7 and now i get this error Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)
08:10.36BadaBingmp is working fine but it show that asterisk is stoped. when i call it works fine but can't connect to CLI
08:10.59BadaBingcan anyone help ?
08:11.53_Vilehow are you trying to connect
08:11.57_VileSIP?
08:12.01BadaBingasterisk -r
08:12.02_Vileor IAX?
08:12.17_Vileprotocol wise from one box to the other.....
08:12.53BadaBingEverything was working fine b4 I follow the wiki
08:13.03BadaBingI am using IAX and SIP
08:13.07_Vileignore then wiki
08:13.11_Viles/then/the
08:13.26BadaBingyeah but how can I fix this problem now
08:13.42_Vileexplain to me what you were doing before the new changes
08:13.47_Vilefirst
08:13.47BadaBingI follow section 3.7
08:13.54_Vileno, explain
08:14.00BadaBingokay
08:14.28BadaBingfirst I changed the the defult FOP Password
08:15.05BadaBingafter that i issued a amportal restart and was still fine
08:15.23BadaBingeverything was working fine...
08:15.56BadaBingthen "nano /usr/sbin/safe_asterisk " changed CONSOLE=yes to no.
08:16.03*** join/#asterisk JohnJacob (n=m00p@pcp0011543623pcs.mainf01.in.comcast.net)
08:16.27BadaBingI also uncomented the line "NOTIFY=your@emailadress.com" and placed my emailaddress
08:16.36_Vilewhat file
08:16.54BadaBing/usr/sbin/safe_asterisk
08:17.09_VileOK anything else?
08:17.32_Vilelook at what you changed, first.
08:17.39BadaBingthen "MACHINE='yourmachinename' " then changed it to MACHINE='asterisk''
08:17.42_Vilereverse what you changed.
08:18.04_VileI'll be here for a couple more minutes
08:18.13_Vilethe MACHINE= doesn't matter.
08:18.21BadaBingeven the amp password?
08:18.25_Vilenor does the NOTIFY
08:18.37_Vilechange back everything.
08:19.50BadaBingokay I changing them back
08:20.16BadaBingone problem though can't remeber MACHINE=`asterisk`what was it
08:20.32_Vileuse localhost.
08:20.39BadaBingokay
08:23.23BadaBing_Vile, okay everything is back the way it was
08:23.35_Vileok, now tell me your problems.
08:24.25_Vileone at a time.
08:24.26BadaBingwhen i issue asterisk -r
08:24.40BadaBingi get Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)
08:25.08testmachinehmmm
08:25.17_Vileasterisk -vvvvr, if you can't get that -- do a ps -auxww | grep ast
08:25.23testmachinei am bussy with some zyxel prestige 2002
08:25.31testmachinein asterisk
08:25.37_Vilelet me know if you get a response from the ps
08:26.04_Vileactually, just do a ps -auxww| grep ast and msg me what you get
08:26.05BadaBingokay got responce from ps
08:26.22_Vile"ps -auxww|grep ast"
08:26.33_Vilemsg me
08:26.34BadaBingokay
08:26.46_Viletest, explain.
08:27.21_Viletestmachine, explain.
08:29.07_Vileor not
08:29.12_VileI'll be in bed in 5m
08:30.56Assidhrmm
08:31.05Assidnote to self.. dont use CFLAGS when compiling asterisk
08:32.04*** join/#asterisk chapeaurouge (n=chap@85.201.81.201)
08:33.35testmachine_Vile:
08:33.35testmachineehm
08:33.58testmachinewith these things you can call over a vpn from server to server, and one has a modem poty
08:34.02testmachineport
08:34.06testmachinewich can call too the outside
08:41.15*** join/#asterisk trixter (n=trixter@65.172.209.246)
08:45.31litageis okay to run a gatekeeper (eg: gnugk) and asterisk on the same machine?
08:47.16*** join/#asterisk habakuk (n=chatzill@c-24-6-173-113.hsd1.ca.comcast.net)
08:49.27*** join/#asterisk psk (n=psk@golia.caltanet.it)
08:49.35*** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it)
08:49.36habakukhi can anyone enlighten me on Manager events? How do you "subscribe" to them?
08:56.40edwar64896events are set by manager interface by default unless you explicitly don't want them
08:57.17edwar64896you do this by setting "Events: Off" in the authentication clause.
08:57.48edwar64896http://www.voip-info.org/wiki-Asterisk+manager+API is a good place to look for this sort of stuff
09:00.00habakukedwar64896: thanks. I just saw that note there. Do you know if it's possible to subscribe only to certain types of events?
09:00.48edwar64896event docco on voip-info is a bit light on in this regard...
09:01.45*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no)
09:02.06*** join/#asterisk Gladius (n=hapchu@82.208.156.94)
09:02.15Gladiushello guys
09:02.40GladiusI have a problem with the queue agents. I use AgentCallbackLogin and it's just not working
09:03.05edwar64896habakuk: there are 7 types of events (include/asterisk/manager.h)
09:03.28*** join/#asterisk oej_ (n=Olle@apollo.webway.se)
09:03.30*** part/#asterisk telmich (i=telmich@gpm/telmich)
09:03.47edwar64896I think you subscribe using the read/write config lines in /etc/asterisk/manager.conf
09:04.12habakukedwar64896: hmm.. interesting. I'll have to take a further look at that
09:04.27edwar64896don't think that the subscription is via the API though.
09:04.58testmachinehm
09:05.17testmachinethese zyxel things are kinda hwphones
09:06.15habakukwell if a user with low privelages will only get a low level of events
09:06.57habakukthere is an eventmask you can set
09:08.23*** join/#asterisk darkskiez (n=darkskie@194.247.78.146)
09:08.38Gladiusis anybody here using agents ? (agents.conf)
09:13.37*** join/#asterisk tengulre (n=tengulre@222.90.66.4)
09:14.46tengulrehi,all!
09:14.51tengulreanybody active?
09:17.18moversure :-)
09:17.49moverGladius yes
09:19.23Gladiusmover, my problem is very weird.... I can see the agent as loged in, but on show queue, it's Unavailable
09:19.57Gladiusthing is, I'm using alphanumeric names for agents, not just digits
09:21.02*** join/#asterisk ptiggerdine (n=ptiggerd@c220-239-93-118.rochd1.qld.optusnet.com.au)
09:21.21ptiggerdinemISDN still has a few bugs :)
09:21.36moverGladius you have added the agents as members in queue.conf?
09:22.18*** join/#asterisk densin (n=x@203-144-187-18.static.asianet.co.th)
09:22.47Gladiusof course :)
09:22.59densinif I connect FXO to line which alread have phone, can it use for record ?
09:23.14densinhello all
09:23.14Gladiustest-agent   (test agent) available at '52' (musiconhold is 'default')
09:23.26GladiusAgent/test-agent (Unavailable) has taken no calls yet
09:26.53viperdudehi all
09:27.04Gladiushi viperdude
09:27.10densinif I connect FXO to line which alread have phone, can it use for record ? (I mean hardware capability , need to programming that ok )
09:27.49viperdudejust upgraded from 1.0.7 to 1.2.1 and found when a caller enters a queue it goes straight to MOH with out a welcome ot queue anoouncment, what do i need to enable it?
09:33.45*** join/#asterisk digime (n=digime@60.49.10.234)
09:40.01*** join/#asterisk rza (i=rza@todellisuus.net)
09:40.11rzaGood morning!
09:43.43*** join/#asterisk t0ke (n=t0ke@51.Red-83-46-136.dynamicIP.rima-tde.net)
09:43.43rzaim sorry for disturbing possibly stupid question, but can i use asterisk to control ericsson md110 ?
09:43.52*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
09:44.20rzaI got this client who wants a "Call this number" button to their intranet and I am lost with how should I initiate a call
09:44.47rzai dont mind paying for detailed help
09:45.01t0keanyone with fedora core 3 and TE410P?
09:45.21t0kewhen I do modprobe wct4xxp I receive this: ZT_CHANCONFIG failed on channel 97: No such device or address (6)
09:45.56*** join/#asterisk zagaya971 (n=almeli@APointe-a-Pitre-102-1-3-9.w81-248.abo.wanadoo.fr)
09:50.44*** join/#asterisk gnosys (n=ksford@ip68-9-201-250.ri.ri.cox.net)
09:50.56*** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com)
09:53.31t0keanyone with fc3 and TE410P?
09:54.37rzat0ke: i presume thats more of a fc issue than asterisk issue
09:55.23rzahave you tried asking #fedora or #linux?
09:55.59rzahttp://www.google.com/search?q=ZT_CHANCONFIG+failed+on+channel+97%3A+No+such+device+or+address+%286%29&sourceid=mozilla-search&start=0&start=0&ie=utf-8&oe=utf-8&client=firefox&rls=org.mozilla:en-US:unofficial
09:56.04rzagives quite many results
09:57.04*** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-85.claranet.co.uk)
09:57.19t0kerza..I looke a possible reason to that fail but I wanted to talk with anyone for to be sure ok?
09:58.49t0keproblem would to be jumpers setting about T1/E1 but I am not sure
10:09.35*** join/#asterisk fulgas (n=fulgas@213.58.130.46)
10:09.52*** join/#asterisk RoyK (n=roy@80.239.107.70)
10:10.47*** join/#asterisk cjk (n=cjk@80.92.64.103)
10:10.59cjkhi, is there any yada user inside? how stable is it?
10:15.26psirachi, i need help about asterfax, is there any asterfax user here ?
10:19.39*** join/#asterisk bangawanga (n=andhecke@ppp-82-135-84-226.mnet-online.de)
10:20.29bangawangahello, can anybody tell me how to listen the same call on more than one phone?
10:21.18*** part/#asterisk bangawanga (n=andhecke@ppp-82-135-84-226.mnet-online.de)
10:21.28*** join/#asterisk bangawanga (n=andhecke@ppp-82-135-84-226.mnet-online.de)
10:21.42bangawangais anybody on line?
10:22.23astcryzalways
10:23.14bangawangawe want to do sales training on our staff and i wonder how to listen to the same call on two phones. is it phone specific?
10:23.29bangawangasomething to configure in asterisk?
10:27.13bangawangano way to get an answer? silly question?
10:31.40bangawangai am new to irc and asterisk, some hints would be useful. i do not know where to start serching. i dont want to set up smoething like conferencing
10:32.07bangawangathe called party should not know that more than one person is listening
10:35.25*** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-85.claranet.co.uk)
10:38.39*** join/#asterisk areski (n=areski@polar.es6.egwn.net)
10:43.31edwar64896bangawanga? you still on?
10:43.38bangawangayes
10:43.45edwar64896Just saw your post.
10:43.58edwar64896<PROTECTED>
10:44.09rzabig butt
10:44.28edwar64896i like big.....
10:44.35edwar64896can you finish the song?
10:44.59edwar64896the bigg butt is that if you do, you _will_ segfault your box.
10:45.09edwar64896(assuming 1.2 svn trunk)
10:45.11rzabutts and i can not lie
10:45.20edwar64896;-)
10:45.35rzaedwar64896: did you see my post?
10:45.45edwar64896check 0005941 on bugs.digium.com
10:46.01edwar64896which has been the subject of much debate for the past week or soo.
10:46.20edwar64896soz - which one?
10:46.25edwar64896just got back on...
10:46.33rzaedwar64896: about md110 integration
10:46.51rzai can explain briefly
10:46.58edwar64896ta
10:47.04rzai need to make application which initiates call from number y to number x
10:47.16rzawith ericsson md110 and application link
10:47.23rzajust wondering if asterisk can help me here
10:49.08edwar64896so, if I get this straight, you want to tell the md110 to "originate" a call from an extension (y) to another number (x)
10:49.17edwar64896a bit like the "asterisk" origination command through AMI?
10:49.22rzayes
10:49.28rzausing csta standard
10:49.43rzaand im willing to pay for detailed help
10:51.45edwar64896does md110 fully support ECMA CSTA std?
10:51.51rzayes
10:51.59rzayou mean all phases ?
10:52.06rzaor just the standard in general?
10:52.36edwar64896where r u Timezone wise?
10:55.18edwar64896I am looking through the ECMA documentation ...
10:55.54edwar64896I think I am probably not the right person for this - no experience with CSTA and even less with the md110
10:56.19edwar64896If the md110 has a published API, it might be possible to interface to this from a client process and originate a call though...
10:56.48bangawangasorry, i was absent because telephone
10:56.54bangawangathank you for your time
10:56.55rzahmm
10:57.12rzaedwar64896: if it has published api
10:57.25rzabut it just says it conforms to ecma csta phase 1
10:57.58rzaim in gmt+2
10:58.36bangawangasegfaul my box? what do you mean?
10:59.16edwar64896banga: er... you box will fall over if you try chanspy one channel from two phones.
10:59.32edwar64896read 005941 for the gory details...
11:00.10edwar64896OK CSTA Phase one refers to Standard ECMA-180
11:00.36rzaedwar64896: i know, but i didnt find those standards very helpfull in implementation wise
11:01.19edwar64896It's all written in ASN.1 format,
11:01.29*** join/#asterisk mmmToop (n=chatzill@196.31.11.194)
11:01.40rzai know that also
11:01.46rzaASN.1 is scary
11:01.55rzaall im looking for is where to start
11:02.00rzawhat apis to use in linux
11:02.04rzaexample applications
11:02.06rzaso on
11:03.02edwar64896you would need to start looking somewhere else - the MD110 docco
11:03.13edwar64896(someone step in If I am getting it wrong...)
11:03.32edwar64896The standard is simply a standard, and if ericsson appear to offer support for this standard
11:03.43edwar64896then there must be some sort of implementation and/or interface to the implementation.
11:04.02edwar64896Your best start is to look for a developers kit for the md110 and identify what tools are provided.
11:04.19edwar64896You will then need to build an application and link in any development libraries provided by ericsson.
11:05.03edwar64896It might be possible that they supply a sockets-type interface, but that would assume your md110 can get itself an IP address.
11:08.41rzai got application link
11:08.49rzawhich listens to tcp/ip port
11:08.59rzaand i should be able to communicate that way
11:10.35edwar64896so you can telnet to your md110
11:12.09rzayes basically
11:12.14rzabut i shouldnt have to do that
11:12.27rzasec
11:12.38edwar64896http://cc.borland.com/Item.aspx?id=22836
11:12.43edwar64896interesting link...
11:12.51edwar64896refers to an intel API which probably does what you want.
11:13.02rzahttp://www.ericsson.com/enterprise/library/brochures_datasheets/MD110/MD110_1022528.pdf
11:15.04edwar64896looks like you need that media kit
11:15.14rzayeah
11:15.21rzathat explains the applicationlink
11:15.25rzawhich i already got
11:17.07mutilatoranyone have experience on a USR TC1000 v92 modem bank?
11:24.54*** join/#asterisk Delvar (n=irc@host-83-146-53-34.bulldogdsl.com)
11:30.33edwar64896Anyone on who has PRI expererince on E1 span?
11:32.53*** part/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com)
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11:35.23taecis there any way to display the original caller id, or at least flag the call in a certain way on a blind transfer?
11:36.00rzaedwar64896: theres no sdk for applicationlink
11:36.12rzatheres zero resources in developers section
11:36.18rzaat ericsson.com
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11:38.57edwar64896looks like p4 of that ericsson flyer has the key
11:39.21edwar64896"communication with the md110...... via tcp/ip over ethernet using ECMA-CSTA protocol"
11:39.37edwar64896time to brush off your ASN.1 ;-)
11:45.51rzaheh
11:45.57rzai was hoping to not go there
11:48.38docelmowhadup whadup?
11:49.31cjkis there a way to set the lastdata field in cdr?
11:50.46*** join/#asterisk dushyanthh (n=chatzill@203.199.114.33)
11:56.54docelmono not really...
11:57.16docelmoBut whatever you want to be there is the last command you need to call before hanging up the call
12:03.39cjkdocelm0: do you have an idea how i can easily add fields to the cdr?
12:12.46*** join/#asterisk dummie (n=dummie@202.155.89.122)
12:13.15dummiecan anyone help me how to set up phpagi? i'm very confused..
12:19.18docelmoyes..
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12:19.49docelmocjk, you have to add them to mysql then go into the cdr_mysql logger and put the information in for the fields your looking for
12:20.14docelmo«dummie» What do you wanna know? I am one of the developers for PHPAGI
12:22.56lehel`s gone..
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12:42.31akrotwhen i start asterisk with "asterisk -vvvgc" everything works fine. but if i start /etc/init.de/asterist start every call is "403 Forbidden" in xlite :/
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12:52.48PakiPenguinevening
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13:03.38dummiehi docelmo
13:03.57dummiedocelmo «dummie» What do you wanna know? I am one of the developers for PHPAGI
13:03.58docelmotook you long enough..
13:04.01RoyKFilesystem            Size  Used Avail Use% Mounted on
13:04.01RoyK/data/var             2.3T  111M  2.2T   1% /var
13:04.04RoyK:D
13:04.15docelmoroy thats sick dude
13:04.21dummiejust come back..
13:04.24docelmoHow mnay HD's do you have in that raid?
13:04.30RoyKdocelm0: just 6
13:04.41RoyK6x500gig in raid5
13:04.50dummiemay i ask you how to set up phpagi? i'm a little confused with that..
13:05.27dummiei just need to require('phpagi.php');
13:05.30dummiethat's all?
13:12.45docelmoI dont DCC chat..
13:12.52docelmonice dude
13:13.02docelmo500? How much did that set you back?
13:13.23docelmoone line needs to be: require('phpagi.php');
13:13.38docelmonext shoud be: $agi = NEW agi;
13:13.39rza<?php require('phpagi.php'); ?>
13:13.47docelmothats it. You didnt download the examples?
13:14.23rzawhats phpagi?
13:14.27taecis there any way to display the original caller id, or at least flag the call in a certain way on a blind transfer?
13:14.52dummieyes, i download it
13:15.02dummiebut it seems to stuck.. don't work
13:15.21dummiedoes it require to insert it to extensions?
13:16.00docelmoexten => s,1,AGI(script.php)
13:16.16*** join/#asterisk coppice (n=chatzill@82.194.17.210.dyn.pacific.net.hk)
13:16.40dummiemy apache document root is in /var/www/html
13:17.32dummieshould i place the script.php in /var/lib/sterisk/agi-bin ?
13:18.06docelmoyes
13:18.20docelmoPHPAGI has nothing to do with Apache just so you kno
13:18.22docelmoknow
13:18.56dummieoh.. ic.. then it seems i've misunderstood about it.. :D
13:19.29dummiethen can i run that script.php from web?
13:19.46docelmoNo..
13:19.51docelmoPHPAGI is ONLY for asterisk.
13:20.12docelmothe only output it produces is for asterisk to interperet. Not the web
13:20.36*** join/#asterisk santiago (n=santiago@208.195.215.160)
13:21.34dummieoh ic.. btw.. if i have to view the sip show peers from web.. can i do it?
13:22.21docelmoyes.. You would use PHP or something and do a $output = `/usr/sbin/asterisk -rx "sip show peers"`;
13:22.47docelmoor use the manager api of phpagi and do it that way
13:23.41dummie<<docelmo>> or use the manager api of phpagi and do it that way
13:24.14docelmoYou didnt read 1 of the docs did you?
13:24.14dummiecan i have a tutorial about it? :)
13:24.23docelmoum, no..
13:24.31docelmohttp://phpagi.sf.net
13:24.37docelmoclick on 2.0 documentation
13:25.18sivanais there a way to tranfer a call using a hook flash?
13:25.30docelmodepends on your ATA
13:25.33*** join/#asterisk Dead-Bum (n=Satan@tor/session/x-d0a5b60ea055d5fd)
13:30.38dummieshould i use the phpagi-asmanager.php?
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13:33.30docelmoyes..
13:33.53docelmoI am not going to go into great detail.. But yes.. Look at the documentation on the site.
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13:34.36dummieumm.. yes.. i will go go read about it
13:34.40dummiethanx. :D
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13:39.22znoGcrazy, i go into voicemail, press 0 then 4 to change my password and it sends me to change my temporary greeting
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13:43.15mutilatorW00T! the insurance company came through, not even a deductable to fix the car
13:43.30iCEBrkrHow'd you pull that off?
13:44.42mutilatori got sideswiped in a parking lot
13:44.45mutilatorand i wasn't around
13:44.58iCEBrkrand you didn't have to pay your deductable?
13:45.09mutilatorfix the while passenger side of the car from bumper to bumper
13:45.10iCEBrkrMerry Christmas
13:45.19mutilatoryea because of the michigan no fault law or something
13:45.23iCEBrkrhaha
13:45.27mutilatorit pays the deductible for me
13:45.40iCEBrkrFlorida is a 'no fault' state, but WTF does that mean when both people get cited for the accident?!?
13:45.48jimmy_deanPBhey [TK]D-Fender, I tried doing 1.2.1 here at work and now I can't get it to load anymore because of chan_zap.so
13:45.58mutilatori dunno
13:46.01mutilatori'm just super happy
13:46.13iCEBrkrmutilator: I would be
13:51.40znoGanyone use voipbuster?
13:51.58iDunnoOK - seriously - I need new eyes.
13:52.06iDunnoI just read voipbuster as vibrator :/
13:52.22znoGok then, anyone use a vibrator?
13:52.25znoGinteresting question all the same
13:52.27znoG:)
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13:53.14ctooleygood morning
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13:56.07[TK]D-Fenderjimmy_deanPB : You'll need to download Zaptel as well
13:56.13[TK]D-FenderAnddo a rebuild of it
13:56.42jimmy_deanPBI did
13:56.46jimmy_deanPBblew everything up
13:56.55jimmy_deanPBnow I'm in panic mode trying to go back to 1.0.7
13:58.44*** join/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net)
14:00.42parylg'morning all
14:00.58Dandangm
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14:04.08Theunihowdi
14:04.26docelmoiCEBrkr more or less in flordia everyone is gonna pay no matter the fault
14:04.36docelmobtw.. WHADUP #ASTERISK!
14:05.18TheuniWeird. When I connect calling from outside (isdn trunk) within asterisk he gets the music on hold. except when i dial the number i want to connect him to and then hangup. he gets the dialtone _but_ is connected after the third party picks up the phone. any idea?
14:06.09iCEBrkrdocelmo: yea, really.
14:06.25docelmokinda sucks.. This is why I avoid them..
14:06.39docelmoor try at least.. But people on the veterans.. geesh
14:06.43iCEBrkrdocelmo: When I got down here, I was debating getting insuranc for my motorcycle and the lady on the phone was saying how 20% of the motorists down here don't have insurance.  That's when I said "CUZ THEY CAN'T F'N AFFORD IT!" and I hung up
14:07.04docelmohaha
14:07.28iCEBrkrDude, the insurance on my motorcycle is what I pay on my Blazer back home.
14:07.37iCEBrkrOk,so it's a crotch-rocket, but still.
14:08.40parylwhen i first set up my te205p, someone helped me get the config straight and had me set pridialplan and prilocaldialplan to 'local' in both cases.  what do those parameters do?
14:09.23iCEBrkrdocelmo: Good news. I just realized during yet another moment of insomnia last night that I'll be able to test my system as soon as I get that Sangoma card.
14:09.23parylthe reason i ask is, my telco has been debugging a call that won't go through from our asterisk system, and they said this: "We see private networking messaging from the PBX going to the network which may be killing the call"
14:09.45docelmohaha
14:10.01docelmoI fell asleep at like 9pm on the couch
14:10.12iCEBrkrNow if I can easily configure that Sangoma card to use CAS is another story
14:10.13*** join/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net)
14:11.02iCEBrkrMan, I got insomnia bad. I can be dead tired and head to bed, then all of a sudden the brain starts thinking about what I gotta do tomorrow, what I wanna do tomorrow, what I gotta get done, which bills I have to pay, etc, etc, the list goes on and on
14:11.22docelmohehe.. fun stuff
14:11.30docelmoUm, I will stick with passing out..
14:11.39iCEBrkrI'd fix it with 2 beers, but there's something wrong with pounding two beers before bed at 2am
14:12.17*** join/#asterisk gpreche (n=gprenche@200-127-6-8.cab.prima.net.ar)
14:12.17docelmoum, ya you belong in Ybor
14:12.28iCEBrkruugh
14:12.39iCEBrkrI haven't been to a single club in Ybor since I've moved her.
14:12.41docelmoincase people are wondering we live about 20mi from each other
14:12.42iCEBrkr+e
14:12.59iCEBrkrI've been to Banana Joes at Channelside tho.
14:13.28docelmoits not bad.. I recommend going to ybor on friday.. around 12am
14:13.32iCEBrkrThe only times I've been to Ybor was on the motorcycle and that's when we all just loiter in the bank parking-lot across from Green Iquana
14:13.35docelmothats when it gets fun..
14:13.42docelmohehe
14:14.11docelmoJoe's is cool.. Im thinking about doing channel side for new years.. Dont know whats happening there tho.. Maybe disney.. they are supposed to have a huge new years bash
14:14.28iCEBrkrOh shit, I never thought about Disney for NYE
14:14.56*** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it)
14:14.59iCEBrkrMy roomie and I were thinking about head'n down to the Keys for NYE, but it'd be $200 a night on top of all the alochol + gas + food.
14:15.16gprecheDoes anybody know if a Dialogic D41/D 4 channels/analog ISA board can be used with Asterisk ?
14:15.26iCEBrkrgpreche: Nope.
14:16.11gnosysHey folks: polycom 501 phones... is the default firmware SIP software or something else (as with Cisco)?  Also, good source of documentation on configuring them?  Do they do they same thing as Cisco phones, searching tftp server for SIPDefault.cnf and SIPmacaddress.cnf?
14:16.20gprecheThanks: Nope meaning "no it wont work" or "not tried" ?
14:16.27parylmy telco has never heard of a PRI dial plan, or overlap dialing
14:16.30paryl:\
14:16.46*** join/#asterisk gambolputty (n=gambolpu@64.74.225.131)
14:16.49iCEBrkrgpreche: Haven't tried and I'm pretty sure they don't work.
14:17.12iCEBrkrparyl: they probably never heard of LCR either.
14:17.31gprecheMe too ... Just found a ton of these and had some hope ....
14:17.39iCEBrkrFound a ton?
14:17.52iCEBrkrat $3k a piece... Hrrm, hit-up eBay
14:18.07gprecheAsterisk does not yet support E1 CAS (MFC R2), right ?
14:18.29parylgnosys... that's a loaded question, but i've gone through it all with the 501
14:18.33iCEBrkrgpreche: I sure hope it supports CAS
14:18.59gnosyswhich question is loaded?
14:19.11docelmo«paryl» its in asterisk.. whats the exact issue?
14:19.52brad_msswgnosys: it depends on where you got the polycom from .... voipsupply ships sip firmware, others don't necessarily
14:20.11docelmo«gpreche» its based on card..
14:20.15docelmobut yes I believe it does
14:20.20gpreche<iCEBrkr>: These old cards ??!! That expensive ??!! We have just replaced several equipments and have a bunch of them
14:20.34paryldocelmo: when we try to dial one of our vendors, 3 toll-free numbers, 1 long distance, it goes onto the PSTN, waits about 10 seconds and comes back with hangupcause 34
14:20.48paryldocelmo: this is the ONLY number we've had issues with
14:20.56gnosysi got these from atacomm
14:21.07docelmohave you tried to diagnose the q.931 error?
14:21.12iCEBrkrgpreche: Seriously, go check out eBay.  I guess it depends on the model, but they're a pretty penny
14:21.15paryldocelmo: but our telco (a CLEC) sees it failing on the local telco's network
14:21.21gprechedocelmo, iCeBrkr: Thank you !
14:21.29docelmofor what?
14:21.46parylgnosys... look here: http://paryl.blogspot.com/2005/10/making-polycom-501-work-we_113034274646163867.html
14:21.59gprechedocelmo: For your answer ... that it supports MFC R2 ....
14:22.00paryli wrote that after screwing with them a lot
14:22.05gnosysthanks, paryl
14:22.10parylnp
14:22.22iCEBrkrgpreche: www.voip-info.org
14:22.23tzafrir_laptopif I want an incoming call from an analog line not to be Answer()-ed immedietly, is there any way to make Dial answer it if the call was actually made?
14:23.47gprecheAny idea if Asterisk would support an Aculab Primary rate E1 cas ISA card ? ((I know it supports the Prosody)) Today I have received old components and hope to be able to use it for anything ...
14:24.26iCEBrkrgpreche: www.voip-info.org
14:24.30*** join/#asterisk fugitivo (n=ajf@209.13.242.149)
14:25.11GladiusI have a problem with agents in queues
14:25.12tzafrir_laptopin short: any way to avoid the "auto-answer" on analog lines?
14:25.17GladiusI'm using AgentCallbackLogin
14:25.31[TK]D-Fendergnosys : Poly's don't ship with the firmware, you typically get it from your vendor (who gest it off Poly's private support site).
14:25.36Gladiusthe login is successfull, but the agent still apears Unavailable in queue
14:25.40gprecheiCEBrkr: thanks
14:25.46[TK]D-Fendergnosys : What version of BR & SIP are you running now?
14:26.05docelmo«Gladius» Did you add them to the Queue?
14:26.19docelmoMember => Agent/???
14:26.26Gladiusof course
14:26.43docelmoWell they will show unavailable untill they take a call
14:26.48Gladiusiuliana      (Iuliana) available at '34' (musiconhold is 'default')
14:26.57GladiusAgent/iuliana with penalty 5 (Unavailable) has taken no calls yet
14:27.03docelmook then whats the issue?
14:27.06docelmoohh
14:27.09gnosysthanks, [TK]D-Fender.  This one reported a version number 1.6.2.0041 (I think) in booting up, but not sure exactly what that's referring to.  From the boot messages, it seemed to have SIP software/firmware loaded.  I got it from atacomm.
14:27.16docelmoWell send her a call and try it..
14:27.21gnosyshow do i find out BR & SIP versions?
14:27.26docelmoWhich Im guessing you didnt being that your here
14:27.40gnosysparyl, at the end of that web page you sent me to, I see this: "The phone will automatically download the latest software and configurations and boot to working state. "
14:27.49[TK]D-Fendergnosys : They always come with SIP installed.  Now you just need the XML sample pack and you're ready to go.
14:27.56Gladiusdocelmo, I actually did call the hotline
14:28.03gnosysDoes that mean that the phone downloads the firmware from my ftp server or from a polycom server or what?
14:28.04Gladiusher extension works
14:28.04docelmoAnd the outcome?
14:28.15Gladiuscan't enter queue
14:28.25docelmothen check your settings for the queue
14:28.29parylso what i posted above... "We see private networking messaging from the PBX going to the network which may be killing the call", that's what the telco says.  they say their software identifies something coming from my asterisk switch as being 'private networking messages'
14:28.36docelmoI setup 6 yesterday and they worked off the bat..
14:28.42docelmowhat version * you using?
14:28.47iCEBrkrBLARRRGH
14:28.47[TK]D-Fendergnosys : Its means that the phone will auto-upgrade if you put a newer firmware on your boot server.  The boot server is your responsibility to set up, be it HTTP, FTP, or TFTP.
14:28.54GladiusI have 3 queues, but I'm only using sip/user
14:28.58Gladiusand they work
14:29.00[TK]D-FenderI use FTP personally and life is good.
14:29.02iCEBrkrmy Cablemodem at my apartment is lagged to shit and back.. 2000ms ping times
14:29.03Seldon1975all your base, your base, base base
14:29.14Gladiusbut the problem is that an agent is still called while he's in a call
14:29.18[TK]D-FenderSeldon1975 : you have no chance to survive.  Make your time.
14:29.23Seldon1975;)
14:29.33*** join/#asterisk klictel (n=klictel@207.107.208.137)
14:29.37Gladius1.2.0 bristuff
14:30.29parylgnosys: yes, that's what i outlined in the blog post... put the files on your ftp server, and the phones dl it at boot
14:30.29[TK]D-Fendergnosys : Click here to get the support pack - http://www.freedomphones.net/polycom/files/SoundPoint_IP_SIP_1_6_2.zip
14:30.43gnosys[TK]D-Fender: how do I discover BR and SIP versions?  Boot messages?
14:30.43[TK]D-Fendergnosys : what paryl said...
14:30.57docelmoiCEBrkr GET FIOS!
14:30.59gnosysthanks, guys.
14:31.21[TK]D-FenderYou just look through the phone's menu (menu/status/platform, menu/status/phone
14:31.28parylgnosys: i created a database and a python script to parse all of the params and create the config files on the ftp server... so now all i have to do is enter a new line in the table with the MAC, and boot the phone ;)
14:31.28docelmo~pastebin'
14:31.30docelmo~pastebin
14:31.32jboti guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/
14:31.38docelmosend your queues.conf and agents.conf there
14:31.41docelmolet me take a look at em
14:31.50gnosysVery nice, paryl!
14:32.04gnosysis it a mysql database or what?
14:32.08paryli think once you get a handle on it, you'll love how easy they are to set up
14:32.09[TK]D-FenderI ust copy and rename mine and change 3 lines!!! Whhe for MEE!
14:32.16iCEBrkrFIOS?
14:32.28docelmoVerizon Fios..
14:32.34docelmo30/5 internet connection
14:32.36iCEBrkrFibre?
14:32.41docelmowhat I have at home.. :P
14:32.50iCEBrkrYou think I'm made of money?
14:33.01docelmostatic-71-251-95-4.tampfl.fios.verizon.net
14:33.05docelmoya arent most?
14:33.07parylD-Fender: well, for all of my users, i've got different SIP params and whatnot.  also i create new directory.xml files for each phone based on the data in the table
14:33.09iCEBrkrPalm tress grow down here, not money trees :P
14:33.40iCEBrkrdocelmo: That's the sad thing, I make good money, and I'm still broke as a joke
14:34.01paryllol... 'broke as a joke'
14:34.03docelmodefine good money to the 10's of thousands..
14:34.27docelmoCause your idea and mine are probably way different..
14:34.29Gladiusdocelmo: http://pastebin.ca/33650 for queues.conf
14:34.42docelmofor instance I make around 70's.. and I like you am broke
14:34.48iCEBrkrdocelmo: I said good money, not great money
14:35.10[TK]D-Fenderparyl : Guess I'm lucky.  I have only 2 profile types for my users.
14:35.14iCEBrkrI should be making $70k/yr
14:35.21gnosysSo what's the general consensus on the preferred server to use with polycom phones?  ftp, http, tftp?
14:35.26Gladiusdocelmo: http://pastebin.ca/33652  for agents.conf
14:35.30iCEBrkrdocelmo: Doesn't matter, I'd still be broke living in a dumpy apartment
14:35.40docelmoYou couldnt put both in the same? geesh
14:35.50[TK]D-Fendergnosys : FTP  is the easiest and more secure than TFTP so it gets my vote...
14:36.01Gladius:)
14:36.30parylgnosys: yes, ftp
14:36.33docelmoWell I own a house on water front w/ 1 ACRE, 2 cars, a lot of technology goodies..
14:36.52[TK]D-FenderDamn, they completely depreciated the ASTDB stuff my dialplan used and now I have some serious re-writing to do :(
14:37.44paryliCEBrkr: do you just like it there?  otherwise... why stay?
14:37.48file[laptop][TK]D-Fender: yup, because we're evil
14:37.49docelmowhat is this: member => sip/horatiu,5 <---- ,5 what does that do?
14:38.10Gladiusdocelmo: it's the priority in the queue
14:38.14paryldocelmo: penalty
14:38.22docelmoahh
14:38.22iCEBrkrparyl: I can't say I totally enjoy it.  But I make a difference and I do see some sort of payoff.  I'm in a good position here to make decent money
14:38.26docelmobut be new
14:38.56iCEBrkrparyl: Like for instance, I think my boss is getting pushed up to administration, my goal is to fill his shoes.
14:38.56docelmoGladius, ok dude.. take out all of your member => sip/.... and change them to member => agent/...
14:39.01docelmothat should fix your issue
14:39.16Gladiusdocelmo, check the #2 queues :)
14:39.39Gladiusthe ones with sip/ are used now, and they work.... with the quirk I told you about
14:39.43docelmoMy goal is to have my bosses job.. VP Operations..
14:39.45Seldon1975can someone elaborate on these messages I got:
14:39.46Seldon1975== Spawn extension (default, t, 1) exited non-zero on 'SIP/226-e710<ZOMBIE>'
14:39.46Seldon1975<PROTECTED>
14:40.01docelmoYou got a internal server error
14:40.10docelmo:)
14:40.17Seldon1975theres noi device on my network with that IP
14:40.28GladiusI just upgraded today from 1.0.8 to 1.2.0
14:40.30Seldon1975does Asterisk think there is?
14:40.39Seldon1975maybe in sip.conf...
14:40.44Gladiusand it behaves the same way
14:41.03paryliCEBrkr: i'm in a privately owned company, lucked out really, and i'm as high as i can get in position, but whenever i feel like i'm not making enough, i normally get a raise because they're scared i'll leave ;)
14:41.21*** join/#asterisk Katty (n=angela@68.112.15.110)
14:41.24docelmohehe paryl same here..
14:41.32Gladiusme too :))
14:41.34sylewhat do you say to get that?
14:41.39docelmospeaking of which I have an evaluation tomorrow.. But I have the day off!!!!
14:41.40Kattyhi lads.
14:41.40iCEBrkrparyl: ew
14:41.43paryldocelmo: yay for being indispensible!
14:42.02iCEBrkrparyl: Lets just put it this way, I was only here 4 months, and I Got a $350 christmas bonus last year.
14:42.15docelmoI am IT Admin(only guy in building who knows all of the quirks of our network), Sr. Data Engineer and Telecom Engineer
14:42.18iCEBrkrparyl: Ya know, that's the only sigificant bonus I've EVER gotten?
14:42.48iCEBrkrI've NEVER gotten a christmas bounus.   Not to mention, I wasn't even here a full year and they still included me
14:42.49docelmoI got a 500 and a raise
14:42.50docelmo:)
14:43.02iCEBrkrI got run of the mill her, I can't complain
14:43.05iCEBrkr+e
14:43.06docelmoI was there for 4 months also..
14:43.51Gladiusdocelmo: do you know a way to stop * from calling an agent while he's already on a call ?
14:44.01docelmoHow long have you been with your comapny ice? I have been with mine since September before last
14:44.19docelmois the agent directly connect to that asterisk box?
14:44.31iCEBrkrdocelmo: Just a little over a year.. Maybe closer to a year and a half.
14:44.43Gladiusdocelmo: yes, using SJPhone (and it doesn't have a setting to take only one call)
14:44.46iCEBrkrdocelmo: I just moved here last Aug.
14:44.46docelmoCause my situation is different than yours.. I have agents in Aussie and tampa.. So my asterisk routes some calls via pstn or voip..
14:45.06iCEBrkrI used Asterisk to get this job :P
14:45.10docelmohehe
14:45.22iCEBrkrie. Picked up a Tampa phone number, used my buddy's Tampa address. :P
14:45.29iCEBrkrMeanwhile, the phone rang in Cleveland
14:45.32docelmoI fell into asterisk as the guy leaving knew it and I was hired cause I heard of it.
14:46.10paryliCEBrkr: before i got into my sitch, i was woo'd by a company in SF, and i went there to tour, put in my notice, etc.  so i took a month to train my replacement (out of the kindness of my heart) and at the end of the month he explained that there was no way he'd be able to do it.  at that point things started getting better because the president realize how important i am
14:46.22parylso i've gotten much better raises, etc
14:46.40docelmoGladius, I Dunno dude.. Sounds like you have a unique situation if they all register to the same box
14:46.54iCEBrkrparyl: Kinda sucks you gotta do things that way
14:46.55paryliCEBrkr: maybe that's what you need to do :)
14:47.15parylvery true, but sometimes the suits don't know who hold the real power in their company
14:47.26parylnot talking revolution or anything... they just don't get it
14:47.28iCEBrkrparyl: I've already sold my soul to this place.  We're changing technology and programming language.  My boss asked me to promise that I'd stick around and make things happen.
14:47.46DandreHello all,
14:47.53paryliCEBrkr: back when i put in my notice i used the following line:
14:48.33iCEBrkrparyl: and like I said, I'm in a good position here-- which also means I'm aware of the financials.
14:48.33DandreI have problem setting moh on my 1.2 installation:
14:48.33paryliCEBrkr: "it sucks to have to choose between eating and paying bills"
14:48.34iCEBrkrparyl: ie. they can't afford me.
14:48.34parylthe owner understood what he was doing at that point
14:48.34docelmoanywho.. off to do some home repairs.. Im out..
14:48.34iCEBrkrdocelmo: l8r
14:48.36paryliCEBrkr: ah
14:48.36iCEBrkrdocelmo: oh!
14:48.38iCEBrkrdocelmo: wait
14:48.42iCEBrkrdocelmo: Mon the 19th
14:48.47iCEBrkrSLUG meeting
14:48.57gnosyshi Katty, wha's up?
14:49.16docelmook will get with you on how the hell to get there
14:49.19iCEBrkrok
14:49.42paryliCEBrkr: automate business processes so that the company saves money.  softcosts are IT paydirt :)
14:49.49iCEBrkrparyl: That's the idea.
14:49.59iCEBrkrparyl: Apparently, there's only a handful of us that understand this
14:50.06iCEBrkrOut with the old, in with the new
14:50.11parylindeed
14:50.14Dandrehttp://pastebin.ca/33654
14:50.56parylDandre: that happens if the box is underpowered
14:51.04paryl(not the ONLY cause, mindyou)
14:51.32DandreThe moh doesn't play at all
14:52.17DandreI don't think it is underpowerd because my other setup with 1.0.7 is same architecture
14:52.20parylDandre: yeah... it'll either play very choppy or not play at all
14:52.40parylDandre: you could try doing a rawplayer type setup
14:52.52parylof course, you'll have to convert the mp3s to raw
14:53.06parylbut i've seen it improve performance quite a bit
14:53.23KriS83Anyone here using Asterisk 1.2.0 with Junghanns cards?
14:53.36DandreI don't understand why moh stops just after starting
14:53.37GladiusKriS83: me
14:53.55KriS83Gladius, may I query you for 2 mins?
14:54.08*** join/#asterisk DaPrivateer (i=Privatee@CRIMSON.OFF-HOURS.COM)
14:54.10*** join/#asterisk trixter_ (n=trixter@65.172.209.246)
14:54.15iCEBrkrdial bitch
14:54.17iCEBrkrWTF?
14:54.28DandreI don't have any zap channel in my setup is that the reason?
14:54.31GladiusKriS83: better talk here
14:54.36KriS83k
14:55.17paryldo changes to the PRI settings in zapata.conf require a total restart of asterisk, or can i just reload a portion?
14:55.36KriS83Gladius, may I ask where you are from? The problem I have is that even when setting Set(CALLERID(name)="") the number is passed to the party being called
14:56.04GladiusKriS83: it can be a "feature" of your telco
14:56.09GladiusI'm from Romania
14:56.31file[laptop]KriS83: you're setting the name, not the number to nothing
14:56.31[TK]D-Fenderparyl : at the minimum you'll need to reload zaptel
14:56.32iCEBrkrahh there we go
14:56.53iCEBrkrOh sure, now he doesn't answer
14:56.54KriS83Well with our normal Agfeo PBX the number is set to anonymous correct...
14:57.11KriS83file, I thought for the name it would be CALLERID(name)
14:57.19file[laptop]it is... that's what you said
14:57.20KriS83ops
14:57.26Gladiusexten => s,1,SetVar(CALLERIDNUM=212312)
14:57.29KriS83sorry.. above I mean number
14:57.32KriS83not name :)
14:57.35file[laptop]Gladius: you can't do that
14:57.46Gladiusfile[laptop]: wanna bet ? :D
14:57.50file[laptop]and setvar is deprecated
14:57.54file[laptop]it's set now
14:57.55parylDandre: have you set up an extension to go straight to MOH?
14:58.13file[laptop]Gladius: yes I'll bet, because the callerid stuff isn't a normal variable
14:58.51Gladiusfile[laptop]: it's been working for me for the last months
14:59.04file[laptop]are you sure?
14:59.10KriS83But as said above with the normall ISDN PBX I can do calls without sending the number..
14:59.16file[laptop]because there's a reason there was functions to change the callerid
14:59.22KriS83with * it always sends a number :(
14:59.22file[laptop]er applications
14:59.26file[laptop]which have now become functions
14:59.39Gladiusfile[laptop]: yes, but on 1.0.8
14:59.51KriS83Gladius, I was asking for 1.2.0 though :)
14:59.52file[laptop]using set to change the variable does it as a variable on the channel, using the application or function actually changes it on the channel
14:59.52GladiusI've just upgraded today to 1.2.0 and didn't test it yet
15:00.18DandreNo I use the hold from another extension
15:00.22Gladiusso, what's the "proper" way to do it in 1.2.0 ?
15:00.45file[laptop]Gladius: you use dialplan functions with set
15:00.47KriS83exten,s,3,Set(CALLERID(number)=12345)
15:01.07parylwhich echo canceller do you guys prefer for sound quality?  i've been using MARK2 and the person on the PSTn always gets broken up
15:02.07KriS83Is there some dial flag to override the set callerid?
15:03.04Kattyit's a beautiful day in the neighborhood, a beautiful day for a neighbor
15:03.45Dandreparyl: I have added a MOH Test extension and same result
15:04.32[TK]D-Fenderparyl : I find the A104d echo canceller to be the next best thing since sliced bread :)
15:04.45KattyiDunno: you, sir, have insaned.
15:05.02iDunnoKatty: I have not! I was always this way!
15:05.02paryl[TK]D-Fender:     >:-|
15:05.11KattyiDunno: oh ah, i see.
15:09.08iDunnosometimes I'm slightly less insane, but that doesn't generally last very long :)
15:10.21*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)
15:10.59iDunnoKatty: I think I can win on the bored stakes, I'm reinjecting mail in to an exim mail spool
15:11.20iDunno(or rather, at the moment I'm waiting for the queue to disolve a bit so that I can reinject some more)
15:11.49*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
15:11.49*** mode/#asterisk [+o anthm] by ChanServ
15:12.44KattyiDunno: i see.
15:13.16anthmhello
15:13.36*** join/#asterisk tRSS (n=tRSS@202.174.142.2)
15:17.11KattyHmmhesays: come on man, it's not even 10 yet
15:17.25Hmmhesaysred bulls were 3 for 5 bucks today
15:17.36Kattyeww, red bull has taurine
15:17.42Kattydo you know what you're drinking?
15:17.46synthetiqwhats wrong with taurine
15:17.48Hmmhesaysbull piss?
15:17.51Kattyye
15:17.59Hmmhesayscrushed baby rabbits?
15:17.59Kattythat's what it is.
15:18.07Kattymade from the bile of oxen!
15:18.13Hmmhesaysmmm good stuff
15:18.20Kattyyou, sir, have insaned.
15:18.33Kattylook at what that stuff is doing to you!
15:18.43Hmmhesayscan't be any worse than that hundreds of cigarettes and gallons of beer I drink every month
15:18.47Kattynext you'll be riding that bike in the ice
15:18.57Kattylike some sort of rebel.
15:19.09synthetiqyea i makes u give attention to katty  which is not allowed in this channel
15:19.10KattyHmmhesays: you should stop smoking
15:19.43Hmmhesaysthe bassist brought over his ibanez rg  guitar ot practice a couple days ago,  I felt like an 80's hair band rockstar
15:20.01*** join/#asterisk coppice (n=chatzill@82.194.17.210.dyn.pacific.net.hk)
15:20.23KattyHmmhesays: you practically are.
15:20.30KattyHmmhesays: except you're too young, and don't have enough hair yet
15:20.42Hmmhesaysi'm going to keep it at this level of spikedness
15:21.18*** join/#asterisk trym (n=trym@062016209171.customer.alfanett.no)
15:21.24Hmmhesaysi remember this summer after ribfest like 10 people came and asked me if I was joan jett's guitarist,  he had the same hair
15:21.31trymis the newest version in the svn rep 1.2.1 ?
15:21.44KattyHmmhesays: haha.
15:21.57KattyHmmhesays: maybe that's why there were following you around at the bar last time ;)
15:22.03*** join/#asterisk Spla4t1 (n=splat1@rrcs-24-172-35-197.midsouth.biz.rr.com)
15:22.06KattyHmmhesays: s/there/they/
15:22.16Hmmhesayslol
15:22.42KattyHmmhesays: of course i was the only SANE one that decided to remain sober
15:22.59HmmhesaysLOL
15:23.14*** join/#asterisk kpettit (n=keith@69.15.174.114)
15:23.18Hmmhesayswe need karaoke at the next cluecon
15:23.24Kattyoh god.
15:23.32Hmmhesaysor open mic night
15:23.39synthetiqcoed naked karokee
15:23.50Hmmhesaysi'd be the nights entertainment
15:23.56KattyHmmhesays: you would.
15:24.01KattyHmmhesays: i'd laugh at you allllll night long.
15:24.10Hmmhesayslol
15:25.01Hmmhesays"this is my artistic interpretation of 80's glam rock as played on a mandolin, thank you"
15:25.12*** join/#asterisk aldem (n=ww@80.237.176.16)
15:26.06*** join/#asterisk ian_k (n=ian@gateway.digium.com)
15:26.40*** join/#asterisk zpn (n=xpn@gateway.digium.com)
15:26.56[TK]D-FenderHmmhesays : My latest toy : http://www.bosscorp.co.jp/products/en/GT-8/index.html
15:27.05*** part/#asterisk zpn (n=xpn@gateway.digium.com)
15:27.27Hmmhesaysdamn you [TK]D-Fender
15:27.29*** join/#asterisk diego_br (n=Diego@200.208.241.178)
15:27.39Hmmhesaysi'm torn between that and a podxt live
15:28.02Kattyspeaking of toys.
15:28.06Kattyi need a new moocow.
15:28.11[TK]D-FenderPod is a great amp/speaker emulator, but they don't have an all-in-one that reeally shines.
15:28.24Hmmhesaysthe xt live does
15:28.33*** join/#asterisk Abbas (i=Abbas@203.81.194.195)
15:28.41Hmmhesays30+ amp models and 80+ effects
15:28.42[TK]D-FenderBehringer's VAMP makes Pod's smaller stuff obsolete at a fraction on the price.
15:28.53KattyHmmhesays: find me one.
15:28.56[TK]D-FenderThe GT-8 replaced my defective Boss VF-1 half-rack
15:29.01KattyHmmhesays: a cute one :>
15:29.12Hmmhesayswe run behringer power amps for our tops
15:29.23[TK]D-FenderGerman audio = good :)
15:29.30Hmmhesaysgod priced audio
15:29.39[TK]D-FenderI need to replace my dead Samson wireless rig and I'm set...
15:29.52Hmmhesaysi need to go wireless
15:29.58Hmmhesaysi hate being tethered
15:30.05KattyHmmhesays: but you're so cute tethered.
15:30.06[TK]D-Fender(andget the electical bits on my Dean fixed up)
15:30.15[TK]D-Fenderrawr ;)
15:30.36Hmmhesaysbut if I go wireless instrument I gotta go  wireless monitors also
15:31.09[TK]D-FenderFor monitors?  Power requirement sucks.. not worth the extra lag
15:31.42Hmmhesaysyeah but its hard to play when you can't hear it
15:31.54Hmmhesaysespecially for me who likes to spend a fair amount of time off stage
15:32.01[TK]D-FenderNo I mean monitors are good, just frget the wireless... that set stage gear
15:32.11[TK]D-FenderI miss the thrill of the spolight....
15:32.50Hmmhesaysi love the thrill of mingling with the audience playing some kickass riff
15:32.59Hmmhesays"you give love a bad name" comes to mind
15:33.10Kattyeww, mingling.
15:33.16*** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br)
15:33.24KattyHmmhesays: that's dirty :<
15:33.29parylhow do you set the rxgain/txgain from the CLI?
15:33.34parylis it possible?
15:33.49[TK]D-FenderI played with a Van Halen tribute band (KB / guitar), and did tons of Bon Jovi, Warrant, Whitesnake, Poison, and so on...
15:33.52Kattyparyl: i've never done it......cause i think you have to stop asterisk and start it again for changes to take effect.
15:34.30paryl:\
15:34.59[TK]D-Fenderparyl : you need to reload ZAptel in CLI  if you don't want a total restart
15:35.06Hmmhesaysthe 80's stuff is a lot of fun once you've leared it.
15:35.17Hmmhesayskiller to learn though especially if you play lead
15:35.19Dandreparyl: Do you know how should I convert from mp3 to raw files?
15:35.36[TK]D-FenderI love my 80's.......
15:35.51Hmmhesaysmy rig right now consists of a mesa boogie, and my les paul
15:36.13Hmmhesayssolid body, dual humbuckers and a tube amp
15:36.27*** join/#asterisk shanky (i=jramirez@217.11.114.145)
15:36.31parylDandre: yes!
15:36.32[TK]D-FenderMy problem is I am hooked on 24 fret w/ floating trems.  Makes my choices more limited...
15:36.34shankyhi, good afternoon
15:36.35paryl:)
15:36.42*** join/#asterisk negatendo (n=negatend@c-24-9-136-152.hsd1.co.comcast.net)
15:36.43Dandre:-)
15:36.50Dandrewould you tell me how?
15:37.28parylDandre: you need sox and lame
15:37.31parylhttp://pastebin.ca/33661
15:37.51[TK]D-FenderHmmhesays : Here's some ancient junk of mine http://www.geocities.com/tk_dfender/
15:37.52paryli wrote that to help automate the task... i put the mp3 file in the same directory as that script
15:38.10paryland use something like ./covert music
15:38.23parylit converts and puts the file in the moh directory
15:39.10DandreI don't have lame :-(
15:39.22Kattysometimes i don't have a clue what you guys are talking about.
15:39.27*** join/#asterisk negatendo (n=negatend@c-24-9-136-152.hsd1.co.comcast.net)
15:40.06parylDandre: what distro?
15:40.16Dandredebian sarge
15:40.26Hmmhesayscool
15:40.52Kattyyay for debian!
15:41.03iDunnodebian++
15:41.22Kattyyay for people helping you set up asterisk too ;)
15:41.23paryljust download the src and compile it
15:41.27parylit take 2 minutes
15:41.35paryltakes*
15:42.04parylKatty: boo-yah
15:42.42Kattyhuh?
15:42.48Kattythat does not parse.
15:43.19parylerm... i agree
15:43.20*** join/#asterisk digime (n=digime@60.49.10.234)
15:43.54parylhttp://www.urbandictionary.com/define.php?term=booyah
15:44.29Kattyclinching fist and thrusting their elbow downward vertically ?!
15:44.35Kattythat sounds painful.
15:44.45parylonly the first time
15:44.49Gladiusis there a way to reload queues without restart ?
15:44.53testmachinedebian is the future :P
15:44.54parylafter the nerve damage it really isn't so bad
15:45.02parylviva la debian!
15:45.11testmachine:)
15:45.11Kattywhy can't people just yay?
15:45.23testmachinebecouse i dont like the word yay
15:45.24Gladiusyay
15:45.24parylbecause then you would /get it/
15:45.24testmachine:P
15:45.51*** join/#asterisk darkskiez (n=darkskie@194.247.78.146)
15:46.39wunderkinGladius, reload app_queue.so
15:46.56Gladiuswunderkin, thanks
15:47.36Gladiusguys, do you know what is the Internet for ?
15:47.37iDunnohow about hoorah!
15:47.49iDunnoGladius: depends on who you believe...
15:48.03iDunnoGladius: if you believe the academics, then it's for sharing information
15:48.22Gladiuswell, these guys know for sure (and I kinda agree with them) that it's for p0rn :)
15:48.25Gladiushttp://www.youtube.com/watch_fullscreen?video_id=lr_HR-iIlYg&l=196&fs=1&title=WOW%20meets%20Porn
15:48.28iDunnoGladius: if you believe the marketting peeps it's to "get your company known, and provide another place to do business"
15:48.41iDunnoGladius: and if you believe the porn industry, it's all about the porn.
15:49.04*** join/#asterisk GXTi (n=scuba@freenode/developer/GXTi)
15:49.20*** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com)
15:49.24GladiusIt's all about the Pentiums, baby :)
15:49.29Gladiusdo you know this song ?
15:49.34parylseriously... what would the internet be without porn?  think about it.
15:49.55shankyhow could I know how many calls (sip or Zap ones) is happening in a asterisk? is there any command for the CLI?
15:50.00Gladiuswatch the movie :)
15:50.11Gladiusshanky: show channels
15:50.38iDunnoparyl: be useful again?
15:51.02Kattyparyl: uhh
15:51.04Gladiuswhat would the internet be without spam ?
15:51.06Kattyparyl: that's cause i AM a girl
15:51.14shankyGladius: thanks
15:51.19paryloh noes
15:51.20iDunnoGladius: less damned annoying and processor intensive?
15:51.27Dandreparyl: I haven't found the sources for lame do you have a pointer?
15:51.29parylthere's a girl in the house
15:51.36parylDandre: one sec
15:51.36Kattyoh grow up
15:51.40Gladius:))
15:51.47_Sam--f'n teliax
15:51.52Gladiusparyl, she didn't say how old she is
15:52.02GladiusI knew once an ircop who was 65 :)
15:52.12Gladiushorny as hell, I might add :)
15:52.23parylhaha
15:52.34parylis sourceforge down?
15:52.46GladiusiDunno, she hit on me A LOT !
15:52.57parylthere it goes
15:53.06iDunnoGladius: ahh - horny *and* insane, then? :)
15:53.10_Sam--sourceforge may be up....but teliax is down
15:53.18_Sam--and so if the phone service for my business
15:53.21parylGladius: i'm notoriously slow, so i wouldn't know when one was hitting on me.
15:53.23_Sam--if = is
15:53.36synthetiqme thinks katty is mid 20s. has a butch hair cut and wears thick glasses
15:53.44parylDandre: http://prdownloads.sourceforge.net/lame/lame-3.97b2.tar.gz?download
15:53.45*** join/#asterisk wrzf (n=chrisf0r@adsl-70-227-44-29.dsl.dytnoh.ameritech.net)
15:53.47synthetiqpale white
15:53.54synthetiqchunky
15:53.57Gladiusparyl: same here, and I figured it out... so it was obvious as hell
15:54.06wrzfGood morning all
15:54.11synthetiqwillign to bet 3$ paypel
15:54.12parylGladius: that's the best kinda flirting, imho
15:54.15synthetiqpaypal
15:54.15[TK]D-Fendersynthetiq : Please leave "thinking" to qualified professionals :)
15:54.16Dandrethanks paryl
15:54.18parylw00t!  slutty girls!
15:54.22parylDandre: np
15:54.33synthetiq3$ paypal bet who wants to take me on
15:54.43wrzfGot a ? if somebody is up to it.. (-:
15:54.57iDunnowrzf: woah - next you'll tell us it's asterisk related?
15:55.08Gladiuswrzf, why do you make me turn my head the opposite direction ?
15:55.16wrzfhahahah
15:55.20iDunnowrzf: and not to do with porn, or (potentially) offending Katty
15:55.23wrzfGood morning to you too 9-;
15:55.31wrzfhahaha
15:55.41wrzfprobably easy for you guys
15:55.52parylyes, probably
15:55.52GladiusiDunno, did you see the movie ?
15:55.57parylIF YOU WOULD ASK IT
15:56.00paryl;)
15:56.07KattyiDunno: possibly.
15:56.29iDunnoGladius: nah - had the word WOW in it, which I was guessing was World Of Warcraft, so I didn't bother :)
15:56.30Kattysynthetiq: fyi, i am pale.
15:56.36Kattysynthetiq: but hardly chunky ;)
15:56.38wrzfI have an inbound did pointing at my asterisk IP address. Do I need to run another program to get that did to ring in. I get Line congestion erros when I call it.
15:56.41Gladiustrust me, it's funny :)
15:56.50parylpale = best ever
15:56.52Kattysynthetiq: but feel free to ask anyone who's met me in person (=
15:56.54synthetiqdamn, why didnt soemone bet me, i would have 3$ more paypal
15:57.06synthetiqanythign bigger than size 6 pants is chunky
15:57.17synthetiq:discuss:
15:57.42iDunnosynthetiq: just which countries sizes are you going by? and you could politely STFU ;)
15:57.43parylwrzf: how can you have a did pointing at an ip address?
15:57.50paryla did is PSTN
15:57.52Kattysynthetiq: perhaps you belong in south korea then (=
15:57.59wrzfits as UK Number just pointing at my IP
15:58.02synthetiqthis country?
15:58.10brettnem"this"
15:58.12iDunnosynthetiq: which is?
15:58.22synthetiqunited states
15:58.38wrzfwhen I call it I get line Congestion errors I put in my inbound routing table but i still get these errors
15:58.41*** join/#asterisk alexhopper (n=a27386@CPE000103d29ae2-CM001225dfdfe0.cpe.net.cable.rogers.com)
15:58.56Kattysynthetiq: girls aren't barbies.
15:58.58iDunnoUK size 6 is insanely small, what the hell would you get hold of?!
15:59.03Kattysynthetiq: if you don't like that, go find a male.
15:59.04parylwrzf: what does the debug show?
15:59.06synthetiqmy ex was a size 4
15:59.08Hmmhesayswww.realdoll.com nsfw
15:59.09_Sam--if anyone knows the guy from teliax tell him his stuff is broke
15:59.19jimmy_deanPBok, just wanted to contribute this: if people complain about echo problems with their zaptel channels, have them upgrade from * 1.0.x series to 1.2.x...completely cured my problems.
15:59.26synthetiqme bets size 10
15:59.32parylwrzf: when the call is coming in... do you see it at all?
16:00.09sivanaheh
16:00.18KattyiDunno: twisted would do it, too
16:00.19wrzfyes
16:00.26sivanaHRISTMAS SPECIAL!
16:00.26sivanaPlace your doll order before Dec. 31st with payment in full
16:00.26sivanaby check, Money Order or Wiretransfer
16:00.29*** join/#asterisk VincentMX (n=vincent@c3eea0597.cable.wanadoo.nl)
16:00.39sivana250 off!
16:00.40VincentMXhi
16:01.09VincentMXcan you use asterisk to call to a normal phone from a computer?
16:01.15sivanayes
16:01.18VincentMXwithout modem?
16:01.24alexhopperyes
16:01.25sivanayou can use your computer to call a normal phone with *
16:01.31VincentMXok
16:01.44VincentMXis asterisk hard to use?
16:01.50sivanadepends
16:01.58fileif it keeps you from using it, and asking questions, yes!
16:01.59alexhopperyeah - it jsut sends the phone signal out over the internet...and then conects youinto the land lines
16:02.02sivana:)
16:02.07[TK]D-FenderVincentMX : You need some sort of interface to the PSTN.  A card in your * server, a VoIP gateway (SPA-3000), a VoIP PSTND provider (take your pick of the dozens out there)
16:02.40[TK]D-FenderHAPPY MORNING TO YOU FILE!!!!!!
16:02.43Seldon1975anyone know if it's possible to rollback the sip.ld on a polycom 301
16:02.52file[TK]D-Fender: ah go to hell :P
16:03.14VincentMXwhat's a PSTN?
16:03.44Seldon1975i updated the sip.ld and now my phone doesnt talk to the * server
16:03.46[TK]D-FenderVincentMX : PSTN is just another term for the normal phone network.
16:03.51Seldon1975i need to roll back
16:03.55Kattyfile: suffocater.
16:03.55[TK]D-FenderVincentMX : Typically just an analog phone line.
16:04.01VincentMXok
16:04.07Kattyalexhopper: paws off, plskthx.
16:04.08VincentMXlike a modem?
16:04.23filemodem != analog phone line
16:04.28filemodem = modulator/demodulator
16:04.40VincentMXok
16:04.49Kattythat was on the a+ test.
16:04.52alexhopperan analog phone line is just the traditional phone system most everyone has
16:05.03Kattymost everyone that's not in here, you mean.
16:05.06VincentMXso i do need a modem for calling to normal phones?
16:05.06fileanalog is silly
16:05.15alexhopperNaturally
16:05.24filewell, if you use a modem to get on  the internet and plan to use a provider - SURE!
16:05.43VincentMXi use cable to go to Internet
16:05.54iDunnoKatty: when putting people in to boxes, don't forget to cut air holes.
16:05.57alexhopperthen thats how you calls will go out
16:06.05fileif you choose to use a VoIP provider.
16:06.06KattyiDunno: file is special and can transcend walls.
16:06.15fileand transcend reality!
16:06.21iDunnocoo.
16:06.30iDunnothat's quite an impressive feat.
16:06.32_Sam--just dont use teliax
16:07.05VincentMXalexhopper, so i just install asterisk and then i can call to normal phones?
16:07.11iDunnohmm - I need to do some asterisk with normal analogue lines at sometime in the not too distant :/
16:07.14DandreOk now moh works with raw files but not with mp3 files. May be I have fogotten sth
16:07.18_Sam--or if you can live with your business and employees not having phone service on a regular basis, then you should choose teliax.
16:07.30fileVincentMX: you have to pay either a VoIP provider, or get hardware that allows you to connect your phone line into asterisk
16:07.38VincentMXok
16:07.45*** join/#asterisk slak- (i=slak@shudup.before.you.get.rewted.biz)
16:07.46*** join/#asterisk Smidge (n=steve@steve.enta.net)
16:08.14slak-hi, i just set up email notification for vm and .wav attachments, i'd like to run it for all old messsages for one mailbox
16:08.15alexhopperIt's just like any phone service, you have to pay for it.
16:08.17slak-can that be done?
16:08.20*** join/#asterisk t0ke (n=t0ke@51.Red-83-46-136.dynamicIP.rima-tde.net)
16:08.24t0kehi
16:08.27fileslak-: not really no
16:08.46_Sam--just forward them from the voicemail options to a new box
16:08.47slak-put it on your list file
16:08.50_Sam--and then have the new box email
16:08.55filemy list is already long enough
16:08.58filelike other stuff of mine!
16:09.19t0keI have connected one E1 to one of my TE410P ports, anyone know if there is any command which show what ports are connected?
16:09.34slak-type "help
16:09.35slak-"
16:09.37slak-in the cli
16:09.56t0kehow I launch cli mode in asterisk? with -g?
16:10.14slak--r
16:10.15fileasterisk -vvvvgc
16:10.18Spla4t1asterisk -r
16:10.20slak-if its running
16:10.24filewill start asterisk running in console
16:10.32fileasterisk -r to attach to an exisiting running asterisk
16:10.43fileand kill -9 to make users scream
16:10.55t0ketnks
16:11.22DandreI haven't installed mpg123 :-(
16:11.32alexhopperOutside the snow is falling and friens are calling to you, come on it's lovely weathr  for a sleighride together with you!
16:11.43*** part/#asterisk VincentMX (n=vincent@c3eea0597.cable.wanadoo.nl)
16:11.50Hmmhesaysanyone know of any uk anac numbers?
16:12.10parylDandre: you don't need to
16:12.16viperdudeanyone know how to get asterisk to stop presenting a call from a queue to a agent who is already on a call?
16:12.54*** join/#asterisk funxion (n=nunya@mtnuser.icgws.com)
16:12.58parylviperdude: the best method would be for their ACD extension to be only one line
16:13.06Hmmhesaysi'm still trying to sort out the BT clid issue
16:13.11DandreI have installed it and it works now Why shouldn't I install mpg123?
16:13.17viperdudeis that configured on asterisk or on the phone?
16:13.30parylDandre: because it sucks
16:13.57Seldon1975Dandre there is a security flaw
16:14.06Seldon1975but it does work
16:14.06Kattyparyl: lolzitSUCKSkthx
16:14.10Kattyparyl: how about a /real/ answer
16:14.15Seldon1975i just installed it yesterday
16:14.29parylalso license issues... you don't want to use it for a busines
16:14.30paryls
16:14.41parylKatty: yes, ma'am
16:14.49DandreSo how should I play mp3 moh ? Is there another tool for that?
16:14.51taecis there any way to display the original caller id, or at least flag the call in a certain way on a blind transfer through asterisk?
16:14.52Kattyparyl: good boy.
16:15.04parylDandre: i just gave you the stuff you need for raw
16:15.40Dandreshure but the sound quality isn't the same. Maybe a configuration issue?
16:15.45klicteluse rawplayer
16:15.52parylDandre: try this: http://www.orderlyq.com/asteriskqueues.html
16:15.59parylit explains using rawplayer
16:16.11alexhoppersont sing
16:16.14alexhopper*dont
16:16.51LostFrogHow can you afford the long distance charges? <grin>
16:16.57alexhopperlmao
16:17.07filethat was bad
16:17.11parylwho's the guy that everyone says is a queues god?  i need help
16:17.12filebut ironically he's just across town
16:17.15file:P
16:17.23filewhich makes it harder to avoid him
16:17.26alexhopperyeah - 20 minute walk
16:17.28alexhopperMWa ha ha
16:17.33alexhopperI'm coming over!
16:17.39fileprobably longer today, considering the storm :P
16:19.05alexhopperhehe = yeah
16:19.05iDunnoheh
16:19.12LostFrogNoone ever calls me..
16:19.17LostFrogI like it that way. :)
16:19.21DandreI have seen that 1.2 has its own mp3 player. Why doesn't it ply mp3 moh on my box?
16:19.25fileLostFrog: you probably have an extension that's hard to get to
16:19.27file:P
16:19.33Kattyfile: !
16:19.38fileKatty: !
16:19.41iDunnonot getting called is good.
16:19.46alexhopperalexhoppper: !
16:19.47iDunnoit means that nothing has broken :)
16:19.51LostFrogI have people who give messages to other people to forward to me, because they are afraid to call me. :)
16:19.56filealexhopper: wow, people sure like you
16:20.01alexhopperThey do
16:20.17parylhey, i found a wierd memory leak yesterday, but i don't know why it happens
16:21.04fileparyl: do you know how it happens, ie: how to reproduce it?
16:21.13Kattyfile: are you not answering your messageseses?
16:21.14fileif so, file a bug report with details and maybe someone will help track it down
16:21.17filemmmkthxbye
16:21.28fileKatty: I got none!
16:21.43parylfile: if you have a timeout in a Dial with the next rule Goto a queue, and that extension is dialed from a queue... it kills * within about 5 minutes
16:21.56filetjat
16:21.59filethat's odd
16:22.04fileKatty: lemme exit and reopen iChat, 1 sec
16:22.11Kattyfile: k'then
16:23.09filethat's very very odd
16:23.20parylfile: depending on the number of incoming calls, natch, but it does go down.  and before it goes down, every call that comes into the queue goes to dead air
16:23.22iDunno# why do you get all the love in the world
16:23.33Kattyfile: and prime.
16:23.44parylfile: but SIP-SIP calls work fine up until * dies
16:24.15LostFrogBeyond 3, are there any prime triangular numbers?
16:24.25*** join/#asterisk ahmuck_jr (n=chatzill@p115n4.ruraltel.net)
16:24.47parylLostFrog: short answer 'yes' with an 'if', long answer 'no' with a 'but'
16:24.54paryl:)
16:24.57*** join/#asterisk fulgas (n=fulgas@213.58.130.46)
16:25.10ahmuck_jrcan one take a standard pc and put three modems in it and use it ?
16:25.20filemodems?
16:25.23filelike regular ol' modems?
16:25.27fileand use them for voice?
16:25.34ahmuck_jryes
16:25.37LostFrogahmuck_jr: with IRQ sharing or full PCI modems, yes.
16:25.41*** join/#asterisk lorinc (n=ang@caracas-1065.adsl.interware.hu)
16:25.54fileno, you can't use modems like that...
16:25.59filenot for asterisk
16:26.04ahmuck_jrso if i have three lines in my house, take a standard pc and put three us robotics or 3com modems in them
16:26.10Kattyfile: what about muffins.
16:26.13LostFrogohh.. not for asterisk.
16:26.16Kattyfile: can you use muffins like that in asterisk?
16:26.24fileKatty: yes, thanks to chan_muffin!
16:26.44Delvaryou lot realy are insane
16:26.59Kattyfile is practically my twin.
16:27.03LostFrogI prefer chan_bakedgoods
16:27.09Kattybut impractically not so.
16:27.18Kattyhe's smarter.
16:27.29Katty.........in voip.
16:27.39Kattybut i'll clearly slingshot around the earth before he does.
16:27.40LostFroglol.. I like the qualification.
16:28.32kpettitI'm trying to use astfax to send faxes through asterisk
16:29.09LostFrogUnfortunately, Katty, you will accidentely reach escape velocity.
16:29.11parylKatty: you better plan your trajectory well
16:29.16kpettitThere is only breif install instructions and they are specific to qmail and not very informitive.  Anybody know of any more general instructions
16:29.26kpettitor how to get astfax to work with a different MTA?
16:29.30parylLostFrog: we had the same thought :)
16:29.41`Sauronkpettit: welcome to open source
16:30.04LostFrogperyl: I am sure she will plan ok.. It's that extra .5 seconds of burn time that will get her.
16:30.11Kattyi'm always well away of escape veloticity requirements, etc.
16:30.16Kattys/away/aware/
16:30.20kpettit`Sauron, yeah it's fun.  But this app dosen't even have a help or a readme which makes it more difficult to experiment
16:30.40kpettitif you rrun the app it just sits there doing nothing so I'm not sure what to do with it to test
16:30.44*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
16:30.45Katty7mps
16:31.10`SauronKatty: Mew
16:31.31Kattyv_esc=sqrt(2*G*M/R) (=
16:31.36Katty`Sauron: mew.
16:31.36parylKatty: is that your major?  (given our previous conversation on photons being split)
16:31.46Kattytwisted[asteria]: !
16:32.34iDunnoKatty: what's R in this case?
16:32.42slak-RATE
16:32.51LostFrogdistance from the center of the earth, iDunno.
16:32.52*** join/#asterisk SpaceBass (n=SpaceBas@static-71-251-230-2.rcmdva.fios.verizon.net)
16:32.54Kattyradius
16:32.55LostFrogRadius
16:33.04slak-roach
16:33.07twisted[asteria]Katty, :P
16:33.08iDunnoright - cool - that's what I thought :)
16:33.12alexhopperKatty: You dont wanna be twins w/ fily
16:33.13twisted[asteria]hard math in the morning makes me combust
16:33.21Kattytwisted[asteria]: oops, sowwie.
16:33.23filebah
16:33.30LostFrogBeans do that to me, twisted[asteria].
16:33.32twisted[asteria]Katty, heh, s'ok
16:33.45iDunnoonly an hour till home time though :)
16:33.54iDunno(or, in this case, pub o'clock)
16:34.27Kattytwisted[asteria]: we really need to go raid nasa :>
16:34.28paryliDunno: can we swap timezones?  just for today?  sigh.
16:34.36Kattytwisted[asteria]: did you ever figure out what to do with the semi?
16:34.38*** join/#asterisk Porks (n=nao@200.231.120.138)
16:34.58slak-mark spencer also works on gaim right?
16:35.00iDunnoparyl: fraid not, I like this timezone
16:35.00alexhopperMy physics teacher used to work for nasa!
16:35.06twisted[asteria]Katty, actually, yea, there's a new road being built, and an overpass that could easily conceal it
16:35.08slak-this homeboy here claims he went to school with mark spencer
16:35.08twisted[asteria]:P
16:35.11alexhopperMaybee he can get us codes!
16:35.13parylstupid central time...
16:35.17slak-in alabama
16:35.25iDunnothough, if someone could give me some sunlight to counteract this flickering flourescant tube, that'd be great.
16:35.32twisted[asteria]slak-, it's not unlikely.
16:36.12*** join/#asterisk mog_work (n=mogorman@gateway.digium.com)
16:36.27*** join/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net)
16:36.34iDunnota :)
16:37.13parylcan upping txgain actually reduce echo?  that seems counter-intuitive to me
16:37.15LostFrogUmm.. that is a metric ruler, Paryl.. It's 50mm.
16:37.27*** part/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net)
16:37.42Kattytwisted[asteria]: excellent
16:37.44Kattytwisted[asteria]: when do we go?
16:37.50parylagain with the stupid place i live!  why can't we freaking use real measurements?
16:37.52Hmmhesaysyes I smoked another pc
16:37.55Hmmhesaysi've got the touch
16:38.31slak-dudes can i do this: PHONE=${MACRO_EXT} exten => ${PHONE},1,Macro(macroname,${PHONE}
16:38.34LostFrogHmmhesays: that doesn't take much.
16:38.35slak-will that work
16:38.41slak-for all my extensions
16:38.57LostFrogI didn't know you could match on a variable.
16:39.25parylslak: what are you trying to accomplish?
16:39.34slak-minimize my extensions.conf
16:39.37slak-right now i have:
16:39.38twisted[asteria]Katty, soon.  when the eagle lands on fertile soil, and the mouse rolls up in pygme oil
16:39.41slak-PHONE15=119
16:39.41slak-PHONE16=120
16:39.45slak-exten => ${PHONE00},1,Macro(oneline,${PHONE00})
16:39.45slak-exten => ${PHONE01},1,Macro(oneline,${PHONE01})
16:39.49slak-i want to just have two lines
16:39.54slak-for all 20 extensions
16:40.01paryldude, it sooo much easier than that
16:40.14slak-and use the extensions dialed for the number
16:40.20*** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)
16:40.31twisted[asteria]Hmmhesays, smoking PC's is bad for your health
16:40.38iDunnoslak-: erm - well, surely you know the range of numbers for the phones, so use...
16:40.41generalhanwhats up guys !
16:40.47*** join/#asterisk mut (n=animenod@65.111.201.79)
16:40.56filetwisted[asteria]: don't you have stupid people to help become the next Vonage?
16:41.09iDunnoexten => _1XX,1,Macro(oneline,${EXTEN})
16:41.17iDunnoor similar.
16:41.21slak-hmm
16:41.33parylslak: http://pastebin.ca/33669
16:41.36slak-ty
16:41.51parylslak: and that even makes it so when they call their own extension it dials VM :)
16:42.07paryli've got a block of 100 DID's, and that works for all of them
16:42.22generalhanI have a quick question ... I have my calls being recorded by ${CALLERIDNUM} but the recordings are being saved as "user" for $CALLERIDNUM anyone have any idea why its not showing the phone number of the caller ?
16:42.46slak-paryl: your extensions are 40XX?
16:43.14twisted[asteria]file, pardon?
16:43.26filetwisted[asteria]: nevermind then!
16:43.39alexhopperDont mind him, he's just an idiot
16:44.03alexhopperfile: just kiding
16:44.09file:p
16:44.29Smacefile: Vonage?
16:44.33iCEBrkrDIE DIE DIE DIE DIE DIE
16:44.39fileI refuse to die
16:44.49iCEBrkrkill -9 file
16:45.00twisted[asteria]nooo
16:45.02twisted[asteria]rm -rf file
16:45.04alexhopperSmace: A little known company that uses something calle VoIP
16:45.07Smacekillall file
16:45.15*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
16:45.16twisted[asteria]you cannot kill a file, but you can delete it
16:45.16iCEBrkrfile: fine.. init 0
16:45.23filelies
16:45.27fileyou can't delete me
16:45.30fileI control your filesystem!
16:45.30iCEBrkrhaha
16:45.31filemuahahahaha
16:45.32twisted[asteria]wanna bet?
16:45.33shmaltzI like this:
16:45.34shmaltzhttp://www.engadget.com/entry/1234000497072377/
16:45.37*** part/#asterisk file (n=jcolp@mctnnbsa24w-142167049176.nb.aliant.net)
16:45.42iCEBrkrLOLOLOL
16:45.42parylslak: sorry, yes
16:45.45*** join/#asterisk file (n=jcolp@mctnnbsa24w-142167049176.nb.aliant.net)
16:45.46*** part/#asterisk file (n=jcolp@mctnnbsa24w-142167049176.nb.aliant.net)
16:45.51generalhanHAHAHAHA
16:45.51*** join/#asterisk file (n=jcolp@mctnnbsa24w-142167049176.nb.aliant.net)
16:45.55filethat's cheating
16:45.55iCEBrkrhaha
16:45.56twisted[asteria]file, told ya :)
16:46.05slak-paryl; i dont understand your macro
16:46.08slak-can you break it down
16:46.13slak-in msgs?
16:46.15twisted[asteria]that's a fun little trick.
16:46.16LostFrogfile: I can just remove the inode entry.
16:46.17iCEBrkrtwisted[asteria]: he's like a virus.. You delete him and he keeps coming back
16:46.40alexhopperAnd throwing muffins at you...
16:47.42twisted[asteria]file, lol
16:47.49twisted[asteria]file, i don't answer calls that don't have ANI
16:48.50file:P
16:49.17filethat was my cellphone
16:50.51iDunnoANI?
16:52.06IkarusAutomatic Number Identification
16:52.09IkarusCaller ID
16:52.45iDunnoahh
16:52.47filewell
16:52.47drumkillatwisted[asteria]: i know, he tries that crap with me too
16:52.53filecallerid and ANI are two different things
16:52.58iDunnoright.
16:53.03generalhancan anyone help me out with my monitor issue ?? i dont understand this at all.
16:53.13iDunnoso what's ANI? it maps it to an entry in a phone book?
16:53.30generalhanI have my calls being recorded, Monitor(), by ${CALLERIDNUM} but the recordings are being saved as "user" for $CALLERIDNUM anyone have any idea why its not showing the phone number of the caller ?
16:53.36Kattytwisted[asteria]: but what if i call?
16:53.42LostFrogWe heard you the first time, generalhan.
16:53.47Kattytwisted[asteria]: i don't think that has ANI on it
16:54.08Kattytwisted[asteria]: though i suppose it does........you do know /who's/ calling
16:54.12Ikarusfile: ANI and CID are used to mean the same thing here
16:55.22fileANI and callerid can be two completely different things
16:55.29iDunnoright - OK
16:55.42filefor example
16:55.43iDunnoso what *exactly* is ANI as apposed to CID?
16:56.01filea US cellphone customer roaming in Canada can have a US callerid, but a Canadian ANI
16:56.36filenow if you bill that off callerid when someone calls a toll-free number in the US, you would be missbilling it
16:56.45filebecause they were actually calling from Canada - which usually has a higher rate
16:56.59Ikarusfile: ah, that is implemented differently here
16:57.49iDunnofile: *ahhh* - I see what you mean. right. cool.
16:57.57Ikarusfile: here the company you are roaming with calculates the extra charge and forwards it to your own telco, the rest of the cost stays the same
16:58.18IkarusSo there is no need for the local telco to billingwise know where the phone is
17:00.18shmaltzanybody seen the treo 650 with Linux?
17:00.40fugitivourl?
17:01.34*** join/#asterisk asteriskmonkey (n=phil@69.156.197.242)
17:02.18asteriskmonkeyCan anyone tell me if asterisk supports presence yet?
17:02.36filewell
17:02.45fileit can tell you if somebody's phone is inuse, whatever
17:02.55asteriskmonkeyyes i know using hints and limits
17:03.18asteriskmonkeybut no official presence support atm?
17:03.29filecorrect, it doesn't accept it from the phone and that
17:04.08slak-hey how do i know what 10x priority needs to be my if busy goto vm line
17:04.15slak-exten => s,4,Dial(SIP/${ARG1},20,t)
17:04.18slak-exten => s,104,Voicemail(b${ARG1})
17:04.18slak-exten => s,105,Hangup
17:04.22slak-is it 104
17:04.22slak-?
17:04.47parylslak: i'm looking for that
17:04.52slak-thanks
17:05.44paryli'm pretty sure it's +101
17:05.46KriS83Isn't it normally prio + 101
17:05.48parylnot +100
17:05.53slak-ok so 105
17:05.58paryljawohl
17:06.04Qwellslak-: What version of asterisk?
17:06.08slak-1.2.1
17:06.19QwellThen you need to do +1, and check ${DIALSTATUS}
17:06.28slak-eh
17:06.31slak-thats that new config
17:06.33slak-i dont like it
17:06.55QwellThen use option j, but it's cheesy and deprecated
17:07.10slak-exten => s,104,Voicemail(b${ARG1})
17:07.10slak-exten => s,105,Hangup
17:07.12slak-er
17:07.17slak-this you mean
17:07.17slak-exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum
17:07.18slak-exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
17:07.18fugitivodial+101
17:07.39Qwellslak-: not exactly, but that would work
17:13.06*** join/#asterisk truescot (n=truescot@j242050.upc-j.chello.nl)
17:14.34slak-hi
17:14.47slak-can someone take a look at http://pastebin.ca/33675 and tell me if that will work
17:15.01slak-its my macro and [extensions] lines...about 10 lines
17:15.06*** join/#asterisk toddf (n=toddf@net-66-210-104-104.theshop.net)
17:15.09*** part/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net)
17:15.42truescotplease can anyone help me, every call on my asterisk box gets dropped within 5 minutes and i get the error PRI got event: HDLC Bad FCS (8) on
17:15.42truescotPrimary D-channel of span 1
17:16.04Delvar<slak->: that should work
17:16.19*** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at)
17:16.21truescoti am using an e1 connection on a te110p card
17:17.15syleasteriskmonkey: even better case statements, check extensions.ael
17:17.29iDunnoslak-: looks good to me
17:17.40slak-okay i just have one question
17:17.44*** join/#asterisk adker (n=adker@67-51-239-152.dsl1.glv.ny.frontiernet.net)
17:17.45slak-extensions 106 is a special case
17:17.49slak-that cant follow that macro
17:17.54asteriskmonkeysyle: ive not gone to 1.2 yet still on 1.0.9 :) looking forward to it though :D
17:18.00slak-but the line exten => _1XX,1,Macro(oneline,${EXTEN}) will send it to the macro
17:18.00viperdudeon the cli, if i do a add queue member xxxx to xxxx can I specify the context for the member ie Local/100@mycontext
17:18.06slak-how can i get around that
17:18.35slak-if i put 106 above exten => _1XX,1,Macro(oneline,${EXTEN}) will that fix it?
17:19.42*** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com)
17:20.45*** join/#asterisk EriSan (n=erisan@81-174-42-85.f5.ngi.it)
17:21.16slak-[extensions]
17:21.16slak-exten => 106,1,Macro(remote,${NICKOFFICE})
17:21.17slak-exten => _1XX,1,Macro(oneline,${EXTEN})
17:21.19*** join/#asterisk Suckysucky (i=Borgon@70-100-55-164.dsl1.tbr.ga.frontiernet.net)
17:21.21Suckysuckyhello
17:21.23slak-will 106 in this case never match _1XX
17:21.28slak-i dont want it to
17:21.33slak-106 follows a different Macro
17:22.02SuckysuckyIf i got a international number to call and am using a softphone, how would i insert the number to call direct for example +639157435091 .. i tried 011639157435091 but nothing
17:23.36*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
17:24.15slak-everyone took off for lunch of what
17:26.11Delvarsorry :)
17:26.24Delvarslak-: asterisk will match 106 before _10XX
17:26.25*** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6)
17:26.40Delvarasterisk as i understand it sorts them alphabeticly
17:26.48Delvarand _ comes after numbers :)
17:27.02*** join/#asterisk jtodd (n=jtodd@dsl027-191-178.sfo1.dsl.speakeasy.net)
17:28.38*** join/#asterisk oej_ (n=oej@apollo.webway.se)
17:29.14LostFrogThere is a comment on the wiki that it isn't determinate which will match first..
17:30.00Ariel_Just wondering looks like the user list is down. I have not gotten any emails in over an hour.
17:30.01LostFrogI don't know if that changed for 1.2
17:32.25nextimecan i set multiple sip friend with same ip?
17:32.37oej_No, one login per peer
17:32.55DelvarLostFrog: i have lots of setups and i can assure you it will amtch 106 before _1XX
17:32.59Delvar100% sure :)
17:33.06loudthey will flap.
17:33.20nextimeoej_ are you sure? this is a bad notice for me...
17:33.21loudalthough both will be able to call though.
17:33.54LostFrogDelvar: works for me.. but whoever wrote the wiki entry probably knows what he/she was talking about.
17:34.02oej_nextime: If you're talking about how many phones you can connect to one account in Asterisk, as a sip peer, I am very sure.
17:34.25Delvarmaybe they have a dodgy version of asterisk... :)
17:35.04oej_The wiki is no longer, I am sad to say, to be fully trusted at all times.
17:35.07nextimeoej_ : they aren't phones, but is the same thing, i'm proxyng h323 to sip with yate, and i need separated login to recognize different h323 users
17:35.24nextimebut as you say this is pretty impossible
17:36.03oej_I don't know how yate translates caller IDs and such, maybe you can separate on something else in the dial plan
17:37.10nextimeoej_: yes i can recognize from clid or maybe from an addictional extension prepended to the called one, but isn't good for accounting and billing...
17:38.04oej_Then use asterisk all the way without involving yate
17:38.18*** join/#asterisk hugov6 (n=foo@p54AD54EB.dip.t-dialin.net)
17:38.24shmaltzlinux on treo 650:
17:38.25shmaltzhttp://www.engadget.com/entry/1234000497072377/
17:38.29hugov6hiho
17:38.38*** join/#asterisk zgor (n=zgor@61.Red-80-36-3.staticIP.rima-tde.net)
17:38.41zgorhi :)
17:39.10*** join/#asterisk heison (n=heison@ns.somanetworks.com)
17:39.26nextimeoej_ h323 support in * is bad, no gk at all and many problems with some h323 remote gateways
17:39.33hugov6is there another option to set the callerid for sip-phones than callerid and fromuser in sip.conf?
17:40.03DelvarsetCIDName
17:40.08DelvarsetCIDNum
17:40.10Delvarin dailplan
17:41.16hugov6Delvar: thx. 'll have a look at it.
17:41.36Smacehello
17:41.51[TK]D-FenderDelvar : the old CID connamds are now depreciated.
17:41.58Delvaryeah
17:42.13[TK]D-FenderI just finished adapting my setup for the DB stuff...
17:42.14DelvarSetVar(callerid(name)=bla) now isnt it?
17:42.24shmaltzhugov6, yes, RPID, or Set(CALLERID(datatype))
17:42.35drumkillayeah, don't use SetVar :)
17:42.40Delvarhehe :)
17:42.45drumkillause Set ... SetVar has been completely removed from the trunk
17:42.52Delvarbeen a while since i used any of that
17:43.12SmaceI'd like to ask for suggestion for one project. We've one large network here, over 50km of wireless networks and 10km of wired networks. We would like to use VOIP here, the point is i'm not sure about which hardware would fit our needs.
17:44.19[TK]D-FenderSmace : Depends on how many end-points, what kind of hardware (wifi-phones needed?), PSTN termination is any, etc....
17:44.22shmaltzSmace, how many phones?
17:44.38SmaceI was thinking about 50 for start...
17:44.38[TK]D-FenderWho are the users, what kind of phones, and any PSTN involved?
17:45.11Smacefirst I need to test it using 3 phones.
17:45.52SmaceThis network covers 3 cities. I want to use this network we have to make calls to IP-IP.
17:46.11*** join/#asterisk wshs (i=screwy@24-52-105-100.bflony.adelphia.net)
17:46.33shmaltzSmace, what is the maximum of users you ever want this system to support?
17:47.12Smaceat least 3 for start. but the right number would be between 50 and 100 users for 2006.
17:47.31jimmy_deanPBhey asteriskmonkey, upgrading to asterisk 1.2.1 pretty much got rid of all echo. we've had one call with echo in it but otherwise no echo.
17:48.12*** join/#asterisk zgor (n=zgor@61.Red-80-36-3.staticIP.rima-tde.net)
17:48.15zgorhi people :)
17:48.25LostFrog1.2.1 is out?
17:48.29LostFrogDuh..
17:48.32shmaltzSmace, then go with a Dual Xeon, and ~2GB RAM, or you could use Dual Core 64bit AMD
17:48.50Smaceshmaltz, but it's not at the same time.
17:48.54shmaltzSmace, that is if you will not use *any* recording or heavy AGI
17:49.06shmaltzSmace, what are you trying to say?
17:49.09*** join/#asterisk junbug (n=junya@c-66-176-211-109.hsd1.fl.comcast.net)
17:49.33zgoris there any way to make SIP extension gain audio ? Im using Sipura 841 with updated firmware and set gain level on web interface, but still being very low level. Any idea ?
17:49.40Smaceshmaltz, I've 50 points in our network I'd like to have ip-phone there.
17:49.42shmaltzSmace, what PSTN connection you going to have?
17:50.02shmaltz~pstn
17:50.03jbotrumour has it, pstn is Pubic Switched Telephone Network, or "please stop the nonsense"
17:50.16shmaltzSmace, how you connecting to the telephone company?
17:50.20Smaceshmaltz, I'm getting started do you suggest me any site?
17:50.32Smaceshmaltz, we have some lines here. We can use that.
17:50.35shmaltz~docs
17:50.37jbotit has been said that docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com
17:50.56SuckysuckyIf i got a international number to call and am using a softphone, how would i insert the number to call direct for example +639157435091 .. i tried 011639157435091 but nothing
17:51.15shmaltzSmace, see you in around 3 days, I'm sure you will be more confused. :P
17:51.22Smacelol
17:51.36shmaltzSuckysucky, ask your provider what they expect
17:52.09Smaceshmaltz, is there any comercial ip-phone that make calls from ip to ip ? something like one access point.
17:52.26Dandanhow do you force first time user to set up his vm?
17:52.36Dandansetting password to his vm number doesn't do it
17:53.00shmaltzSmace, Cisco, Polycom, Grandstream, Sipura, Snom, Zytux, and RTFM on the wiki you should be able to find more
17:53.16Suckysuckyshmaltz: they say do 011 with area code
17:53.24LostFrogRTFW?
17:53.25Smaceshmaltz, is it configured like one access point?
17:53.40rob0hey, off topic but someone here would know ... I have telephone headsets with the itty bitty plug. Is there an adapter/splitter which would allow that to be used for a computer sound card speaker/mic jacks?
17:53.50Smaceshmaltz, I can set up it to use one external voip provider and also make ip2ip calls?
17:53.59shmaltzDanda, you will have to write your own AGI for it, as asterisk DP cannot do that, since asterisk doesnt have an app that will drop you just to the change password in app_voicemail
17:54.07rob0(and what would I ask for?)
17:54.13shmaltzSmace, where you located?
17:54.57shmaltzrob0, visit your local radio shack
17:55.00Smacerio de janeiro brazil
17:55.44rob0:)
17:55.45Dandanfor new years :)
17:56.17shmaltzSmace, you paying fare?
17:57.26*** join/#asterisk mutilator (n=animenod@65.111.201.79)
17:59.22LostFrogJust for the record, the only prime triangular numbers are 1 and 3.
17:59.50Dandanhow do you force first time user to set up his vm?
17:59.56Dandanpls?
18:00.55Smaceshmaltz, we make to much calls between these 3 cities. as we have one wide wireless network that covers it, i thought i could avoid paying for these calls.
18:01.03Smaceusing our own private network
18:01.36shmaltzDandan, you will have to write your own AGI for it, as asterisk DP cannot do that, since asterisk doesnt have an app that will drop you just to the change password in app_voicemail
18:01.55Dandanhm, i thought i had that a long time ago...
18:02.00Dandanbut thx shmaltz
18:02.18shmaltzSmace, you could, but the most CPU power will be needed when converting between PSTN and VoIP, thats why I asked what I asked
18:02.20*** join/#asterisk hugov6 (n=foo@p54AD6B94.dip.t-dialin.net)
18:02.25hugov6hmz.
18:03.47Smaceshmaltz, i believe CPU is not gonna be the problem, the problem is gonna me to do it :)
18:04.03shmaltzSmace, I told you, just pay the fare :P
18:04.21Smacehow much is the fare?
18:04.23Smacehehe
18:05.16Smaceshmaltz, where are you from?
18:05.20shmaltzNJ
18:05.52SmaceNew Jersey?
18:06.12hugov6hmmz. anyone an idea on how to set outgoing callerid for sip-phone and mISDN?
18:06.50shmaltz~nj
18:06.52jbotfrom memory, nj is home to the sopranos.  Fogedaboudit!
18:07.13shmaltzSmace :)
18:10.24Dandanshmaltz: forcename=yes and forcegreetings=yes does that
18:10.32Dandan~ct
18:10.39shmaltzDandan, what?
18:10.49shmaltzwhere? in voicemail.conf?
18:10.54Dandanforces user to record name/greetings
18:10.55Dandanyes
18:12.30*** join/#asterisk ahmuck_jr (n=chatzill@p115n4.ruraltel.net)
18:12.42shmaltzDandan, but nothing for the password
18:12.55asteriskmonkeyis there a protocol you can tie in to some voip phones that will present it with an address book?
18:12.57ahmuck_jri don't quite understand what the hardware is for ?
18:13.02Dandanyes, if your password = your ext. it will ask you to change pass...
18:13.22Dandanshmaltz: i used in in 1.0.6 afair :)
18:13.25shmaltzDandan, is that new as well?
18:13.27Dandanin my testbed :)
18:13.38Dandanshmaltz: no, but it is not documented
18:13.41shmaltzDandan, then why did you ask?
18:13.53*** join/#asterisk genmud (n=genmud@ip68-98-82-206.ph.ph.cox.net)
18:13.59Dandancause I didn't know where it was
18:14.02Dandan:))))
18:14.11Dandanand I have no conf files to grep
18:14.21Dandansorry to bother
18:15.16ahmuck_jri don't quite understand what the hardware is for ?
18:16.58shmaltzasteriskmonkey, thridlane makes an MS Outlook plug in that allows you to click on any phone# in Outlook to dial it thru *
18:17.25asteriskmonkeysmaltz: is it free
18:17.28Dandanor is it *THIRDLANE?
18:17.57Dandanshmaltz: how about a pop-up caller id for your pc?
18:18.00shmaltzasteriskmonkey, also, some phones (Polycom, and Cisco), will allow you to create an xml file that will show up on the phone, no it's not free, and yes its thirdlane
18:18.08shmaltzDandan, use FOP
18:18.14DandanFOP?
18:18.17tainted-anyone have experience doing voip in mexico?
18:18.18asteriskmonkeyi know xten makes a softclient that links into outlook and does the same.. i was looking for something that moreso actuaclly populates a phonebook on a sipphone (hardphone)
18:19.07nextimeis the open source implementation of g729 supporting g729r8?
18:19.13shmaltzasteriskmonkey, thats where the xml files for the Polycom Cisco comes in
18:19.20Dandan~fop
18:19.21jbotAn XSL formatter written in Java that outputs PDF. URL: http://www.jtauber.com/fop/, or the Flash Operator Panel
18:19.22LostFrogI think it just supports 16.
18:19.25nextimeor only g729a like the others?
18:19.41Dandana
18:19.45Dandanthat thing :)
18:20.28nextimeDandan : "a" was for me?
18:21.06asteriskmonkeyshmaltz: tell me more :)
18:21.30asteriskmonkeyim also looking for some hacks info trick for a wip-5000 if anyone has one of those aka hitachi ip 5000
18:21.35shmaltzasteriskmonkey, more about what?
18:22.10asteriskmonkeythe xml extension stuff is there documents on the polycom web?
18:22.27shmaltzasteriskmonkey, yes the docs are in the admin manual
18:22.32shmaltzsame goes for the cisco
18:23.18shmaltzI think the cisco uses http for that, which makes it very easy to be dynamic, while the polycom only supports static from ftp/tftp, or with the newer bootroms http/https
18:24.40asteriskmonkeyso is there a common format for those phones?
18:24.51shmaltzasteriskmonkey, now RTFM
18:25.15SuckysuckyCan anyone direct me on how tom ake an international call? am using voicepuse and my provider said type 011 + country code.. the number am dialing is +639157435091
18:25.41asteriskmonkeyshmaltz: thanks ... but is the syntax same for polycom and cisco ? or is it a completely different xml structure?
18:26.04*** join/#asterisk sysdebug (n=sysdebug@200.163.193.247)
18:26.43shmaltzSuckysucky, so you trying to dial the philipines, first ask voicepulse if they have service to the philipines
18:26.47bsdfreakheh
18:26.55Suckysuckyshmaltz: they do
18:26.58shmaltzasteriskmonkey, I think it's completely different
18:27.11shmaltzSuckysucky, how do you know?
18:29.22*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
18:34.20ahmuck_jrhow do the cards work ?
18:39.04[Lamer]I'm running asterisk with some fxo cards to do the auto-dialout
18:39.35[Lamer]how do I know when the other side picks up the phone?
18:40.42shmaltzhttp://www.local6.com/news/5527928/detail.html
18:41.06shmaltz[Lamer], if you use a PRI, you would know because of the answer state
18:41.10*** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net)
18:42.44*** join/#asterisk marc32422 (n=marc3234@206-248-134-12.dsl.teksavvy.com)
18:43.32*** join/#asterisk jahani (n=BaT@adsl196-219-27-206-196.adsl196-1.iam.net.ma)
18:43.49*** join/#asterisk robl^ (n=robl@dsl093-025-118.hou1.dsl.speakeasy.net)
18:43.50[Lamer]shmaltz: so, there's no way to recognize it using POTS?
18:44.09shmaltzhttp://www.firstcoastnews.com/news/topstories/news-article.aspx?storyid=48816
18:44.59shmaltz[Lamer], no, not with POTS, since its answered when it goes off hook, remember POTS uses inband signalling
18:46.05*** join/#asterisk kuonSama (n=kuon@alragore.goyman.com)
18:46.06shmaltz[Lamer], most SIP porviders, will also give you the answer state, but not all of them
18:46.06kuonSamahello
18:47.11bsdfreakheh
18:47.14[Lamer]shmaltz: I see. I tried with the D41E from dialogic and it seems to recognize those though
18:48.18[Lamer]shmaltz: but only when I called to non-mobile phone users
18:49.16LostFrogDamn.. you can't even buy your friend a hooker in class any more.
18:50.59robl^LostFrog:  buy him a gift certificate
18:52.08LostFrogrobl^: Did you read the site that shmaltz posted?
18:52.38robl^no.. I am at work...
18:52.38*** join/#asterisk sysdebug (n=sysdebug@200.163.193.247)
18:52.41LostFrogIt's not dirty.
18:53.07LostFrogA 18 year old kid paid a 16 year old girl to go down on another minor in class.
18:53.13robl^they have LOTS of "filters" preventing web browsing outisde a few select sites here.
18:53.20rob0Some content filters would block the phrase in the title.
18:53.30rob0namely, Dansguardian.
18:53.55robl^but then again.. since I have remote access to my home computer.. they don't block outbound ssh... :)
18:54.10LostFrogsecure-tunnel.com
18:54.13LostFrog:)
18:54.21rob0"Not indicative of what is happening in schools" .. LOL
18:54.54LostFrogYeah.. we at least used to go into the bathrooms or supply rooms. :)
18:55.09*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
18:55.25*** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net)
18:55.30SpaceBasshey folks
18:55.55LostFrogHey, SpaceBass.
18:56.07SpaceBassdire straits.... trying to order a SSL cert and it requires an automated system to call me... I have to press # which asterisk thinks is me iniating a transfer
18:56.11SpaceBassstupid asterisk@home
18:56.20LostFrogAnyone know of a *good* windoze hylafax client?
18:56.27SpaceBasswith out editing my entire dial plan, is there a quick way around that
18:56.33SpaceBassie disable # transfers
18:57.11sivanaSpaceBass: do you know the macro that rings your phone?
18:57.28sivanaSpaceBass: if so, you can remove the t or T from the dial()
18:57.40*** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV)
18:57.42SpaceBasssivana thats what I'm trying to avoid editing
18:57.56SpaceBassdamn... times like this that A@H is a real pain
18:58.09sivanaI don't use A@H.. so I dunno
18:58.17sivanaI pretty to be able to do things :)
18:58.19sivanaI prefer
18:58.44sivanayou can search for dial( string
18:59.04robl^A@H is useful for a quick base install..  then forget AMP and do it all by hand with custom config files .. :)
18:59.22DandanI used A@H to learn and study config files :)
18:59.22LostFrogrobl^: That's what I did.
18:59.23SpaceBassrobl^ thats pretty much my philosophy
18:59.40SpaceBassbut some of the scripts are so embedded that its hard to manually configure
18:59.50[TK]D-FenderAll * GUI = shit-on-a-stick.  Thats my story... AND I'M STICKING TO IT>
18:59.52*** join/#asterisk tris (i=tristan@camel.ethereal.net)
18:59.52SpaceBassfound it
19:00.01*** join/#asterisk rezEdit (n=rezEdit@zapdos.omnigroup.com)
19:00.21SpaceBassnice... /etc/asterisk/extensions_additional.conf
19:00.30SpaceBasshad a dial string var option
19:00.38SpaceBasswooo pannic over
19:00.41SpaceBassssl cert on its way
19:00.43robl^SpaceBass:  I delete ALL the config files...  then install samples from asterisk source and edit away
19:00.59file[TK]D-Fender: that's a great story
19:01.09sivanaheh
19:03.07[TK]D-FenderI use ScopServ at work, and have helped people get away from A@H & AMP.......
19:03.28SkramXHi all.
19:05.45SpaceBassI hear a lot of knocks agains A@H but frankly I am at home and it works great
19:08.49[TK]D-FenderIt does work, but ... ICK.  the point of * is control, and A@H turns * into another cookie-cutter PBX ableit free.
19:09.09[TK]D-FenderAnd makes nasty to maintain and volatile code.
19:09.35file[TK]D-Fender: !!!
19:09.35*** join/#asterisk asterisk99 (n=dunoowhy@modemcable169.194-130-66.mc.videotron.ca)
19:09.41asteriskmonkeycan anyone tell me if there is a difference between the 406 and 411 quad t1 card from digium other than the voltagwe?
19:10.09robl^SpaceBass:  A@H is really good for experimentation or a VERY VERY basic setup.  Its a pain to upgrade.  It limits flexibility.
19:10.26asterisk99anyone here good at zaptel boot-up (script) procedures (gentoo specifically)
19:10.39[TK]D-Fenderfile: !!!
19:10.40asterisk99(that was a question)
19:10.45file[TK]D-Fender: ...hi
19:14.40jarrodwhere can i get latest polycom ip500 sip firmware
19:15.12asteriskmonkeythe polycom website :P
19:15.17Seldon1975http://www.freedomphones.net/polycom/files/
19:15.22Seldon1975get them from there
19:15.35Seldon1975I believe that is ok for ip500
19:15.44Seldon1975better check
19:16.36Dandancan someone call me over fwd?
19:16.58rezEditasterisk99: I wouldn't say 'good' but what's the issue...?
19:16.59*** join/#asterisk hamish (n=hamish@196-28-87-71.wdsl.co.za)
19:18.26asterisk99rezEdit: after boot, asterisk fails to load... if I manually run ztcfg -vv and the asterisk, it starts up fine    How do I figure out if the zaptel startup script is executing OK or getting an error?
19:19.08rezEditwell ztcfg does not load zaptel
19:19.29hamishAnyone using a BRI hfc card on ubuntu breezy?
19:19.30asteriskmonkeyis there a prituner like fxotune?
19:20.04rezEditit only parses the config.  As long as ztcfg funs fine you should try running the zaptel script
19:20.06asterisk99rezEdit: I know... but there's a startup script(s) somewhere that loads ztcfg and asterisk
19:20.25rezEdityes... /etc/rc.d/init.d/zaptel
19:20.26truescotasterisk99 it tells you whether it loaded or failed in the boot up when its loading all the driver
19:20.53*** join/#asterisk nagl (n=nagl@213.235.241.6)
19:20.54asterisk99rezEdit: does ztcfg have to execute every time after boot?
19:21.11rezEditthe zaptel script runs it
19:21.49rezEditbut it doesn't have to....  I worked aorund some bugs in an older version I was using by altering the script to ingore the failure it was getting from ztcfg
19:22.06*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
19:22.27rezEdityou should also be able to run the zaptel script using /sbin/service zaptel start
19:23.26asteriskmonkeyany digiums people around?
19:23.40mog_worknope
19:23.44mog_workwe are hiding
19:23.46mog_workerrr working
19:24.22asteriskmonkeymog_work: do you know if there is anything to autotune a pri like fxodoes on the pots cards
19:24.23Seldon1975does /etc/rc.d/rc.local get run after boot?
19:24.38mog_workyou cant do that
19:24.41mog_workas its impossible
19:24.44mog_worksorries
19:25.00rezEditasterisk99: so does the zaptel script run OK for you?
19:25.13asteriskmonkeymog_work: :( is there any tools available for aiding in setting right tx rx gain levels?
19:25.46mog_workztmonitor
19:25.56mog_workis levels wrong
19:25.58mog_workor whats going on
19:26.13asterisk99rezEdit: (I'm using gentoo btw) You way ztcfg does not need to run before asterisk?
19:26.18asteriskmonkeymog_work: when setting up the te110p cards most people have mad echo until they get there levels spot on
19:27.07mog_workbah
19:27.09truescoti gotta say i goy a te110p card and have no problems with echos and tx and rxgain are set to default
19:27.18mog_workusing tx and rx are the worst way to attack echo
19:27.21mog_workthey are last resort
19:27.27rezEditasterisk99: the zaptel script needs to run before asterisk, and the zaptel script runs ztcfg
19:27.33mog_worki dont know why everyone hits em first
19:27.52asteriskmonkeymog_work: running asterisk 1.0.9 with 1.2 zaptel.. and kb1 echo can still have minor echo issuse occasionally
19:28.19rezEditasterisk99: what I was saying is that I modified the zaptel script to NOT run ztcfg because there was a bug that was causing errors, and even though everything should have loaded fine, because of the bug it did not.
19:28.20mog_workhrmm you bump up the echocanceling to 256?
19:28.29mog_workor try mg2?
19:29.09asterisk99rezEdit: I found /etc/init.d/zaptel and in it is a command [ -f /sbin/ztcfg ] !! exit 0
19:29.18rezEditasterisk99: The szaptel script basically just runs modprobe zaptel and then modprobe again for each additional driver that needs to load.
19:29.46asterisk99rezEdit: Thos are not !!, they are really the long-stroke   (my keyboard has been remapped on me)
19:29.56rezEditasterisk99: It sounds like ztcfg runs fine for you, so don't worry about that
19:29.58asteriskmonkeymog_work : sec ill pastebin you my zaptel its often far end that hears echo after sometime
19:30.28asteriskmonkeyechos alot on iaxys over 80ms away
19:30.33mog_workokay
19:30.34asterisk99rezEdit: it runs --- but only manually     It looks like it's not running on boot
19:30.50rezEditasterisk99: if anything is failing for you, it seems it's the zaptel script itself.
19:31.02asterisk99rezEdit: how can I tell? Is there a log somewhere?
19:31.07brad_msswthink iaxy's have a tendency to echo ... the single iaxy we have is the only device that echos ...
19:31.20rezEditasterisk99: so do you see a message at startup that zaptel is loading?
19:31.37asteriskmonkeymog_work: http://pastebin.ca/33692
19:31.38rezEditasterisk99: Maybe in boot.log?  or messages.log?
19:31.41brad_msswall our sip phones, and zap (TDM400P FXS connected) phones are fine :/
19:31.49asterisk99machine is 600 miles away... I use Putty for the console
19:31.56asteriskmonkeykb1 is an improvment of mg2 is it not?
19:32.07mog_workbah iaxys dont echo...
19:32.15mog_workmg2 is improvement of kb1
19:32.17truescotmog_work can you tell me if the correct line for my zaptel.conf should be, 1,0,0,ccs,hdb3,crc4 or 1,1,0,ccd,hdb3,crc4 i have a te110p with an e1 line from the telco plugged in
19:32.32mog_workyou want the 1,1 as you are taking timing from telco
19:32.42truescotok cool
19:33.16mog_worki would get rid of gain asteriskmonkey and set echocancel=256
19:33.36truescoti have been bangin my head off the walls at work trying to find out the problem and only just realised that it should prob be 1,1 but now not in front of the machine to check so thanks for putting my mind at rest :)
19:33.45mog_workno prob
19:34.44*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
19:34.53asteriskmonkeymog_work: had that before and it slapped echo no matter what really bad i think :P its a production box so not much time to play this early in day :)
19:35.06asteriskmonkeyshould i just upgrade the whole thing to 1.2.1?
19:36.04asterisk99rezEdit: hmmmmmm I am looking at /var/log/messages   there are messages ... ''rcscripts: /sbin/rc-update /etc/init.d/ztcfg not found; aborting''     That must be it!!!!
19:36.05truescotbtw i changed echocancel in zapata.conf to echocancel=no and my line sounds much better for it
19:36.33asteriskmonkeytruescot: some echo cans must be in place from your provider then..
19:36.34mog_workprobably but it wont change to much
19:36.47asterisk99rezEdit: typo above ... that was rc-scripts
19:36.58asteriskmonkeymost of my clients use the iaxys
19:37.09asteriskmonkeyi see there is some new updates for those things in the 1.2.1
19:37.24rezEditasterisk99: yes, probably.  you need to either specify the fill path in the script or add the path to your $PATH env var
19:38.01asterisk99rezEdit: that would be in /etc/init.d/zaptel ??
19:38.11Seldon1975does anyone know a Windows program that plays Asterisk gsm sound files.  I've been looking for this myself but come up dry
19:38.23asteriskmonkeyjust get the codec
19:38.44mog_workquick time will
19:38.52*** join/#asterisk _-Jon-_ (i=jon@CPE00112f6dfbee-CM00111a232a80.cpe.net.cable.rogers.com)
19:38.55mog_workand you can do wav47 or whatever it is
19:39.00mog_workthat is gsm for windows like codec
19:39.14_-Jon-_hey can someone help me with a nat problem?
19:39.16asterisk99rezEdit: the line has a full path... /sbin/ztcfg    (that fiule exists)
19:39.19rezEditasterisk99: uh wait... I am not sure what rc-scripts is doing there.
19:39.32_-Jon-_Our asterisk server just seems the client (which is behidn NAT) as coming from it's internal IP
19:39.36asterisk99rezEdit: that makes two of us :)
19:40.01rezEditasterisk99: yeah... one sec
19:40.11truescotdoes anyone know of a good dect interface for asterisk that allows roaming?
19:40.13asterisk99rezEdit: 'spose I edit it to /usr/src/zaptel/ztcfg  ?
19:40.33brad_mssw_-Jon-_: did you set nat=yes in sip.conf ?
19:40.34rezEditasterisk99: no, that won't do anything.
19:40.54_-Jon-_brad_massw, I sure did
19:40.55brad_mssw_-Jon-_: also, if your SIP device supports STUN, it may help as well
19:41.01Seldon1975asteriskmonkey: 'just get the codec' can you tell me where from?
19:41.44_-Jon-_brad_mssw, one step ahead of you :)  Tried testing out the client to my personal Asterisk server at home and it worked fine, so it's got to be a problem with the server
19:42.03_-Jon-_The only difference that I can tell is the one that isn't working is the stable build, and the one I'm running is the latest head
19:42.58*** join/#asterisk numbshot (n=numbshot@dsl-201-133-74-110.prod-infinitum.com.mx)
19:43.22rezEditasterisk99: what happens when you run /etc/init.d/zaptel as root?
19:43.25*** join/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net)
19:43.35*** join/#asterisk sysdebug (n=sysdebug@200.163.193.247)
19:43.49asterisk99rezEdit: Permission denied
19:44.20rezEditasterisk99: uh!?! running it as root?  is the file executable?
19:44.31fugitivoof course not
19:44.37asterisk99rezEdit: it's -rw-r--r--
19:44.40fugitivo;)
19:44.48asterisk99rezEdit: chmod 777 it?
19:44.50fugitivoasterisk99: chmod +x it
19:45.13rezEditasterisk99: yeah what fugitivo said!
19:46.15asterisk99rezEdit: got Line 23  /etc/init.d/functions: No such file or directory
19:46.36*** join/#asterisk sysdebug (n=sysdebug@200.163.193.247)
19:47.44rob0777: the executable privileges of the Beast
19:48.21asterisk99rezEdit: the script is plugged  ''system=redhat'' and line 22-24 read is [ $system = 'redhat'] ; then       ./initdir functions      fi
19:48.23rob0asterisk99: what OS?
19:48.39asterisk99rob0: gentoo
19:48.57_-Jon-_So no other ideas?  I set localhost=192.168.2.0/24, set externip, set nat=yes and canreinvite=no, but still no go..  reports something about max retries and then @192.168.0.99
19:50.02rob0did you "emerge asterisk"?
19:50.17paryli have a channel bank, and a fax machine with two lines suddenly stopped making outgoing calls.  every other machine is fine, and this machine will take incoming calls.  outgoing gets congestion tone immediately
19:50.24fugitivoasterisk99: what's the problem?
19:50.34*** join/#asterisk m160858 (n=jsaenz@200.89.12.46)
19:50.55asterisk99fugitivo: looks like ztcfg fails to execute on boot
19:51.01brad_mssw_-Jon-_: hope you mean  localnet not localhost
19:51.08rob0fugitivo: asterisk99 is trying to run a RedHat-style init script in Gentoo
19:51.10brad_mssw_-Jon-_: plus, did you set externip  ??
19:51.19fugitivoasterisk99: did you emerge asterisk or manually install?
19:51.30rob0and there is no /etc/init.d/functions file to source
19:51.39rob0that is what I asked too
19:51.42Ikarusrob0: try selling him a gentoo training or funrolloops
19:51.56rezEditasterisk99: yeah, I am using a Red Hat derivitave here (Yellow Dog) so I will not be much help.
19:51.58m160858hi everyone
19:52.09fugitivoasterisk99: if you emerged asterisk, gentoo installs it's own init scripts
19:52.10_-Jon-_brad_mssw, err yes, localnet, my mistake.  and yes, externip is set.  I just dont' get it..  when I do 'sip show peers' it shows the external IP of the client, but when the errno max retries error comes up, it shows it's internal
19:52.30asterisk99fugitivo: I tried emerge at first ... then later I did the CVS install
19:52.49m160858I need that you recommend a adaptor to me
19:52.50fugitivoasterisk99: ok, then you should use the right init scripts, not redhat init scripts
19:52.59brad_mssw_-Jon-_: and you have   host=dynamic and nat=yes  in sip.conf for your device  ....
19:53.13m160858which adapter is better than the SPA ?
19:53.24asterisk99fugitivo: ok... sounds like a game-plan... where can I download such a beast?
19:53.28brad_mssw_-Jon-_: that should really be all there is to it ... though I'm running asterisk 1.0.10  ... hopefully migrate to 1.2 this weekend
19:53.30fugitivoasterisk99: check your contrib/init.d/rc.gentoo.asterisk script in asterisk sources
19:53.31JonR800_-Jon-_: set nat to yes on the client.
19:54.21_-Jon-_JonR800, that strange thing is, when I connect to a different Asterisk server the same client works.  So I'm think it's an issue on the server side, BUT why would X-Lite report it's internal IP int he first place
19:54.39asterisk99fugitivo: can you give me the full path to that please?
19:54.44JonR800because nat is set to no, so it thinks it's registering locally.
19:54.53fugitivoasterisk99: no, because i don't know where are your asterisk sources
19:55.04slak-hey
19:55.05Seldon1975hey guys
19:55.07fugitivoasterisk99: for example /usr/src/asterisk/contrib/init.d/rc.gentoo.asterisk
19:55.15Seldon1975i think someone is hacking our switch
19:55.17JonR800the other asterisk servers may have ser or some other proxy?.. something to do the footwork themselves and fix up broken clients.
19:55.20Seldon1975can someone have a look at this: http://pastebin.com/464161
19:55.20slak-in exten => 106,1,Macro(remote,${NICKOFFICE})
19:55.21slak-exten => _1XX,1,Macro(oneline,${EXTEN})
19:55.25slak-which will match first
19:55.26slak-?
19:55.29slak-the 106?
19:55.34Seldon1975does it look like an incoming call bridged to the outside world?
19:55.36Seldon1975http://pastebin.com/464161
19:55.41*** join/#asterisk genmud (n=genmud@ip68-98-82-206.ph.ph.cox.net)
19:55.54JonR800_-Jon-_: all else fails do a sip debug and get an idea of what's going on.
19:56.53m160858hello?
19:57.32*** join/#asterisk sshadow (n=sshadow@213-84-101-107.adsl.xs4all.nl)
19:57.50m160858which adapter is better than the SPA ?
19:57.53asterisk99fugitivo: I found that... it's pretty tiny... 3 lines for start() { ebegin "Starting Asterisk"   /usr/sbin/asterisk    eend "Failed to start Asterisk"
19:58.11fugitivoasterisk99: what else do you need?
19:58.28asterisk99fugitivo: there's no line for ztcfg
19:59.08fugitivoyou don't need that
19:59.32slak-heyyy
19:59.32fugitivotell modprobe to load the module and then /sbin/ztcfg
19:59.37slak-come on people
19:59.38asterisk99fugitivo: does ztcfg not need to run always before asterisk?
19:59.38fugitivofor example
19:59.43slak-which one gets matched first
19:59.43fugitivoinstall wctdm /sbin/modprobe --ignore-install wctdm && /sbin/ztcfg
19:59.50slak-exten => 106,1,Macro(remote,${NICKOFFICE})
19:59.50slak-exten => _1XX,1,Macro(oneline,${EXTEN})
19:59.58slak-i need 106 to match first
20:00.03fugitivoasterisk99: that's a single line of modprobe.conf
20:00.04slak-its using a diff macro
20:00.37fugitivoslak-: 106 first
20:00.42slak-no matter what?
20:00.44slak-always?
20:00.54fugitivoslak-: if you dial 106, it'll match with 106
20:01.03slak-but it also matches _1XX
20:01.17fugitivowell, test it yourself
20:01.59*** part/#asterisk sshadow (n=sshadow@213-84-101-107.adsl.xs4all.nl)
20:05.59rob0asterisk99: I bet it's the path to asterisk. Default $PREFIX is /usr/local, so change that to /usr/local/sbin/asterisk
20:06.52asterisk99rob0: how can I see the path?
20:07.14fugitivoasterisk99: do what i told you, it should work
20:07.45rob0asterisk99: you REALLY should go through the Gentoo user guides!
20:08.06asterisk99fugitivo: I tried that command     install: unrecognized option '--ignore'
20:08.13rob0as root do "which asterisk"
20:08.13fugitivoasterisk99: lol
20:08.14asterisk99rob0: yes, I should
20:08.26fugitivoasterisk99: you should put that lines in your modprobe.conf
20:08.40fugitivoasterisk99: so at boot time, it'll load the right modules, and then do a ztcfg
20:08.56*** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br)
20:09.27*** join/#asterisk hhoffman (n=hhoffman@tor/session/x-b198c2110185c500)
20:09.44fugitivoasterisk99: if you want, i have the init script for zaptel used by gentoo
20:10.03hhoffmanhi, just bought a pap2 from staples and trying to flash it... seems to require a admin password... any idea as to what the password is?
20:10.17fugitivohhoffman: blank?
20:10.18rob0asterisk99: so many things are going against you. I am an experienced Linux admin, know my way around /bin/bash very well, and asterisk is a challenge for me.
20:10.22sivanahhoffman: chances are it's locked to Vonage
20:10.39sivanahhoffman: the default I think is admin/admin?
20:10.40rob0but it's fun :)
20:10.50fugitivodefault is admin/blank
20:10.55asterisk99fugitivo: 1. modprobe.conf says not to edit it (says it's generated by modules-update)
20:11.10asterisk99fugitivo: 2. can I see your script?
20:11.29fugitivoasterisk99: http://pastebin.com/464187
20:11.47hhoffmansivana: yeah, it's locked... I'm trying to unlock it... admin/admin doesn't work :-(
20:11.55hhoffmanneither does admin/blank
20:12.07fugitivohhoffman: you should buy the pap2-na
20:12.12fugitivohhoffman: that's unlocked
20:13.34hhoffmanthey're a bit tough to get a hold of
20:13.43asterisk99fugitivo: that file is saved as /etc/init.d/zaptel ??
20:13.53sivananot if you get authorized with a supplier
20:14.10fugitivoasterisk99: yes, then do rc-update add zaptel default
20:17.02parylwhy would i be able to dial into a Zap channel but not out of it?
20:17.24sivanawhat's your Zap channel.. PRI or just a Bell line?
20:17.42*** join/#asterisk bangawanga (n=andhecke@ppp-82-135-14-116.mnet-online.de)
20:18.08bangawangadid anybody succeed configuring asterisk with ser and livecomm??
20:18.14Seldon1975this may seem like a basic question, but how do I check my voicemail messages on the * server from my SIP andset?
20:18.28sivanaSeldon1975: create an extension to VoicemailMain()
20:18.29Seldon1975handset
20:18.37Seldon1975aha
20:18.40Seldon1975ok
20:18.48Seldon1975thx
20:19.48asterisk99fugitivo: K ... did as you suggested... rc-update add zaptel complains zpatel already installed in runlevel default; skipping
20:20.09asterisk99fugitivo: K ... rc-update del zaptel ??
20:20.10fugitivoasterisk99: ok, then reboot and test it
20:20.19fugitivono, leave it
20:20.34asterisk99fugitivo: ok - here goes nuttin'
20:21.23parylsivana: it's a te205p connected to a rhino channelbank
20:21.29*** join/#asterisk razu (n=razu@217-159-240-134-dsl.est.estpak.ee)
20:22.08bangawanganobody who has the challenge to get those programs working together?
20:24.18*** join/#asterisk bangawanga (n=ahecker@ppp-82-135-14-116.mnet-online.de)
20:24.29asterisk99fugitivo: did not work - had to manually do ztcfg and asterisk after boot
20:24.54fugitivoasterisk99: what does it say?
20:28.11asterisk99fugitivo: message .... /etc/init/d/zaptel   syntax errors
20:28.16bangawangais there anybody who tested asterisk <> ser <> livecomm 2005 ?? ser can translate between tcp<->udp, right?
20:28.48asterisk99fugitivo: i think I see it
20:28.49Seldon1975can I get something straight; if you bought an echo cancelling card and you enable echo cancelling; should you still expect to hear echo?
20:29.07Seldon1975is it generally accepted that you cant be completely free of iut?
20:29.09asterisk99fugitivo: vestige of HTML
20:29.20*** join/#asterisk mtupper (n=mtupper@pc-14-172-104-200.cm.vtr.net)
20:30.10mtupperanyone here successfully call SIP > PSTN using * and a spa-3000?
20:30.37asterisk99fugitivo: I fiexed it, and if I manually do a ./zaptel start, I still get syntax error
20:31.01asterisk99fugitivo: or..... is that a script I can't run on my own?
20:31.11*** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de)
20:31.32*** join/#asterisk Igbothom_III (n=HiltonT@203-206-114-129.dyn.iinet.net.au)
20:32.39Seldon1975mtupper yes
20:32.43Seldon1975mtupper: i have
20:33.23m160858which adapter is better than the SPA ?
20:33.26mtupperSeldon1975, i beg you for a quick tutorial!   I am at my wits end...
20:33.37Seldon1975i had trouble with it too,
20:33.50*** join/#asterisk |cleric| (n=dacleric@p54829A91.dip0.t-ipconnect.de)
20:33.55Seldon1975until i found out that you HAVE to turn use authentication with the * server
20:34.07fugitivoasterisk99: if i don't know the error, i can't help you
20:34.14*** join/#asterisk nagl (n=nagl@213.235.241.6)
20:34.19Seldon1975ie: set a username and password on the SPA3000 that matches username and secret in your SIP.conf for that device
20:34.33parylwill reload chan_zap.so reset the active zap channels?
20:35.11*** join/#asterisk nagl (n=nagl@213.235.241.6)
20:35.26hugo-v6ok. found out how to get cid screened on outgoing calls with misdn.
20:35.36mtupperSeldon1975, forgive me for being such a rookie, but could you specify more there...  where do I modify this?  in which .conf?
20:36.06Seldon1975ok first you need to set your configuration on the SPA3K by using a browser to visit the IP address of the SPA device
20:36.11mtupperSeldon1975, by the way, I am using astbill on top of my asterisk...
20:36.27Seldon1975there you can go to Advanced settings and sell it where your SIP server (*) is
20:36.36Seldon1975i have no idea about Astbill
20:37.08Seldon1975after that you just need to edit Asterisk's sip.conf and add an entry there corresponding to the SPA3K's IP address
20:37.24asterisk99fugitivo: syntax error
20:37.27mtupperSeldon1975, yes, I have configured the spa-3000 about 50 times in the last 24 hours...  I can successfully call  SIP > SIP from the phone plugged into the SPA to my x-lite softphone...  but I cant for the life of me get my SPA to use the PSTN line...
20:37.55Seldon1975hmm
20:37.57Seldon1975how odd
20:38.05hugo-v6i guess thats better than every existing tool
20:38.07Seldon1975I have no issue calling PSTN
20:38.08asterisk99fugitivo: ''/etc/init.d/zaptel' has syntax errors in it; not executing...''
20:38.23Seldon1975mtupper I didnt need to do any thing 'special' to achieve this
20:38.35sylerrdtool is kewl, used that like 5 years ago, i thought there would be something better by now :)
20:38.49Seldon1975mtupper do you have other SIP devices calling PSTN lines on your network?
20:38.50hugo-v6Seldon1975: its more reliable
20:38.53Nivexsyle: sometimes you just get something good an it endures
20:38.54asterisk99fugitivo: 'i found it!!!!!!!
20:39.02mtupperSeldon1975, not yet
20:39.05hugo-v6syle: nothing better than rrdtool :)
20:39.15Seldon1975ok so your issue may be nothing specific to the SPA3000
20:39.20asterisk99fugitivo: one & became &amp; ... I didn;t see it
20:39.31sylegood to know
20:39.34sivanaI need an AGI script that query's a databas and returns 1 or 0... who wants to do it
20:39.56Seldon1975mtupper: but you CAN make SIP-SIP calls on your network?
20:40.45hugo-v6sivana: write it yourself. not that hard
20:41.00mtupperSeldon1975, yes, I have completed many calls from SIP to SIP via combinations of softphones and SPAs (1001, 2100 and 3000)...
20:41.00sylesivana: perldoc DBD::mysql
20:41.01sivanahugo-v6: if I wanted to do that, do you think I would have posted it here
20:41.07sylebig 5 lines of code to do that
20:41.08sivananame your price
20:41.36hugo-v6sivana: well whats to do exactly? i get 75euros/h + taxes
20:42.13*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
20:42.16hugo-v6syle: shhhhhh!
20:42.18sivanaI need to query a table with the CID area code and return 1 if its canada or 0 if not
20:42.30hugo-v6syle: i would bill him 1h fpr each line ;)
20:42.35sylelol
20:42.42shmaltzanybody here know what is normal to pay for an independed agent for mr charges?
20:43.01*** join/#asterisk asteriskmonkey (n=phil@69.156.197.242)
20:43.28hugo-v6sivana: well ull see we're not that cheap here ;) better do it yourself. and its really easy
20:44.00sivanaalright.. clearly you're not interested your too busy
20:44.10syledepends
20:44.12sylehow much?
20:44.27sivanaI need an AGI script that query's a databas and returns 1 or 0... who wants to make a couple of bucks (besides hugo-v6)
20:44.27hugo-v6syle: dont do it cheaper or i d0s you ;))
20:44.49syle#DEFINE coupleofbucks
20:44.59sivanaif it's such an easy script.. should be easy money for you.. $25-30
20:45.09asteriskmonkeymog_work: does the iaxys only update there bin when its a new asterisk or when a newer .bin file is found... what is the process also.. when an iaxy connects does it do a request for a bin update?
20:45.17*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
20:45.20hugo-v6i dont understand why u dont write it yourself?
20:45.22sylelol, i wouldn;t bother doing anything for 30 bucks
20:45.34sivanasyle: is it really worth my time to spend the 2-3 hours learning Perl?
20:45.42sivanano.. my time is better spent making money
20:45.43hugo-v6lol
20:45.50mog_workyes
20:45.56hugo-v6sivana: then pay the price
20:45.59asterisk99fugitivo: BUDDA BING BUDDA BOOM!!!!!     That did the trick!!!!!! :)    Thanks fugitvo and rob0 ... I owe you BEERS!!!!! :)
20:46.01puzzledevening all
20:46.06Seldon1975mtupper have you configured your fxs modules in zapata.conf?
20:46.21mtupperSeldon1975, NO!!!
20:46.23asteriskmonkeydoes anyone here have a te406p?
20:46.23sivanahugo-v6: you've yet to give me a serious price...
20:46.31Seldon1975mtupper you gotta do that
20:46.37*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
20:46.39Seldon1975mtupper to make PSTN calls
20:46.41mtupperlet me look at that, can you add more details...
20:46.52sylethen i;d suggest you don;t insult people with 30 dollars :)
20:47.00sylei think people in here are worth more than that
20:47.05asteriskmonkeyis kram around?
20:47.24mog_workin france
20:47.30sivanasyle: are you kidding me... it's a simple script
20:47.36mog_workor actually he is probably somewhere over the ocean
20:47.38hugo-v6sivana: well. i got enough todo in my workinghours which brings 75euros/h + taxes (own comp.) so ill better do my own job ;)
20:47.43Seldon1975I have configured all my fxs modules (18-24) to use fxs_ks signalling, you might as well try that
20:47.47sivanahugo-v6: ya, you should
20:47.49*** join/#asterisk trym (n=trym@062016209171.customer.alfanett.no)
20:47.52asteriskmonkeymog_work: did you read my msg above re the the iaxys?
20:47.53mog_workwhat do you want sivina?
20:47.56mog_workyes
20:47.59mog_workthey will autoupdate
20:48.12asteriskmonkeybased on what the bin or asterisk version
20:48.17mog_workbin
20:48.19sivanamog_work: I need a simple script to query a table with the first 3 digits of CID and return 1 if it's canada value or 0 f not
20:48.25sivanaagi script
20:48.36Seldon1975mtupper then you need a rule in your dialplan (extensions.conf) to route outgoing calls throuugh those Zap iterfaces which represent your PSTN lines
20:48.46mog_workwhy does it need to be agi
20:48.50mog_workand what kind of tablee
20:48.53asteriskmonkeysivana: just make a php agi script :)
20:48.56sivanait doesn't
20:48.56mog_workyou can do it in dialplan most likely
20:48.57sivanalol
20:49.04sivanapostgresql table
20:49.33mog_workis there app_postgres in addons?
20:49.37mog_worklike app_mysql?
20:49.40filemog_work: !!!
20:49.53mog_workfile: !!!!!!!!!!!!!!!!!!!!!
20:50.00sivanaI don't really want to load another module just for that one call
20:50.25mog_workagi is much more work than a module
20:50.30mog_workits simple logic
20:50.42mog_worki dont generally do agi, as C agi just seems dumb
20:50.55fileC agi
20:50.56mog_workand I only write in c or dial plan for free these days
20:50.59mog_workit exists
20:50.59syleyou smoking dope?
20:51.03mog_worki had to do it for someone
20:51.03sivanaI'm not asking for free
20:51.06sivananever did
20:51.14sivanahehe
20:51.16mog_workits a conflict of intrest
20:51.21mog_workyou can pay digium
20:51.22mog_workto pay me
20:51.27mog_workbut that can be expensive
20:51.30mog_workas im not cheap
20:51.31sivana:)
20:51.47filemog_work: you're not a cheap programmer whore? :(
20:51.52sylehigher than a 2 cent hooker?
20:52.01mog_workim more of a programming slut
20:52.04mog_workill give it away
20:52.12mog_workbut offering me cash insentives is pointless
20:52.39mog_workmmm muffinlicious
20:53.03filemog_work: did you get one of the cookies I sent Mark?
20:53.12mog_workyou sent mark cookies
20:53.14*** join/#asterisk nagl (n=nagl@213.235.241.6)
20:53.19mog_worki cant believe he didnt share.....
20:53.20asteriskmonkeyim cheap ill work for food/sex
20:53.26mog_worklol
20:53.39sylei think he should jsut get billing software anyways, from what he described hes just selling tollfree numbers based on a CDN or US rate
20:53.46sylenot much more
20:54.31hugo-v6asteriskmonkey: well. for sex is a point... but what if its your worst dreams type?
20:54.41mog_workeh its not hard
20:54.45mog_workbut shouldnt be agi
20:54.56sylei have it done in c
20:54.59syleas a module
20:55.10mog_workwell you dont even need a specific
20:55.11mog_workone
20:55.18mog_workisnt there app_postgresql
20:55.31mog_workor did i imagine it
20:55.37hugo-v6mog: afaik ther is at least a app_mysql
20:55.40syleyeah its in the source tree
20:55.44mog_worki know of app_mysql
20:55.49syleunder cdr directory i beleive
20:55.58*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
20:56.00asteriskmonkeyhugo-v6: if she be hidious its sex pluse tdmxxx or texxx cards
20:56.04mog_worki just wasnt sure if there was app_odbc and app_psgql
20:56.18harryvvis there any DID providers for the 604 area code?
20:56.29*** join/#asterisk lesouvage (n=lesouvag@82.74.114.137)
20:56.35sivanasyle: it's much more than that
20:56.39asterisk99My next big question is... how can I use something like cronolog to switch Asterisk's logs daily at midnight?   Or if not cronolog, what?  :) :) :)
20:56.46hugo-v6asteriskmonkey: hrhr
20:57.04sylesivana is there? tollfree is only thing i know of that uses callerid of incomming call
20:57.21syleor maybe some attempt at a calling card solution
20:57.32sivanasyle: any PSTN interface sends CID
20:57.33mog_workthere is cron rotate or something
20:57.47lesouvageI have this output in the cli: "Dec 14 15:54:52 WARNING[3247]: chan_oh323.c:4147 oh323_gk_check: Gatekeeper discovery failed." What should I do to get rid of this h323 isn't used.
20:57.50hugo-v6sivana: not in .de
20:57.52mog_workumm no sivana
20:57.55syleyes but normally you route based on what they called not where they're calling from
20:57.59mog_workmost CAN send
20:58.01mog_worknot all do
20:58.04sivanatrue
20:58.07*** join/#asterisk nagl (n=nagl@213.235.241.6)
20:58.12asterisk99syle: Inbound CallerID is useful... I use it to route calls to different extensions
20:58.19sivanasyle: you have no idea the purpose of the script :P
20:58.23*** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt)
20:58.45mog_worklogrotate
20:58.51slak-hey what does Answer do
20:58.52mog_workthats what its called i thought
20:58.56slak-is it harmless?
20:58.58mog_worklol
20:59.00mog_workdepends
20:59.01parylwhat does "progress with cause code 127" mean?
20:59.04mog_workbut mostly yes
20:59.11sylewell talking about asterisk not SER, yes you;d use SER to route calls wherever on what areacode dialed
20:59.20slak-mog_work: i have Answer as the s,1 in my macro for local extensions
20:59.28slak-s,1,Answer
20:59.32slak-s,2,Wait
20:59.33mog_workthats harmless
20:59.36*** join/#asterisk limo (n=limo@limonet.opf.slu.cz)
20:59.38*** join/#asterisk file[desk] (n=jcolp@mctnnbsa24w-142167049176.nb.aliant.net)
20:59.39slak-s,3,Dial
20:59.49mog_workharmless
20:59.49slak-why would one want an Answer there
20:59.54harryvvtraditional pbx passes cid?
21:00.07mog_workits a just in case
21:00.07*** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk)
21:00.19slak-ok
21:00.23sylebtw what are you guys using for SER stateless redirects or stateful?
21:00.46Seldon1975how cna i stop a specific Zap channel
21:00.51asteriskmonkeydo alot of people use ser in conjunction with asterisk?
21:00.59Seldon1975i cant do 'zap stop 19'
21:01.00filesyle: depends how evil you get...
21:01.23limoplease can somebody help me with message "Bridge stops bridging" ?
21:02.10syleasteriskmoney, anyone big does
21:02.10KattyFILE
21:02.33paryl"progress with cause code 127" brings up 4 results on google, all from the same thread :\
21:02.42*** join/#asterisk santiago (n=santiago@208.195.215.154)
21:02.48paryland none of them really explain anything
21:03.00syleits fairly new to me, but i;m quite impressed at their c code
21:03.37fileKATTY
21:03.45Kattyfile: hi.
21:03.52fileKatty: hi.
21:04.02*** join/#asterisk nagl (n=nagl@213.235.241.6)
21:04.41*** join/#asterisk nagl (n=nagl@213.235.241.6)
21:04.45*** join/#asterisk stoffell (n=stoffell@d51A4D148.access.telenet.be)
21:05.37Kattywow, i've not gotten any email in 5 hours.
21:05.40Seldon1975ive been looking for ages and still cant find a GSM codec for windows to play the asterisk sound files
21:06.04*** join/#asterisk loick (n=loick@APuteaux-151-1-74-135.w83-204.abo.wanadoo.fr)
21:06.10fileyup
21:06.12fileit's all my fault
21:06.16Kattyi /knew/ it!
21:06.20syleecho "here you go baby" | mail katty@pornstar.com
21:06.38Katty...
21:07.30*** join/#asterisk Seldon1975 (n=someone@CPE0013105d0913-CM0014e8b6162c.cpe.net.cable.rogers.com)
21:09.28sylepeople still use windows for mail :(
21:09.35Seldon1975can someone tell me where the vulnerability is in this dialplan?  it seems that someone is dialling in, and then dialling out onto the PSTN again somehow: http://pastebin.com/464260
21:09.41gnosyscould someone point me (again) to that config & xml zip file for polycom phones?
21:10.29Seldon1975gnosys: http://www.freedomphones.net/polycom/files/
21:10.51gnosysthx!
21:10.57*** join/#asterisk EriSan (n=erisan@81-174-42-85.f5.ngi.it)
21:11.11Seldon1975np
21:11.24Seldon1975now maybe someone will look at my dialplan issue
21:11.31Seldon1975http://pastebin.com/464260
21:11.39Seldon1975D-Fender?
21:11.43Seldon1975asteriskmonkey?
21:13.07gnosyswish i could help with your dialplan, but i'm sure it's more sophisticated than I could manage at this point in my learning.
21:13.30Seldon1975;)
21:14.16hamishanyone using hfc based bri cards?
21:14.38*** join/#asterisk bartpbx (n=bartpbx@p54B0411C.dip0.t-ipconnect.de)
21:14.54bartpbxhello
21:15.07fugitivoSeldon1975: well, your dial out logic is inside your default context where your incoming calls go
21:15.24Corydon-wSeldon1975: Yeah, anybody calling in can dial an external extension
21:15.42Corydon-wSeldon1975: you should instead be using a different context for incoming than for internal calls
21:15.42bartpbxhow can i convert an existing wav file to get it accepted by asterisk? I always recive " Unexpected header size 16"
21:15.51Seldon1975ok
21:15.53Seldon1975thanks guys
21:16.03Seldon1975I set up these contexts in Zapata.conf?
21:16.04sylefile: what can i use to compare times, ie if current hour is between 9pm and 6am etc
21:16.12fugitivoSeldon1975: you should create another context for dialing out, and another context for local extensions, include the dialout context in your extensions context
21:16.13filegotoiftime?
21:16.13*** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd)
21:16.16Seldon1975ie: the Zap channel/Context correspondence
21:16.22syleusing time.h
21:16.32sivanamog_work: I'll try the app_sql_postgres.so
21:16.34Seldon1975ok thanks
21:16.35*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
21:16.47Seldon1975do you by any chance have an example of this type of dialplan?
21:16.52Corydon-wNo, you set up contexts in extensions.conf
21:16.58bartpbxoh.. I'm to silly.. I've just fixed it my self.. sorry
21:16.59mog_workawesome
21:17.03fugitivoSeldon1975: check asterisk config samples
21:17.23sylesuppose i could use that builtin function, but i noticed a problem with gotoiftime to that it doesn;t work after 11:59 pm
21:17.33Seldon1975Corydon-W: but dont I map Trunk lines (specific Zap interfaces) to the extensions.conf contexts in zapata.conf?
21:17.42Corydon-wSeldon1975: you already have a separate context for faxes
21:17.54Seldon1975yes; I set that up in zapata.conf too
21:18.03syleso if i say gotoiftime sun-thurs 9pm-8am  its not accurate
21:18.04Seldon1975so that Zap21 uses that context
21:18.05Corydon-wYou map channels, whether or not they're in a trunk or not
21:18.13Seldon1975right
21:18.15*** join/#asterisk zotz (n=zotz@24.231.47.168)
21:18.29Seldon1975so in zapata.conf I mapped Zap/21  to 'faxes'
21:18.40sylewould have to do fri 00-00 to 8am :(
21:18.41*** join/#asterisk cyberjew (n=jhochber@adsl-065-007-156-054.sip.asm.bellsouth.net)
21:18.42tainted-Seldon1975 is this your dayjob or a client?
21:19.06cyberjewdoes anyone have the cisco 6.3 firmware?
21:19.42tainted-cause u might not want to attempt any more production stuff until u get a better understanding of asterisk
21:20.09asteriskmonkeyseldon1975: you rang?
21:20.32Seldon1975asteriskmonkey yes its ok thank others have stepped up
21:21.23SkramXHi all
21:21.27mrinvaderhi all! I have a goofy question - how/where would i go to set the dtmf tone length/spacing? i have a ksu that i believe cant handle digits as fast as asterisk sends them. myasterisk ver is 1.0.9 .
21:22.09asteriskmonkeyseldon1975: ok cool :D
21:22.37cyberjewhowdy y'all...i've got a cisco 7960g here, but i need the 6.3 SIP firmware
21:23.05cyberjewcant find it anywhere
21:23.13syleso login to cisco site and get it
21:23.22cyberjewmy support contract is not setup yet
21:23.37mtupperbit-torrent?
21:23.46cyberjewordered from cdw weeks ago...they are slacking
21:23.56cyberjewand i'm getting frusterated
21:23.59mrinvaderdern slackers :)
21:24.01SkramXheh, hi cyberjew
21:24.04*** part/#asterisk m160858 (n=jsaenz@200.89.12.46)
21:24.37cyberjewany of you be able to help me out?
21:24.52cyberjew<<would
21:25.00parylso i tried the MARK2 echo cancellation and it chopped up the PSTN side, MARK3 is unreliable and sometimes screams at us, MG2 acts as though it's not even there... can anyone suggest another to try?
21:25.09kuku5How can I  make a dial command AFTER 5:30 pm   and a different one after 8:30a m
21:25.14mog_workwe do not promote piracy cyberjew
21:25.15mog_worksorry
21:25.17stoffellparyl, doesn't MG2 help you?
21:25.27mog_workgotoiftime kuku
21:25.32Kattyright. some people need to ping me.
21:25.32kuku5ok
21:25.33parylstoffell, not really... same ol' echo
21:25.33mog_workwhich is now just a function in trunk
21:25.38*** part/#asterisk cyberjew (n=jhochber@adsl-065-007-156-054.sip.asm.bellsouth.net)
21:25.40kuku5mog_work: thanks
21:25.41stoffellwhat phones you using paryl?
21:25.45parylpolcom 501
21:25.46LostFrogI don't understand how companies can charge for firmware updates.
21:25.49parylpolycom*
21:25.50mog_workyou can do show functions to see that
21:25.57mog_workyeah it doesnt make sense
21:26.03syleparyle KB2?
21:26.13mog_workmg2 not kb2
21:26.21sylehe renamed it?
21:26.39asteriskmonkeywhat there is a kb2 now?
21:26.45stoffellparyl, i just read this today, haven't tried it yet : http://sangoma.editme.com/wanpipe-linux-asterisk-debugging
21:27.00mrinvaderwouldnt a firmware update sometimes count the same way as say going from win95 to win98 tho?
21:27.05mog_workno mg2
21:27.10*** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net)
21:27.15kuku5mog_work: what if i dont want to GOTO  - I want to just DIAL
21:27.27*** join/#asterisk Katty (n=angela@68-112-15-110.dhcp.cpgr.mo.charter.com)
21:27.34mog_workthen you want the fuctionstuff
21:27.41kuku5which is what stuff ?
21:27.48Kattymog_work: would you tell me what my ping reply is please.
21:27.48mog_workother wise you are just gonna have to prefix that dial with the gotoitftimes
21:27.48*** part/#asterisk bartpbx (n=bartpbx@p54B0411C.dip0.t-ipconnect.de)
21:27.49asteriskmonkeymog_work: damn had my hopes up
21:28.18mtupperSeldon1975: i could use your assistance again...
21:28.21mog_work2 seconds katty
21:28.25mog_worksorries
21:28.29Kattythanks.
21:28.34*** join/#asterisk backblue (n=moo@87-196-38-64.net.novis.pt)
21:28.38mog_workand kuku do show functions
21:28.41mog_workin asterisk
21:29.11mtupperSeldon1975: so I have uncommented signalling=fxo_ls in my zapata.conf is this what you were referring to?
21:29.25Seldon1975mtupper not quite
21:29.39mtupperummm...
21:29.42Seldon1975mtupper: your fxs devices are the ones to the PSTN
21:30.09Seldon1975you need an entry in there to mirror what you have in zaptel.conf
21:30.18Seldon1975can you pastebin your zaptel.conf?
21:30.33Seldon1975http://pastebin.com/
21:30.55mtupperk, one sec...   but FXO to PSTN  FXS to phone, no???
21:31.19mrinvaderanyone here play with dtmf timings??
21:31.20stoffellparyl, you're readin it now, aren't ya ? :)
21:31.32Seldon1975it depends if you're talking about the hardware modules themselves, of the signalling
21:31.34harryvvno need to
21:31.34KattyHmmhesays: FOP is being evil :<
21:31.36parylstoffell: yeah :) interesting
21:31.38Seldon1975which are exactly opposite
21:31.42LostFrogmtupper: on a FXO device, you use FXS signalling, and vice versa.
21:31.43Seldon1975i know - its confusing
21:31.45KattyHmmhesays: you should stabbity it.
21:31.50stoffellyes paryl, I'm gonna try it out asap also :)
21:31.59mtupperahhhh!   thx
21:32.20*** join/#asterisk nagl (n=nagl@213.235.241.6)
21:32.25Seldon1975so the setup in zapata.conf is the OPPOSITE of what you have in zaptel.conf
21:32.32*** join/#asterisk heath__ (n=root@12-215-32-62.client.mchsi.com)
21:32.33Beirdonothing like breaking stuff to release some tension
21:32.34Seldon1975...which makes for fun for newbies
21:32.51mtupperLostFrog, Seldon1975: but I ummmm, dont have a zaptel.conf, just a zapata.conf ....   same thing?
21:32.53mrinvaderi want to stabbity asterisk for trying to talk to my phone system too fast :P LOL
21:32.54kuku5anyone have a iftime() example?
21:33.20parylstoffell: thing is, i don't know if my echo issues are what they're describing
21:33.25KattyBeirdo: :>
21:33.25Seldon1975mtupper: you do have it - just not in the same place
21:33.37Seldon1975mtupper: /etc/zaptel.conf
21:33.39stoffellparyl, indeed, first thing one should do is, find the 'source' of the echo
21:33.46stoffellwich is not always an easy task
21:33.47Seldon1975mtupper: you need to set up both
21:33.51mtupperk, damn, I feel like the rookie I am...
21:34.24stoffellparyl, this page has good resources: http://www.voip-info.org/wiki-Asterisk+echo+cancellation
21:34.30stoffellat the bottom some nice links
21:34.31parylthey mention muting the other side.  if the other side mutes, the echo goes totally away.
21:34.39parylyeah, i've been over that page over and over
21:34.44stoffellaha
21:34.55parylproblem is the echo is only present on about 30% of our outside calls
21:35.04stoffellhm, same problem here :(
21:35.10kuku5same here
21:35.13heath__true or false: there's no way to do command substitution in the dialplan without doing something hack assed like System(OUTPUT=`echo "whatever"` ; /usr/sbin/asterisk -rx dbput family key ${OUTPUT})
21:35.21paryland of course, those agents with headsets are CAUSING echo issues
21:35.29parylPRI
21:35.35kuku5PRI
21:35.55*** join/#asterisk genmud (n=genmud@ip68-98-82-206.ph.ph.cox.net)
21:36.37heath__i'd rather give my server Adderall
21:36.48paryli'd rather go to the bar
21:37.01paryla nice glass of Wild Turkey on the rocks... mmm
21:37.03parylrocks
21:37.06Beirdoeek
21:37.14asteriskmonkeywould be sweet to map audio to key presses so people can be like 1... one etc :D
21:37.21mtupperSeldon1975: not there either!!! i'm running Asterisk 1.2 on Debian Sarge with MySQL 5, and Astbill (and if you didn't notice I am a total rookie, only 6 weeks using linux)!!!
21:37.22mrinvadershotgun it with acapulco gold...
21:37.27stoffellparyl, you won't have any echo then :D
21:37.28BeirdoI should make some more bourbon sauce
21:37.36Beirdothat Jim Beam needs to get finished
21:37.40parylstoffell: lol
21:37.55mrinvadermmmmm bourbon sauce...
21:37.57parylyou guys see the Good Eats about bourbon?  i never knew that stuff
21:38.07parylAlton Brown is the bee's knees
21:38.08Beirdonot yet, when was it on?
21:38.19paryla couple weeks ago
21:38.21paryli tivo'd it
21:38.22Seldon1975mtupper: if you dont have etc/zaptel.conf then I have no idea how you are set up.... sorry!
21:38.31Beirdoah, it will take a while to be shown in Canada
21:38.37parylactually.. i guess it was egg nog, but a large part was about bourbon
21:38.42Beirdoah
21:38.45Beirdothe egg nog one?
21:38.48Beirdolet me look
21:39.05Beirdobourbon and egg nog do NOT go in the same sentence for me
21:39.23LostFrogRUM.. only RUM.
21:39.24parylyeah... the color/flavor of bourbon comes from the wood of the burned out barrels they age it in
21:39.38Beirdoyeah, I knew that
21:39.39Beirdo:)
21:39.43Beirdosame goes for rum
21:39.59Beirdothe color's from the barrels... that were previously used for bourbon
21:40.02mrinvaderrum is brown sugar  based tho isnt it?
21:40.02Beirdoin some cases
21:40.07paryli never knew they burned them!  that's so odd
21:40.13Beirdomolasses-based, yes
21:40.22mrinvaderya
21:40.51mtupperSeldon1975: thanks anyway
21:40.53mrinvaderso the darker rums have more cane solids in them.
21:41.00Beirdoumm, no
21:41.06Beirdothe darker rums are from aging
21:41.14mrinvaderahhh. k.
21:41.23parylBeirdo: 1/4 coconut run, 1/4 creme de banana, 1/4 peach schnapps, 1/4 mountina dew code red
21:41.26Seldon1975Im about to try tweaking my SIP Phones Audio Processing settings to reduce echo.  My JitterBuffer min/shrink/max are 40/500/160.  can someone suggest other values to try?
21:41.26parylrum*
21:41.32*** join/#asterisk jsolares (n=jsolares@200.6.233.132)
21:41.51BeirdoI like my rum in a glass. :)
21:41.54Beirdojummy
21:41.54paryli call it hawaiian punch
21:42.13jsolareshmmm how is ringing indication supposed to work with E1/T1's?
21:42.30mrinvaderrum in a vanilla/eggnog milkshake. w00t
21:42.43Beirdommmm, pint of Havana Club dark.
21:42.49jsolaresif i answer the incomming zap channel before dialing a sip phone i get ringing, if i dont answer i dont get ringing, am i doing something wrong? :S
21:42.52mrinvaderXD
21:43.17Beirdoand my (now) fiancee witnessed that on the webcam
21:43.18Beirdohehe
21:43.31Beirdoboozy goodness
21:43.45mrinvaderrofl
21:43.53paryljsolares: with either a ring or a polarity reversal
21:44.43mrinvaderor power fluctuations in the trans-warp oscillators
21:44.52jsolareshehe
21:45.04mrinvader%P
21:45.34jsolaresparyl: but this is an E1 not a fxo/fxs channel :S
21:46.19jsolaresmaybe i should make asterisk believe that it has answered the channel when in fact it hasnt so it sends the ringing audio... weee
21:46.51jsolaresor get the sip phones to send ringing audio? hmmm this is confusing
21:46.57mrinvadereverything i have is up and working.. except that when i dial from the server or from a sip phone it randomly will not make it out to the pstn ... im figgerin its gotta be asterisk sending digits too fast.
21:47.41paryljsolares: where are you?
21:47.50jsolaresin central america
21:48.00syleno shit
21:48.03sylewhere abouts?
21:48.18jsolaresin guatemala city
21:48.20mrinvader[sip-phone] to [asterisk server 1.0.9] to [vodavi triad 2 ksu] to pstn
21:48.21paryljsolares: ah... i don't know what the network does there.  i've never seen that problem
21:48.30*** join/#asterisk Gamercjm (n=gamercjm@71.254.175.163)
21:48.31jsolarescrap
21:48.36*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
21:49.10sylehehe kewl
21:49.13jsolareswell if i make the line digital so asterisk wont send that there's audio inband to the E1 it works if i call in from a big telco, seems they get to generate the ringing tone for me
21:49.20sylearen;t they holding a survivor series there?
21:49.32jsolaresthey were, it ended yesterday or so
21:50.00jsolaresbut that was way up north in the jungle, the migthy jungle where the lions sleep at night.... or something
21:50.08jsolares:)
21:50.11sylelol
21:50.12GamercjmI tried setting up DISA and when i test i get the dialtone and call other extensions, but wont allow outbound calls, get busy sound. Is DISA able to make outbound calls?
21:50.15*** join/#asterisk Dio_ (n=dima@82.207.0.44)
21:50.48BeirdoDISA goes into the context you tell it to
21:50.56sylenot sure what the problem is but have you tried the r option to dial to get the audio to the sip phones?
21:51.09Beirdoif the context you put it in can't do outbound calls, then DISA won't do outbound calls
21:51.27jsolaresthe problem is not getting the audio to the sip phones, but getting it to the incomming calls via the zap pri
21:51.30brettnemTHE 'r' OPTION!!!
21:51.42sylelol
21:51.42mrinvaderthe only time ill ever watch realitard tv is when they have thr unning man in real life and ppl actually die from eating rancid whale anus or fighting a rabid randy water buffalo barehanded.
21:51.49*** join/#asterisk shido (n=shido@d221-68-216.commercial.cgocable.net)
21:51.50*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
21:51.54Gamercjmoh I just set to defualt, ill check into context then
21:52.00sylethe evil 5979 right :)
21:52.07jsolarestried with and with no r, no luck
21:52.09Beirdomrinvader: or when they do "Survivor: Alaska"
21:52.11LostFrogDamn, steal my idea for a show, why don't you, mrinvader.
21:52.25mog_workdo you get calls on this sip phone
21:52.32LostFrogmrinvader: "Survivor: North Pole"
21:52.34mrinvaderLO friggidy L
21:52.43jsolareseverything works, it's just ringing indication that wont
21:52.56mog_workgoing to or from this sip device
21:53.07jsolaresfrom the sip to zap
21:53.11Beirdoa lot harder to survive in the north than in the frigging rain forest/paradise island
21:53.12jsolaresE1 digium card
21:53.14*** join/#asterisk toddf (n=toddf@net-66-210-104-104.theshop.net)
21:53.30mog_workwell if you have r it should give you ring nomatter what
21:53.35mog_workwhat if you do m
21:53.39mog_workdo you get musiconhold
21:53.58*** join/#asterisk Igbothom_III (n=HiltonT@203-206-114-129.dyn.iinet.net.au) [NETSPLIT VICTIM]
21:55.14*** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn)
21:56.06*** join/#asterisk jsolares (n=jsolares@200.6.224.146)
21:56.15jsolaresmeh this internet is killing me
21:56.47jsolaresi'd rather be in a frozen tundra with a reliable internet than this paradise jungle with crappy connection...
21:57.29BeirdoSurvivor: IRC?
21:58.17jsolaressince those would all be geeks, they'd die within minutes of the show XD
21:58.27LostFroglol.. all the lamers would be killed the first day.
21:59.21Beirdoall that would be left would be the ops
21:59.23LostFrogI know.. take 12 average windows users, install Linux to their machine..
21:59.28LostFrogSurvivor: Linux
21:59.39Beirdoto make it more fun...
21:59.47mrinvadernah the geeks would fashion a nuclear furnace out of radium watch dials and old at power supply..
21:59.49Beirdogive them a Gentoo CD, no web access.
22:00.00Beirdoand watch em squirm
22:00.02LostFrogowww.
22:00.06LostFrogYou are mean, Beirdo.
22:00.08Beirdo:)
22:00.19Beirdowith the docs, Gentoo's easy enough
22:00.23LostFrogBy web, I assume you mean internet..
22:00.26Beirdowithout...  have fun
22:00.36Beirdono, I mean specifically web
22:00.38mrinvadernah.. microsoft evangelists.. not just winders users..
22:00.43LostFrogOhh.. ok then.
22:00.49LostFrogI can do it without the web.
22:00.52Beirdothey can emerge -sync all they want
22:00.55NuggetMicrosoft Flight Simulator is a great product!  :)
22:01.05Beirdoyeah, but you ain't a Microsoft-only gumby
22:01.11mrinvaderyeah im glad they bought it arent you?
22:01.16LostFrogWithout internet, it would suck.
22:01.54Beirdohmmm
22:01.58mrinvadernow are we talking the cheapbytes complete gentoo set or one from jerkit city..
22:02.05Beirdowell, I've had enough of work for today
22:02.07Beirdostage 1
22:02.08Beirdoduh
22:02.18Beirdono shortcuts allowed
22:02.56Beirdoor if you are especially cruel.
22:03.08Beirdogive em a box of 50 floppies with OLD Slackware on it
22:03.11LostFrogLFS with no host OS.
22:03.11Beirdocirca 1992
22:03.12twisted[asteria]wake me up
22:03.15twisted[asteria]can't wake up
22:03.17twisted[asteria]save me
22:03.48LostFrogBeen there, Beirdo.. done that.
22:03.52mrinvaderYAH!!! watch em fall to the ground and seize/soil themselves..
22:03.58BeirdoLostFrog: me too :)
22:04.14LostFrogI want to buy a Yggdrasil CD.
22:04.17Beirdo"what's LaTeX"?
22:04.26nextimeanyone with a call termination with DialMex?
22:04.33LostFrogUmm.. that's the stuff that rubbers are made of.
22:04.42Beirdono, that's "latex" :)
22:04.55Beirdonot even pronounced the same
22:04.55LostFrogOhh.. then it's a document markup language.
22:05.01mrinvaderi was gonnasay Phylactixx Pro v.69
22:05.03LostFrogWell.. TeX is.
22:05.11Beirdoexactly, that's what the nooobs would be wondering though
22:05.50Beirdomake em compile gcc and then ghostscript
22:05.51Beirdoheh
22:05.57Beirdooh, and glibc
22:06.03LostFrogok.. gcc can be a pain.
22:06.08Beirdoby hand, no emerging, that's for wusses
22:06.13mrinvaderanyone have any ideas about tone timings? no.. XFree 4.1
22:06.15LostFrogIt seems not to compile 50% of the time.
22:06.18Beirdoand qt
22:06.20fugitivolinuxfromscratch.org
22:06.32LostFrogfugitivo: been there, got the t-shirt.
22:06.40fugitivoi use lfs for my servers
22:06.45fugitivono crap inside
22:06.49LostFrogI use minimal debian.
22:06.59Beirdosure there's crap, but you put it all there :)
22:07.10Beirdoso you know WHAT crap is there :)
22:07.10mrinvaderme II. minimal debian runs my firewall.
22:07.37mrinvaderdebian potato.
22:07.49mrinvaderon a p133/32mb machine
22:07.57BeirdoOpenBSD 3.6
22:07.58LostFrogI run sarge everywhere now.
22:08.00fugitivohmm, that's nice for openbsd
22:08.00Seldon1975Im about to try tweaking my SIP Phones Audio Processing settings to reduce echo.  My JitterBuffer min/shrink/max are 40/500/160.  can someone suggest other values to try?
22:08.01Beirdoanyways.
22:08.15Beirdohave fun, people
22:08.24mrinvadersarge is cool. did a fw in indiana with sarge.
22:08.32mrinvaderc ya beirdo
22:08.34fugitivoopenbsd and pf
22:08.45Nuggetpf+altq is tasty.
22:08.51fugitivoi love pf
22:08.56fugitivoi want a pf's tshirt
22:09.14mrinvaderque es pf?
22:09.21mrinvaderjust curious
22:09.31Nuggeta really great packet filter/firewall stack
22:09.39mrinvaderahhhh
22:09.39fugitivomuch better than iptables
22:09.54fugitivobetter syntax
22:10.06mrinvaderbsd is very reliable for stuff like that too. i use it for mailservers.
22:10.11LostFrogIf I could get over my SYSV racism, I would install BSD>
22:10.17LostFrog.
22:10.18fugitivoand you can do bridging
22:10.29fugitivotransparent
22:10.30LostFrogYou can do bridging in Linux.
22:10.36iDunnoFreeBSD isn't so bad :)
22:10.38LostFrogYes.. transparent.
22:10.52mog_workbsd and linux can all do same things now
22:10.52fugitivoLostFrog: but, pf is cool :)
22:10.54mog_workwoopity do
22:10.58iDunnoand linux bridging is fucking easy these days.
22:11.00mog_worknot the place for this fight
22:11.13fugitivowhat fight?
22:11.14mrinvadergotta love those holy wars.. i love it all.
22:11.18fugitivolinux and bsd are friends
22:11.19Nuggetopenbsd (and freebsd to a lesser degree) is still more capable of protecting itself from local users.
22:11.23mrinvaderyaa
22:11.27iDunnoand iptables is nicer (for me) than pf/ipf/ipfw/fuckingputtheevilbsdtoolofdoomhere
22:11.38astcryzAnyone had issues with asterisk realtime and md5password?
22:11.39fugitivopf is 37337
22:11.44Nuggetlinux is better at using crap like $9 tape drives that you bought at hamvention
22:11.46mog_worki dont know i think linux and bsd can be pretty pissy with each other
22:11.47mrinvaderlololiololol
22:11.53Nugget(which is great for users who have $9 worth of data)
22:12.28LostFrogI think BSD is good.. for those who like it.
22:12.44fugitivoi like openbsd soundtracks
22:12.45mog_worki think bsd/linux is not much more than prefrence these days
22:12.50fugitivolinux doesn't have soundtracks
22:12.50mog_workobsd songs rule
22:12.54mog_worklol
22:12.57mrinvaderi like native bsd process accounting for some things and linux's native baremetal speed for others
22:13.00mog_workill make one for you fugitivo
22:13.07Nuggetlinux bridging cheats inside, which makes it unworkable in some situations.  (like needing to do bidirectional filtering in bridged mode)
22:13.15fugitivomog_work: thank you
22:13.29LostFrogNugget: that works now, as well. As well as accounting.
22:13.31Nuggetto be fair, so does freebsd's
22:13.33tdonahueSIP/2.0 407 Proxy Authentication Required
22:13.38Nuggetwhen did they fix it?
22:13.39mog_worklike i said
22:13.42tdonahuewhat setting would cause asterisk to start sending a 407 response?
22:13.54mog_worklittle advantages here and there
22:13.59mrinvaderit ran out of 406'es?
22:14.03LostFrogSome time midway through 2.4.
22:14.04mog_worki use linux because zaptel hardware works better in linux
22:14.06Flautomog_work, how are you doing
22:14.10mog_workGrand
22:14.10*** join/#asterisk heison (n=heison@ns.somanetworks.com)
22:14.17fugitivowell, but that's like mysql vs pgsql, now mysql has a lot of new functions, but i moved to pgsl years ago, i'll not try again mysql
22:14.17LostFroglol@mrinvader.
22:14.36synthetiqi use macos to run asterisk because i want to be leet
22:14.43mog_workyeah
22:14.51mog_worki agree gutiov
22:14.58mog_workerr fugitivo
22:15.00synthetiqv.9 !!!!!!!!!!!11one
22:15.01fugitivolol
22:15.01LostFrogSend me OSX86, and I'll run * on it.
22:15.17fugitivogive me solaris for x86!
22:15.17iDunnoLostFrog: why on earth?!
22:15.26iDunnoOSX is just hideously broken ;)
22:15.28LostFrogiDunno: just to say I could.
22:15.33synthetiq* on pico bsd =x
22:15.36NuggetOSX is doubleplusgood.
22:15.48iDunnoand OpenSlowarsis has a semi broken licence
22:15.51fugitivoosx is a os for sissy girls
22:15.57synthetiqosx double playchild toy good
22:16.15iDunnoOSX is fine for 3 year olds that only want bouncy icons and no productivity...
22:16.16fugitivois the "barbie" theme out for osx?
22:16.19mog_workugh...
22:16.25mog_workos fight
22:16.26mog_workover
22:16.28mrinvaderosx can get ppl off winders.. an enemy of my enemy is my friend :LOL
22:16.32iDunnokinda like XP is fine for all telly tubbies ;)
22:16.47mrinvaderidunno : XD XD XD
22:17.01NuggetOSX is unix on the desktop that doesn't totally suck.
22:17.06Nuggetit's the *only* one, imho.
22:17.07mrinvaderi think all the *nices need to ally
22:17.20iDunnomrinvader: the figures actually show that OSX takes people away from the real operating systems to OSX, they were generally linux or FreeBSD users before hand ;)
22:17.22fugitivoNugget: kde doesn't suck
22:17.26tdonahueso no one knows why my account settings that worked 2 days ago no longer work?
22:17.30brookshirekde is great!
22:17.32iDunnoNivex: no it isn't.
22:17.41iDunnoerm - Nugget: no it isn't.
22:17.46fugitivokde is functional for me, and i do a lot of work with it
22:17.47NuggetI find KDE to be pretty miserable, but a lot of the blame lies with x.org/xfree86.
22:17.56NuggetX11R6 is just too crude to do anything modern with
22:17.56iDunnoNugget: OSX is a bastardisation of a Unixoid.
22:18.17iDunnoit's FreeBSD with all the useful bits ripped out and replaced with shit.
22:18.26fugitivoNugget: why? i have 3d, dual head
22:18.28Nuggetclipboard sucks, it can't even do font kerning properly.
22:18.31iDunnoalso, it has a broken python *by default*
22:18.34mog_workugh
22:18.36mog_worklets move on
22:18.47iDunnomog_work: agreed.
22:18.49fugitivoNugget: clipboard? i have 2 clipboards with my kde
22:19.07Nuggetso you can copy an image out of mozilla and paste it into openoffice?
22:19.16mog_workso there is this guy
22:19.20mog_workhe walks into a bar
22:19.27mog_workand orders a small drink and a cactus
22:19.37LostFroglol.. a cactus.
22:19.54mog_worknow think about that for a minute
22:20.02mrinvaderit _is_ all just preference outside of doze. i unite with all the unixes/alikes. we have the entirety of the non-billy world now.
22:20.36fugitivoNugget: no, but it copies the link to that image :)
22:20.47Nuggetheh
22:20.53Nuggethow almost useful.
22:21.02fugitivoanyways, i never had to do that
22:21.04mog_workback to the cactus
22:21.08Nuggetprobably because you can't.  :)
22:21.30LostFrogI've never done it in windoze.. Always saved the image and imported it.
22:21.34fugitivoi didn't know i can't :)
22:21.39fugitivome too
22:21.50brad_msswcan anyone in here verify that teliax is down right now ?
22:21.58fugitivoi don't like that drag'n drop stupid thing, hehe
22:22.07*** join/#asterisk Seyr (n=Seyr@rrcs-24-227-133-226.sw.biz.rr.com)
22:22.09file[laptop]brad_mssw: call and ask if they're down? lol
22:22.27brad_msswfile[laptop]: haha ... not like they answer their phones even when they're up
22:22.31fugitivobrad_mssw: simple, does it work?
22:22.40LostFrogThere was a time when you couldn't call the broadvoice support line from a broadvoice number. :)
22:22.47fugitivoif it doesn't then the service is down
22:22.55brad_msswfugitivo: want to make sure my ISP didn't just start blocking this outgoing port ...
22:23.09fugitivoare you in china?
22:23.12Seyrim using Realtime and set mailbox=100@mycontext , but when i call Voicemail(u100@mycontext) the SQL query that Realtime does, has context="" .. so it is not getting the context. anyone seen this before?
22:23.13brad_msswfugitivo: they've been known to do stupid things like that ...
22:23.37brad_msswfugitivo: no, not in china ... in US, but my ISP has primarily windows boxes, so they close down about any port they see a lot of traffic on
22:23.39LostFrogDo a nmap from another machine.
22:23.44brad_msswfugitivo: don't ask :/
22:23.58fugitivobrad_mssw: ok :)
22:24.07*** part/#asterisk mtupper (n=mtupper@pc-14-172-104-200.cm.vtr.net)
22:24.17brad_msswjust looking for anyone else that uses teliax that can confirm
22:24.26mrinvaderbrad_mssw: what are you trying to connect out on?
22:24.46brad_msswmrinvader: voip-co3.teliax.com .. iax2 registration is failing ... port 4569
22:25.35tainted-brad_mssw teliax was having issues
22:25.45mrinvaderyeah i doubt that it would be blocked.. thats an unprivileged port. they usually only kill ports like 25 and such.
22:26.10tainted-mrinvader what do u mean 'unprivileged'
22:26.14mrinvadermy isp unblocked me because i host a real mailserver
22:26.23LostFrogtainted-: less than 1024.
22:26.32mrinvaderroot/admin access .. thanks lost!
22:26.48synthetiqwhat you need is a good iax2 termination service brad_mssw =]
22:26.50mrinvaderi cnt type fast enuf *nix geek or not.
22:26.57tainted-yea but how does that mean 'unprivileged'
22:27.10mrinvaderor one not down right now at least LOL
22:27.13tainted-brad_mssw does teliax go down a lot?
22:27.36LostFrogtainted-: only root-owned process can open any port less than 1024 in *nixes.
22:27.42mrinvader1025+ do not req. admin privs on a server to open up.
22:27.50mrinvaderzacly
22:28.03brad_msswtainted-: this is the first time it's gone down on me
22:28.04tainted-i did not know what
22:28.05synthetiqyiu need izx2 termination on a loadbalanced set up =]
22:28.09iDunnoLostFrog: not true, there are kernel patches that give privileges to certain users for other ports.
22:28.20tainted-s/what/that
22:28.30mrinvaderwell right but under normal circumstances...
22:28.43tainted-brad_mssw i was going to port a # to them when they had this downtime
22:28.45LostFrogright.. and the kernel runs at ring 0.
22:28.49tainted-brad_mssw their portal went down too
22:28.51iDunnounder a usual distro, in usual circumstances, granted ;)
22:28.52LostFrogIt can do anything it wants.
22:29.08mrinvaderlike httpd runs as a user and so does ssh with privilege separation.
22:29.16brad_msswtainted-: hmm, their portal is up ...  this is for business unfortunately, so it hurts more ... ported our 800 and local numbers over
22:29.25iDunnobut the selinux patches, IIRC, allow you to actually provide a more open policy on ports
22:29.33LostFrogmrinvader: httpd starts as root and chroots as a normal user.
22:29.39iDunnoand thus you don't have to even start daemons as root ;)
22:29.58*** join/#asterisk JakBeatZ (n=JakBeatZ@gw2tor1nat161.beanfield.net)
22:30.07*** join/#asterisk nagl (n=nagl@213.235.241.6)
22:30.15mrinvaderand mahcrusawf jes allows anyone to do that stuff as a user... including frobbing packets :) LOL
22:30.33tainted-brad_mssw do u terminate through them or do u have others?
22:30.38mrinvaderlostfrogtrue that
22:31.52*** join/#asterisk clarity (n=chatzill@84.255.196.166)
22:31.55brad_msswtainted-: we have a single PTSN line coming in for 911 routing but everything else is through teliax ... we have a direct 100Mbps fiber connection to our ISP (only 2 doors down), so we thought reliability wouldn't be a problem, at least from our end
22:31.57JakBeatZIs it possible to use a variable as an extension in a context?  For example assume a [global] EXT => 100 can you have exten => $EXT,1,Dial(SIP/100) or is that an invalid use of an extension?
22:32.06JakBeatZer.. invalid use of a variable, rather
22:32.59LostFrogI don't believe you can.
22:33.34mrinvaderbrad_mssw: it might not be on your end.. | wow... MMMmmm.. Fiber... i need to get more fiber in my network diet.
22:33.34claritygood evening everyone... i have a bit of a strange question: I'm trying to force plain authenthication to a SIP phone, does anyone know if this is possible? i tried auth=plain but asterisk that I use doesn't send anything different with that option, and phone still does md5 auth
22:35.17tainted-brad_mssw yea origination u can't do anything about.. but termination u should probably look into some redundancy
22:35.26*** join/#asterisk brookshire (n=nubb@gateway.digium.com)
22:35.29SeyrAnyone know why Realtime is giving me: [SELECT * FROM vmail WHERE mailbox = '555' AND context = '' ] even though I am calling it with "555@mycontext" ???
22:35.36LostFrogHow can you have redundancy for incoming termination?
22:35.42Seyrits cutting out the context
22:35.56tainted-termination meaning outgoing calls
22:36.03_Sam--f'n teliax, down for the 2nd time today....been down a total of over an hour today...got 15 employees sitting around twiddlin thumbs
22:36.17mrinvaderidunno: why would one want to run a demon on a 1024- port as a user.. isnt that kinda dangerous?
22:36.36tainted-_Sam-- i think there are several people peeved with teliax
22:37.23iDunnomrinvader: think httpd
22:37.44lesouvageHow can I prevent hisax and isdn modules to load during startup. I can remove them with rmmod but I don't want them to load at all.
22:37.50iDunnomrinvader: that's a user. not one that I'd *ever* give a password or login access, but a user all the same.
22:38.28LostFrogWhy would you want to run a daemon as root, if you didn't want to, mrinvader?
22:38.36LostFrogdidn't need to, even.
22:39.37Seldon1975do the 'context's in zapata.conf correlate to the 'context's in extensions.conf?
22:39.45brad_mssw_Sam--: yeah, i'm pissed at teliax right now ...
22:39.52brad_mssw_Sam--: guess they're not a good business solution for us
22:40.08FaithfulHey guys, I have looked on google... I have a problem when I use a particular provider I get chan_iax2.c:5067 socket_read: meta trunk cmd 1 received, I only understand 0 (perhaps the remote side is sending trunk timestamps?)
22:40.47_Sam--brad_mssw:  i feel your pain and then some.
22:40.51Faithfuland I don't get incoming sound... outgoing is fine
22:40.55_Sam--who will you switch to?
22:42.10*** part/#asterisk Seyr (n=Seyr@rrcs-24-227-133-226.sw.biz.rr.com)
22:43.47*** join/#asterisk testmachine (n=assink@ip237-239-58-62.adsl.versatel.nl)
22:43.48clarityperhaps someone can tell me what does auth=plain actually do? just allow plain auth? doesn't force it?
22:43.57clarityany help is much appreciated
22:45.34mrinvaderwell all its been a lot of fun.. ill be back tamarra at the butt-crack o' 10am (EST). gnight all.
22:46.42brad_mssw_Sam--: looks like teliax just came back online
22:47.37_Sam--indeed
22:48.01*** join/#asterisk |Vulture| (n=V@216.84.158.117)
22:48.48Seldon1975hey guys, if i have contexts [incoming] and [outgoing] in extensions.conf, how do I configure the context in zapata.conf for my Zap interfaces to the outside world
22:48.48Seldon1975which are used for  both incoming and outgoing
22:48.48|Vulture|Anyone seen an instance where a polycom phone keeps trying to get an IP from a DHCP server? The server issues the DHCPDISCOVER then DHCPOFFER until it fails?
22:49.44_Sam--nice of teliax...call our toll free number it says it has been disconnected
22:50.23Sedoroxahaha
22:50.59Sedoroxdunno how far you are from canada..... but I know a good canadian place that does toll free... never had a problem with them
22:51.46_Sam--they fixed it now
22:52.30_Sam--if you can recommend a good IAX provider id love to know
22:52.40_Sam--or maybe i should go sip?
22:53.15Sedoroxhttp://www.thinktel.ca
22:53.18Sedoroxthey do both sip and iax..
22:53.30Sedoroxthink they prefer to do sip tho.. but not sure :p
22:54.05_Sam--what would be the downside to switching server communications from IAX to SIP?
22:54.37SedoroxI was told there are still issues with sip trunking and stuff on asterisk...
22:54.41Sedoroxbut I haven't done any tests myself...
22:57.08*** join/#asterisk ldnblk (n=Just@212.183.128.185)
22:57.47gnosyshey folks, Polycom phones.  paryl sent me to his blogspot site: http://paryl.blogspot.com/2005/10/making-polycom-501-work-we_113034274646163867.html but I have a question about this... what are folks doing for the ftp server?  vsftpd?  proftpd?  other?  setting it up with local users in /etc/passwd?  a "virtual user" database?  ldap repository?  other?
22:58.20*** join/#asterisk JakBeatZ (n=JakBeatZ@gw2tor1nat161.beanfield.net)
22:58.40Flautois iaxtel still working
22:59.05mog_worksometimes
22:59.37Flautomog_work, i can never get it registered
23:01.53parylgnosys: i had success with pure-ftpd
23:01.54Flautoitis always not registered
23:02.17gnosyshow did you set up the user list?  in /etc/passwd?  other?
23:02.18parylgnosys: http://paryl.blogspot.com/2005/10/setting-up-pure-ftpd.html :)
23:02.33gnosysoh!  thanks, paryl.
23:02.38paryli knew logging my experiences would come in handy :)
23:02.46JakBeatZIf I have a callback context configured in voicemail.conf and the callback context requires a dialing prefix, how would that work?  The number called back would be based on the callerid so the dialing prefix wouldn't be added to the callerid.  Conversely, if I had an outbound context configured in my voicemail.conf and I wanted to dial an outbound number that required a different dialing prefix, will that "just work"(tm) or would I need
23:02.47JakBeatZ<PROTECTED>
23:03.37Flautoor, the context you can use by default is fromvm
23:03.50gnosysit definitely did for me, paryl!  thanks.
23:03.52Flautoyou can create that context in your exensions.conf
23:04.05parylgnosys: awesome!  glad to know i could help
23:04.17Flautoto define what dialout channel you want to use
23:05.13Flautomog, thanks for your answers last night. i have some more questions
23:05.17Flautoif you dont' mind me asking
23:05.47Flautowhatis the difference in between enumlookup and dundi?
23:06.06Flautoi know that i can set up some channels for dundi
23:07.00mog_workdundi is kinda like distributed enum
23:08.23_Sam--if i have 12 incoming trunks via IAX, would there be any problems switching them to SIP?
23:08.30mog_workno
23:08.45_Sam--ty
23:08.53*** join/#asterisk kshumard_home (n=ksh@pcp01931374pcs.huntsv01.al.comcast.net)
23:08.56Flautomog, how exactly should i use dundi
23:09.05Flautothat is the part i am not sure
23:09.18Flautothere is examples in the extensions.conf
23:09.21sivanamog_work: http://pastebin.ca/33709
23:09.45sivanathank you :)
23:10.49mog_workyay!
23:10.59mog_worknow isnt that better than an ugly agi?
23:11.04sivanaI hate the module, but it works :)
23:11.23mog_worksorries
23:11.27mog_workim glad it works though
23:11.49mog_workdundi is not a real trivial thing to do honestly
23:11.50sivanaya.. the parameter thing is weak, maybe some day I'll make it a function instead
23:12.02mog_worktook me a bit of time to set up first time
23:13.23mog_workhehe im a genius, the bug is gone in my library now i just need to code it
23:15.33Flautomog, i know people can monitor calls to see where the call is from
23:15.37*** join/#asterisk xachen (i=justin@magnum.thisgeek.com)
23:15.41Flautowhat should i do to set it up
23:15.48Flautolike when a call is coming in
23:15.55Flautoyou see it on your cli
23:16.01Flautobut you don't see where the call is from
23:16.56mog_workyou can do a noop(${CALLERIDNUM})
23:17.14Flautowhere should i set it though
23:17.23mog_workzap show chanel
23:18.15Flautoi am not using zap channel much
23:18.19Flautoi use broadvoice
23:18.22*** part/#asterisk Jestre (n=ack@dargo.trilug.org)
23:18.27Flautolet me try
23:19.06mog_workoh iax2 show channels
23:19.11mog_workor your channel
23:20.57Flautobroadvoice is sip. so it should be sip show channels?
23:21.04*** join/#asterisk jsaunders (i=js@S01060060971c5817.va.shawcable.net)
23:21.06mog_workbroadvoice i thought was both
23:21.08mog_workbut yes
23:21.43jsaundersAnyone know of any articles discussing Sangoma Shark cards?
23:21.48shido.
23:22.01Flautosip only
23:22.14mog_workhavent seen any jsaunders
23:22.15mog_workjust picks
23:22.23mog_workvapor drivers i think still
23:22.35Flautodone
23:22.38jsaundersLikewise
23:22.52mog_worklooks spiffy though
23:23.00jsaundersIndeed
23:23.37Flauto147.135.12.128   3129620228  53732565368  02691/00000  unkn  No
23:23.41Flautothis is what i see
23:24.10mog_workokies
23:24.13mog_workwhat do you want to know
23:24.24drrayI want a 24xxp
23:24.59jsaundersWhat the heck is a 24xxp?  Heheh
23:25.08mog_worktdm 2400p
23:25.21jsaundersMake/model?
23:25.29mog_workhttp://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM2400P
23:25.31drraydigiums new FXS/FXO 23 port card
23:25.34drrayer, 24
23:25.34jsaundersTnx
23:26.08mog_worki want for my apt
23:26.18mog_workand just have a phone in every knook and cranny
23:26.18jsaundersQuite the beast.
23:26.22mog_workyes
23:26.25jsaunders:D
23:26.26mog_workfull height full length
23:26.40drrayit's expandable, which is what I like, I can grow to 24
23:26.48mog_workyup
23:26.51drrayas opposed to just buying 24 and lumping it
23:27.07mog_workor change modules out
23:27.10jsaundersThat backplane bus idea from Sangoma is pretty slick.
23:27.17mog_workits intresting
23:27.25mog_workbut i dont have a machine with 4 pci slots
23:27.28jsaundersHeheh
23:27.38mog_workand wouldnt want to use em all for a card
23:27.46mog_workrather have 96 ports of tdm ^_^
23:27.48jsaundersMakes sense.
23:28.00jsaundersBiased is ok.
23:28.03jsaundersDon't beat yerself up.
23:28.07mog_workno if you do a whois on my ip
23:28.20gnosysparyl, you still here?
23:28.46jsaundersmog_work:  Heheh.  Gotcha.  ;)
23:29.19mog_workjust pluggin away
23:29.33gnosysanybody here know polycom phones?
23:29.42mog_workyeah
23:30.24gnosysparyl pointed me to his blogspot on configuring them, but he mentions files that I don't see in the xml zip file: 000000000000-phone.cfg
23:30.37drraymac address?
23:31.06Flautomog, noop works
23:31.08mog_workthats the default one it will spray to all
23:31.11mog_workyay flauto
23:31.22Flautoi see callerid when someone is calling in
23:31.24Flautothank
23:31.25Flautos
23:31.28gnosyswell, he says to fill in the mac address into the zeros here, but there's no file by that name.  There is a phone1.cfg, is that it?
23:31.34mog_workno prob
23:31.44mog_workthats probably sameone
23:32.04gnosysso i should rename it following the mac address naming convention i guess?
23:32.20mog_worki thik all 000s flushes any phone
23:32.39shidono
23:32.39shidoman
23:32.42shidono 0'
23:32.48shidoth phone will use the 0's file
23:32.50shidofor default
23:32.55shidoif no mac address cfg file is found
23:32.59mog_workrigth
23:33.01shidou can read that in the -log file in the same dir
23:33.03shidoif u use ftp
23:33.06shidoinsetead of tftp
23:33.07mog_workotherwise you can use teh mac of the phone
23:33.13shidoI suggest using ftp because when u make a change to a file
23:33.14mog_workand use ftp
23:33.15*** join/#asterisk CyberPony (n=CyberPon@cpe-069-132-017-022.carolina.res.rr.com)
23:33.16shidothe phone reboots
23:33.18mog_workas tftp is unreliable
23:33.21shidoand updates
23:33.39Seldon1975if I have issued 'restart when convenient' in # console; can I cancel it?
23:33.50mog_workhmm i dont know
23:33.51gnosysthanks folks... i'll try it.
23:33.54mog_workwhy would it matter?
23:34.00shidoits not convenient for asterisk to cancel a command like that
23:34.11Seldon1975i dont want it to restart
23:34.18Seldon1975when the lines all stop
23:34.18kshumard_homemaybe `abort halt`
23:34.20mog_workhmm maybe no or done
23:34.24shidobrb
23:34.41jsaundersGolf anyone?  http://img.photobucket.com/albums/v493/odog/golfcarrier.jpg
23:34.50kshumard_home`help abort halt` makes it seem like that's what you want, Seldon1975
23:35.21drrayrestart when convenient, stops incoming calls.. waiting for zap to clear.  so it's not a good option for us
23:35.25mog_workkshumard_home:  has mad asterisk cli skills
23:35.42kshumard_homemog_work, giggity giggity
23:35.45Seldon1975yes
23:35.47drrayoh yeah
23:35.51Seldon1975i think abort halt might work
23:36.15ldnblkjsaunders do the  Sangoma cards do the polling on the card as apposed to doing it by software ?
23:36.28mog_workldnblk: ?
23:36.55*** join/#asterisk apardo (n=apardo@29.Red-81-39-85.dynamicIP.rima-tde.net)
23:37.29ldnblkwell what do the  Sangoma do ?
23:37.37mog_work???
23:37.42jsaundersHeheh
23:37.43Corydon-wOoops... help <tab> no longer works
23:37.50twisted[asteria]lol
23:38.14jsaundersHonestly, couldn't tell ya ldnblk.  I've only seen pictures, no details thus far.
23:38.46Corydon-wHow exactly would you do 'polling on the card'?
23:39.10twisted[asteria]by smoking a large blunt.
23:39.24Corydon-wYou're reading from the card...
23:39.45Corydon-wmog_work: I don't think he does, either.
23:39.51parylgnosys: i'm back, sorry
23:39.56paryldid you figure it out?
23:40.06gnosysnp... i may have solved it...
23:40.12ldnblkjsaunders: I was at a linux expo, and the card that u mentioned was billed as an equivelent to the digum cards
23:40.15parylah, great
23:41.20mog_workyeah its equiv to tdm400p
23:41.26mog_workwhen they get drivers done
23:41.34jsaundersWell, Digium does have more presence.
23:41.37Corydon-wSo you can buy a card from Sangoma... or you could buy an equivalent card from Digium and finance further Asterisk development....
23:41.46Corydon-wHmmmm, what to choose, what to choose...
23:41.50mog_workkeep mog off the streets
23:41.52mog_workand in college
23:41.53jsaundersHeheh
23:41.58mog_worksounds like a good plan to me ^_^
23:42.19jsaundersChi eyes there mog?  Heh.
23:42.38mog_workim big fan of that smiley face as apposed to :)
23:43.13*** join/#asterisk bjohnson (n=bjohnson@i216-58-58-83.cybersurf.com)
23:44.45Corydon-w"Keep Other Matt in college.  Buy Digium."
23:44.59jsaundersmog: Did you get to do the photoshoot for that chick on digium.com?  :D
23:45.23jsaundersNot bad, not bad indeed.
23:45.24mog_worklol
23:45.25mog_worki wish
23:45.29mog_workbut i got a woman
23:45.48Corydon-wmog_work: damn.  There goes my chances.
23:46.03mog_worklol
23:46.05*** join/#asterisk riddlebox (n=james@24-217-15-91.dhcp.stls.mo.charter.com)
23:46.19mog_workkeep the 6 matts off the streets is a good reason to support digium ^_^
23:46.34Corydon-wThere are now 6 Matts?
23:46.37mog_workyes
23:46.48mog_work4 in support
23:46.51mog_work1 in web dev
23:46.52Corydon-wCrap.  4 of you are "Other Matt" ?
23:46.54mog_work1 in shipping
23:47.03xachencreepy
23:47.10jsaundersWhat about the 1 in front of the door?  Heheh.
23:47.17xachenIf I change my name to Matt can I wor there? :D
23:47.20Corydon-wI thought Matt F was Asterisk development
23:47.21jsaundersOh, that was good.
23:47.25DaPrivateergah
23:47.29mog_workyeah its a gen rule
23:47.30DaPrivateermy nick alert wont stop going off!
23:47.31mog_workwell he is
23:47.32DaPrivateerlol
23:47.37mog_worki guess really
23:47.40mog_work3 matts are in dev
23:47.42*** part/#asterisk DaPrivateer (i=Privatee@CRIMSON.OFF-HOURS.COM)
23:47.44Corydon-wand Matt B is the web dev
23:47.44mog_work1 in support
23:47.46mog_work1 in shipping
23:47.49mog_work1 in web dev
23:48.03jsaunders1 on ground in front of door
23:48.27mog_workheh
23:48.34mog_worknah all matts are always on the move
23:48.39mog_workwe are raid 0
23:48.42mog_workno redundancy
23:48.55Corydon-wDo you have more than one Matt with the same initial last name?
23:49.08mog_worknope
23:49.22mog_workwe have b,r,n,o,f,s
23:49.26mog_workfnorbs
23:49.56drrayYate is interesting
23:50.05jsaundersShall I run and hide?  Heheh.
23:50.11nextimeanyone using chan_h323 on SVN-trunk-r7230 ?
23:50.14mog_workmeh we dont care
23:50.16drrayYate has windows drivers
23:50.19mog_workbe happy
23:50.22parylyate? huh?
23:50.23jsaundersExactly.
23:50.24mog_workyay!
23:50.40jsaundersToo bad he went and killed himself though.  Sheesh.
23:51.00mog_work?
23:51.05*** join/#asterisk Druken (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com)
23:51.07jsaundersWhat was his name?  Bobby McFaren?
23:51.09jsaundersSomethin' like that.
23:51.11Corydon-wand there's what, 2 people developing yate?
23:51.29jsaundersIf that.  Heh.
23:51.38mog_workhey incourage our oss breathren
23:51.40*** join/#asterisk GaVa (n=GaVa@201.14.147.30)
23:51.41Corydon-wNo, it's Diana and her man
23:51.44mog_workdont knock em
23:51.46drrayIf that is how they want to spend their time..
23:51.47jsaundersHe's doin' a pretty good job I must say.
23:51.52mog_worki mean people should do what makes em happy
23:51.56drrayamen mog
23:51.56jsaundersI like Paul, nice guy.
23:52.13jsaundersDiana means well, but high strung at times though.
23:52.18Drukendrray: haven't gotten that board yet... should i have?
23:52.29drraythey said 5 days
23:52.34drraybusiness days
23:52.46Corydon-wjsaunders:  She comes off as manic
23:52.51jsaundersIndeed
23:53.02Drukendrray: when was it sent again?
23:53.14drrayI want to say last wednesday maybe?
23:53.22drrayI don't recall for sure
23:53.26drraymaybe longer
23:53.41Drukenaight.. no biggie
23:53.42litageis okay to run a gatekeeper (eg: gnugk) and asterisk on the same machine?
23:53.53Drukeni'm sure it'll get here when it does :)
23:54.04drraywell, I declared it at zero value
23:54.11drraymaybe they opened it?
23:54.33Drukenthat's possible... OH NO IT'S A BOMB!!!!!!
23:54.43drraywell, drugs
23:55.05Drukendid you pack it in cofee grounds? :)
23:55.14drrayno, the stinky ass newspaper
23:55.48Drukenahh, so i'll know the news from where ever your from
23:55.59drraywell seattle
23:56.09drrayand you'll know what date I boxed it and mailed it
23:56.11drray:)
23:56.17Drukenpfft, nothing special there... :)
23:56.32Drukengood point..
23:56.38Drukenunless it was an old paper
23:56.58Drukensomething tells me ya didn't buy the paper to use it for packing
23:57.02drraydid I email you telling it was sent?
23:57.15drraywe get the paper everyday here
23:57.16Drukenno, i think you msg'd me
23:57.18drrayit was snowing
23:57.22drraythe day I mailed it
23:57.23drrayI think
23:57.48jarrodhey what is the format for a subscribecontext as specified in sip.conf?
23:57.55jarrodis it an extensions context or a sip context?
23:58.36drrayYate, is interesting to me because it uses windows versions of the tor2 drivers
23:58.45drrayand it's just interesting
23:58.53mog_work?

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