irclog2html for #asterisk on 20061223

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00:22.55kink0hello
00:23.16kink0I have this problem: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel cong
00:23.29kink0I have unloade and reloaded the zaptel module, but still the same.
00:23.32kink0any sugestion ?
00:23.55kink0even I did a shutdown -r now, but that did not fix the problem.
00:25.00kink0there any way to reset my TE405 card ? ( remotelly )
00:25.03[TK]D-Fenderkink0 : the person you are calling is busy
00:25.24kink0[TK]D-Fender, I experience a problem with my TE405, not busy really
00:25.41kink0I am calling myself to testing, also all calls returns the same.
00:25.49kink0appears a problem with my TE405 card
00:26.28kink0I did: stop asterik, remove module, reload module... restart asterisk. That did not fix the problem.
00:26.48kink0then, I did a rebot shutdown -r now, that did not fix the problem.
00:26.59Mavviekink0: do incoming calls work?
00:27.04kink0the problem is the same for the 3 PRI ports I have in use
00:27.18kink0Mavvie, I have not tested incoming, are dissabled.
00:27.31Mavviekink0: it would be a good first step to see if that direction works.
00:29.11kink0Mavvie: I think this TE405 card does stupid things, and some time, time ago, I was need to hardware reboot, but at this time I am away from this asterisk
00:29.34kink0I pretend to find anyway to do a full reset on the TE405 card, if possible.
00:29.53Mavvielogin to the ilo, do power off, do power on :-P
00:30.29kink0ilo ?
00:31.57kink0I have REC and RED alarms on PRI :( , even shutdown -r now
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01:16.24ektohey all
01:16.43ektoanyone know anything about TDM400P's?
01:16.53ektoive got a bit of a problem with using 2 in the same server at once
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01:26.27Sline78Is it possible to set the duration of the DTMF tones sent by asterisk as RTP Events (RFC2833)?
01:39.44kink0any idea about how to do near hardware reset on TD405 ?
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01:43.52ektoUsing 2 TDM400P's in the same server seems to be impossible.  Only one shows up at a time 0.o
01:44.19ektoLSPCI shows both, but cat /proc/interupts only lists one
01:46.19ektotoggling APCI doesnt make a difference either
01:46.30ektoand digium havent been a whole lot of help :-/
01:47.44Juggiei might be totally off base but isnt there a thingy on them to assign each card a unique number
01:47.54Juggieto each like an id to each card
01:47.59Juggieor maybe i'm thinking of dialogic
01:48.05ektohmmm
01:48.10ektoyou mean like a hardware switch?
01:48.14Juggieyes
01:48.26ektonot that ive seen, or digium has said anyhting about
01:48.37ektobut that would make sense for them to have an IRQ switch
01:48.38Juggieall my servers have just one card so i might be off base.
01:48.43ektoahh ok
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01:49.03Juggiewhat i'm thinking of isnt an irq switch perse just like an ID for when there are more then one of the same card.
01:49.25ektook, ill have a look and see if i can find one.  Thanks
01:50.22Juggiei might be totally off base so i appologize if i am :)
01:50.55ektolol all good
01:51.10ektoand no i cant seem to find one
01:51.19ektothere are some jumpers on there though
01:51.46ektobut they might be self destruct jumpers for all i know lol
01:52.39Juggiehave you tried different slots?
01:52.45Juggieon the motherboard
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01:56.51ektoyeah
01:56.55ektothe motherboard only has 2 slots
01:56.58ektoand we have 2 cards
01:57.13ektoso ive tried swapping them, rebooting with only one in either slot
01:57.49Juggiecan you force an irq to each slot in the bios?
01:58.17ektonah
01:58.31fileJuggie: ack
01:58.38ektobois has very few options in it, none to do with IRQ's
01:59.32ektomost of the slots on the board are PCI-X so all the configuration options are for those slots
02:00.40jlimbdoes anyone know how to read backtraces?
02:01.34Juggiefile: ping
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02:03.44filepong
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02:14.42ekto*sigh* now neither of the cards show up lol
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02:43.38tracyout of interest (fairly new to gentoo), but is there a gentoo emerge to install freePBX?
02:46.24ManxPowertracy: try asking on #freepbx
02:46.43tracyoh ok, soz didnt know they had thier own channel
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02:58.33horsesgofasterekto:  you still here?
02:59.00horsesgofasterekto: still having problems with 2 tdm400p's?
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03:05.47bkruse_homehello all
03:05.53bkruse_homejbot: file++
03:05.53bkruse_home:]
03:06.36bkruse_homeeveryone checked out the pimped out http://asterisk.org/ i hope
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03:18.29c4t3lhello out there!
03:20.02c4t3ldoes anyone know how to force callers with anon CID to enter their number before accepting calls?
03:23.33horsesgofasterbkruse_home:  wow, I am wide awake now!!!
03:24.47horsesgofasterasterisknow.org...   I need shades...
03:24.47c4t3lanyone
03:25.02horsesgofasterwe're all asleep...
03:25.10c4t3lim looking
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03:25.16c4t3lplz wake up
03:25.20c4t3l:)
03:25.46horsesgofasterI don't have experience w/caller id in that respect... sorry...
03:25.56c4t3ldarn
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03:42.15horsesgofasterok, is everyone quitting?...
03:42.46horsesgofasterhas anyone tried the beta2 iso or live CD from asterisk?....
03:42.47horsesgofaster..
03:43.26horsesgofasterno?....
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04:01.26tracywhat does this mean?
04:01.27tracySTARTING FOP SERVER
04:01.27tracyThis account is currently not available.
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04:03.27Strom_MFOP == flash operator panel
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04:05.05tracyis that part of freePBX or asterisk? i did an emerge asterisk as per the wiki and got that error :(
04:06.50Strom_Mum, it shouldn't be part of regular asterisk
04:06.56Strom_Mit's a third-party addon
04:08.41tracyok, so i must have screwed up something
04:09.34Strom_Mwhat I've always done is used the package manager to handle the dependencies and then installed asterisk from source obtained directly from digium
04:09.35c4t3li think Gentoo screwed up something
04:10.15tracyi seem to have mysql error messages in the logs
04:11.56tracylooks like: MySQL RealTime: Failed to connect database server asterisk on  (err 2002). Check debug for more info.
04:12.09tracyis my error
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04:14.36Strom_Mtracy: which page's instructions did you follow?
04:15.03tracylol, i had 2, let me get them, although they very simple
04:15.10tracyhttp://www.voip-info.org/wiki/index.php?page=Asterisk+Linux+Gentoo
04:15.12tracyand
04:15.25tracyhttp://gentoo-wiki.com/HOWTO_Asterisk#Installation
04:16.12tracyalthough
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04:16.35tracyi have this in ps: /usr/sbin/asterisk -U asterisk
04:16.46tracyso i guess it is running
04:16.48Strom_Mit sounds like you've emerged more than just asterisk
04:17.04tracyStrom_M: i emerged about 10 things :(
04:17.14tracyaddons, and sounds
04:17.25Strom_Mdidn't I tell you to ONLY install asterisk, zaptel, and libpri?
04:17.38tracyyip, i did those
04:17.45Strom_Moh hey, yes I did
04:18.01tracythen i tackled freePBX, and that is a bit of a b1tch to install
04:18.06Strom_Mum
04:18.11Strom_MI told you not to install that
04:18.42tracybut being abit of a noob in asterisk and voip/trunks etc
04:18.50Strom_Mlook
04:18.55tracyand only got it working in trixbox
04:18.57Strom_Mfreepbx isn't going to help you learn shit
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04:19.31Strom_Mit makes a very small number of tasks easier and makes the rest sixty-five times more difficult
04:19.55ektohorsegofaster: back now ;)
04:20.50CunningPikeMY EYES!! MY EYES!!!!
04:21.07ekto*implodes*
04:21.12ektoyeah lol
04:21.15tracyis there any quick check to see if asterisk is running ok?
04:21.18ektoit dont work in firefox too well
04:21.33Strom_Mtracy, type "asterisk -r" at the prompt
04:21.33ektoand the colour is... off..
04:21.35CunningPiketracy: sudo /usr/sbin/asterisk -r
04:22.11tracythanks, looks like it is running
04:22.20[R]urouni:)
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05:07.15buddyMI_what about openser?
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05:08.25TheCopshi
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05:18.03tengulreanybody here?
05:18.13Juggiedependson what your going to ask me
05:19.29TheCopsI have echo problem right now, when I receiv call all is ok, when I'm calling, all is ok, but if I put Dial(Zap/1) without number to get the full dialtone of the line, I have echo. someone know why?
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05:29.16CunningPikeTheCops: The dial tone has echo?
05:29.35TheCopsand when I'm dailing on the line
05:30.24CunningPikeIs it possible that the telco's e/c doesn't kick in until the call is connected? What equipment are you connecting to the PSTN with?
05:34.29TheCopslol I just changed for MG2 module, all my outgoing call are not working, but all incoming are perfect sound
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05:49.46SwKdongs
05:50.03fileSwK: clothes aren't optional in here
05:50.12SwKwrong channel
05:50.19SwKfile: its a bot trigger on another network
05:50.25filemakes sense
05:50.28SwKwhats up
05:50.47fileI'm looking at a boxing day sale flyer :(
05:50.57fileu?
05:51.09SwKsetting up a call router
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05:54.09TheCopsSomeone using Bell canada Voicemail? I have a slutter tone telling me new messages. But now when I have message, I can't do any call. It seem that Asterisk don't recognize this slutter tone I guess
05:56.25CunningPikeTheCops: You may have to play with the indication settings
05:57.00TheCops:/ err how can I get the information about slutter frequencies
05:57.16Qwellslutter tone?
05:58.00TheCopsyeah from Bell
05:59.06Qwellstutter, maybe?
05:59.28TheCopshaha
05:59.31TheCopsyup
06:00.03SwKslutter tone/stutter tone whats the difference
06:13.36shellsharkone letter?
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06:14.25robl^slutter tone?  "Five dollah, have ya hollah!"
06:15.24shellsharki thought slutter tone was more like "oooooh yeah right there... dont stop, dont stop, dont stop" ?
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06:15.59robl^hehehe
06:17.15robl^why is it always holiday weekend when you IAX provider has problems...and everyone is on "holiday leave"
06:17.29shellsharknot us :)
06:17.38shellsharkwe're open 24/7/365
06:17.51robl^shellshark: who are you? ;)
06:18.03robl^er.. who is "us"
06:18.48robl^I may be looking to switch if they can't resolve the issues.
06:19.49shellsharkShellShark Networks -- http://shellshark.net/
06:21.00shellsharkif you have any questions, feel free to email support@shellshark.net
06:21.06shellsharki'm gonna get off IRC for a bit :P
06:21.49robl^'k!  thanks for the info
06:21.53shellsharkno prob
06:22.06robl^I have 12 DIDs I may be moving. ;-)
06:22.55shellsharkwoo :)
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06:30.45fx0that clock is a replica
06:30.53fx0:p
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06:38.19hoobastoobai am looking for a solution to have my callerid name displayed to people i call. I am using nufone.net and they do not subscribe my caller id to a directory listing. are there other alternatives?
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06:46.09hoobastoobais there any way to get my number listed in the directory if my provider does not offer that feature.
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06:56.37Ezsome know a good voip provider us or canada ; wil name and number display
06:56.43Ezwil = with
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06:57.53hoobastoobai use nufone, but they do not offer a directory listing, so your name will not display to people you call.
06:58.07hoobastoobathey are cheap and I have been happy with them except for that one feature.
06:58.27hoobastoobathere are other more expensive services which offer an outgoing name in addition to the number.
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07:10.41AsteriskExperthello everyone
07:11.12AsteriskExpertis there a way to get call-id header of sip message of outgoing call?
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07:15.28AsteriskExpertanybody there?
07:21.49xheliox${SIPCALLID}
07:21.59xhelioxIt's in that variable.
07:23.22xhelioxAnd I assume (pray) that root@host for your IRC ident is actually a joke and you're not running an IRC client as root.
07:25.51brookshirei think it's ${CALLERID(num)}
07:26.29xhelioxThat's not what he technically asked for.
07:27.10xhelioxAnd here I go again, assuming... but an AsteriskExpert would know how to use the ${CALLERID()} variable.
07:27.33brookshireit's actually a function, but whatever :)
07:27.47xheliox;)
07:28.12AsteriskExpertno i can get call-id or originating channel
07:28.16AsteriskExpertwhich made a call
07:28.18AsteriskExpertto asterisk
07:28.24AsteriskExpertbut after i execute dial
07:28.32AsteriskExpertwhich creates new sip request
07:28.38AsteriskExperti want call-id of that
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07:29.02xhelioxhmm.
07:29.38brookshireso you just want the originating channel?
07:30.52AsteriskExpertcall-id of that channel
07:31.24AsteriskExpertthe 2nd leg of call
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07:31.42brookshiredo you need it during the call or after the call?
07:32.06AsteriskExpertafter the call is preferable but anyway will do
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07:41.41AsteriskExpertanybody????
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07:45.33AsteriskExpertpossible to get call-id header of sip request of outgoing call from asterisk?
07:46.04AsteriskExpertI think I can do that by some tweaking in source but I prefer to do it from application level
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07:47.57AsteriskExpertbrookshire: is that possible?
07:48.11AsteriskExpertxheliox: what do u think
07:50.31xhelioxNo ideas on how to do that come to mind..
07:52.08AsteriskExpertbrookshire:what do u say?
07:52.21[TK]D-FenderAsteriskExpert : Yes.  in * CLI "show channel [channel]"
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07:53.38brookshireAsteriskExpert: still looking
07:53.48brookshire[TK]D-Fender: i think he wants it from dialplan
07:53.55brookshireafter the call is completed
07:54.08[TK]D-Fenderbrookshire : I try not to assume anything, and continue handing out rope :)
07:54.11ilan`/win close
07:54.12AsteriskExpertonce Dial is executed
07:54.16*** part/#asterisk ilan (i=ilan@69.60.110.251)
07:54.56[TK]D-FenderAsteriskExpert : How & when do you expect to capture & process this information?
07:55.02AsteriskExpertI want it from dialplan
07:55.27AsteriskExpertit would be better if it can be done on hangup (i.e. at h extension)
07:55.38AsteriskExperti used macro to get it
07:55.41AsteriskExpertit works fine
07:55.46AsteriskExpertbut only if channel answered
07:55.58AsteriskExperti want to get call-id of unanswered channels also
07:56.04AsteriskExpertthats the main thing
07:56.29[TK]D-FenderAsteriskExpert : I seriously doubt it.  The call-out channel no longer exists.
07:56.37AsteriskExpert:D
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07:59.29AsteriskExpertHo I think  i got it
07:59.35AsteriskExpertI'll do one thing
07:59.37AsteriskExpertImportvar
07:59.47AsteriskExpertand channelname and sipcallid will do ;)
08:00.07Strom_Mnow the next challenge is to get you to stop pressing "enter" fifteen times per sentence.
08:00.09AsteriskExpertbut still it won't be available at hangup
08:00.12AsteriskExpertlet me check
08:00.27RouteTableCan anyone point me in the right direction regarding clustering multiple asterisk servers behind OpenSER? (load balanced configuration)
08:00.39AsteriskExpertStrom_M:sorry
08:01.48[TK]D-FenderStrom_M : I ... graduated... from the James..... T Kirk... school..... of overacting!
08:02.24brookshireroutetable: www.google.com
08:02.24Strom_M[TK]D-Fender, so you have a Di  Plo   Ma then
08:02.44RouteTablebrookshire, so you don't know, i guess?
08:02.49Strom_Mbrookshire, I got the hang of my turntables :D
08:03.01brookshireStrom_M: awesome!
08:13.45AsteriskExpertbrookshire: any way to get variable just before hangup of channel?
08:14.05AsteriskExpertbrookshire:this is the only thing i m missing now
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08:46.48AsteriskExpertanybody????
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08:58.06linageehrm. this sort of sucks. i get a lower latency to my new voip provider, but i tried a test call and it "sounds" more latent.
08:58.30Strom_Mwell, that's just one hop
08:58.45linageeStrom_M: true.
08:58.48Strom_Mtotal end-to-end latency may very well be greater
08:58.49linageeStrom_M: it's just i was hoping
08:58.51linagee:-/
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09:15.05Brixiusdoes asterisk work on x64 linux?
09:16.45XxxXYes.
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09:30.12orestyle9is asterisk a feasible option for call centers?
09:33.37Brixiuswe use it as part of our call center application
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09:34.23orestyle9do you use it for incoming calls ( in a round robin) setting?
09:34.43Brixiusgranted it's only 1 part of our call center app, we also use a comecrial app.
09:35.18orestyle9we are using commercial hardware and software which seem to work only some of the time...
09:35.39BrixiusI didn't write it, but I believe it's handeling all the phones in our call centers, it's not doing the routing decisions
09:35.45orestyle9especially the software and look out for those licensing charges..
09:35.58Brixiuswe use cosmocall as well
09:36.18orestyle9Thought maybe it would route calls to the most appropriate call taker...
09:36.28orestyle9ie longest available etc...
09:36.32Brixiuswe do use asterisk for our conf bridge
09:37.44orestyle9We are using a Mitel 200sx at the moment with PraireFyre reporting software...
09:38.04Brixiusif you can come up with the logic to do it, asterisk will handle It I believe, you might have to do some external scripting in perl to make the routing decisions.
09:39.19Brixiusyou can setup agent groups, and come up with a menu that users can follow.  or if you are doing a lookup against the ani or a sn type thing you could do that in perl.
09:39.56orestyle9Cool..
09:41.09*** part/#asterisk orestyle9 (n=orestyle@64-13-89-220.mfd.clearwire-dns.net)
09:42.42Brixius"There's a fine line between genius and insanity.  I've erased that line...." HAHA
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11:01.17AsteriskExpertI just fixed it. found sipcallid of leg 2 :D
11:14.40AsteriskExpertoops unanswered channels dont give sip call id
11:14.42AsteriskExpert:(
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11:43.30Hali_303hi
11:44.15Hali_303does anyone use * with wrt54gl router? what is the point of that? how can I do voip with that router?
11:46.06nibbler_deyou can do voip with it but you don't have any interfaces to connect to pstn
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11:50.13Hali_303nibbler_de: but can I connect some kind of USB phone device to it?
11:50.36Hali_303my main concern is that I dont wanna have the PC turned on for making a call
11:50.45Hali_303so I'm looking for a voip router
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11:57.09asdxHali_303: is there a voip router?
11:57.22asdxthat could run asterisk of course
11:59.20nibbler_deHali_303: usb phone device? you can "connect" a sip phone to it
11:59.26brookshireasdx: the asterisk appliance :)
11:59.55brookshirehttp://www.digium.com/en/products/hardware/aadk.php
12:00.10brookshireunfortunately, it's not available yet...
12:00.32Hali_303nibbler_de: hmm true. I just thought because usb devices are cheaper
12:00.33brookshireyou can get one, but it includes training and such.. ment to be for developers
12:02.29nibbler_deHali_303: usb "phones" are cheaper because they're not an actual phone but an usb audio device where you need a softphone for to use it
12:09.54xnonanybody knows about a nice case for one asterisk server with digital LCD Front in the frontal case!
12:09.56xnon?????
12:10.28xnonhow this http://pbxpress.com
12:11.32monstedxnon: silverstone makes one
12:11.41xnonsilverstone?
12:11.51xnonhaven the link?
12:12.03xnonhave u the link url?
12:12.15monstedhttp://www.silverstonetek.com/products-lc18.htm
12:12.34asdxhow much the asterisk appliance do cost?
12:12.52monstedthat one has a complete VGA monitor on the front
12:12.57xnonwoao monsted thanx so much i like it
12:13.43monstedhttp://www.silverstonetek.com/products-lc16m.htm "only" has a small VFD display
12:14.36monsted$600 for the one with the VGA screen is a little much :)
12:16.14xnon$600 ? :S woao
12:16.44tsurk0hello everybody
12:16.46monstedwell, what do you expect - it's got a bloody VGA screen in it :)
12:16.49xnonbut this cases have a motherboard too really?
12:17.04tsurk0i'm receiving this in CLI -> res_config_mysql.c:361 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info.
12:17.20tsurk0COuld you tell me how to check debug? Is it some kind of log file or what?
12:17.36xnonsip  debug
12:17.44tsurk0thank you
12:18.22xnonor sip debug ipnumber
12:18.58monstedxnon: no motherboard, just a case
12:19.00xnonor sip debug peer where peer is the exten for debuger
12:19.45xnonmonsted, look at LC04 Model !
12:21.13xnonsay Riser Card 1 x PCI (Included), 2 x PCI or 1 PCI-Ex16 (optional)
12:21.31xnonmonsted, leet to see it
12:21.58monstedstill not a motherboard
12:22.14monstedonly the riser boards needed to use the pci/agp slots
12:22.26asdxso, if i have a pbx connected through my isdn line, and i'm outside with my wifi voip phone, i can make calls through my pbx?
12:23.36xnonu need a inbound route for your extn
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12:27.30asdxinbound?
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12:40.52tsurk0hmm, i am using iax2 so may be i was supposed to use iax2 debug, i did so but it says nothing about db
12:41.09tsurk0is there any way to check wich querry realtime is unable to perform?
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12:51.02Drukenoh man, am i a sucker for punishment.....
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13:27.01Teeliis Gizmo project SIP based?
13:30.58xnonyes
13:31.24xnoni dont be sure but i think that its a IAX2 Project too
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13:54.15Drukenwhat is gizmo?
14:02.32xnoni dont be sure but i think that its a IAX2 Project too
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14:38.25zotzxnon, as I am interested in IAX2 phones, I checked on Gizmo when I saw your post, from what I can tell, it is SIP and not IAX2...
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15:50.40tracyneed advice, i will be away from home for an extended period of time. would like to pick up voicemails to home phone via email, and to be able to make remote calls from internet through asterisk (voip). is a X100P good enough for what i want to achieve?
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16:01.01Hali_303hi
16:02.37Hali_303I know that it's not the best channel to ask, but maybe you guys ahve sg like that: I'm looking for a home ADSL router, that does wireless and is SIP capable (it contains an ATA). what do you recommend?
16:03.27Hali_303if it contains linux firmware, that would be superb
16:07.12bsdfreakhmm
16:07.17bsdfreaki know linksys has stuff like that
16:07.26bsdfreaknot that i can recommend it, since i've never used it
16:07.54bsdfreaki'm sure smc and d-link (and others) have their own, too.
16:07.59Hali_303ok. then the alternative: can you recommend a good Wifi SIP phone? or wifi IAX?
16:08.07bsdfreakhmm
16:08.18Hali_303then I'll buy a simple router and use it with wifi phone
16:08.52bsdfreaki don't have a wifi sip phone either, but i've seen some setups that seem pretty cool.  uniden has one that has a base unit that allows you to connect a phone (or phone line if you're using several phones) with it and it also comes with a wireless phone
16:09.21bsdfreakpretty cool concept.  i haven't used them, but i saw them online for around $100.  you do have to be careful though, because i heard certain models are locked by vendors like 8-packet or whatever.
16:09.36bsdfreaki know linksys has them, too.
16:10.07bsdfreakand iirc i heard of one a while ago that allows you to use sip, pots and a cell phone (via bluetooth) as your outgoing, define routes (based on times or number dialed/both), etc.
16:10.11bsdfreakseemed like an interesting idea.
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16:11.12bsdfreakjust having a way to deploy access to a cell phone housewide is neat.
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16:11.23riddleboxis it possible, to have a que made up of agents on different asterisk servers?
16:11.30bsdfreaki think so
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16:11.51bsdfreaki think you have to have your servers as peers or something
16:12.01riddleboxI guess as long as the servers are talking it should work
16:13.21riddleboxspeaking of a cell phone look at this http://www.voipsupply.com/product_info.php?products_id=1755
16:13.45Qwelleww, dlink?
16:13.58QwellI barely trust them to make networking gear
16:14.04riddleboxwell I am just saying it looks like a cellphone
16:14.30Qwellit's kinda ugly
16:14.48riddleboxyeah, I like the linksys one better
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16:15.32riddleboxwow tdm cards a pretty cheap $149 unless you know where to get them cheaper
16:16.02riddleboxthat is for a card with one fxo
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16:21.13guptaaI would like to create a macro that will place a call (to a cellphone) to notify someone that a voicemail was left.  I planning on using the 'h' extension after the voicemail is left to run a DeadAGI that will create a .call. Does anyone know better way to do this?  The thing don't like about the .call is that I want to use a 'or 'memoryhunt' dialparties call a number of until one answers call.
16:24.13riddleboxguptaa, this may help you http://www.voip-info.org/wiki/view/Asterisk+tips+Wake-Up+Call+PHP+with+Snooze%252FAnnoy
16:24.51riddleboxreally, http://www.voip-info.org/wiki/index.php?page=Asterisk+tips+Wake-Up+Call+PHP
16:25.43guptaariddlebox: I was looking at that and it used .call files
16:26.10riddleboxyeah thats why I posted it, I know it uses .call files
16:26.50riddleboxI just use the voicemail.conf to send my cellphones a text message saying a voicemail was left
16:27.49guptaaThis is actually going to call a number of people that are in the field, and it needs to continue calling them until someone confirms the call was answered
16:27.59guptaaor that the voicemail was received rather
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17:10.47stolpenHi I have a small problem, anyone have a moment so help?
17:12.45stolpenI have 2 accounts in sip.conf, setup the same way. Different usernames of course. One is used on X-lite, the other on a nokia e61. The x-lite user can register using the username, the e61 user only registers when using the extension (numbers) as account. Why?
17:12.48PMantisHello, I just built a server, Ubuntu, asterisk 1.2.14. Added the zaptel/zapata configs or the quad T1 card, and the 3FXS/1FXO card.... BUT, no dialtone on the analog phones.
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17:13.32SwKpmantis: did you add the zap stuff after you build and installed asterisk?
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17:13.57stolpenIf I register with the username on the e61 it won't accept calls.
17:14.06PMantisSwK, To clarify:
17:14.07PMantis<PROTECTED>
17:14.08PMantis<PROTECTED>
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17:14.42PMantisSwK,and i can dial, ring other phones, accept incoming calls... just the card (asterisk?) is not playing a dialtone to the phone.
17:15.04SwKbeats me
17:15.19SwKI'm assuming TDM400 ?
17:15.23PMantisye
17:15.25PMantisyes
17:15.27SwKhavent seen that happen
17:15.30PMantisOhhhhhhhhh
17:15.39SwKunless your inidications are screwed up for those channels
17:15.39PMantisThis is an older card...
17:15.41PMantisperhaps Rev h.
17:16.24PMantisShould I pastebin my /etc/zaptel.conf for ya?
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17:18.32Strom_MPMantis, are your T1 spans configured?
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17:21.51PMantisStrom_M, yes
17:22.11Strom_Mis at least one of them generating timing or receiving timing from somewhere?
17:22.16PMantisStrom_M, Completely unused right now, the server is in my office for now, until it's time to install on-site
17:22.47PMantisStrom_M, Dunno... used the autoconfig binary that came with the zaptel stuff
17:23.08Strom_Mcheck zapata.conf
17:23.14Strom_Mand zaptel.conf
17:23.32PMantisSTANDBY FOR PASTEBIN
17:23.36PMantisWhhos.
17:23.42PMantissorry - caps :)
17:23.48Strom_Mjeez, now i'm expecting a pastebin from NASA
17:24.02PMantislol
17:24.25PMantishttp://pastebin.ca/290058
17:24.34Hali_303question: if I have asterisk running on a wrtg65GL router, what are my options for a headset?
17:24.52PMantisHali_303, SIP or IAX2 based clients.
17:24.55Hali_303I've seen that someone got a bleutooth headset working for it
17:25.03Strom_Mah, I think I see the problem, PMantis - all your T1 spans are set up to receive timing and none are set to generate it
17:25.05Hali_303PMantis: I mean using the router's USB port
17:25.11PMantisI doubt that router speaks bluetooth.
17:25.27PMantisStrom_M, that will affect the TDM4xxp?
17:25.44Hali_303PMantis: in a forum a guy said that he attached a bluetooth adapter, and got a headset working
17:25.49Hali_303using chan_bluetooth
17:25.50PMantisHali_303, Oh, maybe... that's a new one to me. :)
17:25.59Strom_MPMantis, yeah, i've run into this before - if asterisk is set to look for a timing source from a t1 and can't find it, the TDM card won't pass audio.
17:26.12Hali_303is there any device, which can be attached to a USB port, and enabled someone to talk using asterisk?
17:26.23PMantisStrom_M, It passes audio fine - once I dial. :)
17:26.29Hali_303like a USB headset or something like that
17:26.37PMantisHmm, come to think of it, I never tested RECEIVING audio
17:27.07PMantisYeah, works fine...
17:27.17Strom_MPMantis, try this:  disable the T1 card completely and see if you get dial tone
17:27.21PMantisThe *only* thing missing s a dialtone. :)
17:27.21sweeperHali_303: there are like 8 billion USB SIP phones
17:27.37PMantiss/ s / is /
17:27.46Strom_Modd
17:27.52Hali_303sweeper: what do you recommend? (=cheap?)
17:28.18PMantisStrom_M, Ok, I'll unload the module, uncomment the T1 stuff, and try...
17:28.54Strom_MPMantis, anyway, just as a matter of nitpickiness, the T1 card should be configured before the TDM card; that way the T1 card grabs channels 1-96 and the TDM card gets 97-100
17:29.06sweeperHali_303: I can't, never used one before
17:29.19Hali_303but do you know of any that is compatible with asterisk?
17:29.23sweeperand there's always the option of a nice usb headset + softphone
17:29.23PMantisStrom_M, It's not that way...
17:29.34sweeperHali_303: anything that does SIP is compatible with asterisk
17:29.58Hali_303sweeper: ok thanks
17:30.08PMantisStrom_M,  What determines the order? When the modules are loaded to the kernel, or the /etc/zaptel.conf ?
17:30.26Strom_MPMantis, good rule of thumb is always go from high->low density, digital -> analog
17:30.37PMantisok
17:30.54Strom_Myeah, the order you load the modules
17:31.15PMantisI think udev is loading them.. 'cause I never set that up.
17:34.55PMantisStrom_M, Yeah, removed the T1 card module, reconfig, and I have dialtone...
17:35.11Strom_Myep, as I thought
17:35.33Strom_Mmake sure you're generating or receiving timing on at least one T1 span
17:35.53PMantisMakes some sense...
17:35.56Strom_Mand make sure you load the module for the T1 card first and configure that card as spans 1-96
17:35.57Strom_Mer
17:36.00Strom_Mchannels 1-96
17:36.01PMantisI need to learn more about T1 stuff.
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17:37.44Strom_MT1 is fun
17:38.07PMantisYeah, I configured one other PBX system witha  T1 card.
17:38.13PMantisand that was remotely...
17:38.31Strom_Mthen again, I'm a telecom junkie
17:38.36PMantisI built the server, shipped it to Canada, and SSHd in whe on the phone with their phone provider. :)
17:38.48PMantisheh
17:39.00PMantisStrom_M, Thanks for the tips - I think i got in under control for now.
17:39.04Strom_Myou are a braver man than I - I would want to be there to configure the T1 in person
17:39.08Strom_Myou're welcome
17:39.43PMantisThe call center has been runing since July and i've never been there. :) heh
17:40.11PMantisSwK, Thaks for your help as well
17:40.22SwKhuh
17:40.23SwKsorry
17:40.24SwKphone
17:40.26PMantiss/Thaks/Thanks/
17:40.29SwKwhat did I miss?
17:40.32PMantisheh
17:40.41SwKget it working?
17:40.59PMantisSwK, I yanked the T1 kernel module, and the TDM behaved.
17:41.33PMantisI need a timing source on at least 1 span.
17:41.33PMantisStrom_M, pointed that out.
17:41.33Strom_Mor you need to generate timing on one span
17:42.07PMantisStrom_M, Yeah... I not gonna worry about that until it's installed with a real T1 connection.
17:42.29PMantisMight as well pull the card from the server for now.
17:42.54SwKyou can leave the card in there it wont hurt anything w/out the module loaded
17:43.13PMantisSwK, Except that udev ( or something) is loading it on bootup..
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17:49.13birrddog?
17:50.15PMantisHmmm
17:50.19horsesgofasterhello, has anyone tried the asterisknow LiveCD?
17:50.26PMantisOne other thing to note...
17:50.46PMantisI dial, use Zap/4 (TDM FXO port)... Zap/4-1 answered SIP/999-081ef8c0
17:51.02PMantisBut... there's no phone cable in that port... shouldn't it NOTICE that?? :)
17:51.33PMantisIf they call 911, this should pass a *real* port state.
17:52.24Strom_MPMantis, *shrug* it's your responsibility to make sure there's a phone line there
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17:53.04PMantisHeh - just thought it would be smarter than that...
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17:55.28birrddoganyone here familiar with Shared Line Appearance in 1.4 ?
17:55.50russellbbirrddog: it's not quite done ...
17:56.00russellbunfortunately, it won't be until the release of 1.4.1 it looks like
17:56.10birrddogBeen reading all the "adverts" for 1.4 claiming it's there...
17:56.15russellbright.
17:56.29russellbit is "there", but it doesn't work correctly
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17:56.45Qwellrussellb: !
17:57.00russellbit has been pending to get fixed for a long time now
17:57.06birrddogrussellb: thanks, any reference sites I can look at for more info?
17:57.07russellbbut we decided not to hold off 1.4.0 for it any longer
17:57.14russellbnot that i know of
17:57.34birrddogcan you guess my next question? ;-)
17:57.34Qwellrussellb: Don't forget libpri and zaptel :D  (yay, btw)
17:57.44russellbQwell: yep ;)
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17:57.55birrddog... any eta on 1.4.1 (ball park) ?
17:58.03russellbend of january?
17:58.09russellbthat's my guess ... but who knows
17:58.10Qwellbirrddog: some time after 1.4.0, at the earliest.  heh
17:58.21birrddogyeh
17:58.27birrddogwow, this irc brings back memories
17:58.27Qwellend of Jan sounds about right though
17:58.47russellbQwell: yeah, unless something is reallllllllly broken in 1.4.0
17:58.50birrddogcool, thanks for the update
17:59.09Qwellrussellb: like the compile error I almost missed in chan_skinny? :(
17:59.25filerussellb: why aren't you doing... stuff... elsewhere!
17:59.32russellbQwell: hehe
18:00.00russellbfile: i'm hanging out with family.  waiting on my brother to get ready to go to lunch
18:00.05filerussellb: excellent
18:00.48birrddogThanks all
18:00.54russellbbirrddog: you're welcome ..
18:01.40Qwellrussellb: file: 4 of 6!
18:01.56filewhatttttt
18:03.51fileah yes
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18:22.36ManxPowerSprint wireless makes a lot of money by leasing out time on their network to smaller cell companies (mostly prepay) that do not have their network.  Does anyone where I can find out what companies use Verizon Wireless's network for their service?
18:23.24filegettalktime.com has a good listing and usually says who the underlying network is
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18:24.58Strom_Mah, MVNOs
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18:25.00Strom_Mone moment
18:25.15ManxPowerWhere I live the ONLY network available is Verizon Wireless.
18:25.23QwellManxPower: my condolences
18:25.29ManxPowerNo Nexten, No Cingular, no Sprint, etc
18:26.14robl^tmobile?
18:26.24Strom_Mhttp://en.wikipedia.org/wiki/List_of_United_States_mobile_phone_companies
18:26.34Strom_Mthat tells you which network each one uses
18:26.48ManxPowerStrom_M: It never occured to me to look at the Wiki
18:26.50ManxPowerThanks!
18:26.56ManxPowerrobl^: no t-mobile either
18:28.02ManxPowerAha!  the term I'm looking is MVNO
18:28.10robl^that's what ya get for living out in the wilderness with all the bears.  ;-)
18:28.18Strom_Mhahaha
18:29.24coppiceMobile Very Nearly an Operator
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18:30.22Qwellis it bad to drop a bottle cap into a heating duct?
18:30.45Strom_Mprobably
18:30.49Qwellcrap :D
18:31.01Strom_Mtime to call Central Services
18:31.25Pete__Not really sure where im going wrong, calling out to gizmo users works, calling in there is no audio either direction. Im sure its something stupid im missing ;)
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18:59.11Pete__any ideas?
18:59.45hwta
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19:02.28zmef420<PROTECTED>
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19:06.59NetviewHello! Anybody nows soemthing about the releasedate of bristuff-0.3.0-PRE-1x (asterisk- 1.2.14) and (maybe) if kapejod plans a bristuff-version for asterisk-1.4?
19:07.33DUAL_MANhey everyone
19:07.37DUAL_MANwhat is asterisk now
19:10.22ManxPowerDUAL_MAN: ask on #asterisk-gui
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19:13.00Netviewwithout typos :-)  Anybody knows something about the releasedate of bristuff-0.3.0-PRE-1x (asterisk- 1.2.14) and (maybe) if kapejod plans a bristuff-version for asterisk-1.4?
19:20.08tzafrir_laptopNetview, we heard you the first time
19:20.58tzafrir_laptopI have no idea what are kapejod's plans
19:25.01Igbothom_IIIAndyCap, like 99% of other things, it will be released when the project maintainer deems it to be ready for release, not on a particular date that has no relationship to the code being actually ready (like Windows Vista was released for)
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19:31.03StyleWarzHey guys, is it somehow possible to define the order agents in queues get called?
19:32.06ManxPowerStyleWarz: yes
19:32.39StyleWarzManxPower: Well with placing ",<somenumber" only works if the guys are busy, so what is your idea? :)
19:36.39ManxPowertry using rrmrmory method
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19:38.24tparcinaStyleWarz: define priority for every agent
19:42.20StyleWarztparcina: Well i have my phone at home, my voip connected cell and my cell phone number if it doesn't have a wifi network in range
19:42.25StyleWarzi want to ring it in this order
19:42.31StyleWarzwith all features of queues
19:43.40XxxXHey, anyone know IAX providers that allow callerid Spoofing please tell me?
19:45.19*** join/#asterisk ApEtc (i=apetc@ip70-162-197-214.ph.ph.cox.net)
19:51.42*** join/#asterisk [Airwolf] (n=airwolf@cp1141755-a.landg1.lb.home.nl)
19:52.36NetviewXxxX: try voipjet.com
19:56.44*** join/#asterisk kink0 (n=k@161.pool62-37-205.static.orange.es)
19:56.47kink0good night
19:57.16*** join/#asterisk stealth (n=stealth@6.206.204.68.cfl.res.rr.com)
19:57.25kink0the problem when some calls are inmediatelly or with fews seconds hanged, is a know issue ?
19:58.42kink0-- Zap/59-1 is making progress passing it to OOH323/208.2.2.2-320a and inmediatelly Hangup Zap 59-1
19:58.50kink0that happens with ooh323 as well as SIP
19:59.15*** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust337.leed.cable.ntl.com)
20:00.32*** join/#asterisk Gr1ncheux (n=devine@AToulouse-257-1-119-227.w82-125.abo.wanadoo.fr)
20:02.55kink0in the event I assume the other part is hanging the line, any idea why ? I got something like this in CDR: 3,0,"NO ANSWER","DOCUMENTATION"
20:03.48kink0of course I call my self ussing a F3000 wifi ip phone, and then there no any problem.
20:07.43*** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net)
20:08.40*** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar)
20:09.38linageewifi phone?
20:09.46linageemaybe you are using your neighbor's wifi. :-O
20:10.17kink0linagee, no.. no hehehe, I am ussing a wifi IP phone registered ussing my home wifi net to my remote asterisk
20:11.32linageekink0: what is the purpose of wifi voip?
20:13.25*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
20:13.36kink0linagee, nothing, I just test to compare results, and I used a SIP phone I have at hand
20:13.54kink0I used a F3000 phone I bought from digiumcards.com
20:14.00kink0that works fine.
20:14.27kink0but when calls from my peers arrives to my Asterisk, many calls are hanged without much duration.
20:14.28linageemaybe some day evdo unlimited plans will become so cheap that you could use it at a replacement cellphone
20:14.38linageejust push voip over the evdo/3G
20:14.59linagee(only evdo has horrible latency)
20:15.23kink0may be in USA, but here ( Spain ) that is some utopic
20:16.14linageekink0: since it's spain and there is no FCC, why not get a huge power increaser thing for the wifi
20:16.16kink0while you in USA can transmit with 4W power on antennas, here is limited to 100mW, that will give you an idea about wifi penetration on the market
20:16.23linageekink0: turn your 100mW into 50W!
20:16.42linageethe FCC is stupid
20:16.50kink0no FCC but is limited by law to 100mW
20:17.19linageekink0: invent a good mesh network
20:17.27kink0anywise , everybody uses high gain antennas and so
20:18.11kink0my problem is not about wireless now, is about calls, comming to my asterisk that is connected to a DS3 line
20:18.22linageekink0: the best mesh network will be the most compatible
20:18.27kink0the wireless is only here, for home, and I used it for testing
20:18.53linageekink0: basically, you need to steal bandwidth from free wifi access points in existance, and bring the network farther out, and still allow people's 802.11b embedded in their laptop to work
20:19.09kink0the question is about calls comming from external peers, arriving to the DS3, and then placed on my Asterisk, that have a very little duration
20:19.51linageekink0: maybe if you had a wifi repeater with two transcievers, one having a directional antenna and the other having an omni antenna. that might work.
20:20.18kink0I don't know why the caller party is hanging, if lack of ring, or so, even I place a priority with Ring ( some times gives dual rings, but that is better than nothing )
20:21.11kink0see you later, I have to go now
20:23.36*** part/#asterisk Netview (n=Netview@212.60.156.145)
20:27.41nibbler_deamplifyers are for people who can't build decent antennas ;9
20:28.47*** join/#asterisk ScottyTM (n=ScottyTM@marlin.42h.de)
20:28.57nibbler_deScotty!
20:29.15nibbler_delong time no see ;)  when will you arrive in berlin?
20:30.11ScottyTMnibbler: 26. in the evening
20:30.13ScottyTMmy plane lands at 18:30
20:31.23nibbler_dehmkay - i'm still undecided if i should go tomorrow or on monday
20:34.32ScottyTMmy family wants to see me once in a while ...
20:34.55ScottyTMsince when does xchat do spell-checking?
20:35.10QwellScottyTM: for a while now - as long as it's linked with aspell
20:35.15*** join/#asterisk GiantPickle (n=GiantPic@S0106006008bd147d.gv.shawcable.net)
20:35.22Qwellmine doesn't, because gentoo is stupid sometimes
20:35.25ScottyTMthat's awful
20:35.38Qwells/is/acts/
20:35.49ScottyTMfeel's like using M$ Word
20:36.40Qwelloh, heh, it needs to be enabled is all
20:37.12Qwellit's not as good as the gaim spell checking though...  kinda lame...I'm gonna re-disable it
20:37.33Qwellin gaim, it doesn't highlight until you finish spelling a word..  xchat does it as you spell...that's silly
20:40.49matt_hiya, does anybody know why i never get a phone call when i register my number with e164.org
20:41.01matt_its a 00448444 number
20:41.05Qwellmatt_: I'd ask them..  there is no way we can know
20:41.28matt_Qwell, i have i posted on the forum, just wondered if somebody else has had the same issue
20:42.02Qwellno, I mean ask their support folks
20:43.29matt_Qwell, support people ?
20:43.36Qwellat e164.org
20:43.40*** join/#asterisk daysmen3 (n=primus@host86-138-239-159.range86-138.btcentralplus.com)
20:43.41matt_i didn't see anything about that
20:43.50Qwellwell, somebody runs it
20:44.20matt_yea, looks like a guy called evilbunny runs the forum
20:44.34Qwellno, not the forum..  nevermind
20:45.24matt_Qwell, it seems to be all part of the same system
20:45.36*** join/#asterisk obnauticus (n=aao_pwne@c-24-21-116-29.hsd1.wa.comcast.net)
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20:59.41dlynes_laptopHas anyone else had issues installing zaptel 1.2.14 on a 2.4 kernel?
21:00.07XxxXdlynes_laptop works for me
21:00.17XxxXi had it on my slakcware box before it was running 2.4
21:00.22dlynes_laptopDoes it require devfs, then?
21:00.56XxxXmeh, to be quite honest, i dont exactly rember, i dont rember installing or doing anything else other then what the manual said i untar'd and then installed that was it
21:00.59dlynes_laptopOr maybe something else that's not default on Slackware?
21:01.09sevardsup dlynes
21:01.15dlynes_laptopheya sevard
21:01.50dlynes_laptopsevard: you never say hi anymore :(
21:01.54XxxXno dlynes it was all default
21:01.55XxxXwich i used...
21:01.57sevardpishaw
21:02.00sevardyou never hug me anymore
21:02.01dlynes_laptopOk, weird
21:02.10shellsharkhah
21:02.26dlynes_laptopsevard: I said sup bitchness, the other day, and you didn't even say nada
21:02.30*** join/#asterisk ameyavp (n=ameya@221.134.1.165)
21:02.31sevardhey x86
21:02.34shellsharkheya
21:02.38dlynes_laptopoh
21:02.40sevarddlynes_laptop: maybe i was busy with your mom
21:02.42dlynes_laptopshellshark's x86?
21:02.44sevardbecause i didn't see it
21:02.50shellsharkdlynes_laptop: *nod*
21:02.56shellsharkshellshark: you didn't know?
21:03.02dlynes_laptopI think you mean wanking to my mom's pic
21:03.08shellsharkdlynes_laptop: url?
21:03.09sevardx86: i heard my ex-roommate canceled
21:03.19sevardsorry to hear
21:03.20shellsharksevard: not that I'm aware of
21:03.27sevardoh? interesting
21:04.10*** join/#asterisk awannabe (n=brad@ip24-251-135-202.ph.ph.cox.net)
21:04.16shellsharkhe's used the service today, as a matter of fact :)
21:04.27sevardinnnnnnteresting
21:04.43shellsharkso... ex-roomie huh? what happened?
21:05.05sevardi couldn't really afford the apt anymore, plus.. the damned bugs drove me insane
21:05.15*** join/#asterisk Tili (n=tili@172.Red-88-0-147.dynamicIP.rima-tde.net)
21:05.16shellsharkbugs--
21:05.23sevardbox elders, you know of them?
21:05.30shellsharksevard: oh yes
21:05.33sevardin older houses, durning the winter months, they live in the walls
21:05.39shellsharkthey are near impossible to get rid of ;)
21:05.44*** join/#asterisk ameyavp (n=ameya@221.134.1.165)
21:05.46shellsharkyeah
21:05.52sevardand when you turn on the central heating they crawl into bed with you
21:05.53shellsharkso you got a new place now?
21:05.53XxxXdurning?
21:06.03sevardduring, i just woke up :PO\
21:06.06sevardgrr, :P
21:06.06XxxX:P
21:06.27sevardshellshark: yeah, i'm living with this old couple ... it's nice since they went on vacation for 5 months
21:06.39sevardpretty sure they went to utah to die, but that's not my business.
21:06.48*** part/#asterisk ameyavp (n=ameya@221.134.1.165)
21:07.20sevarddlynes: when the hell are you going to party on up with the sevard
21:07.28awannabehey guys, im planning on doing a failover * server, the easiest way is to have all the phones have static IPS and not make them register with the server, right? so the server ALWAYS know how to call the phone, correct?
21:07.28sevardbecause you never live up to your promises
21:07.50shellsharksevard: oh man that's awesome lol
21:08.05shellsharksevard: so are you wanting service from us?
21:08.13sevardword, i had a tiny room in this house to myself, but since they're gone, i have the whole place to myself
21:08.21EzWayanybody can give a minute or 2 to add voipjet account to my current asterisk server ^
21:08.23XxxXhey dlynes you must know of a iax provider that allows callerid spoofing or doesn't block it at least?
21:08.27sevardshellshark: in 2 months, when i'll need it, i know where to go.
21:09.00sevardXxxX: most VoIP providers let you set your own outbount CNUM.  IIRC shellshark allows that ;)
21:09.05sevardoutbound*
21:09.17XxxXIIRC? :P?
21:09.25sevardif i recall correctly, shellshark.net
21:09.29XxxXahh ok
21:10.09XxxXits unlimed outbound calling for 15$ umm thats kinda shady lol
21:10.17XxxXnext thing you know service wil be crap :p
21:11.24sevardthat would be subjective, if anything, they're worth a trial
21:11.50XxxXum i guess i'll try them out for a month and see :p
21:12.01sevardthat's the voip spirit!
21:12.29sevardx86: i couldn't remember, do you guys offer E911?
21:13.05XxxXsevard know of any other ones incase shellshark doesn't work out :p?
21:13.44sevardclip-no-screening?
21:13.56nibbler_desevard: you can send an arbitrary caller-id
21:13.57XxxXsevard?
21:14.11nibbler_dewhich is nice if you want to forward calls to pstn
21:14.17XxxXnibbler_de URL?
21:14.17sevardXxxX: I've had experience with teliax, they wern't that bad, good DID availibility,  expensive though.
21:14.41XxxXthey allow you to setyoure own CNUM?
21:14.41nibbler_deXxxX: it's .de only
21:15.01nibbler_dehttp://www.bellshare.com/de/
21:15.12XxxXsevard?
21:15.14XxxXthanks nibbler_de
21:15.21*** join/#asterisk zotz (n=zotz@24.244.163.157)
21:15.47sevardXxxX: correct, most VoIP providers offer you service designed for a PBX, which would, in turn, give you the ability to set your own outbound CNUM
21:15.59XxxXahh thanks :)
21:16.39shellsharksevard: cool deal
21:16.52sevardx86: i couldn't remember, do you guys offer E911?
21:17.51XxxXhah
21:17.56XxxXeveryones doinf that now days :(
21:18.10shellsharksevard: E911, no
21:18.18sevardah.
21:18.30shellsharknibbler_de: you own a CLEC?
21:18.52nibbler_de"CLEC"?
21:19.09shellsharkif you dont know what a CLEC is, you can't get SS7 ;)
21:19.14awannabehaha
21:19.47nibbler_dedo you mean: "Competitive local exchange carrier"?
21:19.56sevardheh.
21:19.58shellsharkwhat else would I mean?
21:20.00shellshark:p
21:20.03nibbler_deshellshark: i don't know
21:20.17*** join/#asterisk hohum (n=dcorbe@c-71-62-76-68.hsd1.va.comcast.net)
21:20.19sevardhe's in germany.
21:20.26XxxX-Clausterphobic Land Enemy Carrier :p?
21:21.31nibbler_dehere in .de we have a nearly fully open market - most telcos operate nation-wide networks - only last-mile to the customer is often an issue - but the former state-owned network has been given to dtag who have to sell it for a fixed price
21:22.00*** part/#asterisk Pete__ (n=pete@nock.demon.co.uk)
21:22.17*** join/#asterisk Pete__ (n=pete@nock.demon.co.uk)
21:23.42*** join/#asterisk ameyavp (n=ameya@221.134.1.165)
21:31.02ManxPowernibbler_de: in the USA, the last mile companies are not required to lease their lines to other companies anymore.
21:31.15ManxPowerMost of them do, but they are not required to
21:32.04nibbler_deManxPower: i have to get hold of a decent paper (or likewise) about how the us market works - it's a complete mystery to me
21:32.32ManxPowernibbler_de: it is a complete mystery to us in the usa too
21:32.44nibbler_dehehe ;)
21:32.46nibbler_dei thought so
21:33.04XxxX-lol
21:34.17ManxPowernibbler_de: in the USA there are 2 basic types of voice telecom companies.  Local and long distance.  For the most part local (LEC) companies are not allowed to provide interstate (or internata) service
21:34.27Gr1ncheuxhi there
21:34.44Gr1ncheuxi have some digium products to sell
21:34.53ManxPowernibbler_de: the rules in the ISA change every few years as well.
21:35.03nibbler_deManxPower: so how do you book a long-distance carrier? do you tell your local carrier?
21:35.09Gr1ncheuxif someone is interested pm me :)
21:35.12ManxPowernibbler_de: yes.
21:35.30ManxPowerusually you actually get the LD carrier and then tell your local carrier what it is.
21:35.43ManxPowerthe LD carrier will noticy the local carrier in many cases
21:35.46*** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net)
21:37.30ManxPower<PROTECTED>
21:37.44nibbler_demhh, sweet ;)
21:38.09ManxPowereach state has a Public Service Comission aka Public Utilities Commission that set local rates and regulations.
21:38.18nibbler_dethat's how you keep your jurists busy - in .de we do it with the tax system ;)
21:38.36ManxPowerFor example in Tennessee ISDN BRI is about $20/month.  In Louisiana it is about $109/month for the same service from the same company
21:38.52nibbler_demh, i see
21:39.14awannabeyeah in Arizona its like $110 as well
21:39.25sevardI thought BRIs were only used in Europe, where in the US they used PRIs
21:39.29ManxPowerin most states it is a little over $100/month
21:39.56ManxPowerthe local teleco apparently made the local regulators in tennessee angry and they do not allow the local company to charge very much.
21:40.09awannabewhat telco is that?
21:40.30sevardManxPower: So you're saying if you want to start up a VoIP company Tennessee is a good place.
21:40.32ManxPowerIn Louisiana at least one of the people on the regulatory board is good friends with someone on the local telecom's board of directors
21:40.55ManxPowersevard: only if you want to only provide service in TN
21:41.03sevardwhat about LD?
21:41.16ManxPowerMany VoIP companies i the USA contract out to a company like Level 3, which already has local numbers in most places.
21:41.33*** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net)
21:41.36ManxPowersevard: what about LD?  The LD market is more or less deregulated for inter-state calls.
21:41.56sevardso you buy five PRIs in TN and sell VoIP to the world
21:42.13ManxPowersevard: only for outgoing since you will only be able to get TN local numbers for incoming.
21:42.16awannabei can get PRIs for 350 a month here in arizona
21:42.17sevardword
21:42.23ManxPowerThere are a zillion VoIP long distance companies.
21:42.29nibbler_dethat's pretty expensive
21:42.30sevardi think in MN a PRI ranges between 400 and 450
21:42.44awannabe350 is dirt cheap for here, its from 500 to 900 here
21:42.55dlynes_laptopsevard: $1200 here, after you add on all the other shit you have to get with it
21:42.59ManxPowerLocal Bell PRI in Louisiana is about $1,500/month
21:43.01awannabenow that depends on loop across, the one for 350 would be going to a data center, so its all on net :)
21:43.15ManxPowerlocal CLEC PRI can be as low as $350
21:43.17dlynes_laptopsevard: that's $1200 cdn, though, but still...that's about $1100 USD
21:43.20ManxPowernibbler_de: that is for 30 channels?
21:43.24nibbler_deManxPower: yup
21:43.26awannabenibbler_de, thats 27 channels right?
21:43.31awannabe30, i always forget, heh
21:43.36sevard24 channels...
21:43.38nibbler_deawannabe: channelized E1 ;)
21:43.41ManxPowernibbler_de: Telecom in Europe is in many ways far ahead of telecom in the USA
21:43.43*** join/#asterisk psy65535 (n=psy65535@24-180-25-172.dhcp.gldl.ca.charter.com)
21:43.45nibbler_de-> 30 channels you can use
21:43.46awannabeyeah, E1 is diff then T1,
21:43.49sevardPRI, 24 channels, 24th is the D chan
21:43.55nibbler_deManxPower: depends - but generally - yup
21:43.57awannabe23 in US, 30 for others
21:44.04dlynes_laptop24 in us
21:44.05nibbler_deManxPower: what i dislike is the dtag monopoly over here
21:44.08awannabe23 useable....heh
21:44.21dlynes_laptop24 usable, if you skip the pri, and just got to a regular t1 :)
21:44.30ManxPowerFor example, in europe if you want to change carriers you can just change your SIM card.  In the USA ALL phones are locked to the carrier and you are required to buy a new phone when you move carriers
21:44.40dlynes_laptopsame as in europe....29 usable, but 30 if you go to a regular non-pri e1
21:44.47nibbler_deManxPower: if you have a number from the voip-block (032) you have to terminate through dtag - or 0900 - or 0700...
21:44.54sevardManxPower: there's a group of people trying to move away from that
21:44.58nibbler_deManxPower: we have simlock here too
21:45.05nibbler_deManxPower: but only on prepaid-offers
21:45.09fileManxPower: T-Mobile will give you your phone's code after a certain amount of time
21:45.09Pete__ManxPower: I don't know of any providers in the uk who don't lock phones
21:45.16ManxPowernibbler_de: yes, but in the USA it is for ALL service.
21:45.29nibbler_dePete__: i can send you a locked phone from 3 if you like ;)
21:45.32ManxPowerPete__: can you move cerriers if you unlock the phone?
21:45.46nibbler_deManxPower: ah - you don't have simcards, right?
21:45.47Pete__ManxPower: yes
21:46.07Pete__don't have the same frequency problems, most phones sold these days here are atleast tri band
21:46.19ManxPowerOf course in the USA we have many cellular standards.  CDMA, TDMA, iDEN, GSM.  Only GSM uses SIM cards)
21:46.45fileiDEN uses SIM cards too
21:46.47ManxPowerPete__: in the USA even if you unlock your phone you cannot usually move carriers and if you can move carriers you can only move to a carrier with the same standard
21:46.54sevardi've never used a cell in the US that wasn't GSM
21:47.11ManxPowerMost of the TDMA carriers are moving to GSM
21:47.11nibbler_deManxPower: that's why a friend of mine recently had my iridium phone - he was traveling to the us and confused about what service he should use ;)
21:47.35nibbler_deManxPower: well - technically speaking gsm is tdma ;)
21:47.39ManxPowernibbler_de: he should have done what I did.  Buy a pre-pay phone when you get to the destination
21:47.41awannabeand CDMA as well
21:47.48nibbler_deitym amps
21:47.55sevard<PROTECTED>
21:48.02nibbler_deManxPower: he was traveling across the nation
21:48.05sevardif you're only going to be in the country for less than a month
21:48.13ManxPowernibbler_de: yes.  a nationwide prepay would work
21:48.18nibbler_deManxPower: there are?
21:48.21sevardAMP nation wide, so are those tmobile phones, iirc.
21:48.25psy65535Any recommendations for computer cards? anyone have experience with bonding? how many lines can you get out of a decent system?
21:48.30ManxPowerno matter WHAT he did, it would have been cheaper than iridium.
21:48.39sevardwtf is iridium
21:48.46ManxPowersevard: satellite phone
21:48.49sevardoh jesus
21:48.53ManxPower$2.99/mon and up
21:48.54nibbler_deManxPower: he paid 1.20$/min to germany - that was ok for him ;)
21:48.57ManxPowermon == min
21:48.57awannabewhich are like 900 bucks a minute, haha
21:48.58sevardthose are for going to the amazon, dude
21:49.12Pete__hah thats expensive
21:49.32nibbler_deManxPower: long time ago... you pay about $550 for a 500 minute prepaid card
21:49.39sevardi had a friend who traveld to remote parts of africa with a sat phone, paid a flat rate of $8.00/minute
21:49.39ManxPowernibbler_de: nifty.
21:49.41nibbler_deso - price is 1.10/min
21:49.53nibbler_dewhich is rather ok for global coverage ;)
21:50.21ManxPowerIf you are going to travel between the USA and Europe frequently I would recommend a QUAD-band GSM phone.
21:50.29nibbler_degood to know
21:50.31sevardi second that recomendation
21:50.48ManxPoweror just get a phone for Europe and a phone for USA
21:50.49filethe T-Mobile To Go prepaid stuff will actually roam internationally now
21:51.03ManxPowerfile: but for how much/mon
21:51.24sevardI'm looking to spend three weeks in cali and was going to buy 2 prepaids for my gf and I, but they're to damned expensive
21:51.25nibbler_deone cool thing about the us is - you have competiting pager systems
21:51.32nibbler_dein .de there's just one
21:51.39sevardi would have ended up paying about 150 in activation/minutes/phones for just 3 weeks of usage
21:51.44fileManxPower: same as postpaid customers afaik
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21:52.01ManxPowerMy problem with T-Mobile is coverage
21:52.18filetheir prepaid is now capable of roaming on other networks in the US too
21:52.20sevardso i can't find a single prepaid that I can recommend, unless anyone has any insight / cheaper deals
21:52.22ManxPowerI use verizon for DATA, as it has (or had) better coverage than Sprint
21:52.24nibbler_deManxPower: that was his problem too - he needed 100% coverage on his tour since he runs a datacenter
21:53.45ManxPowerPrepaid roaming really sucks in the USA
21:54.11ManxPowerI jump from 10 cents/min to 60 cents/min when I roam on my prepaid
21:54.54nibbler_dewe have just one provider that does national roaming - the difference is 60/1 vs. 10/10 increments
21:55.26nibbler_debut there are a remarkable number of mvnops lately
21:56.24ManxPowerI may move to an MVNO
21:57.18nibbler_deis there any umts network in .us? or do you only have evdo?
21:57.38robl^umts??
21:57.47robl^we have evdo and EDGE
21:58.09nibbler_deumts=3G in europe/asia
21:58.21ManxPowerrobl^: I think the GSM people are moving to UMTS
21:58.21nibbler_de1.8Mbit/S downlink, 384k uplink
21:58.36*** join/#asterisk SomeOne1 (n=SomeOne1@pool-71-126-167-19.washdc.fios.verizon.net)
21:58.50robl^nice!  I can get about 2020k up and down on edge
21:59.03robl^sorry.  200k up and down
21:59.15SomeOne1i keep get this
21:59.47SomeOne1Dec 23 11:31:18 DEBUG[32515] channel.c: Didn't get a frame from channel: SIP/67.15.180.14-087817e0
21:59.47SomeOne1isnt there some way i can get it to ignore that and not drop the call
21:59.48ManxPowerSomeOne1: you can turn off that message by NOT RUNNING IN DEBUG
21:59.48SomeOne1and keep trying to bridge?
22:00.17SomeOne1ManxPower: the reason i turned it ON in the first place was to see why the calls are being dropped
22:00.19ManxPowerSomeOne1: your problem is somewhere ELSE.
22:00.35SomeOne1ManxPower: suggestions?
22:00.35ManxPowerperhaps a codec issue or a transcoding issue or a reinvite issue or or or
22:00.43SomeOne1i see
22:00.46ManxPowerSomeOne1: there should be OTHER messages that will help
22:00.55SomeOne1the calls starts and goes fine for a while
22:01.01SomeOne1but then it drops all of a sudden
22:02.32ManxPowerWhat messages do you get with debug off?
22:03.51*** join/#asterisk BSDTech (n=RNeese@ppp-69-238-216-111.dsl.irvnca.pacbell.net)
22:03.56SomeOne1hmm
22:03.58SomeOne1im trying to see
22:03.59*** join/#asterisk _BOBWEEVER (n=chatzill@adsl-156-72-86.msy.bellsouth.net)
22:08.18SomeOne1whats that service that allows you to paste?
22:08.26SomeOne1instead of pasting into the cahnnel
22:08.27*** join/#asterisk Gr1ncheux (n=devine@unaffiliated/gr1ncheux)
22:08.28psy65535Anyone care to address performance experiences? How many lines can you get out of a server?
22:10.51*** join/#asterisk wils0r (n=wilson@c-71-234-37-243.hsd1.ct.comcast.net)
22:11.15_BOBWEEVER`pm
22:11.18_BOBWEEVER~pm
22:11.25jbotextra, extra, read all about it, pm is private message, or perl mongers, or pathetic moron: when you see someone say pm, they're asking if you think that they're a pathetic moron, or something you don't do without asking permission
22:11.28_BOBWEEVER~pb
22:11.30jbotmethinks pb is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste
22:11.37_BOBWEEVERahh yes  :)
22:16.46*** join/#asterisk mega (n=mega@217.201.75.212)
22:19.28Sline78Is it possible to set the DTMF duration for RTP Events (rfc2833) sent by asterisk?
22:19.47*** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net)
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22:30.04EzWayanybody can help me setup voipjet
22:34.57*** join/#asterisk Gershwin (n=Bonzi@c-76-16-183-209.hsd1.il.comcast.net)
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22:43.21*** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com)
22:43.46EmleyMoorIs there a discussion anywhere of the pros and cons of various codecs?
22:46.54*** join/#asterisk copantl (n=galel@207.13.77.83)
22:47.03copantlhi everyone
22:48.20copantli tried to install asterisk-ss7 and libss7 but i have this error when the make is done: /usr/bin/ld: unrecognized option '--exclude-libs'
22:48.26copantlany idea?
22:49.00copantlis any library missing?
22:49.44*** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir)
22:50.50copantlany bodu there
22:51.11EmleyMoorIs your ld up to date?
22:51.12*** join/#asterisk DUAL_MAN (n=dwayne@64-42-247-120.mb.skyweb.ca)
22:51.42*** join/#asterisk resistance (n=dwayne@64-42-247-120.mb.skyweb.ca)
22:51.43copantlEmleyMoor: what is the name of the ld library?
22:52.14resistancei'm trying to compile asterisk 1.4 and i get a termcap support not found when i run the configure script
22:52.17copantlbecause if i use apt-cache search ld... i see to many info
22:53.45EmleyMoorWell, I believe it's part of binutils
22:53.47matt_is there a 800 number for the uk that will allow me to dial through to a number with an enum lookup ?
22:54.25*** part/#asterisk tparcina (n=tparcina@83-131-154-86.adsl.net.t-com.hr)
22:54.55*** join/#asterisk posixisposi (n=andy@user-0c93jo3.cable.mindspring.com)
22:56.17resistancei've already made sure all the packages i need are installed, why would it still give me a termcap error
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23:00.37copantltrieing to install asterisk-ss7 but i got this :/usr/bin/ld: unrecognized option '--exclude-libs' any idea?
23:03.24*** join/#asterisk brut- (n=brut@66.173.4.254)
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23:12.32psy65535resistance: you need ncurses-devel
23:14.26*** join/#asterisk ShellMan (i=xnon@200.82.222.86)
23:15.53copantlany body can help me with asterisk-ss7?
23:17.25*** join/#asterisk notyou (n=naereb@blk-89-245-103.eastlink.ca)
23:36.04*** join/#asterisk SimoAmi (n=SimoAmi@ip67-91-253-242.z253-91-67.customer.algx.net)
23:36.09SimoAmihi there
23:36.41SimoAmiI need someone to dial a number for me
23:37.17EmleyMoorSimoAmi: What kind of number?
23:37.39SimoAmi7186262209
23:37.47SimoAmisee if it gets through
23:37.53EmleyMoorWhat "kind" is that?
23:38.11SimoAmi?? meaning?
23:38.27Igbothom_IIIwhich SIP carrier
23:38.29robl^US?  other counry phone number?
23:38.35SimoAmiUS
23:38.43SimoAmi17186262209
23:38.46EmleyMoorOK - whose number is it?
23:39.07SimoAmimine
23:39.18EmleyMoorI'll give it a shot
23:39.22Igbothom_IIIis it a landline, or SIP trunk?
23:39.24SimoAmijust switched the number from verizon to Broadvoice
23:39.45EmleyMoorRinging tone
23:39.53SimoAmibut can't get incoming calls. However one of their customer care agents could reach us
23:40.04SimoAmisip trunk
23:40.23EmleyMoorNot ringing there?
23:40.48SimoAmiyou should get an welcome message if it works
23:40.58SimoAminope
23:41.01EmleyMoorIt rings and hangs up
23:41.06SimoAminot here in new york
23:41.28SimoAmiok, just an advice:
23:41.48robl^they accidentally re-routed your number to Dresden ;-)
23:42.23SimoAminever ever go for Broadvoice. They'll put your business bankrupt and don't care
23:42.48SimoAmirobl^: what do you mean?
23:43.00robl^SimoAmi: sorry. I was joking
23:43.06SimoAmi:)
23:43.22SimoAmiman these people are unreliable
23:48.42*** join/#asterisk copantl (n=galel@207.13.77.88)
23:50.45copantlhi
23:53.59*** join/#asterisk aao_pwner|oper (n=aao_pwne@c-24-21-116-29.hsd1.mn.comcast.net)
23:54.46matt_can anybody here phone 0844 uk numbers from their voip service ?
23:56.47Igbothom_IIII *have* an 0844 number, but can't call 'em  :)
23:57.01EmleyMoorI don't know - but I can call them over my BT line using *
23:57.12matt_humm ok
23:57.34matt_i was talking to the person of e164.org and he cant phone me lol
23:57.40matt_and i cant phone myself :(
23:57.48matt_well i can but only from my mobile

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