irclog2html for #asterisk on 20060619

00:00.07drraynow that cisco is wising up
00:00.07Strom_Ccan you run SIP on the 70s now?
00:00.13CerealVorei don't think so yet
00:00.13drray8.2 firmware
00:00.14drrayyes
00:00.21CerealVorewow, they finally released it
00:00.24Strom_Coooooooooh
00:00.31Iamthemantks
00:00.51CerealVoreis there anything particularly special about the 7970s? apart from having a funky background on your phone?
00:00.56drraycolor
00:01.03drrayand 5 menu buttons
00:01.04CerealVoreyeah
00:01.08CerealVorebut they don't do video
00:01.09drrayso no
00:01.12CerealVoreso why would you bother?
00:01.19drraybut I still want one
00:01.50*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
00:01.50*** mode/#asterisk [+o russellb] by ChanServ
00:01.52CerealVoreheh
00:01.57CerealVorei can't deny, they are funky
00:02.52Iamthemanand what's the best quality/price/ease of use model ?
00:03.05Strom_C7940 / 7960
00:03.08Strom_C:)
00:03.18drraythe ciscos are a pain getting sip firmwares
00:03.21drjones1yea
00:03.25drjones1but asfterr you do
00:03.26drjones1they are great
00:03.28drrayyeah
00:03.31drjones1i was recommened some polycomms outta here
00:03.36drjones1i bout 15 of those bitches
00:03.38drjones1just the other day
00:03.40drjones1i picked one up
00:03.45drjones1fried transitors
00:03.45Strom_Cthe polycoms are a huge pain in the ass to set up
00:03.48drjones1i didn't even touch anything
00:03.50drjones1fuck a pollycom
00:03.56drraycisco is a good solid phone
00:03.59drjones1indeed
00:04.02drjones1thanks guys
00:04.02drjones1later
00:04.12Iamthemanand 3Com ?
00:04.12*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
00:04.12drraybut any voip phone will be persnickity
00:05.32*** join/#asterisk coppice (n=chatzill@18.162.17.210.dyn.pacific.net.hk)
00:05.36dlynes_homeso i guess no such thing as a voip phone with firmware that works reliably?
00:05.39Iamthemanwhat's the difference btwn G and G-GE, only Gigabit ?
00:06.08Strom_Cdlynes_home, if you want reliability in a phone, get a Western Electric :)
00:06.19dlynes_homeIamtheman: nbx works with sip?
00:06.21drrayand an adit
00:06.23drray:)
00:06.42dlynes_homeStrom_C: what's western electric? an analog phone?
00:06.56Strom_Cdlynes_home, yow, and you call yourself a phone person :)
00:07.05Strom_Cwestern electric was the manufacturing arm of the Bell System
00:07.08dlynes_homeStrom_C: i call myself a computer guy
00:07.18dlynes_homeI know dick about American manufacturers
00:07.26Strom_Cmade every piece of AT&T's hardware from 188x through 1984
00:07.37dlynes_homeah.
00:07.42drrayI miss my old southwestern bell phone
00:07.50drraythat was a solid device
00:07.52dlynes_homenorthern telecom is better, anyways
00:07.58drrayexhibit a your honor
00:08.04dlynes_homeThat's who made almost every single phone I've ever used
00:08.07Strom_Cbecame "AT&T Technologies" after divestiture, and today they're called "Lucent"
00:08.14Strom_CNortel is awesome
00:08.17h3x0rand then Avaya
00:08.17dlynes_homewell now
00:08.18Qwellugh
00:08.20dlynes_homeI've heard of Lucent
00:08.28dlynes_homeand AT&T Bell Labs
00:08.32dlynes_homebut never western electric
00:08.35Strom_Ch3x0r, Avaya is the business systems division of western electric
00:08.37h3x0rand nortel sold the phone stuff to Aastra
00:08.40Strom_Cor was, anyway
00:08.47Strom_Cdlynes_home, how old are you?
00:08.53dlynes_homeStrom_C: 37
00:09.01Qwellold men...
00:09.06dlynes_homeold?
00:09.07dlynes_homewtf?
00:09.10Strom_Cheh, and never heard of western electric
00:09.11*** join/#asterisk SuperLag (n=aaron@gentoo/developer/SuperLag)
00:09.14Strom_Cthats funny
00:09.16dlynes_homeStrom_C: like i said
00:09.20dlynes_homeStrom_C: i'm not american
00:09.44Strom_Cok, but you heard of something called "Northern Electric" at one time, right?
00:09.54dlynes_homenope
00:10.00dlynes_homeNorthern Lights, yeah
00:10.06Strom_CNorthern Electric -> Northern Telecom -> Nortel
00:10.36dlynes_homehow many decades ago was it called Northern Electric?
00:10.46Strom_Cname change happened in 1975
00:11.00dlynes_homeyeah...I was 6 at that time
00:11.11Strom_Cbut there was tons of Northern Electric stuff in service for years after that
00:11.24coppiceBell Northern Research + Northern Telecom -> Nortel
00:11.32QwellStrom_C: how much of that is now currently in your house?
00:11.32*** join/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net)
00:11.33Qwell;)
00:11.34dlynes_homeStrom_C: in the 70's, we didn't have a phone line, electricity, plumbing, or any of that stuff
00:11.47Qwellsilly canadians
00:11.49Strom_CNorthern Electric was once a division of Western Electric
00:11.50Strom_CQwell, all of it
00:12.07dlynes_homeStrom_C: coppice sounds more correct
00:12.10Iamthemangood place to buy cisco phones in Canada ?
00:12.13dlynes_homeStrom_C: nortel was never an american company
00:12.19QwellStrom_C: how large IS your collection?
00:12.40coppicecompanies are so jumbled up if they are old. daimler chrysler can't call a car a daimler, as ford have the tracemark for that
00:12.42h3x0rits hard to find cisco routers in canada let alone phones
00:12.52Strom_Cdlynes_home, Nortel was once Northern Electric, a subsidiary of Western Electric, and Bell Canada was once a subsidiary of AT&T.
00:12.56dlynes_homeh3x0r: you must not look very hard
00:13.15dlynes_homeStrom_C: bell canada is a subsidiary of BCE
00:13.27h3x0ri spent much time trolling bell canada, telegloba, etc
00:13.30h3x0rteleglobe
00:13.32h3x0rback in the daty
00:13.35h3x0rand uunet canada
00:13.40Strom_Cdlynes_home, you're not thinking far enough back in time
00:13.45h3x0reverything was nortel or 3com
00:13.53dlynes_homeStrom_C: BCE also owns teleglobe, HK Telecom, UK Telecom, Transcanada Pipelines, ...
00:14.09dlynes_homeh3x0r: uunet canada
00:14.14dlynes_homeh3x0r: those guys were a major ripoff
00:14.19h3x0ryep
00:14.26coppiceHK telecom is owned by the Li family
00:14.33coppiceand its called PCCW now
00:14.38QwellThe Li Family is owned by BCE
00:14.40Qwell:D
00:14.54dlynes_homecoppice: perhaps, but last time i checked my annual report, BCE owned 20 or 30% of HK T&T
00:14.54dongswhat
00:15.06Strom_CBCE is owned by my left ring finger
00:15.23Iamthemanwhat's a good price for a used 7940 and a 7960  &
00:15.28h3x0rbce can suck my left nut
00:15.36QwellIamtheman: $50
00:15.39Strom_CIamtheman, I got one for free once
00:15.39h3x0rtelus can suck my right nut
00:15.54dlynes_homeh3x0r: they did...that's why they moved to First Canadian Place from Montreal :p
00:15.54Strom_Ch3x0r, so where does that leave MTS?
00:16.03h3x0rlicking my asshole?
00:16.10russellbh3x0r: enough
00:16.11coppiceIts sad when your main telco ends up with a name like Pacific Century Cyber Works. quite embatassing, really :-)
00:16.36dlynes_homeCyber works?  wtf?
00:17.34dlynes_homeh3x0r: you don't like telus?  wtf is wrong with you man?
00:17.45h3x0ri didnt say that
00:17.48Strom_Cdlynes_home, maybe he just isnt fond of GTD-5 EAX :)
00:17.52h3x0ri just said they could balance out the nut sucking
00:17.55coppiceThe run the worlds only successful IPTV business. successful means they get lots of subs by more or less giving the ervice free with broadband :-)
00:17.59dlynes_homeStrom_C: ok, whatever that is
00:18.18Strom_Cdlynes_home, class 5 digital switching office that GTE used a lot
00:18.24Strom_Cdlynes_home, and since BC TEL was a subsidiary of GTE...
00:18.24dlynes_homeStrom_C: ah
00:18.28h3x0rim thinking about moving to vancouver
00:18.31dlynes_homeStrom_C: yeah...that's old news
00:18.36dlynes_homeh3x0r: dood
00:18.40dlynes_homeh3x0r: stay in Toronto
00:18.44dlynes_homeh3x0r: we don't want you here :p
00:18.49h3x0rthen i can pee on telus
00:18.51h3x0ri live in vegas
00:18.55dlynes_homeh3x0r: oh
00:18.56h3x0ri used to work in montreal for a while
00:18.56dlynes_homeheh
00:19.05Strom_Cjeez, how can you make telephone jokes in #asterisk and have people not get them?!:)
00:19.22Strom_Ch3x0r, so how's Embarq these days?
00:19.22dlynes_homeStrom_C: i don't know if you noticed or not
00:19.23coppicetelus is a pretty dumb name too
00:19.32dlynes_homeStrom_C: but a lot of us are computer geeks, not telco geeks :)
00:19.32coppiceTelus - we enjoy a good laugh!
00:19.37h3x0rit reminds me of telstra
00:19.49Strom_Cdlynes_home, bah!
00:19.52*** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn)
00:20.30dlynes_homeStrom_C: the only thing i like about at&t
00:20.43dlynes_homeStrom_C: is when they freely released korn shell :p
00:22.05JoseBravoIm getting this error triying to call from ext to ext. Jun 19 04:22:59 WARNING[23803]: app_dial.c:803 dial_exec_full: Dial requires an argument (technology/number)
00:23.46SkramXanyone going to HOPE6? Or even heard of it :P
00:24.12*** join/#asterisk Strom_C (n=strom@12.150.239.132)
00:24.20orlockEurgh telstra
00:24.20Strom_Cdid I mention this hotel's wifi sucks balls?
00:24.27SkramXStrom_C: :(
00:24.35Iamthemanhow to change the default ring sound with eyebeam or xten ?
00:24.36Strom_Cso we were on AT&T and ksh when I dropped off
00:24.43SkramXI have been on the road the last  week (just got home) and I totally know what you mean
00:24.45Strom_Cwhat'd I miss
00:24.49Strom_Chey SkramX
00:24.50SkramXI said:
00:24.51SkramXanyone going to HOPE6? Or even heard of it :P
00:25.18Strom_CSkramX, dunno.  HOPE is a long way away from los angeles
00:25.21SkramXI need to make my final decision if I want to present about Asterisk in a VPS environment or not
00:25.32hadsJoseBravo: So what does your dial command look like?
00:25.33SkramXStrom_C: I hear ya.. Im coming from Central Texas
00:28.13Iamthemanhow to change the default ring sound with eyebeam or xten ?
00:34.38Strom_CSkramX, go to defcon instead, so you can see my talk :)
00:34.48QwellStrom_C: get me in free :p
00:34.58SplasPoodHeh, I think I've completely rewritten asterisk's agent system /w realtime, app_addon_mysql, some macros and some dialplan
00:35.14QwellSplasPood: good..  publish it, so we can remove chan_agent :P
00:35.21SplasPoodQwell: heh
00:35.47SplasPoodit'd have to be hacked up a bit to be ripped away from the rest
00:36.00SkramXStrom_C: about what?
00:36.01coppiceis this a secret agent system?
00:36.09SplasPoodcoppice: I suppose
00:36.21SkramXI think greyarea is presenting..
00:36.22Strom_CSkramX, fedex kinko's hack, plus another talk which has yet to be approved
00:36.39SkramXhmm sounds good
00:36.46SkramXim only going to hope since i have family up there
00:36.48QwellWhat kind of hack?
00:36.51SplasPoodif only asterisk's realtime extensions wasn't so lame
00:36.54Strom_CQwell, free xeroxes!
00:36.58Qwellahh
00:36.58SkramXhehe
00:37.10Qwellreverse engineer the card system?
00:37.13Strom_Cyes
00:37.16SkramXim deciding if i should do a talk about asterisk and vpses, etc
00:37.18Qwellnice
00:37.31coppicexeroxes are almost free. the toner on the other hand.....
00:37.40Strom_Casterisk + vps == ew
00:37.47SkramXStrom_C: why do you say that?
00:37.50Qwellanything + vps == eww
00:37.54Strom_Cyes, exactly
00:37.55SkramXA number of my clients like it
00:38.08SkramXvps == Virtual Private Server == wee
00:38.11SkramXwee == good
00:38.13SplasPoodVPS for dev == great
00:38.19QwellSplasPood: That's about it
00:38.25Qwellbut, even then...
00:38.36SkramXI actually agree, but it can also be used in the production environment
00:38.40SkramXbut, its debatable
00:38.46SplasPoodI actually love running VMs on my macbook in parallels
00:38.48SplasPoodgot asterisk in one
00:38.52Strom_Cif you're a cheapskate, sure
00:38.55SplasPoodopenser in the other...
00:39.04SkramXStrom_C: ah, I guess so
00:39.20SkramXMeh, dont tell our customers!!
00:39.25SplasPoodI wish I could figure out how to put them all on their own subnet, and then route that through the macbook as a gw
00:40.23SplasPood~paste
00:40.25jbothmm... paste is see http://paste.husk.org
00:41.18drraydo you use vmware for vps?
00:41.26SkramXdrray: nopers.
00:41.31drraywhat do you use?
00:41.44SkramXlinux-vserver.org
00:41.57SkramXstill in some development but we have it stabilized
00:42.00drraythank you
00:42.03SkramXsure
00:45.28*** join/#asterisk SwK (n=Silik0nJ@12-219-147-107.client.mchsi.com)
00:48.05SplasPoodwhen I do a NoCDR() I get these console warnings:
00:48.06SplasPoodJun 18 20:46:10 WARNING[29399]: cdr.c:443 ast_cdr_free: CDR on channel 'SIP/C1000-000f34fa1d16-1-6539' not posted
00:48.06SplasPoodJun 18 20:46:10 WARNING[29399]: cdr.c:445 ast_cdr_free: CDR on channel 'SIP/C1000-000f34fa1d16-1-6539' lacks end
00:48.11SplasPoodis that something to be concerned about?
00:54.10*** join/#asterisk pigpen (n=mark@fw.seamans.cc)
00:54.18*** part/#asterisk pigpen (n=mark@fw.seamans.cc)
00:58.05*** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net)
00:59.14*** join/#asterisk Strom_C (n=strom@12.150.239.132)
01:01.05*** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)
01:01.35paolobHi guys! Where do I specify the format (gsm, ulaw, etc.) of the sound files to play? thank you!
01:03.40*** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net)
01:05.00russellbpaolob: you don't specify it
01:05.15russellbpaolob: asterisk will try to find the best format based on what format the calling channel is using
01:05.16*** join/#asterisk Strom_C (n=strom@m215e36d0.tmodns.net)
01:05.21Strom_Cok
01:05.24Strom_Cto hell with wifi
01:05.27Strom_Cedge for the win
01:05.52paolobrussellb, I get a "Unable to open MenuEspanol (format ulaw): No such file or directory" error: what does it mean?
01:06.08paolobdo I lack the ulaw format file MenuEspanol?
01:06.55russellbpaolob: it means it couldn't find that file at all
01:07.30russellbfile: go party
01:08.35paolobrussellb, I don't know what I changed in a config file, that file is in /usr/share/asterisk/sounds/es.  Before the change asterisk found it, now it doesn't. However in sip.conf I have specified language=es. Shall I specify somewhere else?
01:09.03russellbtry putting it in /var/lib/asterisk/sounds/es
01:09.49SplasPoodhrm... I wonder how I can somehow pass the name of the queue making the call to the Local/ channel I have setup as a member
01:10.34russellbSplasPood: you can probably do it using variable inheritance
01:10.55russellbbefore a caller enters the Queue, do ... Set(__QUEUENAME=whatever)
01:11.00russellbthat is two underscords
01:11.08russellbs/underscords/underscores/
01:11.14SplasPoodrussellb: hrm, lemme try
01:11.36russellbthen, in the extension for the local channel, NoOp(Coming from queue: ${QUEUENAME})
01:11.54*** join/#asterisk CoffeeIV (i=rgr@cpe-70-112-100-20.austin.res.rr.com)
01:12.53SplasPoodrussellb: bingo, thanks
01:13.03russellbs/chers/cheers/
01:13.05russellbi can't type
01:13.12russellbyou're welcome :)
01:13.21Strom_Ci believe you have my vocoder
01:13.22Qwell"Do you belieeeevve .."
01:13.38russellbeep!
01:14.12*** join/#asterisk neilbags-work (n=neilbags@149.171.94.134)
01:14.28CoffeeIVI am about to set up an * server to try some stuff out, if it works it will go into production -- any reason to stick to Fedora Core 4 instead of using FC 5 ?
01:14.55QwellCoffeeIV: none
01:15.14russellbunless you have FC4 cds and not FC5
01:15.28CoffeeIVcool -- that's what I thought -- I have to download the CDs either way
01:15.40Qwellin that case...get debian :P
01:15.46russellbword :-p
01:15.51Qwell(or gentoo..)
01:16.01paolobrussellb, no the issue is that it looks for it in /usr/share/asterisk/sounds: if I put the sound file there it find it.
01:16.27CoffeeIVI prefer Debian or Slackware, but the other guy on this project is used to redhat/fedora systems
01:16.42russellbpaolob: try adding Set(LANGUAGE()=es) before playing the file
01:17.19*** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net)
01:20.23SplasPoodNice, I now have Cepstral based queue agent announcements "Call in from <queue name>, press 1 to accept"
01:20.32russellbSplasPood: nice
01:20.41SplasPoodrussellb: ALL THANKS TO YOU! :P
01:20.48russellbwoohoo
01:21.01russellbat least i accomplished something
01:21.01*** join/#asterisk Iamtheman222 (n=fddfffsa@modemcable237.246-83-70.mc.videotron.ca)
01:21.09Iamtheman222I have a problem with the callback
01:21.11russellbmy Makefile changes aren't quite as successful here ...
01:21.20Qwellrussellb: sucks
01:21.26russellbQwell: i'm pretty close, though ..
01:21.28Qwellcool
01:21.40Qwellcommit it!  "close" is close enough ;)
01:21.53russellbbut the default paths lost their prefixes ... so running asterisk -r complains that there is no file /run/asterisk.pid
01:21.58Qwellahh
01:21.59SplasPoodIamtheman222: I think people are gonna need a lil more info..
01:22.05Qwellthat could be an issue
01:22.05Iamtheman222when I try to use the callback, it seems to look for callback.config file
01:22.05russellbQwell: that's kinda bad :)
01:22.24russellbIamtheman222: there is no such thing as callback.config in asterisk
01:22.27SplasPoodIamtheman222: what is 'the callback' ?
01:22.56Iamtheman222callback function that I have added in the extensions.conf to call me back
01:23.05russellband perhaps you should say hi first :)  IRC is a social place
01:23.22Iamtheman222Hi
01:23.27russellbyay
01:23.40russellbIamtheman222: pastebin your extensions.conf entry ... www.pastebin.ca
01:24.34Iamtheman222http://pastebin.ca/67014
01:25.06Iamtheman222the callback function has been found in freepbx...
01:25.19Qwell~freepbx
01:25.23jbotmethinks freepbx is NOT supported here!  People using it should join #freepbx (FreePBX is the new name of AMP)
01:25.24paolobrussellb, There is something strange with my problem with playing sound file: the sound file is looked for in the sounds directory on incoming calls, while it is looked for in sounds/es if I call from an extension. Why?
01:25.54Iamtheman222I know but what's the problem, it's not about freepbx
01:26.10russellbpaolob: because the language isn't getting set on the channel for incoming calls
01:26.12Qwell<russellb> paolob: try adding Set(LANGUAGE()=es) before playing the file
01:26.46paolobrussellb, isn't there a way to get it set for incoming calls?
01:26.53*** part/#asterisk Iamtheman222 (n=fddfffsa@modemcable237.246-83-70.mc.videotron.ca)
01:27.02*** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com)
01:27.02russellbgeez, impatient
01:27.15russellbpaolob: yeah, you just have to get it into the right entry of your config
01:27.16Qwellrussellb: he's been doing that all day
01:27.18SplasPoodpaolob: your sip peer in sip.conf, that the call comes in on
01:27.22SplasPoodpaolob: or iax..
01:27.37paolobSplasPood, iax...?
01:27.46SplasPoodhow do your inbound calls come in
01:27.51SplasPoodor zap for that matter..
01:27.58russellbor ... mgcp
01:28.04SplasPoodyea, but who uses that :P
01:28.05Qwellor ski..wait, nm
01:28.06*** join/#asterisk tlowe_ (n=tlowe@omfg.wtf.no)
01:28.11russellbQwell: ;)
01:28.19Qwellactually though..
01:28.35QwellI have briefly thought about making chan_skinny a client implementation too :)
01:28.40paolobSplasPood, russellb, but in sip.conf I have set language=es , the comment says "Default language setting for all users/peers". Why doesn't it work for incoming calls?
01:28.51Qwellrussellb: How incredibly pointless would that be?
01:28.51russellbQwell: people would use it, i'm sure
01:28.53SplasPoodpaolob: hrm, good point
01:29.07russellbQwell: what about for extra features for an existing call manager install?
01:29.20Qwellrussellb: yeah, perhaps
01:29.31russellbpaolob: perhaps it's a bug
01:29.35russellbpaolob: is this 1.2 ?
01:29.51russellband are you using allowguest=yes ?
01:29.52paolobrussellb, 1.2.7.1
01:30.16paolobrussellb, it's commented out
01:30.19russellbQwell: i think it would be cool.  lots of people have asked for the client implementation of mgcp
01:30.35Qwellit's fairly easy to do, I think
01:30.42Qwellwe already know all the messages
01:31.10SplasPoodpaolob: uncomment it?
01:31.31paolobSplasPood, what does it makes precisely?
01:31.46russellbno, that probably won't help
01:31.52russellbi was just trying to think places there could be a bug
01:32.55JoseBravoI have lost all my day triying to call from ext to ext with astbill. But I get this f....k error, Jun 19 05:32:41 WARNING[30085]: app_dial.c:803 dial_exec_full: Dial requires an argument (technology/number). Plese if anoye know why is that, I'll be very thanks.
01:33.07russellbpaolob: is this an installation from source?  i have a patch for you, i think ...
01:33.20paolobrussellb, no, it's the debian package
01:33.35paolobrussellb, will the patch enter the debian archive?
01:34.03russellbpaolob: eventually, yes, if it fixes your problem
01:34.11*** join/#asterisk Iamtheman222 (n=fddfffsa@modemcable237.246-83-70.mc.videotron.ca)
01:34.20russellbpaolob: if you want to download source, then i have a patch you can try ...
01:34.36paolobJoseBravo, You must put dial(SIP/resource,60,Tt), for example
01:34.47russellbpaolob: if you have svn installed .... svn co http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2
01:34.48QwellDO NOT USE tT!
01:34.50Qwellever
01:34.58Qwellwell...mostly
01:35.00russellbQwell: oh shush
01:35.10paolobrussellb, no, thanks, I'm going to put the set language in the incoming extension
01:35.12Qwellrussellb: make me :p
01:35.19russellbpaolob: alright
01:35.26Qwellwait, no, forget I said that
01:35.34*** mode/#asterisk [+b %Qwell!*@*] by russellb
01:35.39russellbpwned
01:35.46Strom_Cha
01:35.54*** mode/#asterisk [-b %Qwell!*@*] by russellb
01:35.55Qwell:D!
01:36.50JoseBravopaolob exctly?
01:36.56*** part/#asterisk tlowe_ (n=tlowe@omfg.wtf.no)
01:37.59paolobJoseBravo, I put an example, see http://www.asteriskguru.com/tutorials/dial.html
01:38.01russellbpaolob: well i'm pretty sure i found the bug, and i'm committing the fix ... just fyi
01:38.25paolobrussellb, ok, thank you, what version will it be fixed?
01:38.34russellbpaolob: 1.2.10
01:38.53paolobrussellb, ok, you're fantastic!!!!
01:38.58russellbthanks :)
01:39.16Qwellrussellb rocks :D
01:39.46paolobrussellb, (excuse me) when will 1.2.10 be released?
01:39.48[TK]D-FenderOh God.. another release within a week?
01:39.52Strom_Coh, speaking of which, 1.4 is in beta now, right?
01:40.05russellbpaolob: sooner if you test my fix :D
01:40.13[TK]D-FenderStrom_C :Yup, and due to be released with Windows Vista I hear ;)
01:40.14paolobrussellb, :-(
01:40.17russellbStrom_C: not yet
01:40.34paolobguys, and what about stun support in asterisk? is there anyone working on it?
01:40.37*** join/#asterisk DMark (n=kk7cu@pool-70-105-217-25.scr.east.verizon.net)
01:40.42russellbpaolob: it's in trunk already
01:41.02russellb(to be 1.4)
01:41.15paolobrussellb, "trunk" (I'm not english speaking) what is it?
01:41.22JoseBravopaolob it was a account addedwith AstBill
01:41.22[TK]D-Fenderrussellb : A lot of remarkable stuff going on for 1.4...
01:41.25Strom_Cwill chan_skinny be in 1.4?  (I assume the answer is no)
01:41.32*** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg)
01:41.33russellbpaolob: the development branch of code that will be released as asterisk 1.4 in the near future
01:41.45[TK]D-Fenderrussellb : Whats the status of SIP-B Shareld Line Appearances?
01:41.46russellb[TK]D-Fender: we hope so :)
01:41.56*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-246-145.buckeyecom.net)
01:42.11russellb[TK]D-Fender: that was something that was talked about a lot at our developers conference ... no code in trunk yet, i don't think
01:42.25russellbit's pretty complicated to do
01:42.26gambolputtyHi.  Can * detect if an incoming call is already muted?
01:42.27[TK]D-Fenderrussellb : DAMN... thats a huge selling point...
01:42.44QwellStrom_C: the answer is yes, actually
01:42.50Strom_CQwell, oh awesome!
01:42.54russellbStrom_C: it went in today, because Qwell r0x0rz
01:42.56QwellStrom_C: which is why I committed the branch today :)
01:43.20russellbok, i better log off of IRC before people find more bugs
01:43.21Strom_Cdouble awesome on a stick with sugar and caramel and pralines and holy shit now I want ice cream
01:43.33Strom_Cis there a good ice cream place in huntsville? :/
01:43.43QwellStrom_C: are you in huntsville?
01:43.46Strom_Cyep
01:43.52Qwellfun
01:44.03Strom_CI go back to los angeles on tuesday
01:44.29Strom_CQwell, you can do shared line appearances with skinny, cant you
01:45.34Qwellnope..not yet
01:45.45Strom_Cah ok
01:46.56QwellStrom_C: easy
01:47.04Qwellasterisk should automatically create your dialplan for you
01:48.02Strom_Cwill that be res_bippity_boppity_boo.so?
01:56.06littleballhello, i want to enable "invite" from any sip client without authorization. How to?
02:03.47*** join/#asterisk pbd (n=pbd@c-67-163-20-134.hsd1.il.comcast.net)
02:04.14pbdEvening, all.
02:04.58*** join/#asterisk dlynes_home (i=1000@S0106001217014b92.vc.shawcable.net)
02:05.14*** join/#asterisk Snake-Eyes (n=blog@203.220.55.70)
02:11.57*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)
02:12.41*** join/#asterisk rbd (n=rbd@cpe-066-057-011-095.nc.res.rr.com)
02:13.13rbdhey guys, anyone using asterisk on debian with debian's asterisk packages? any idea if ztdummy comes precompiled anywhere, or if I have to compile that from source?
02:13.47Qwellrbd: It's best not to use the packages..
02:14.57rbdQwell: seems like it, I'll grab the source
02:15.28Strom_Chmm, is there a #asterisk FAQ page?
02:15.38neilbags-workQwell: why is it best not to use packages? are the packages broken?
02:15.43pbd~wiki
02:15.46Qwellneilbags: usually, yes
02:16.13neilbags-workQwell: so who is the maintainer? is he/she doing a poor job?
02:16.15pbdThey're usually old, if nothing else.
02:16.20neilbags-workok
02:16.26Qwellneilbags-work: there is at least one maintainer for each distro
02:16.46Qwelland they all have wildly different ways of packaging it
02:16.57neilbags-workQwell: well i use gentoo and the ebuilds work great and they are totally up to date, i guess other distros lag
02:17.22Qwell!!! All ebuilds that could satisfy "asterisk" have been masked.
02:17.24Qwellsorry, no
02:17.55*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)
02:19.20neilbags-workQwell: well i mean the ones in ~x86 work great, and are up to date. 1.0.11_p1 is the latest unmasked version on x86, are you using a differerent arch?
02:19.45Qwell1.0.11.1 is old
02:19.51QwellI mean...it's new...but it's old code
02:20.33pbdA 1.0 release? Ick. Current stable is 1.2.  A loooong road in between.
02:20.45neilbags-workQwell: i know, but the 1.2.9.1 ebuild works fine ... there would be no reason not to use the ebuild ...
02:20.56Qwellexcept that asterisk packages suck :)
02:21.46*** join/#asterisk littleRalphie (n=ralph@c-67-162-230-107.hsd1.tx.comcast.net)
02:22.27neilbags-workQwell: its bad form to compile it from source yourself, i write my own ebuild for things that don't exist, if they suck then they should be fixed, not avoided
02:22.46neilbags-workQwell: i havn't come across any 'sucking' yet so i havn't had to make any changes to the ebuilds
02:22.54littleballhello, i want to enable "invite" from any sip client without authorization. How to?
02:22.56Qwellit's not really possible to make good asterisk packages.
02:23.15Qwellit makes it incredibly difficult, by design
02:23.22littleRalphielooking for a quick tip
02:23.32neilbags-workQwell: how come? what about just a basic-ebild that does ./configure && make && make install?
02:23.57Strom_ClittleRalphie, just ask your question
02:24.02Qwellbecause it's far more complex than that
02:24.08neilbags-workQwell: surely its better to use the ebuild even if only to track what files have been installed!
02:24.10Qwellespecially now, with menuselect
02:24.22littleRalphieI added gnome to my trixbox 1 system, but I dont get how to start the gui once booted
02:24.29Qwell~trixbox
02:24.53Strom_C#freepbx
02:24.58Strom_C~freepbx
02:25.00jboti guess freepbx is NOT supported here!  People using it should join #freepbx (FreePBX is the new name of AMP)
02:25.08neilbags-workQwell: why not have an ebuild call menuselect?
02:25.18Qwell~trixbox
02:25.19jbothmm... trixbox is NOT supported here!  People using it should join #trixbox or #freepbx (FreePBX is the new name of AMP)
02:25.33Qwellneilbags-work: which would do what?
02:26.02neilbags-workQwell: huh? i don't quite understand what you're asking
02:29.41neilbags-worki'm interested to hear any good reasons why using the gentoo ebuilds is bad compared to compiling manually ... i'd be really surprised if any reasons warranted not modding the ebuild.
02:30.05drraywhy not just use the svn?
02:30.14drraywhat do you gain?
02:30.17Qwellneilbags-work: because it would be impossible to get every single option in make menuselect, as a USE flag
02:31.21JoseBravoWhy im getting this error,  ast_channel_make_compatible: No path to translate from SIP/dominet-9b29(256) to SIP/70103-ed68(4)?
02:31.28JoseBravoTriying to do a out call
02:31.31neilbags-workQwell: ok. but if menuselect is so important, why not let the ebuild run menuselect? at least then you can keep track of what files asterisk is installing for ease of removal/upgrades
02:31.48Qwellneilbags-work: because no other ebuilds are interactive
02:31.50fileJoseBravo: one side is G729, one side is ULAW, and your Asterisk can't transcode between them
02:32.24Qwellfile: You aren't allowed to remember codec numbers
02:32.35*** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com)
02:32.36file:(
02:32.37Qwellonly 4 and 8..no more
02:33.04neilbags-workQwell: yes valid, but but even if asterisk is an exception, why not make that exception? if the gentoo people don't like it then keep it as an 'unofficial' ebuild in an asterisk overlay.
02:33.21Qwellbecause then it becomes an unofficial ebuild
02:33.36SplasPoodmuahahhaa... I can now have queue agents on cell phones
02:33.36neilbags-workQwell: i still don't think that arguement warrants dodging portage to install asterisk
02:33.37Qwellwhich, NOBODY will use
02:35.16*** join/#asterisk RF_MIA (n=Administ@69-172-194-16.miamfl.adelphia.net)
02:35.48neilbags-workQwell: i would ... wouldn't you? ... if menuselect is preventing you from using portage isn't it better to have an ebuild that is a little non-standard (not like its going to break anything) than doing it manually from source?
02:36.10Qwellneilbags-work: no
02:36.11neilbags-workQwell: gentoo is designed to be all-encompassing ... if asterisk needs an interactive build then it should have it
02:36.16SplasPoodmailbags: asterisk isn't all that messy with it's files
02:36.18drrayagain, what is wrong with svn?
02:36.26SplasPoodand I think just no one cares enough about gentoo to .. care
02:36.32Qwellneilbags-work: Then why doesn't gentoo let you compile your kernel, from just emerge? :)
02:37.10pbd(while being supportive of the 'compile from source' bunch- I have to say that, adding svn to a machine is a PITA, with dependancies and, if left alone, mangled httpd.conf when complete)
02:37.21neilbags-workdrray: when you compile a kernel it doesn't scatter its files through /usr and become a mess in the future
02:37.38neilbags-worksorry, i meant Qwell not drray
02:37.57Qwellneither does asterisk
02:38.36neilbags-workyes it does, it puts plenty in /usr/lib and /usr/share
02:38.57drrayso you've been using asterisk how long?
02:39.31neilbags-workQwell: i'm not saying the asterisk build is bad, its really clean ... but i still don't see how you justify dodging portage
02:40.42SplasPoodpbd: wtf are you smoking?
02:40.51JoseBravofile Thanks
02:41.22pbdNot to be purely argumentative here, but why *should* the asterisk-dev community support any particular distro over another?
02:41.24SplasPoodneilbags-work: I justified it... no one cares enough to deal with it...  You could...
02:42.01pbdSplas- I've installed svn on several different distros, versions, kernels, and machines- and it was never 'smooth'.
02:42.23SplasPoodpbd: Odd, I've installed it everywhere and its been perfectly smooth.. which distros
02:42.34neilbags-workSplasPood: i guess i'm more careful than most with my gentoo boxes ... i won't put *anything* in /usr unless portage installs it
02:42.37SplasPoodpbd: as a client, I mean
02:42.50pbdMosty slack and suse- also some FC.
02:42.54SplasPoodneilbags-work: I don't run gentoo
02:43.11SplasPoodpbd: heh 3 distributions I haven't run in quite some time
02:43.12pbdOnly as a client- haven't delved into using it as a repository for my own work yet.
02:43.22SplasPoodpbd: I use it for both, on debian
02:43.28SplasPoodas a client on macosx (darwin)
02:43.30pbdSplas: Hence the difficulty. :)
02:43.33SplasPooddebian
02:43.34SplasPoodcentos
02:43.39neilbags-workSplasPood: 'i justified it' ? this implied you were using gentoo
02:43.48littleRalphieok, well trixbox seems cool, but is there a better approach to speed up my learning curve?
02:43.56SplasPoodneilbags-work: No I justified why there's no working ebuild or whatever.. by saying "no one cares"
02:44.37SplasPoodI think the problem is that an asterisk compile doesn't A) take long enough or B) have enough ansi eye candy to warrant it's own ebuild
02:44.54drrayheh
02:44.56neilbags-workSplasPood: there is a working ebuild, but it doesn't run menuselect for you to fiddle with ... i'm just wondering whether its worth having an ebuild that does run it
02:44.59*** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com)
02:45.15SplasPoodportage is only for 24+hr seizure inducing ansi nightmares
02:45.29SplasPoodmenuselect?
02:45.53drraywould portage take longer than SVN asterisk?
02:46.08neilbags-workdrray: define 'take longer'
02:46.16SplasPooddrray: depends on how much "optimization" you want and how "k-rad" you wish to feel
02:46.27SplasPoodwww.funroll-loops.org
02:46.38neilbags-workSplasPood: Qwell said he doesn't use ebuilds because of menuselect
02:46.57Qwellno, that is one reason
02:47.03QwellI have many other reasons...such as being a developer
02:47.22neilbags-workQwell: well that makes sense for sure, i'm talking about production use
02:47.34SplasPoodChief among them is not caring, I bet
02:47.43SplasPoodPRODUCTION MUST HAVE ANSI
02:47.59QwellI use svn trunk in prod
02:48.07QwellThat's just how I roll
02:48.24SplasPoodword.
02:48.35neilbags-workdo you remove prior versions before an update or install over the top?
02:48.42drrayover
02:48.49pbdQwell likes to live on the edge. :)
02:48.55SplasPoodif you keep copying binaries over one another there's buildup
02:48.57SplasPoodcan't have that
02:49.36Qwellneilbags-work: I just make install, and it works
02:49.44Qwellperhaps I just know what I'm doing
02:49.48SplasPoodif asterisk wasn't such a moving target i'd be all for proper binary packages
02:50.15neilbags-workSplasPood: gentoo doesn't use binary packages
02:50.25pbdIf asterisk wasn't such a moving target- it would be like thousands of other OSS projects out there.
02:50.26neilbags-workQwell: well you know the source and build better than me
02:50.27rbdguys, I try to appexec festival, and I get the voice but it's very choppy
02:50.30SplasPoodneilbags-work: I know, hence my use of the word 'proper'
02:50.50SplasPoodrbd: you wan cepstral
02:50.54SplasPoods/wan/want
02:54.10neilbags-workQwell: askerisk is just one piece of software i maintain on gentoo boxes i can't know every file that every package installs (thats what portage is for!) so not being an asterisk developer, there is no reason for me to treat asterisk any different. if i need to write an ebuild that checks out svn and runs menuselect then so be it
02:55.33SplasPoodneilbags-work: you're probably better off with asterisk in a box anyway
02:56.02*** part/#asterisk RF_MIA (n=Administ@69-172-194-16.miamfl.adelphia.net)
02:56.06rbdSplasPood: cepstral? googled it..looks interesting, looking for a trial/free link
02:56.47Corydon76-homeYou can download any Cepstral voice for free
02:57.02SplasPoodrbd: you can use it as a demo, but it adds "blah blah, this isn't paid for" to your stuff
02:57.05Corydon76-homeThe only thing is, all phrases are prefixed with a demo phrase
02:57.27*** join/#asterisk Cerlyn (i=ALEIN@pdpc/supporter/sustaining/Cerlyn)
02:57.28SplasPoodits reasonably priced tho
02:57.41Corydon76-homeCan't argue too much with $30/voice
02:57.42neilbags-workSplasPood: well i'm happy with the way things are going now, i just didn't know about the ebuilds 'sucking'. i don't have dedicated boxes for asterisk anyway.
02:57.42SplasPoodI wonder how the price drops off when you get into volume, I didn't look
02:57.56Corydon76-homeNo, the price is actually cheaper one at a time
02:57.57SplasPoodneilbags-work: who said they suck?
02:58.03neilbags-workQwell
02:58.23Corydon76-homeI truly love domain names for $9.99/year, but 19.99/2years
02:58.23SplasPoodCorydon76-home: I saw something about buying "ports" ... seemed to be simultaneous use stuff
02:58.39Corydon76-homeDon't they realize that the price for multiple years is more expensive?
02:58.58*** join/#asterisk h0 (n=h0@ool-44c69453.dyn.optonline.net)
02:59.02Corydon76-homeSplasPood: the same is true for Cepstral, but it's only pennies
02:59.08*** join/#asterisk Strom_C (n=strom@m615e36d0.tmodns.net)
02:59.14Strom_Cno no, the answer is "cake"
03:00.19h0if i have a TDM400P then i can use any linux network interface card to conect to the asterisk server with a softclient corect?
03:00.21drraythe answer is always cake
03:00.30Strom_Ch0, yes
03:00.34h0k thanx
03:00.36drrayyou need to rethink the question if that does not work for you
03:05.26*** join/#asterisk [koss] (i=koss@adsl-68-76-106-229.dsl.bcvloh.ameritech.net)
03:06.01*** join/#asterisk inv_Arp (i=junya@c-67-191-62-53.hsd1.fl.comcast.net)
03:06.16*** part/#asterisk pbd (n=pbd@c-67-163-20-134.hsd1.il.comcast.net)
03:07.22*** join/#asterisk erwinism (i=erwin@61.9.118.37)
03:07.29erwinismhello
03:08.19erwinismi have asterisk server, one x100p single FXO card, can anyone help me to make my asterisk work. :)
03:09.03Strom_Cerwinism, sure.  I can do everything for you at my standard hourly rate :)
03:10.03erwinismStrom_C,  does that mean its hard to configure this thing?
03:10.08*** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)
03:10.22Strom_Cno, it means that you are encouraged to read the documentation :)
03:10.29SplasPooderwinism's question is topic worthy
03:11.01Corydon76-homeThe difficulty of configuring Asterisk increases only with the scale of complexity of what you want it to do
03:11.11fileand the IQ requirement increases too
03:11.22Strom_Cso if you just want it to route calls, easy.  If you want it to bake cake, tough.
03:11.33fileres_oven!
03:11.34erwinismhaha, ok, im not a tech guy. so maybe i need to read more
03:12.17Strom_Cerwinism, if you have specific questions, we're more than happy to answer them
03:12.19erwinismi just want to make a simple pbx at my office using PC, and be able to accept calls from PSTN
03:12.26*** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn)
03:12.43SplasPooderwinism: I encourage you to checkout trixbox or one of the other asteriskInABox type things
03:12.43Strom_Cbut I doubt you will find anyone here willing to regurgitate the entire asterisk book from memory :)
03:12.47erwinismStrom_C, ok, sorry :) i have to read the manual :)
03:12.55Strom_Cand I encourage you not to do that
03:13.04Strom_CGUIs are the devil
03:13.05SplasPoodStrom_C: Why not?
03:13.07SplasPoodtrue
03:13.10*** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)
03:13.11fileI will do it, but I will say exactly what happens in the core
03:13.14Strom_Cespecially for something like asterisk
03:13.17fileright down to the concepts and API calls
03:13.33SplasPoodStrom_C: Seems to be exactly what this person wants tho..
03:13.50*** join/#asterisk vivek (n=vivek@unaffiliated/tintin)
03:13.53Strom_CSplasPood, yes, but the problem is that he'll learn to do things the wrong way
03:13.59erwinismSplasPood, ok i will search about trixbox
03:14.02vivekhello all
03:14.13Strom_Cdamnit, there goes another one
03:14.16Strom_Chello vivek
03:14.27erwinismhahaha
03:14.28Strom_Cerwinism, i would advise you not to waste your time with trixbox
03:14.44SplasPoodStrom_C: Sorry
03:14.45viveka little off topic. Can i transfer a call from pstn to sip numbers using a sipura or some other sip hardware ?
03:14.51erwinismok, im reading the version2 of the asterisk handbook
03:14.56SplasPooderwinism: trixbox won't teach you anything.. but at first it'll be "easy"
03:15.01file~thebook
03:15.08Strom_C~book
03:15.11jboti heard book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
03:15.11filedarn it I do that every time
03:15.32pjchilds~just-buy-a-key-system
03:15.42pjchildshmm.... :)
03:15.48Strom_Cpjchilds, ooh, Western electric 1A2? :)
03:16.17Strom_Cfile, 5ESS or 5ESS-2000?
03:16.29filejust 5ESS
03:16.33Strom_Cah ok
03:16.42*** join/#asterisk L|NUX (n=linux@202.5.145.56)
03:17.20fileCerlyn: 'tis a telco switch
03:17.35pjchildsor the 'VCDX' the (very compact digital exchange...)
03:17.50CerlynYou mean they don't just fill it up with a bunch of SPA-2000's? :)
03:17.53pjchildsminimum footprint... three cabinets 6 feet heigh...
03:18.16*** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net)
03:18.31Strom_Cfeh.  DMS-10 for the win
03:18.47fileStrom_C: can you only do centrex off a DMS100?
03:19.00Strom_Cfile, I think DMT can do centrex as well
03:19.17fileI've been trying to figure out what my telco uses
03:19.33filewell, not really trying... just randomly wondering
03:19.35Strom_Cfile, PM me your NPA/NXX
03:19.53fileI'm Canadian don't forget :D
03:20.28Strom_Cclaims DMS-100
03:20.38Strom_CAlliant Telecom
03:20.41filethought so
03:22.55*** join/#asterisk droops (n=droops@adsl-065-005-212-128.sip.jan.bellsouth.net)
03:23.25Strom_Cdroooooops!
03:28.50littleRalphiequestin about sipp_additional.conf?
03:29.04*** part/#asterisk justdave (n=dave@unaffiliated/justdave)
03:30.03littleRalphieI can connect to telregister to telasip.com with a sip client but not in asterisk
03:30.21rene-~amp
03:30.22jbotit has been said that amp is NOT supported here!  People using it should join #freepbx (FreePBX is the new name of AMP)
03:30.53*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
03:30.53*** mode/#asterisk [+o russellb] by ChanServ
03:32.01rene-littleRalphie: you could try to see what errors are you getting by issuing a sip reload from your asterisk CLI (asterisk -r and then sip reload)
03:32.17erwinismSplasPood, you there?
03:34.04littleRalphiejust reattemmpts, is this loggd somewhere?
03:35.45*** join/#asterisk TheCops (i=nobody@got.securebinary.com)
03:36.09littleRalphieREGISTER attempt 81 to rshumway@gw4.telasip.com
03:36.09littleRalphieREGISTER attempt 81 to ralphshumway@yahoo.com@sip.stanaphone.com
03:36.10littleRalphieasterisk1*CLI>
03:36.32littleRalphiethis is alli see
03:36.57TheCopsSomeone have 1.6.6 firmware of polycom SIP phone?!
03:37.07*** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net)
03:37.53rene-before that you can enable a sip trace  with sip debug on, and that will give you the error you are getting
03:39.02rene-is stanaphone working for you?
03:39.08littleRalphienope
03:39.14rene-telasip?
03:39.18littleRalphiedebug on now
03:39.22littleRalphieneither
03:40.15litageis it possible to limit the size of an extension's voicemail box?
03:40.26rene-but you are using AMP/FreePBX/ or something like that right?
03:40.42sevardas far as i know, no.. but you can limit the max amount of messages in said box
03:42.00rene-littleRalphie: go here for an example: http://forum.stanaphone.com/viewtopic.php?t=2342&highlight=asterisk
03:42.52*** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com)
03:43.07rene-you should be able to find sample configurations for your other voip provider unless it is very new or very little known
03:43.27rene-~google
03:43.28jboti heard google is a search engine found at http://www.google.com/
03:44.07littleRalphieI followed mr nerdvittles docs, but perhaps I missed somthing
03:44.43rene-some of the users posted a working configuration
03:45.01rene-try to match that to your specific acct details
03:45.31SplasPooderwinism: yes
03:47.11*** join/#asterisk dlynes_home (i=1000@S0106001217014b92.vc.shawcable.net)
03:49.06erwinismSplasPood, i installed trixbox :)
03:49.31sevardsad.
03:51.06erwinismi have to make everythings working before wondering how these things works hehe
03:52.12sevardthat's true, i started with AMP and after I had a vague understanding deleted it
03:58.32*** join/#asterisk CrashHD (n=crashhd@c-67-182-167-222.hsd1.ca.comcast.net)
03:58.41CrashHDhow can I use a gotoif to tell if a file exists?
03:59.15*** part/#asterisk rene- (n=rene@201.152.34.100)
03:59.18CrashHDSTAT doesn't seem to be registered
04:01.14CrashHDhow is EXISTS used?
04:03.20*** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135)
04:12.34*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
04:15.42Corydon76-homeSTAT is only in trunk
04:15.52CrashHDahh
04:16.08Qwellheh, I used a little hack in 1.2
04:16.30QwellSystem(ls filename) with SYSTEMSTATUS
04:16.36russellbyou dialplan h4xor
04:16.43Qwell:D
04:16.53russellbhey, that's a pretty cool way to do it
04:17.11QwellI like func_backtick...
04:17.16Qwellhaven't used it, but I like the idea
04:17.34russellbwhat's the idea
04:17.55Qwellbacktick..  like running a subcommand from a shell
04:18.06Qwellls -l /lib/modules/`uname -r`/
04:18.10Qwellso, like
04:18.11Corydon76-homeIsn't that what System does?
04:18.18QwellCorydon76-home: it doesn't return a value though.
04:18.30*** join/#asterisk subdolus (n=subby@subby.afraid.org)
04:18.32russellbok, yeah, i gotcha
04:18.33Corydon76-homeAh...
04:18.41Qwellexten => s,1,System(ls -l ${BACKTICK(uname -r)})
04:18.45Qwellor something
04:19.01russellbheh, would be fairly straight forward to write
04:19.13littleRalphieI currently run a sunrocket gizmo direc to my cable modem, culd that be my issue since the device uses port 5060?
04:19.26Qwellprobably
04:19.29Corydon76-homeYou could have done ${CURL(http://localhost/exists.cgi?${filename})}
04:19.51QwellCorydon76-home: heh, hack
04:19.55QwellI like it :D
04:21.15russellbi think the MacroExclusive app is a cool idea
04:21.21Qwellhuh?
04:21.27QwellI think I saw that...  on -dev?
04:21.28russellbit was something discussed on the dev list
04:21.34russellbbut it's on mantis now, i think
04:21.37russellbhaven't looked at the code, yet
04:22.18Corydon76-homeYou could do the same thing with Groups
04:22.43Corydon76-homenonblocking, though
04:23.12Corydon76-homeThat's always nice
04:30.19*** join/#asterisk P-NuT (n=nut@fw.office.unitedip.net.au)
04:31.04P-NuTHi all, has anybody in australia got an x100p card to sell me?
04:31.49*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
04:36.49*** part/#asterisk Cerlyn (i=ALEIN@pdpc/supporter/sustaining/Cerlyn)
04:49.57*** join/#asterisk Strom_C (n=strom@m615e36d0.tmodns.net)
04:50.23*** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone)
04:52.43Qwellbrimstone: y0
04:53.05Strom_Cbrimstone!
04:57.13*** join/#asterisk janeNarak (i=chonlada@jane.lru.ac.th)
04:57.48janeNarakhi all, Asterisk are support Skype?
05:00.35Strom_Cnot natively
05:04.58CrashHDhow can I stop the dial function from bridging the call (I'm trying to use the M option but there is about a second where the channels are bridged)????
05:05.12CrashHDbefore the macro starts
05:06.09*** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com)
05:10.06erwinismin order to accept calls from pstn and forward to extention, what configuration should i edit? i have x100p already installed
05:11.01vivekjaneNarak: do you have some skype hardware ? if you don't you can do without skype ...
05:13.49*** join/#asterisk eset (n=eset@203-114-177-203.dsl.sta.inspire.net.nz)
05:14.00esetdoes speex work in a conf room w asterisk?
05:14.38Qwellyes
05:15.03sevardit ought to
05:15.54Strom_Cwheeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeee
05:17.07sevardhappy drugs?
05:17.25Strom_Ceven better
05:17.30Strom_Cive got a date on friday
05:17.38sevardmailman?
05:17.39*** join/#asterisk SuperLag (n=aaron@gentoo/developer/SuperLag)
05:18.09Strom_Cwhat?
05:18.20sevarddid you get a date with the mailman?
05:18.25Strom_Cno
05:18.41sevardcop?
05:18.44Strom_Cno
05:18.52sevardanyone from the village people
05:19.01Strom_Clet me check
05:19.02Strom_Cno
05:19.05sevardwtf
05:19.21sevardI thought you liked uniforms.
05:19.59Strom_Conly if the uniform has a telephone company logo on it
05:21.30*** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)
05:22.03*** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk)
05:25.12*** part/#asterisk neilbags-work (n=neilbags@149.171.94.134)
05:28.33*** join/#asterisk Eggplant (n=none@dsl-72-19-42-179.cascadeaccess.com)
05:33.14erwinismStrom_C, i only have x100p card, what should i name it for outgoing channel?
05:33.34Strom_CZAP/1
05:33.53erwinismok i will try that one ;)
05:34.01sevard"Sandy"
05:34.04sevardor HAL
05:35.28Strom_Cdave
05:35.32Strom_Ci'm sorrry dave
05:35.37Strom_Cyour call did not go through
05:36.08sevardhahaha
05:36.21sevarddaisy daisy, your sip client won't register
05:36.30sevardk mine wasn't as good.
05:36.47Strom_Cseriously, how cool would it be to get Douglas Rain to do voice prompts for asterisk
05:36.58sevardwho the hell is that
05:37.14Strom_Cthe guy who did the voiceover work for HAL9000
05:37.15Strom_Cduh
05:37.21sevardjesus like i'm fucking imdb
05:37.38sevardbut no, that wouldn't be cool.  that would be scary.
05:37.46sevardthere's a reason why we have cheerful women doing our prompts.
05:38.15erwinismStrom_C, i already named it ZAP/1 when i tried calling 9 plus 2971274 it tells me "busy" ...my pstn is not busy, maybe i missed something
05:38.34sevardit's your pstn?
05:38.35Strom_Cerwinism, are you using asterisk or are you using trixbox?
05:39.09erwinismStrom_C, i installed trixbox, but i configured it manually
05:39.50Strom_Cerwinism, pastebin your extensions.conf, zaptel.conf, and zapata.conf files
05:39.53Strom_C~pb
05:39.55jbotfrom memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/
05:40.04erwinismStrom_C, ok :)
05:41.47erwinismStrom_C, brb, boss is calling me hehe
05:42.03*** join/#asterisk neilbags-work (n=neilbags@149.171.94.134)
05:42.42CrashHDI can't seem to get mp3's to play for music on hold, anything special I need to do?
05:42.53Strom_Cdid you install mpg123
05:42.55CrashHD(I just uploaded one of my mp3's I usually listen to for testing)
05:42.58CrashHDno I did not
05:42.58*** join/#asterisk znoG (n=gs@205-17-235-201.fibertel.com.ar)
05:42.59Qwellor asterisk-addons
05:42.59CrashHDrunning 1.2
05:43.05CrashHDno asterisk-addons either
05:43.11Qwellwell...
05:44.32Strom_Cthis is kind of like asking "why won't my car start?" when you have never purchased gasoline for it
05:44.39CrashHDlol
05:45.00CrashHDif I knew a car ran on gasoline that would make sense, but if I didn't...I would still be where I'm at now...
05:45.19*** join/#asterisk Sponge_bob (n=None@cpe-66-27-162-13.socal.res.rr.com)
05:46.47CrashHDmod=quitemp3
05:46.58CrashHDthis will not work by default
05:47.02CrashHDI'm guessing
05:47.03CrashHD?
05:47.26Strom_Cwell, especially if you misspell mode like that
05:47.35CrashHD*nods*
05:50.21QwellYou mean it doesn't just know what you want?
05:50.46Strom_Cmaybe he forgot to compile res_esp.so
05:50.54Qwellmust be it
05:55.20Strom_COOOH
05:55.28Strom_CThere's Waffle House in Arizona!
05:55.45Strom_Cat this rate, there will eventually be one in Los Angeles
05:56.28sevard...
05:56.48dlynes_homesevard: ow!  wtf was that for?
05:57.25sevardHOLY CRAP MAN
05:57.35sevardthat was delayed... how many hours?!
05:57.52*** join/#asterisk aclark78 (n=aclark@cpe-70-116-103-202.houston.res.rr.com)
05:58.35mitchelocStrom_C: we already have ihop and arthurs ;)
05:58.53Strom_C"arthurs"?
05:59.14mitchelocwhat city are you in?
05:59.45Strom_Calso, can you walk into an ihop, order hash browns scattered, covered, diced, and be greeted by someone who even has a clue what the hell you're yammering on about?  no!  :)
06:00.04Strom_Cmitcheloc, I live in Los Angeles, though right now I'm in Huntsville AL
06:00.26mitchelocarthurs is in whittier, are you visiting digium?
06:00.58Strom_Coh ok, i'll have to schlep out to Whittier when I get back
06:00.58Strom_Cyes, I am
06:01.53sevardStrom_C: the only ihop i was at was in las vegas and our waitress was clearly a huge meth head
06:02.08Strom_Cdoesn't surprise me in the least
06:02.16Strom_Cthat's vegas for you
06:02.21sevardshe had about 5 teeth left
06:03.03*** join/#asterisk mrtwister (n=manopulu@107.250.broadband5.iol.cz)
06:03.21mitchelocStrom_C: i'm supposed to visit there next month ;)
06:03.28Strom_Cmitcheloc, ah cool
06:03.39Qwellmitcheloc: huntsville?
06:03.53mitchelocQwell: yep
06:04.01Qwellwhat timeframe?
06:04.30mitchelocthe 6th to the 10th, why do you ask?
06:04.42Qwellbecause I'll be there on the 9th..
06:04.54Qwellseems like a happenin' place, heh
06:05.14*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal)
06:05.22mitchelocah, yea, are you going on business? (i'm just going for fun)
06:05.27Strom_Cthere's waffle house and bumpers and...um...what else is there to do here besides phones and grease and booze and 8-ball?
06:06.28sevardyou're a coke fiend?
06:06.42Strom_Cno
06:06.47Strom_Cneither coca-cola nor cocaine
06:06.47sevardliar.
06:11.41dlynes_homesevard: i was sleeping..sheesh
06:11.51Strom_Cwaffle house!
06:11.56*** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it)
06:12.51mitchelocheh Strom_C is it really that good? i'll have to check it out then when i go ;)
06:13.14Strom_Cmitcheloc, yes
06:13.17Strom_Cit's addictive
06:13.49dlynes_homemitcheloc: the coke, or the meth?
06:14.04Strom_Cthe meth waffles
06:14.13Strom_Cand the coke hash browns
06:14.21mitchelocsounds good then, it's worse weather though isn't it? =(
06:14.30Strom_Chot and muggy
06:14.32Strom_Cbring shorts
06:14.33Strom_Ctshirts
06:14.38Strom_Cand no underwear
06:14.47mitchelocaww/eww
06:14.56Qwellw00t - 2.6.17 == niagara support
06:15.08dlynes_homeniagara?
06:15.16Qwellnew Sun chip
06:15.18dlynes_homeyou can go over the falls in a barrel with linux now?
06:15.19dlynes_homeoh
06:15.25*** join/#asterisk P-NuT (n=nut@fw.office.unitedip.net.au)
06:15.34P-NuTHi all.
06:15.58*** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg)
06:16.18P-NuTDoes anyone know what the cheapest FXO card is I  can get in Australia?
06:16.27dlynes_homeP-NuT: x100p
06:16.29P-NuTsingle line.
06:16.46dlynes_homeyou didn't say how many ports
06:16.48Strom_Cof course, the X100P will make you stab your eyes out with white-hot knitting needles
06:16.49dlynes_homeand you said cheapest
06:17.06mitchelocwhat i never understood is why asterisk doesn't support regular old modems...
06:17.20dlynes_homemitcheloc: it does
06:17.26dlynes_homemitcheloc: but only two specific models
06:17.33Strom_Cbecause most regular old modems don't really work all that well for voice communications
06:17.50dlynes_homemitcheloc: which is what the x100p, and x101p cards are
06:17.54neilbags-workdlynes_what models are supported, and do they work well?
06:17.57sevardwhat about voice modems?
06:18.03mitchelocdlynes_office: yep but i can't pick those up at Fry's
06:18.13dlynes_homeneilbags-work: intel and motorola
06:18.26Strom_Cmitcheloc, just buy a tdm400 and quit kvetching :)
06:19.23dlynes_homeP-NuT: if you don't want x100p, i'd suggest a sangoma a200 or a200d with 1 to 12 fxo modules (2-24 fxo ports)
06:19.26mitchelocnah, the real truth is a conspiracy to get us to buy tdm400's ;)
06:19.40dlynes_homeP-NuT: a200 is the regular card; a200d is the same card, but with hardware echo cancellation
06:23.44sevardfreaka you
06:23.46sevardfreaka me
06:24.23P-NuTwell, I have a x100p and it works totally fine for me.
06:24.30P-NuTI've had no issues at all.
06:24.50neilbags-workhi, is anyone using chan_bluetooth? is it un the current stable asterisk releases?
06:25.08P-NuTBut I want to buy a card for a mate's asterisk box, but x100p's are hard to come by here is Oz. Anybody wanna sell me one? lol
06:25.25Strom_CP-NuT, the x100p has been discontinued
06:26.21*** join/#asterisk satlan32 (n=pargit@212.150.142.211)
06:26.35neilbags-workP-NuT: i'm having a weird problem with my x100p on telstra, if you call in just after the line has been hung up (say 10 seconds later) asterisk answers the call and emits a high-pitched tone ... considering trying different hardware
06:27.05Strom_CTDM400P with single FXO module will be reasonably inexpensive and work well for you
06:29.17P-NuTneilbags-work: I'm going to write an article about x100p and telstra soon, so hang out for a few weeks.
06:29.27orlockjust blame telstra
06:29.35P-NuTa few settings you need in zapata.conf apparently
06:29.36orlockthere wont be anybody there knowledgable enough to deny its their fault
06:29.38orlock:)
06:29.40neilbags-workP-NuT: do you know about this problem?
06:30.11neilbags-workP-NuT: i get an error saying that the caller ID module exited with an error, something like that
06:30.21P-NuTno, but I had a few weird issues with telstra and x100p until I stuffed about with zapata.conf
06:30.28P-NuTtry this...
06:31.01neilbags-workis the x100p obsoleted by x100m, or am i confused?
06:31.17Strom_Cx100p is replaced by TDM400P
06:31.38mitchelocor you could use a t100p to create a channel bank
06:31.45neilbags-workok, P-NuT i have a TDM400P with 1 FXO module, not a x100p
06:31.54mitchelocand yes i'm serious, i've got one for my house ;)
06:32.01neilbags-worki think
06:34.10neilbags-workP-NuT: i used http://www.voip-info.org/wiki/view/Australia+Asterisk+Details, is this info correct?
06:36.59*** join/#asterisk oej (n=oej@apollo.webway.se)
06:38.58*** join/#asterisk littlejohn (n=little@82.53.4.87)
06:39.21P-NuTneilbags-work: I use this in zapata.conf
06:39.23P-NuThttp://pastebin.ca/67058
06:40.26P-NuTtry that.
06:40.37P-NuTbut change the context to your context.
06:41.22neilbags-workP-NuT: yeah but there is nothing in there that looks like it'll solve my obscure problem
06:41.29neilbags-workhere's mine: http://pastebin.ca/67059
06:41.47nextimealeeee
06:41.51nextimeops
06:42.10P-NuTwell, mine looks kinda different..
06:42.11neilbags-workwhat does callwaitingcallerid=yes do?
06:42.32neilbags-worki have call-waiting turned off so it shouldn't matter
06:42.40P-NuTwhy dont you try my config? but.... where is your context?
06:43.06neilbags-workyeah they are different but mine does work in every other situation .... default context
06:43.23P-NuThmm..
06:43.34P-NuTgive mine a try. You have nothing to lose..
06:44.10neilbags-worki'll try it next time i'm on site, cant break things in business hours :(
06:44.30neilbags-worki just can't see anything in there that would solve my problem
06:44.36neilbags-workwhat does callreturn=yes do?
06:44.46P-NuTAhh yes...
06:44.49P-NuTumm...
06:44.51P-NuTyes..
06:45.11P-NuTno idea.
06:45.15neilbags-workand i assume all the callwaiting stuff is irrelavent since i have it disabled on the line
06:45.37neilbags-workmy gains are different, but again, they work fine for me
06:45.42P-NuTbut all I know is all of my weird call hanging issues, low volume and not hanging up properly were magically solved by that/
06:45.59neilbags-workby what exactly?
06:46.57P-NuTchanging the zapata conf to that
06:47.09*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal)
06:47.11*** join/#asterisk akke (n=akke@85.158.211.235)
06:47.11neilbags-workP-NuT: ok, where did you get that config from?
06:47.15P-NuTbut hey, I hope it works for you.
06:47.17P-NuTAhh
06:47.24P-NuTbits from here...
06:47.33P-NuThttp://www.loligo.com/asterisk/current/
06:47.43P-NuTgreat config example that one..
06:47.44neilbags-workP-NuT: can you confirm that you don't have the same problem as me? try calling the line as soon as it has been hung up ...
06:47.49akkecan anyone recommend me a provider that offers a 'dial-in number' from belgium?
06:48.23P-NuTI do that and it's still offhook. asterisk doesn't hangup the call imediatley
06:48.40P-NuTso therefore I dont get the beeping that you have.
06:48.48P-NuTBut hey, it could always be dodgy hardware.
06:49.08P-NuTso what has replaced the x100p now? What's the new el-cheapo card?
06:49.12neilbags-workP-NuT: yes thats what i'm currently putting it down to
06:49.28P-NuTI'd just do that
06:49.36neilbags-workP-NuT: its more like a screaming than a beeping :(
06:50.31neilbags-workP-NuT: so if asterisk hasn't hung up the line yet, what happens, do you get a busy signal? how long does asterisk take to hang up your line? why the delay?
06:53.21P-NuTbusy ssignal for about 30 secs to 1 min
06:53.44P-NuTthen asterisk resets the line and we're good to go again.
06:54.02P-NuTso where do you get your VOIP hardware from?
06:54.12neilbags-workP-NuT: hmmm .... i don't think i have a delay like that but i've never really tested it well.
06:54.28neilbags-workP-NuT: i'm pretty sure my asterisk hangs up straight away
06:54.49*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no)
06:54.57neilbags-worki got my TDM400P from techtopia (i think) and I buy linksys PAP2s from a computer stop down the street
06:56.08P-NuTgot a link for techtopia?
06:56.51neilbags-workgoogle it
06:56.53P-NuThow hard are the linksys's to setup? and how many lines do they take?
06:57.11neilbags-workthere is 2 FXS ports on each device
06:57.20P-NuTso.. 2 incoming lines?
06:57.40neilbags-work2 sip lines, 2 handsets
06:58.19neilbags-worki havn't changed anything from the defaults on the linksys pap2s except for IP address, sip proxy and dial plan
06:59.24*** join/#asterisk lorinc (n=ang@caracas-0691.adsl.interware.hu)
07:01.01P-NuTcool
07:01.57*** join/#asterisk yxa (n=diablo@58.185.90.101)
07:06.06*** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net)
07:06.35FuriousGeorgeappears to work
07:06.45FuriousGeorgesomeone say my name
07:06.48*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
07:07.14FuriousGeorgeplease?
07:07.29Strom_CFuriousBill
07:07.36Strom_CFuriousJames
07:07.44FuriousGeorge:|
07:07.47hads|homeFuriousTom
07:08.15FuriousGeorge:|
07:08.41FuriousGeorgeseriously, i wanna see what the client does
07:09.41hads|homeFuriousGeorge!
07:09.48FuriousGeorgethanks hads|home
07:09.49*** join/#asterisk prh (n=paul@X80.mjr.org)
07:09.55hads|home:)
07:09.59Strom_CFuriousJorge!
07:10.04FuriousGeorgenow say my name again, but this time while you talk dirty to me
07:10.15Strom_Coh FuriousGeorge, dial me harder
07:10.18*** join/#asterisk h0 (n=h0@ool-44c69453.dyn.optonline.net)
07:10.20FuriousGeorgelol
07:10.28hads|homeha
07:10.36Strom_Chello, FuriousGeorge, i'm here to punch my big tool into your connecting block
07:11.07FuriousGeorgei buiilt gaime with my new gcc 4.1 based toolchain and now it gets all core-droppy
07:11.26Strom_Cit doesnt like you misspelling its name
07:11.54FuriousGeorgethat must be it
07:12.07hads|homegaym
07:13.25*** join/#asterisk [pyro] (i=pyro@tor/regular/bracketed-pyro)
07:13.34liloFuriousGeorge: ember
07:13.35lilo?
07:13.42*** join/#asterisk Sonderblade (n=mah@host-213.131.147.169.addr.tdcsong.se)
07:14.26akkecan anyone recommend me a provider that offers a 'dial-in number' from belgium?
07:14.37*** join/#asterisk jonnysupersonic (n=jonny@dsl-145-56-131.telkomadsl.co.za)
07:14.46FuriousGeorgelilo: arent you famous or something
07:14.50x86akke: that'd be DID
07:14.54x86akke: what's the country code?
07:14.59liloFuriousGeorge: no, I'm not a boot loader :)
07:15.06akkex86: +32
07:15.21FuriousGeorgei was thinking the disney cartoon chacter
07:15.28FuriousGeorgelilo: what you mean ember
07:15.28liloI'm named after a favorite science fiction character, in John Varley's THE OPHIUCHI HOTLINE. I am *not* named after the LInux LOader, and I didn't write it! :)  -- lilo, winter 1993 || Nor am I named after the little girl in the movie, nor the air mattress, nor Last In Last Out. ;) -- lilo, summer 2003
07:15.36liloFuriousGeorge: cartoon humor, ignore it 8)
07:16.08FuriousGeorgefair enough
07:16.10FuriousGeorge:)
07:16.27x86akke: i'm looking now
07:16.33akkex86: thanks
07:16.46x86akke: you just need termination in the US, and origination in belgium?
07:17.23x86for example, people can call you locally from belgium, but you only dial the US
07:17.50akkex86: i live in belgium but I want some extra numbers that come into my asterisk box. I don't need to dial out using that number...
07:18.06x86ah ok
07:18.09x86i can hook you up :)
07:18.15x86akke: talk in private?
07:18.22akkeokay
07:22.08SkramXeeeks, I need to be up forwork in ~4 hours
07:22.09SkramXpeace out
07:26.32*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220)
07:29.27*** join/#asterisk Shoragan (n=shoragan@134.169.175.72)
07:35.00*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
07:41.12*** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net)
07:43.11*** join/#asterisk cypromis (n=michal@voiceworks.pl)
07:43.39*** join/#asterisk astar` (n=astar@ANantes-154-1-7-185.w81-53.abo.wanadoo.fr)
07:51.07*** join/#asterisk QuAtRo[NL] (n=QuAtRo_@dsl-083-247-051-039.solcon.nl)
07:51.34QuAtRo[NL]Hi all
07:51.51QuAtRo[NL]I'm searching for a Astrisk 'command' which rings all phones...
07:52.01mitchelocdial
07:52.22*** join/#asterisk martijn_ (n=martijn@dbcorp.xs4all.nl)
07:52.24QuAtRo[NL]In that case i have to specify all phones, haven't i?
07:52.37QuAtRo[NL]don't i?
07:52.47dpryoSet up a queue with ringall strategy?
07:54.06QuAtRo[NL]I'm developping my own Control Panel voor Asterisk, i want a default rule for all numbers which no dialplan assigned to them..
07:54.20*** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it)
07:54.27FuriousGeorgecall group?
07:54.37*** join/#asterisk MooseAble (n=ismoose@203-59-195-5.dyn.iinet.net.au)
07:54.51QuAtRo[NL]In that case i still have to assign the users to a group...
07:55.13QuAtRo[NL]It should be enough to create an extension for an user...
07:55.16QuAtRo[NL]Is that possible
07:55.17QuAtRo[NL]?
07:55.28FuriousGeorgedynamically?
07:55.46MooseAbleI'm trying to solve an extention registration problem at the moment... i would like to know what "Auto destroying call" means in the asterisk logs first off... can't find documentation anywhere covering this.
07:55.46QuAtRo[NL]Yes
07:56.05FuriousGeorgea meetme and an api script that makes it happen
07:56.31QuAtRo[NL]Ok, thanks!
07:56.48*** join/#asterisk saftsack (n=saftsack@p54A7FCEB.dip.t-dialin.net)
08:00.41motuhow do i make my fxo device detect pickups? right now it just times out even if someone answers
08:11.08af_there is something could I put in the dialplan like autohangup <tme> ?
08:19.08*** join/#asterisk Modcuts (n=bob@82.133.98.155)
08:22.05erwinismwhen i add this line "channel => 1" to my zapata.conf, it gives me error chan_zap.c: Unable to open channel 1: No such device or address
08:22.12erwinismwhat did i miss?
08:24.47*** join/#asterisk qdk (n=qdk@213.237.44.34)
08:24.52motuwhat does the various wink settings in zapata.conf mean?
08:25.13dec;)
08:25.46motuwhere can i find documentation on them
08:26.08*** join/#asterisk kmilitzer (n=km@office-gw.westend.com)
08:26.38kmilitzerGrrr, never book an EasyJet flight ...
08:27.14saftsacklo
08:27.15saftsackl
08:27.37*** join/#asterisk vivek (n=vivek@unaffiliated/tintin)
08:43.18MooseAbleanyone managed to get a adsl router with built in voip to register on their LAN hosted asterisk server? Just having a little trouble here.
08:43.46MooseAblewhilst the asterisk server registers to an external sip server for inbout/outbound calls that is
08:45.35MooseAblerouter keeps reporting "TRYING:No Response" and asterisk keeps giving it "Auto destroying call 'DC4C-D416-8172465-DD2A58ED28D6-0008@203.59.xxx.xxx'" (despite the fact that the routers lan address is "10.1.1.1", 203.59.xxx.xxx is the wan ip address)
09:01.40*** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg)
09:08.17*** join/#asterisk pappu (n=giridhar@60.254.116.238)
09:09.39pappuhi every on .. i need some help on the AGI .. i have written a agi script which does some database check and return some value
09:10.06pappuhow do i capture that in dialplan ?
09:10.58Eric-xx-- Got SIP response 302 "Moved temporarely" back from 194.221.62.207
09:11.03Eric-xxdoes anyone know what this means?
09:12.04pappuhow do i capture the return value of a agi script in a dailplan ?
09:12.34QuAtRo[NL]Are there any some known problems with timeouts, which seem not work?
09:16.55erwinismhello, i have Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface. but when i run ztcfg i got the error: ZT_CHANCONFIG failed on channel 1: No such device or address (6)
09:17.01erwinismany ideas?
09:17.04*** join/#asterisk RoyK (n=roy@80.239.107.70)
09:22.19*** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com)
09:23.38*** join/#asterisk andrebarbosa (n=andrebar@62.48.215.93)
09:27.25andrebarbosaasterisk does g729b encoding\decoding? cant find any info about this..
09:27.36andrebarbosacan make pass-thru, tough
09:28.20RoyKrotfl http://karlsbakk.net/fun/kebab.jpg
09:28.41RoyKandrebarbosa: don't think there's such a thing as a g.729b codec for  *
09:29.10Luke-Jrhow about h264? =p
09:30.29RoyKLuke-Jr: methinks libavcodec integration would be neat
09:30.50andrebarbosaok, thanks RoyK
09:30.51andrebarbosa:)
09:35.23tzafrirerwinism, a "Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface" can be one of quite a few cards. Is it a X100P FXO card?
09:35.50tzafrirpastebin the output of the following:
09:35.55*** join/#asterisk kay2 (n=ashdown@sd-420.dedibox.fr)
09:36.02*** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198)
09:36.08Dr-Linuxhi guys
09:36.19tzafrirlsmod | grep zaptel; cat /etc/zaptel.conf
09:36.22erwinismyes, it works when i "modprobe wcfxo" do i have to do this everytime i start asterisk?
09:36.23Dr-Linuxwhat these Alarms mean? http://pastebin.com/718016
09:36.44tzafrirerwinism, what distro do you use?
09:36.46Dr-Linuxtzafrir: please have a look, i just installed new digium te210p calrds
09:36.55*** join/#asterisk Pj_ (n=pj@fernande.happycoders.org)
09:36.56erwinismtzafrir, centos
09:37.03Dr-Linuxs/calrds/cards
09:37.04Pj_G'morning people
09:37.41tzafrirerwinism,  if you use the standard zaptel.init init script, just set MODULES=wcfxo in /etc/sysconfig/zaptel
09:38.09Eric-xx-- Got SIP response 302 "Moved temporarely" back from 194.221.62.207
09:38.10Eric-xxdoes anyone know what this means?
09:38.13[pyro]evening :)
09:38.24Dr-Linuxtzafrir: any clue for my question?
09:38.30Pj_evening too yeah ;)
09:38.30tzafrirDr-Linux, not a PRI expert, but "red" basically means something like "no cable", right?
09:39.01Dr-Linuxtzafrir: i'm not sure my friend
09:39.11erwinismtzafrir, thank you very much
09:40.40Dr-Linuxtzafrir: but my FXO asterisk server, it shows 1 line for each card
09:40.41Dr-LinuxWildcard TDM400P REV I Board 1           OK         0          0          0
09:40.41Dr-LinuxWildcard TDM400P REV I Board 2           OK         0          0          0
09:41.08Dr-Linuxbut not sure how PRI works in this matter
09:41.45Dr-Linuxtzafrir: would you like to see my zaptel.conf?
09:43.27nextimered for pri is no cable or d-channel down
09:45.47Dr-Linuxnextime: can i show you my zaptel.conf?
09:46.26nextimeDr-Linux sure
09:47.51Dr-Linuxnextime: have a look >> http://pastebin.com/718024
09:48.19Dr-Linuxnextime: i have two TE210P card installed
09:48.23Dr-Linux2 port each
09:49.05erwinismtzafrir, i have another problem, when i call from my voip to pstn, i can hear my own voice echoing at one time only, how can i fix this?
09:49.17RoyKDr-Linux: http://karlsbakk.net/fun/kebab.jpg
09:49.21nextimeDr-Linux use only one time source, so, set the first span to 1, the second to 2 and others to 0
09:49.55nextimeyour have now all 4 spans set to 1 that mean "use this span as a primary timiming source"
09:49.58Dr-Linuxnextime: i didn't understand :S
09:50.27nextimeyou have span=1,1,0,esf,b8zs
09:50.37nextimeand span=2,1,0
09:50.42nextimespan=3,1,0
09:51.07*** join/#asterisk UlbabraB (n=UlbabraB@host-84-222-44-158.cust-adsl.tiscali.it)
09:51.08Dr-Linuxnextime: yes
09:51.13nextimeuse the first one as span=1,1,0, the second one span=2,2,0 the third and the fourth span=X,0,9,
09:51.14Dr-Linuxso how should i use?
09:51.36*** join/#asterisk RoyKa (n=roy@80.239.107.70)
09:51.57nextimethe second parameter is the timing source, you need to have only one time source, and for failover purpose a secondary one
09:52.07Dr-Linuxwht do you mean by span=X,0,9, ?
09:52.14nextimeso, set the span 1 to be the primary time source, the span 2 to secondary, others to 0
09:52.44nextimeDr-Linux : span=3,0,0,esf,b8zf and span=4,0,0...
09:53.37nextimei don't see any other error in your config, are you sure that 24 is the right d-channel?
09:53.57nextime( i don't have t1 pri, i'm in europe )
09:55.41Dr-Linuxnextime: check now please >> http://pastebin.com/718034
09:55.44Dr-Linuxis tht what you said?
09:55.46*** join/#asterisk pjo (n=pjo@212.88.98.114)
09:56.13nextimeDr-Linux : yes, exactly
09:56.16Dr-Linuxnextime: yes 24 is the right d-channel
09:57.43pjohi all, when the docs say 's' isn't a "catchall extension" does it mean doing something like  exten => 12345,1,Dial(IAX2/HOSTNAME/s@default) will not just go through the default context on iax peer HOSTNAME?
09:57.57Pj_I've been testing the BT-100 and I was wondering if you had any recommandation for other phones in that price range... Those looks cheap, act "user-unfriendly" (speaker button) and offer only 10-baseT connectivity
09:57.58nextimeDr-Linux : sorry i was thinking that you have one quad-pri card, as i can see now you have 2 double span cards, so, you can set a primary and a secondary time source for any card, it's meaning that you can set span 1 to 1, span 2 to 2, span 3 to 1, span 4 to 2
09:58.27nextime( not a must, but you can if you prefer )
09:58.42nextimeanyway, your config is correct
09:59.11Dr-Linuxok
09:59.20nextimeif you get a red error, it mean that you cable isn't right connected or your dchannel is down
09:59.51Dr-Linuxnextime: what you prefer, should i change them back to the previous setting?
10:00.39nextimeDr-Linux : no, previus setting was all 4 spans to primary time source, isn't correct. Only one primary source for card, one primary and one secondary for card
10:01.45Dr-Linuxokey great i understand
10:01.59Dr-Linuxnextime: now would you like to see my zapata.conf?
10:02.20MooseAbleanyone know what "Jun 19 05:59:06 DEBUG[1388]: Auto destroying call 'DC4C-D416-25900427-67DA7E70EEDE-0019@203.59.xxx.xxx'" means?
10:02.25nextimeDr-Linux it's not related with red alarm, but yes, paste it
10:02.55Dr-Linuxnextime: it's my new setup, so just wanna verify..
10:03.04nextimeDr-Linux : ok
10:03.50Dr-Linuxhttp://pastebin.com/718048
10:04.06Dr-Linuxnextime: looks like there are few errors, please verify
10:06.22Dr-Linuxwhat should be here:
10:06.23Dr-Linux#
10:06.24Dr-Linuxpridialplan=unknown
10:06.24Dr-Linux#
10:06.24Dr-Linuxprilocaldialplan=unknown
10:08.06nextimerxgain and txgain are obsolete if i remember right, and you need to specify context ( one for all groups, or one for group, as you prefer ), for the latest group you can use channel => 49-71,73-95, i suggest to add overlapdial=yes, pridialplan and prilocaldialplan are ok to unknown, or you can set national / local
10:09.08*** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net)
10:09.11Bert-hello there
10:09.34Dr-Linuxnextime: so should i remeve rxgain/txgain?
10:09.47Dr-Linuxand what it does? overlapdial=yes
10:10.32Bert-Well When I call someone through my *, when the called party hangup, Asterisk don't, and maintains the connection between my sipphone and himself. Is there a way to make * asterisk detecting disconnection plz ?
10:10.57Bert-timeout issue ?
10:10.58nextimeDr-Linux : overlapdial is for calls come from old analogic phone source, to catch the right extension on your primary, if you don't specify overlapdial, calls come from old analog lines come all to your shirt number
10:11.06nextimes/shirt/short
10:11.41Dr-Linuxok great
10:12.00Dr-Linuxwhat about rxgain/txgain, i'm using them in other servers with FXO's
10:12.13Dr-Linuxdon't i need them with PRI setup?
10:12.23QuAtRo[NL]nextime saw too many soccer
10:12.25QuAtRo[NL]:P
10:13.03nextimeDr-Linux : no, you don't need them
10:14.04Dr-Linuxnextime: what if i'd need increase or decrease rxgain/txgain ... then what i will do? :S
10:14.17Dr-Linuxnextime: or i won't need that with PRI?
10:14.52nextimeDr-Linux : in my opinion you don't need them on a isdn channel, pri or bri is indifferent
10:14.55Bert-hmm noone can help me ?
10:15.18nextimeanyway, you can use it if you think, no matter about that
10:15.19Bert-Is a way to detect disconnection on a SIP trunk ?
10:15.32Dr-Linuxi see
10:16.15Dr-Linuxnextime: actually i'm not sure if "cables" are plugged to my cards or not, that's why i have doubt about RED alarams
10:16.28Dr-Linuxnextime: i'm in pakistan and my servers are in the US datacenter
10:17.04Dr-Linuxnextime: i just wanna know, what possible things can make it "RED Alarm"
10:17.19Dr-Linux?
10:18.16nextimeDr-Linux : a bad cable, a too long cable ( if it's too long, you can set the third parameter of span= to something different to 0 ), or a dchannel down from carrier
10:18.31nextimeor, of course, a disconnected cable
10:19.30nextimeor a wrong dchannel number, but as you say it isn't your case
10:20.20Dr-Linuxnextime: i see
10:20.28nextimeor a wrong framing/coding combination, but i don't think that is your case, cause your first span is ok
10:20.46Bert-<PROTECTED>
10:20.58Dr-Linuxbut didn't understand the logic if the cable is too long i can set third parameter of span= to somehting different than 0 :S
10:21.04Bert-it is my softphone which doesn't detect disconnect ... :)
10:21.43Dr-Linuxnextime: well, my first Span is "OK" but not others
10:21.54Dr-Linuxnextime: so what you think, what could be the reason
10:22.13Dr-Linuxas you saw, my zap config is fine. that what's new :S
10:23.14Dr-Linuxnextime: if the cable is not plugged it should show the 1st span also "RED" but in my case 1 span is OK but 2nd is RED :S
10:23.23nextimeDr-Linux : if your cable is very long, you can receive a "low signal" on that. So, using a different set on the third parameter you say to zaptel to be more "sensitive", you must use 1 if your cable is from 133 to 266 feet, 2 from 267 to 399 and so on
10:24.26nextimeDr-Linux : you will have 4 pri with 4 different cable, if your first cable is plugged and the others 3 cable aren't connected, you get ok on the first span and red on the others 3 spans
10:24.54nextimeany span have a different cable
10:25.06Dr-Linuxnextime: i understand
10:25.07Mw3hm, i hear double ringing on my sip device when dialing out on a PRI. what can the cause of this?
10:25.28Dr-Linuxbut really the long/short cables matter is very good to know :S
10:25.50nextimeMw3 : maybe are you answer before the dial command to call over pri?
10:26.23MooseAbleanyone know what "Jun 19 05:59:06 DEBUG[1388]: Auto destroying call 'DC4C-D416-25900427-67DA7E70EEDE-0019@203.59.xxx.xxx'" means?
10:26.39Pj_that resistence is futile
10:26.56nextimewe will be assimilated?
10:27.22MooseAbleyour head a-splode
10:28.13pjocan anyone recomend a good IAX2 client for macos (no not * .. that's a server :-D )
10:28.21pjoerr OS X
10:28.27nextimepjo : wengophone
10:28.33pjonextime: thx.
10:28.40nextimeoh
10:28.44nextimeyou say IAX
10:28.49nextimesorry wengo is sip
10:29.02pjoahh.. yes. i meant iax
10:29.07nextimeso, no idea for iax, sorry
10:29.33pjok.
10:30.18Mw3nextime: no, i dont. my outgoing dialplan is just a SetAMAFlags(billing) and a Dial(ZAP/g1/${EXTEN},60)
10:30.29Eric-xxhow do i change my User-Agent: Asterisk PBX ?
10:32.53nextimeok, lunch time, bye all
10:33.48*** join/#asterisk Delvar (n=irc@host-83-146-53-46.bulldogdsl.com)
10:36.31*** join/#asterisk yacyac (n=yac@203.145.159.44)
10:36.48andrebarbosaEric-xx, on sip.conf
10:36.58andrebarbosaunder [general]
10:37.18andrebarbosauseragent=XXXXXX
10:37.19andrebarbosa;)
10:37.44pappuhi how do we enable echo canlellation on zaptel card ?
10:42.45pjopappu: echocancel=yes in zapata.conf
10:43.05erwinismtzafrir, i have another problem, when i call from my voip to pstn, i can hear my own voice echoing at one time only, how can i fix this?
10:43.19erwinismanyone can help me?
10:44.10Eric-xxthanks andrebarbosa
10:46.23tzafrirerwinism, for starters, set opermode (the parameter to the driver) appropriate to your country
10:46.41tzafrironce that is done, read about echo cancelling
10:47.10tzafrire.g: echocancell=yes in zapata.conf
10:47.50erwinismi already have that echocancell=yes, tzafrir , i am in philippines right now
10:50.21pjoI have a voicetronix openswitch 12 card.. setting callerid=yes in vpb.conf results in a lot of static noises on the FXS ports which doesn't go away till I reload the driver module. Any ideas on how to fix that?
10:52.10erwinismtzafrir, how can i set opermode?
10:55.48*** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net)
10:56.40erwinismby the way, i use the clode x100p card
10:57.33*** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com)
11:01.25tzafririt is a module parameter. You can set it manually when loading the module or through modprobe.conf
11:03.20erwinismok thank you :)
11:07.05*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
11:16.27*** join/#asterisk [pyro] (i=pyro@tor/regular/bracketed-pyro)
11:22.16*** join/#asterisk clive- (n=pirch@dsl-165-165-00.telkomadsl.co.za)
11:22.53*** join/#asterisk beyond (n=beyond@200.192.160.100)
11:28.21*** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no)
11:29.22zwelchpuzzled: yes, sort of (in answer to your question in #minisip)
11:30.22puzzledzwelch: great, let me get back there
11:39.50*** join/#asterisk Falle (n=falle@213.141.80.88)
11:45.34*** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka)
11:50.33*** join/#asterisk oej (n=oej@apollo.webway.se)
11:54.51MooseAblewell, looks like nobody knows teh answer. seems I'll never get this router to register with asterisk o_O
11:56.56*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)
11:59.10*** join/#asterisk fbucher (n=krp@hub.xunix.org)
11:59.12fbucherhi
12:01.15*** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net)
12:03.08Mavvieanybody here familair with the Outlook Dialer from Third lane?
12:08.23*** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br)
12:09.42*** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net)
12:11.51*** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198)
12:12.12Dr-Linuxanybody using spa-2100 or 2000?
12:13.04Dr-Linux[TK]D-Fender: can i access spa-2100 device from the local network?
12:17.10*** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com)
12:19.01[TK]D-FenderDr-Linux : Should be able to (for admin), bup SIP is only functional on the WAN interface which I why I would never buy it.
12:19.09*** part/#asterisk Ludo_ (n=Ludo@obelix.zoxx.net)
12:20.16Dr-Linux[TK]D-Fender: i can access spa-2100 device when i plug a cable in PC port of device, wheather device is on network, but i can't access it from the local network.
12:22.42*** join/#asterisk tomtom_ (n=root@83.217.70.166)
12:22.50tomtom_Hi
12:22.54*** join/#asterisk Webboarder (n=rubentul@dsl-083-247-051-039.solcon.nl)
12:22.55littleRalphieanyone have an x100p o sell?
12:23.14tomtom_Anyone can help me with DID's in the Phillipines? pvt msg pls
12:23.34[TK]D-FenderDr-Linux : maybe you didn't enable outside access to it...
12:26.43WebboarderHello, I have a problem with queue's. I am logged in as an agent at queue 'support'. But when i log off as an agent, my phone still rings when someone is in the queue 'support'. How can i solf this problem?
12:27.13Dr-Linux[TK]D-Fender: i see, i'd need to see from where can i enalble this access :S
12:28.00*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
12:30.13clive-littleralph ebay is your friend
12:31.55[TK]D-FenderWebboarder : Pastebin your queues.conf, agents.conf, and extensions.conf
12:32.42[TK]D-Fender~pb
12:33.00jbotfrom memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/
12:34.05*** join/#asterisk coppice (n=chatzill@18.162.17.210.dyn.pacific.net.hk)
12:34.29*** join/#asterisk Mavvie (n=edwin@252-131-222-203.static.techex.net.au)
12:39.25Webboarder[TK]D-Fender : http://pastebin.com/718264
12:40.14Dr-Linux[TK]D-Fender: please have a look and let me know, why 3 span's alarms are "RED" http://pastebin.com/718048
12:40.36Dr-Linuxhttp://pastebin.com/718268 << here
12:42.15*** join/#asterisk speedwagon (n=FreePBX5@70.46.87.158)
12:43.09iCEBrkrW3rd.
12:46.46*** join/#asterisk smurf (n=smurf@debian/developer/smurf)
12:46.51[TK]D-FenderWebboarder : How are you "logging out"?
12:48.05*** join/#asterisk Darthclue (n=Darthclu@adsl-69-152-233-213.dsl.snantx.swbell.net)
12:48.27Webboarder[TK]D-Fender : I am calling 1000, then pressing 1, enterring my extension number followed by '#'. And finally when asked to enter a new extension number I only enter a '#'.
12:48.50Webboardersorry i'm pressing 2 instead of 1
12:50.50[TK]D-FenderWebboarder : Hmmm...
12:51.08[TK]D-FenderDr-Linux : Have you tried a loopback test?
12:51.45Dr-Linux[TK]D-Fender: sir, how can i do that?
12:53.17iCEBrkr[TK]D-Fender: You're like Asterisk Tech support 24/7 :P
12:53.27[pyro]lol
12:53.35*** join/#asterisk hongtien (n=hongtien@203.162.100.12)
12:53.45Dr-Linuxhe glads to help :)
12:54.14hongtienHi
12:54.57[TK]D-FenderDr-Linux : Go read up on how to make a loopback connector dongle.
12:55.33hongtienHi all
12:56.03hongtienI want to use Asterisk with Voice Gateway
12:56.11[TK]D-FenderDr-Linux : May want to verify your wiring as well.  When in doubt plug your other lines onto port 1 and see if they come up.
12:56.11Dr-Linux[TK]D-Fender: ok, and your thoughs about my configs?
12:56.30Dr-Linuxok
12:57.06[TK]D-FenderDr-Linux : Not sure on your 2nd cards timing, but go verify the PRI's first
12:57.20hongtienie: Asterisk didn't act Media Gateway
12:58.09hongtienHow many concurent call can make on that
12:58.11hongtien?
12:59.46*** part/#asterisk hongtien (n=hongtien@203.162.100.12)
13:00.37*** join/#asterisk hongtien (n=hongtien@203.162.100.12)
13:00.50*** part/#asterisk hongtien (n=hongtien@203.162.100.12)
13:02.45Dr-Linux[TK]D-Fender: what's possible things that cause of "RED" alarm?
13:03.31[TK]D-FenderDr-Linux : Nothing connected to the port.  Use your imagination and test what I jsut told you to.
13:04.07Dr-Linux[TK]D-Fender: ok thanks
13:04.10QuAtRo[NL]Problem of Webboarder is solved
13:05.57_4d4m_i've just been reading around about options for LCR on *.  I'm unsure what I need to use.. Should I run MySQL or execute it through AGI? 4 points of termination max, and dont envisage altering this structure for a while.
13:06.15_4d4m_any assistance appreciated
13:08.00speedwagon_4d4m_, I would use your dialing rules, maybe use something like dialparties.agi But your dialing rules can do this for you.
13:08.03*** join/#asterisk acrg (n=aragon@decoder.geek.sh)
13:08.06acrghi
13:08.16speedwagonmorning everyone
13:08.22inv_Arp~seen ariel
13:08.42jbotariel <n=kvirc@host253.200-82-113.telecom.net.ar> was last seen on IRC in channel #debian, 47d 11h 39m 37s ago, saying: 'Does anyone know how to capture my  current X session to any video format ?'.
13:08.52acrgcan anyone confirm that sending/receiving faxes works with asterisk and a sangoma pri card ?
13:09.15speedwagoninv_Arp, how are you doing?
13:09.16acrgfrom spandsp
13:09.43inv_Arpspeedwagon: ello...
13:09.45[TK]D-Fenderacrg : SpanDSP seems to.  Fax machines behind a channel-bank are "OK"
13:10.09speedwagoninv_Arp, also the nick is ariel_, or me speedwagon... or abatista one and the same..
13:10.18inv_Arpahh ok... lol
13:10.40*** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane)
13:10.57acrgFender thanks
13:10.59inv_Arpbeen lookin through my career builder and I thought I seen something you might like...
13:11.05speedwagonacrg, it should I have a customer using a sagoma pri card.
13:11.18speedwagoninv_Arp, OK
13:11.38acrgspeedwagon and they send faxes over it with something like asterfax ?
13:12.23*** join/#asterisk Dovid (n=none@barak.cellcom.co.il)
13:12.35DovidCan anyone help me with a meetme issue ?
13:13.11*** join/#asterisk Ariel_ (n=Ariel@70.46.87.158)
13:13.29*** join/#asterisk speedwagon (n=Ariel@70.46.87.158)
13:13.41*** part/#asterisk Ariel_ (n=Ariel@70.46.87.158)
13:13.55Dovid?
13:14.32*** join/#asterisk Ariel_ (n=Ariel@70.46.87.158)
13:15.10speedwagoninv_Arp, sorry network issues.
13:15.12[TK]D-Fender~suggestions
13:15.16jbot[suggestions] 1) Don't ask to ask. Just say your problem, 2) Don't repeat until 5 mins after, 3) Read and re-read the docs first, then admit it if you REALLY don't understand. You're wasting your time and ours if you haven't at least tried. 4) If your problem ain't solved, come back in 12 hrs or 24 hrs later. We're very international. 5) Be polite and patient.  6) ...
13:15.40DovidWhy does meetme dump the call if the extension enterd is in valid (I.e. Exten _5xxx,1,Meetme)
13:16.18[TK]D-FenderDovid : Pastebin the code that isn't working, and evidence of its failure in CLI.
13:16.51[TK]D-FenderDovid : And if its dialplan related, the entire context and linked contexts.
13:17.56*** join/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com)
13:18.12DovidK, on dial up. Give me a sec :)
13:20.27Dovidhttp://pastebin.com/718381
13:22.14*** part/#asterisk acrg (n=aragon@decoder.geek.sh)
13:22.34[TK]D-FenderDovid : Sorry, can't help with realtime...
13:22.43Dovidk
13:23.01DovidCan u take a guess as to y it dumps the call ?
13:24.07*** join/#asterisk Faithful (n=Faithful@202.6.145.116)
13:25.28Dovid${EXTEN}:1 will subtract the first number from the exten ?
13:25.35tzangerno
13:25.38tzanger${EXTEN:1} will
13:26.31*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1)
13:26.42*** join/#asterisk Samoied (n=Samoied@200.187.153.130)
13:26.59Dovidthx
13:48.52*** join/#asterisk Spy000007 (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net)
13:51.59*** join/#asterisk mercestes (n=merceste@69.15.174.114)
13:52.40*** join/#asterisk Hmmhesays (i=negative@66.173.103.110)
13:56.16*** join/#asterisk kshumard_ (n=kshumard@gateway.digium.com)
13:56.38*** join/#asterisk Katty (n=angela@64.82.232.54)
13:57.00Kattymorning
13:59.30*** join/#asterisk Pointy (n=chowell@brain.xilogix.net)
13:59.35*** join/#asterisk bkervaski (n=bkervask@adsl-072-149-159-016.sip.bhm.bellsouth.net)
13:59.55bkervaskiHi all.  Is it possible to have a voicemail delivered to multiple mailboxes without copying the file manually with a script?  i.e., in extensions.conf?
14:01.11Pointyanyone have success using the follow-me option in * version 1.2.9.1?  Having an issue where the timeout seems to be ignored.
14:01.13*** part/#asterisk kmilitzer (n=km@office-gw.westend.com)
14:01.14*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
14:03.28HmmhesaysMorning
14:03.44*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)
14:03.47PakiPenguinhello everyone
14:04.01*** join/#asterisk Cyon (n=cyon@216.179.31.170)
14:05.34[TK]D-FenderKatty : Mew.
14:05.58[TK]D-Fenderbkervaski : Yes, its all part of the Voicemail application
14:06.16[TK]D-Fenderbkervaski : go read the instructions again
14:06.42*** join/#asterisk JoseBravo (i=JoseBrav@200.119.32.47)
14:07.21HmmhesaysWe saw a kickass live band this weekend
14:07.28*** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.141.Dial1.SanJose1.Level3.net)
14:08.18DovidUsing meetme in real time with _5XXX if invalid room is enterd meetme dups the call. Anyone know y ?
14:08.44JoseBravoI have a sip peer for outgoing calls that use g729 codec, and my sip clients use x-lite that don't have g729 codec. Then when I tried to do a call, I get this message: channel.c:2693 ast_channel_make_compatible: No path to translate from SIP/dominet-f844(256) to SIP/70103-6092(4). Where dominet is my provider and 70103 is the client.
14:08.54Ahrimaneshttp://bash.org/?654797
14:08.59[TK]D-FenderDovid : Maybe because the room is invalid.. you should try to avoide passing applications garbage values you know....
14:09.20DovidWell the room is invalid
14:09.22[TK]D-FenderJoseBravo : Did you pay for licenses at Digium.com?
14:09.54Katty[TK]D-Fender: mew.
14:10.06vader--is it normal to see asterisk say  B-channel 0/23 successfully restarted on span 1
14:10.06DovidClient enters *5000 to start a room. Problem is if a client of his calls and enters the wrong room he gets dumped. I am tryin to send em back to the main menu instead
14:10.10vader--for all the channels
14:10.13vader--every so often
14:10.14Katty[TK]D-Fender: my company is moving.
14:10.18Katty[TK]D-Fender: and i get my own office!
14:10.21Katty[TK]D-Fender: with a door!
14:10.24filevader--: that's normal
14:10.24JoseBravo[TK]D-Fender I need to pay licences?
14:10.29drraygood morning fender
14:10.29Dovidyay
14:10.35Dovidyes
14:10.38*** join/#asterisk stephane_ (n=stephane@merlin.cabale.net)
14:10.41DovidTo digium $10.00 per channel
14:10.43Kattyfile: i get my own office :>>>>
14:10.55fileKatty: yay!!!
14:11.00Kattyi know.
14:11.06Kattyi feel all speshul.
14:11.15fileand now... I run away!
14:11.23[TK]D-FenderKatty : Cool...
14:11.53JoseBravoI need to pay $10 per use sip peer to aoutgoing calls?
14:11.54Dovid[TK]D-Fender: If room is invalid it automaticly dumps the call ? Can I set it to go to the next pri. In the dial plan ?
14:12.14Dovidyes
14:12.17[TK]D-FenderJoseBravo : G.729 is a patented codec and * will not translate it without paying a licensing fee.
14:12.20Kattyfile: byebye
14:12.25Kattyfile: CALL ME
14:12.30DovidIt works 2 ways. Pretty much one channel per concurent call 2 eays
14:12.44*** join/#asterisk jeremib (n=netnameu@c-71-203-209-162.hsd1.tn.comcast.net)
14:13.10Dovid[TK]D-Fender: did u get my last ?
14:14.05znoGdid anyone else get spammed by LisaZhang?
14:14.12*** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198)
14:14.13znoGfrom Telecomoutsourcing in China or something
14:14.16Dr-LinuxJun 19 07:13:00 NOTICE[10627]: app_dial.c:1040 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)
14:14.16Dr-Linux<PROTECTED>
14:14.18DovidFor which list ?
14:14.25bkervaskiThanks, Fender.
14:14.28bkervaskiHelpful as always.
14:14.29Dr-Linuxwhy can't i dialout via PRI line? :S
14:15.13jeremibhow can I troubleshoot my iax2 trunks not being listed in "iax2 show registry"?  I have three different register= lines in my iax config file connected to 3 different servers.  When i do iax2 reload it shows it parsed the file, but if i do iax2 show regstry nothing is listed
14:15.18*** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr)
14:15.28jeremiband obviously my iax2 connections aren't being made
14:15.40DovidIax2 debug
14:15.48*** join/#asterisk feld (n=feld@66-188-108-178.static.mdsn.wi.charter.com)
14:15.55[TK]D-FenderDovid : nO IDEA.
14:15.56filejeremib: or make sure they're in the right place in iax.conf
14:16.14DovidThanks tk
14:16.23[TK]D-FenderDr-Linux : wHATS THE LINE STATUS?
14:16.40JoseBravoDovid no other solution for not pay per channels?
14:16.50[TK]D-FenderJoseBravo : Stop using G.729
14:16.56jeremibfile... you're a genious :)
14:17.01Dovidnope
14:17.13DovidUnless u just pass thru
14:17.18jeremibi'm using freepbx, and it was loading my _custom.conf file before the _additional.con file, which i guess was messing it up
14:17.23DovidTk: You are using caps LOCK
14:17.24[TK]D-FenderDovid : Which he can't...
14:17.29jeremibswitched it around and it worked
14:17.31jeremibthanks!!
14:17.33JoseBravo[TK]D-Fender but its depends for my provider.
14:17.43[TK]D-FenderDovid :WhAt ArE YoU TaLkInG AbOuT?!?!
14:17.51Dovidhehe
14:17.55PakiPenguinDr-Linux, got a pri ?
14:18.06file[TK]D-Fender: Hi TkDeFeNdEr HoW aRe YoU?
14:18.19[TK]D-FenderJoseBravo : They don't support G.711?  I'd be surprised.
14:18.30znoGcan anyone get to www.freshmeat.net? it gets content from falkag.net which doesn't seem to respond, quickly
14:18.33bkervaskiHey tk: give me a hint on the voicemail->multiple mailbox delivery.. coming up short.. probably not sure what to search for.. I've looked through voicemail.conf, no dice.. (* 1.2.x)
14:19.08[TK]D-Fenderfile : Feeling colourful
14:19.14file[TK]D-Fender: uh oh
14:19.37[TK]D-Fenderbkervaski : "
14:19.45[TK]D-Fenderbkervaski : "show application voicemail"
14:19.46Dr-LinuxPakiPenguin: yes
14:19.56bkervaskiThanks.
14:19.57PakiPenguinDr-Linux, ptcl ? cool ? what for ?
14:20.50bkervaskiSometimes, it's just too easy to fathom... Thanks, TK
14:21.06[TK]D-Fenderbkervaski : SCARY isn't it :)
14:21.13Dr-LinuxPakiPenguin: not here, i got PRI in USA
14:21.16Dr-LinuxPTCL sux
14:21.28PakiPenguinyup ;)
14:21.32JoseBravo[TK]D-Fender g723 will help to me?
14:21.33PakiPenguinit does!
14:22.06*** part/#asterisk jeremib (n=netnameu@c-71-203-209-162.hsd1.tn.comcast.net)
14:22.09Dr-LinuxPakiPenguin: PakiPenguin look here what PTCL did > www.syednetworks.com
14:23.26*** part/#asterisk clive- (n=pirch@dsl-165-165-00.telkomadsl.co.za)
14:25.35JoseBravo[TK]D-Fender g723 will help to me?
14:26.25*** join/#asterisk copland (n=stonecol@209.216.65.10)
14:26.42*** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler)
14:26.43coplandis there a way to have a h323 ext  in asterisk
14:27.37JoseBravoI need pay licences for use g723?
14:27.52fileJoseBravo: Asterisk currently can't transcode G723.1
14:28.32RoyKJoseBravo: I beleive a g.723.1 license starts at about $100k or so
14:28.41*** join/#asterisk ruza (n=ruza@81.0.238.58)
14:29.48JoseBravoMy provider says that g711 use much bandwidth
14:31.10JoseBravoThats right?
14:31.13fileyes
14:31.30*** join/#asterisk marv[work] (n=timr@64.89.118.139)
14:32.57HmmhesaysSomeone needs to read some more
14:33.18[TK]D-FenderJoseBravo : No, even worse
14:33.49[TK]D-FenderJoseBravo : So go pay for a lincense or two from Digium.
14:34.12[TK]D-FenderHmmhesays : EVERYBODY needs to read more.  Just most more than some ;)
14:34.28Hmmhesaystrue
14:34.28*** join/#asterisk harpermood (n=harpermo@24-180-0-138.static.snlo.ca.charter.com)
14:34.33file[TK]D-Fender: Professor [TK]D-Fender, how much do I need to read?!?
14:36.00harpermoodCan anyone help me with a connection problem.... I have a digium TE110P and am attempting to connect with a Channelized T1 with 6 channels using E&M signalling with wink
14:36.12fileoooooooh
14:36.59*** join/#asterisk gaupe (i=rmo@slogen.sunnmore.net)
14:37.36JoseBravo[TK]D-Fender ok, im purchasing
14:39.36*** join/#asterisk ToyMan (n=stuq@74-32-6-50.dsl1.mdl.ny.frontiernet.net)
14:40.10*** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.141.Dial1.SanJose1.Level3.net)
14:40.16*** part/#asterisk harpermood (n=harpermo@24-180-0-138.static.snlo.ca.charter.com)
14:41.07Dr-Linuxanybody have a look >> http://pastebin.com/718487
14:41.10Dr-Linuxnot sure why i'm getting this
14:41.39Hmmhesaysuclinux is not fun
14:41.55[TK]D-FenderDr-Linux : And is the port still red?
14:42.01*** join/#asterisk harpermood (n=harpermo@24-180-0-138.static.snlo.ca.charter.com)
14:42.02SplasPoodhrm, if i wanted to test what will become 1.4, whats the current best route?
14:42.23Dr-Linux[TK]D-Fender: i'm getting this from "OK" port
14:42.24[TK]D-FenderSplasPood : Install chan_fluxcapacitor
14:42.38SplasPood[TK]D-Fender: :P  You know what I meant ;)
14:43.08[TK]D-FenderDr-Linux : I believe 34 is a notice that the remote end is busy
14:43.15Dr-Linux[TK]D-Fender: for not i forgot other 3 spans, but if it should dialout via 1 span, but it gives me this error
14:43.24*** part/#asterisk harpermood (n=harpermo@24-180-0-138.static.snlo.ca.charter.com)
14:43.42[TK]D-FenderSplasPood : And you know what I meant, thus is symmetry attained!
14:44.56fileSplasPood: wait until the beta is out
14:45.24*** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net)
14:45.34[TK]D-Fenderfile : Whats the ETA looking like for it these days? (R, not RC)
14:46.19Dr-Linux[TK]D-Fender: by dialing any number, i get the same "Notice"
14:46.24Dr-LinuxJun 19 07:37:52 NOTICE[10694]: app_dial.c:1040 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)
14:46.24Dr-Linux<PROTECTED>
14:46.55[TK]D-FenderDr-Linux : Maybe you should pastebin the relevent parts of you config, and not jsut the 1 line error....
14:47.14Katty[TK]D-Fender: corporate made a terrible, terrible decision today.
14:47.24[TK]D-FenderKatty : rut roh....
14:47.27Katty[TK]D-Fender: they want me to take down the asterisk box and put up a vonexus box.
14:47.33Katty[TK]D-Fender: because microsoft is /clearly/ better.
14:47.42Katty[TK]D-Fender: save me :<
14:48.23file[TK]D-Fender: this year!
14:48.25tzangerKatty: put it on a 10mbit hub with a p2p client
14:48.35pjolol
14:49.17*** join/#asterisk harpermood (n=harpermo@66-215-122-11.dhcp.atsc.ca.charter.com)
14:49.32[TK]D-FenderKatty : And why would they want such a thing?
14:50.03Dr-Linux[TK]D-Fender: i have nothing more then this >> http://pastebin.com/718500
14:50.13harpermoodI need to setup my phone system for my company... got my TE110P card on Saturday, but can't seem to dial out or in
14:50.17Dr-Linuxthis time i pasted while trying different number
14:50.46[TK]D-FenderDr-Linux : perhaps you aren't awake yet... I said PASTE YOUR CONFIG FOR IT
14:50.53[TK]D-Fender</subtle>
14:51.03Dandan[TK]D-Fender u r being sooo nice today :)
14:51.04Dr-Linuxok
14:51.08Dandanis it your birthday? :D
14:51.14Katty[TK]D-Fender: they seem to think that since i'm an mcp, working on my mcse, and because sell microsoft products.....that we should be using them.
14:51.33Dandanthere is nothing wrong with mcse and asterisk
14:51.46*** join/#asterisk anonymouz666 (i=anonymou@200.218.196.5)
14:51.52fileKatty: how works your Asterisk install right now anyway?
14:52.03Kattyfile: it works.
14:52.08[TK]D-FenderKatty : and MS hosts on Linux servers... go figure... your admins are looking to waste money for nothing... what you REALLY need to do it ditch that hybrid channel bank BS you've got and just go PRI.
14:52.08Kattyfile: it's a neat as a button too
14:52.33Katty[TK]D-Fender: corporate doesn't share my views.
14:52.35Dandan[TK]D-Fender: yeah, and their experts.microsoft.fr has been hacked into and defaced :)
14:52.36Dr-Linux[TK]D-Fender: here >> http://pastebin.com/718506
14:52.36*** join/#asterisk wunderkin (n=wunderki@69.26.192.234)
14:52.46Katty[TK]D-Fender: no surprise there tho (=
14:52.47Dandanthat's what you get for hosting stuff on windows... :)
14:53.01Dandanover the weekend :)
14:53.22harpermoodIs there anyone that can help me with my TE110P problem.. I am really stuck.
14:53.39Dandanharpermood: as I am with sangoma and zapata.conf :)
14:53.52Dandanand even asterisk-users can't help :/
14:55.25[TK]D-FenderDr-Linux : that isn't your config... thats just the dialpland and I can SEE what its dialing... I want to see what its USING.
14:55.26Katty[TK]D-Fender: in other mews, my banking is getting better :>
14:55.53Kattyfile: wrong banking, deary.
14:56.01Dr-Linux[TK]D-Fender: you wanna see  zap configs?
14:56.10*** join/#asterisk salviadud (n=ralfalfa@201.145.29.99)
14:56.13[TK]D-FenderKatty : Laws of Star Trek : "Bank left, lurch right"
14:56.27[TK]D-FenderDr-Linux : ..... perhaps some coffee is in order....
14:56.36Katty[TK]D-Fender: lurch?
14:56.51Dr-Linux[TK]D-Fender: http://pastebin.com/718048
14:57.04harpermoodI am on the phone now with Digium support...
14:57.13harpermoodThey are having phone system troubles HAH ;)
14:57.30[TK]D-FenderKatty : a jerking lean.
14:58.26fileharpermood: yeah... we are...
14:58.28[TK]D-FenderDr-Linux : Dunno... verify that your lines aren't all in use, try calling those #'s from other lines to confirm they're OK...
14:58.52Dr-Linux[TK]D-Fender: okey thanks
14:59.33harpermoodPhone system troubles happen.. it is just ironic, and a bit embarrassing,  I would guess !!
14:59.56*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
15:00.34DandanI am looking for a GOOD knowledge of PRI for hire for ONE (SMALL?) glitch?
15:00.38harpermoodso, file, if I am in the queue, listenening to some fantastic music, how long might I expect to stay there?
15:00.39Dandanblah
15:00.44Dandan*expert of course
15:00.54fileharpermood: less then a year
15:02.12harpermoodgot someone!!
15:02.12harpermoodit has been a short year :)
15:02.32JoseBravoI have a problem registering my g729 codec, because my interface is not eth0. Then I can't register my codec?
15:02.58fileJoseBravo: are you having an actual problem or are you just asking
15:03.26Katty[TK]D-Fender: what's that?
15:03.31Katty[TK]D-Fender: that does not parse.
15:03.32*** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au)
15:04.38[TK]D-FenderKatty : Picture getting cross-checked by an NFL quarterback, then you'll understand "lurch"....
15:04.55[TK]D-FenderKatty : Or try http://dictionary.reference.com/browse/lurch
15:07.02JoseBravofile I have an actual problem.
15:07.13JoseBravofile Unable to determine hostid.  You must have at least one ethernet card
15:07.39[TK]D-FenderJoseBravo : You don't have a NIC in that box?
15:07.50filewhat OS and what's the interface name?
15:08.34*** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1)
15:08.59JoseBravoThe OS is Fedora Core 4. But I patched the kernel with OpenVZ. Then Its a VPS. The interface name is venet0:0
15:09.10fileinteresting
15:10.00Ahrimaneshm maybe the : that confuses it?
15:11.11JoseBravoThe alias of the first
15:11.12JoseBravo<PROTECTED>
15:11.12JoseBravo<PROTECTED>
15:11.12JoseBravo<PROTECTED>
15:11.29JoseBravoSorry for the flood, it wasn't my intension.
15:11.40fileyeah I'm just trying to think of how you can make it work
15:11.56JoseBravoThanks file
15:12.10*** part/#asterisk harpermood (n=harpermo@66-215-122-11.dhcp.atsc.ca.charter.com)
15:12.18[TK]D-Fenderfile : Alias it to ETH0 ?
15:12.47Ahrimanesor look at the freebsd reg util? it doesnt look for eth0..
15:12.59JoseBravoI can't create new virtual interfaces into VPS
15:13.10fileregistration utility is the same, just compiled for FreeBSD
15:14.07Ahrimaneshm but eth0 is not present on freebsd...
15:14.14fileit uses a different method to grab the info
15:14.35filereally think the API calls would be the same between Linux and BSD? :D
15:16.13lunkman asterisk is so much faster without all that A@H overhead
15:17.00*** join/#asterisk feld (n=feld@12.148.212.157)
15:17.09mitcheloclunk: go post that on the tb forums ;)
15:17.15mitchelocevangelize!
15:17.22*** join/#asterisk mzeltner (n=eaon@62.96.102.155)
15:17.24*** join/#asterisk akke (n=akke@85.158.211.235)
15:17.25lunktb?
15:17.43mitchelocaah=tb
15:18.04filemzeltner: hope you're having a good time
15:18.08lunkoh
15:18.08mitchelocmzeltner: that's just mean!
15:18.14lunkthat new project with the stupid name?
15:18.17akkeanyone here can offer flat fee SIP/IAX dial-out to belgium landlines?
15:18.18mitchelocyes
15:18.23lunkthe name is the reason i stopped using it
15:18.31mitchelocgood for you
15:18.34lunkcan't go into a client's office with a 'trixbox'
15:18.45*** part/#asterisk cods (n=cods@tuxee.net)
15:18.53mitchelocwell.... there are worse names forprojects
15:19.07lunkMs Bob?
15:19.17mzeltnerfile: Yeah, it's nice, would've expected more people though - maybe tomorrow
15:19.19mitcheloclike yahoo or google
15:19.29[TK]D-Fendermitcheloc : Another sign of * GUI underacheivement ;)
15:19.40filemzeltner: well it's spread out across three places, so everyone doesn't have to go to one location...
15:19.47*** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx)
15:19.49mitcheloc[TK]D-Fender: what are you referring to?
15:21.06[TK]D-Fendermitcheloc : concerning their failure to have come up with the WORST possible name for TrixBox :)
15:21.56mitchelocahh, well, name's rarely matter for products/projects as long as they are solid
15:22.06rene-it is pretty lame tho
15:22.13rene-(the name)
15:22.13mitchelocthe thing that kicks me is that you can't use "asterisk" in your project name
15:22.18mitcheloc*** or aren't supposed to
15:22.26[TK]D-Fenderrene- : its flaming
15:23.09mitcheloc[TK]D-Fender: do you have any name suggestions? i'm working on a gui, and i need a name
15:23.34*** join/#asterisk dgilmore (n=dennis@fedora/dgilmore)
15:23.46rene-i was going to work on a GUI and i was going to name it Masterisk but i dont know when i am going to ship this if ever so if you like it
15:24.14mitchelocrene-: what kind of gui?
15:24.22*** mode/#asterisk [-o file] by ChanServ
15:24.28mitchelocM* doesn't really give it away ;)
15:24.29MikeJ__heh
15:24.35MikeJ__file you got -o'd
15:24.40fileChanServ restarted ;(
15:24.59rene-just a front for Asterisk Realtime Mysql tables
15:25.26rene-but i found a really good one what it is already in my language of choice so, i much better use that one
15:25.33mitchelocthat's a ton of work =/
15:25.46rene-you can find it under rubyforge telephony apps
15:25.52rene-it is very well done
15:26.05Dr-Linuxfile: question, what i suppose to get, if PRI cable is not connect and i try do dialout?
15:26.07mitcheloc[TK]D-Fender: i'd still love to hear your name suggestion ;)
15:26.09akkeanyone here can offer flat fee SIP/IAX dial-out to belgium landlines?
15:26.32fileDr-Linux: out of a zaptel channel? no clue
15:27.01Dr-Linuxfile: in FXO case, it makes me conneted to the channel, even phone line is not connected
15:27.31fileFXO case is different, for a PRI it would know it's in red alarm and probably refuse to allocate a channel
15:27.31Dr-Linuxcurrently i gives me "notice"
15:27.32Dr-Linux<PROTECTED>
15:27.35filewith whatever reason code
15:27.42tomtom_so nobody can help me with philipine did's?
15:28.17Dr-Linuxfile: but my first span is not "RED" it's "OK"
15:28.31fileDr-Linux: well you never told me what you tried to dial out on
15:29.00[TK]D-Fendermitcheloc : Pick something vaguely normal is business-like  like "QuickPBX", "InstaPhone", or something catchy that doesn't sounds some gay punk created it in his basement...
15:29.33Dr-Linuxfile: have a look > http://pastebin.com/718048
15:30.01Dr-Linuxand
15:30.01mitcheloc[TK]D-Fender: so i take it, you don't like "druid" (the name of an * web gui)?
15:30.09Dr-Linuxfile: i'm trying this >> http://pastebin.com/718506
15:31.03fileI would go there except it's not loading
15:31.07fileah there we go
15:31.09[TK]D-Fendermitcheloc : Cheap twist on "Wizard" (TM'd?  that'd be stupid), and the entire concept of quick setups as "wizardry" is what perpetuates stupid users.  Then again, thats what GUI's do so.... hrm
15:31.24*** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net)
15:31.41fileyou're dialing out on group 3, which I assume is down
15:31.50*** join/#asterisk momelod (n=momelod@HSE-Montreal-ppp133997.qc.sympatico.ca)
15:31.53asterboydawson
15:31.56momelodhello peoples
15:32.04filewell, down in that all the spans part of the group are down
15:32.15Dr-Linuxfile: when i dialout throug group 1  , it gives me same busy/congestion notic though
15:32.16*** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca)
15:33.02mitcheloc[TK]D-Fender: i'm not keen on gui's either for configuring asterisk, however some people would argue that modern software should have gui's, so who knows on that one, that's why i like to work on guis for end users and not administrators
15:33.30fileDr-Linux: so try to dial out an exact channel on the span that's up and see
15:33.42fileand try out the group that supposedly has a span that's up, and pastebin it too
15:33.50*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)
15:33.56momelodi've got a question about my zaptel card.  I have been experiencing echo on my telephone when i call out on a zap interface.  The person im calling doesnt hear the echo, only i hear my own voice echoed back to me on my reciever.. I used ztmonitor as described here: http://www.voip-info.org/wiki/view/Asterisk+PSTN+interface+debugging and saw that my transmit signals are way too high.  How can i manipulate the transmit signal stren
15:34.20Dr-Linuxfile: ok, wait
15:34.30*** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com)
15:34.35feldmomelod: zapata.conf i believe in your asterisk folder
15:34.41feldyou can do rxgain and txgain settings
15:34.51momelodthanx
15:34.52felds/folder/directory/
15:35.00rene-mitcheloc: the way i see it, asterisk is a pbx toolkit, and it you are using a subset of it for specific functionality then a GUI is a must, since you cant realistically expect you users to be fluent in technologies like asterisk and linux, so a GUI is the way to go for specific purpose apps, and the name should let people know what the app is about
15:35.14feldnobody says "folder" except when they're using windows... which i do too often.......
15:35.49coppicei thought a folder was a dot com
15:36.12asterboygui for asterisk? fuck that
15:36.12coppicea manila folder is surely a bankruptcy in the philipinnes
15:36.41Dr-Linuxfile: here i dialout through group 1, but same >> http://pastebin.com/718578
15:36.57mitchelocrene-: well, for specific purpose apps yes, but general purpose, not so much.. and that's just my opinion
15:37.06JoseBravoHow can I contact ro digium for fix my problem?
15:37.13mitchelocasterboy: you would say no to a gui for users?
15:37.16l-fyJoseBravo > call at 500
15:37.17asterboypickup a phone and call them.
15:37.24asterboyyep
15:37.39asterboyno gui for users...for asterisk anyway.
15:38.00mitchelocasterboy: by user's i mean people sitting at workstations...not admins
15:38.35asterboymaybe the flashop stuff would be nice.
15:38.45mitchelocasterboy: would you rather be using telnet for irc =P
15:40.21Dr-Linuxfile: no clue? ... how can i dialout through s specific channel? exten => _91NXXXXXXXXX,1,Dial(Zap/1-1/${EXTEN:1})
15:40.24asterboya web interface is sufficient...but then I'd argue that it's not gui in that you don't have to run XWindows or something intensive on resources
15:40.27Dr-Linuxis that correct?
15:40.29*** join/#asterisk Meaty (n=cp_simbu@office.abi.ca)
15:40.34fileDr-Linux: Zap/1/blah
15:40.46akkeanyone here can offer flat fee SIP/IAX dial-out to belgium landlines?
15:40.58l-fyakke > not me
15:41.09pjoakke: check voip-info
15:41.11mitchelocasterboy: a "web interface" for asterisK? if so you can run it on a different machine (and should)
15:41.23akkeok
15:41.24Dr-Linuxexten => _91NXXXXXXXXX,1,Dial(Zap/1/${EXTEN:1}) < is this correct? :S
15:41.25mitchelocof coures it has to be programmed correctly
15:41.30Dr-Linuxor i need to mentioned group as well?
15:41.31asterboyhow do you mean run on different machine?
15:41.41asterboydifferent binaries?
15:41.44akkeanyone tried voipcheap.com? supports SIP protocol but i wonder how usable it is ?
15:41.45fileyes that's fine as far as I know, but as I said before... I don't do zaptel :P
15:42.08mitchelocasterboy: i mean, lets say for a call log analyzer, like ast-stat, it just needs a mysql connection
15:42.08Dr-Linuxlol
15:42.17mitchelocasterboy: both pieces can be on a seperate server
15:42.20asterboyah, yes...thats nice
15:42.34Dandanneed an PRI expert for hire... please /msg me
15:43.01pjoakke: nope, but to save you some searching http://www.voip-info.org/wiki-VOIP+Service+Providers has quite a list. (and i think there are some user comments at the end)
15:43.04mitchelocasterboy: the problem is most people won't bother setting up the second machine =/
15:43.19Hmmhesaysbah
15:43.22Dr-Linuxaww
15:43.23Dr-Linuxthis time:
15:43.24Dr-LinuxNOTICE[10843]: app_dial.c:1040 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown)
15:43.24Dr-Linux<PROTECTED>
15:43.34mitchelocHmmhesays: ?
15:45.32*** join/#asterisk LH-euhost (n=LH-euhos@82.131.200.80.pool.invitel.hu)
15:45.54KattyHmmhesays: the company is /finally/ moving! we're getting a new building and i'm getting my own office (=
15:46.09Hmmhesayscool
15:46.14Hmmhesaysi'm still getting the shaft here
15:46.19Kattydamn :<
15:46.36Hmmhesaysohwell
15:47.15Dandanneed an PRI expert for hire... please /msg me
15:47.33salviadudthe shaft?
15:48.01HmmhesaysDandan: what a cryptic message
15:49.42*** join/#asterisk catlee (n=catlee@Z-pc1-959-S1.gw2.tor1.rogerstelecom.net)
15:49.45catleeGood morning
15:50.21SplasPoodHey all, question..   Say I had a call listening to MusicOnHold() and then I wanted to command asterisk to drop MusicOnHold and instead bridge it to another channel.. possible?
15:51.50catleeIf I wanted to use * to connect to my POTS at home, would things like call-waiting, and voicemail if the line is busy still work?
15:56.10Dr-Linuxfile: have a look, maybe it helps > http://pastebin.com/718611
16:00.24[TK]D-Fendercatlee : yes
16:00.55catleeHow does voicemail work if the line is busy?
16:01.00[TK]D-Fendercatlee : if by that you mean a phone connected to *.  if the call never makes it to * because of something telco related well that out of its hands.
16:01.25[TK]D-Fendercatlee : Depends on your definition of "line is busy".  You referring to a PHONE on your system, or an incoming "line"?
16:01.39catleewell, let's say I have one incoming POTS line
16:01.44catleeand I'm using it to talk to somebody
16:02.03[TK]D-Fendercatlee : then no.  Forget call waiting on that line, and if its busy, TFB
16:02.33catleeright now, if somebody else calls me, I will be notified that there's another caller on the line, and if I don't answer they will be sent to my voicemail
16:03.03catleeok, that's what I thought
16:03.24catleebut different with incoming calls via SIP/IAX?
16:03.59[TK]D-Fendercatlee : depends.
16:04.13JoseBravoThe mail support of digium is fast?
16:04.56[TK]D-Fendercatlee : think of it in terms that your telco (whatever tech) HAS to be able to send the enxt call to *.  That means and extra PSTN line or being permitted multiple VoIP calls simultaneously.  Every call takes a line.
16:05.51catleeah, so that's why VoIP providers list # of simultaneous calls?
16:06.08[TK]D-Fendercatlee : In typical ITSP's you get to have 2 calls at a time.  If you're using an analog phone you can switch/conference between them by use of callwaiting style functionality.  On a SIP phone you may be able to treat them more naturally like in a normal PBX.
16:06.14[TK]D-Fendercatlee : Yup...
16:06.29[TK]D-FenderBecause if you want * to do the VM it obviously has to be the one answering that call.
16:06.34catleeyup
16:07.09*** part/#asterisk faberk64 (n=faberk@213.199.15.249)
16:07.22*** join/#asterisk faberk64 (n=faberk@213.199.15.249)
16:11.22catleeSo that's why it may make sense to have voicemail provided by the telco
16:11.55[TK]D-Fendercatlee : Nope... thats why it makes sense to have more lines than calls :)
16:12.06[TK]D-Fendercatlee : Or feel safe in knowing people will call you back.
16:12.24[koss]are there any turn-key asterisk solutions i can buy to replace a PBX for about 50 phones and 10 telco lines?
16:12.42dlynes_office[TK]D-Fender: btw...just thought I'd let you know
16:12.54dlynes_office[TK]D-Fender: everything seems to be working fine (for the most part) with the a200d
16:13.03dlynes_officein a 2.6.15.5 kernel, too
16:13.41dlynes_officethe only thing that doesn't seem to be working is the HWEC
16:13.56catlee[TK]D-Fender: that's probably not an option for many residential users, is it?
16:14.03dlynes_officeDo I not use echocancel=yes in zapata.conf if i'm using the hwec?
16:14.27*** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com)
16:14.32*** join/#asterisk Egonis (n=Egonis@207.245.14.10)
16:15.41akkeanyone here can offer flat fee SIP/IAX dial-out to belgium landlines?
16:15.44EgonisI am using a TDM400P w/ 2 FXS Cards, and I notice that I get lots of cracks/beeps during conversations -- when I load ALSA, it gets worse.. although i have noload => chan_alsa.so -- has anyone else experienced this?
16:16.32dlynes_officeEgonis: what's your cpu load like?  what kinda cpu are you using?
16:16.32Egoniswhen I try load => chan_oss.so and noload => chan_alsa.so and the reverse, it makes no difference
16:16.45catleealthough it makes sense for businesses, and most businesses should already have enough lines
16:17.28dlynes_officecatlee: probably not an option for residential users, no
16:17.29Egonisdlynes_office: 99.3% idle, PIII 1.0
16:17.40DandanEgonis: and your irq?
16:17.46EgonisDandan: HOw do I find out?
16:17.51dlynes_officecatlee: but then again, pbx is probably unnecessary for a residential user, too :)
16:17.57Dandanstatic is usually caused by irqs and buffer underruns
16:18.04Dandanegonis with stuff like mpstat
16:18.07Dandanafair
16:18.17EgonisDandan: mpstat?
16:18.28dlynes_officeEgonis: lspci -vv
16:18.34dlynes_officeEgonis: cat /proc/interrupts
16:18.34Dr-Linuxdlynes_home: hey :)
16:18.37dlynes_officehey
16:18.40Dandanyeaf, from sysstat package
16:18.42Dandanre dlynes_
16:18.43Dandanre dlynes_office
16:18.44Dandan:)
16:19.02*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
16:19.03Dandani offered some money for anyone willing to resolve my issue
16:19.05Dandanwith PRI
16:19.18dlynes_officeEgonis: oh..you're using bsd, not linux?
16:19.31dlynes_officeDandan: which problem was that again?
16:19.33Egonisdlynes_office: And what am I looking for in cat /proc/interrupts? I get:
16:19.33Egonis193:  781748155   IO-APIC-level  wctdm
16:19.34catleedlynes_home: true :)  I was thinking it would be cool to set up so I could phone my family using VoIP...And maybe even use each server as a termination point into its local area code
16:19.40DandanPRI calls going out, not coming in
16:19.42Egonisdlynes_office: Gentoo Linux
16:19.43DandanBLAH
16:19.53DandanPRI calls *NOT* going out, but coming in...
16:19.54dlynes_officeDandan: ah...and did you get the problem fixed?
16:19.55Dr-Linuxdlynes_home: my both TE21OP card has been succesfully installed. but can't dialout
16:20.10Dr-Linuxdlynes_home: http://pastebin.com/718611
16:20.11dlynes_officeDr-Linux: can peeps call in?
16:20.22Dandandlynes_home: no, i just posted a follow up asking to step forward if you know PRI and I will pay you to have it resolved
16:20.31dlynes_officeah
16:20.32Dr-Linuxdlynes_home: didn't check that yet, don't have numbers, still waiting for
16:20.39Dr-Linuxdlynes_home: also have a look > http://pastebin.com/718048
16:20.55dlynes_officeDr-Linux: if you don't have numbers yet, what makes you think the pri is even functional?
16:21.14Dandandlynes_home: http://lists.digium.com/pipermail/asterisk-users/2006-June/156266.html
16:21.24Dr-Linuxdlynes_home: i mean, i don't know number yet, but it should dialout
16:21.24Egonisdlynes_office: So what should I check next?
16:21.28Dr-Linuxbut i gives .. :
16:21.49*** join/#asterisk mog (i=ejabberd@68.62.237.103)
16:21.49dlynes_officeEgonis: zttest
16:21.51Dr-Linux<PROTECTED>
16:21.52Dr-Linux<PROTECTED>
16:22.09Egonisdlynes_office: Should I shutdown asterisk prior?
16:22.16dlynes_officeDr-Linux: yeah...sounds to me like your pri is down
16:22.21dlynes_officeDr-Linux: type zap show status
16:22.51Dr-Linuxdlynes_home: that should 1 span "OK" other 3 "Alarm"
16:23.01dlynes_officeDr-Linux: and what span are you trying to dial out on?  span 1 or span 2?
16:23.06Dr-Linuxsorry "RED"
16:23.07dlynes_officespan 2 is in alarm
16:23.15Dr-Linuxdlynes_home: span 1
16:23.20Dr-Linuxthe "OK" one
16:23.22dlynes_officeand both spans are down
16:23.27dlynes_officenot up
16:23.33Dr-LinuxDescription                              Alarms     IRQ        bpviol     CRC4
16:23.35Dr-LinuxT2XXP (PCI) Card 0 Span 1                OK         0          0          0
16:23.35Dr-LinuxT2XXP (PCI) Card 0 Span 2                RED        0          0          0
16:23.44dlynes_officeDr-Linux: read your pri show span 1
16:23.51Dr-Linuxdlynes_home: the one i pasted in ?
16:23.53dlynes_officeStatus:  Provisioned, Down, Active
16:23.57Dr-Linuxyeah, you are right
16:24.02dlynes_officeThe 'down' means it's down
16:24.06Dr-Linuxbut why it showing "OK"  the span 1 ?
16:24.06dlynes_officeas in it's not operational
16:24.13dlynes_officeBecause it's not in alarm
16:24.17Egonisdlynes_office: Does zttest end? I am getting mostly 100.00% readings, but worst appear to be 85%
16:24.30dlynes_officeiow, your eq and your telco's eq are talking to each other
16:24.37dlynes_officebut you're not getting service
16:24.45Dr-Linuxdlynes_office: can you explain a bit, "bcoz it's not in alarm" ?
16:24.47dlynes_officeEgonis: it should run for about 2-3 minutes
16:25.11dlynes_officeEgonis: it should run at 99.875% or higher
16:25.13Egonisdlynes_office: Okay, thanks.. should I specify for it to not run on the zap pseudo interface? should I force it to measure on a Wildcard Channel?
16:25.14Dr-Linuxdlynes_office: what do you mean it's not in alarm?
16:25.29Egonisdlynes_office: How would I adjust it to run above or at that level?
16:25.32dlynes_officeEgonis: iow, you're probably having irq issues
16:25.38dlynes_officeEgonis: you don't
16:25.42dlynes_officeDr-Linux: hold your horses
16:25.48dlynes_officeDr-Linux: i've only got one keyboard
16:25.53Egonisdlynes_office: nice... any suggestions for fixing irq issues?
16:26.00Dr-Linuxheh :) ok
16:26.04dlynes_officeEgonis: how many pci slots do you have?
16:26.20Egonisdlynes_office: two, both full -- one is an IBM ServeRAID, the other is the TDM400P
16:26.23*** join/#asterisk JoseBravo (i=JoseBrav@200.119.32.47)
16:26.51dlynes_officeEgonis: ok, does your bios allow you to force slots onto certain irqs?
16:26.58Egonisdlynes_office: nope! :(
16:27.06[TK]D-Fenderdlynes_office : Yes, you are supposed to use echocancel=yes for the HWEC.  pastebin your wanpipe1.conf
16:27.53dlynes_officeEgonis: yeah...the digiums are really picky about interrupts; the apic is only masking the problem
16:27.55*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
16:28.05dlynes_officeEgonis: can you try the card in a different machine?
16:28.17asterboy~echo
16:28.19jbotwell, echo is an issue which can be best fixed using this link: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html#AEN1718, or fixed with fxotune: http://www.voip-info.org/wiki/view/Asterisk+fxotune, or best fixed by troubleshooting your pci bus: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting, or of ...
16:28.21Egonisdlynes_office: Theoretically, yes.. but with lots of work, this is my only asterisk box
16:28.28Dr-Linuxdlynes_office: look for your quick and last answer, that's why i'm looking for then i need to run.
16:28.43Egonisdlynes_office: Is there a kernel option I should use?
16:28.46dlynes_officeDr-Linux: there's not much you can do...once you get your dids, try it again
16:29.00dlynes_officeDr-Linux: if it's still not working, then complain to your telco that your pri is not up yet
16:29.12asterboyI just fixed my digim blues.
16:29.26Dr-Linuxdlynes_office: yes i understand that, but why first span is "OK"
16:29.27dlynes_officeasterboy: you bought a sangoma?
16:29.30asterboyYou really need a tight kernel, no acpi
16:29.31Dr-Linuxthat's my questin
16:29.33dlynes_officeDr-Linux: it's not ok
16:29.35dlynes_officeDr-Linux: it's down
16:29.38asterboyhave sangoma as well
16:29.38Dr-Linux<dlynes_office> Because it's not in alarm  << please explain it
16:29.52Egonisasterboy: So if I have crack/beep noises during calls, I should kill acpi? i.e. noacpi in kernel options?
16:29.55Dr-Linuxdlynes_office: i know it's down, but it shows "OK" but other span not
16:30.06asterboyhell ya
16:30.08dlynes_officeDr-Linux: both spans are down, one is in alarm
16:30.16dlynes_officeDr-Linux: alarm means the other end is not connected to anything
16:30.18Egonisasterboy: And that can possibly fix irq issues?
16:30.26asterboyif zttest has 99.975 or lower ... you'll get that
16:30.33asterboyyep
16:30.35Egonisasterboy: Mine bounces from 100 to 85
16:30.41asterboy85?
16:30.41dlynes_officeEgonis: yeah...sangoma has almost no issues with sharing interrupts
16:30.43Dr-Linuxawwwwwwwwwwwwww nice answer :)
16:30.43asterboyyikes!
16:30.51Egonisdlynes_office: Yeeeah, I've heard
16:30.52asterboyyou have a serious problem with 85
16:30.57Dr-Linuxdlynes_office: thanks,
16:31.00Egonisasterboy: I would tend to agree.. :)
16:31.04asterboyEgonis, follow this:
16:31.06asterboy~echo
16:31.07jbotwell, echo is an issue which can be best fixed using this link: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html#AEN1718, or fixed with fxotune: http://www.voip-info.org/wiki/view/Asterisk+fxotune, or best fixed by troubleshooting your pci bus: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting, or of ...
16:31.16asterboyespecially the last entry
16:31.16Dr-Linuxdlynes_office: what i suppose to get while dialing out, if my PRI cables are not connected?
16:31.29dlynes_officeDr-Linux: you can't dial out...your pri is down
16:31.34JoseBravoI need to go to take a coffe or go to vacations until Digium Support response my mails?
16:31.37dlynes_officeDr-Linux: what part of 'down' do you not understand?
16:31.53asterboyIn BIOS I turned off every unnecessary option
16:31.56Dr-Linuxdlynes_office: yes, i understand :)
16:32.02dlynes_officeEgonis: ideally you shouldn't have acpi or apic enabled
16:32.03asterboySound, Para Port, Serial Ports, blah blah
16:32.07*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
16:32.08Dr-Linuxdlynes_office: i'm asking things just for learning purpose
16:32.12dlynes_officeEgonis: also, you should enable dma on your hard drives
16:32.19dlynes_officeDr-Linux: I've already told you everything
16:32.28dlynes_office[TK]D-Fender: http://pastebin.com/718701
16:32.30asterboyThen compiled the kernel with no mouse, no everything I did not need.
16:32.31Dr-Linuxdlynes_office: bcoz i used FXO not PRI, and they are kinda different
16:32.44Dr-Linuxdlynes_office: ok thanks
16:32.46dlynes_officeDr-Linux: umm...you were showing me pri, not fxo
16:32.46Dr-Linuxbrb
16:32.49asterboyThat brought my zttest scores within 5 9's
16:33.01asterboynow the system is working perfectly.
16:33.18Dr-Linuxdlynes_office: yes, but in the FXO case if the phone line is not connected, it will still connect you to the channel,
16:33.20asterboyotherwise you get crack, drops, noice, bad dtmf
16:33.24dlynes_officeJoseBravo: no, not usually
16:33.37dlynes_officeJoseBravo: emailing them is generally not very efficient
16:33.38Dr-Linuxdlynes_office: but PRI  doesn't go this way :S
16:33.42dlynes_officeJoseBravo: try calling them
16:33.55dlynes_officeDr-Linux: no it won't
16:34.02dlynes_officeDr-Linux: it'll tell you the line is congested also
16:34.28JoseBravoWhat is the digium support line?
16:34.41Dr-Linuxdlynes_office: yes, correct, that's what i was trying to asking since 6 hours :)
16:34.45dlynes_officeJoseBravo: it's on their website, but you can also call them via iax
16:34.49[TK]D-Fenderasterboy : If you aren't 5-9's, you aren't using enough decimal places :D  bound to be a few more 9's in the next hundred digits or so of decimal precision....
16:34.54Dr-Linuxi knew you will have an answer :)
16:35.06dlynes_officeDr-Linux: and i gave you an answer so get over it :p
16:35.22Dr-Linuxdlynes_office: yeah, thanks
16:36.05[TK]D-Fenderdlynes_office : Pastebin.com is being completely slow for me... can you use .ca....
16:36.20dlynes_office[TK]D-Fender: yeah..one sec...going to have to find the ip address for it first
16:36.30dlynes_office[TK]D-Fender: stupid dns servers that use ipv6 anyways :p
16:36.48CunningPikeMorning all
16:36.54*** part/#asterisk Egonis (n=Egonis@207.245.14.10)
16:37.44dlynes_officemorning, cp
16:38.12*** join/#asterisk terrapen (n=cjs@166.70.183.108)
16:38.29CunningPikeHey, dlynes_office
16:39.07akkeanyone here can offer flat fee SIP/IAX dial-out to belgium landlines?
16:39.16dlynes_office[TK]D-Fender: http://159.18.52.69/index.php
16:39.22dlynes_officeerm
16:39.32dlynes_office[TK]D-Fender: http://159.18.52.69/67245
16:39.49terrapenI wonder, is there any way to set up Asterisk server roaming for softphone users?  Like, where they get the local asterisk server for the office that they happen to be in at the time
16:39.58terrapenI suppose it could be done with DNS but that's kind of ugly
16:40.23*** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com)
16:40.43dlynes_officeterrapen: Yeah, you could set something like that up with an ettercap or ethereal parser, or something like that
16:40.44[TK]D-Fenderdlynes_home : Has Sangoma logged in to confirm the DSP is up and running?
16:40.51dlynes_office[TK]D-Fender: nope
16:40.57terrapenwhat does ethereal have to do with it?
16:41.06dlynes_office[TK]D-Fender: the logs all indicate the hwec is up
16:41.23dlynes_office[TK]D-Fender: however, I get an error during bootup
16:41.32[TK]D-Fenderterrapen : set them to use DNS at whatever site they are at.
16:41.49*** join/#asterisk zamba (i=marius@2001:730:5:21e0:0:0:0:1)
16:42.06dlynes_officeterrapen: yeah...another way you can do it too, is if the phones support SRV entries in the DNS
16:42.08terrapend-fender, yeah, that's the only solution i can come up with...i was wondering if there was something more elegan
16:42.09terrapent
16:42.37zambai'm planning to set up a voicemail service in an already running telephony environment.. i want one of the extensions to point to an asterisk server.. what kind of hardware do i need for this?
16:42.37dlynes_officeterrapen: srv entries are more elegant, but not all sip phones support them
16:42.38[TK]D-Fenderterrapen : Slightly less elegent, but you could just set up 2-3 peers and let them hand-pick.
16:43.04[TK]D-Fenderzamba : Depends how you want to interface the 2 systems.
16:43.13zamba[TK]D-Fender: what are my alternatives?
16:43.24[TK]D-Fenderzamba : Typically you'd use analog ports, or preferably T1/E1
16:43.37terrapenI'm probably going to be using X-LITE and Ekiga
16:43.39zambaE1 is the same as ISDN, right?
16:43.41[TK]D-Fenderzamba : How many ports do you want for it?
16:43.42dlynes_officezamba: what phone system, specifically?
16:43.49zambadlynes_office: some alcatel-stuff..
16:44.00dlynes_officezamba: e1 is pri, not bri
16:44.02[TK]D-Fenderzamba : E1 is ISDN PRI for EU.
16:44.04zambadlynes_office: "no one" in the office knows about it, it has just been running there for years :)
16:44.28zamba[TK]D-Fender: "many ports" means how many will be able to call the service at the same time, right?
16:44.36[TK]D-Fenderzamba : Correct.
16:45.00zamba[TK]D-Fender: just one to begin with, but i want the system to scale
16:45.14[TK]D-Fenderzamba : I had an old Nortel phone system with a 4 port VM which obvious how many people can be picking up / leaving VM at a time.  That usit also served as our IVR
16:45.33zamba[TK]D-Fender: too many abbrievations :) VM and IVR?
16:45.35[TK]D-Fenderzamba : What kind of ports do you have available on that PBX?
16:45.50*** join/#asterisk websae (n=websae@209-252-79-66.ip.mcleodusa.net)
16:45.56[TK]D-Fenderzamba : VoiceMail.  Interactive Voice Response (auto-attendant)
16:45.57zamba[TK]D-Fender: i'm not quite sure, i'll have a look at it later today.. take some pictures and stuff..
16:46.01zambaah
16:46.08*** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com)
16:46.10[TK]D-Fenderzamba : Pictures = no good... need line specs.
16:46.18zambayeah, i'll get that as well
16:46.49[TK]D-Fenderzamba : T1/E1 users RJ48 which is indistinguisable from Ethernet & Norteles proprietary stuff really...
16:47.17zambai think it's rj11
16:47.23[TK]D-Fenderdlynes_home : I take it you compiled the HWEC tools seperately?>
16:47.24zambaso analogue?
16:47.31[TK]D-Fenderzamba : Quite likely.
16:47.33dlynes_office[TK]D-Fender: ummm...no?
16:47.51zamba[TK]D-Fender: but what are the alternatives for hardware on the asterisk box?
16:47.56dlynes_office[TK]D-Fender: i unzipped the main sangoma tarball, and then i unzipped the hwec tarball into the same directory
16:48.06[TK]D-Fenderzamba : For that you'd use an analog TDM card.  Probably 4-port cardwould serve your needs to start
16:48.15zamba[TK]D-Fender: got some names for me?
16:48.17dlynes_office[TK]D-Fender: and then ran the main setup
16:48.26[TK]D-Fenderdlynes_home : Go inspect before I start swinging the trout :)
16:48.36dlynes_office[TK]D-Fender: go inspect what?
16:48.42[TK]D-Fenderzamba : Digium TDM400P, Sangoma A200
16:48.50[TK]D-Fenderdlynes_home : the HWEC tools packacge
16:49.00dlynes_officewhat should i be looking at in it?
16:49.10[TK]D-Fenderdlynes_home : just go look....
16:49.39dlynes_office[TK]D-Fender: you mean the wan_ec subfolder off of /etc/wanpipe?
16:49.47dlynes_office[TK]D-Fender: or the original directory I installed from?
16:51.08zambaoh shit, those cards are expensive :)
16:51.29dlynes_officezamba: well, they're not video cards :)
16:51.32zambahehe
16:51.34*** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)
16:51.41generalhanwhats up all
16:51.53dlynes_officeni zao, generalhan
16:52.22generalhanI still cant get this stupid MWI issue fixed with 2 of my 7960s ....
16:52.23[TK]D-Fenderdlynes_home : Look in the wanpipe/util/wan_ec from your source folder for some of the diagnosis binaries.
16:52.26generalhanits driving me insane !
16:52.41[TK]D-Fendergeneralhan : Get a pair of pliers and yank them out..
16:53.12generalhanlol ... for that matter i can just turn them off by removing the 'mailbox=' line in sip.conf. the goal is to get it to function properly ! lol
16:53.25dlynes_officegeneralhan: thought you said you were gonna try chan_skinny as soon as you got home yesterday?
16:53.49generalhandlynes_home: wasnt me ... i havent been on since Friday ( i only get on while im at work )
16:54.02[TK]D-Fendergeneralhan : Go make sure the VM box has no files in it...
16:54.03generalhani try as hard as i can to forget about everything related to work when i leave
16:54.16dlynes_officegeneralhan: ah...anyways...if you check trunk, Qwell's committed a bunch of changes to chan_skinny
16:54.29*** join/#asterisk Qwell[] (i=north@unaffiliated/qwell)
16:54.30dlynes_officegeneralhan: it might be much more stable than trying to run sip on a cisco now
16:54.35dlynes_officespeak of the devil
16:54.42dlynes_officeyour ears must've been ringing, qwell
16:54.45generalhan[TK]D-Fender: nothing in it ... and ive even removed that line from voicemail.conf reloaded ... then put it back in so that it would create a new VM folder
16:55.19generalhanthat stupid red light just wants to stay on
16:55.37dlynes_office[TK]D-Fender: what should I supply as a parameter for the devname for wan_ec_client?
16:55.43dlynes_office[TK]D-Fender: the if_name is obvious
16:56.15generalhani thought that it was odd because they all have the same firmware .. so i actually downgraded that phone, and then reloaded the new version and still the stupid red light stays on
16:58.09[TK]D-Fenderdlynes_office : Don't know the details.. I just remember the tech using that do diagnose problems with the DSP not firing up properly.
16:58.24dlynes_office[TK]D-Fender: ah..ok
16:58.27dlynes_office[TK]D-Fender: thx
17:01.15*** join/#asterisk LokeshIndian (n=lokesh_k@estrela.nortenet.pt)
17:02.21*** join/#asterisk JoseBravo (i=JoseBrav@200.119.32.47)
17:02.24*** join/#asterisk paryl (n=chatzill@216-201-177-82.res.logixcom.net)
17:02.36LokeshIndianHello, can anyone please help me???
17:02.39paryldoes anyone know what "Don't know what to do if second ROSE component is of type 0x6" means?
17:03.15[TK]D-FenderLokeshIndian : www.drphil.com
17:03.35inv_Arp[TK]D-Fender: lol
17:04.14*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
17:04.32dlynes_officeparyl: i've searched all over the damned place on asterisk-dev, asterisk-users, google, ... I haven't been able to find a damned thing on it, either
17:04.54LokeshIndian[TK]D-Fender: If my softphone is off and i try to call it then i gets some wierd entries in asterisk cdr
17:05.10LokeshIndian[TK]D-Fender: how i can get rid off with that ?
17:05.23Hmmhesaysprobabably because by defaults your cdrs reflect parts of your dialplan
17:05.27dlynes_officeLokeshIndian: weird entries....good description
17:06.01LokeshIndiandlynes_office:hang on plz i m pasting them here
17:06.16dlynes_office~pb
17:06.18jbotwell, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/
17:06.18*** join/#asterisk Delvar (n=irc@host-83-146-53-46.bulldogdsl.com)
17:07.09LokeshIndiandlynes_office:-- Executing Hangup("Local/4365@from-sip-ec13,2", "") in new stack
17:07.09LokeshIndian<PROTECTED>
17:07.38CunningPikeLokeshIndian: Looks like normal call progression to me........
17:07.39zamba[TK]D-Fender: Intel Dialogic D/4-PCI ok?
17:07.42dlynes_officeno kidding
17:07.51dlynes_officezamba: dialogic is absolute crap
17:07.57dlynes_officezamba: probably worse than digium
17:08.04zambadigium is crap as well?
17:08.13zambawhat's no crap(tm)?
17:08.18dlynes_officezamba: not really, but it does have problems
17:08.25dlynes_officezamba: dialogic is horrible though
17:08.29*** join/#asterisk Katty (n=angela@64.82.232.54)
17:08.31[TK]D-Fenderzamba : Avoid Dialogic.... barely usable at best...
17:08.38LokeshIndianCunningPike: if the softphone is off then i had 60-10 lines like i pasted here in the asterisk logs
17:08.41[TK]D-Fenderzamba : Whats your idea of "expensive"?
17:08.53CunningPikezamba: Digium can be fussy to set up (some people have no problems, some do). The only real issue we had with Digium cards is the on-board EC
17:08.54LokeshIndianand all those entries are also gets logged in asterisk cdr database
17:09.05zamba[TK]D-Fender: well, since this is merely for testing, i'd prefer to be under 100$
17:09.34paryli'm getting some weird things with IAX... i've got two asterisk boxes, and we make calls back and forth without any issues
17:09.34dlynes_officezamba: go with an ATA or an analog gateway then
17:09.54[TK]D-Fenderzamba : Would you like fires with that sir? :)  Were talking about PBX's here....
17:10.01parylbut in the last couple days , during calls, we've been getting just dead air... like 5-10 seconds, totally dead, and the call comes back and proceeds normally
17:10.07parylnothing is in the logs
17:10.09zamba[TK]D-Fender: i know, but i'm new to this, and i really have no budget :)
17:10.15[TK]D-Fenderzamba : around $200 for 2 line Sangoma A200
17:10.26zamba[TK]D-Fender: and since i can't guarantee success it's hard for me to sell this to my company :)
17:10.27[TK]D-Fenderfries*
17:10.46dlynes_officezamba: you can get Yoda G620's, Sipura 2000's (ATA), Sipura 3000 (analog gateway)
17:10.53[TK]D-Fenderzamba : How about you just replace the entire PBX with *.... that'd be effective...
17:10.58dlynes_officezamba: but they're not hardware cards...they're external units
17:11.03LokeshIndianCunningPike: its not normal call progression..can u plz help me
17:11.06zamba[TK]D-Fender: then i definitely can't guarantee success :)
17:11.15[TK]D-FenderTalks does Yoda funny hhmmmMMMMMM?!?!
17:11.19dlynes_officeLokeshIndian: it is normal call progression
17:11.46dlynes_officeCunningPike: you mean the lack of an onboard ec?
17:11.54zambadlynes_office: how do i connect those to asterisk? LAN?
17:12.01dlynes_officezamba: exactly
17:12.08zambainteresting
17:12.18dlynes_officezamba: they're sip devices
17:12.25dlynes_officezamba: the yoda devices can also do h323
17:12.30*** join/#asterisk Gabriel25 (n=gabe@user-12ld5f7.cable.mindspring.com)
17:12.34CunningPikedlynes_office: No - we found the VPM less effective than software EC, and it introduced call quality issues (clicks and dropouts)
17:12.34Gabriel25Hi hys
17:12.39Gabriel25[root@server zaptel]# make
17:12.39Gabriel25You do not appear to have the sources for the 2.6.16-1.2133_FC5smp kernel installed.
17:12.39Gabriel25make: *** [linux26] Error 1
17:12.51*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-80-10.red.bezeqint.net)
17:12.51Gabriel25I have this error but I installed kernel-devel
17:12.54dlynes_officeGabriel25: pretty self explanatory, don't you think?
17:12.56Gabriel25any idea ?
17:13.20Gabriel25dlynes_home I installed kernel-devel
17:13.26CunningPikeGabriel25: symlink in /usr/src?
17:13.32dlynes_officeGabriel25: obviously not hte right kernel-dev
17:13.41dlynes_officeGabriel25: you probably installed the wrong version of kernel-dev
17:13.52JoseBravoGabriel try to install kernel-smp-devel
17:14.16tzafrir_laptopthe kernel-devel package has the symlink /lib/modules/`uname -r`/build ?
17:14.31dlynes_officetzafrir_laptop: one would think
17:14.48Gabriel25[root@server gabriel]# uname -r
17:14.48Gabriel252.6.16-1.2133_FC5smp
17:14.48Gabriel25[root@server gabriel]# rpm -qa kernel-devel
17:14.48Gabriel25kernel-devel-2.6.16-1.2133_FC5
17:15.03[TK]D-FenderCunningPike : I hear the VPM rev 2 is a noticable improvement.
17:15.13dlynes_officeGabriel25: see the lack of the 'smp' appended to that kernel version?
17:15.14[TK]D-FenderCunningPike : And that a REAL DSP is pending.
17:15.20CunningPike[TK]D-Fender: So is a Ditech box ;)
17:15.34dlynes_officeGabriel25: like i said...you don't have the correct version of kernel-dev installed....listen to JoseBravo
17:15.41Gabriel25ok
17:15.43[TK]D-FenderCunningPike : So is my Otasic ;)
17:15.48JoseBravoGabriel25 yum isntall kernel-smp-devel
17:15.50CunningPike[TK]D-Fender: Heh heh
17:16.54*** join/#asterisk Bullseye_Network (n=Kyle@216.143.192.69)
17:16.57Gabriel25ok
17:17.07Gabriel25thank you so much
17:17.13CunningPikeActually, here is a question for the group - all of a sudden, it seems that the far end of our PRIs are receiving any CID number any more - anyone ever seen that happen?
17:17.37CunningPikeAll the 'pri intense debug' looks normal, but the far end isn't seeing a number
17:17.46CunningPikeIt's weird
17:18.12dlynes_officeCunningPike: is it every far end?  or just one particular location?
17:18.29CunningPikedlynes_office: Every - our Nortel, and the telco
17:18.36CunningPikedlynes_office: Name is fine - just number
17:18.46dlynes_officeCunningPike: try giving me a call through it...see if I get your caller id
17:18.51CunningPikeOK
17:19.04dlynes_officeCunningPike: you've got my cell number, right?
17:19.38lunkhow do you setup internal routing between SIP extensions? (or a link to some info)
17:19.46lunki'm missing something :/
17:20.03Gabriel25thank you guys is working
17:20.06Gabriel25stupid me !
17:26.28JoseBravoANoye have installed AstBill without all expample info?
17:27.16CunningPikelunk: Explain?
17:27.30*** join/#asterisk wingman_sg (n=Wingman@bb219-74-103-65.singnet.com.sg)
17:28.30lunkCunningPike: i plead ignorance
17:28.52CunningPikelunk: Explain what it is you are trying to do
17:29.16wingman_sganybody know how to instal/configure l Dialogic card into Asterisk platform ?
17:29.21*** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx)
17:29.26lunki just want ext 500 to call ext 501
17:29.33lunki don't have any internal routing
17:29.46rene-is it possible to have agents defined like agent/johnny as opposed to numeric only agent channel names?
17:29.53*** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com)
17:29.54CunningPikelunk: So, have you set up a basic dialplan?
17:30.05lunkthat must be what i'm missing
17:30.14lunki can call out through voipjet, but not internally
17:30.35CunningPikerene-: It sure is
17:31.05CunningPikelunk: So, you need to have each UA in sip.conf, and then a dialplan that handles calls between them
17:31.26[TK]D-Fenderlunk : ...
17:31.27[TK]D-Fender~book
17:31.34jboti heard book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
17:31.43rene-thanks, question 2, thereis agentlogin and agentcallbacklogin, can i use addqueuemember and still have the agentlogin functionality? (always in line, new calls preceeded by beep, as opposed to telephone onhook new call is new ring)
17:32.26JoseBravolunk
17:32.28lunkCunningPike: do you have simple dial plan example? I'm looking on voip-info, but they're all pretty complex
17:32.39*** join/#asterisk ManxPower (i=ewieling@53.sub-70-219-19.myvzw.com)
17:32.42ManxPowerJun 19 12:28:51 WARNING[902]: chan_zap.c:8394 pri_dchannel: Ring requested on channel 0/2 already in use on span 1.  Hanging up owner.
17:32.43ManxPoweryippee
17:32.57CunningPikelunk: The samples that come with asterisk are good, and I highly recommend The Book
17:33.01CunningPike~book
17:33.03jbotbook is probably a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
17:33.03ManxPoweraccording to "show channels" channel 0/2 was NOT in use.
17:33.07JoseBravoYou need to add exten => 500,1,Dial(SIP/sip_user)
17:33.12rene-~rene-
17:33.14jboti guess rene- is THE MAN
17:33.22rene-i agree
17:33.38CunningPikejbot, rene- is vain
17:33.39jbot...but rene- is already something else...
17:34.02CunningPikejbot, rene- is also vain
17:34.03jbotCunningPike: okay
17:34.07CunningPike:D
17:34.08[TK]D-Fender~[TK]D-Fender
17:34.10jbotmethinks [tk]d-fender is rockin' the casbah !!!
17:34.14[TK]D-Fender:D
17:34.20mitcheloci forgot mine...
17:34.21mitcheloc~mitcheloc
17:34.22jbotsomebody said mitcheloc was your master
17:34.31mitchelocheh, i like that
17:34.33CunningPikeSettle down, children
17:34.59*** join/#asterisk sg_wingman (n=Wingman@bb219-74-103-65.singnet.com.sg)
17:35.11mitcheloc~CunningPike
17:35.15generalhan!^caret Outcast (Goodie Mobb) - They Don't Dance No More.mp3
17:35.20generalhanHAHAHA !
17:35.20mitcheloc~<insert nick here>
17:35.30generalhani Love random spam mess. on mIRC
17:35.48sg_wingmanCLEAR
17:35.53mitchelocjbot, CunningPike is invisible
17:35.54jbotmitcheloc: okay
17:35.57CunningPike:D
17:36.04lunkJoseBravo: that's all it was, thanks dude, knew it was something small
17:36.04mitcheloc~CunningPike
17:36.06jboti heard cunningpike is invisible
17:36.09rene-jbot, CunningPike is also ledgar se cae
17:36.11jbotrene-: okay
17:36.15mitchelocwhere'd he go!
17:36.18*** join/#asterisk syzygybsd (n=chatzill@66.226.228.204.cpe.speedyquick.net)
17:36.32*** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)
17:37.14paolobHi guys! Is there a string in asterisk a function returning the lenght of a variable?
17:37.20rene-this is funny :)
17:37.25syzygybsdI have a couple polycoms nated behind routers.  When I connect to asterisk (not nated) and try to make a call it is trying to send it to my local 192.168 addresses instead of my external address
17:37.40syzygybsdI am connecting via sip and have nat=1 in the sip.conf
17:37.46*** join/#asterisk Qb3rt (n=jhgjkgui@kyle.colba.net)
17:37.47rene-paolob: most likely, try show applications like len in the asterisk cli
17:38.07dlynes_officerene-: and show functions
17:38.23dlynes_officerene-: such as for 'if'; it's a function, not a dialplan application
17:38.36*** join/#asterisk thock (n=thock@63.133.144.2)
17:38.39dlynes_officeno idea what the difference is but...
17:38.43rene-jbot, CunningPike went to watch edgar se cae http://www.youtube.com/watch?v=0Ab9ERPhLh8
17:38.51ManxPowersyzygybsd, Dial by peer name, not IP address
17:39.01paolobrene-, I can't see it
17:39.13thockAnyone here pretty knowledgeable with E&m/PRI's that i can bug for a few minutes?
17:39.20CunningPikerene-: What does ledgar se cae mean?
17:39.25syzygybsdManxPower: it happens if I dial out zap from my sip phone too
17:39.38rene-i meant edgar se cae: http://www.youtube.com/watch?v=0Ab9ERPhLh8
17:39.48syzygybsdor if I do dial(sip/Gary)
17:39.50dlynes_officepaolob: show functions, you'll see 'LEN'
17:40.04paolobdlynes_home, ok, thank you!
17:40.47dlynes_officepaolob: if you don't, it's because you don't have one of the func_*.so modules loaded
17:44.41paolobwell, the command Dial(SIP/${EXTEN:LEN(${VAR})}@skypho,60,t) doesn't work. What am I wrong?
17:45.53CunningPikerene-: So which one am I? The mean orange guy, or the poor red one? :)
17:46.17thockHere's something really strange. my LDT1 is constantly calling my extension every minute and a half
17:46.19thockleaves a message
17:46.21thockand then hangs up
17:46.32thockCID shows as "unknown"
17:47.53paolobdlynes_home, could you tell me if I can put Dial(SIP/${EXTEN:LEN(${VAR})}) ?
17:48.15*** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au)
17:48.38[TK]D-Fenderpaolob : Do you have any clue what that does?  Basically its returns a BLANK.
17:48.45*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
17:48.56*** join/#asterisk Egonis (n=Egonis@207.245.14.10)
17:48.57rene-i dont know you get to pick victim or saddistic cousin
17:49.02[TK]D-Fenderpaolob : Actualy... what's {var}.. I missed something there..
17:49.21CunningPikerene-: Probably victim. [TK]D-Fender is the sadistic cousin ;)
17:49.26rene-heh
17:49.32[TK]D-Fenderpaolob : And you need to call a function inside of ${}
17:50.25[TK]D-Fenderpaolob : And what are you trying to strip a variable # of chars off an exten?
17:50.33paolob[TK]D-Fender, I want to use it to strip away a code (like #123545) of variable lenght. The code is used to permit international calls only to the persons that receive the code
17:50.47Dr-Linuxdlynes_home: around?
17:51.25[TK]D-Fenderpaolob : pastebin your code segment (everything related)
17:51.34paolob[TK]D-Fender, for example I set the code at "#567", and I give it to the person I permit to make itnl calls, marking #567 to be able to make the code
17:52.06[TK]D-Fenderpaolob : Show me exactly how you're going to dial it.
17:53.21EgonisFYI: I asked for help earlier about IRQ issues with my TDM400P, I added 'noacpi, acpi=off, and routeirq' to my kernel options, and disabled assign IRQ to VGA Controller in the bios -- and  bingo.. works like a charm
17:55.00*** join/#asterisk ToyMan (n=stuq@74-32-6-50.dsl1.mdl.ny.frontiernet.net)
17:55.03paolob[TK]D-Fender, http://pastebin.com/718909
17:55.35*** part/#asterisk Egonis (n=Egonis@207.245.14.10)
17:59.38*** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1)
17:59.38*** part/#asterisk kshumard_ (n=kshumard@gateway.digium.com)
18:02.13paolob[TK]D-Fender, I added the result in the CLI
18:02.36*** join/#asterisk dec_ (n=tom@ppp147-40.lns3.adl2.internode.on.net)
18:02.55Gabriel25Jun 19 14:04:41 ERROR[6260] chan_zap.c: Unable to load config zapata.conf
18:02.55Gabriel25Jun 19 14:04:41 WARNING[6260] loader.c: chan_zap.so: load_module failed, returning -1
18:02.55Gabriel25Jun 19 14:04:41 WARNING[6260] loader.c: Loading module chan_zap.so failed!
18:03.00thockwhat would be the first thing to look for when you get a channel unavailable message?
18:04.55rene-~rene
18:04.56jbot[rene] always happy to learn new words! what's antsy?
18:05.08rene-~rene-
18:05.10jbotit has been said that rene- is THE MAN, or vain
18:05.29rene-jbot, rene- is not vain
18:05.30jbot...but rene- is already something else...
18:05.52rene-damn
18:09.45tzafrir_laptopjbot, no, rene- is just a nick floating around on #asterisk
18:09.47jbotokay, tzafrir_laptop
18:10.51tzafrir_laptopGabriel25, touch /etc/asterisk/zapata.conf
18:11.10tzafrir_laptopalternatively, maybe this is a permissions problem?
18:11.31tzafrir_laptop(touch: in case it didn't exist)
18:12.38*** join/#asterisk h3x (i=hex@ip70-189-236-254.lv.lv.cox.net)
18:12.41PakiPenguinjbot, tzafrir_laptop's company makes astribanks
18:13.18PakiPenguinah not listening to me
18:17.04Corydon-wjbot:  astribank is <reply>Ask tzafrir about the Astribank.  I dunno anything about it.
18:17.05jbotokay, Corydon-w
18:17.48Corydon-w~botsnack
18:17.48jbotCorydon-w: aw, gee
18:17.52*** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie)
18:28.06*** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx)
18:28.32*** join/#asterisk gandhijee (n=gandhije@mail.win-ent.com)
18:28.43gandhijeehas anyone check the dlfnc.c file?
18:28.57*** join/#asterisk h0 (n=h0@ool-44c69453.dyn.optonline.net)
18:29.24gandhijeethe static int isValidStatus(struct dlstatus *status) looks kinda jacked
18:34.21[TK]D-Fenderpaolob : You are using a constant, not a variable, so why not just use a constand for your trim as well.  And you aren't calling the LEN function properly.
18:38.48gandhijeei should say it seems jacked for x-compiles
18:38.58*** join/#asterisk C4T3l (n=robert@cpe-24-175-57-117.houston.res.rr.com)
18:39.22[TK]D-Fenderpaolob : You also seem to not have a "1" priority for that exten... another "no-no"
18:39.32*** join/#asterisk zeppelin_ (n=zeppelin@201-35-79-132.paebv700.dsl.brasiltelecom.net.br)
18:40.21*** join/#asterisk clive- (n=pirch@dsl-145-34-91.telkomadsl.co.za)
18:41.38clive-is " PCadach" here ?....under another Nick
18:42.49clive-found him
18:46.01*** join/#asterisk TheMonoTone (n=tburdick@unaffiliated/themonotone)
18:46.21TheMonoToneso I've been trying out various echo cancellation methods and they all seem to get rid of the echo
18:46.24TheMonoTonebut are adding noise
18:46.35TheMonoToneis there some setting I should be adjusting in zapata.conf ?
18:46.51TheMonoToneI'm using a pair of zaptel tdm400 cards
18:49.42SplasPoodit'd be nice to be able to do ifAppisLoaded(app_milliwatt.so,something,something)
18:49.44feldTheMonoTone: look into the fxotune tool to "tune" your ports to the analog lines and also try the echolearning or whatever it is
18:50.01feldecholearning is a setting in zapata.conf i believe
18:50.04*** join/#asterisk angom_w (n=angom@red-corp-201.130.138.29.telnor.net)
18:52.27*** join/#asterisk tbacevic (n=tbacevic@MTL-ppp-156239.qc.sympatico.ca)
18:53.38TheMonoTonefeld: I'll check in to fxotune first
18:54.48*** join/#asterisk Beighto (n=chatzill@64.160.113.130)
18:55.09vader--do you guys know if there is a sound file that comes with asterisk that says please record your message?
18:55.41dlynes_officevader--: it'd be /var/lib/asterisk/sounds/vm-something-or-other.gsm
18:55.48[TK]D-FenderSplasPood : There are ways.... not elegent though...
18:56.23SplasPood[TK]D-Fender: yea...
18:56.56*** join/#asterisk tbacevic (n=tbacevic@MTL-ppp-156239.qc.sympatico.ca)
18:59.13*** join/#asterisk flynux (n=flynux@2a01:38:0:0:0:0:0:1)
19:01.12*** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com)
19:02.15*** part/#asterisk mog (i=ejabberd@68.62.237.103)
19:05.24*** join/#asterisk mog (i=ejabberd@68.62.237.103)
19:05.47*** join/#asterisk Navire (n=navire@200164021182.user.veloxzone.com.br)
19:06.29NavireSomeone how I test SIP Post Dial delay?
19:10.08*** join/#asterisk roche (n=roche@crsj-dc1-fw001.accuhosting.com)
19:15.06dlynes_officeHas anyone encountered a problem whereby all incoming iax2 calls and all incoming zap calls are not passing audio?
19:15.20dlynes_officeI can't seem to figure out the problem, for the life of me
19:15.31dlynes_officesip to sip is working just fine
19:15.49dlynes_officeThe iax2 is set up exactly the same way I have it on another box, that's working just fine
19:15.55ghenrydoesn't sip pass audo via rtp?
19:15.59dlynes_officeyep
19:16.11ghenryso they do audio indepant of *
19:16.26dlynes_officeyeah?
19:16.46ghenrymaybe your sound card then
19:16.50TheMonoToneugh, why is there so much noice in this phone
19:16.57TheMonoTonea regular phone line doesn't have this much noice
19:16.58TheMonoTone*noise
19:17.00dlynes_officeghenry: sound card?  huh?  I'm not even using a soundcard
19:17.21*** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl)
19:17.23dlynes_officeghenry: it's two sip hardphones
19:17.36ghenryah right
19:17.50*** join/#asterisk shaun2222 (n=ndci@ip68-5-63-223.oc.oc.cox.net)
19:18.00shaun2222any of you guys used the linksys WRT54GP2
19:18.16shaun2222does it require vontage or can i use it to connect via sip to my asterisk server
19:18.17ghenrydo you only need a sound device for ivr?
19:19.01dlynes_officeghenry: you only need a sound device for paging
19:19.20dlynes_officeghenry: i.e. for chan_alsa, chan_oss, and I think chan_phone as well
19:19.30ghenrypaging?
19:19.30dlynes_officeghenry: chan_phone is a softphone built into asterisk I think
19:19.36ghenryah
19:19.39dlynes_officeghenry: yeah...paging out over a loudspeaker
19:19.41Dr-Linuxdlynes_home: why this now? >> Jun 19 12:18:22 WARNING[13423]: loader.c:554 load_modules: Loading module chan_zap.so failed!
19:19.45ghenrygotya dlynes_home
19:23.19dlynes_officeDr-Linux: i have no idea...you're not showing me the whole error
19:23.37thockDr-Linux: pastebin your zaptel.conf? and possibly the last bits of /var/log/asterisk/full ?
19:23.51dlynes_officeDr-Linux: and zapata.conf
19:23.55thockthat too
19:24.25Dr-Linuxdlynes_office: thanks, i figured out
19:24.37dlynes_officeDr-Linux: jumped the gun?
19:24.42Dr-Linuxdlynes_office: now it showing my 2 ports "OK"
19:24.46thock[TK]D-Fender: can i pick your brain a bit re: sangoma stuff
19:26.51*** join/#asterisk blaylock (n=seth@snap.helixsystems.com)
19:27.08Dr-Linuxdlynes_office:
19:27.09Dr-LinuxDescription                              Alarms     IRQ        bpviol     CRC4
19:27.09Dr-LinuxT2XXP (PCI) Card 0 Span 1                OK         0          0          0
19:27.09Dr-LinuxT2XXP (PCI) Card 0 Span 2                OK         0          0          0
19:27.36Dr-Linuxdlynes_office: i skiped the 2nd span
19:28.04thockDr-Linux: is ztdummy running?
19:28.40dlynes_officethock: he's not having an issue now....he figured it out
19:28.52thockoh.
19:28.55Dr-Linuxthock: lsmod doesn't show "ztdummy" is it not running right?
19:29.07Dr-Linuxdlynes_home: noooooooo
19:29.16thockwhat was the original problem?
19:29.34Dr-Linuxi just figured out the error after changing the zaptel.conf structure
19:30.07Dr-Linuxthock: when i dialout, i get busy congested
19:30.16thockAh.
19:30.24thockI'm having that same problem with my LDT1
19:30.47thockbut it's probably the fault of my insessent newbitizim editing my dialplan :<
19:31.09Dr-Linuxthock: mine is not dialplan issue,
19:31.36Dr-Linuxlooks like my pri lines are connected, but my telco didn't put the DID's or D channels stuff yet :S
19:31.48thockDr-Linux: open up the CLI and type pri show span #
19:31.51thockand then the span of your PRI
19:32.37Dr-Linuxthock: that shows down, active
19:32.39Dr-Linuxlemme show ya
19:32.42thockpm it
19:33.39Dr-Linuxok
19:34.29*** join/#asterisk beyond (n=beyond@200.192.160.100)
19:37.16*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
19:39.02*** part/#asterisk clive- (n=pirch@dsl-145-34-91.telkomadsl.co.za)
19:41.34*** join/#asterisk pagec (n=cpage@h-74-0-107-178.nycmny83.covad.net)
19:42.18pagecwhat if i want to show if person x is on the phone to phone y?
19:45.21*** join/#asterisk KranZ (n=user@imail.bestline.net)
19:45.23*** part/#asterisk KranZ (n=user@imail.bestline.net)
19:45.25*** join/#asterisk KranZ (n=user@imail.bestline.net)
19:46.15*** join/#asterisk Nugget (i=nugget@dazed.slacker.com)
19:47.26*** join/#asterisk nagl (n=nagl@86.59.54.237)
19:51.57*** part/#asterisk Navire (n=navire@200164021182.user.veloxzone.com.br)
19:55.00*** join/#asterisk zotz (n=zotz@24.244.133.115)
19:59.03*** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net)
19:59.20*** join/#asterisk hads|home (n=hads@mail.nice.net.nz)
20:00.14*** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
20:00.16*** join/#asterisk dacleric (n=dacleric@p548227D3.dip0.t-ipconnect.de)
20:04.18JoseBravoIm triying to do a call from 70103 tp 71462 extension. But its my CLI output: http://pastebin.com/719296 any idea?
20:06.58Bullseye_NetworkI cant get anything to come up on pastebin.com hmmm...
20:07.25*** join/#asterisk mrbnet_ (n=sureal@cust-static-blk197-45.BHI.COM)
20:07.42*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)
20:08.17*** join/#asterisk d-tech (n=dtc@72.245.233.107)
20:08.57TripleFFFFcan we have playback use mp3 ?
20:11.32*** join/#asterisk Skumlos (n=youl@nat.kollegienet.dk)
20:11.59Skumlosmjallo
20:15.10*** join/#asterisk mtaht4 (n=m@reserve-64-79-114-30.wiline.com)
20:16.00TripleFFFF>??
20:19.37*** join/#asterisk nortex (n=nortex@ama-wldhcp.696130103.amaonline.com)
20:20.46*** join/#asterisk MedozasSVR (n=MedozasS@p549B8087.dip0.t-ipconnect.de)
20:20.55MedozasSVRHi guys!
20:21.17MedozasSVRIs anybody here with knowledge of asterisk realtime?
20:21.46*** join/#asterisk vivek (n=vivek@unaffiliated/tintin)
20:22.11TripleFFFFyyeah
20:22.34vivekhello all, can i use tor & privoxy on a linksys router to get across the voip blocks put up by some countries like uae ?
20:23.04LoReznot if you want to actually use VoIP
20:23.43vivekhmmz ok i don't get why though ...
20:23.50MedozasSVRtripleffff: you mean you have knowledge with asterisk realtime?
20:24.41JoseBravoHow I see the agi debug?
20:24.56MedozasSVR@jose bravo: agi debug
20:25.14MedozasSVRthats the command
20:25.38vivekLoRez: lets say i have a spa-2k connected the the linksys router and a gizmo or some other voip provider. and linksys passes all the info via the http port ... it won't work ?
20:26.35MedozasSVRi have varoius questions regarding it:
20:26.35MedozasSVR1. how about support for "s" and "i" extensions within realtime - are they now builtin to 1.2.9.1 or still only availiable via SVN?
20:26.35MedozasSVR2. are there any implementations regarding conference rooms (meetme.conf) in realtime except from asterisk realtime static (like sip table or so)?
20:27.02dlynes_office[TK]D-Fender: found out the problem...there was a stale socket file for wanpipe in /var/run
20:27.05TripleFFFFno idea
20:28.26dlynes_office[TK]D-Fender: btw...for future reference, it's like wan_ec_config wanpipe1 w1g1 stats
20:28.42[TK]D-Fenderdlynes_home : Thought so.. I did mention that to you a while ago.. from a lockup quite likely
20:28.57dlynes_office[TK]D-Fender: ah...didn't see you mention the stale file
20:29.05*** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com)
20:29.06dlynes_office[TK]D-Fender: i got that from David
20:29.09[TK]D-Fenderdlynes_office : That was like 2 weeks ago :)
20:29.12dlynes_officeoh
20:29.13dlynes_officehahaha
20:29.22dlynes_officei was having problems with sangoma in general then
20:29.24[TK]D-Fenderdlynes_home : Before I got stage 1 working for you :)
20:29.37dlynes_officeso i wasn't even looking at ec at that point
20:30.14[TK]D-Fenderyeah... echo would be a sign of life back then :)
20:30.21dlynes_officeexactly
20:30.32dlynes_officeanyways...it's up and running now, with hwec enabled
20:30.45dlynes_officeI guess when it says it's using mg1 echo canceller, you can ignore that, right?
20:31.07dlynes_officeas long as the hwec reports back correctly?
20:31.40justinu|laptopi'd guess you'd want to disable any software EC in zaptel
20:31.49[TK]D-Fenderdlynes_home : exactly... the Otasic kicks in when the wanpipe drive sees the call for EC.
20:31.55*** join/#asterisk tsurk0 (n=tsurko@85.187.160.157)
20:32.29[TK]D-Fenderdlynes_office : which is why you need to specify "echocancel=yes", not nothing regarding echotraining, and you'll never have to screw around with gains again.
20:33.33[TK]D-FenderZaptel EC never really gets called.  Wanpipe sees to that (a change in the more modern firmwares)
20:34.34justinu|laptopcool
20:34.46[TK]D-Fenderok, heading home, back in a few.
20:35.56MedozasSVRanother question: are there any tags for noise reduction for sip and/or zap?
20:36.16MedozasSVRto reduce background noise
20:36.45dlynes_officeMedozasSVR: for sip, you can replace your phone
20:36.57dlynes_officeMedozasSVR: for zap, you can try adjusting the gains
20:37.04dlynes_officeMedozasSVR: also for sip, you can try a different codec
20:37.13justinu|laptopor replace those irritating CPU fans with quiet models
20:38.10*** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net)
20:38.45MedozasSVRphone: a good cisco, no way :) --- codec is 729 with a 1 m/bit line with traffic shaping (module hsfc)
20:39.38*** part/#asterisk terrapen (n=cjs@166.70.183.108)
20:39.43MedozasSVRwell its actually not because of cpu fans - much more because other people talking in the background, and some people are irritated by that
20:40.09justinu|laptophmm, mic is too sensitive?
20:40.31justinu|laptopor you txgain on your zap channels is too high
20:40.37justinu|laptopif you're doing sip->zap calls
20:40.49MedozasSVRits same with sip>sip
20:41.23justinu|laptopin that case you can only blame the phones, since asterisk is just forwarding rtp, not modifying it
20:41.26MedozasSVRi know the 7960 can be changed a bit in its sensitivity, but are there any tags for asterisk? would save time....
20:41.37*** join/#asterisk GarethTheGreat (n=gareth@unaffiliated/gareththegreat)
20:41.39justinu|laptopno, asterisk doesn't modify the rtp packets unless it's transcoding
20:41.52MedozasSVRdarn, thanks!
20:41.54justinu|laptopnp
20:42.02GarethTheGreatanyone here had problems building on Fedora Core 4?
20:42.36MedozasSVR@justinu: as you seem to know a lot - do you know things of asterisk realtime? can you help me there?
20:42.57GarethTheGreat/usr/bin/ld: cannot find -lssl
20:43.03GarethTheGreatgetting that when trying to make install
20:43.09GarethTheGreatthough openssl is installed alright
20:43.11MedozasSVRyou should install openssl
20:43.17MedozasSVRopenssl-devel
20:43.34justinu|laptopyep, you'll need a lot of *-devel packages perhaps
20:43.48MedozasSVRi can give you an exact listing - one moment
20:43.51GarethTheGreatcompiling now
20:45.14MedozasSVRgcc, ncurses-devel, openssl-devel, patch (not neccessarily), bison, zlib-devel, kernel-source, kernel-syms
20:45.33MedozasSVR@gareth: did it work now?
20:45.48sevarddamnit
20:46.05sevardIf somebody enters 4000# how would I strip off the #
20:46.27MedozasSVR${EXTEN:-1}
20:46.42MedozasSVRsorry, wrong
20:46.42sevardi've already tried that
20:46.43sevardhmm
20:46.44*** join/#asterisk MarcPtz (n=MarcPtz@18.Red-80-35-146.staticIP.rima-tde.net)
20:46.58justinu|laptopsevard: get that ATA?
20:46.58MedozasSVR${EXTEN:0:4}
20:47.09MedozasSVRif 4 digits
20:47.22sevardjustinu|laptop: Yes I did, did you catch my email from yesterdayand the one from today?
20:47.30sevardMedozasSVR: what if it's an undefined amount od digits
20:47.37sevards/od/of/g
20:48.01MedozasSVRhmm
20:48.13MedozasSVRthen it should be able to work with LEN
20:48.38justinu|laptopservard: not until just now :)
20:49.00justinu|laptopso you got it working well?
20:49.02justinu|laptopthat's awesome
20:49.10sevardlike ${EXTEN:$LEN:{MATH:LEN-1:}}
20:49.13sevardor something crazy?
20:49.19sevardjustinu|laptop: yeah :D:D
20:49.23justinu|laptopwerd
20:49.28sevardjustinu|laptop:I played with it for a couple of hours
20:49.32MedozasSVRim testing at one system at the moment
20:49.42MedozasSVRgive me some minutes, ok?
20:49.52sevardright on
20:50.07sevardjustinu|laptop: if you come across another one I can show you how to do it
20:50.31justinu|laptopi will take you up on that
20:50.48*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
20:51.18sevardjustinu|laptop: apparently it will take a day or two for funds to be transfered, my first time using paypal ;/
20:51.33sevardjustinu|laptop: i'm really appreciative though man, it's pretty awesome of you.
20:52.14MarcPtzHi all , one question , when working with a2billing all generated calls are by default redirected to context [from-sip-external] ?
20:52.43justinu|laptopit's ok, i'm not waiting on your money to eat or anything :)
20:53.05sevardhehe
20:53.06justinu|laptopjust glad it has a good home :)
20:53.22sevard:D it's awesome
20:55.43Qwell[]justinu|laptop: what, no guilt trip?
20:56.06sevardjustinu is a good guy
20:56.17Qwell[]bah! :p
20:56.33justinu|laptopnah, i wouldn't guilt him even if he didn't pay... life's to short to be pissed off over 12 bucks
20:56.43Qwell[]oh, heh
20:56.50justinu|laptopbut i wasn't gonna tell him that before he paid :)
20:56.57*** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx)
20:57.50rene-what is the deal with realtime agents? is it something that will be available for * 1.4? is it available for head?
20:58.11Qwell[]rene-: I believe it's in trunk..which means it'll be in 1.4
20:58.50justinu|laptopbe a man, backport it to 1.2
20:58.59mogrighttttttttttttt
20:59.03Qwell[]real men backport to a pre 2005 cvs
20:59.19Qwell[]mog: Did you see, did you see?!
20:59.22justinu|laptopheh
20:59.28mogwha Qwell ?
20:59.30justinu|laptopi backported res_snmp to 1.2
20:59.32Qwell[]pfft
20:59.34justinu|laptoptook about 30 minutes
20:59.36Qwell[]didn't even see...
20:59.42Qwell[]mog: skinny in trunk :p
20:59.48mogOH that
20:59.50mogyes i did
20:59.55Qwell[]:D
21:00.45rene-thx
21:01.09justinu|laptopso who's running gentoo?
21:01.15sevardnobody important
21:01.17justinu|laptoplol
21:01.19Qwell[]justinu|laptop: You know I do :p
21:01.20sevard:)
21:01.25sevardi rest my case.
21:01.26Qwell[]why?  because that's what real men run!
21:01.33justinu|laptopgentoo seems pretty slick
21:01.46Qwell[]mog: I don't have access to the "other" file ;)
21:01.52CunningPikejustinu|laptop: yes - oil slick ;)
21:01.57justinu|laptoplots of good docs for running on laptops
21:01.58*** join/#asterisk Spy000007 (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net)
21:02.01justinu|laptoplots of control
21:02.18Qwell[](CREDITS, that is)
21:02.21mogohhh
21:02.22mogyeah
21:02.49fileeh?
21:02.52Qwell[]I wish he did too
21:02.58MedozasSVR@sevard: hmmm - tricky
21:03.11*** join/#asterisk MatsK (i=MatsK@83.233.97.229)
21:03.22MedozasSVRyou can make one thing: match with if or gotoif
21:03.32MedozasSVRis better anyways
21:03.57MedozasSVRso you can match with ${EXTEN:-1:1} if this is '#'
21:04.11MedozasSVRand use if or gotoif as case what to do
21:10.48*** join/#asterisk alystair (i=Alystair@CPE001109c15241-CM00407b8794db.cpe.net.cable.rogers.com)
21:11.15Poincareanyone knows why 'call-limit' won't work with some sip-providers?
21:11.53generalhanPoincare: what ver. of * you using ?
21:12.16Poincaregeneralhan: Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1q
21:12.37Poincareit works fine with a test account on a friends *
21:13.00Poincarebut fails when I try limit calls one of those sip-providers
21:13.40*** join/#asterisk X-Gen (n=X-Gen@dsl-145-254-10.telkomadsl.co.za)
21:14.27generalhanPoincare: in your sip.conf are your phone entried listed as "friend" or "peer" ?
21:15.05Poincarepeer, it aren't phones but accounts on another server
21:16.11generalhanPoincare: im not familiar with this issue ... but read here and see if this seems to be your issue   http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+incominglimit
21:16.40Poincarealmost know that page by hard :-)
21:16.42generalhanPoincare: all the way to the bottom they use a group function to make it work
21:17.08generalhanare you trying to only allow one or so calls at a time through one provider ?
21:17.10sevardI have a question about phone.conf
21:17.14sevardDo we really even need it?
21:17.38*** join/#asterisk Lino` (n=Lino@i577BD2A6.versanet.de)
21:17.43Poincarethat's the idea yes, to make sure i don't make a second outgoing call through a provider
21:18.22alystairanyone here heard of Comwave?
21:21.23*** join/#asterisk zeppelin_ (n=zeppelin@201-35-78-44.paebv700.dsl.brasiltelecom.net.br)
21:21.54*** join/#asterisk Mattwj2005 (n=Matt@user-12l3n74.cable.mindspring.com)
21:22.24Mattwj2005hey guys I just wanted to let you guys know about something that I discovered
21:23.07Mattwj2005one of the programs that comes with mythtv, mythfrontend conflicts with Asterisk....that is why I was getting all those errors before
21:25.07*** join/#asterisk los415 (n=los415@sfca-office.corp.race.com)
21:33.06*** join/#asterisk Corydon-w (n=tilghman@pdpc/supporter/sustaining/Corydon76-home)
21:35.04*** join/#asterisk rollot (n=rollotom@c-68-37-168-10.hsd1.pa.comcast.net)
21:35.44*** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin)
21:37.04justinu|laptophey paki, ltns
21:37.15*** join/#asterisk smackus (n=smackus@63.149.122.94)
21:37.44*** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka)
21:38.02smackusam i going to have any issues using asterisk and zaptel on a 64 bit machine? I will be running Red Hat Enterprise
21:38.04dlynes_officesevard: i think that's for using the cli 'dial' application
21:40.16*** part/#asterisk rollot (n=rollotom@c-68-37-168-10.hsd1.pa.comcast.net)
21:43.26*** part/#asterisk smackus (n=smackus@63.149.122.94)
21:45.06*** join/#asterisk flujan (n=flujan@internet.nube.com.br)
21:46.08flujanguys, I'm trying to use chan_spy and monitor a specific channel... is it possible? I want do record the conversation and chan_spy the channel at the same time.
21:47.35*** join/#asterisk existx (i=existx@sniff.ttyp.net)
21:48.39*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)
21:49.17*** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
21:51.18redder86Hi.  We were using Asterisk 1.0 up until a few weeks ago at which point I upgraded to 1.2.7.1 (and now 1.2.9.1).  After doing so we started having some problems where the call audio would be grossly distorted.  I'm ready to revert the installation to use 1.0 again, but I would be delighted to work with someone to root-out the problem.
21:51.54dlynes_officeare you using zaptel?
21:54.17redder86yes
21:54.28X-Rob?centosbug
21:54.33Qwell[]~
21:54.40X-Rob~centosbug
21:54.41jbot[centosbug] a problem with the latest Centos kernels (4.2 and 4.3).  To fix it, paste everything inside the quotes into a root shell:  "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h"
21:54.41redder86The OS is Linux 2.4.20 on RedHat 9.
21:54.53X-Robeven
21:54.58Qwell[]heh
21:55.08Qwell[]I love how those get incrementally more complex
21:55.32redder86what is that centos bug?
21:55.36Qwell[]..that
21:55.48X-RobI re-wrote to auto-fix the correct file
21:55.53Qwell[]X-Rob: nice
21:56.01X-Rob<-- sick
21:57.03dlynes_office~redhatbug
21:57.04jbotredhatbug is probably is a problem with the latest RedHat Enterprise Linux and CentOS kernels (4.2 and 4.3).  To fix it, paste everything inside the quotes into a root shell:  "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h"
21:57.54redder86why are we messin' with kernel headers?
21:58.01redder86are the kernel headers faulty?
21:58.05Qwell[]becayse they're broke
21:58.10X-Robyes
21:58.17X-Robspinlocks don't work on RHEL
21:58.26X-Rob(without that fix)
21:58.28redder86and they make zaptel audio quality crappy?
21:58.35Qwell[]no
21:58.37X-Robno, zaptel doesn't complile
21:58.38X-Rob8)
21:58.38Qwell[]they make zaptel not compile
21:58.45redder86zap compiles fine
21:59.53redder86ls: /usr/src/kernels/2.4.20-8smp-i686/include/linux/spinlock.h: No such file or directory
22:00.18dlynes_officeredder86: what do you get when you run zttest?
22:00.18X-Robyou're not running centos 4.2 or 4.3
22:00.29redder86I didn't say when I run zttest.
22:00.38dlynes_officeredder86: are you getting any %'ages lower than 99.975?
22:01.46redder86--- Results after 27 passes ---
22:01.46redder86Best: 100.000000 -- Worst: 99.975586 -- Average: 99.989601
22:02.10MikeJ[Laptop]greetings redder86...
22:02.12redder86We're using Asterisk as a PRI bridge between the telco and a Patton 2977 for faxing
22:02.16redder86MikeJ: hi
22:02.39dlynes_officeredder86: are you sharing any interrupts?
22:02.54redder86after upgrading to 1.2 Asterisk will make the audio in the fax call turn very ugly about 1 in 10 calls ... it varies.
22:03.26dlynes_officeredder86: and did you let zttest run for 2 or 3 minutes?
22:03.29redder86<PROTECTED>
22:03.29redder86<PROTECTED>
22:03.30redder86<PROTECTED>
22:03.30redder86<PROTECTED>
22:03.30redder86<PROTECTED>
22:03.30redder86<PROTECTED>
22:03.32redder86<PROTECTED>
22:03.34redder86<PROTECTED>
22:03.34dlynes_officeredder86: also, have you run patlooptest?
22:03.36redder86<PROTECTED>
22:03.38redder86<PROTECTED>
22:03.40redder86NMI:          0          0
22:03.42redder86LOC:   92637658   92637657
22:03.44redder86ERR:          0
22:03.47redder86MIS:          0
22:03.48redder8627 passes is much less than 2 or 3 minutes
22:03.49dlynes_officeummmm
22:03.52redder86but remember ... things were fine with 1.0
22:03.54dlynes_office~pb
22:03.55jbotit has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/
22:04.36redder86what is patlooptest?
22:04.42dlynes_officeyeah...things were fine for me with 1.0, too
22:04.44brad_msswwhat version of the zaptel driver are you using ?
22:04.46dlynes_officebut 1.0 was next to useless
22:04.59redder86zaptel-1.2.6
22:05.03robin_szredder86, probably a pathlooptest spelt wrong
22:05.06dlynes_officepatlooptest is another utility that doesn't get compiled by default with zaptel
22:05.34dlynes_officeit runs a pattern loop test on each span of your pri crd
22:05.36dlynes_officeit runs a pattern loop test on each span of your pri card
22:06.03dlynes_officeyou need a loop module connect to each span you want to test
22:06.25*** part/#asterisk mog (i=ejabberd@68.62.237.103)
22:06.47dlynes_officeand you need to use clear x-n, where x is the min chan number and n is the max chan number for the particular span (make sure you've got the b chans and d chan commented out)
22:06.52redder86dlynes_office: so you had the same kind of problems as I'm having after upgrading to 1.2?  screwed up audio (that begins at the beginning from the time the call is first bridged) on only some small percentage of the calls?
22:06.58dlynes_officethen reissue a ztcfg before running patlooptest
22:07.13dlynes_officeredder86: i had issues with the pri in general after upgrading
22:07.30dlynes_officeredder86: i've traced it down to a bad card that never reared its head in 1.0, but did in 1.2
22:07.45redder86you replaced the zap card and all is well?
22:07.59dlynes_officeredder86: i'm going to be replacing hte card within the next day or so
22:08.07redder86how do you know it's a bad card?
22:08.37dlynes_officeredder86: because i switched spans, and found out span 1 was totally fubar (patlooptest threw up an endless stream of errors on it)
22:09.05dlynes_officeredder86: and span 2, 3 and 4 i get less than 99.875% the odd time on zttest
22:09.43*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)
22:09.55dlynes_officefor all i know, there might not be anything wrong with 2, 3 and 4, but because 1 is bad, it's affecting the other three
22:09.58dlynes_officei don't really know
22:10.04dlynes_officei just know it needs to be fixed
22:10.11Dr-Linuxhi again
22:10.16dlynes_officebye again
22:10.21Dr-LinuxCunningPike: hey
22:10.25Dr-Linuxdlynes_office: bye
22:10.31dlynes_officeDr-Linux: don't disturb him
22:10.34dlynes_officeDr-Linux: he's sleeping
22:10.49Dr-Linuxdlynes_office: with whome? :S
22:10.52redder86why wouldn't the problem have come up in 1.0?
22:10.56dlynes_officeDr-Linux: you know how those government employees are
22:11.03dlynes_officeDr-Linux: they never do any real work :)
22:11.29dlynes_officeredder86: i have no idea, but I can't stay on 1.0, myself
22:11.31Dr-Linuxdlynes_office: i see, i thought Canada guys work hard :P
22:11.34dlynes_officeredder86: it's lacking too many features
22:12.01Dr-Linuxdlynes_home: i got DID's from my telco, but my problem is still there
22:12.02dlynes_officeredder86: so instead, i broke down and bought a sangoma pri card instead
22:12.10dlynes_officeDr-Linux: so phone up your telco and bitch
22:12.26dlynes_officeDr-Linux: are you able to dial in to your asterisk box, using one of the dids?
22:13.28redder86hehe ... well, that's not going to tell you for-certain that the card was defective or whether the fault was a built-in Digium issue
22:13.31Dr-Linuxdlynes_office: when i call on the DID, it do not come to my server
22:13.33Dr-Linuxivr1*CLI> pri show span 1
22:13.35Dr-LinuxPrimary D-channel: 24
22:13.35Dr-LinuxStatus: Provisioned, Down, Active
22:13.42dlynes_officeBECAUSE YOU'RE STILL DOWN
22:14.16Dr-Linuxdlynes_office: yes, is it my configuration fault or telco fault? :S
22:14.18dlynes_officephone up your telco, and tell the idiots that your pri is still down
22:14.46dlynes_officeit looks like it's a problem on their end, to me
22:15.05dlynes_officebut you better hurry up
22:15.13dlynes_officei think their office might have already closed
22:15.30Dr-Linuxdlynes_home: no they are 24/7
22:15.42Dr-Linuxi have opened a ticket to the Datacenter
22:15.45dlynes_officeah...even their business center?
22:15.53*** part/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com)
22:16.02Dr-Linuxdlynes_home: i asked them 4 question in the ticket
22:16.06dlynes_officeah
22:16.15Dr-Linuxquestions
22:16.20dlynes_officeyeah...our crappy telco's business center is only open 9-6
22:16.23dlynes_officem-f
22:16.48*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
22:16.52Dr-Linuxdlynes_office: that's a Datacenter
22:17.11Dr-Linuxdlynes_office: i asked them:
22:17.11dlynes_officeoh...a colo?
22:17.21Dr-Linux1. PRI cables connectvity
22:17.26Dr-Linux2. PRI numbers/DID's
22:17.43dlynes_officeForget all that crap
22:17.48dlynes_officeJust tell them your pri is down
22:18.01Dr-Linux3. Framing and coding (currently i'm using esf,8bzo something)
22:18.16dlynes_officeyeah...you can tell them your framing and coding
22:18.18Dr-Linux4. it's only inbound or outbound as well
22:18.24dlynes_officeDr-Linux: ummm
22:18.32dlynes_officeDr-Linux: what else is there besides inbound and outbound?
22:18.43Dr-Linuxi asked them those 4 question
22:19.06*** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
22:19.09Dr-Linuxdlynes_home: maybe the allow us only inbound calls
22:19.18dlynes_officeDr-Linux: ummm
22:19.23dlynes_officeDr-Linux: YOUR INBOUND IS NOT WORKING, EITHER
22:19.33Dr-Linuxdlynes_home: yes
22:19.39Dr-Linuxdlynes_office:
22:20.42*** join/#asterisk AlexCTI (n=alex@adsl-074-238-025-003.sip.mia.bellsouth.net)
22:20.47Dr-Linuxdlynes_office: they gave me that number beside 22 more numbers
22:21.46*** join/#asterisk philv (n=bleep@cowpig.ca)
22:21.56Dr-Linuxdlynes_home: when i call this number, it says "all circuit are busy please try later"
22:22.03philvAnyone here ever used an i2004 with chan_unistim?
22:22.58AlexCTIHi Dr-Linux, Are you familiar with the varible ${PRIORITY} ?
22:23.31Dr-LinuxAlexCTI: nope, sorry
22:24.17MedozasSVRcan anyone tell me how to turn on debug mode to also see mysql queries been done with asterisk realtime?
22:24.36Dr-Linuxdlynes_office: PRI down/up is also related with framing and coding?
22:24.59dlynes_officeDr-Linux: don't think so, but then again, i don't know enough about pris, either
22:25.30dlynes_officephilv: nope...how well does it work though?
22:25.40dlynes_officephilv: i'd love to have those working with asterisk
22:25.50Dr-Linuxdlynes_home: CunningPike told me that i can't do anything untill my PRI status is UP and green.
22:25.53dlynes_officephilv: but i don't want to spent $600 to find out it's a paperweight, either
22:26.08dlynes_officeDr-Linux: isn't that what I've been telling you ALL DAY, too?
22:26.18philvdlynes_office: Well, I have the i2050 working with no troubles, but my i2004 tries to locate the server, does so, and sends a few bad UDP packets
22:26.31philvdlynes_office: I'm fortunate, I picked it up on fleaBay for 20 bucks ;)
22:26.39dlynes_officephilv: ah...how well does the i2050 work?
22:26.43Dr-Linuxdlynes_office: yes, but i was still thinking that my configuration is wrong.
22:26.48dlynes_officethat's probably why it was $20
22:26.53philvdlynes_office: the i2050 softphone works very well
22:26.54dlynes_officeit probably won't work on a bcm, either :p
22:27.02dlynes_officeoh...it's a softphone, not a real phone
22:27.03philvOh it does :P
22:27.26dlynes_officeIt does sip though too, doesn't it?
22:27.26philvThe i2050 works well on Asterisk, and the i2004 works on a BCM50 no trobules.
22:27.28philv*troubles.
22:27.33dlynes_officeor do you need a super expensive module for sip?
22:27.43philvI think the i2050 doesn't do SIIP
22:27.45philv*SIP
22:27.49philvMCS does.
22:28.11dlynes_officehow much does the i2050 cost?
22:28.13philvActually, MCS is entirely SIP, iirc, with just a few of our special proprietary options tacked on.
22:28.18philvThat I'm not sure of.
22:28.30dlynes_officeah....you've got a pirated copy?
22:28.38philvHaha no no.
22:28.41philvI work for Nortel :)
22:37.17X-Rob~centosbug
22:37.18jbotcentosbug is probably a problem with the latest Centos kernels (4.2 and 4.3).  To fix it, paste everything inside the quotes into a root shell:  "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h"
22:38.56*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
22:41.18Qwell[]philv: heathen ;)
22:42.08MedozasSVRi have one weird thing i have a question for:
22:42.08MedozasSVR200/200                    192.168.254.37   D          5060     OK (6 ms)
22:42.08MedozasSVR100/100                    192.168.254.75   D          1720     OK (150 ms)
22:42.08MedozasSVRSIPGATE_VOIP/1999XXX       217.10.79.9          N      5060     UNKNOWN
22:42.08MedozasSVR301/301                    192.168.250.25   D          18680    OK (132 ms)
22:42.46MedozasSVRfollowing: im sitting about 100 km away from this asterisk box - i have a vpn connection to there
22:43.02*** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca)
22:43.18MedozasSVRmy location is 192.168.250.0/24, the other location is 192.168.254.0/24
22:43.41MedozasSVRboth are completely switched 100 M/bit networks
22:44.41MedozasSVRwhen i ping 192.168.254.75, i get a respone time of (from the other location!) 112 ms
22:45.31MedozasSVRwhy is the response time from within the network itself (192.168.254.75 <> 192.168.254.0/24) slower than it is even through a vpn from completeley somewhere else?
22:45.44MedozasSVRthe switches are fine
22:46.08MedozasSVRPING 192.168.254.75 (192.168.254.75) 56(84) bytes of data.
22:46.08MedozasSVR64 bytes from 192.168.254.75: icmp_seq=1 ttl=128 time=2.41 ms
22:46.08MedozasSVR64 bytes from 192.168.254.75: icmp_seq=2 ttl=128 time=2.42 ms
22:46.51MedozasSVRand even ping from the >same< machine with the listing of peers
22:47.07MedozasSVRis looking fine
22:47.11MedozasSVRanybody can help?
22:47.32X-Robsounds like the phone is crappy
22:48.53MedozasSVRhmmm - but what does asterisk use to measure this?
22:49.01MedozasSVRnot a simple ICMP?
22:49.08*** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane)
22:49.08MedozasSVRor SIP response time?
22:49.28X-Robsip response
22:49.39MedozasSVRthanks
22:52.18*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)
22:52.31*** join/#asterisk sumdingwong (n=sumdingw@adsl-66-137-178-195.dsl.spfdmo.swbell.net)
22:53.46sumdingwongDoes anyone know how asterisk knows to send mwi light off?
22:54.33*** join/#asterisk dant (n=dan@84.9.188.2)
22:55.12sumdingwongis there anyone else on this channel?
22:55.33*** part/#asterisk sumdingwong (n=sumdingw@adsl-66-137-178-195.dsl.spfdmo.swbell.net)
22:58.54MedozasSVRhere
23:01.45*** join/#asterisk harpermood (n=harpermo@24-180-0-138.static.snlo.ca.charter.com)
23:02.12harpermoodI have a question, regarding my TE110P card.
23:02.16harpermoodI can dial out...
23:02.25harpermoodbut when I try to dial in, I get all busy trunks.
23:02.39harpermoodthe Telco guy, whom I trust, says everything works up to my card..
23:02.49harpermoodzttool shows no change to any of those bits.
23:03.05harpermoodhow can I get low level debug information from the driver or card to see what signales I have.
23:03.32Dr-Linuxquestion: i'm doing "zttest" it's doing and doing, so should i stop it manually or it will be stopped by it self?
23:04.05*** join/#asterisk hads (n=hads@mail.nice.net.nz)
23:04.47*** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com)
23:13.57*** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net)
23:14.30RoyK<PROTECTED>
23:15.13*** part/#asterisk droops (n=droops@adsl-065-005-212-128.sip.jan.bellsouth.net)
23:25.33*** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com)
23:28.25*** join/#asterisk Eggplant (i=No@dsl-72-19-44-253.cascadeaccess.com)
23:28.27*** join/#asterisk Beighto (n=chatzill@64.160.113.130)
23:29.52harpermoodWhat does it mean when my Card doesn't return a wink to the telco?
23:31.00Qwell[]harpermood: that the telco needs to buy it another beer
23:31.33*** join/#asterisk Beighto (n=chatzill@64.160.113.130)
23:33.16*** part/#asterisk Beighto (n=chatzill@64.160.113.130)
23:36.37*** join/#asterisk MoutaPT (n=MoutaPT@a83-132-239-109.cpe.netcabo.pt)
23:37.01MoutaPTNon-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event)                                              , combined - 0x1 (telephone-event)
23:37.18MoutaPTi'm getting this msg in sipdebug with voip provider
23:37.31MoutaPTany one has any idea what might be wrong?
23:37.33harpermoodany idea why my card wouldn't return the wink to the telco (and I have had plenty of beer ;)
23:37.54fileMoutaPT: why do you think there is something wrong?
23:38.03*** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net)
23:38.29MoutaPTbecause after that i get congestion message
23:38.30*** join/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au)
23:38.40MoutaPTCHANUNAVAILable
23:38.45P-NuTHi all.
23:38.49filewell pastebin the complete sip debug and console output
23:38.57MoutaPTok brb
23:38.57CunningPikeDr-Linux: ctrl-C
23:39.18Dr-LinuxCunningPike: already done
23:39.30Dr-Linux--- Results after 918 passes ---
23:39.31Dr-LinuxBest: 99.987793 -- Worst: 99.975586 -- Average: 99.976639
23:39.41P-NuTCan anyone recommend a cheap 2 port FXO gateway for asterisk?
23:39.55Dr-LinuxCunningPike: someone told me it's bad result
23:40.14CunningPikeDr-Linux: It's not that bad.........
23:40.34CunningPikeDr-Linux: It's only a bad result if you are having problems
23:40.37Dr-LinuxCunningPike: i see
23:41.07Dr-LinuxCunningPike: i'm having problems, but not sure what's down :S
23:41.37CunningPikeDr-Linux: Your PRI is. The problems I was referring to are call quality problems, not a Down ORI
23:41.48CunningPikes/ORI/PRI/
23:42.29Qwell[]P-NuT: tdm400p
23:44.08*** join/#asterisk iq|mobile (n=iq@71-215-58-212.omah.qwest.net)
23:45.52*** join/#asterisk rikstah (n=quirc@213.205.195.254)
23:47.05MoutaPTfile: http://pastebin.com/719777
23:47.16P-NuTthink so?
23:47.23P-NuTdoes it detect without drivers?
23:47.30P-NuTis it widely supported?
23:47.32Qwell[]no, you need zaptel instaled
23:47.34Qwell[]installed
23:47.53MoutaPTa=rtpmap:101 telephone-event/8000 is this event a problem?
23:47.58fileno
23:48.08filethat's DTMF
23:48.13Qwell[]only if you think getting dtmf is problematic
23:48.32MoutaPTno problem with that
23:48.37fileMoutaPT: and you're hearing a congestion?
23:48.43MoutaPTyes
23:48.49fileit's coming from your provider
23:48.50MoutaPTi mean
23:48.52fileor wherever this is going
23:48.53P-NuThmm..
23:49.02philvAny of you kind folks ever use chan_unistim with a Phase I i2004?
23:49.02MoutaPTnot hearing congestion tones
23:49.10P-NuTanybody have 2 x100p's they want to sell me?
23:49.11MoutaPTjus no chan available
23:49.12MoutaPTmsg
23:49.13*** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net)
23:49.18*** part/#asterisk harpermood (n=harpermo@24-180-0-138.static.snlo.ca.charter.com)
23:49.27Qwell[]P-NuT: $75 each
23:49.27fileprovide me with the complete console output too
23:49.31*** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka)
23:49.46P-NuTQwell[]: is that $AUD?
23:49.48MoutaPTbrb
23:49.55Qwell[]P-NuT: USD
23:50.02P-NuThmm..
23:50.12P-NuTDo you have the dodgy asian knock-offs?
23:50.15P-NuTthe clones?
23:50.25Qwell[]That is a clone
23:51.03*** join/#asterisk oceanlan|dustin (n=info@cpe-69-133-109-130.woh.res.rr.com)
23:51.24oceanlan|dustinHey all...I have an outdated version of AAH (Asterisk @ Home)
23:51.33Qwell[]oceanlan|dustin: see topic
23:51.40oceanlan|dustinHow can I update it in the CLI? where should I look?
23:51.46*** join/#asterisk adker (n=adker@70-97-137-155.dsl1.glv.ny.frontiernet.net)
23:51.47Qwell[]in the channel topic
23:51.55oceanlan|dustin? channel topic?
23:51.59Qwell[]type /topic
23:52.13oceanlan|dustini did..
23:52.24*** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net)
23:52.25fileread it
23:52.35oceanlan|dustinnothing is happening..
23:52.54fileFreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx
23:53.07orlockHmm..
23:53.11oceanlan|dustinneat. thanks!!
23:53.15orlockdo Cisco's have a habit of locking up with the sip firmware?
23:53.19oceanlan|dustinwhat is the /topic thing though?
23:53.19P-NuTAUD$100 + postage for a clone!
23:53.21P-NuTdamn.
23:53.23Qwell[]P-NuT: yep
23:53.35Qwell[]or, just slightly more for a "real" fxo
23:53.36rikstahorlock not here
23:53.48P-NuTwouldnt the spa3000 be a cheaper one?
23:53.58Qwell[]P-NuT: two of them
23:54.11P-NuTyeah
23:54.18Qwell[]at $75 each?
23:54.25Qwell[]$75-1000
23:54.28P-NuTtouch?
23:54.33P-NuThmm...
23:54.34Qwell[]touch what?
23:54.38P-NuTLOL
23:54.41P-NuTnever miond.
23:54.50P-NuTbut thanks.
23:54.54P-NuThaha
23:55.27X-Rob~centosbug
23:55.30jbotcentosbug is, like, a problem with the latest Centos kernels (4.2 and 4.3).  To fix it, paste everything inside the quotes into a root shell:  "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h"
23:55.36X-RobBlah.
23:55.36oceanlan|dustin#freepbx is a dead chat...anyone know where I can find a walkthough on upgrading *@home via the CLI?
23:55.44rikstahno apostrophy !
23:55.55X-Roboceanlan|dustin, you know, it does say 'wait'
23:56.39oceanlan|dustinwhat? says wait?
23:56.45X-Robwhen you joined the channel
23:57.18oceanlan|dustinooohhh...I gotcha
23:58.10*** join/#asterisk hads (n=hads@mail.nice.net.nz)

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