irclog2html for #asterisk on 20051117

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00:03.31TheCopsSomeone have played with RDNIS on PRI?
00:05.08justinui suppose this won't work in a macro: exten => ${ARG1},hint,${ARG2}
00:05.09IronHelixnope
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00:18.06dippocan anyone recommend a GUI/web-based config tool for asterisk?
00:20.56dippoamportal?
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00:22.44ginventMy outgoing fax starts to negotiate, then asterisk just hangs up... I am using SpanDSP
00:22.44ginventAny suggestions?
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00:22.45ginventI am using a clone X110P card.
00:23.08ginventAnyone around to help me with my fax problem?
00:24.30Ariel_spandsp has always been 80% ok. But depending on the fax machine it works sometimes sometimes not. Just make sure you have the correct timing setup for the board.
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00:24.51ginventHow do I do that?
00:25.01ginventI am trying to send out a fax...
00:25.12ginventit calls... beeps... starts to go... then just hangs up.
00:25.30ginventHow to I check/change the timing?
00:25.38ginventCould it be my X100P sucking?
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00:27.05Ariel_is this the clone fxo board not the 110?
00:27.11Ariel_do you have it on it's own irq?
00:28.03ginventIt's a clone
00:28.14ginventit's an X100P clone... simple modem card.
00:28.20ginventIt works fine for voice.
00:28.27ginventpicks up calls... makes calls... no problem.
00:28.33ginventBut the fax outgoing... no good.
00:28.42ginventIt could be tx rx gain settings... or timing...
00:29.13Ariel_what are the tx and rx gains set to? for fax they should be 0.0
00:29.26ginventHmmm... I tried that...
00:29.44ginventI had to turn the gain up a touch... rxgain at 15% and txgain at 5%
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00:33.30ginventYeah, it is just weird. I start to get the "fax noises" then asterisk just hangs up.
00:33.40ginventHow can I debug to see any messages on the fax?
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00:35.57ginventHow I can debug the zap channel to see what is going wrong with the fax?
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00:37.48Samoied1.2.0 today?
00:38.21ginventWhat does faxdetect=both mean? if I turn that on... it detects my own outgoing fax and thinks it is incoming.
00:38.36docelm0Who wanted DC DID's?
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00:38.54mog_workwhen is it hitting
00:39.43ginvent?
00:39.52mog_workasterisk 1.2
00:42.45ginventAnyone know how I can debug my outgoing fax from spandsp... it starts to send, then just hangs up.
00:44.40dzahnhi. has anyone experience with an avm b1 isa and a hfc-s card under asterisk? calls routed from extern isdn via avm b1 via hfc-s to a phone are in very bad quality
00:46.19dzahni'm running Asterisk 1.0.9-BRIstuffed-0.2.0-RC8o, chan_capi-0.3.5, kernel 2.6.14 on a k6-300
00:46.48Samoiedmog_work: you work at digium?
00:47.18mog_workyeah
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00:47.51hardwiremog_work: hi
00:47.57mog_workhi
00:48.01YoMamayo mog
00:48.04mog_workyomama
00:48.06hardwirehows the moggy woggy?
00:48.06Samoiedmog_work: You are working for release asterisk 1.2 now?
00:48.14InfraRedBid on ma phone : http://cgi.ebay.co.uk/ws/eBayISAPI.dll?ViewItem&item=6453583074
00:48.23mog_workits commin
00:48.25hardwireinfinity1: don't be a poop
00:48.30YoMamapewp
00:48.43hardwireinfinity1: you can be a poop.. InfraRed don't be a poop
00:49.01Drukenanyone know where i'd find a NPANXX list for the us?
00:49.14YoMamaDruken: google it..they're all over the place
00:49.19hardwiremog_work: is the makefile in a state of migration toward something more sane yet?
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00:49.37hardwireor is donated code welcome?
00:49.39mog_workwhats wrong with the way it is hardwire
00:49.48mog_workgo right ahead to the  bug tracker
00:49.51mog_workbugs.digium.com
00:49.55hardwireI am in the lists..
00:49.59mog_workbut no code is accepted without disclaimer
00:50.02hardwirerather.. I have an account there w/ agreement
00:50.10hardwireagreement/disclaimer
00:50.22YoMamamog_work: so if they aren't supporting wcfxsusb, why don't they just take it outta the distribution?
00:50.42mog_workwell
00:50.51infinity1hardwire: huh
00:50.51mog_workthere is a lot of stuff in there digium doesnt support
00:50.55hardwiremog_work: there are a few things that need to be changed.. including changes to zaptel and libpri's Make system
00:50.59hardwireinfinity1: no worries
00:51.00mog_worklike r2
00:51.06mog_workwhats wrong hardwire
00:51.21hardwiremog_work: I am afraid this list is too daunting.. and you would hate me forever and ever.
00:51.25infinity1mog_work: he's not hard wired.
00:51.28hardwireI am not a nazi about these things :)
00:51.34infinity1heh
00:51.34mog_work<PROTECTED>
00:51.36ginventARGH!!! this stupid fax thing is killing me.
00:51.41hardwiremog_work: I will make a list.. check it twice..
00:51.46mog_workokies
00:51.49ginventThe fax beeps, starts going... then just hangs up.
00:51.51*** join/#asterisk alephcom (n=Miranda@207.34.97.130)
00:51.52hardwirethen you can deside if I'm naughty or nice.
00:51.54ginventNo message... nuttin.
00:52.00mog_workokies
00:53.59YoMamaanyone ever tried this follow me thing?
00:54.14mog_workit does look sexy yomama
00:54.19IronHelixurl?
00:54.30mog_workits on the bugtracker
00:54.52IronHelix:)
00:55.13YoMamamog_work: there's a wiki for it...so it doesn't work yet?
00:55.21mog_workhuh?
00:55.23mog_workoh
00:55.31mog_worki thought you were talking about app_followme
00:55.38mog_workwhich is a new feature in bug tracker
00:55.41mog_workthats sexy
00:55.48YoMamamog_work: i think i might be...http://www.voip-info.org/wiki/view/Asterisk+Tips+follow+me
00:56.19mog_workhttp://bugs.digium.com/view.php?id=5574
00:56.22mog_worklook at that
00:56.27mog_workit takes all the work for ya
00:56.45*** join/#asterisk logicalonline (n=upirc@m495e36d0.tmodns.net)
00:56.53Ariel_well does not look like 1.2 is actually released yet like the heading says.
00:57.26YoMamamog_work: cool
00:57.45YoMamamog_work: it's be cooler if u had the option of recording the person's name
00:58.16IronHelixor would saydigit the caller id number
00:58.17infinity1is it possible to set the caller id num on the sip device? i don't see a way to do it. it seems to inherit the autentication information
00:59.39docelm0Is Zoa in here?
00:59.44YoMama"Press 2 to follow me", User presses 2, and then is asked "Please say your name"
00:59.53YoMamathen it records their name and then tries a list of #'s
00:59.58YoMamaand will announce the caller...
01:00.40*** join/#asterisk ginvent (n=joseph@adsl-68-125-225-108.dsl.sndg02.pacbell.net)
01:00.53ginventanyone know how to debug asterisk with spandsp?
01:01.06ginventMore importantly can someone tell me how?
01:01.38docelm0«ginvent» whats your prob?
01:02.12docelm0ohh ya.. for people who dont know.. IAX/SIP Termination US48 for .0092 @ www.plainvoip.com :) TY
01:03.18IronHelixwow
01:03.19IronHelixthats cheap
01:03.39ginventI use teliax.
01:03.41IronHelixalthough they charge more for tollfree termination (calling 1800 numbers) than they do for calling normal #s
01:03.45ginventIs plainvoip better?
01:03.52docelm05 month old startup havent done much advertising
01:03.57_Sam--hmmm any place with a grammar error on their first sentence....
01:03.59docelm0I am gonna fix that.. :) have to setup the route..
01:04.02_Sam--"to see where we have came from"
01:04.15_Sam--plainvoips site
01:04.19docelm0Huh? I typed that..
01:04.25_Sam--Being that we are new to customers check out our history to see where we have came from.
01:04.27IronHelixdoc- plainvoip is your company?
01:04.34_Sam--take much english?
01:05.04docelm0Its a run on.. My bad.. I will go fix it. I wrote this site at 2am gimme a break.. didnt have any Mt. Dew..
01:05.09docelm0No.. Company I consult for..
01:05.14IronHelixah
01:05.14IronHelixcool
01:05.33*** join/#asterisk nitestarr (n=knightst@cpe-24-33-15-250.midsouth.res.rr.com)
01:06.36asterboyAnyone try that new yak service? http://www.yak.com
01:06.50asterboy$20 unlimited NA
01:07.04Ariel_what a name yak
01:07.21asterboyDID can end up being more expensive.
01:07.40docelm0better?
01:07.53asterboydunno
01:07.54IgbothomI wonder if the Tibetans will eat up this service?  :)
01:08.05_Sam--wonderful
01:08.10docelm0I try
01:08.11docelm0:)
01:08.16docelm0And the comment is true..
01:08.19_Sam--but its still not grammatically correct
01:08.27_Sam--service and support are two things...plural
01:08.31_Sam--so you cant use the word "is"
01:08.34_Sam--service and support ARE
01:08.47docelm0Hence the reason I fixed it.. What are you a fucking english major?
01:08.51asterboyIf they keep lowering the price like that, the little A-Z companies will loose ground.
01:09.02_Sam--well you could be grammatically right too...because the goal is #1..one thing
01:09.36docelm0I think is sounds better..
01:09.40infinity1is regexten depricated? the voip-info wiki page is gone
01:09.47_Sam--yeah im just being a d*ck anyway
01:09.49docelm0Im not a english major.. I have a BS in EE, IT Network Engineering
01:09.50_Sam--sorry
01:10.03docelm0and certs out the ass
01:10.17_Sam--pet peeve of mine is spelling and grammar cant help it
01:10.24asterboygrammer/spelling and IRC....oxymoron?
01:10.45docelm0Look now.. betteR?
01:10.58docelm0:)
01:11.19_Sam--what about this blurb
01:11.20_Sam--Personal accounts
01:11.20_Sam--What Email Service Provider has
01:11.20_Sam--to offer you.
01:11.23_Sam--what does that mean
01:12.09YoMamastupid thing
01:12.13YoMamai shoulda just compiled regular ircii
01:12.20_Sam--also you have a link "Sales Support Center
01:12.26_Sam--is it for sales or for support
01:12.31*** join/#asterisk steve___ (n=steve@66.38.193.254)
01:13.02docelm0Ya.. I dont know.. I must have been outa it.
01:13.18*** join/#asterisk ekuminek (n=eduardok@202.55.146.70)
01:13.29_Sam--hey no worries its not like i critique websites all day or anything
01:14.06docelm0What do you do for a living?
01:14.11_Sam--its alot of what i do actually
01:14.12docelm0critique websites? :)
01:14.14_Sam--i own a dotcom thing
01:14.17docelm0ahh
01:14.23_Sam--so i mostly critique our own
01:14.55docelm0Well this is fine.. I am not a english person.. I just put it together and let 'er rip
01:15.03docelm0But should be happier now
01:15.10docelm0the website that is..
01:15.15_Sam--i bet you'll get 1000 signups
01:15.54Kattymew.
01:16.06docelm0Would be nice.. Since I just built a 10 server asterisk cluster
01:16.09steve___"not an english person"   :)
01:16.17steve___ruff.
01:16.20Katty...
01:16.27Igbothomyour mother was a hampster...
01:16.30KattyFor all of you who don't speak kat, mew means hi.
01:16.38docelm0ohh ya.. for people who dont know.. IAX/SIP Termination US48 for .0092 @ http://www.plainvoip.com :) TY
01:17.02docelm0and to think I just thought all this time my cat was dumb..
01:17.04_Sam--id like to make some test calls
01:17.06steve___for all those who do not speak steve, ruff means i need to shave  :)
01:17.10_Sam--can you give me a testing account?
01:17.12Kattydocelm0: i said kat, not cat. kthx
01:17.19docelm0You get .25c free.. Just signup
01:17.37Kattydocelm0: also, cats are /quite/ intelligent.
01:17.41Igbothoma whole quarter of a cent free?
01:17.44docelm0Tell that to mine..
01:17.50docelm0what?
01:17.52_Sam--what is the hostname i would connect to?  id like to check the route and latency
01:17.52Kattydon't have to be entertained like a big dumb dog
01:17.53steve___docelmo, steve is not :)
01:18.04docelm0no.. 25c my bad..
01:18.08docelm025cents
01:18.09Igbothom:)
01:18.10Kattywho slobbers and digs, and eats anything
01:18.17docelm0MY FUCKING LAB!
01:18.25Igbothomand appreciates it when you feed it
01:18.39docelm0I had a flannel blanket.. Now I have an affigian..
01:18.47docelm0damn dog ate the shit outa it
01:18.47Kattydocelm0: simmer down dude, we don't need that sort of lanugage in here (=
01:18.48steve___hahaha
01:19.02*** join/#asterisk Vivaldi (n=Antonio@adsl-99-237.37-151.net24.it)
01:19.05Vivaldihello
01:19.20Kattyhihi
01:19.22docelm0Yes Ma'am.. forgot there are now females using asterisk.. damnit..
01:19.28steve___mewmew
01:19.28docelm0I cant cuss anymore.. :P
01:19.30Vivaldidoes asterisk have the features of gnoememeeting, so that i can ip to ip voice calls?
01:19.33Kattydocelm0: that's a good boy.
01:20.00docelm0ok I am gonna leave the comment off the channel I am thinking of right now..
01:20.03steve___Vivaldi, snicker
01:20.10KattyVivaldi: you mean sip?
01:20.19docelm0You think I am bad here.. Ask twisted about our adventrue @ astricon
01:20.28VivaldiKatty, i do not know...
01:20.29Kattydocelm0: i can't, he's in vegas
01:20.37docelm0ahh
01:20.38Vivaldiis it possible to call another user on internet?
01:20.38Kattydocelm0: i take that back, i can
01:20.46IronHelixviv- what feature do you need?
01:20.49docelm0I can call him too.. :P
01:21.01Kattyyes, but i can beepbeep him
01:21.10docelm0ewww.. sprint
01:21.12docelm0cucca
01:21.16Kattyno, sprint does not beepbeep
01:21.19Kattythat is nextel, kthx
01:21.27docelm0And who do you think owns nextel now?
01:21.37Kattywe're an indirect dealer
01:21.37IronHelixgnomemeeting uses h.323, asterisk supports that protocol with a bit of tweaking...  SIP supports much of the same stuff
01:21.47Kattyand it is nextel who still uses the iden network
01:21.48Kattynot sprint
01:21.53docelm0I am not a 6'6" dog..
01:22.04docelm0iden.. haha
01:22.05IronHelixi'd be happier with sprint if they would open their PTT network
01:22.08IronHelixbut it will never happen
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01:22.25IronHelixeven their latest tv ad was pointing out how all the PTT users are on nextel
01:22.25docelm0I prefer GSM.. The ONLY way to go
01:22.27Kattyoh, you're not? ;)
01:22.33Katty</sarcasm>
01:22.37IronHelixyeah i like gsm as a technology better
01:22.45IronHelixbut gsm data has crappy latency
01:22.53IronHelixUMTS might be a bit better tho
01:22.58docelm0I have used it in 4 countries.. I have no complaints
01:22.59IronHelixi get 800-1300ms over gprs
01:23.06Kattydocelm0: two bad GSM can't handle two way radio
01:23.19IronHelixactually i think umts migh have it as an extension
01:23.25Kattydocelm0: unless you can its crappy 6 minute lag two way radio
01:23.37IronHelixi saw some PTT phones at a cingular store the other day...  not sure if they do it as a UMTS thing or voip like verizon did
01:24.26KattyIronHelix: Nextel has the patent on that
01:24.34KattyIronHelix: and nextel has sued several companies over it too...and won
01:24.46KattyIronHelix: while some companies, like verizon, have a similiar thing...it's not on IDEN
01:24.54KattyIronHelix: which means the latency is horrible...being around 6 minutes.
01:25.16KattyIronHelix: it's crap compared to nextel
01:25.34IronHelixwell the verizon one was just an app on the phone that worked over the data link...  i know nextel / motorola iden division has a patent on iden PTT but do they have a patent on PTT as a concept?
01:25.38infinity1er ...how does regexten work? i can't find any documentation that breaks it down
01:25.54KattyIronHelix: from what i hear, apparently they do
01:26.12IronHelix:\
01:26.17Kattythey've even sued vonage
01:26.56IronHelixthat was over a voip thing tho
01:27.26Kattyindeed
01:27.34Kattybut nextel has been suing everyone they can get their hands on
01:28.08IronHelixhmmm
01:28.34docelm0Vonage just sucks.. :) They diserve it.
01:29.04docelm0also on the Plainvoip site I should have some DID's available tonight for locations
01:29.34IronHelixFootball City DID's coming soon!  ---?
01:29.49*** join/#asterisk CarstenG (n=irc@cpe.atm2-0-1011085.0x50c67b02.kd4nxx11.customer.tele.dk)
01:30.03asterboyYa but can they beat $10/month for unlimited talk?
01:30.08IronHelixalso i love the Bullerins and Updates on Network forum
01:30.14IronHelixasterboy where are you seeing that?
01:30.25asterboyhttp://www.yak.com
01:30.30xbmodderDoes anyone here have "Iconnecthere" or "deltathree"
01:30.34docelm0$8 per DID
01:30.37asterboy$25 for unlimited NA
01:30.38docelm0with E911
01:30.57asterboydocelm0: but is there a per minute charge?
01:31.15docelm0hay iron where you from?
01:31.23IronHelixasterboy 10/mo and 25/mo are a big difference
01:31.35asterboy$10 is for local
01:31.37IronHelixConnecticut
01:31.38IronHelixahh
01:31.42asterboy$25 is anywhere
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01:32.35asterboy$80 for equipment though...but thats cheap.
01:32.42IronHelix$10 gets you '10 hours' of calling...
01:32.52docelm0ok well time to setup the toll free route.. :)
01:32.55CarstenGi am trying to use an x-ten eyebeam with xtunnels2 and asterisk, but i can't register on my asterisk, i get the message "Registration from 'Carsten<sip:103@example.com>' failed for '127.0.0.1'
01:33.00IronHelixaltho if it wont let me register my * box its worthless IMHO
01:33.21barretjhi, i am interested in setting up a system for a small business where customers can call a single number and get a menu that lets them choose a department/representative to talk to, and then get forwarded to the person they chose
01:33.23asterboy$10 from another company...seen it on bigreddeals or something like that.
01:33.27barretjcan i do this with asterisk?
01:33.31IronHelixyes
01:33.32IronHelixyou can
01:33.33IronHelixvery easily
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01:33.44barretjdo i need multiple phone lines?
01:34.02asterboylol um thats the whole idea behind a pbx.
01:34.06tamp4xhow do i prevent users from dialing to each other (same context)
01:34.08IronHelixonly if the people that they will talk to are not on the system
01:34.28barretjno, i mean, do i need to buy multiple lines from the phone company?
01:34.39IronHelixIE if you have employees that have IP phones (like a normal phone but theres an ethernet plug where the phone plug would be) you only need one
01:34.42IronHelixone line that is
01:34.53barretjcool
01:35.02IronHelixif you have two lines you can forward a call, ie so a caller can transparently call an employees mobile phone by extension
01:35.40IronHelixas a general rule- asterisk will take any type of line or channel (including T1/e1/j1/PRI/BRI or voip such as SIP provider or phone or IAX) and take calls however you want and do whatever you want with them
01:35.44asterboybut that line will be tied up.
01:36.07barretjah, ok
01:36.08IronHelixthe limitation is that analog lines will only support one call per line.  This is a limitation of an analog line, not *.
01:36.17tamp4xit seems exten => s doesnt do that trick??
01:36.33CarstenGwhat does the message Nov 17 02:33:22 NOTICE[81925]: chan_sip.c:7561 handle_request: Registration from 'Carsten Gram<sip:103@cgw.dk>' failed for '127.0.0.1'
01:36.34IronHelixso if you have one analog line, and a call comes in, you can talk to that guy but if another guy calls they wont be able to talk to somebody else
01:36.35IronHelixsee?
01:36.43barretjyes
01:37.07IronHelixif you have a robust internet connection, you can purcahse phone service over the internet and it will go straight to * as data
01:37.47barretjhow much does the hardware that asterisk needs cost?
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01:41.32docelm0damn now thats a split
01:41.32steve___that quieted things down  :)
01:42.08IronHelixsome more robust business-class providers do not have a channel limit
01:42.08CarstenGwhat does the message mean: Nov 17 02:33:22 NOTICE[81925]: chan_sip.c:7561 handle_request: Registration from 'Carsten Gram<sip:103@cgw.dk>' failed for '127.0.0.1'
01:42.22nick125w00t /me got his asterisk box working :D:D
01:42.22IronHelixso if you have the bandwidth for it you can have 100's of calls going in and out at the same time, they just charge you more
01:43.03Math`IronHelix: providers that charge per-minute are usually allowing a lot of concurrent calls too :P
01:44.05shido6we dont
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01:44.35asterboywhat happened?
01:44.40IronHelixnetwork problems
01:44.49IronHelixfreenode is de-node-ifying
01:45.11asterboyhaven't seen that eva
01:46.00Math`you never saw netsplits?
01:46.20barretjyou obviously havent been using irc for long
01:46.23asterboySeen that...but I lost all channels and every user was cut
01:46.24docelm0hay iron just updated tollfree on plainvoip..
01:46.25docelm0:)
01:47.00Pikorois there a walkthrough somewhere on how to set up a TDM400P for just incoming calls?
01:47.01asterboySo is 1.2.0 out yet? Nothing on the front page.
01:47.02*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
01:47.27Pikoroall i get when i do a zap show channels is:  pseudo            from-outside
01:48.04Pikorothe kernel module is loaded and /etc/zaptel.conf seems to be configured correctly...
01:48.26asterboyWhere is the 1.2.0?
01:49.20asterboyTopic: Asterisk 1.2.0 will be released today!!! || http://www.asterisk.org
01:49.46docelm0The great kram awakens
01:49.52kramsomething like that
01:49.53Math`asterboy they didnt say for which timezone :P
01:50.00asterboylol
01:50.03docelm0kram how's capt crunch?
01:50.18kramheh okay i think, i haven't seen him since internet telephony
01:50.25docelm0ohh my bad.. That was kristians problem.. :)
01:50.32asterboythat cereal gives me bad gas
01:50.37kramheh
01:50.58docelm0Ya I wanted to come to that.. but I was too busy setting up Mera over the last week
01:51.10kramit's all good
01:51.11hypa7iaoh haha apparently he caused a lot of trouble at HOPE last year too eh
01:51.24docelm0hypa who?
01:51.29docelm0Crunch?
01:51.41hypa7iayup
01:51.53docelm0What was HOPE?
01:51.54hypa7iacamped out at the hacker halfway house and wouldn't leave :s
01:52.04hypa7iahackers on planet earth (2600 conference)
01:52.30docelm0I wanna go to that someday!
01:52.39docelm0Brush up on my linux skills.. :P
01:52.41hypa7iait was so much fun last year
01:52.57docelm0I bet not as much fun as astricon
01:53.16hypa7iait was right up there
01:53.23hypa7iai partied way harder at astricon tho :p
01:53.23docelm0haha
01:53.43docelm0You were there this year?
01:53.52*** join/#asterisk adelas (n=booger@rrcs-24-199-21-141.west.biz.rr.com)
01:53.58docelm0I am horrible with names.. Did we meet up?
01:53.59hypa7iait's only every two years
01:54.26hypa7iadocelm0: i was out drinking with you the friday night
01:54.31hypa7ia:p
01:54.49docelm0Ya.. I was supposed to goto a hockey game ended up getting plastered off of 8 Long Island Ice Teas w/ Double shots
01:54.56docelm0I dont remember much of that night.. Who are you?
01:54.59Math`lol
01:55.08IronHelixheh
01:55.25docelm06 Bud Weiser/Rolling rocks and 2 Greatful Dead's
01:55.37hypa7iai was the girl
01:55.40docelm0Man Matt needs to be in here
01:55.48docelm0ohh NO SHIT! WHATS UP!
01:55.55hypa7iahahaha
01:56.11docelm0Considering you were the only one I do remember that much
01:56.18hypa7ialol
01:56.23docelm0Remember Matt the Aussie?
01:56.26docelm0dark hair?
01:56.39hypa7iayuppers
01:57.02docelm0and daniel.. :) Dude.. those guys could bathe in alchol and not get drunk.. They can put it down
01:57.29docelm0ok thats cool.. Now I remember you.. its coming back to me a little.. hehe
01:57.33docelm0:P
01:57.45docelm0ok need Mt. Dew and Niccotine.. brb
01:57.47*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
01:57.58hypa7ialaters :)
02:01.53*** join/#asterisk pandect (n=ender@69-162-25-193.stcgpa.adelphia.net)
02:01.59pandectHi
02:02.04IronHelixwelcome
02:02.18pandectI'm about at wit's end trying to get an SPA-1001 to work with an Asterisk system
02:02.36IronHelixthat we can help with, we get that alot
02:02.45IronHelixso tell me about your wit, and how close to the end you are
02:02.48pandectEssentially -- the asterisk system is properly connected to the SIP trunk, and incoming calls are routed to the phone properly
02:03.09pandectThe SPA-1001 is saying that it is registered properly with the asterisk system
02:03.32pandectBut, when I pick up the phone, no dial tone... and when I dial the number, the phone rings, but when I pick it up, it doesn't answer, and the calling phone keeps ringing
02:03.43pandectI'm thinking a communication issue between the two? :-/
02:03.53pandectnot sure...
02:04.17harryvvasterisk can talk to pulse style phones?
02:04.22harryvvInteresting
02:04.30IronHelixpandect- do you get any wierd messages on asterisk console?
02:04.57pandectaha, now that is interesting
02:05.02pandectI just discovered the 9th console :)
02:05.17IronHelixor do asterisk -r from any other console
02:05.24*** join/#asterisk hhoffman (n=hhoffman@tor/session/x-4a0d33b8950dc7b1)
02:05.37pandect"== spawn extension (...) exited non-zero at ..."
02:05.46IronHelixthats normal, it means you hung up
02:05.59IronHelixdo set verbose 3
02:06.05IronHelixthen make a call and see if anything stands out
02:06.18IronHelixalso- is nat involved anywhere with this?  are the spa and * on the same netowrk
02:06.45pandectspa and * are directly connected
02:06.51pandecthow can I set verbose to 3?
02:07.06IronHelixtype at the asterisk console
02:07.08IronHelixset verbose 4
02:07.11IronHelix*set verbose 3
02:07.50pandectok, got it
02:08.01IronHelixnow make a call and see what pops up
02:08.06IronHelixsee if theres anythign interesting
02:08.18IronHelixdon't paste more than 2-3 lines of it here
02:08.27IronHelixif you dont know what it means, copy it all and put it on www.pastebin.ca
02:09.03pandectno, nothing seems interesting really
02:09.20pandectit all works properly, no error messages when I try to answer the phone and it doesn't work
02:09.23*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
02:09.29pandectand then the messages continue when it forwards me to voicemail etc
02:09.54IronHelixpastebin your extensions.conf and sip.conf (take out passwords of course)
02:09.56pandectWhen I try to answer the phone, I hear a click, then another click, like it was hung-up, but there aren't any messages saying what happened
02:10.01pandectokie dok
02:10.17Flautowhta is crontab
02:10.41IronHelixcron is a system that will execute tasks at predefined times
02:10.49IronHelixvery useful for running automated upgrade or backup systems
02:11.05Flautooh, how would i use it
02:11.16IronHelixtype crontab -e
02:11.17Flautoi am trying to use wakeups.agi
02:11.20{zombie}pandect: sounds to me like an issue between the phone and the sipura - sipura hasn't noticed you've taken the phone offhook
02:11.20JonR800man crontab
02:11.25IronHelixthat brings up a text editor that lets you edit the crontab file
02:11.29Flautocrontab -e and the file
02:11.30IronHelixtheres a specific syntax for how that works
02:11.41IronHelixno
02:11.43IronHelixjust crontab -e
02:11.55pandect{zombie}: The sipura... hmmm... actually could be true, as I'm using upgraded firmware on a SPA-1001 that has 2 lines, and using line 2
02:12.12IronHelixthat lets you edit your crontab
02:12.25docelm0BACK!
02:12.28IronHelixwb
02:12.34IronHelixhow was your chemical intake?
02:12.46{zombie}pandect: does line 1 work?
02:13.15docelm0Good.. working on caffeen now
02:13.17Flautoyes, that gives me a file to edit
02:13.28IronHelixthat file si your crontab
02:13.34IronHelixin there you put stuff for cron to act on
02:14.04pandect{zombie}: awesome. Just reconfigured the SPA to use line 1 instead of line 2, and it works now!
02:14.05JonR800Flauto: man crontab or google crontab. :)
02:14.08IronHelixhttp://www.adminschoice.com/docs/crontab.htm#Crontab%20file  info on how
02:14.12nextime<PROTECTED>
02:14.14IronHelixthats the syntax you need
02:15.51Nugget<G.W.Bush> How many is a brazillion?
02:16.05IronHelixlol
02:16.51alephcom:-)
02:17.15mog_homeheh
02:17.29Pikoro2
02:17.54barretjhahaha
02:19.11Pikorois there a walkthrough somewhere on how to set up a TDM400P for just incoming calls?
02:19.48IronHelixdifficult, because the call then has to go somewhere
02:19.54IronHelixdo you have zaptel set up?
02:19.59Pikorowell, all my outbound is via sip
02:20.04docelm0«Pikoro» whats the issue? Its all in your dialplan
02:20.14Pikoroall i get when i do a zap show channels is:  pseudo            from-outside
02:20.16docelm0«Pikoro» Your probably paying too much.. :)
02:20.30Pikoropaying too much?
02:20.34docelm0Did you configure Zaptel and Zapata?
02:20.40IronHelixi was waiting to see if docelmo would pimp his company...
02:20.47Pikorothe kernel module is loaded
02:20.49IronHelixpik- you need to configure zaptel.conf and zapata.conf
02:20.50docelm0Hay minutes are minutes
02:20.58Pikorozaptel.conf is configured
02:21.07Pikoroand i think zapata.conf is configured properly
02:21.08docelm0What about Zapata?
02:21.16Pikorono errors in zapttool
02:21.26Pikorobut no channels show up in the cli
02:21.28docelm0Well if your seeing pseudo then your good to go..
02:21.46Pikoroand i got channels => 1-4 in zapta.conf
02:21.46IronHelixwhat do you get when you do ztcfg -v
02:21.51docelm0then your Zaptel.conf is probably not right I had the same issue with my TDM410P
02:21.53*** join/#asterisk supaigtr (n=yurplsl@152.53.17.1)
02:22.08IronHelixztcfg -v should show your channels
02:22.26PikoroChannel map:
02:22.26PikoroChannel 01: FXO Kewlstart (Default) (Slaves: 01)
02:22.26PikoroChannel 02: FXO Kewlstart (Default) (Slaves: 02)
02:22.26PikoroChannel 03: FXO Kewlstart (Default) (Slaves: 03)
02:22.26PikoroChannel 04: FXO Kewlstart (Default) (Slaves: 04)
02:22.27Pikoro4 channels configured.
02:22.49IronHelixare your channels configured as signalling fxsks?
02:22.57Pikoroyes
02:23.00[TK]D-FenderPikoro : If your * box is accessable I'll volunteer to help get you set up.
02:23.29IronHelixor if its not, pastebin your zapata.conf and zaptel.conf files and i'll look them over
02:23.30Pikorofender: it is.. just a moment :D
02:24.31docelm0What a guy!
02:24.50docelm0I had so much trouble I ended up calling Digium for a bit of help to set mine up..
02:25.08docelm0My channel bank was a pain in the ass
02:25.44harryvvWhat does digium charge for consulting fees
02:25.59hypa7ia$175/hr i think
02:26.06hypa7iafor support anyway
02:26.08IronHelixyou get an hour or so of free phone time if you buy one of their interface boards i think
02:26.15harryvvI see
02:27.19IronHelixyeah if you buy a digium card they will help you get it going if you need help
02:27.37IronHelixbut if you need more help (dialplans, scripts, agi's, etc) you gotta pay them
02:28.36harryvvI went to talk to the head of a church and bridge all of there employment offices in different provinces and reduce the cost of all there calls. We was very interested in all the things it can do for his phone systems but I said there are consulting fees and it was cheap he kinda looked at me like what the $%? he never called back. He wants to update there pbx for more phones but nortel wants to charge a arm and a leg for there extra fe
02:29.49IronHelixhttp://www.digium.com/index.php?menu=service_category&category=installation  digium will compile install and configurize your stuff for about $750-1200 depending on what you need
02:30.07harryvvOhh I am aware of it.
02:30.08hypa7iawhich compared to bell / nortel is peanuts
02:30.13IronHelixvery true
02:30.17harryvvbell of canada?
02:30.17hypa7iacrushed peanuts even
02:30.21hypa7iaharryvv: yup
02:30.24harryvvI see
02:30.24IronHelixnortel will probably charge you like $2500 just to answer the phone
02:30.48IronHelixplus an extra $500 if you want there to be a *person* on the other end
02:30.54*** join/#asterisk ceph (n=amit@adsl-065-012-197-229.sip.bct.bellsouth.net)
02:30.56IronHelixplus another $350 if you want them to speak english
02:30.58hypa7ialol IronHelix
02:31.01Nuggetplus a $100 "english language facilitation charge"
02:31.07Nuggetheh
02:31.08hypa7ianah, they haven;t done much offshoring
02:31.10harryvvI was disapointed but not suprised by the responce I got from this head manager of these employment offices. He said alot of people that enter the door make alot of long distance calls and wanted to cut down on the cost of those calls.
02:31.29harryvvLong distance calls for job searches.
02:31.35hypa7iahaha yea right
02:31.38hypa7ia:s
02:31.56IronHelixharry- even if you dont upgrade your pbx, you can still save money on phone service by replacing their existing lines with ATAs and service from any one of a hundred ITSPs
02:31.56docelm0Geesh.. I only charge $1500 down here for a base asterisk install and that includes the server..
02:32.07harryvvWhere is down there?
02:32.11hypa7iadocelm0: never compete on price alone, man
02:32.13docelm0Tampa, FL
02:32.22harryvvis tampa close to miami
02:32.23hypa7iatampa, nice
02:32.29docelm0Yes.. 3 hours
02:32.32hypa7iamy grandparents used to have a place in venice
02:32.37harryvvArial lives in miami
02:32.44docelm0ahh..
02:32.55hypa7iai've been to tampa like 15 times
02:33.00hypa7iaover the years :)
02:33.09cephIm in Boca...are there many people that do asterisk offerings in St Pete/Tampa area?
02:33.11docelm0But for the 1500 that includes the dialplans, IVR, config etc. No phones or TDM cards..
02:33.24docelm0No.. I am about the only one
02:33.28cephsweet
02:33.43docelm0And I dont have much time during the day since I work full time
02:33.56harryvvdoc, so thats good. what do the local proprioratory companys charge?
02:34.37hypa7iaharryvv: usually around 5-10x that
02:34.43hypa7iaat least in my experience :/
02:34.50cephWhat kind of phones do you use docs?  I am looking to do an install for a friend.  Cisco too much.
02:34.52harryvvis that with phones?
02:35.00hypa7iaharryvv: nope
02:35.11docelm0Local's charge abour 175 per hour
02:35.20hypa7iaa cisco install for like 50 users is going to be like 60k CAD plus.
02:35.22IronHelixceph- grandstream, snom, aastra/sayson, all work good
02:35.36IronHelixsipura/linksys too
02:35.40docelm0and traditionals PBX's start around 10-15K or so
02:35.53[TK]D-Fenderceph : how many phones, used where?
02:36.03barretjis it true that 2.4Ghz wireless phones can interfere with 802.11b networks?
02:36.16harryvvdoc, local propriotory companyes charge 175 per hour?
02:36.23docelm0yes
02:36.23*** join/#asterisk TheCops (i=nobody@got.securebinary.com)
02:36.26TheCopsHi
02:36.26IronHelixbarretj- yeah, but it depends on which phone and what wifi channel
02:36.31harryvvdoc, nice
02:36.35IronHelixsadly, theres no easy guide to what will or wont interfere with what
02:36.36harryvvleaves room to charge less
02:36.40docelm0The company I work for has a NEC when dude comes out to look at the system he charges us 175 per visit
02:36.48TheCopsHow can I recognize a 3.3v slot and a 5v slot PCI for T1 card ?
02:36.49harryvvgood grief
02:36.49harryvv:)
02:36.57docelm0I finally talked my boss into a 4 partition Asterisk system with 100 Grandstream GXP2000's
02:37.00barretjwhat about microwave ovens, do they interfere with wifi?
02:37.07IronHelixhttp://www.digium.com/index.php?menu=whatpcislot
02:37.12harryvvnooo barretj
02:37.14alephcomyeah, big time. :-)
02:37.19IronHelixbarretj- only crappy ones taht leak.  i've never had a problem
02:37.20cephA small office PBX....one phone per user...prob VoIp phones.
02:37.31IronHelixhow many users
02:37.41ceph5
02:38.21cephI am just learning this stuff though....read half of the Oreilly book this afternoon..lotta loose ends trying to figure
02:38.32IronHelixgrandstream gxp2000, sipura spa-841/941, aastra 9122i,
02:39.04barretjceph: what book?
02:39.05IronHelixpolycom 301 maybe
02:39.13docelm0How many users what?
02:39.20IronHelixfor ceph's install
02:39.28IronHelixhe's diving in head first
02:39.30IronHelix:)
02:39.31docelm0ceph where are you?
02:39.41docelm0at least he isnt using A@H
02:39.49IronHelixyeah :)
02:39.51ceph.  the asterisk: future of telephony.    About 5 users.  I am in Boca.
02:40.14docelm0Im just north of you if you need any help drop me a line sometime.. I love helping out new asterisk users..
02:40.53barretjw00t! that book is free online!
02:41.05*** join/#asterisk asteriskmonkey (n=phil@HSE-Toronto-ppp300017.sympatico.ca)
02:41.06IronHelixyuppers
02:41.08asteriskmonkeylo
02:41.09docelm0it has been for awhile now
02:41.11cephShut up!  I just discovered that about 15 minutes ago.
02:41.13IronHelix<3 creativecommons
02:41.14barretjoreilly is nice like that
02:41.14Pikoroook ook
02:41.25docelm0I got mine @ astricon
02:41.37cephI mean I was reading it online.  Emmm that;s right.
02:41.43IronHelixsigned by anyone cool?
02:42.03asteriskmonkeyanyone know alot about iax TOS ?
02:42.08*** part/#asterisk Samoied (n=Samoied@201.14.169.178)
02:42.20*** join/#asterisk xbmodder_lappy (i=nobody@unaffiliated/xbmodder)
02:42.48cephThanks doc for the offer.
02:42.53[TK]D-Fenderceph : I'd probably suggest Linksys SPA-941's for a small office.
02:43.04xbmodder_lappyin the auto-dial files for "Application" can I put "Application: /usr/bin/test.asterisk"
02:43.20barretjlol, Paperback: 404 pages
02:43.24IronHelixapplication is the asterisk application
02:43.29IronHelixnot the asterisk binary
02:44.03IronHelixasteriskmonkey- http://www.voip-info.org/wiki/view/asterisk+config+iax.conf has a good explanation of iax TOS
02:45.17cephI suppose, I am supporting Oreilly anways...a good cause.
02:45.39*** join/#asterisk SplasPood (i=nobody@paravolve.net)
02:46.25*** join/#asterisk SplasPood (i=nobody@paravolve.net)
02:51.22*** join/#asterisk marc324 (n=marc3234@206-248-157-92.dsl.teksavvy.com)
02:51.57marc324is there a method for storing voicemail message0000.txt into a db table, instead of txt file?
02:54.22*** join/#asterisk ceph (n=amit@adsl-065-012-197-229.sip.bct.bellsouth.net)
02:57.35Pikorogrr
02:57.41IronHelix?
02:57.54Pikoroloader.c: Loading module chan_modem.so failed!
02:57.55*** join/#asterisk file[laptop] (n=jcolp@mctnnbsa24w-142167049176.nb.aliant.net)
02:58.05Pikoroi just rebuilt from head
02:58.11IronHelixwhy are you usign chan_modem anywya?
02:58.36Pikorodunno.. i tried noload and it fails with some broken pipe audio error
02:58.43asteriskmonkeyironhelix: thanks read that one already was looking for more so .. something along the lines of when is it good to use TOS
02:58.56asteriskmonkeyand in what circumstance do you use what type :P
02:59.12IronHelixthe main reason for TOS is to give your voice packets priority
02:59.16Pikoro[chan_modem_i4l.so]Ouch ... error while writing audio data: : Broken pipe
02:59.29IronHelixnot all routers and switches understand TOS tags
02:59.37Pikorothat's if i noload chan_modem
02:59.57IronHelixthe idea is that the ones that do will put your voice data in front of everybody else's kazaa downloads
03:00.06asteriskmonkeyby default if no tos is defined in iax.conf its set to none right?
03:00.15IronHelixi think so
03:00.33asteriskmonkeysp it would have no adverse effect enabling it to lowdelay/high through put?
03:00.34IronHelixpik also noload chan_modem_i4l.so
03:00.38Pikorok
03:00.57IronHelixmay also have to noload chan_modem_aopen.so chan_modem_bestdata.so
03:01.12IronHelixcorrect, setting it wont make anythign worse
03:01.24Pikorook, that worked
03:01.25IronHelixits mostly useful if you're in a corporate witha  limited amount of bandwidth
03:01.25Pikorothanks
03:01.31Pikoronow to test my config
03:01.33IronHelixor you have a router that understands it on your dsl line
03:01.48Pikorook.. outbound sip works
03:01.49Pikorogood deal
03:02.17Pikoronow for incoming pstn
03:03.22Pikoro[TK]D-Fender:  still here? :D
03:03.25[TK]D-Fenderyup
03:03.33Pikorolet me pm you the login info
03:03.34[TK]D-Fenderbarely.  My attention is pretty divided :)
03:03.36[TK]D-Fendersure
03:03.47IronHelixbbl
03:04.00IronHelixeh, wrong network
03:04.42marc324ne1 knows how to parse the voicemail 'message.txt' files, and display into db?
03:05.13Pikorowholly crapola
03:05.18Pikorothe zap channels show up now
03:05.20Pikorow00t!
03:05.51cephsorry to ask this as iax config file questions fly by...
03:06.12cephis there a router type device for plain ole' phones?
03:07.05docelm0RT31P2-NA
03:07.08docelm0Linksys makes em
03:08.09cephso there is such a thing....then is using VoIP phones a silly proposition for SOHO?
03:09.21*** join/#asterisk justinu (n=justinu@cpe-72-129-86-208.socal.res.rr.com)
03:09.27*** join/#asterisk samueltc (n=samuel@modemcable211.200-203-24.mc.videotron.ca)
03:09.40harryvvnot really
03:10.45docelm0I use Sip Phones for my home office and my RT31P2-NA for running my house
03:11.01xbmodder_lappyIs this a valid extension:
03:11.02xbmodder_lappy[inbound]
03:11.02xbmodder_lappyexten => 8883145779,1,Answer
03:11.02xbmodder_lappyexten => 8883145779,2,MP3Player(/mnt/music/Rob_Zombie_-_Feel_So_Numb.mp3)
03:11.03xbmodder_lappyexten => 8883145779,3,Hangup
03:11.20mog_homeahh dialplan
03:11.37Sedoroxhmmm
03:11.51harryvvkinda to simple
03:12.08harryvvtrying to understand dial plans?
03:12.11xbmodder_lappyyeah
03:12.24harryvvwell there is of course alot more then that.
03:13.00cephCool....thanks for the info.
03:13.52asteriskmonkeyso the tos throughput option is to give it best through put or to say minimize thorughput?
03:14.50*** join/#asterisk Inv_arp (i=junya@adsl-11-59-72.mia.bellsouth.net)
03:18.07cephthanks for help docelmo, et al.  Really appreciate it!  Be back soon.
03:18.21docelm0ok..
03:18.27docelm0Im cheap onsite also.. :)
03:18.54cephhaha.  Will keep you in mind  ;)
03:19.22docelm0for you man.. $50 a hour.. I usually charge $85 for onsite.. :)
03:19.25docelm0aww damn
03:20.11*** join/#asterisk santiago (n=santiago@208.195.215.124)
03:20.33*** join/#asterisk freespace-in (n=special@ppp-70-225-142-30.dsl.ipltin.ameritech.net)
03:21.47*** join/#asterisk Trionnis (i=fsn@12-215-250-176.client.mchsi.com)
03:22.07Trionnisany suggestions on where to get polycom or grandstream phones in bulk?
03:22.27docelm0Polycom or Grandstream?
03:22.29*** join/#asterisk RazorJack (n=RazorJac@d235-172-135.home1.cgocable.net)
03:22.34SkramXxbmodder_lappy: i feel for you.. a lot of dicks are going to call your 800 number excessively.
03:23.15xbmodder_lappyoh crap SkramX
03:23.27justinulol
03:23.35RazorJackFor the last week, I've been trying to debug why people behind a NAT do not have any audio..... I am using IPTEL Ser..... anybody able to help me debug whats going on?
03:23.41RazorJackDunno where to start debugging it.
03:23.54Trionnisyes docelm0
03:23.58Trionnisi.e. cheap
03:24.45docelm0I was saying buy directly from the company
03:24.47harryvvrazor, your firewall is not open for sip/rtp and your configureation may also be not configured properly.
03:24.51docelm0depending on unit size
03:24.52RazorJackcaller---[nat]---internet---asteriskserv..... I dont need RTPPROXY right? since the asterisk is on a public ip
03:24.53Trionnisah
03:24.54RazorJackright?
03:24.54Trionnisheh
03:24.57justinuRazorJack: are you running nathelper/rtpproxy?
03:25.01Trionnisgood point
03:25.13RazorJackjustinu; yup, dont need rtpproxy do i?
03:25.23RazorJackjustinu; thought rtpproxy was for if the asterisk was behind a nat
03:25.40justinuyou should use it
03:26.25RazorJackjustinu; how? just run it, doesnt seem todo anything... documentation on it seems far and few between
03:26.43justinuyeah, you just run it and tell it the interface(s) to listen on
03:26.58RazorJackI am right now, no change... :)
03:27.04RazorJackrtpproxy -l ip here
03:27.06justinuyou might need to restart SER
03:27.47RazorJacknope
03:27.54RazorJackill pastebin my ser.cfg, 1 sec
03:28.02justinuyou'll need to describe your topology more then
03:29.14[TK]D-FenderTrionnis : Polycom, hands down over alomst any other choice if your budget permits.
03:29.17RazorJackjustinu; eyebeam client, on a private ip, connected to a NAT, connected to internet then into a cisco router, and into the voip/asterisk box on a public ip
03:29.32justinurazorjack: so where does SER fit in?
03:29.50Trionnis[TK]D-Fender, ok thanks :)
03:29.54RazorJackjustinu; eyebeam connects to port 5080 which is ser
03:29.58harryvvwhich wifi phone has the longest lasting battery?
03:30.03harryvvlongest talk time?
03:30.05justinutrionnis: i agree with fender, polycom rocks!
03:30.12[TK]D-FenderTrionnis : I run 26 at my office.  they are rock solid stuff.
03:30.14RazorJackhttp://pastebin.ca/28974
03:30.21Trionnisok, good
03:30.33justinurazorjack: before we look at that....
03:30.40Trionnisthis is for a casino, so it's gotta be pretty solid, so that sounds like a plan
03:30.42justinuRazorJack: phone registers to ser, then what?
03:30.59Trionnisthanks for the input all :)
03:31.00RazorJackjustinu; you can call anywhere, just no INBOUND audio
03:31.07RazorJackjustinu; people can hear you
03:31.18harryvv[TK]D-Fender to bad the polycom phones have a backlit display
03:31.20justinuok, but tell me what happens after the phone registers to ser...
03:31.26justinuwhy are you running ser?
03:31.27harryvvdont have one that is
03:31.46justinuharryvv: the 601 display is amazingly bright even under dim lighting
03:31.52justinuit's a great LCD
03:31.54[TK]D-Fenderharryvv : I know, but for everything else you get, #1
03:31.58RazorJackjustinu; to get around a nat
03:32.09justinuRazorJack: you don't need SER to do that...
03:32.18justinuRazorJack: did you set nat=yes in sip.conf?
03:32.28RazorJackjustinu; yup
03:32.33harryvvjustin so the 601 is the only model that has a backlit display?
03:32.43*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
03:32.50justinuharryvv: 601 isn't backlit, it's just nice and reflective
03:32.57harryvvokay
03:33.11harryvvI just think thay should have a option to have a backlit display
03:33.13*** part/#asterisk Trionnis (i=fsn@12-215-250-176.client.mchsi.com)
03:33.34justinuRazorJack: why don't you turn on sip debug peer <phone_name> and make a call, and pastebin that.
03:33.46*** join/#asterisk tengulre (n=tengulre@222.90.66.156)
03:33.52RazorJackjustinu; ok, gimmie a few, that might take awhile
03:34.01justinui'd like to see the registrations too
03:35.11RazorJackjustinu; actually, before I do that... I can make my eyebeam work... but cannot receive calls
03:35.30RazorJackjustinu; using ser to try and fix that, but getting the no audio
03:35.43nick125anyone here know of a command line tool to change a ogg into a wav?
03:35.58RazorJackjustinu; the lesser of the two evils I guess is being able to dial, but cant receive calls
03:36.02justinuRazorJack: the sip trace will tell all :)
03:36.08nick125would sox do it?
03:36.13RazorJackjustinu; with or without ser?
03:36.22justinuRazorJack: leave it in
03:36.29RazorJackokey, gimmie a few
03:38.20pandectHere's a quick question: I've set up a "Digital Receptionist" (in AMP) -- how can I set where it will forward to if no options are selected?
03:38.30RazorJackhttp://pastebin.ca/28979
03:38.49RazorJackpandect; I created in extensions_custom.conf this entry
03:39.11RazorJacksorry [aa_1-custom]
03:39.11RazorJackexten => t,1,Goto(ext-group,1,1)
03:39.25RazorJacker take out sorry
03:39.25RazorJackheh
03:40.49pandectRazorJack: And that won't be reset by AMP next time it reloads the config?
03:41.13justinuRazorJack: looking
03:41.19RazorJackpandect; nope, what amp ver?
03:41.51RazorJacklater versions added the custom file
03:42.22justinuRazorJack: for whatever reason, your ser setup isn't rewriting the local IP of your phone in the SDP
03:42.37RazorJackjustinu; weird.
03:42.49RazorJackjustinu; where do you see that?
03:42.57justinuline 21
03:43.14RazorJacko=- 39001434 39001586 IN IP4 192.168.1.3
03:43.20RazorJackthats my local ip
03:43.47justinuso your * box is trying to send RTP to that IP
03:43.50justinuwhich obviously won't work
03:43.59RazorJackahhh
03:44.04RazorJackso is it the fault of my eyebeam?
03:44.30justinuwell, ser is supposed to fix that
03:44.45RazorJackI have a stun server.
03:44.49pandectRazorJack: I'm not exactly sure what version I have...
03:44.56pandectStill trying to figure it out
03:45.07pandect1.10.008
03:45.19RazorJackok
03:45.42RazorJackfixed
03:45.53RazorJackeyebeam had an advanced setting
03:46.00justinuRazorJack: not sure, your ser.config looks basically default
03:46.05RazorJack!@#$ send internal ip -> always
03:46.12justinuRazorJack: you shouldn't need SER then
03:46.20xbmodder_lappyDoes anyone here have nufone DID over SIP?
03:46.47RazorJackjustinu; well using ser because trying to make all these different voip devices work in harmony
03:46.59*** join/#asterisk MGSsancho (n=user@ppp-67-126-243-173.dsl.irvnca.pacbell.net)
03:46.59RazorJackjustinu; but its difficult... pap2-na, grandstream, cisco, eyebeam
03:47.01RazorJack...
03:47.04justinuRazorJack: unfortunatly, it's not a magic bullet
03:47.17RazorJackjustinu; I am realizing that
03:47.30RazorJackjustinu; I appreciate the work vonage network guys do right now
03:47.47justinuwell, they have a controlled environment
03:47.56RazorJackjustinu; exactly, was just gonna say
03:48.06justinufairly controlled, at least
03:48.11justinuvonage does work well tho
03:48.33RazorJackjustinu; yup it does..... Apparantly rackspace hosts some of their network
03:48.41justinuRazorJack: look into siproxd
03:48.48justinurackspace is awesome
03:48.51justinubucks tho
03:49.05RazorJackI work for a professional game development company.....
03:49.10RazorJackphoned rackspace for pricing
03:49.15RazorJackthey refused to talk to me
03:49.19mog_homelol
03:49.23justinuit's like 800 bucks a month
03:49.23asterboylol
03:49.25RazorJackbecause they thought I was a gamer
03:49.30harryvvheh
03:49.30mog_homethats sad
03:49.30RazorJackit was humerous
03:49.48harryvvyet games exceed hollywood in revenue
03:50.02RazorJackwell, its sad for them.... we were slashdotted after we got our new host
03:50.21*** join/#asterisk graphyx_home (n=mike@c-67-169-246-4.hsd1.ut.comcast.net)
03:50.28RazorJackcould have had a little rackspace btn on the bottom of the page
03:50.29RazorJack:)
03:50.30justinuwhat site?
03:50.41graphyx_homeWhat is the default folder for where asterisk puts the default recorded sounds?
03:50.49justinu<PROTECTED>
03:50.53graphyx_homety
03:51.29RazorJackthis was the slashdotted page -> http://www.legendsthegame.net/
03:51.31pandectRazorJack: after I write that in, do I need to reload asterisk
03:51.32*** join/#asterisk bmg505 (n=leon@rndf-146-15-15.telkomadsl.co.za)
03:51.38justinuah
03:51.40justinucool
03:51.47RazorJackpandect; reload now... right justinu?
03:51.58RazorJackpandect; or just amportal stop, amportal start
03:51.59graphyx_homeso if the file is in the digits directory do I have to call Playback("digits/2")?
03:52.08pandectRazorJack: Sounds good...
03:52.34graphyx_homenvm.  got it.
03:52.51*** join/#asterisk CpuID (n=nathan@dsl-202-173-176-82.qld.westnet.com.au)
03:53.30CpuIDhey ppls, quick question, could i use like exten => _X.,1,Dial(...) as a rule for all outbound dialled numbers to be sent to there?
03:53.44RazorJackjustinu; anyway, thanks for the help....
03:54.04CpuIDeg. if an analog handset was plugged into an FXS port, it would be just like dialling out their landline as normal
03:54.10CpuIDcomplete passthrough
03:56.11graphyx_homeIf you are using one asterisk server as a gateway for another one, what channel type would be best to have the two talk with?
03:56.23CpuIDgraphyx_home, iax2 probly
03:56.28graphyx_homeok.
03:56.40CpuIDinter-asterisk exchange from memory it stands for
03:56.44graphyx_homeCan that stream multiple calls over the same channel or would it just be doing one at a time?
03:56.54CpuIDyou can use iax trunking for that
03:56.59graphyx_homeOk.
03:57.00graphyx_homeThanks.
03:57.29CpuIDdo some research on it, trunking allows asterisk to drop the overhead when multiple calls are going between 2 installations
03:57.31LostFrogDoes drinking and * mix?
03:57.39CpuIDlol no LostFrog :)
03:57.46LostFrogDamn..
03:58.00LostFrogAnyone want a Yuengling?
03:58.06dippoi do :(
03:58.34LostFrogWhere be you, dippo?
03:58.35docelm0anyone know what Damin's number?
03:58.38LostFrogI have a 1/2 keg.
03:58.41*** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net)
03:58.48LostFrogI'm aferaid it won't all get drunk.
04:02.10hhoffmanwondering if someone could take a look at: http://pastebin.ca/28955 . I'm still battleing with the problem of getting a unknown context. If I put the extension directly into [incoming] then everything works
04:02.44LostFrogI'm sorry, my drunkness is off-topic.
04:03.43Pikorowhen running mc from within a screen window the F keys don't work properly.  anyone know of a quick workaround for that?
04:04.12*** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn)
04:04.33LostFrogAren't there M-codes equivalents?
04:04.40LostFrogYou would have to read the man pages.
04:04.46LostFrogAlt- = Esc
04:04.49LostFrogin screen
04:05.45Pikorohmm
04:05.51Pikoroi'll czech
04:05.59LostFroglol.. czech.
04:08.12graphyx_homedoes SET(COUNT=COUNT+1) work?
04:08.36konfuzedgraphyx_home: only in a loop
04:08.51graphyx_homeok.
04:09.17xbmodder_lappywhat is the "s" variable in the dial pan?
04:10.57*** join/#asterisk marc324 (n=marc3234@206-248-157-92.dsl.teksavvy.com)
04:11.02graphyx_homekonfuzed: That didn't increment it in a while section.  What would be another way of incrementing a variable?
04:11.28konfuzedgraphyx_home: perhaps your syntax is off
04:11.48asteriskmonkeyRazorJack i do rack hosting :D
04:11.49marc324ne1 can help with reading voicemail 'message0000.txt' file ?
04:11.56graphyx_homeexten => 9998,9,Set(COUNT=COUNT+1);
04:11.57konfuzedxbmodder: only n00bs that didnt rtfm ask that question
04:12.26graphyx_homeshould I put it as count = ${count} + 1?
04:12.40konfuzedgraphyx_home: how do you count already has a value
04:12.50konfuzedhow do yoou know
04:12.51asteriskmonkeymarc324 : ssh in and go cat message0000.txt
04:12.57graphyx_homekonfuzed: I set it already to one.
04:13.06konfuzedgraphyx_home: good start
04:13.25konfuzedxbmodder: the s stands for start
04:13.44konfuzedbut thats meaning less with out rtfm
04:15.42konfuzedgraphyx_home: perhaps Set(count = (${count} + 1))  but try them
04:16.28konfuzedtoo many different uses of syntax for me to keep track of
04:16.54graphyx_homekonfuzed: I tried that and it didn't work either.  tried it with [ ${count} + 1 ] and still didn't work.
04:17.03graphyx_homeSo I'll keep plugging along.
04:17.14graphyx_homeIt isn't critical  Just learning all the options to do in the dial plan.
04:17.17graphyx_homeknow what I mea.
04:17.17konfuzedwell I dont do any programming my self but
04:17.37konfuzedthere has to be some consistent examples of syntax for incrementing a variable
04:17.55konfuzedsome where
04:19.31*** join/#asterisk DrDeke (n=rdekema@dsl093-002-245.det1.dsl.speakeasy.net)
04:19.43DrDekeso what's up with asterisk 1.2.0?
04:19.55DrDekeeastern us time only has 41 more minutes left todya...
04:20.22graphyx_homeMove to California and things would be different... ;)
04:20.26DrDekethat is true
04:20.34DrDekeor better yet australia
04:20.42graphyx_homeNow you're talking.
04:20.54DrDekeerr wait, thats the other way aroudn isnt it
04:20.57LostFrogDrDeke: I said before, I didn't believe release today.
04:21.03DrDekeyeah it looks like you were right :)
04:22.00*** join/#asterisk fiftyCal (n=b@69-160-145-156.ontrca.adelphia.net)
04:22.30LostFrogRC2 was too close.
04:22.37LostFrogNeed to stabilize.
04:22.40DrDekeyeah
04:22.51DrDekeoh well, either way i'm just happy that someone told me how to fix my pulse dialling problem
04:22.53LostFrogTalking in short phrase.
04:22.56LostFrogMust be drunk
04:23.01LostFrogduh]
04:23.09DrDekenow as long as there are no more kernel panics, i should be stable and good to go
04:23.09DrDekehaha
04:23.34marc324ne1 familiar with voicemail?
04:23.41Damintwisted: You around?
04:23.41shido6yesh
04:23.54wasimmarc324: i get some from people at times
04:23.56shido6whats wrong wiff yer vercemail?
04:24.10fiftyCalI'm trying to remove the realtime modules from asterisk (I compiled 1.2, and now am trying to go back to 1.0).  Where are the realtime modules installed?
04:24.38shido6remember to checkout the "/include" dirs for lingering asterisk code
04:24.44shido6and whipe them out
04:24.46shido6all of them...
04:25.01fiftyCalshido6 is that for me?
04:25.18shido6yesszzzir!
04:25.45fiftyCalthanks
04:26.32bsdfreaksuckers
04:26.58fiftyCalahh..  /etc/asterisk.conf tells me /usr/lib/modules/asterisk instead of /var/lib....
04:27.37fiftyCalshould I just rm -rf the whole lot, make clean && make install to get only the modules I need?
04:27.44wasimfiftyCal: yep
04:27.49fiftyCalgracias
04:28.33shido6no
04:29.05shido6because you STILL need to remove the other /include dir
04:29.14harryvvman telus is being nasty and really giving local voip companies a run for there money by dropping there rates.
04:30.05shido6<PROTECTED>
04:30.16wasimharryvv: here as well, rates here have quartered since last year
04:30.17Qwellharryvv: That's a good thing, right?
04:30.33DrDekei guess telus is trying to tellus to stick with them *bow bow bow*
04:32.33harryvvqwell, its not good if you want to jump into the voip retail market.
04:32.45mog_homeqwell!!!!!
04:34.18marc324is it possible  to store content of message0000.txt in a db table?
04:34.37infinity1anyone know why the park button doesnt work on the xten client?
04:37.15LostFrogAnyone know why I have to use Dial,t and DTMF #700 with snom320/360s?
04:37.25LostFrogFor call parking?
04:37.51LostFrogWhy can't I have destination 700?
04:40.17LostFrogAnyways.. night all.
04:41.19*** topic/#asterisk by drumkilla -> Asterisk 1.2.0 has been released! -- ftp.digium.com || http://www.asterisk.org
04:43.06IronHelixsw33t!!!
04:43.25SkramXoo lala
04:44.45Dr_Rayis 1.2 now standard?
04:44.57mog_homeyes
04:45.30*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal)
04:47.52mog_homenow go report to slashdot....
04:48.05drumkillamog_home: wait for the notices to make it out
04:48.16mog_homeyeaha i guess...
04:48.50*** join/#asterisk GXTi (n=omgwtfbb@freenode/developer/GXTi)
04:49.26*** join/#asterisk Moc- (n=mochouin@modemcable181.215-82-70.mc.videotron.ca)
04:49.29*** part/#asterisk Moc- (n=mochouin@modemcable181.215-82-70.mc.videotron.ca)
04:49.45xbmodder_lappyWhat are some good flat rate providers (cheap)
04:50.11hypa7iadrumkilla: post it to slashdot :)
04:50.11DrDekeflat rate to where?
04:50.34*** join/#asterisk diclophis (n=diclophi@199.35.96.52)
04:50.40DrDekeha ha ha ha! ho ho ho ho! asterisk 1.2.0 is definitely in the house!
04:50.41DrDekew00
04:50.51drumkillahypa7ia: nooo, not yet - let the press release and the notice email go out
04:50.54diclophishello
04:51.07hypa7iahehe okay :)
04:51.10diclophisso... i have a  question
04:51.17drumkilladiclophis: HELLO!
04:51.23DrDekeYou've got questions... We've got blank stares!
04:51.30diclophishaha nic
04:51.31diclophise
04:51.37xbmodder_lappyDrDeke, my city
04:51.44xbmodder_lappypreferably IAX2
04:51.53diclophiswell anyhow, through AGI i am running a Background command right
04:51.57diclophisand the sound plays fine
04:52.08DrDekewhat city are you in?
04:52.17xbmodder_lappysilicon valley
04:52.19xbmodder_lappyamerica
04:52.25DrDekeoh
04:52.25DrDekewell
04:52.27diclophismy problem is when the caller presses a key to cancel the playback
04:52.28DrDekethere's broadvoice (sip)
04:52.34diclophisit discards that digit
04:53.00DrDeketeliax.com is iax but only pseudo-unlimited
04:53.05DrDekecheap tho
04:55.11diclophisis there a way for it to not discard the first digit?
04:56.07*** join/#asterisk IcePenguin (n=G4Master@207-118-182-184.dyn.centurytel.net)
04:56.27IcePenguinHi asll
04:56.31IcePenguin*all
04:56.46diclophishello
04:56.57Qwelldrumkilla: YAY!!!
04:57.02IcePenguinIs anyone willing to give a newbie a little advice?
04:57.18IronHelixsure
04:57.20IronHelixwhats up
04:57.35drumkillaw00t
04:57.35IcePenguinI'm looking at putting together an asterisk system for my church
04:57.37IronHelixit will cost you though.... do you need ALL 10 fingers?
04:57.52xbmodder_lappyI hate sip.
04:57.52IcePenguinIron: sadly, yes ;)
04:58.17IronHelixhehe
04:58.32IcePenguinIron: I'm looking at putting together an Asterisk system for my church
04:58.34IronHelixsure
04:58.48mog_homeyay
04:59.01IcePenguinOn voipsupply.com, I've selected a digium card that will work with what we need
04:59.06IronHelixwhich card?
04:59.16IcePenguinTDM21B
04:59.20IcePenguin2FXS 1FXO
04:59.27IronHelix1 line, two extensions
04:59.29IronHelixsure
04:59.30IronHelixkeep going
04:59.49IcePenguini've noticed that the extensions are RJ-11 jacks
04:59.54QwellThey're rj45
05:00.00drumkillaQwell: not anymore
05:00.00IcePenguinthey are?
05:00.05Qwelldrumkilla: really?
05:00.12drumkillathey are RJ11 now, yes.
05:00.14Qwelloh
05:00.25Qwellwell, they were :p
05:00.25*** join/#asterisk dudes (n=dudes@12-215-32-62.client.mchsi.com)
05:00.30Math`Qwell heh
05:00.30drumkillaQwell: yep :)
05:00.38Math`why the change?
05:00.38IronHelixanyway whats your Q ice
05:00.40Qwelldrumkilla: How many people put POE to them?  heh
05:00.49drumkillalol ...
05:00.50QwellThere has to be a reason it was changed. ;]
05:00.52IcePenguinSo, how do I connect a digital phone (such as VOIP or Nortel or Cisco) to this card?
05:00.54QwellDon't lie to me now. :P
05:01.00IcePenguinthe digital phones use rj45's
05:01.04Math`uh
05:01.05IronHelixwell ice those are different things
05:01.05drumkillaIcePenguin: you don't
05:01.11Qwellanalog phones only
05:01.20Math`IcePenguin: those cards are for connecting traditional phones
05:01.21IronHelixa nortel digital phone will probably be proprietary and thus useless
05:01.31QwellIf you want a voip phone, you won't even need the fxs ports
05:01.36Math`if you want to connect an ip phone just plug it into the ethernet switch
05:01.37IcePenguinIron: I was just using Nortel as an example
05:01.38IronHelixvoip or cisco phone connects to your church's lan and connumicates with the server over its network card
05:01.38Qwellthose plug right into your switch
05:01.57drumkillaeveryone talk at once!
05:01.59IronHelixyou dont need a card for those, just plug the * box and the phones into a switch and setup sip.conf
05:02.01IronHelixhehehe
05:02.01Qwellman, I almost bricked my bosses 7970 the other day
05:02.12Qwellsomehow, it decided it wanted to do a factory reset...
05:02.24IcePenguinIron: if I get a card with 1 FXO, and then install 2 PCI Ethernet cards, can I achieve my goal?
05:02.30IronHelixwasting time
05:02.31IronHelixhangon
05:02.31QwellI have zero control over the dhcp servers, so I couldn't set the tftp server for it to grab the firmware
05:02.39IronHelixicepenguin- you need a digium card to connect to an ANALOG line or telephone
05:02.49IronHelixlike a standard wall mount phone
05:02.57IronHelixor a line from your telco that such a phone would connect to
05:03.00QwellIcePenguin: if you get voip phones, just buy a switch or something
05:03.03Math`IcePenguin: just get a switch
05:03.05IcePenguinIron: I need to connect to a staqndard POTS telco
05:03.12IronHelixthat you need fxo for
05:03.13distortionqwell, you can overide that on the 7960's but not the 7970??
05:03.15IronHelixhowever
05:03.16IronHelixfor extensions
05:03.24IronHelixif you are using a SIP or IAX phone, you do not need a special interface card
05:03.31Qwelldistortion: no, you can...but not when you do a factory reset.  It won't load until it has firmware, which means no settings menu
05:03.43Qwellfactory reset purges the firmware
05:03.52IronHelixthe asterisk server, becomes part of your ethernet network, as do all of the IP phones
05:03.53distortionoh yeah- that bad eh
05:04.01IcePenguinOkay Iron I think I'm getting what you're saying
05:04.02IronHelixthey communicate with each other much like computers on the internet
05:04.06IcePenguin(Block diagram)
05:04.10IronHelixyou don't need a separate port for each one
05:04.43IronHelixtelco -- phone line -- digium fxo port --  asterisk -- ethernet -- switch -- any number of ip phones
05:04.50konfuzedIcePenguin: [Block DiAgrAM]
05:04.57IcePenguinsweeet
05:05.09IronHelixyou will need to start by running CAT5E or cat6 cable to each place where you want a phone
05:05.09IcePenguinThanks so much, Iron
05:05.12IronHelixyou may already have this
05:05.19IcePenguinyep
05:05.19IronHelixbecause many phone jacks are wired with cat5 anyway
05:05.21DaminSo.. 1.2.0 is released?
05:05.27IronHelixmake sure there is an rj45 end on each of them
05:05.31QwellDamin: nope, just a sick joke
05:05.35IronHelixthen they all come out in a wiring closet or something
05:05.40IronHelixbuy an ethernet switch
05:05.45IronHelixplug all the lines into that
05:05.48IronHelixas well as the * server
05:05.58IronHelixmake sure you have a router or something providing internet and DHCP
05:06.04Qwellwow...you know what would be kinda cool?
05:06.07DaminQwell: There is a 1.2.0 tar.gz file on ftp.digium.com
05:06.12konfuzedHybrid Helix Hub
05:06.13Qwell(and perhaps extremely sacreligious?)
05:06.23IronHelixthen use sip.conf to define each phone
05:06.26IronHelixand configure the phones
05:06.33IcePenguinIron: do I need web access on that server if I'm using telco POTS?
05:06.36Damin<PROTECTED>
05:06.40Qwelltake a church, right...  turn it into a wisp...put your directional APs into the cross. :D
05:06.48IronHelixuse extensions.conf to make a dialplan and control who can dial what
05:07.06QwellIcePenguin: you don't
05:07.09konfuzedchurch towers I great for community mesh points
05:07.09IronHelixicepenguin- you dont need internet per se... however you will need to distribute IP addresses somehow
05:07.20IcePenguinokay
05:07.24IronHelixyou can static assign each phone and the server Ip addresses
05:07.35IronHelixor you can set the linux server * runs on to be a dhcp server
05:07.43Qwelldrop dhcpd on the asterisk box, and just get a little switch...problem solved
05:07.45*** join/#asterisk implicit (n=implicit@216.13.124.132)
05:07.46IcePenguinYeah... I think I'll do that
05:08.04IcePenguinExcellent... thanks a bunch, IronHelix
05:08.10*** part/#asterisk graphyx_home (n=mike@c-67-169-246-4.hsd1.ut.comcast.net)
05:08.11IronHelixno problem
05:08.14IronHelixgood luck with your project!
05:08.20IcePenguinI might be back if the church decides to go through with it  :)
05:08.22IcePenguinthanks!
05:08.29IronHelixand you can keep your fingers....  for now
05:08.36IcePenguindouble thanks! ;)
05:08.40IronHelix:)
05:08.46IcePenguinbye, guys
05:09.13IronHelixhmmm
05:10.13IronHelixone of these days i might write a '15 minutes to understanding asterisk' document...
05:10.18IronHelixmight make things easier
05:10.24Qwell~thebook
05:10.26jbotfrom memory, thebook is Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online!
05:10.26Qwellit's been done
05:10.42IronHelixfrom what i've seen its pretty good, but its long
05:10.53IronHelixi mean something ~5pages long that gives you just hte basics in like 10 mins
05:11.14*** join/#asterisk bweschke (n=bweschke@65.162.120.25)
05:11.23*** join/#asterisk fiftyCal (n=b@69-160-145-156.ontrca.adelphia.net)
05:14.29*** join/#asterisk M-I-A (n=chaguy42@209.161.199.142)
05:20.10*** join/#asterisk IcePenguin (n=G4Master@207-118-182-184.dyn.centurytel.net)
05:20.19IcePenguinHi guys, back again :)
05:20.45mog_homeyay!
05:20.52mswhttp://www.rpath.org/project/asterisk/release?id=180
05:20.58mswhttp://www.rpath.org/project/asterisk/release?id=182
05:21.08IcePenguinQuick question: If there is CallerID on the incoming analog POTS, will * forward that info to the receiving VOIP phone extension?
05:21.41IronHelixit will
05:21.52IronHelixyou can also manipulate it however you want
05:21.58IcePenguinHow so?
05:22.10IronHelixfor example, if a user serves more than one department (sales, service,etc)
05:22.16*** join/#asterisk Astinus (i=iBook@freenode/staff/gentoo.astinus)
05:22.32IronHelixyou can change the caller id to have a number before it (1-theircallerid) or to change the name (SALES: theirname)
05:22.56marc324in * voicemail i get one .txt for every voicemail.  is it possible to change this?
05:23.09IronHelixthat way if you only have 2 or 3 employees they can answer with 'sales department, this is jack, how can i help you' and make it look like there is a dedicated department instead of just a few guys answering phones
05:23.10Qwellmarc324: You could use ODBC_STORAGE, and store your VM in a database
05:23.34IcePenguinWow, playing with * is gonna be fun! Thanks again, Iron!
05:23.39IronHelixit is :)
05:23.44IronHelixas a general rule
05:24.02IronHelixasterisk will take any type of call coming in on any interface, do whatever you want to it, and put it out again on any other kind of interface
05:24.20IcePenguinrock on! thanks, iron!
05:24.24IronHelixnp
05:24.26IronHelixhave fun
05:24.30marc324odbc_storage stores the actual voicemail recording in db.... instead of putting a link to it....
05:24.49Qwellmarc324: yes
05:24.57IcePenguinThanks again, iron! bye
05:25.21ikarusIronHelix: unlike old PBXs which will take a call coming in on a single interface, mangle the hell out of it, drop it unpredictably and don't leave a log, and with some luck put it out through the right interface
05:25.36IronHelixhehehe
05:25.52*** join/#asterisk Assid (n=assid@59.183.51.0)
05:25.55marc324no one actually uses odbcstorage.
05:26.10drumkillaplenty of people use it
05:26.12Qwellmarc324: I do
05:26.24Qwellit works great
05:26.27marc324how well does it scale?
05:26.37Qwellas well as anything else using a database
05:26.53Qwellif you know how to scale it, it can grow as large as you'd like
05:26.54DaminQwell: Asterisk 1.2.0 HAS been released..
05:27.02Assidis the firmware for pap2-as the same as pap2 and pap2-na?
05:27.03QwellDamin: You believed me? :p
05:27.23QwellDamin: sorry, I assumed you knew I was kidding
05:27.39DaminQwell: No.. I'm actually building it on the venerable old p133 as we speak!
05:27.43Qwellnice
05:28.02QwellI just got head...I figured it was close enough
05:28.09Vcoo,O
05:28.13Assidi read that they released a firmware with gsm support
05:28.24Assidbut i cant seem to find it
05:28.29ikarusIronHelix: back to something I asked a while back, I mailed Grandstream about if they could add an option in the firmware to disable doing flash by actually flashing the hook and only keep the button activated
05:28.39IronHelixdid they reply?
05:28.42bweschke#ifdef NOTANOTHERVMSTORAGECODEOPTIONINTHECODE!
05:28.45marc324i want to have the recording.wav on file.... not in blob....  email forward..
05:28.51bweschke#endif UGH
05:29.08ikarusIronHelix: no idea yet
05:29.20DaminNow I gotta build RPMS for 1.2.0
05:29.37IronHelixi've only mailed them once, they replied in about 2 days
05:29.47Vcoanyone have any bright idea about this one.......incoming call , goes to voicemail, when i have it go to Voicemail without options, it prompts to enter the mailbox to leave a messsage for....i enter the correspnding mailbox and asterisk segfaults...
05:29.53marc324with odbcstorage no recording file is created on the drive
05:30.12Vcoif i pass the call to say... Voicemail,u2  it gets to the mailbox just fine..
05:30.45ikarusIronHelix: no reply yet, I sure hope they do reply, because with the way my users currently use their phones it would be an issue
05:31.03IronHelixthey throw the handset in the general direction of the phone?
05:31.14Vcothats what i was going to ask
05:31.47bweschkeVco: is this with odbc storage or just the standard vm?
05:31.53Vcostandard
05:32.01ikarusIronHelix: they bash the hook to hang up on a PBX and then want to call out immediatly
05:32.16ikarusIronHelix: which the BudgeTone sees ofcourse as a flash
05:32.28IronHelixahh
05:32.29Vcoi had this pop up in 1.0.9, out of frusteration i moved to 1.2.0rc1 and it went away..
05:32.39bweschkeoh
05:32.46Vconot it appears to have returned in rc2 and 1.2.0
05:32.54bweschkehmmm...
05:33.04bweschkeSo you just do VoiceMail() and it works?
05:33.16bweschkeor rather - crashes?
05:33.21Vcoyea
05:33.28Vcosegfaults
05:33.35Vcono matter what mailbox
05:33.57ikarusHmmmm, won't see my cow-orker in 2 weeks, will need to engage in some vocal combat to decide if we keep the old answering machine on an ATA, or do as I want and use Asterisk for that
05:34.07Vcochecked for  even simple stuff like file permissions or something...everything is fine
05:34.33bweschkek
05:34.44bweschkeu have a backtrace on the crash?
05:35.48Vconot now...i had gone through it in the past and it wasn't showing anything usefull..
05:35.50bweschkevco: can I put in an invalid extension?
05:36.04bweschkevco: or it has to be a valid mbox?
05:36.44DaminI must go to sleep now!
05:37.07bweschkesleep? bah
05:37.12bweschkeoverrated
05:37.17bweschke:)
05:39.08Vcoit's just using mbox
05:43.12Flautohello, would anyone tell me how to dial an sip exten and use a macro to dial my cell phone in the same line in the dial plan
05:43.52wasimDial(Sip/Flauto&Zap/1/239471239048719234)
05:44.09Flautocan i use macro to do that?
05:44.12wasimyes
05:44.25Flautocan i use
05:44.25bweschkevco: i'm not crashing here when I try it
05:45.51Flautodial(sip/myexten&macro(callmycell,${mycell}))?
05:47.09Flautoor, it has to be dial(sip/blah&iax/blahlah)
05:49.21*** join/#asterisk froguz (n=froguz@212-43-112.adsl.terra.cl)
05:49.28froguzhi
05:49.30Vcohmm...
05:49.58Vco#0  0x40908d59 in leave_voicemail (chan=0x818d8b8, ext=0x0, options=0xbd7f9458) at strings.h:155
05:49.58Vco155     strings.h: No such file or directory.
05:49.58Vco<PROTECTED>
05:50.00Vcoerr..
05:50.24bweschkedo you have an /etc/asterisk/voicemail.conf ?
05:50.30Vcoyea
05:51.04bweschkeso what happens? it asks you for your extension and then it crashes?
05:51.07bweschkeor something after that?
05:51.10Vcothe thing is i can send it straight to the mailbox with Voicemail,u(somevalidmailbox)
05:51.11froguzsomebody can tell me if it's possible to make "call hold" using normal phones + an ATA??
05:51.42Vcoruns the "vm-whichmailbox" prompting to enter the maibox to leave a message for..
05:51.50*** part/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
05:51.59froguzi can't figure out hoe to implement such function within the extentions.conf
05:52.15bweschkeand then a crash?
05:52.23Vcoenter the mailbox, as soon as it gets the dialed digit, or i hit # to send, it segfaults
05:53.02hellopMy * recently decided to stop working.  I hear no sounds on phones when dialing out.  But they work fine when dialing between phones.
05:53.08hellopHow can I diagnose this?
05:53.26[TK]D-Fenderfroguz : yes, Sipura's support "hold" like that
05:53.48*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
05:53.53[TK]D-Fenderfroguz : its not in extensions.conf, its in your ATA's setup
05:54.01[TK]D-FenderWhich one do you have?
05:54.23froguz[TK]D-Fender, i'm not using IP phones, just normal phones with an ATA (gateway in proxy mode)
05:54.32[TK]D-FenderWhat model of ATA?
05:54.51froguzmicronet 5004 or something like that
05:55.03froguzFXS gateway with 2 phone lines
05:56.02[TK]D-Fenderfroguz : well I can't tell you what they support but other ATA's have their own "dialplan".  This is often to support both their internal features, and to control what/when/how to route calls to SIP servers they are registered to.
05:56.19[TK]D-FenderBut its not part of *'s job for that
05:57.03bweschkevco - i'm not crashing here - but I can see how you are.. congrats.. you've just found the 1st "crash" bug of 1.2
05:57.24froguzbut i wish to implement a fuction that allow me to mute the mic pressing for example *1 while a conversation is in progress
05:57.41Vcono, i think it's something else here..
05:57.52froguzis that possible?
05:57.58Vcoas to WHAT or WHERE to even look is whats annoying me
05:58.01*** join/#asterisk oej (n=Olle@apollo.webway.se)
05:58.14Qwelloej: morning
05:58.16Qwellor something
05:58.22bweschkevco - no ... you're spot on
05:58.27bweschkeit's a guaranteed segfault
05:58.28Vcoi just don't understand how it could work fine before and now it has reappeared again on a static box
05:58.41bweschkewell - I'm not sure ....
05:59.01oejQwell: Morning!
05:59.02bweschkethe code in app_voicemail is .... ummmm... how shall we say .... complex.
05:59.03[TK]D-Fenderfroguz : good odds your ATA supports hold, you'd have to read up on it, but in case I might mut them on speakerphone and mute that.
06:00.00hellopWhy suddenly after like 8 months, I now have to reset my zaptel modules periodically?  Maybe the card just died...
06:00.19[TK]D-Fenderok, WAY too late for me to hang around, I'm off to collapse....
06:00.33froguzit seems i'll have to upgrade my ATA's software
06:00.51froguzthanks [TK]D-Fender
06:00.56froguzhave a good rest
06:01.22[TK]D-Fendernp, another happy customer... *yawn*
06:03.16marc324ne1 here has parsed the 'message.txt' file for voicemail/
06:04.19*** part/#asterisk diclophis (n=diclophi@199.35.96.52)
06:10.04infinity1can someone look at my sip debug? http://pastebin.ca/28984  for the life of me i can't get a gizmo call into asterisk. I can call from asterisk to gizmo though.
06:10.51IronHelixLooking for s in default (domain 208.184.76.77)
06:10.51IronHelixReliably Transmitting (no NAT) to 198.65.166.131:5060:
06:10.51IronHelixSIP/2.0 404 Not Found
06:10.54IronHelixis your problem
06:10.56xbmodder_lappyack.
06:11.00xbmodder_lappyouch man
06:11.08infinity1whoa shit. what?
06:11.15IronHelixcheck your context for the gizmo sip.conf entry
06:11.19infinity1whas is S default domain.
06:11.22infinity1hmm. ok.
06:11.43IronHelixalso when you register
06:11.54IronHelixyou can specify where it goes
06:12.35IronHelixat the end of the register statement is /blahblah
06:12.47IronHelixthe blah blah is what extension in the context specified it should go to
06:13.11infinity1um. i have context=outside-iax
06:13.25infinity1and exten => _X.,1,Macro(cidfix-ael)
06:13.49infinity1do i need to specify the phone # instead of _X. ?
06:14.11infinity1oh. i need s,1,macro(...
06:14.16IronHelixpastebin your sip.conf minus passwords
06:14.48xbmodder_lappyanyone here have a teliax account?
06:14.54xbmodder_lappyor work for teliax?
06:15.25infinity1k.
06:18.04infinity1http://pastebin.ca/28987
06:19.22IronHelixah
06:19.23IronHelixgot it
06:19.36IronHelixthere is nothing linking your register statement to the [gizmo] config block
06:19.48IronHelixtry 17476226092:xxxx@gizmo
06:19.53IronHelixeh
06:20.00IronHelixregister => 17476226092:xxxx@gizmo
06:20.08IronHelixit should pull the host and stuff from the config block below
06:21.22infinity1k
06:21.59pandectI used AMP to set up a "Digital Receptionist", but I can't find any settings on where to forward the user when they don't enter anything... any ideas?
06:22.16infinity1er ..that didn't do it.
06:22.24Qwellpandect: suggestion - don't use amp
06:22.29bweschkevco - u still around?
06:22.46infinity1Nov 16 22:22:41 WARNING[21868]: chan_sip.c:1966 create_addr: No such host: gizmo
06:23.14IronHelixah
06:23.23bweschketry the patch at: http://bugs.digium.com/view.php?id=5774   that should fix it for you
06:23.24IronHelixthen change [gizmo] to [proxy-01.sipphone.com]
06:23.36IronHelixand make a copy of it so you can still use dial/gizmo/exten
06:24.04bweschkethank goodness for bugs ... .what else would I do while waiting for my delayed flight? oh.. that's right doxygen docs!
06:24.16pandectQwell: Really? that bad :-/
06:24.22Qwellbweschke: doxygen docs is for on flights
06:24.34*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
06:24.38IronHelixonce you learn the config files, gui's like amp are more of a hobbling thing
06:24.39bweschkeya - well... they've gotta start boarding before I get that far yet
06:24.54Qwellpandect: yep
06:24.57xbmodder_lappyHow long do transactions on credit cards take to show up?
06:25.06Qwellxbmodder: 10 seconds, or 6 weeks
06:25.13Qwellanywhere in between is normal
06:25.17xbmodder_lappygret.
06:25.19xbmodder_lappygreat
06:25.38IronHelixdepends on the way they process cards
06:25.38Qwellsome companies wait until the item ships, and is delivered, before they'll charge the card
06:25.42xbmodder_lappyanyone here have a teliax account or a one time use credit card.
06:26.01Qwellusually credit cards take a day or two though
06:26.08Qwelldebit cards are almost always instant
06:26.14IronHelixit happens in two parts, AUTH and CAPTURE.  the auth authorizes the company to charge you that much, and once an AUTH goes through, the company is guaranteed access to that money until the AUTH expires
06:26.20IronHelixthat can be anywhere from 6 hours to 4 days
06:26.31IronHelixif you use a debit card, an AUTH reduces your available balance
06:26.43IronHelixif you use a credit card, AUTH depletes your available credit
06:26.44infinity1IronHelix: wait. if i cahnge [gizmo] to proxy01.sipphone.com then register => 17476226092:xxxx@gizmo doesn't have a reference
06:27.02xbmodder_lappyIronHelix, I got the AUTH
06:27.02Vcoyea..stilll around
06:27.05QwellYou use hostname in the register
06:27.12xbmodder_lappyI see a little confirmation letter
06:27.15QwellI don't think you can use the peer name in the register
06:27.23IronHelixthe second part is the CAPTURE, where money is actually yoink'd from your acct and put in theirs.  THIS is what shows up on the bill.  Most companies do this for all pending transactions once a day
06:27.25IronHelixthus the delay
06:27.33IronHelixinfinity- change the register back to what it was before
06:28.25Vcodammit
06:28.28*** join/#asterisk heath__ (n=root@12-215-32-62.client.mchsi.com)
06:28.31bweschkewhat?
06:28.47infinity1so now i have the original register line with [proxy01.sipphone.com] in the sip.conf. but the results are the same.
06:28.48Vco*CLI> is much less usefull to type to when it's the frickin' log file
06:29.02infinity1no errors on the CLI, but it doesn't work.
06:29.12IronHelixdo sip show registry
06:29.17IronHelixare you registered?
06:29.29infinity1proxy01.sipphone.com:5060       17476226092        105 Registered
06:29.33heath__anyone got a sip gateway w/ t38 i could connect to to test the new spandsp stuff?
06:29.37*** join/#asterisk YoMama (n=tchen@pcp02689850pcs.roylok01.mi.comcast.net)
06:29.48bweschkeall right gang... flight is getting rdy to board... see you in the morning
06:29.53IronHelixnow turn off sip debug, do set verbose 4 and call
06:29.56YoMamawhere ya flyin' to?
06:29.59IronHelixpastebin what you get
06:30.04Qwellbweschke: What, no 'net in the air? :)
06:30.04YoMamahey Iron
06:30.08IronHelixsup yomama
06:30.13bweschkehome from the IP4IT show - las vegas to NJ
06:30.16YoMamaQwell: sure there is...at like $5/minute
06:30.18Pikorowow.. something else to drive me crazy
06:30.20bweschkeheehee
06:30.27Pikoronow chan_zap.so won't load
06:30.29Pikorowtf
06:30.33YoMamabweschke: next time yer in LV..i know a girl that works at Club Paradise..hahaha
06:30.37infinity1the cli shows nothing
06:30.39Pikoroworked fine 5 minutes ago... rebooted... and now nothing
06:30.51Dr_RayI've eaten at that taco bell in the LV airport many a time
06:30.52QwellPikoro: modules loaded?
06:30.55bweschkehaha.. nice
06:30.58YoMamashe'll hit ya extra hard in the face with her funbags
06:30.58Pikoroyah
06:30.59*** join/#asterisk svenna_ (n=svenna@p548D289C.dip0.t-ipconnect.de)
06:31.06QwellPikoro: You sure? ;]
06:31.15YoMamabweschke: fly safe
06:31.24IronHelixhmmm, add /17476226092 to the end of the register statement
06:31.26IronHelixtry it again
06:31.35bweschkewill do. thx. later all
06:31.44IronHelixso its register => user:pass@proxy01.sipphone.com/17476226092
06:31.50Pikoroyah...
06:31.54Pikorolet me try something
06:31.55Pikorobrb
06:32.01IronHelixnow add to your default context and from-iax context an extensions 17476226092
06:32.10YoMamaIronHelix: yup..then it'll send incoming calls to that extension
06:32.33YoMamaand make sure sip.conf has type=friend too
06:32.39Pikorohmm.. how did my fxo get changed to fxs
06:32.40IronHelixwhich would be an improvement cuz they are going nowhere atm
06:32.53infinity1restarted, did set verbose 4 (was 3) and retried the call. nothing on the cli.
06:32.53IronHelixpikoro, remember that it should be defined as the opposite of what it is
06:33.03Pikoroyah, it was before
06:33.06Pikorofixed it now
06:33.36YoMamaanyone know a good place to get TDM400 cards used?
06:33.41Qwellebay
06:33.43IronHelixahh found another problem
06:33.54IronHelixinfinity- it should be disallow=all then allow=ulaw
06:33.56YoMamaQwell: they're in hot demand...
06:34.06IronHelixif you allow, then disallow=all you will have no available codecs
06:34.15YoMamadoh
06:34.21YoMamano codecs...what is one to do?!?!?!
06:34.37YoMamadoes fax work over GSM?
06:34.41IronHelixNO
06:34.41Qwellhell no
06:34.44YoMamahaha..k
06:34.48infinity1IronHelix: fixed that. nothing new.
06:34.51YoMamaonly over G.711?
06:34.57Qwellif that
06:35.10IronHelixpastebin a sip debug as well as what configs you ahve so far
06:35.27YoMamahmm
06:35.35infinity1k
06:35.51YoMamathose TDM cards are only 3.3V PCI tho right?
06:35.59IronHelixTDMxxB is 5v
06:36.02IronHelixas i recall
06:36.12YoMamai have an old P2...i bet it's got old PCI slots in it..i opened it up and there's freakin' ISA slots in it
06:36.16YoMamaso it's some old crap
06:36.19IronHelixits the multiple t1 cards that are 3.3v and there are 5v versions of all fo them
06:36.39YoMamawhich one came first...5v or 3.3v?  it's 5 isn't it?
06:36.40IronHelixhttp://www.digium.com/index.php?menu=whatpcislot
06:36.47IronHelix~pcislots
06:36.56IronHelix~pci
06:36.57jbotpci is, like, (Peripheral Connect Interface) This interface was designed to supplant the VL-Bus architecture and provide a standard slot with a reduced size for high-speed peripherals. It normally runs at 33 MHz on a PC, but can run faster. PCI slots are 32-bit slots, like EISA or VL-BUS, but in a compact form factor. or way faster than agp coz the little slot is ...
06:37.23Vcothis is f'ing retarded
06:37.36IronHelixjbot, pcislots is A good explanation of which types of PCI slots look like what is available at http://www.digium.com/index.php?menu=whatpcislot
06:37.37jbotokay, IronHelix
06:38.12Pikorook, this is wierd... asterisk says it's picking up zap/1-1
06:38.30Pikorobut the phone just keeps ringing on the client end instead of passing it off to the menu
06:38.38Pikoroeven though asterisk says it's playing the menu
06:38.55infinity1IronHelix: http://pastebin.ca/28990
06:39.24Vcois that Zap or from a DID?
06:39.34YoMamaIronHelix: i doubt my system is PCI 2.2 compliant
06:39.41Vcoerr. Pikoro
06:39.42YoMamawhich means if i want some working FXS ports..my best bet is an ATA
06:39.43Pikorozap i believe
06:39.46infinity1IronHelix: sip.conf http://pastebin.ca/28991
06:40.09Pikoroi see incoming call, answer, play(), etc...
06:40.26Pikorobut my cellphone just keeps ringing like nobody picked up
06:40.33IronHelixchecking, infinity
06:41.52IronHelixinfinity- line 40 Found no matching peer or user for '198.65.166.131:5060'
06:42.08IronHelixthen line 54
06:42.42IronHelixin sip.conf set in [general] context=outside-iax
06:43.19IronHelixor in extensions.conf in [default] put exten => 17476226092,1,Goto(outside-iax,s,1)
06:43.28QwellNO, bad, bad, bad
06:43.35QwellDon't put outgoing routes in default
06:43.54Qwelldon't put any routes in default, for that matter...not unless you like anonymous callers being able to use those routes
06:44.01YoMamaso what kinda box do u need to be running four T1s into one machine?
06:44.01infinity1IronHelix: it worked. wholy shit.
06:44.03IronHelixits an incoming route
06:44.05IronHelixnot outgoing
06:44.23QwellYoMama: something a bit burly, depending on where the T's are going
06:44.26IronHelixoutside-iax is his incoming context
06:44.27infinity1IronHelix: how come it ignores contect=?
06:44.38YoMamaQwell: well, let's say they're all PRIs
06:44.46IronHelixahh
06:44.49QwellYoMama: yeah, but what are you doing with the calls?
06:44.51IronHelixalso set it to type=friend
06:44.54Pikorogrr
06:44.58IronHelixthat may also help and remove the requirement for the above
06:45.02IronHelixbut either way it works right?
06:45.08YoMamaQwell: eh...calling people using a mixture of G.711 and G729
06:45.17infinity1IronHelix: yea. i'm just trying to learn so i can figure this out next time :)
06:45.26YoMama24 x 4 = 96 calsl
06:45.27YoMamacalls
06:45.27QwellYoMama: You'll probably need a pretty burly box then, if you're gonna transcode
06:45.34YoMamacan one box handle 96 calls?
06:45.36Qwellespecially to g729
06:45.37Qwellsure
06:45.47Qwellone box could handle hundreds of calls
06:45.47YoMamalike dual or quad proc?
06:45.50Qwellnah
06:45.51IronHelix96 calls yes, 96 calls with transcoding... if its a big box
06:45.58Qwelldual tops
06:46.11YoMamawhere do u set up transcoding in asterisk?
06:46.17QwellYoMama: it's all automatic
06:46.23YoMamashit...cisco makes u pay big buckaroos for transcoding
06:46.33YoMamaand if u do it on the hardware...it eats a lot of juice
06:46.35Qwellif a g711 device needs to connect to a g729 device, it'll "Just Work"
06:46.38infinity1IronHelix: so whats the deal with context= and type=user? that is not allowed?
06:46.44QwellYoMama: the latter is still true with asterisk
06:46.53IronHelixtype=friend is one who makes and recieves calls
06:46.55Qwelltranscoding can be a bit of work for a box
06:47.00YoMamaQwell: yeah..transcoding is nasty bsiness
06:47.07IronHelixthat may be why it wasnt matching correctly
06:47.15infinity1IronHelix: user is just receive calls
06:47.36IronHelixeither way, sipphone should be friend
06:47.57YoMamais there a page anywhere that describes what this "realtime" is?
06:48.13infinity1k.
06:48.30*** join/#asterisk froguz (n=froguz@212-43-112.adsl.terra.cl)
06:48.32infinity1IronHelix: i'm going to save all this an clean it up.
06:48.32IronHelixrealtime is configuring asterisk from a database in such a way that the db is queried as calls go through, so changes to the database take effect in 'realtime'
06:48.38IronHelix~realtime
06:48.39jbotrumour has it, realtime is http://www.voip-info.org/wiki-Asterisk+RealTime
06:48.48infinity1IronHelix: thanks a ton!
06:48.51IronHelixnp
06:48.52IronHelixgood luck!
06:49.20YoMamaIronHelix: which is good for providers 'cause they can make changes on the fly without having to reload asterisk
06:49.43IronHelixexactly
06:49.46YoMamaslick
06:50.03YoMamaanyone used the Grandstream ATAs?  they seem like the most economical
06:50.12IronHelixalso a sql database is much easier to manage if you have 1000's of users than huge .conf files
06:50.13QwellYoMama: there is a reason for that
06:50.31YoMamaalthough if i'm gonna blow $60 on one..i should just spend $85 for the grandstream GX-2000 phone
06:50.40IronHelixget the phone
06:50.50QwellYoMama: if you're gonna spend $85, you might as well spend $120
06:50.51IronHelixand more than one agent or administrator can edit a database at a time, whereas if two people open a conf file, whoever saves it second will overwrite the changes of the first
06:50.58YoMamaQwell: on what?
06:51.07Qwellsomething that isn't complete crap. :)
06:51.16IronHelixthe gxp isnt COMPLETE crap...
06:51.18YoMamaQwell: I heard the GX-2000 ain't bad
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06:51.30QwellIronHelix: but, lowercase complete
06:51.41froguzit's possible to execute asterisk commands (Wait(2) for example) during a conversation by pressing a key ???
06:51.45YoMamaok..then what's not gonna bust my bank and isn't a piece
06:52.06Pikorocan someone look at my extensions.conf and tell me if something in here might be causeing my problem
06:52.11QwellYoMama: I hear the spa841 is decent
06:52.20QwellI've never seen or used one though, so...
06:52.27IronHelixi say go with the gxp...  its pretty easy to setup, not quite as tweakable as teh 841 but it has a much better backlit display
06:52.33IronHelixthe 841 is a SMALL phone
06:52.33Qwellgotta be better than the gs
06:52.41YoMamaI heard that the SIP images for the Cisco phones are crap..that true?  I might be able ot get my hands on one for free..which is why i ask
06:52.42{zombie}841 has lousy shielding
06:52.50{zombie}you do NOT want to use a mobile (cellular) phone in the same building as oen
06:52.51{zombie}one
06:52.59infinity1IronHelix: making type=friend removed the requirement to have the general context=outside-iax ...yer right
06:53.00Pikorohttp://pastebin.ca/28992
06:53.07{zombie}I'm hoping they fixed that in the 941
06:53.10IronHelixcool
06:53.14QwellYoMama: sip images are alright.  I personally am digging sccp
06:53.18Pikoroproblem is, the phone rings, and asterisk says it picks up, but it never does
06:53.35YoMamaQwell: so u just use the default skinny client on the cisco phone eh?
06:53.45YoMamaQwell: hmm...that'd be nice 'cause then i don't even have to flash it
06:53.46QwellYoMama: no, I upgrade the sccp firmware
06:54.06IronHelixi have a grandstream bt100...  thing is noisy as hell.  it kills all FM radio in the same room, turn on a radio and you hear patterns of noise as the LEDs blink
06:54.13YoMamaQwell: ah..and the advantage to SCCP?
06:54.15IronHelixthe gxp is better tho
06:54.22QwellYoMama: cleaner, heh
06:54.22{zombie}yeah gxp is pretty good
06:54.24Pikorogxp seems to be pretty good
06:54.26Pikoroi got 10 of em
06:54.33YoMamaPikoro: good to know
06:54.36YoMamaimight order one tomorrow then
06:54.37QwellSIP is ugly...really robust...too robust perhaps
06:54.41{zombie}and the latest beta (alpha?) gxp firmware supports the busy led feature
06:54.47{zombie}like the Snoms
06:54.53{zombie}very handy
06:54.56YoMamabusy leds?
06:55.00IronHelixyomama- grab a gxp, they are useful, reliable, and with firmware .12 and later the speaker phone doesnt totally suck
06:55.06{zombie}yeah, so you can see when other people are on the phone
06:55.10YoMamaoooh
06:55.13IronHelixlights next to speed dials light up when ppl are on the phone
06:55.13{zombie}because their speed dial LED lights up
06:55.18IronHelixits not perfect
06:55.19YoMamaso u could use the GXP as a receptionist phone
06:55.21froguzi'll put it on a different way
06:55.21froguzfroguz  it's possible to make call hold usig a phone without hold key? (X-lite for example)
06:55.27{zombie}YoMama: yup, I wouldn't yet though
06:55.37{zombie}few too many nasty bugs in this firmware
06:55.39IronHelixi'd wiat for firmware .14 for that
06:55.41Qwellget a 7914 on a 7960 and use sccp ;]
06:55.46YoMama{zombie}: well, that'd be sweet...because then u could actually go and start replacing traiditional PBXs
06:55.47{zombie}wait for it to be non-beta
06:55.50Pikoroco can someone czech my extensions.conf paste and see if it looks ok?
06:55.52IronHelix.12 is pretty solid tho
06:55.55YoMamawhat about intercom...is there anyway to do that thru SIP phones?
06:55.56{zombie}YoMama: if you want to do that now, get a Snom
06:56.00{zombie}they do it already
06:56.02{zombie}and work great
06:56.15YoMamaSnom = more expensive
06:56.17QwellYoMama: most phones have an autoanswer thing
06:56.18IronHelixyomama- yup, grandstream supports it with a hack as of .12 on the gxp and supports it correctly as of .13
06:56.22YoMamai'm unemployed right now..can't be buying shit left and right
06:56.34{zombie}yeah but you were talking about replacing receptionist phones
06:56.36IronHelixcorrectly = you set a sip header in the dialplan
06:56.49{zombie}for home/small office use the GXP is great
06:56.57{zombie}if you want a proper receptionist phone get a Snom
06:57.03YoMama{zombie}: cool
06:57.17IronHelixotoh for receptionists... the new Polycoms look SEXY
06:57.17Qwell7914 + hints = <3
06:57.29YoMamathe budgetphones from grandstream are crap tho?
06:57.30QwellI want to talk my boss into getting me one
06:57.34QwellYoMama: yes
06:57.37YoMamathey look like they were made by a kid
06:57.44{zombie}I should get a polycom for eval
06:57.47{zombie}haven't seen one in the flesh yet
06:57.53QwellYoMama: You'll often hear them referred to as "barbietones"
06:57.55IronHelixyomama- pretty much, they look like they were designed by toddlers for toddlers...  they are fun to play with tho
06:58.17YoMamareminds me of those old conair phones u can buy at target for like $8...big huge buttons
06:58.24IronHelixalot of ppl here (including myself) learned about voip on the 'barbietel' cuz at the time they were the only non-$300 ip phones you could get
06:58.37Qwellconair?  same people who make blow dryers?
06:58.41YoMamauh huh
06:58.48YoMamathey used to make handsets
06:58.51Qwellscary
06:58.53Pikoroand crimial transportation aircraft
06:58.54YoMamawith big red and blue and green buttons on it
06:58.56IronHelixhttp://www.voipsupply.com/images/SP_IP601_3x.gif   sechsay for a receptionist :)
06:59.01QwellPikoro: :P
06:59.27YoMamawow..that's pretty badass
06:59.45YoMamaand where do u set up line appearances...in sip is the config on the phone itself or is it in one of the config files?
06:59.46Qwellokay, thats hardcore
07:00.02YoMamaCisco only lets you have up to two sidecars
07:00.11Qwellthe polycoms support proper presence, right?
07:00.30QwellI still won't buy a polycom though...they're ugly ;/
07:00.47YoMamathey make the best speakerphones on the market though
07:01.07IronHelixyeah, polycom speakerphones are awesome
07:01.17Qwelldoesn't cisco license the speaker from them?
07:01.20YoMamathe only time i'll hold a phone conf is when u got one of those starfish lookin' suckers in the middle of the table
07:01.24IronHelixthat used to be (and still is) one of their main businesses, making large expensive speakerphones for conference rooms
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07:01.38YoMamaQwell: yeah..7940's and 7960's have their chip in them
07:02.19YoMamawhich is why even tho the cisco phones are expensive as shit..it's still teh cheapest way to get a really decent speakerphone
07:02.47YoMamareally decent = people not being able to tell you're on speakerphone
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07:04.40harryvvpolycom has a conferance style speakerphone
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07:05.10harryvvqwell, ugly or not thay work. mine works well only gripe though is no backlite display
07:06.46Pikorohttp://pastebin.ca/28992
07:06.52Pikorodoes that look ok?
07:07.06Pikoroasterisk says it's picking up but the phone keeps ringing on the caller's side
07:07.37IronHelixpik your problem isnt in extensions.conf
07:07.44Pikorono?
07:07.48Pikorozapta.conf?
07:07.52IronHelixif asterisk says it picks up but it doesnt stop ringing, you have a zaptel problem
07:07.57Pikorohmm
07:08.10IronHelixpastebin the output of ztcfg -v, zapata.conf and zaptel.conf
07:11.45Pikorohttp://pastebin.ca/28995
07:12.49IronHelixyou are in japan i assume
07:13.28IronHelixand you have 4x fxo card (red)
07:13.58Pikoroyes
07:14.59UberbotSo, why doesn't the output of this show up correctly on by cid box?  exten => s,4,setcidname("${name}" <${number}>)
07:15.00xbmodder_lappyDoes festival have to be patched on gentoo to work with asterisk?
07:15.14Qwellxbmodder_lappy: didn't for me
07:15.36xbmodder_lappyQwell, well it didn't earlier but now I started the festival server--
07:15.40xbmodder_lappyand it wants me to
07:15.45UberbotAllI get is number number
07:15.55IronHelixuber- are you sure you dont watn ${CALLERIDNAME} and ${CALLERIDNUM}....  ${name} and ${number} will have to be set by something else
07:16.57Uberbot${name} and ${number} are set from an agi script...  The log indicates that the setcallerid is working.... It's just not displaying correctly on the phone.
07:17.15Uberbot<PROTECTED>
07:17.41QwellUberbot: quotes are bad
07:17.57UberbotBut they are specified in the syntax...
07:18.08Qwellwhat syntax?
07:18.19UberbotSetCallerID("Name" <Number>[|a])
07:18.26QwellNorth Antara <1234>
07:18.28Qwellis valid
07:18.30Qwell"North Antara" <1234>
07:18.33Qwellis not cool
07:18.35Uberbothttp://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SetCallerID
07:18.48Qwellwell, it's wrong
07:18.49UberbotI'll try it.
07:18.55UberbotSounds like it.
07:19.07maikUberbot: setCIDName != SetCallerID
07:19.15Qwellyeah, and that
07:19.18xbmodder_lappyQwell, do you have festival running?
07:19.22Qwellxbmodder_lappy: no
07:19.31xbmodder_lappyfestival --server I mean
07:19.38Qwellmaik: good catch...it's too late for this :p
07:19.52Pikoroso my zap.conf files look ok right?
07:20.27Qwellsetcidname, setcidnum, setcallerid, are all pretty deprecated in 1.2
07:20.41Qwellthe former two, at least
07:20.57xbmodder_lappyQwell, can you send me your festival.conf
07:21.12Qwellxbmodder_lappy: I don't have it setup here
07:21.35UberbotQwell, any pointer to documentation on these functions being deprecated in 1.2?  News to me.
07:21.42IronHelixpikoro- yeah they look ok
07:21.52QwellUberbot: show application setcidname
07:21.55IronHelixi found a mention in voip-info about enabling a JAPAN option when compiling zaptel
07:21.58IronHelixim looking into that atm
07:22.04Pikorothanks :D
07:22.13Qwellset(CALLERID(name)=blah)
07:22.21Pikorofunny thing is, it was working before and the only thing modified was the extensions.conf
07:22.39IronHelixwhat did you change?
07:22.58PikoroI didn't.  [tk]D-Fender changed it
07:23.09Pikorobut he said all he did was comment out one line
07:23.09UberbotGood enough.  Thanx.
07:23.25Pikoroso i uncommented it and same problem
07:23.27IronHelixbecause http://pastebin.ca/28992 looks fine
07:23.28Pikorokinda odd
07:23.30Pikoroyah
07:23.35IronHelixi dunno
07:23.39Pikoroi think i'll unplug and replug the lines or something
07:23.47IronHelixdid you try the usual 'black magic' troubleshooting?
07:24.00IronHelixie restart asterisk completely, reboot machine, kick side of server case, etc
07:24.09Pikorowell, i stripped down and did a rain style dance around the asterisk rack
07:24.10Qwellsacrifice any goats yet?
07:24.15Pikorobut that did not seem to help much
07:24.22Pikorofull reboot, yes
07:24.26IronHelixhmmm
07:24.34Pikorolet me try a shutdown instead of a restart
07:24.37Pikorocut power to the card
07:24.40Pikorobrb :D
07:25.29Pikorook, this is debian btw.. where can i put the modules so they start on reload automatically
07:25.41IronHelixin zaptel or asterisk source dir
07:25.42IronHelixdo make config
07:25.56tzafrir_laptopIronHelix, but it will only work if you kick the server with a boot that has proper bootstrapts
07:25.58IronHelixit will add entries in /etc/rc.d/init.d for zaptel and asterisk
07:26.02IronHelixhehehehe
07:26.12Pikoroyah, i don't have an asterisk init script
07:26.14tzafrir_laptopPikoro, /etc/modules
07:26.15{zombie}Pikoro: /etc/modules
07:26.18{zombie}snap!
07:26.25UberbotGot it working.  The agi script had to quote the value, since it had spaces in it, but the setcidname did not.
07:26.27IronHelixso you can stop/start/restart asterisk or zaptel with service _____ restart
07:26.32UberbotThanx.
07:26.38Pikoroyah.. i'm not to init script savvy
07:26.56tzafrir_laptopPikoro, if you don't have an asterisk init.d script htan you don't use a proper asterisk package. shame on you
07:27.00IronHelixpikoro- it does it for you
07:27.04Pikoropackage?
07:27.08Pikoroi'm using cvs
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07:27.32tzafrir_laptopPikoro, bad excuse
07:27.32Pikoron/m.. i got it
07:27.36Pikoroi'm an idiot
07:27.37Pikorohaha
07:27.37Pikorobrb
07:27.43IronHelixwhat was wrong?
07:27.49Pikorothe init thing
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07:27.54Pikoropowering off right now to reset the card
07:28.30harryvvinteresting platinum is almost $1,000 per ounce
07:28.56UberbotIs there any reason NOT to upgrade to 1.2 under Gentoo?  It's currently masked.
07:30.35Pikorook.. restarted
07:30.38Pikorotime to try this out
07:30.43QwellUberbot: shouldn't use packages.  just download the tarball from asterisk.org
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07:31.01*** join/#asterisk Timoti (n=asqsa@85.99.166.94)
07:31.14UberbotQwell, which distro do you use?
07:31.17Qwellgentoo
07:31.33UberbotIs there a reason you don't use the package manager (only for *?)
07:31.43Qwellonly for *.
07:31.54Qwellasterisk is pretty complex...it's best not to use packages for it
07:32.05TimotiHi I am new and have problem from extension to extension call .. whenever I call , rings the other party but whenever other party hang off then I get busy tone ... what would be the problem ?
07:32.15Pikorodo i need to run ztcfg each time i restart?
07:32.20IronHelixtimoti- are you using grandstream phones?
07:32.28IronHelixie bt100 bt101 bt102 GXP-2000
07:32.39IronHelixpik- yeah i think so
07:32.41Timotino ... I use sipura 3000 and Fritzbox fon
07:32.52Timotialso I use firefly softphone just to make sure
07:33.22Timotioubound call over the sip and iax works perfect .. but internal calls are not working
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07:34.02IronHelixso you can call another internal extension
07:34.04IronHelixand it rings
07:34.10IronHelixyes?
07:34.22Timotiyes but wheneever it hangs off .. I get busy tone
07:34.30Pikorodamn... incoming pstn still just rings
07:34.50IronHelixyou mean hangs up... so you can call and talk but when the other guy hangs up you get a busy signal?
07:35.05Timotiyes exactly ..
07:35.22Timotinoo .. wait I am confused
07:35.28Timotilet me explain like this
07:35.40IronHelixwhat would you prefer it do?  if you hang up the phone then you are expected to hang up on the other side anyway
07:35.45TimotiI call the other extension .. it rings .. and everything is perfect till now
07:36.04Timotibut whenever other party take the phone to talk to me .. I start to get busy tone
07:36.17IronHelixah
07:36.17IronHelixok
07:36.29Timotisorryy I got you wrong that is why I wrote wrong
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07:36.32IronHelixso when the person you call picks up, you get a busy signal instead of talking to them
07:36.35tzafrir_laptopQwell, nothing is too complex for a package manager. apache is one good example. There are simply too many packages developped outside of asterisk that depend on the asterisk codebase
07:37.23Pikorowow this is pissing me off
07:37.24Timotias far as I know that is not a codec problem .. boths ATA has ULAW
07:37.32Pikoroeverything works but this one thing
07:37.39IronHelixtimoti i think you have a media path problem
07:37.52IronHelixyou have a sipura 3000 and a fritzboxfon, both are registered to * correct?
07:37.53Timotihow can I solve this problem ?
07:37.57Timotiyes
07:38.12Timotiand even my firefly softphone also
07:38.17IronHelixdo the following
07:38.47IronHelix1. in sip.conf, in [general] set localnet= and externip= to be your correct *external* ip address (go to www.ipchicken.com to find it)
07:39.08IronHelixlocalnet= is your internal network / mask ie 192.168.1.1/255.255.255.0
07:39.25Timotino my router is 192.168.0.1
07:39.31IronHelixthen use that
07:39.36IronHelixi was saying an example
07:39.39IronHelixfor the format
07:39.41Qwellmask would be .0
07:40.09Timotiis it this one
07:40.10Timoti<PROTECTED>
07:40.16Timotisorryy
07:40.24Timotibindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
07:40.29Qwellno
07:40.30IronHelixno
07:40.34Timotiok wait
07:40.35IronHelixscroll down about 1 page
07:41.45TimotiI can not find it !
07:41.50Timotithere is nothing like this
07:41.58IronHelixits there, find for it
07:42.12harryvvlet me guess, no audio heard on one side of the phone?
07:42.13harryvv:)
07:42.20IronHelixfor me its on line 117
07:42.33IronHelixharry- no, his call disconnects as soon as the other side picks up
07:42.42harryvvohh thats strange
07:42.44IronHelixwhich can happen if no media path either way
07:43.02harryvvno rtp getting though
07:43.37Timotibut ... I really can not find it
07:44.21Timotishould I add it ?
07:44.28IronHelixnah, hang on with that for a sec
07:44.35IronHelixgo on to step 2
07:44.36IronHelix2. go into the setup pages for the sipura, fritzbox, and firefly softphone.  Look for information about NAT and STUN, anywhere you see anything about STUN and turn it off or disable it or make it blank
07:44.51Timotiok
07:46.25Timotiok .. you got it .. it works ...
07:46.53Timotibut my question will be .. I got my sound 0,5 sec later that my speech
07:47.07IronHelixits lagged?
07:47.18Timotino but only a bit delay
07:47.19IronHelixis your * box fast at all?
07:47.23Timotiis it normal
07:47.32IronHelixnot usually
07:47.39IronHelixesp. on a LAN
07:47.46IronHelixwhat kind of machine is asterisk running on
07:47.50Timotihimm
07:47.55Timotihp pavilion
07:47.58Timotinotebook
07:48.15IronHelixyou're running asterisk ona  LAPTOP?
07:48.16IronHelixhahaha
07:48.18TimotiP4
07:48.23IronHelixthats a new one
07:48.28Timotiyes .. is it bad
07:48.35IronHelixanyway, go back in sip.conf
07:48.39IronHelixfor each of the internal clients
07:48.45IronHelixie sipura, friz, firefly
07:48.49IronHelixadd canreinvite=yes
07:48.57IronHelixfor the SIP telephony providers
07:49.00IronHelixadd canreinvite=no
07:49.19Timotimay I ask what does that mean ? ( i am really new
07:49.34IronHelixthe STUN thing= STUN lets a sip client work through NAT.  You don't want this if you arent working through nat, just staying behind it
07:49.42IronHelixfor canreinivite
07:49.53IronHelixboth sides of the convo make a SIP connection to the * server
07:50.09IronHelixbut the audio doesnt go over SIP, it goes across another system called RTP, SIP just sets up the call
07:50.11*** join/#asterisk hadi57 (i=al_moghr@62.3.44.61)
07:50.32IronHelixonce the call si going, sip can send a 'reinvite', which redirects the RTP stream directly from one client to the other
07:50.33Timotishould I also add canreinvite= no for IAX
07:50.37IronHelixno
07:50.40IronHelixiax doesnt use reinvite
07:50.46Timotiok
07:50.50Timotionly for sip
07:50.54Timotilet me do that
07:51.09IronHelixyou do want internal clients to reinvite each other, but you DONT want the sip service provider to reinvite because of the NAT
07:51.28IronHelixbecause the ports are forwarded to the asterisk box, not clients
07:52.32Timotiit works great now ...
07:53.03Timotiso for every sip voip provider ... I should say " no for canreinvite
07:53.05*** join/#asterisk shmooz (n=shmooz@H142.C72.B0.tor.eicat.ca)
07:53.05Timotiright ?
07:53.22IronHelixexactly
07:53.24*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)
07:53.33Timotiok ... many thanks iron
07:53.33IronHelixnot because they're a provider, but because they arent on your internal network
07:53.49IronHelixno problem
07:53.57IronHelixbuy a real server! :)
07:53.59Timotiiron .. my last question is .. when i try to register my ATA to asterisk ... it takes for a while to register
07:54.19IronHelixhow long is a while
07:54.21Timotibut even for the remote SIP provider .. it takes much more faster
07:54.36Timotisomething like 6-8 sec
07:54.42UlbabraBgiorno
07:55.00IronHelixthats not taht bad...  by register i assume you mean log in when you plug it in?
07:55.21Timotiby the fritz fon I have different sip provider ...
07:55.40UlbabraBehm...I meant "hi!"
07:55.44IronHelixhi
07:55.49Timotieach of them register almost 3-4 second but to asterisk ( at the same network ) it register 7-8 sec
07:56.13IronHelixi think your server may be lagged or slow a bit
07:56.16Timotii see them at the summary page
07:56.29Timotihimm .. by which meaning ?
07:56.31IronHelixthats why setting canreinvite fixed the audio thing
07:56.38IronHelixlike your server takes a while to deal with packets
07:56.52Timotiyou mean my notebook is slow ?
07:56.59IronHelixor something is slowing ti down
07:57.13Timotiwell only asterisk is running on it
07:57.30IronHelixat the command prompt (not asterisk) do
07:57.32IronHelixps -A
07:57.36IronHelixand pastebin what you get
07:57.41IronHelixhttp://www.pastebin.ca
07:58.08marc324whats the best way of having voicemail recordings displayed on a webpage?
07:59.38IronHelixhttp://www.voip-info.org/wiki/view/Asterisk+gui+vmail.cgi  try that
08:00.14Timotiiron .. I did it .. but I have no idea how to copy and paste it .. :-) I am really new at this linux stuff
08:00.36IronHelixjust look at the times next to each process
08:00.40TimotiI do that chat over windows .. but on the other machine asterisk is running
08:00.47TimotiI did
08:01.01Timotialmost all of them 00:00:00
08:01.04IronHelixwhat are the 5 processes with the highest times
08:01.07IronHelixthats good
08:01.08IronHelixwhich ones arent
08:01.18*** join/#asterisk xtr (i=01928375@S01060012174cc0e1.vf.shawcable.net)
08:01.30Timotihow can I use this command with " pause "
08:01.43TimotiI only use the last 20
08:01.47*** part/#asterisk bumblefsck (n=bumblefs@69-160-145-156.ontrca.adelphia.net)
08:01.49Timotisorry see
08:01.50IronHelixps -A | more
08:02.09IronHelixthat will display them so you can go through them
08:02.16Timotiinit 00:00:04
08:02.39Timotirest are 00:00:00
08:02.41IronHelixanything using more than 00:01:00
08:02.47Timotinoo
08:03.26IronHelixis your network segment overloaded at all?
08:03.32IronHelixgot any huge downloads, transfers, etc?
08:03.32Timotinot really
08:03.42Timotinot at all .. I hate huge downloads
08:04.01TimotiI only use 2 gb transfer ( at most )  per month  :-))
08:04.15IronHelixwhat about between computers on the lan
08:04.28Timotiwell  I think it is something related fritz boy or
08:04.44TimotiI have only 2 nothebook on the network
08:04.49Timotiboth I use them
08:04.56Timotione asterisk another my own
08:05.11Timotiand also couple of ATA
08:05.44TimotiI think I need to change some stuff at fritz box .. or .. but at the fritz box not that much to change
08:05.57IronHelixthey are registering to the internal ip right?
08:06.03IronHelixie 192.168.xxxxx
08:06.12Timotiyes to 192.168.0.60
08:08.19YoMamaIronHelix: Grandstream only has 1.0.1.9 firmware up on their website for the GXP...is there another place to get the .12 u were talking about?
08:08.24Timotihimm .. well as long as it register to network .. no problem .. but it is really odd that it register much later than any other one
08:08.30IronHelixhttp://www.grandstream.com/BETATEST
08:09.05YoMamaoh momma...beta test :-P
08:10.12Timotiare there any way to connect to my asterisk server from windows  and make use the command over that
08:10.13{zombie}YoMama: and version 13 is at http://www.voip-info.org/users/557/15557/images/116/GXP2000_Release_1.0.1.13.zip
08:10.28{zombie}which is the version with the BLF functions
08:10.35*** join/#asterisk ComputerWarm (i=Computer@rddrpx27-port-44.dial.telus.net)
08:10.38IronHelixtimoti- SSH
08:10.43ComputerWarmEvening all
08:10.44IronHelixgoogle for a program called PUTTY
08:10.56Timotiok .. I'll do that
08:10.56*** join/#asterisk mcnobody (n=laaksola@laaksola.net)
08:10.59YoMama{zombie}: BLF?
08:11.09IronHelixbusy lamp field
08:11.13YoMamaoooh...
08:11.15IronHelixaka lights that light up when ppl are on the phone
08:11.21YoMamaperty
08:11.28Timotiiron  many thanks for your kind help
08:11.33IronHelixno problem
08:11.34IronHelixgood luck!
08:11.42Timotilet me go and work my fun machine a bit more
08:11.50Timotibye
08:11.58IronHelixcya
08:12.08YoMamais anyone working on a login for the phones so u can get your extension
08:12.16YoMamadude..i'm totally clueless but if he doesn't know what ssh is...
08:12.30YoMamalike what cisco's got with extension mobility
08:13.10IronHelixSSH = secure shell
08:13.15IronHelixlogin to asterisk over the internet
08:13.28IronHelixyomama- you can do that in asterisk with a bit of configuation
08:13.39IronHelixset up yourself a call queue using ringall style
08:13.41IronHelixthen log into it
08:13.48YoMamaIronHelix: no..i know what ssh is..i'm saying if he doesn't know what it is...that's sorta scary
08:14.03IronHelixahh
08:14.05IronHelixhes new
08:14.18YoMamayeah...yer a more patient man than i
08:14.21ComputerWarmeveryone has to start some where..
08:14.57YoMamaif someone came on here and asked me for help..i'd be ok with it until i figured out they don't know how ports work or what ssh is..etc..i'd tell 'em to go get Unix in a nutshell first and then come back
08:15.02ComputerWarmi remember when i first started with ASterisk bkw helpped me ALOT lol
08:15.16*** join/#asterisk cjk (n=cjk@80.92.64.103)
08:15.25YoMamahmm..looks like atacomm's got the GXP for the cheapest price
08:15.38IronHelixhow much?
08:15.41YoMama$85
08:15.44YoMamai froogled it
08:15.56YoMamadude..i love google...those guys are effin' geniuses
08:15.57IronHelixthat is pretty good
08:16.07IronHelixi generally use voipsupply,com, they're about $100
08:16.20YoMamayeah..iv'e heard good things about voipsupply
08:16.46YoMamai gotta get me a switch that does QoS
08:16.52YoMamato do this right :)
08:17.13IronHelixnot for a small LAN
08:17.35IronHelixno need
08:17.50IronHelixwhen the combined total of all traffic wont overload your gateway or exceed 100mbit you're fine
08:18.00IronHelixmaybe get a gateway that has qos
08:18.02IronHelixthat always helps
08:18.10IronHelixlinksys wrt with sveasoft = pretty good
08:18.31*** join/#asterisk high| (n=morten@80.239.93.22)
08:18.55Pikoroanyone know if there are any asterisk-sounds packages in japanese?
08:19.04YoMamakonichiwa
08:19.28YoMamaIronHelix: eh...u can get a cisco 2924 for pretty cheap these days
08:19.35Pikoro??????
08:19.39Pikorohmm..
08:19.43Pikorono japnaese support in mirc
08:19.43IronHelixpiko try this http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international
08:20.07YoMamaPikoro: arrigato
08:20.18YoMamathat's about all i know in japanese
08:20.21PikoroYoMama: yo ma ma
08:20.29YoMamaoh..and i know lots of sushi lingo
08:20.49YoMamauni..ebi...hamachi...
08:20.56Pikoroheh
08:21.01Pikorohotate
08:21.05Pikoroscallop
08:21.10YoMamammm...hotate is good stuff
08:21.24Pikoroyah
08:22.55YoMamak..i just ordered the GXP
08:23.07IronHelixwhich one?
08:23.18YoMamadoes anyone get any popping noises with x-lite?
08:23.32YoMamai was using it today to call a friend and he complained it was a little warbly and had some popping
08:23.34IronHelixyomama- whcih model did you order?
08:23.39YoMamaIronHelix: the GXP-2000
08:23.42shmoozlol, forgot your popcorn in the microwave ?
08:23.44IronHelixdude thats the wrong one
08:23.46YoMamawhat?!
08:23.51IronHelixyou want the GXP-2200
08:23.56IronHelixthe 2000 doesnt work wtih asterisk
08:24.00YoMamaare u serious?
08:24.13Pikoroi have a 2000
08:24.13IronHelixnah, just fucking with ya :)
08:24.19Pikoro:p
08:24.23YoMamaBAAAAAAAAAH
08:24.31YoMamaIronHelix: thanks for the mild heart attack
08:24.32IronHelixthere is no gxp2200
08:24.37IronHelixyou're welcome :)
08:24.37Pikoroheheh
08:24.48Pikorowow.. there's lots of funny sounds that come with asterisk
08:24.52Pikorolike dem monkeys
08:25.18YoMamai'm trying to think of a proper greeting when someone calls me at home
08:25.25IronHelixyeah, i built a whole ivr out of nothing but the funny noises that come with asterisk-sounds
08:25.31YoMamathat doesn't sound like a cheesy business..but if i get a business call..they aren't like "that's stupid"
08:25.48PikoroYoMama:  Play(goodbye)
08:25.52PikoroHangup
08:26.00IronHelixyomama- Hello, you have reached (PUT YOUR NAME HERE).  Please wait while I am located.
08:26.19Pikorothen put em on hold with some zamfir pan-flute music
08:26.33YoMamaha..no..i got a local radio station pipedin
08:26.48IronHelixyomama- fyi that can create possible copyright issues
08:26.51Pikorocan you do that?  like point it at a stream?
08:26.57IronHelixmusic on hold is considered a 'public performance'
08:27.01IronHelixthats why everybody uses classical music
08:27.13IronHelixcuz the copyright expired before there was such a thing as copyright
08:27.19YoMamaIronHelix: bah..no one's gonna come after me...and it's pointed to a public radio station
08:27.28IronHelixyeah pik- you can use a stream for music on hold
08:27.40YoMamaPikoro: shoutcast baby
08:27.42*** join/#asterisk opus_ (n=opus@dahphish.org)
08:27.42IronHelixthat wont protect you, its consider a redistribution, just hope nobody comes after it
08:27.48YoMamai gotz it werkin'
08:27.52YoMamahmm
08:28.42Pikoroi need to get this incoming pstn stuff to work
08:29.04YoMamaif i wanna do fax...probably need to get a real FXS port eh?
08:29.20Pikorovoip fax?
08:29.29YoMamano..just regular fax
08:29.34IronHelixyomama- an ata will work, sort of
08:29.41YoMamayeah..sort of
08:29.48IronHelixfxs wont be much better
08:29.54IronHelixunless the call is coming in on another zap chan
08:29.54YoMamai've tried using Cisco ATA 186's running with G.711u
08:29.59*** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it)
08:30.06YoMamaIronHelix: that's what it'd be...Zap to Zap
08:30.06*** join/#asterisk implicit (n=implicit@216.13.124.132)
08:30.11*** join/#asterisk rking (n=rking@ip68-1-234-152.dl.dl.cox.net)
08:30.17IronHelixah, then turn off echocancelwhenbridged and you're set
08:30.22YoMamanice
08:30.28opus_async patch == make fxs port work foo good
08:30.44IronHelixyou can also isntall spandsp and rxfax and have asterisk email you your faxes
08:30.50YoMamaIronHelix: what about FXS -> SIP proxy?  ugly?
08:30.56ComputerWarmCan anyone recommend any billing system for asterisk that works?
08:31.10opus_computerwarm does a2billing work?
08:31.28YoMamaIronHelix: can u make asterisk recognize a fax coming in and route it to the correct agent?
08:31.33ComputerWarmopus_  not for me... but i think thats because i am running php5
08:31.34IronHelixyuppers
08:31.41IronHelixexten => fax,1,Dial(faxport)
08:31.45opus_computerwarm that sucks
08:31.53IronHelixBUT you have to answer the channel
08:31.59ComputerWarmopus_ been working on it for about a week, not trying to fix it up. but there is still ALOT to do
08:32.11YoMamaIronHelix: whaddya mean?
08:32.17IronHelixcall comes in
08:32.19YoMamak
08:32.21IronHelixif you just dial(somebody)
08:32.32IronHelixthe zap channel wont be actually answered until somebody picks up
08:32.32YoMamawhat if it's going to an AA?
08:32.36*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
08:32.38IronHelixthus asterisk wont hear the fax tones
08:32.49IronHelixyou can do
08:32.52IronHelixanswer()
08:32.59IronHelixthen dial(somebody,,r)'
08:33.01*** join/#asterisk Brumle (n=brumle@brumle.com)
08:33.05IronHelixhowever that will actually pick up the line
08:33.10YoMamaright
08:33.20IronHelixand if the guy is calling from a payphone, they will lose their quarter, even if YOU dont answer
08:33.20YoMamahmm
08:33.35YoMamasomeone should write something similar to background
08:33.35opus_they'll get answer supervision
08:33.42opus_unless the ground the green wire!
08:33.57YoMamabackground listens for tones...BackgroundPlus should listen for fax tones or something
08:34.09IronHelixbackground will answer the channel i think
08:34.10YoMamaopus_: you've been watching too much wargames
08:34.25YoMamaIronHelix: u know..but how do u get it to recognize the fax tones and send it to the right program
08:34.28IronHelixit should listen for fax tones during bcakground
08:34.46IronHelixcreat an extension called 'fax' in the same context
08:34.48opus_mcfax
08:34.55Pikorodialup?
08:34.55opus_ironhelix
08:34.58Pikorono way
08:35.04IronHelixsup opus
08:35.21opus_what are you doing for a job
08:35.33IronHelixlike right now or like in general
08:35.42opus_like right now
08:35.43Pikoroyou're a general?
08:35.48Pikoro;p
08:35.53ComputerWarmPikoro don`t ask lol some weird friend of mine wants me to route him an extension to his dial up phone...  I have it working not bad with GSM, thinking of trying G729
08:35.54*** join/#asterisk psk (n=psk@golia.caltanet.it)
08:35.58IronHelixright now i'm sitting here chatting on irc blowing off time
08:36.02IronHelixin general im a consultant
08:36.05opus_hahaha blah blah blah
08:36.16opus_are you accepting any new work
08:36.20IronHelixi fix, setup, configure, or otherwise screw with anything containing or related to computers
08:36.30IronHelixsure
08:36.32IronHelixwhatcha got?
08:36.35Pikoroincluding legos
08:36.37opus_cuz i seen your work
08:36.47IronHelixheh where?
08:36.51IronHelixohhh
08:36.59IronHelixgotcha
08:37.01IronHelix:)
08:37.26*** join/#asterisk testmachine (n=assink@ip237-239-58-62.adsl.versatel.nl)
08:37.29opus_i want to quote some work out but i am to buzy
08:37.35opus_they need help
08:37.43IronHelixyeah i can help out, whats the client
08:37.54opus_old client of mine
08:38.15opus_1.2 with realtime though, you know how to do that stuff
08:38.35IronHelixdont have any realtime experience (yet) :(
08:38.46Qwellrealtime is pretty easy
08:38.52opus_ok well whats your addr i will ask you offline
08:39.22IronHelixygpm
08:39.39Qwellwell, there's a new acronym
08:39.46IronHelixYou Got Private Message
08:39.48Pikororipe?
08:39.52Qwellyeah, I got that
08:39.52Pikororealtime is pretty easy
08:40.01QwellPikoro: stop with the bad puns. :P
08:40.05Pikorohaha
08:40.08opus_no ygpm no bueno
08:40.11Qwelleven _I_ can't take any more...
08:40.19IronHelixlol
08:40.24Pikorowell i gotta hang out on here until i can figure out my problem
08:40.31Pikoroor i get kicked off
08:40.34Pikoroheh
08:40.39opus_whats your problem?
08:40.40Qwellwell, pun away then
08:40.46opus_btw, asterisk sucks. real men run openpbx
08:40.50Pikoroincoming pstn over a zap channel
08:40.51IronHelixheh
08:41.03Pikoroasterisk says it picks up, but the caller's phone keeps ringing
08:41.08Qwellreal men route the SIP traffic by hand
08:41.12IronHelixalso- hes in japan
08:41.17Pikoroyah.. japland
08:41.21IronHelixhas zap set to japan settings
08:41.30opus_Pikoro your zapata.conf is farked-ified
08:41.47Pikorohttp://pastebin.ca/28995
08:41.51Pikorocan you unfark it?
08:41.51*** join/#asterisk BladeRunner05 (n=feelme@adsl-ull-90-66.44-151.net24.it)
08:41.53IronHelixthe only thing i can think of is if they dont use kewlstart
08:41.54shmoozopus_ openpbx is entirely written in perl
08:41.57Pikoroor tell me how to unfark it?
08:42.13opus_shmooz did you design it?
08:42.14BladeRunner05IronHelix: but u don't sleep never !
08:42.27shmoozopus_ yes ;]
08:42.30IronHelixi do occasionally
08:42.33opus_shmooz hehe
08:42.36Pikoroouch.. imagine a transcoder written in perl
08:42.36IronHelixi was actually going to go to sleep soon
08:42.39opus_shmooz Well cool:)
08:42.43YoMamawow...very cool
08:42.46Pikorotalk about a cpu usage nightmare
08:42.46Pikoroheh
08:42.51YoMamathe fax thing that is
08:43.07IronHelixyeah
08:43.16opus_shmooz what do you think of anthm's efforts w/ openpbx.org 2.0?
08:43.17*** join/#asterisk lme (n=lme@gw-r-ifc.infoclip.fr)
08:43.19IronHelixif you set it up right you can email the fax to your * box and it will be sent
08:43.29lmehi guys
08:43.41MozartsGhostholy crap
08:43.41MozartsGhost: idle     : 165 hours 57 mins 20 secs (signon: Thu Nov 10 12:31:49 2005)
08:43.44IronHelixsup lme
08:43.50MozartsGhostlook at my idle time.
08:43.50opus_ironhelix, i didn't get any pm from you btw
08:43.58MozartsGhostneways.
08:44.00PikoroMozartsGhost, that's along time in the bathroom
08:44.08shmoozopus_ jk I just heard of it now, just reading about it
08:44.10IronHelixstrange... resending
08:44.12MozartsGhostlol
08:44.28IronHelixget that?
08:44.34opus_do you see my /msg pm?
08:44.41IronHelixno
08:44.43IronHelixhow strange
08:44.51YoMamaIronHelix: that's pretty phat too
08:45.03IronHelixohhh
08:45.04opus_ok send me e-mail to opus@dahphish.org real quick
08:45.04IronHelixhangon
08:45.09*** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com)
08:45.11IronHelixfor some reason it un identified me
08:45.12IronHelixstand by
08:45.17opus_standing by
08:45.21MozartsGhosteish, having a strange problem on one of my boxes. runnin TDM04P. "Unable to create channel of type 'Zap'" I get that after its been runing for about 20 hours
08:45.42MozartsGhostworks perfect for the first 20 hours tho.
08:45.47MozartsGhostanyone know about that ?
08:45.53*** join/#asterisk IronHelix (n=irc@ool-45785cfe.dyn.optonline.net)
08:46.10YoMamaMozartsGhost: exactly 20 hours?
08:46.18opus_MozartsGhost run 'valgrind' on it its obviously not closing a socket or somethign
08:46.28MozartsGhostyea well, normal round about 20 hours, sometimes more
08:46.36opus_your running out of FD
08:46.50sylemozart i use to get that, then i redid my dialplan to make sure hangup was called everywhere and that solved it
08:47.33*** join/#asterisk steff (n=steff@80.125.254.220)
08:47.48steffhi all
08:47.58YoMamaok
08:48.01YoMamai'm going to bed
08:48.04YoMamahave a good night all
08:48.14Igbothombed with yomama?
08:48.36Igbothomyomama's a BitchX?
08:48.43opus_Pikoro did you figure it out?
08:50.00*** join/#asterisk zcat[1] (n=zcat@203.89.166.230)
08:50.14opus_aww later
08:50.15*** part/#asterisk opus_ (n=opus@dahphish.org)
08:50.37Pikoronot yet
08:51.15cjkhi, are there any known problems with ilbc in 1.2
08:51.54cjkshow translation shows me extremly high transcoding times and its not rellay working to g729
08:52.58*** join/#asterisk IgorG (n=gia@195.162.32.126)
08:53.20MozartsGhostmy default "outgoing" did not have Hangup on it, only Congestion.
08:53.31Pikorothat'll teach you to put a pi symbol in the lower right corner of your page
08:53.42MozartsGhostlol
08:53.52MozartsGhostsyle: u think that would be my problem ?
08:54.51IronHelixlol i thought i was the only one that'd seen that movie
08:55.33Pikoroyou were.. i had to google it
08:55.36Pikoro;p
08:56.25Pikorowell i think i'll drop this and try again in the morning
08:56.33*** join/#asterisk MuppetMaster (n=MuppetMa@81.184.73.169)
08:56.39*** part/#asterisk MuppetMaster (n=MuppetMa@81.184.73.169)
08:56.40*** join/#asterisk MuppetMaster (n=MuppetMa@81.184.73.169)
08:56.44Pikoroalmost 6pm.. time to go home
08:58.04IronHelixalmost 4am, time to hit the sack
08:58.15MGSsancholater dude
08:58.20IronHelixcya
08:58.23Pikorothanks for the help
08:58.24Pikoronight
08:58.26IronHelixnp
08:58.28IronHelixgood luck
09:01.30*** part/#asterisk MuppetMaster (n=MuppetMa@81.184.73.169)
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09:01.33*** part/#asterisk MuppetMaster (n=MuppetMa@81.184.73.169)
09:02.10*** join/#asterisk zgor (n=zgor@61.Red-80-36-3.staticIP.rima-tde.net)
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09:03.53*** join/#asterisk grimse (n=grimse@p5481FF5F.dip.t-dialin.net)
09:07.46ComputerWarmquestion. on sending via sip. is there anyway to force asterisk to send with one ip address but still receive on another?
09:11.21Vhataerr...  why?
09:12.26ikarusComputerWarm: sending what via SIP ?
09:12.48ComputerWarmbecause i screwed up when i signed up with a company. and i can`t get them to change the ip address right now because they are not open
09:12.54ComputerWarmikarus Long distance route
09:13.23ikarusComputerWarm: it is more that SIP is only session setup, RTP is data
09:13.58ComputerWarmok well all they care is they see the attempts coming from my back up ip address
09:14.02ComputerWarmnot my primary one
09:15.43*** join/#asterisk lme (n=lme@gw-r-ifc.infoclip.fr)
09:20.36ComputerWarmok next question there was someone i was talking to a few nights ago that told me where i can go and get the source for G723 for testing my asterisk system. if anyone knows where i can get it please tell me. as i lost the url
09:29.54*** join/#asterisk grimse (n=grimse@p5481EEF1.dip.t-dialin.net)
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09:38.41lmewell well... time to implement dundi....
09:39.54lmegonna peer with myself !
09:41.08*** join/#asterisk steff (n=steff@80.125.254.220)
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09:44.12*** join/#asterisk delox99 (n=delox99@206-248-149-59.dsl.teksavvy.com)
09:44.22delox99hi all
09:45.10schulahi, is there any posibility to use chan_bluetooth to handle incoming calls? i want to recieve incoming calls from my mobile phone
09:46.57delox99can someone tell me why in my musiconhold.conf file the layout is [default]
09:47.05delox99mode=quietmp3
09:47.20delox99directory=blabla...
09:48.04lme??
09:48.08delox99on voip-info they suggest to configure using default=quietmp3:/var/lib/aserisk/mohmp3
09:48.18*** join/#asterisk Jas_Williams (n=Jason@host86-129-251-140.range86-129.btcentralplus.com)
09:48.21delox99oups sry
09:48.38delox99default => quietmp3:/var/lib/asterisk/mohmp3
09:49.00*** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com)
09:49.01delox99i think i have an old moh sample conf file
09:49.35delox99how can i install sample conf files to another directory using make samples?
09:51.37*** join/#asterisk grimse (n=grimse@p5481D645.dip.t-dialin.net)
09:52.21lmedelox99: you can grab files directly form asterisk sources
09:52.41delox99oh yeah true
09:52.51testmachine:)
09:53.03lmedelox99: configs subdir
09:53.11testmachineanybody here uses amportal?
09:55.20delox99thanks
09:55.27delox99i just checked and it s the good file
09:56.03delox99the problem is that the syntax in the conf file is completely different then what is on voip-info
09:56.38sylethen update it plz
09:57.01delox99update voip-info you mean?
09:57.08syleyeah
09:57.20*** join/#asterisk darby_t (n=merci@80.72.34.118)
09:57.26delox99bu the thing is that i just dont know how it now works
09:58.31testmachineits just a wiki
09:58.49delox99it seems obvious but i had mpg321 installed on my Debian sarge so i removed it and i want to go with madplay
09:58.53testmachinePlease update this page with new information, just login and click on the "Edit" or "Add Comment" button above. Get a free login here: Register Thanks! - support@voip-info.org
09:58.56testmachine:)
09:59.01testmachineat the bottom of the site
09:59.04delox99no not the weiki but the new syntax for musiconhold.conf
09:59.37delox99=)
10:00.08delox99i installed madplay and i m trying to tell asterisk via musiconhold.conf to play the files
10:01.07testmachineyou can edit those config files verry east
10:01.10testmachineeasy
10:01.12testmachinewith AMP
10:01.21testmachinehttp://sourceforge.net/projects/amportal
10:01.32testmachineits an php based webinterface
10:02.21delox99asterisk says that i need to specify an application to play the default class
10:02.30delox99but it should be well configured
10:02.37testmachinehm
10:02.53schulaor is there any working channels for gsm modems/mobile phones? didnt found any one :(
10:02.59*** join/#asterisk Moc- (n=mochouin@modemcable181.215-82-70.mc.videotron.ca)
10:03.06testmachinedelox99: why you dont want to use mpg123?
10:03.06*** part/#asterisk Moc- (n=mochouin@modemcable181.215-82-70.mc.videotron.ca)
10:03.32delox99with AMP i m afraid i ll have to go through a package dependency kinda mess
10:03.56testmachinei didnt have really problems with it
10:05.21delox99id mpg123 gpl licensed?
10:05.25delox99is*
10:06.02delox99it s not available on the debian package mirror im using
10:07.00delox99yeah looks like its gpl
10:07.28lmeahhh debian....
10:07.55lmei will burn mine one day
10:07.59*** join/#asterisk test34 (i=1000@unaffiliated/test34)
10:10.28delox99i just found that mpg123 is for non comercial use
10:10.44delox99thats why i need an alternative
10:12.10*** join/#asterisk Timoti (n=asqsa@85.96.7.62)
10:12.20TimotiHi everybody
10:12.21TimotiHi I have a newbei question related asterisk .. about codec conversion ... does asterisk do the codec conversion by itself or should I do something special ..
10:13.10konfuzedusing asterisk is kinda special
10:13.27Timotiexample .. my Sipura has only couple of codex except GSM codec.. when I connect to asterisk .. and over the asterisk can I connect to voip provider with GSM codec or not
10:14.25sylesock-ra-tees
10:14.45Timotino answer ?
10:15.35konfuzedTimoti: each device node handling the call in the chain manages its own com link
10:16.20konfuzedboth devises on each side of each link have to use the same codec protocol combo
10:16.30Timotiso it means that ATA >>> G711>>>> Asterisk >>> GSM >>> Voip provider is possible ?
10:16.45konfuzedyes
10:17.14tzafrir_laptopdelox99, it is a bit more complicated. its license sort of has changed
10:17.33Timotiwouww .. but a question .. some companies are offering codec conversion to charge the customer .. how is that ?
10:17.41tzafrir_laptopsort of: the site claimes that the license changed. The tarball was not updated yet
10:17.45konfuzedas long as your voip provider hands out gsm
10:18.19konfuzedwith computers any thing is possible
10:18.42Timotihimm ...
10:18.45tzafrir_laptopcheck http://bugs.debian.org/mpg123 ...
10:18.55konfuzedputting money on the table gets those techies to be much more creative and focused
10:19.02konfuzedor at least focused
10:19.05konfuzedusually
10:19.11konfuzedwheres my chocolate
10:19.44konfuzedbah
10:19.48Timotihim
10:19.54konfuzedmore ice cream then
10:20.04Timoti:-)
10:20.24*** join/#asterisk pif (n=ldm@zenon.apartia.fr)
10:20.25tzafrir_laptopdelox99, also note that mp3 *encoding* generally requires a license for a patent if you're in the US
10:20.27pifhi, what 24 port POE switch would you recommend for a VoIP installation?
10:20.33Timotiis an notebook  really bad for an asterisk server .. ?
10:21.27tzafrir_laptopTimoti, why?
10:22.02Timotino just a question ...
10:22.27Timotijust to know how important is the hardware for asterisk
10:23.39konfuzedhardware capacity directly corelates to number of calls and call quality
10:23.51*** join/#asterisk adelas (n=booger@rrcs-24-199-21-141.west.biz.rr.com)
10:24.30konfuzedthe only factor really out side that is the out side lines coming in and the quality of your internet bandwidth
10:24.52konfuzedeverything else is asterisk running on the os on your Hardware
10:25.26delox99tzafrir_laptop: true for encoding licenses
10:25.28Timotiok ..thanks for info konfuzed ..
10:25.40delox99have you ever used format_mp3?
10:25.51delox99i m just confused here
10:25.54Timotifor G729 .. if I pay the licence fee.. how can I install it ?
10:26.00konfuzedtheres a section at voip-info for selecting hardware to match needs
10:26.17konfuzedI forget the word but similar to provisioning
10:26.33delox99my musiconhold.conf syntax is just not the same as the one on voip-info and i dont know where to find new documentation for that file
10:26.42viperdudei have a background() message that asks for an extension however when you enter an extension and the other party answer the CLI reports Forbidden - wrong password on authentication for INVITE and the called party can't hear
10:26.53viperdudeany ideas?
10:27.05cjkhi, anyone know an easy way to do some asterisk-stressing
10:27.05delox99if i could just get that docs maybe i could just use madplay or fomat_mp3
10:28.00Timotithanks for info everybody
10:28.03Timotihave a great day
10:28.05Timotibye
10:29.03delox99i have to go to bed its 6 am here iiii
10:29.14delox99see y all later
10:32.35*** join/#asterisk rikard_bj (n=rikard@81-236-29-213-no10.tbcn.telia.com)
10:32.45rikard_bjhello
10:32.47rikard_bjis there any sample application anywhere that places a call, then places another call and connects the two calls? Using the manager interface for example?
10:37.48*** join/#asterisk grimse (n=grimse@p5481D853.dip.t-dialin.net)
10:39.53c0wcjk: depends on how you want to stress it.
10:40.11c0wcjk: like number of calls per/sec number of concurrent calls etc
10:46.35*** join/#asterisk sysdebug (n=sysdebug@200.163.193.247)
11:10.29*** join/#asterisk flenders (n=chatzill@b03DE.static.pacific.net.au)
11:17.18Falleis there a nicer way to reload asterisk from an internal extension then a system(asterisk -rx reload)?
11:17.38*** join/#asterisk grimse (n=grimse@p5481DFAA.dip.t-dialin.net)
11:23.32c0wtell ya what
11:23.41c0wit doesn't have cost a fortune to send stuff to the US
11:23.58c0wand the paperwork you have to fill in is just stupid.
11:24.26c0w6 invoices/ and $250 ish to send 3 quad e1 cards to the us.
11:24.49*** join/#asterisk areski (n=areski@polar.es6.egwn.net)
11:27.24*** join/#asterisk santoshr (i=1063@203.199.110.93)
11:30.17santoshrpeople asterisk 1.2 is registering inband dtmf where as the same instrument tht i use sends dtmf properly to 1.0.x asterisk
11:30.32santoshrpeople asterisk 1.2 is not registering inband dtmf where as the same instrument tht i use sends dtmf properly to 1.0.x asterisk
11:30.54cjkc0w: i want to stress concurrent calls
11:31.08santoshranybody seen a similar behavious
11:31.10santoshrr
11:31.32nextimewich gui you suggest for a customer care oriented call center ?
11:31.48santoshrafter demo-instruct it does not register "500" but asterisk 1.0 does why ?
11:32.50santoshrpeople
11:32.52santoshr.
11:34.37santoshrhelooooo
11:35.26tzafrir_laptopsantoshr, which phone exactly?
11:36.22santoshrits fxs box
11:36.26santoshrwellgate
11:36.43santoshrto fxs box is connected a normal phone.
11:37.17ikarusI am going to kill someone, to be exact, the guy that was supposed to get me an ISDN card here
11:37.50santoshrwhen i dial 500 after demo instruct it does not do nething.. but when runnig asterisk 1.0.x it does stuff. i mean it atleast acknowledges tht i pressed some digits
11:38.37*** join/#asterisk blop (i=blop@VoIP-with-Asterisk.mgcp.h323.sccp.sip.iax.be)
11:41.15santoshrtzafrir_laptop: dude u around
11:42.25cjkis there a way in iax or/and sip to tell a client to register at a different server?
11:44.24santoshrhas anybody get dtmf to work with asterisk 1.2
11:45.31santoshri mean inband to work with asterisk
11:47.36c0wcjk: then just get as many people as poss to ring others on your box.
11:47.50c0wusing h323, 2xe1 - 60 channels inbound
11:47.58c0wi had 48 concurrent last night
11:48.07c0wwith a load of .7
11:48.16*** join/#asterisk sysdebug (n=sysdebug@200.163.193.247)
11:55.45*** join/#asterisk jbot_ (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
11:55.45*** topic/#asterisk is Asterisk 1.2.0 has been released! -- ftp.digium.com || http://www.asterisk.org
11:56.21*** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com)
11:56.31santoshryes it is.. setup
11:56.43santoshrproperly in h323.conf dtmf=inband ..
11:57.03santoshrthe fxs boxis a h323 based 1fxs device
11:57.28*** join/#asterisk ManxPower (n=eric@adsl-67-65-233-194.dsl.lgvwtx.swbell.net)
11:58.29santoshrthe same setup is working with asterisk 1.0.x
12:01.17ManxPower~docs
12:01.19jbotdocs is, like, probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com
12:01.20ManxPower~mailinglist
12:01.21jbotmethinks mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives/search.php
12:06.40tzafrir_laptophow can I save/restore /var/lib/asterisk/astdb through an external program? any standard berkeley db dump program on debian?
12:07.10tzafrir_laptopI tried db*_dump from libdb[23]_util , and both fail to recognize the file
12:07.21ManxPowertzafrir_laptop: I would assume so
12:08.15tzafrir_laptopfile(1) claims it is Berkeley DB 1.85/1.86 (Btree, version 3, native byte-order)
12:09.33{zombie}db_dump185 -p /var/lib/asterisk/astdb
12:09.33*** join/#asterisk fishboy1669 (i=proxyuse@62.69.81.129)
12:10.15tzafrir_laptopso if I want to backup /var/lib/asterisk/astdb while asterisk is running and later restore it, what do I need to do?
12:10.32{zombie}I don't know about restoring it, but you can back it up with the above command
12:10.38{zombie}redirect it to a file with >
12:11.38santoshrinband dtmf on asterisk 1.2
12:12.45tzafrir_laptop{zombie}, does not work with db_dump185 from libdb2-util
12:13.08lmetzafrir_laptop: astdb is quite simple. A dirty way is to strings it
12:13.29tzafrir_laptopIt claims that the file is and "Invalid Argument"
12:13.56tzafrir_laptopditto db3_dump185
12:14.05{zombie}works for me
12:14.18{zombie}but I compiled asterisk myself, not using the deb package
12:15.07{zombie}you could also do: asterisk -rx 'database show'
12:16.15tzafrir_laptop{zombie}, but can I restore from that?
12:17.03tzafrir_laptopI'm not sure I can count on values there not to contain all sorts of binary values
12:17.08{zombie}sure, you could turn that into a bunch of "database put" commands
12:17.22{zombie}binary values? from what?
12:17.26{zombie}what are you putting into the database?
12:18.03tzafrir_laptopor nice chars like '"'
12:19.23{zombie}bit of shell/perl/python scripting should sort the restore for you anyway
12:20.56*** join/#asterisk ManxPower (n=eric@adsl-67-65-233-194.dsl.lgvwtx.swbell.net)
12:21.12ManxPowerSomeday I'll have reliable internet access.  Someday.
12:21.27santoshrwould inband not registering cuz asterisk to not accept the *2 which was tht call pickup
12:21.28AhrimanesManxPower: you're rather optimistic?
12:21.32mutyep, fios everywhere
12:21.34mutthen maybe
12:22.06SkramXIs it possible to make something simultaneously ring two sip addresses?
12:22.19AhrimanesSkramX: queues can
12:24.35*** join/#asterisk Frosted (n=Procrast@buzzbud.plus.com)
12:24.39ManxPowerSkramX: you mean like Dial(SIP/phone1&SIP/phone2)
12:24.51ManxPowerAhrimanes: no.  I'm stubborn.
12:24.59AhrimanesManxPower: ah
12:25.11ManxPowermut: I'm looking at moving to a place 11 miles from the CO
12:25.26AhrimanesCO ?
12:25.44Dr_Raycentral orrifice
12:25.46FrostedAnyone know what the port field in realtime has changed to?
12:26.29santoshrhow to setup call pickup
12:26.37*** join/#asterisk zotz (n=zotz@24.231.47.168)
12:26.45ManxPowersantoshr: you need to make sure your DTMF is working first.
12:26.52santoshrexactly.
12:27.32santoshrin the default there is a demo instruct ..which should recognize the basic extensions whcih come as samples.
12:27.40santoshri mean 500 and 1234 and stuff.
12:27.51ManxPowersantoshr: what codec do your devices use?
12:28.01santoshrmu law and a law
12:28.35ManxPowersantoshr: do you have a reason for using both?
12:28.45santoshr1.0.x works i mean asterisk acks tht some digits were dialed .. i checked this twice.. but asterus 1.0.x wont
12:28.48mutthats alright manx
12:28.49SkramXManxPower: Tes.
12:28.50ManxPowerAsterisk has (maybe it's fixed) had issues with transcoding bterrn alaw and ulaw.
12:28.51SkramX:)
12:28.52SkramXWell
12:28.54mutjust lay your own fiber
12:28.56santoshrits the priority set inb the bozx
12:29.00mutonly cost 30k
12:29.41santoshrthe priority is ulaw and alaw in the box. it would try ulaw first
12:29.47SkramXI guess the best way would be to register the voip carrier, and have all calls immediately do Dial(SIP/phone1&SIP/phone2) to the sip phones..
12:30.04SkramXWould that work?
12:30.28*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
12:30.57santoshrManxPower: but. see a dials b , and when c sends a *8 it comes up in the asterisk konsole but it doesnt find the extension and goes into default
12:30.58puzzledhi
12:31.02puzzledw00t 1.2.0. congrats all!
12:31.30santoshrso i  feel its something wron tht i have doen in the h323.conf or extensions.conf for this behavious
12:31.31ManxPowerpuzzled: Yeah, but it mostly includes Cool Stuff, not the stuff I need.
12:32.16ManxPowerI need jitterbuffers for SIP
12:32.52santoshrManxPower: any suggestions
12:33.19ManxPowersantoshr: not with H323
12:33.28santoshrwht r the things tht i need to set for the following
12:33.40santoshra ---- > b and c wants to pick up
12:33.55ManxPowersantoshr: on my systems I set pickupgroup= and callpickup= in zapata.conf
12:34.04ManxPowerBut that, of course, won't work for h323
12:34.08santoshrin the features.conf pickupexten = *8
12:34.34santoshrwhere does this go pickupgroup
12:34.45tzafrir_laptopsantoshr, which h323 channel do you use? chan_h323? I'm quite surprised to learn it works for current 1.0
12:34.55ManxPowertzafrir_laptop: I do NOT use H323
12:35.08santoshrcuurenyl it is working only with 00h323
12:35.18ManxPowersantoshr: what version of Asterisk?
12:35.24santoshr1.2.0
12:35.45tzafrir_laptopsantoshr, oh323 or ooh323c ?
12:35.57santoshrooh323c
12:36.17testmachinehey, my voicemail is working now
12:36.18tzafrir_laptop(which did not exist in 1.0)
12:36.19*** join/#asterisk Tili (n=Tili@213.132.60.182)
12:36.33testmachinei only have the problem that it wont come into my webbased voicemailbox
12:36.46santoshrya 1.0 had chan_h323 in itselg onyl this i had to download and put
12:36.49testmachineit is coming into /var/spool/asterisk/voicemail/*/*
12:36.57testmachinebut i cant see it webbased
12:37.06testmachinein the vmail.cgi script
12:37.23ManxPowertestmachine: Well.....the web server userid must be able to see those files.
12:37.34testmachineah
12:37.36testmachineofcourse
12:37.38testmachinehehe
12:37.44puzzledManxPower: yup I could use that too. but afaik it's not included cause zoa & friends decided to go another route with the jb so there wasn't one ready on time to be included
12:37.49santoshrthe features.conf has this context => parkedcalls ,  where is this sipposed to be included in the extensions .. in the context which contains a and b and c right
12:37.58testmachineManxPower: so it are permissions
12:37.58testmachine:)
12:38.56santoshrManxPower:   the features.conf has this context => parkedcalls ,  where is this sipposed to be included in the extensions .. in the context which contains a and b and c right
12:39.38ManxPowersantoshr: call pickup is NOT call parking
12:41.12santoshrok. i understnad tht. but. features.conf in the general context thesre are the thingscontext => parkedcalls and also there is   pickupexten = *8
12:42.07ManxPowersantoshr: I cannot help you.
12:42.35santoshrwhy
12:43.01santoshri am slightly new to this. i understnad i dont know alot of stuff but..
12:43.01ManxPowersantoshr: because I do not have time to teach you about Asterisk.
12:43.05tzangermorning
12:43.26santoshrcan u atleast guide me ons setting up call pickup
12:43.32puzzledManxPower: and the latest sip jb no longer applies to 1.2.0(-rc2)
12:43.39tzangersantoshr: I'll put $5 on ManxPower showing you the way to the asterisk handbook, the asterisk physical published book, the wiki and the asteriskdocs website...
12:43.59ManxPowersantoshr: perhaps you did not understand me.  I cannot help you with setting up call pickup with H323 because I do not use H423.
12:44.03puzzledI'll throw in some stroopwafels
12:44.12tzangerdamn you and your stroopwafels
12:44.17puzzledhehe
12:44.25tzangerI'm already buying my girl those super expensive "real chcolate" truffels
12:44.44tzangerpuzzled: I want a christmas card with stroopwafels in it from you
12:45.24puzzledI should start a stroopwafel export business. goodbye asterisk, hello stroopwafels
12:45.38tzangerhehe and hello 350lbs
12:45.43SkramXSo I have someone asking me for the following: They have 2 ATA's and want to have their voip service ring on both ATA's. Would the best way to do this be: Register both ATA's to the asterisk box (via SIP) and then register the voip carrier on the asterisk box. And immediately when someone calls the voip #, the asterisk box does something like "Dial(SIP/ata1&SIP/ata2)"
12:46.00puzzledsounds good to me
12:46.06lmesantoshr:  are you trying to setup that any extension call pickup any other with asterisk ? via a feature ?
12:46.08ManxPowerUgh.  I'm getting tired of the depressing post-Katrina news stories.
12:46.31*** join/#asterisk chapeaurouge (n=chap@85.201.80.249)
12:46.34testmachinepuzzled: lol stroopwafels?
12:46.36testmachinehehe
12:46.38testmachinemafkees
12:46.44puzzled:)
12:46.44tzangerSkramX: yes that's definitely a way to do it if the VOIP provider doesn't support multiple UA registrations (which I am 99% sure it won't)
12:47.05tzangerwhat's a mafkee?
12:47.19SkramXtzanger: they do not.
12:47.22santoshrlme: yes, but does it have to be thru a feature or how ?
12:47.31SkramXThe user is using Broadvoice.. but on a BYOD plan
12:47.35tzangerSkramX: then use SER or asterisk, yes.
12:47.41puzzledtzanger: nutcase
12:47.43SkramXI just dont remember how to get the ata's to register with asterisk
12:47.44tzangerif SER will let you register to a VOIP provider, which I don't think it can
12:47.49SkramXbut i am sure i can figutr it out
12:47.50lmesantoshr: in fact, asterisk can't do that... it does only support groupcall pick up
12:48.14lmeso you have to set each phone on a pickupgroup
12:48.27puzzledanyone happen to have a variation of allpage.agi that allows you to page a single phone?
12:48.28lmegroups are limited to 65 i think
12:49.18ManxPowerpuzzled: I don't know of one, but paging to a single phone is really "call phone and make it autoanswer" and that's pretty easy with some phones.
12:49.48santoshrok..so if a exten is in the pickup group and some does  a *8 which is present in features.conf is should pick up coorect
12:49.50puzzledManxPower: yup have seen some stuff about Polycom phones
12:50.09ManxPowerpuzzled: I hope to implement some paging once I get moved.
12:50.13SkramXI am confused with sip, would I make both ATA's friends or peers of the asterisk box?
12:50.27puzzledsantoshr: I have a 1.2.0-rc2 box here and can do *8 without any callgroup/pickupgroup defined
12:50.51ManxPowerSkramX: for many reasons SIP endpoints are usually type=friend, where servers are usually type=user and type=peer
12:51.17lmesantoshr: correct... but you can only pickup phone's of your group
12:51.27SkramX[sip.broadvoice.com]
12:51.28SkramXtype = peer
12:51.31santoshrcoorec tht is not an issue ..
12:51.35SkramXSo i guess peer for this customer :)
12:51.38SkramXThanks, ManxPower
12:51.52tzangerpuzzled: probably because everything's in a default pickupgroup
12:52.45santoshrbut how do i tell asterisk to give me tht call.. i mean wht i did was in hte context in features.conf i put the context = mycontext which has the exten's in them in the extensions.cong
12:52.54santoshris this correct or stupid
12:53.01lmepuzzled: yes... but you pickup the first ringing phone... not the one you want
12:53.38puzzledtzanger: guess so. need to check the config
12:53.57puzzledlme: you could use Pickup(<exten>) for that
12:54.30lmepuzzled: it's only work for a group number or am i wrong ?!
12:54.42puzzledlme: dunno. try it
12:54.48santoshrno no people.. lme and puzzled..
12:55.01*** part/#asterisk Geert (i=geert@irssi/staff/geert)
12:55.02lmeokay... already tried :)
12:55.04santoshrwht am i doing wrong
12:55.35ManxPowersantoshr: maybe your VoIP endpoint is not allowing you to dial *8
12:56.17santoshrno it does .. *8 which i can c in the asterisk konsole
12:56.30santoshrbut it goes to default as it does not find the exten
12:56.53ManxPowersantoshr: You have a phone ringing when you dial *8?
12:57.11lmesantoshr: with that, yo canno't specify groupeid you want
12:57.48lmeouch... weird behaviour wiuth chan_sccp and astdb corrupted...
12:57.59santoshryes ManxPower
12:58.08*** join/#asterisk X-Files (i=x-files@x-files.lv)
12:58.52lmedamn frenchies & damn alcatel !!!
12:59.01Zeeekalcatel sucks!
12:59.28santoshrpeople were tellin me tht exten who wants to pick should be in hte same group.. how do i do tht .. i mean . yes they are in the same context in the extensions.conf
12:59.39Rowterthere is any way to detect when a fax machine answeres an outgoing call. NV_FaxDetect  and the zaptel fax detect seem to only work in calls originated FROM a fax machine, not for calls ANSWERED by a fax.
12:59.51lmesantoshr: you have to specify that for each phone/line
13:00.06santoshrwhere ?
13:00.11lmesantoshr: but if u'r using h323, i can't help you... Only knowing about sccp & sip
13:00.50lmeZeeek: jealous
13:01.05lmeforgot to say : damn boss
13:01.54santoshrhow did u implemetn it with sip
13:02.44lmesantoshr: you can specify two things : callgroup & pickupgroup
13:03.16lmesantoshr: you have to do this for each peer... pickupgroup is the groups you can pickup.... and callgroup is the group you're member of
13:03.25santoshrshouldi put in ip address of the extensions or name of the guys in h323.conf
13:04.43lmedon't know about h323
13:05.06*** join/#asterisk tdonahue (n=tdonahue@208.51.101.201)
13:05.53santoshrlme:  wht did u  do in sip.conf ?
13:06.00Zeeekalcatel sucks!
13:06.01lme!!!
13:06.23santoshri mean in the pickupgroup wht did u put.. peer names..  ?
13:06.27lmeZeeek: oh come on... don't be so jealous... It's the only pbx which can make you cofee while calling...
13:06.48Zeeekwhat pbx, I'm talking about the phone on my desk! Alcatel phones suck!
13:07.11Zeeekevey chinese phone I own has a better speaker than the $50 alcatel
13:07.13lmea yeah... it's a possibility
13:07.39Zeeekonly one thing is psoitive about it, it needs no AC power :)
13:07.40lmesantoshr: in the peer section... so -> below the peername
13:08.29lmeZeeek: not the only one... If you press severals time on the hangup key, you can transform it in chrismas tree
13:09.15*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)
13:09.18ZeeekI don't need a christmas tree at the moment
13:09.34ZeeekI'm not sure that *8 works between different technologies, does it ?
13:10.14lmein fact i never use features.conf... all stuff are in extensions in my configuration...
13:10.57santoshrlme: u would really not put up ur extensions.conf
13:11.20lmeouch
13:12.01santoshri know it does not sound right.. but i am not well versed wit the syntax..
13:12.05lmeit will take me many time to mask things i do not want to be public...
13:13.02santoshrok.
13:13.06lmesantoshr: so you'll have to work on it ! extensions.conf is about 80% of my work with asterisk...
13:13.52lmeI consider it as the most important file...
13:14.01santoshri totally understan .. and it indeed is.
13:14.23santoshrbut can u help me with the pickupgroup syntax..
13:14.35Zeeekas far as I can see *8 does NOT work with IAX
13:15.35*** join/#asterisk Broom (i=Broom@jescobar.ayustar.net)
13:15.48lmethe 2nd line of the original features.conf make me thinks that it should not work with anythings but callparks
13:15.49santoshrhas any one ever in theier entire lives has heard of it workin with h323  ,,, if yes plese tell me .. there are people in grief here. ;)
13:16.15lmeI hate h323
13:16.29lmerecallin' me alcatel :)
13:16.35Zeeekalcatel sucks
13:16.40lmeah ah :)
13:16.44Broomhello all, I am having a nasty crackling noise on calls coming trough a TE110P, any ideas?
13:16.54lmeBroom: again ?!
13:16.57Zeeek<PROTECTED>
13:17.00Broomlme: yes man
13:17.08Zeeekcontext does not exist
13:17.35lmealcatel ? (or is this a bot) ?!
13:17.54Broomlme: if i have a box running how can I upgrade to 1.2.0 without loosing configs? (Other question)
13:18.10lmeBroom: just not make samples
13:18.17santoshrlme: can u tell me the pickupgroup in sip.conf
13:18.22Broomk
13:18.34lmei think i swa an upgrade.txt or something like that bundled with 1.2
13:18.47lmesantoshr: but... i already told you....
13:18.56Zeeekguys if you want to test *8 just put every phone in pickupgroup=1
13:19.42lme(14:02:44) lme: santoshr: you can specify two things : callgroup & pickupgroup
13:19.42lme(14:03:16) lme: santoshr: you have to do this for each peer... pickupgroup is the groups you can pickup.... and callgroup is the group you're member of
13:20.35lmeBroom: have you got crackling when you make internals calls ?
13:21.21Broomlme: no
13:21.53*** part/#asterisk ManxPower (n=eric@adsl-67-65-233-194.dsl.lgvwtx.swbell.net)
13:22.26*** join/#asterisk ManxPower (n=ewieling@adsl-67-65-233-194.dsl.lgvwtx.swbell.net)
13:22.51ikarusWhat is the sensible solution to make a group of phones ring on a single extension, and keep it simple te administrate (IMPORTANT, I won't be around forever to maintain this)
13:23.50ManxPowerikarus, Depends.  Usually I use Dial(SIP/phone1&SIP/phone2) but if you need a more complex way use queues
13:25.33X-FilesManxPower: hello, where i can found full howto for compile oh323 protocol for asterisk
13:25.38X-Files?
13:26.05*** join/#asterisk RoyK (n=roy@80.239.107.70)
13:26.11santoshrlme: can u please look at this ..       http://sial.org/pbot/14336.
13:26.18*** join/#asterisk hadi57 (i=al_moghr@62.3.44.61)
13:27.54lmesantoshr: you need to add them to the callgroup 1 too
13:28.48lmeif i consider only your h323.conf file, everybody as the right to pickup group 1
13:28.49ikarusManxPower: hmmm, that looks horrid, I wanted to split it so that I direct an extension to a group (say reception) and in that group I configure a list of phones or something to ring
13:29.01lmebut there isn't anybody in it
13:29.07ManxPowerikarus, That would be call queues
13:29.44ikarusManxPower: hmmmmm, k
13:30.08ManxPowerikarus, Asterisk doesn't ring extensions.  Asterisk rings DEVICES.  A device is not an extension.
13:30.11ikarusManxPower: and I can still do a normal dial plan with that (so first try a queue, if no one answers, or all busy fall through to voicemail)
13:30.14ikarustrue
13:30.19ikarussorry :)
13:30.21ikarusI knew that
13:30.57ikarusand they still think traditional telecom
13:31.07lmeahhh
13:31.11lmegot the same problem
13:31.22muti got the solution!
13:31.31ikarusManxPower: what config files and commands are related to queues ?
13:31.31mutquit your job
13:31.35lmethey want another phones and another pbx, but the same things that before
13:31.43Ahrimanesikarus: queues.conf ?
13:31.52ManxPowerikarus, queues.conf  See the Wiki.  Don't use Agents unless you REALLY need to.
13:32.00Ahrimanesikarus: http://www.voip-info.org/wiki/view/Asterisk+call+queues
13:32.12stbainthe all-knowing... never sleeping ManxPower
13:32.39ManxPowerstbain, I'm waiting for the upgrades of a couple of my servers to 1.2RC2 to finish.
13:32.56stbainwhy RC2?
13:33.36*** join/#asterisk Morpheus68 (n=chatzill@a20030.upc-a.chello.nl)
13:33.41Morpheus68hi all
13:33.58ManxPowerstbain, because I leave for a long weekend this evening, we need to upgrade the servers because of issues with RC1 and I am NOT going to reapply all our custom patches to 1.2.0
13:34.00Morpheus68i'm trying to get into sip again
13:34.10ManxPowerAlso, only a moron would upgrade to a .0 release of anything.
13:34.32Morpheus68i'm running asterisk with 3 ISDN cards in AT and NT mode
13:34.32stbainI'll be installing 1.2.0 here in a new installation soon
13:34.45Morpheus68works fine
13:34.45*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-242-4.buckeyecom.net)
13:34.49stbainare you using an internal CVS to manage your patches against the core code?
13:34.59ManxPowerI KNOW RC2 works, since I've been testing it for 3 days.  I don't know if 1.2.0 will work for us and if it doesn't my customers are going to want to lynch me when I'm not available this weekend
13:35.06ManxPowerstbain, gads no.
13:35.14Morpheus68now i am trying to configure sipgate.de account and this fails with:
13:35.25Morpheus68handle_response: Forbidden - wrong password on authentication for REGISTER
13:35.41ManxPowerResults 1 - 10 of about 223 from lists.digium.com for  sipgate.de. (0.28 seconds)
13:36.07Morpheus68when i use the same info in X-Lite software ip phone i manage to connect and to make phone calls
13:36.16stbainYMMV... we use an internal CVS (in the process of switching to SVN) to keep track of our internal changes to phpBB and it makes life much, much easier
13:36.18Morpheus68anyone an idee?
13:36.32puzzledManxPower: agree with the point release stuff but in what way would 1.2.0 be less than 1.2.0-rc1/rc2?
13:36.55ManxPowerpuzzled, Excelent question.  I don't want to find out when I'm on holiday.
13:37.38ikarusManxPower: erm, what do you mean with don't use Agents unless you have to (as far as I can see you need agents to have a queue
13:37.40puzzledManxPower: couldn't agree more :)
13:38.10ikarusoh, wait, no
13:38.12ikarusnow I see
13:38.15ManxPowerikarus, you do not need agents.  Agents are people that are sometimes in the queue and sometimes not in the queue.  member= are devices that are always in the queue.
13:38.23ikarusyou can also have member => SIP/phone1
13:38.29ikarusetc
13:38.49ManxPowerpuzzled, I'm already breaking my policy of "never make any changes less than 2 weeks before I go on holiday" rule.
13:38.58ManxPowerikarus, yes
13:39.17puzzledManxPower: yeah that's pushing it a bit
13:39.48ManxPowerbut forwarding a voicemail with a prepended attachment doesn't work in RC1
13:40.01ManxPowerAlso there are zaptel flash transfer DTMF issues with 1.2RC1
13:40.41ManxPowery'all know that the initial release of 1.2.0 was totally broken, right?  They replaced it with a working tar.gz
13:41.02puzzledyes the chan_modem stuff was not commented out
13:41.02santoshrlme:  "lme: but there isn't anybody in it" in the pickupgroup..  can u please exemplify ?
13:41.07ManxPowerYay!  I'm down to only 3 trouble tickets assigned to me.
13:41.09santoshrsory for buugin u so much
13:41.26gambolputtytime to recompile I guess then
13:42.04ManxPoweralso RC1 has an issue with DACSRBS
13:42.32ManxPowerWe use DACSRBS on many of our servers
13:45.00ManxPowerI would love to know what this means "Nov 17 07:44:41 WARNING[1013]: chan_sip.c:11248 do_monitor: chan_sip: ast_sched_runq ran 23 all at once
13:45.00ManxPower"
13:46.29tzangerManxPower: yes I am seeing that too
13:46.36tzangerI think it has something to do with the iax2 jb
13:46.39tzangerI haven't turned it off ot se though
13:46.41tzangerer to see
13:47.00ManxPowertzanger, I'm not going to complain because people will just say "WARNING != ERROR" and then ignore me.
13:47.40tzangerManxPower: well I'm going to complain :-)
13:47.45tzangerit's something that needs investigating
13:47.58tzangerWARNINGS shouldn't be ignored, they indicate a potential problem
13:48.03tzangerERROR indicates a problem
13:49.25*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
13:50.31FrostedI just got a WARNING that realtime couldn't finish a query :P
13:51.05*** join/#asterisk lehel (n=asd@82.79.20.17)
13:51.07lehelhello
13:54.05*** join/#asterisk tracinet (n=tracinet@24-50-29-205.atlsfl.adelphia.net)
13:54.55gambolputty<PROTECTED>
13:55.01gambolputtyis yours similar to this?
13:55.08X-Filesneed help ! why i can't compile oh323 ?? http://pastebin.ca/29009
13:56.23Broomlme: anything on the crackling noise? what is weird is that it is on some calls only not all of them
13:58.08*** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net)
14:00.43tracinetjust read on the users list that someone reported an issue using iax2 jitterbuffer between 1.2 and 1.0.x machines - is this a known issue?  I tried searching mantis and could not find anything - I am planning on upgrading my gateway asterisk servers to 1.2.0, but will have a few 1.0.x asterisk servers running behind it for some time until I can migrate them all over to 1.2 - should i expect issues if I am using iax jitterbuffer betwe
14:01.22MikeJ[Laptop]tracinet, are they 1.0.9 or older 1.0.x?
14:01.29ikarusHmmmmm, can ringall queues work together with not actually picking the incoming line (which is ISDN) until one picks up, or do I have to use Dial for that ?
14:01.54tracinetAsterisk CVS-v1-0-11/12/04
14:01.59MikeJ[Laptop]there are issues with the iax timestamps sent from older 1.0.x servers that will break many things, including the 1.2 iax jitterbuffer
14:02.28tracinetgotcha - and i assume my version (Asterisk CVS-v1-0-11/12/04) is one of those older ones
14:02.31MikeJ[Laptop]tracinet, yes, you will have an issue between 1.2 and those
14:02.37tracinetthanks - good to know
14:02.46MikeJ[Laptop]the issue is, older ones send bad timestamps
14:02.52MikeJ[Laptop]newer 1.0.x will not
14:03.21MikeJ[Laptop]you really want to get those old ones all up to 1.0.9 before you do anything else...
14:03.38tracinetthat makes sense - at least my old configs will still work
14:03.51MikeJ[Laptop]for the most part, yes, they should
14:04.28tracinetthanks for your input (i wonder if those bad timestamps are why i occassionally get audio garble between my old servers)
14:05.07MikeJ[Laptop]well.. that, and the several 1000's of issues that have been fixed in the last 12 months, maybe the fix was in there :)
14:05.14tracinetLOL
14:05.57ManxPowertzanger, have you seen excessive restarting of the bchannels with 1.2 port RC1?
14:06.01ManxPowerport = post
14:06.53ManxPower1.2 is going to force me to have my own release of asterisk - one that includes sip JB and zaptel timing for RTP audio, and the ISX2 JB patch
14:07.09ManxPowerAll the stuff that REALLY should have been in 1.2
14:07.46tzangerManxPower: nope
14:07.56tzangerI set my forced restart time to something like 6 hours
14:07.57ikarusAnyone on the queues + no pickup stuff, or will I just have to accept I'll have to use Dial, or bill people :)
14:08.17tzangerManxPower: IAX2 JB patch?
14:08.19tzangerwhich patch?
14:08.47*** join/#asterisk fri (n=fri@port84.ds1-sdb.adsl.cybercity.dk)
14:08.51ManxPowerikarus, I would assume that if you don't issue an answer and don't play audio to the caller it won't pickup.  Remember many carriers will terminate a call if it's not answered after a while.
14:09.18ManxPowertzanger, see bug 5197
14:10.07ikarusManxPower: I know, but this is just to replace the functionality of a current PBX (which is braindamaged)
14:10.12tzangerManxPower: yes I knwo that one
14:10.32tzangerI have not had IAX2 stop answering calls for about 2 weeks now
14:10.54ManxPowertzanger, you turned off the new IAX2 JB?
14:11.22tzangerno
14:11.41tzangerI am running a modified version of the -ve timestamp detection patch though
14:11.49tzangerthat doesn't allow it to go -ve
14:11.54RoyKManxPower: you know, i'm paying zoa to write a new channel-based generic JB for */opbx
14:11.56ManxPowertzanger, one might hope if they put the fix into CVS there would be a noteof it on the bug report
14:12.09tzangerit isn't a confirmed fix
14:12.27ManxPowerah
14:13.45X-Filesin asterisk 1.2.0 included pickup() ?
14:18.04xbmodder_lappyis this valid in the dial pan:
14:18.05xbmodder_lappyexten => _XXXXXXXXXX,n,AGI(meetme.py)
14:18.05xbmodder_lappyexten => 8686,1,Meetme(666)
14:18.32tzangerxbmodder_lappy: sure it is, but I have no idea what you're trying to do there
14:18.38xbmodder_lappythe AGI application takes in some input and if the input is correct it runs "SET EXTENSION 8686"
14:18.55xbmodder_lappyin agi debug:
14:18.57xbmodder_lappyAGI Rx << SET EXTENSION 8686
14:18.57xbmodder_lappyAGI Tx >> 200 result=0
14:19.03xbmodder_lappyand the whole damn thing dies
14:19.18tzanger'n' means 'next' -- I don't know if it is intelligent enough to start at '1' if not specifically told to
14:19.30tzangerand 8686 does not match _XXXXXXXXXX
14:19.38tzangerso if 8686 is your '1'
14:19.44tzanger_XXXXXXXXXX will not follow next
14:19.45xbmodder_lappytzanger, but "SET EXTENSION 8686"
14:19.54tzangerxbmodder_lappy: stop
14:19.54*** join/#asterisk Guggemand (i=Guggeman@80.198.131.42)
14:19.57tzangerwhat is it you're trying to do
14:20.04xbmodder_lappyme 1 is: exten => _XXXXXXXXXX,1,Answer
14:20.06cpatryu wanna join ur meetme in ur agi right?
14:20.07ManxPowerxbmodder, What do you expect "SET EXTENSION 8686" to do?
14:20.14xbmodder_lappytzanger, cpatry yeah
14:20.23xbmodder_lappyManxPower, join the meetme
14:20.25cpatryjust dial to ur exten
14:20.37cpatrywhen meetme will finish, u will continue where u were before.
14:20.41ManxPowerxbmodder, Try using a goto with extension and priority.
14:20.42cpatry(in ur agi)
14:21.23xbmodder_lappyManxPower, i don't see any "goto" in AGI
14:21.35cpatryexec goto?
14:21.39ManxPowerxbmodder, exec the goto app
14:21.45xbmodder_lappyoh, ok
14:21.56ManxPoweror why not have a goto right after the AGI?
14:22.11xbmodder_lappySo I should output "EXEC GOTO(24)" if i want the meetme
14:22.25ManxPowerxbmodder, you have a priority 24?
14:22.31*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
14:22.44xbmodder_lappyManxPower, I can make one
14:22.49cpatryjust dial to ur meetme instead.
14:23.03xbmodder_lappyManxPower, or would it be better to "EXEC Meetme(666)"
14:23.08cpatrywill be much simpler + allow to continue in ur agi.
14:23.12ManxPowerxbmodder, seems like you are making things more complicated than it needs to be.
14:23.13xbmodder_lappycpatry, oh.
14:23.36xbmodder_lappyManxPower, sorry i am a little new to this all.
14:24.15lmebump
14:24.52Kattylump.
14:25.09Kattywho authorized this cold front moving through the states?
14:25.13Kattyi'm going to have someone's head over this.
14:25.23Kattyor at least complain.
14:25.26cpatrykatty: i told you, snow were here 2 days ago :)
14:25.37Kattycpatry: well that's there.
14:25.50Kattycpatry: i should move to flordia.
14:25.58jontowand get blown away?
14:26.06Kattybetter than hurting all day long
14:26.11jontowi suppose so, yes
14:26.19Kattyi mean /all winter/
14:26.25jontow37F here when i left for morning
14:26.31jontowand thats the high for the day
14:26.32jontowmeh
14:27.00ManxPowerKatty, move to the west coast
14:27.02xbmodder_lappyManxPower, now my last question where do I declare these extensions?
14:27.11KattyManxPower: you've got a solution for everything, don't you?
14:27.20ManxPowerxbmodder, I have no idea what you mean.  You define extensions in....extensions.conf
14:27.33ManxPowerKatty, The solution to MOST things is "move to the west coast"
14:28.47xbmodder_lappyManxPower, but what section?
14:29.08ManxPowerxbmodder, That is up to your design of the dialplan.
14:29.33KattyManxPower: unless you're a female
14:29.37jontowwtf, man.. boss claims that me monitoring the softswitch using the builtin log-parsing utility is causing too much load -- i login once every 15 minutes and run the command once
14:29.40KattyManxPower: and then you're not supposed to offer solutions.
14:29.42ManxPowerI add all MY extensions in a [extensions] context, I define all my incoming calls from outside the system in [incoming]
14:29.44Broomwell, i'm upgrading to 1.2 lets see how this goes
14:29.53xbmodder_lappy*CLI> Nov 17 06:29:34 WARNING[23014]: pbx.c:2332 __ast_pbx_run: Channel 'IAX2/NuFone-2' sent into invalid extension '8686' in context 'nufone-IAX', but no invalid handler
14:29.59jontowand checking the load averages while its runnign.. it shows that the load isn't gaining at all -- its DECREASING :(
14:30.10xbmodder_lappyexten => 8686,1,Meetme(666)
14:30.15xbmodder_lappywtf.
14:30.40ManxPowerxbmodder, You do not have an extension 8686 in the nufone-IAX context (which is where you told asterisk to send those calls by setting it in iax.conf
14:31.03jontowxbmodder; its saying you don't have the 'i' extension in 'nufone-IAX', and also .. put that extension (8686) in the correct context as well :D
14:31.29ManxPowerxbmodder, learn to walk before you try to run, grasshopper.
14:31.39docelmoYIPPIE!
14:31.39*** join/#asterisk coppice (n=chatzill@40.199.17.210.dyn.pacific.net.hk)
14:32.36xbmodder_lappyManxPower, now everything works. except ones it changes the extension it crashes
14:32.39*** join/#asterisk christo (n=chris@brezhnev.spiration.co.uk)
14:32.45christohi all
14:32.58docelmohola..
14:33.13xbmodder_lappynot crashes to say, but hangs up which is equally bad
14:36.27Morpheus68hi all
14:36.45Morpheus68anyone has sipgate working on asterisk?
14:39.48ManxPowerxbmodder, you must ALWAYS have a priority when doing a goto
14:40.07ManxPowerMorpheus68, none of the 200 or so messages on the mailinglist archive were helpful?
14:40.43Morpheus68no
14:41.07Morpheus68:(
14:41.52*** join/#asterisk aRJAy (n=aRJAy@adsl-130-112.swiftdsl.com.au)
14:41.53Morpheus68i get: handle_response: Failed to authenticate on REGISTER to
14:42.51[TK]D-FenderBad user/pass <-
14:43.07RoyKbad user! BAD BAD user
14:43.11Morpheus68and then <sip:sipgatenr@sipgate.de>;tag=as477b62d6
14:43.28Morpheus68no because when i use X-lite it is working
14:43.33[TK]D-Fenderbut file does NOT fall down!
14:43.47file[laptop]Morpheus68: you do realize you were supposed to put your number in? I assume sipgatenr is literal...
14:43.51*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
14:43.55Ariel_Morning all.
14:43.56Morpheus68no
14:43.58KattyAriel_: :>
14:44.07Ariel_Nice to see the 1.2 has been released.
14:44.12Morpheus68is't normaly my number bud i am not going to post it ;)
14:44.22KattyAriel_: is it freezing cold there too?
14:44.25ikarusHmmmm, if an extension is BUSY then asterisk should fallthrough to the next priority in the dial plan, right ?
14:44.45Kattyikarus: whatever the next number is.
14:44.47Ariel_Katty, really. Sorry to hear it. it's 82 outside today here.
14:44.55jontowariel :(
14:44.56KattyAriel_: jealous :<
14:45.00ManxPowerAnyone else think this sounds weird?  http://pastebin.ca/29012  No changes to the zap config
14:45.05Kattywe should all go to Ariel_'s house today
14:45.09ikarusKatty: hrm, that is not happening
14:45.16Kattyikarus: sads.
14:45.20Ariel_great you can help me move then.
14:45.25KattyAriel_: ok!
14:45.30KattyAriel_: you can help me move later (=
14:45.33ikarusKatty: it just hits timeout
14:45.40Kattyikarus: sad.
14:45.44Ariel_sure if I get up there.
14:45.54jontowikarus; with or without autofallthrough enabled?
14:47.27[TK]D-Fenderautofallthrough... *shudder*  I still don't see the point...
14:47.27jontow;)
14:47.27[TK]D-Fenderwe have "hangup", USE IT...
14:47.27jontowautoBREAKSHIT
14:47.28jontow:)
14:47.28Ariel_ManxPower, wow I have not seen them before are you using 1.2 or the 1.0.9
14:47.28Morpheus68by the way outgoing calls are working only incomming calls not!
14:47.28ManxPowerAriel_, 1.2RC2
14:47.28[TK]D-FenderI helped a guy who had it on and his IRV DIES screaming...
14:47.28[TK]D-FenderIVR*
14:47.28Morpheus68same user/pass
14:47.30ManxPowerAriel_, The odd thing is the channel numbers are wrong when unloading chan_zap
14:48.15ManxPower<PROTECTED>
14:48.36X-FilesManxPower: ?
14:49.03*** join/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net)
14:49.05ikarusjontow: happens in both cases
14:50.31*** join/#asterisk cnet2 (n=jjohn@adslnat-sanjose-4.ice.co.cr)
14:51.41ikarushmmm, me not typoing might help, let's see
14:52.44*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal)
14:54.44*** join/#asterisk db48x (n=db48x@dsl093-026-247.hou1.dsl.speakeasy.net)
14:55.23xbmodder_lappyanyone here work for teliax?
14:55.53*** join/#asterisk tamp4x (n=kkdkdkkk@204.124.238.248)
14:55.54db48xdoes anyone have any open source softphone software they like to recommend, for linux and windows?
14:56.07tamp4xhow can i prevent static echo when a sip phone calls an analog phone?
14:56.08xbmodder_lappygnome-meeting for linux
14:56.54jontowx-lite runs on windows and linux
14:56.59testmachinedb48x: linux kphone too
14:57.00testmachine:)
14:57.11cnet2there's cisco router called Catalyst Express with QoS for VoIP only..  Does all/most of voip phones and voip software(like asterisk) mark the packets so they can be identified?
14:57.19db48xkphone is one of the ones I've tried already - it's problem was that the sound dropped out after 5 or 6 seconds
14:57.33xbmodder_lappy*GRR
14:57.58wasimmoziax rocks
14:57.59xbmodder_lappyjontow, xlite ain't open
14:58.07jontowthats true
14:58.44wasimwe use moziax at over 200 call center seats
14:59.00*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
14:59.00*** join/#asterisk clive- (n=pirch@ndn-165-137-90.telkomadsl.co.za)
14:59.13fishboy1669how come the release of version 1.2.0 hasnt been announced on the website
14:59.16jontowhttp://moziax.mozdev.org/
14:59.35wasimand the developer (jdg) is very responsive as well
14:59.43db48xthat's always a plus
15:00.26*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal)
15:00.59ZeeekHi wasim
15:01.16fishboy1669hi zeek
15:01.18Zeeekbye everyone
15:01.27lmeZeeek: bye alcatelman !
15:01.29Zeeekho fishboy1669sh
15:01.38fishboy1669hows things
15:01.39jontowwonder if moziax works in netbsd under linux emulation
15:01.47Zeeekthat Alcatel phone is the worst piece of crap I've ever seen
15:01.48ManxPowerfishboy1669, it was announced on three Asterisk mailing lists.
15:02.00ManxPowerI guess if you are not on the mailing list you are out of luck
15:02.04*** part/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek)
15:02.06ManxPowerWhich is the way it should be
15:02.26fishboy1669i know but as a marketing ploy i whould have thougth there would have been firework display etc on web site
15:02.41fishboy1669seems a very hush hush affair
15:02.48fishboy1669i saw it on web site
15:02.51lmeah ah ah :)
15:03.10*** join/#asterisk gr0mit (n=w10277@dhcp4.zuk40.mot-tools.co.uk)
15:04.30*** join/#asterisk gniretar (i=mark@152.160.35.157)
15:04.50*** part/#asterisk pif (n=ldm@zenon.apartia.fr)
15:05.16gniretarhi people
15:05.28gniretarcan anyone point me to a "whats new" page on asterisk 1.2?
15:05.31fishboy1669hi gn
15:05.42ManxPowergniretar, Changelog and UPGRADE.txt
15:05.45*** join/#asterisk heath__ (n=root@12-215-32-62.client.mchsi.com)
15:05.55parylwhenever i try to compile 1.2.0, i get "pbx_dundi.c:1409: error: âZ_OKâ undeclared (first use in this function)"... this happens when compiling the source from tar file and cvs both
15:06.07ikarusHmmmmm, I wish I could borrow the standard telco no such number tape here, would make it a little bit more userfriendly (the tones alone are nice, but not so clear)
15:06.07fishboy1669gniretar http://www.asterisk.org/node/19
15:06.09fishboy1669is all i could find
15:06.40*** join/#asterisk froguz (n=froguz@212-43-112.adsl.terra.cl)
15:06.41fishboy1669manx gniretar is asking same questions as me
15:06.51gniretarlol
15:06.53fishboy1669i would expect all this info on the website
15:06.57gniretargreat minds...
15:07.13heath__anyone get the new app_txfax to work with t38? we can get it all compiled, but have yet to successfully transmit
15:07.21fishboy1669maybe its just the opensource business methology that im not used to
15:07.28parylanyone know why that error occurs?  what can i do to fix it?
15:07.31fishboy1669ie have to look in the code to find out whats new ;-)
15:07.37*** join/#asterisk XTR-II (n=xtr@staff-nat.netnation.com)
15:07.42gniretarparyl: your using SpanDSP?
15:07.51parylgniretar: nope
15:08.06gniretarfishboy1669: Nah, usually there are changelogs
15:08.25fishboy1669just seems a bit wird that something that most other businesses would publisise like mad as marketing ploy
15:08.36ikarushrm, a goto has to go to an actual extension, or can it also go to something that matches
15:08.40fishboy1669ie the new version astierisk doesnt seem to bother
15:08.55ManxPowerIt looks like 1.2 resets the B channels once every min
15:09.22parylgniretar: i don't understand... is spanDSP the issue?
15:09.25gniretarlol, well Asterisk is confident in its market share.  It is a good enough product that word of mouth along does its marketing
15:09.39fishboy1669mmmm
15:09.53gniretarparyl: I dont know, when i did the fax system for where I work i ended up using HylaFax
15:09.54fishboy1669as a tecky driven business
15:09.57ManxPowerand by not announcing it on the web site their FTP servers won't become overloaded
15:10.11gniretarAsterisk makes it money from Digium
15:10.11fishboy1669but if it wants to get as big as microsoft etc then it needs to do a bit more
15:10.17parylhopefully no one 'announces' it on slashdot
15:10.19gniretaror from the cards that Digium sells
15:10.37fishboy1669manx surly that is what mirrors are used for ;-)
15:10.46fishboy1669although cvs may struggle
15:10.53ManxPowerfishboy1669, what mirrors/
15:11.12tamp4xhow can i prevent static echo when a sip phone (only heard here when voice is projected)  calls an analog phone?
15:11.26parylgniretar: i don't understand... i don't need a fax system in asterisk... where did that question come from?
15:11.31ManxPowertamp4x, you spend a few months learning about echo, then fix it using the info you have gained.
15:11.45ManxPowerEcho is one of the hardest things to fix in all of telecom
15:11.57parylgniretar: i was talking about an error in compiling
15:12.02gniretarparyl: what are you trying to do?  Did not you mention trying to use txfax?
15:12.07*** join/#asterisk kippi (n=chris@untrust-gct.equinoxit.net)
15:12.08kippihey
15:12.12fishboy1669manx maybe they should look into creating some!
15:12.17*** join/#asterisk wunderkin (i=kev@12-201-105-27.client.mchsi.com)
15:12.17gniretaroh crap
15:12.19gniretarlol
15:12.23gniretari looked at the wrong message
15:12.24fishboy1669if they are worried about too many hits
15:12.29gniretarsoddy paryl
15:12.30parylhaha
15:12.32gniretarsorry*
15:12.32parylnp
15:12.50parylbut again, i must ask, has no one else gotten the error when they try to compile?
15:13.00parylpbx_dundi.c:1409: error: âZ_OKâ undeclared (first use in this function)
15:13.25db48xah
15:14.20kippiI have a TE110P and the light on the back keeps on flashing and the BT box dosn't change, any ideas why this might be happing?
15:14.53froguzi'm sorry for my insistence, i've been looking for an answer 8 hours ago: is there anyway to implement "call hold" with phones without hold key (ie Xlite or no IP phones connected to an ATA)?
15:15.38*** join/#asterisk dos000 (n=dos000@CPE00119572fd49-CM00137186e53a.cpe.net.cable.rogers.com)
15:16.23*** join/#asterisk marc324 (n=marc3234@206-248-133-72.dsl.teksavvy.com)
15:16.38dos000anyone explain why i keep getting "channel.c:1703 ast_set_write_format: Unable to find a path from gsm to g729" all my end points are g729 and the gateway does g729 as well.
15:16.40tamp4xmanxpower i dont think there is a way to tune this type of echo
15:16.46[TK]D-Fenderfroguz : Maybe in features.conf .... but did you check your ATA's manual?
15:16.46fishboy1669manx ftp or cvs which would u recomend?
15:17.10dos000is there a way to figure who decided to use gsm here ?
15:17.13[TK]D-Fendertamp4x : Echo is the bane of digital telephony.  What is your analog interface?
15:17.21marc324ne1 knows if its possible to have the voicemail messages stored in a remote server, instead of local?
15:17.38tamp4xwhat do u mean what is my analog interface
15:17.56tamp4xthe nalog interface is a telephone  fxs
15:17.59tamp4xanalog
15:17.59[TK]D-Fendermarc324 : change *'s voicmail storage folder to one that is mounted remotely.
15:18.13[TK]D-Fendertamp4x : TDM400?
15:18.15tamp4xonly the sip phone hears this static echo
15:18.33tamp4xthe analog phones sit on channel banks
15:19.28tamp4xi use 4 port te411p cars
15:19.36froguz[TK]D-Fender, it's seems my ATA doesn't support Hold, but i'm going to upgrade the firmware on friday to see if i get luck
15:19.49*** join/#asterisk _SMP_ (n=SMP@pandora.burned.net)
15:20.03[TK]D-Fenderfroguz : hmmm, whats that model again?  5xxx somthing or other....
15:20.12ManxPowertamp4x, correct.  You will only hear the echo on the hich latency device.  See echocan= echotraining= rxgain= and txgain=
15:20.42xbmodder_lappywhy do I get this error:
15:20.42xbmodder_lappyhttp://www.voip-info.org/wiki/index.php?page=Asterisk%20AGI
15:20.46xbmodder_lappyack wrong paste
15:20.49xbmodder_lappy*CLI> Nov 17 07:20:27 NOTICE[23428]: chan_iax2.c:6793 socket_read: Rejected connect attempt from 208.139.204.232, request '9252712130@default' does not exist
15:20.52xbmodder_lappythat error
15:21.01froguz5002
15:21.16tzangerxbmodder_lappy: you don't have a 9252712130 matching extension in the [default] context
15:21.19[TK]D-Fenderxbmodder : that exten doesn't exist in that context <-
15:21.27tzangerpretty straightforward error
15:21.32ManxPowerxbmodder, it means that the extension 9252712130 does not exist in context [default] in extensions.conf.  The device that requested it was at IP 208.139.204.232
15:21.33froguzmicronet sp5002 i think, is an fxs ATA with two phone line
15:22.03ManxPowerxbmodder, The context a call hits in extensions.conf is controlled by the [whatever] setting for the device in iax.conf or sip.conf
15:22.17xbmodder_lappytzanger, hmm... shouldn't defuault/demo work always?
15:22.24ManxPowerxbmodder, no.
15:22.24tzangeruh no
15:22.30ManxPowerxbmodder, it's only there if you configure it.
15:22.32*** join/#asterisk toddf (n=toddf@ns0.fries.net)
15:22.48ManxPowerYou're not doing something REALLY STUPID like using the default sample config files, right?  They don't actually work.
15:22.49*** join/#asterisk Mother_ (n=m@53.Red-217-126-93.staticIP.rima-tde.net)
15:22.59xbmodder_lappyok odd part:
15:23.01xbmodder_lappy[teliax]
15:23.01xbmodder_lappycontext=nufone-IAX
15:23.11*** join/#asterisk ful|work (n=fulgas@209.8.233.141)
15:23.25Mother_multiple greetings
15:23.26tzangeris the caller authenticating into that user though?
15:23.41xbmodder_lappy*CLI> Nov 17 07:23:22 NOTICE[23474]: chan_iax2.c:6793 socket_read: Rejected connect attempt from 208.139.204.232, request '9252712130@default' does not exist
15:24.36fishboy1669ftp or cvs which would people recomend for getting the latest version for a production platform
15:24.48*** join/#asterisk mkrufky (n=mk@68.160.103.77)
15:24.51ManxPowerxbmodder, the incoming call is not matching the [teliax] section
15:24.58db48xfishboy1669: since you'll get the same thing either way…
15:25.05ikarusexten => _0Z. should match 0123456789, right ? even if you goto to it (Goto(extern(${EXTEN},1,))
15:25.26fishboy1669db48 i thought that the cvs was always slightly ahead of the ftp
15:25.31ManxPowerNo, it would match 0[2-9]
15:25.34xbmodder_lappyManxPower, umm how do I make it match? I put in the default a config file by teliax
15:25.36ManxPoweractually
15:25.41ManxPowerNo, it would match 0[2-9]anything
15:25.50ManxPowerxbmodder, that is NOT the default config from teliax
15:25.50db48xfishboy1669: if you use cvs to get a release, you won't be checking out from the HEAD, you'll be checking out from some tag
15:26.00ManxPowertwo lines is NOT a configuration
15:26.00fishboy1669cool
15:26.06db48xthat tag identifies the exact versions of all the source files that went into the tarball to go on the ftp site
15:26.27ikarusManxPower: no, according to my doc Z is [1-9]
15:26.37ikarusManxPower: but the problem is more that the . isn't working
15:26.46ikarusit doesn't match anything if you goto appaerently
15:26.52xbmodder_lappyManxPower, the whole thing is:
15:26.55ManxPowerikarus, try pasting the ACTIAL exten line
15:27.02ManxPowerxbmodder, DO NOT FLOOD THE CHANNEL!!!!!!!!!!!!!!!
15:27.03fishboy1669is there any sites reviewing the 1.2.0 version yet
15:27.04[TK]D-Fenderikarus : just pastebin your extensions.conf for us to see....
15:27.06ManxPower!pastebin
15:27.08kippianyone got anyideas on my Wildcard TE110P card?
15:27.26[TK]D-Fenderfishboy1669 : it just came out.. who has time to give it a qualified review? :)
15:27.48fishboy1669well its been tested for a while
15:27.48xbmodder_lappyhttp://pastebin.ca/29014
15:27.57xbmodder_lappyManxPower, i am not THAT stupid
15:28.18fishboy1669i know testing involves errors but usually people get a feel of wether its an improvement or not
15:28.48froguzwow!! 1.2.0 released! and i just noticed
15:28.54[TK]D-Fenderxbmodder : EXTENSIONS.CONF <-
15:29.22ManxPowerxbmodder, the first thing youi need to do is comment out all the unised stuff.
15:29.47fishboy1669see what i mean about marketing
15:29.57fishboy1669another user thats found out the hard way
15:30.19fishboy1669maybe ive been living in a commercal world to long ;-)
15:30.21froguzthe release process between rc1, rc2 and the stable versions was really fast
15:30.22ikarusManxPower, [TK]D-Fender: a cut down version, but functionally identical http://pastebin.ca/29015
15:31.23fishboy1669frog is that a good thing or not?  did the bugs get found out prpperly?
15:31.32*** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net)
15:31.33[TK]D-Fenderwell.. that Goto is DOA....
15:32.30ikarus[TK]D-Fender: why would it be, it should match the _Z., right
15:33.25[TK]D-FenderWhere is your target context named [extern?
15:33.38ikarus[TK]D-Fender: |:(
15:33.41ikarusthank you
15:33.52froguzfrog is that a good thing or not?  did the bugs get found out prpperly? < it's a very good thing, is just like it allways shoud be in every opensource app
15:33.58ikarusI should not code this late a day
15:34.27ikarus[TK]D-Fender: this is the 2nd stupid thinko I made
15:34.56[TK]D-Fendernp, glad to help....
15:35.07[TK]D-Fenderbut dear God I hate the sample files....
15:35.23[TK]D-FenderI un-AMP'd a guy yesterday!  To a NICE setup now!
15:35.25xbmodder_lappyextensions.conf: http://pastebin.ca/29017
15:35.59fishboy1669btw i stumbleld on this, im probably the last to know but maybe of use to some others of u
15:36.01fishboy1669http://www.internet2.edu/sip.edu/
15:36.15[TK]D-Fenderxbmodder : So what's not working in there>
15:36.23fishboy1669although there just using asterisk as a sip to legacy gateway
15:36.43xbmodder_lappy[TK]D-Fender, i can't recieve a call
15:36.50fishboy1669but interesting to see that its getting used in such a large environment
15:37.22[TK]D-Fendernow pastebin theconfig file with your incoming interfce defined
15:37.28xbmodder_lappyiax.conf?
15:37.31xbmodder_lappyhttp://pastebin.ca/29014
15:38.20Rawplayerhmm, i'am wondering if preemtible kernel has something with asterisk
15:38.31Rawplayeror any hz value's that you can change
15:39.09ikarusAh well, time to go home
15:39.29[TK]D-Fenderok, too much of a mess for me to do here right now.... maybe someone else?
15:40.41*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal)
15:40.50*** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br)
15:40.53*** join/#asterisk ceph (n=amit@68.216.29.70)
15:41.13*** join/#asterisk Evanrude (n=david@wsip-68-15-251-34.dl.dl.cox.net)
15:42.32ManxPowerxbmodder, http://pastebin.ca/29016
15:43.46*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfl99.dialup.mindspring.com)
15:44.33[TK]D-FenderManxPower : You are so nice to remove 99% of his file for him :)
15:44.41[TK]D-FenderI can read it now!
15:44.59froguzhahaha [TK]D-Fender is recieving all the bullets, somebody helps him
15:45.53Samoiedhello all
15:46.08Samoiedanyone know why E3 card isnt for sale at digium?
15:46.16froguz[TK]D-Fender, maybe a way to just mute/unmute the phone microphone by pressing a key?
15:46.59brad_msswanyone in here use e164.org ??  what's the best way to set up a 'guest' sip context for non-authenticated incoming calls?
15:47.48X-FilesPpls, why not working MusicOnHold in protocol OH323 ? small debug http://pastebin.ca/29018 . Please
15:47.48[TK]D-Fenderfroguz : only if you have a mute key on your analog phone from what I can see
15:47.50ManxPower[TK]D-Fender, *I* could not read it 8-)
15:48.16froguzbuh, i think i better should change my ATA
15:48.16ManxPowerSamoied, Billions of reasons.  But the fact is that it is not available and not shipping
15:48.34jontowsangoma has a DS3 card yes?
15:48.59[TK]D-Fenderjontow :Yup, but I'm not sure of its usability in * for TDM voice
15:49.04froguzwell, i have to go
15:49.11jontowyeah, no idea
15:49.13tzangerjontow: not yet I don't think
15:49.16jontowno cash for such a venture :)
15:49.20tzangerlike Digium's card, "it's coming"
15:49.22froguzthanks a lot mr stratocaster ;)
15:49.24tzangerbut personally I'd just use a MaxTNT
15:49.30SamoiedManxPower: But this card was released? Exists in fact?
15:49.34tzangeryou have failover and hardware DSPs
15:49.54ManxPowerSamoied, no, the Digium T3/E3 card was never released.  It was anounced but never released.
15:50.07*** join/#asterisk dalfry (n=dalfry@gateway.ishisystems.com)
15:50.15puzzledI would use a MaxTNT for E3 stuff anyway
15:50.19jontowhttp://www.sangoma.com/products/p_aft-et3-specs.htm ?
15:50.20*** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw)
15:50.26*** join/#asterisk santiago (n=santiago@208.195.215.124)
15:50.40ManxPowerI suspect Digium realized that there was no way in hell they could do a T3/E3 card with their current PCI interface design
15:50.55puzzledjontow: that's a clear channel card. you need channelized T3/E3
15:51.03puzzledManxPower: nod
15:51.07jontow:)
15:51.15X-FilesManxPower: check it http://pastebin.ca/29018 , Please
15:51.30SamoiedManxPower: but, why they have announced?
15:51.45ManxPowerSamoied, Because they thought they could make it work
15:51.54pcmsamoied: they wanted people to start saving hehe
15:52.14jontowManxPower: i heard that it does work; but i also hear that they don't have the resources to make it work in the field, for a number of reasons
15:52.35ManxPowerpcm, Most E3/T3 hardware costs more than the GDP of a small country
15:52.50*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal)
15:52.54pcmmanx: you're talking about channelised though ?
15:53.03pcmmanx: and more than just one port ... ?
15:54.00SamoiedFrom voip-info.org:
15:54.07Samoied"The DS3000P will definitely support PCI-X, and probably bus speeds of 100MHz or higher, so at least if your system has that you will have plenty of bus capacity. Many servers nowadays actually have their NICs on a separate PCI bus as well, so the TDM and NIC cards won't be contending for the same resources.  "
15:55.12*** join/#asterisk copantl (n=galel@205.240.205.192)
15:55.27*** join/#asterisk DrDeke (i=dekemar@auriaria.engin.umich.edu)
15:56.03*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal)
15:57.20ManxPowerSamoied, VOIP-INFO is a lot like the National Enquirer - they have lots of information that sounds good, but it totally wrong.
15:57.26DrDekelol
15:57.47SamoiedManxPower: OK. I use voip-info like bible.
15:57.51SamoiedManxPower: :)
15:58.10tamp4xlol
15:58.39pcmit's a more organized mailng list ...
15:58.49Juggiehas anyone managed to get mpg123 to compile on 64bit?
15:58.54*** join/#asterisk bweschke (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net)
16:00.25*** join/#asterisk swm_ (n=admin@digitaldatabits.net)
16:00.28hypa7iaJuggie!
16:01.33^HowlerJuggie: I have mpg123 on my AMD64 3500+
16:01.52Juggiehypa7ia
16:02.05hypa7iahow goes man
16:02.07Juggie^Howler, i bet your running a 32bit kernel then :)
16:02.11Juggienot bad, working away.
16:02.30Juggieordered 5 digium cards yesterday :P
16:02.37puzzledJuggie: I tried (on FC4) but did not succeed in getting it work with alsa instead of OSS. There is a Mandrake rpm floating on the Net that has most of it
16:02.43^HowlerJuggie: everything on my system has been compiled 64-bit
16:02.55gniretarFC4 is teh sucks
16:03.07Juggie^Howler, for me it dies with asm errors
16:03.24hypa7ianice Juggie
16:03.51^HowlerJuggie: not sure what to tell you...I'm running Gentoo, so everything on my system was compiled from scratch, in 64-bit.
16:04.04puzzled^Howler: does it use alsa or oss?
16:04.12Juggie^Howler, are you sure its mpg123 you have installed, and not 321
16:04.20puzzledgniretar: whatever floats your boat
16:04.22*** join/#asterisk Gerriall (n=NonYa@209.42.198.18)
16:04.25*** join/#asterisk blop (i=blop@pushpop.be)
16:04.30Juggiehypa7ia, whats new w/ you
16:04.32^Howlerpuzzled: alsa, I believe.
16:04.50hypa7iaJuggie: i've only got 2 weeks left at bell :)
16:04.52puzzled^Howler: cool, I should find their src+patches I guess
16:05.20Juggiehypa7ia, what you doin next?
16:05.21^Howler*  media-sound/mpg123
16:05.21^Howler<PROTECTED>
16:05.21^Howler<PROTECTED>
16:05.39chapeaurougeanyone read that book? http://www.amazon.fr/exec/obidos/ASIN/1904811159/qid=1132243502/sr=8-2/ref=pd_ka_1/171-9074066-4115412
16:05.42_Sam--to upgrade from 1.0.9 to 1.2 is there anything in 1.0.9 configurations that neesd to change?
16:05.52puzzled^Howler: the * docs specifically state that you need to use 0.59r
16:05.54Qwell_Sam--: yes
16:06.00Broomqwell: tell
16:06.02Broomjaja
16:06.06puzzled_Sam--: read UPGRADE.txt in the source
16:06.07^HowlerJuggie: I don't have mpg321 installed
16:06.07Broomcuz i recently upgraded
16:06.21hypa7iaJuggie: going back to school and doing contract work
16:06.21Broompuzzled: only what it says there?
16:06.34_Sam--what would be the 'killer app' that would make a person want to upgrade from 1.0 to 1.2?
16:06.39hypa7iaJuggie: got a gig teaching infosec courses for IBM
16:06.41*** join/#asterisk Kort (n=james@65.211.216.202)
16:06.44*** join/#asterisk blop (i=push@pushpop.be)
16:06.47Qwell_Sam--: proper realtime
16:06.58*** join/#asterisk Kokey (n=Kokey@201.153.63.79)
16:06.59Korthey does anyone have their provider giving CID information over a Frame Relay T1?
16:07.04puzzledBroom: afaik yes
16:07.05ManxPowerWell it isn't a SIP jitterbuffer, that's for sure.
16:07.11^Howlerpuzzled: that may be..I really don't know. I'm an asterisk noob. I'm just trying to learn =)
16:07.19puzzled^Howler: ok
16:07.30Broomthere is something weird going with the call groups and 1.2
16:07.32tzangerKort: over a frame relay T1?
16:07.40Broommust be something wrong on my configs
16:07.41tzangerI thought frame relay was a data service, not a voice service
16:07.44KortI have a frame relay t1 that's split voice/data
16:07.47tzangerahh
16:08.04tzangerwell they only way they can signal that is with DTMF before pickup
16:08.26*** join/#asterisk YoMama (n=tchen@pcp02689850pcs.roylok01.mi.comcast.net)
16:08.41Kortsome people mentioned feature group D...
16:08.53Broomwell the configs are ok, it is weird that when i call a group I always get the fall back solution..
16:08.59YoMamaanyone know of a way to find out which NXX's are local to you?
16:09.06ManxPowerKort, You can't do non-VoIP voice over Frame Realy
16:09.09ManxPowerRelay too
16:09.22YoMamaManxPower: not true :)
16:09.25YoMamaManxPower: VoFR
16:09.26ManxPowerYoMama, you call your carrier.
16:09.32ManxPowerYoMama, Not with Asterisk.
16:09.33YoMamait exists and it's a mess but it does work
16:09.34YoMamaoh
16:09.43YoMamawell, duh...asterisk is 100% IP-based
16:09.44KortManxPower: so basically I"m screwed when it comes to having any CID service..?
16:09.45YoMama:-P
16:09.54tzangerYoMama: yes
16:09.55ManxPowerKort, you're screwed, since you can't get any service.
16:10.01tzangerlook up the calling area guide on dandy.net
16:10.05Kortsweet
16:10.10Kort...
16:10.12YoMamatzanger: sweet..that's what i was lookin' for
16:10.15tzangerhttp://members.dandy.net/~czg/lca_index.php
16:10.26ManxPowerKort, you either need channelized T-1/E-1 or a SIP/IAX connection to your carrier.
16:10.41ManxPowertzanger, that's only for ILEC's right, and only if you don't have a special callingplan?
16:10.45Broomanyone, is this a valid command: Macro(rg-group,ringall,60,,ext1-ext2-ext3) ??
16:10.46KortSo the setup I have now is a Frame Relay T1 split data/voice
16:10.53tzangerManxPower: not sure
16:10.57Kort(I think)
16:10.59tzangerlook up your # and see :-)
16:11.07ManxPowerKort, I suspect you have a T-1, with some channels for voice and some channels for Frame-Relay
16:11.08tzangerKort: so how are you getting voice out of this?  a T1 going to *?
16:11.17tzangerI understand that it's split voice/data but how is the voice getting to *
16:11.20Kortmanxpower: correct.
16:11.20ManxPowerin which case your voice channels have nothing to do with Frame Relay.
16:11.22tzangerover ethernet or over TDM?
16:11.26Korttzanger: DSU/TSU
16:11.29YoMamaManxPower: even certain channelized T configurations CID won't work right
16:11.42Kortsplits the channels out
16:11.43ManxPowerFor example we have a T-1 in Gulfport MS with PRI on channels 1-17 and 24, and Frame Relay on channels 18-23
16:11.53YoMamaManxPower: yummy
16:12.08ManxPowerYoMama, Well your carrier has to support FSK for CallerID
16:12.21*** join/#asterisk MacDeath (n=davidn@david.tsol.co.za)
16:12.21tzangerYoMama: odd, I owuld have figured out for any CAS T1 the only way you're gonna get CID is with a wink start
16:12.31ManxPowerYoMama, not bad, we'll just toss an Adtran 120 between Asterisk / Router and the T-1
16:12.40KortSo basically I'm wondering if there's anything I can tell MCI to get CID working on the Voice channels
16:12.46Kort(yes I"m new at this VoIP stuff)
16:14.04tzangerI'd imagine you call them and say you want a wink-start on your voice channels, and hope that * handles CID fetching via DTMF in the wink
16:14.11tzangerbut generally that is reserved for DID
16:14.17MacDeathI have a small problem
16:14.18tzangeryour CID may still come through between ring 1&2 with FSK
16:14.24fishboy1669anyone here tryied setting up * in "high availability" on linux?
16:14.31MacDeathI cant transfer incomming calls.....
16:15.20Kortheh and pray :)
16:15.26*** join/#asterisk yxa (i=empty@cm121.gamma228.maxonline.com.sg)
16:16.04MacDeathmy incoming context looks like
16:16.05MacDeath[incomingisdn]
16:16.05MacDeathexten => s,1,Dial(SIP/1237&SIP/1241&SIP/1238,60,Tt)
16:16.18MacDeathwhen i press the transfer button, the caller gets put on hold
16:16.28YoMamatzanger: i don't even wanna mess with channelized T's anymore..PRI's just work
16:16.32MacDeathI dial the extension to be transfered to
16:16.40MacDeathit rings
16:16.50MacDeaththe person picks up, I can speak to them
16:17.07YoMamabut they can't speak to you
16:17.08MacDeathbut when I put the phone down, the call comes back to me, not to the person who I dialed
16:17.09brad_msswif I wanted to allow 'guests' to come in via sip, unauthenticated, and just hit a menu system ... what would the  type=   be ?  and would I set host=dynamic still ?  I assume no secret would be specified
16:17.30YoMamaMacDeath: i think you're accidently parking the call
16:17.37YoMamaor you're not completing the transfer
16:17.41tzangerYoMama: amen to that
16:17.49DrDekehmm
16:17.52MacDeathdo i have to press something to complete the transfer?
16:17.54*** join/#asterisk DeeJayTwo (i=deejay2@215-238.sh.cgocable.ca)
16:17.57DrDekeman, this iLBC codec sounds really good for voice
16:18.01YoMamapark will bounce your call back to you if no one picks up the call
16:18.07KortI curse the cheap-asses who said they couldn't pay for separate voice/data T1s
16:18.08YoMamatzanger: this website rocks...thanks
16:18.19MacDeathor have more than i line in the exten => s,x?
16:18.22[TK]D-FenderDrDeke : if you like Styx maybe ;)
16:18.32gniretarKort: I used to have to deal with a 1/2 data and 1/2 voice T1
16:18.34gniretarit sucked
16:18.34YoMamaalthough..this is gonna make my dialplan really complicated
16:18.39MacDeathDrDeke : I only use Ilbc
16:18.47DrDeke[TK]D-Fender: what about Styx? ;)
16:18.48Kortgniretar: that's what I have
16:18.56Kortso I don't have CID (at the moment)'
16:19.07YoMamatzanger: the local NXX's is about three pages long..and i want all other calls to go out my SIP gateway
16:19.23YoMamaso...does this mean i gotta put ALL of these NXX's in?
16:19.26tzangerYoMama: yup I did the same thing
16:19.28*** join/#asterisk dalabera (n=dalabera@146.82.190.164)
16:19.31tzangerif you can figure out how to group them, go for it
16:19.51YoMamahmm..maybe i can load it into the AstDB
16:19.54YoMamaand have it do a lookup
16:20.02tzangeri.e. if 5551210000 through 5551290000 are there just replace it with _55512NXXXX
16:20.20*** join/#asterisk rculp (n=rculp@66.173.240.20)
16:20.35YoMamatzanger there's different NPA and NXX's..
16:20.36[TK]D-FenderDrDeke : Domo Arigato Mr. Roboto :)
16:20.43DrDekehahaha
16:20.44YoMamai think i know how to do it..i just don't wanna
16:20.44DrDekeright
16:20.52DrDekeI dunno; It doesn't sound robotic at all to me...
16:21.00YoMamai can take this file and copy and paste it in there with some search and replace strings and create a dialplan out of it
16:21.01tzangerYoMama: I just did it in vim with a few regexes
16:21.03[TK]D-FenderIlbc onlyworks with Skype.  *
16:21.05tzangerprety straightforward
16:21.08DrDeke...
16:21.11[TK]D-Fendernicely anyways
16:21.12tzangermy unlimitel one is pretty bug
16:21.14YoMamatzanger: yup..that's what i'm gonna do
16:21.17tzangerbut then again so is my tollfreetleecom one
16:21.17DrDekeilbc is working fine right now between my grandstream and asterisk box
16:21.24[TK]D-FenderGSM = best lowend choice.
16:21.35DrDekei would use that but the grandstream does not appear to support it
16:21.38tzangerYoMama: # wc -l unlimitel.ext
16:21.38tzanger3107 unlimitel.ext
16:22.03rculpok, here's a fun potential flame question. If you're absolutely being forced into a web interface by your boss whether paid or not, what would be the best most flexible choice?
16:22.04tzangertollfreetelecom was significantly shorter (500 lines) because I could replace all of 416 with _416NXXXXXX :-)
16:22.14tzangerrculp: roll your own
16:22.29YoMamatzanger: yeah...this is a regular LEC...it's not that simple
16:22.30LostFrog<PROTECTED>
16:22.38YoMamarculp: kill your boss
16:22.49[TK]D-Fenderrculp : Oh no.....
16:22.50rculpwhile those options so far are viable
16:22.51tzangernah
16:22.53bbzrculp: A@H
16:22.59[TK]D-Fenderyes, death is a valid option....
16:22.59rculpI can't quite go that route
16:23.01YoMamarculp: or at least maim him badly enough for him to reconsider
16:23.10rculplol
16:23.20YoMamaa good bat across the knees will remind him who really does the work
16:23.41rculpand while I would like to do that for past reasons included
16:23.45rculpit's not my personality
16:23.46[TK]D-Fenderrculp : A@H = kiddy style.  I'd have to say the one I'm on was the best I've seen anywhere.
16:23.49marc324someone knows where to find docs on vmail.cgi ?
16:23.55[TK]D-FenderAnd I wish I didn't have it :)
16:24.00rculpheh
16:24.09bbzTK: what do you use?
16:24.17[TK]D-FenderI just helped a guy UNDO the damage that is A@H
16:24.24*** join/#asterisk L|NUX (n=linux@202.5.145.58)
16:24.31[TK]D-Fenderbbz : ScopServ
16:24.32*** join/#asterisk BladeRunner05 (n=feelme@adsl-ull-90-66.44-151.net24.it)
16:24.33tzangerjust go to your boss and say "I've researched the list of possible candidates for this paradigm instantiation but unfortunately their prime action items aren't inclusive to our goals and obligations.  It would behoove us to rethink our priority focus and synergize on a new method of attack."
16:24.39Vivaldidoes "asterisk" have something to do with "asterix"?
16:24.40tzangerwhile he's trying to parse that, use the CLI.
16:24.52LostFroglol
16:24.54tzangerbonus points if you can fit the word "zeitgeist" in there
16:25.07`SauronAsterix and OBELIX!
16:25.32*** join/#asterisk Bentley (n=Bentley@S0106000f3d016dd2.cg.shawcable.net)
16:25.58ManxPowerYay!  I leave for Birmingham this evening
16:26.00rculptzanger: he's a little more tech saavy
16:26.17LostFrogrculp: not if he wants a GUI.
16:26.21rculptzanger: or I'd throw a transconfigure the discombobulator to it's full capacity
16:26.28rculpat him
16:26.39DeeJayTwoVivaldi: From a french speaking point, it appears too... but from an english speaking point... it's probably only the asterisk digit of a phone ;)
16:26.47DeeJayTwoappears to..
16:27.08shido6yes
16:27.15*** join/#asterisk Himeko (i=himeko@S01060040ca128fc3.ed.shawcable.net)
16:27.19tzangerrculp: you have to be careful, they're on to words such as "polyphasic wobulator" and especially "discombobulator" -- you need to use new fresh terms like "hyperbarion positive reinforcement inverse pulse rectification"
16:27.23LostFrogrculp: Can you get him to believe that the token feel out of the network?
16:27.31Vivaldifrom an italian point of view as well ..and spanish too
16:27.37*** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca)
16:27.37LostFrogfell, even.
16:27.46[TK]D-Fenderrculp : I see you're on an older gen EXTENSIONS.CONF version of mine.  I do have one with a lot more feature I jsut hacked upfor a guy yesterday.
16:28.21Bentleydoes anyone know what the "chan_modem" problem with * 1.2 was?  I've got the latest tarball and cvs v1-2-0, both won't start after complaining about chan_modem.so
16:28.26rculpI'll bug you via pm
16:28.29[TK]D-Fenderheh
16:28.49Bentleyeven after using noload => chan_modem.so in modules.conf
16:28.50Kortgod damn my boss is going to kill me
16:28.50ManxPowerBentley, chan_modem was removed from building, but not from installing, hence the error (or something like that)
16:28.50DrDekeBentley: I don't think that chan_modem is supported any more in 1.2.0; you need to have a noload line in modules.conf for all the chan_modem_* modules
16:28.58tzangerLostFrog: I love that dilbert comic
16:28.59DrDekeoh
16:29.05rculpI'll keep the beat the boss idea on the table though
16:29.08rculpthanks for that suggestion
16:29.10rculp:p
16:29.11Bentley_* .. ahhh
16:29.21ManxPoweror just see the install messagge about the modules that were not built with this asterisk and remove them
16:29.43Kort"Sorry you probably won't have any CID service because of the cheap bastards who told the other IT guy to get a Voice/Data T1)
16:29.56Bentleywell .. i did.  there is no chan_modem* files
16:30.18YoMamawhcih dilbert?
16:30.19YoMamalink?
16:30.20syzygyBSDwhat speech recognition software would someone suggest?
16:30.36YoMamasyzygyBSD: there's one that works with asterisk...go into the wiki
16:30.41YoMamafor some reason..i can't place my finger on it right now
16:30.48LostFrogYoMama: the one where dilbert disconnected his bosses network connection.
16:30.53syzygyBSDya, I read about it, heard it takes a ton of processing power
16:30.56YoMamaoh yeah... :)
16:31.06YoMamasyzygyBSD: eh...anything that does that is gonna take a lotta juice
16:31.18YoMamadude..no speech recog..i hate that shit
16:31.37KortsyzygyBSD: best I've used is Dragon Naturally Speaking - no idea if that is compatible at all with *
16:31.42tzangerno it's not
16:31.43DrDekeBut without speech recognition, how am I supposed to put my entire mp3 library on my asterisk "dial a tune" line? ;)
16:31.55syzygyBSDya, I was thinking about trying to write one, just to get a handle on how to process sound
16:31.58YoMama"Hello.  Welcome to Northwest Airlines flight arrival and departure automated information system.  Please say the flight number you're checking on.  Are you still there?  Could you please repeat it?"
16:32.02YoMama"two forty three"
16:32.09YoMama"Did you say two forty five?"
16:32.11YoMama"No"
16:32.14YoMamaAAAAAUGH
16:32.23DrDekeyeah everyone can tell when i am talking to one of those things; I always sound pisse
16:32.26DrDeked*
16:32.33DrDekeand you have to speak in a very mechanical fashion usually
16:32.36syzygyBSDwellsfargo is nice
16:32.37YoMamaDrDeke: i'd rather use touchtone
16:32.41Kortthe worst part is when they won't let you key it in
16:32.46LostFrogI love the ones that want yes or no and never recognize my voice..
16:32.47syzygyBSDi can't say my card number and make it screw up
16:32.48tzangerKort: yes
16:32.48DrDekeand heaven forbid you are using a cell phone or anything else with voice-activity-detection; it will always not hear your first syllable
16:32.50YoMamaKort: werd
16:32.56tzangerI hated Bell Canada's "emily" service
16:33.00tzanger"Please tell me what the issue is"
16:33.04syzygyBSDalways says, "thanks, let me look that up" right every time
16:33.09tzanger"YOU DO NOT UNDERSTAND ENGLISH YOU PATHETIC EXCUSE OF A MACHINE"
16:33.10KortI had to speak a 7 digit number into one of those things
16:33.11YoMamaDrDeke: hahaha.....
16:33.14Kortit took me 10 tries
16:33.26tzanger"I'm sorry, I do not understand the nature of the request.  I'll pass you along to an operator now."
16:33.27YoMamahaha..i like it when it asks me to repeat myself..
16:33.29`SauronKort: type it on the keypad
16:33.29YoMama"Fuck you."
16:33.32`SauronDuh
16:33.35YoMama"I'm sorry.  I didn't get that.  Can you repeat yourself?"
16:33.38KortSauron: couldn't, system didn't support it.
16:33.39YoMama"Fuck you"
16:33.41YoMamahahaha
16:33.44`SauronEh
16:33.45ManxPowermost of those systems accept DTMF for numbers
16:33.50YoMamaManxPower: some of them don't now
16:33.55`SauronI'd be surprised if they don't
16:33.55syzygyBSDya, that is normally what I say when I want to get to an operator
16:33.57BeirdoI'd always get belligerant with those systems
16:34.04YoMamai just keep hitting zero until i get a human
16:34.13Beirdo"give me a human you worthless piece of shit"
16:34.15syzygyBSDsome of the systems catch for swear words to see if you are getting frustrated and route you to an operator
16:34.16YoMamathen u end up with someone in india
16:34.25`SauronWhen I worked at IBM, the first thing you heard calling sup[port was the "you call may be monitored .... blah blah"
16:34.29YoMama"Hello..this is Raj..can I help you?"
16:34.32DrDekeYoMama: That is what I usually do too, but then I get REALLY mad on those systems that just go "Thank you for calling. Goodbye!" after three invalid responses.
16:34.38`SauronSo one guy said "well, monitor THIS! <cussing>"
16:34.45YoMama`Sauron: IBM's Lotus support always emptied to a human which was nice
16:34.49Bentley<PROTECTED>
16:34.57`Sauronthe quality control lady came on the line and said "Sir, is there anything I can help you with?"
16:35.00ManxPowerUntil recently when someone tried to "0" out of the IVR it played "No, No, No!  Try again smart guy!"
16:35.04bweschkeBentley: noload all of the chan_modem modules in modules.conf
16:35.05syzygyBSDI like the guys that have a heavy indian accent "Hi, this is Bob"
16:35.08ManxPowerThen someone complained and I had to remove it.
16:35.09YoMamauh huh
16:35.12syzygyBSDbullshit you are bob
16:35.14tzangerManxPower: who's IVR was that?
16:35.15YoMamayeah like their name is Bob
16:35.16`SauronYoMama: Well, it had that recording, and then went to a human
16:35.16tzangeroh yours
16:35.17tzangerhahaha
16:35.23YoMamaand heaven forbid u wander off the script of what they're expecting
16:35.32ManxPowerbweschke, why not rm -f /usr/lib/asterisk/modules/chan_modem*
16:35.34YoMamacall Dell sometime
16:35.40YoMama"I have aproblem with my PC"
16:35.41DrDekeor better yet, don't
16:35.46Bentleyi did for: chan_modem.so, chan_modem_aopen.so, chan_modem_bestdata.so, chan_modem_i4l.so
16:35.46ManxPowertzanger, the medical clinic customer
16:35.49YoMama"What is your express code?" (heavy indian accent)
16:35.54YoMama"I don't know, can't you look it up by customer?"
16:36.06YoMama"...(long pause)...please hold"
16:36.23LostFrogNo sigh, YoMama?
16:36.31Bentleybweschke: there are no chan_modem* files in /usr/lib/asterisk/modules
16:36.32YoMamaLostFrog: hahahahahah...they don't sigh
16:36.46LostFrogMy people don't either.
16:36.57YoMamaLostFrog: u run a call center?
16:36.58LostFrogI wish they didn't sound so mechanical.
16:37.08LostFrogYoMama: of 2 people, yes.
16:37.09YoMamaLostFrog: stop having them read from a script hen
16:37.16LostFrogYoMama: they don't.
16:37.22YoMamathen..what's their deal?
16:37.24*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal)
16:37.30LostFrogTheir Indian.
16:37.31LostFrog:)
16:37.37LostFrogThey're, even.
16:37.39YoMamafirst u get identification information..and then hire some people who can read people over the phone to tell how formal or informal to be with someone
16:37.44YoMamaooh
16:38.10LostFrogThey should always be formal.. but not machine like.
16:38.11syzygyBSDI like this page http://www.paulenglish.com/ivr/ how to get a human operator
16:38.12YoMamasometimes u can tell when someone wants to be called by their first name..for example, i do..i'm very uncomftable being called Mr.
16:38.28YoMamawell, formal is fine..but machine-like...yeah..that's boring
16:38.31LostFrogEveryone is Mr. or Ms. or Mrs. for us.
16:38.42YoMamaLostFrog: every company i've worked for in the last 10 years was first name basis only
16:38.49YoMamaeven if you were talking to the CEO worth $200 million
16:38.56syzygyBSDI am only 23 and I love Mr as long as it is followed by my last name
16:39.00YoMamau call him Mr. so-and-so and you'd get picked on for a week
16:39.00LostFrogWithin the company, yes..
16:39.07syzygyBSDMr. Blood just sounds good
16:39.26*** join/#asterisk hhoffman (n=hhoffman@tor/session/x-c0dd11c36701ae1d)
16:39.29LostFrogNot for clients though.
16:39.36YoMamaLostFrog: well, yeah..sometimes they'll ask "Can I call you Tony?"  "Sure"
16:39.45YoMamaLostFrog: werd...but the mechanical thing..get some people who can think :)
16:39.50MacDeathYoMama : how can i see if the call is parked?
16:39.55LostFrogOh.. they use our first names.
16:40.11bweschke<PROTECTED>
16:40.11bweschke> noload => chan_modem_i4l.so
16:40.11bweschke> noload => chan_modem_bestdata.so
16:40.11bweschke> noload => chan_modem_aopen.so
16:40.13bbzwhat number do you add to your dialplan when something is busy?
16:40.14bbzis it 101?
16:40.16YoMamalast time i worked near a call center..all the guys used to call each other "Fucker"
16:40.22bweschkeBentley: put that into modules.conf
16:40.49YoMamaMacDeath: i don't think you're completing the transfer
16:40.50Bentleybweschke: that's what i have, i took it out of modules.conf.sample
16:40.53YoMamaso when u hang up..it bounces back to you
16:40.55Bentleyi don't get it
16:41.04Broomhello all, is there any way i can send a notify or change the cid or anything to a call that's being transfered?
16:41.17bweschkeBentley: when did you download the 1.2 code?
16:41.33Bentley20 minutes ago
16:41.45BentleyI've tried both the tarball and from v1-2-0 cvs
16:41.59bweschkeok... and you've rm -rf /usr/lib/asterisk/modules/* and then make install'd ?
16:42.05*** join/#asterisk gorauskas (n=gorauska@66-224-20-131.atgi.net)
16:42.09LostFrogIt's almost butterball time.
16:42.12YoMamaoh what the hell..i can't view the fortune 500 without paying $4.95?
16:42.17Bentleybweschke: that's what I did, yes
16:42.39ManxPowerI'll be on a business trip to New Orleans on thanksgiving
16:43.28DrDekewoot!
16:43.30bweschkecan you pastebin your modules.conf to me?
16:43.40Bentleysure, 1 sec
16:44.01YoMamaLostFrog: lunch rules
16:44.27YoMamaanyone know anything in detail re: the digium hardware?
16:44.35YoMamathe cards are pci 2.2 compliant..what if your PC's not
16:44.38YoMamawill they still work
16:44.45Bentleybweschke: http://pastebin.ca/29028
16:44.52syzygyBSDpci is backwards compatible both ways I think
16:45.04Bentleybweschke: DAMN I'm STUPID!
16:45.06Bentleyi see the prob
16:45.14Rawplayerdoes hzvalue en preemtible kernel invalve with asterisk?
16:45.19bweschkeBentley: you're explicitly loading it before noloading it
16:45.26Bentleyya
16:45.27Bentleydoh
16:45.33YoMamablah
16:45.41YoMamasee...u screw one little thing up and nuttin' works
16:45.46LostFrogYoMama: I wasn't talking lunch.. I was talking rape-the-natives day.
16:45.48Bentleysorry - my bad ...
16:45.56bweschkenp - that load isn't in the modules.conf.samples is it?
16:46.10YoMamaLostFrog: not familiar with that...you gonna go hump some fat girl in the closet during lunchtime?
16:46.25LostFrogThanksgiving, YoMama?
16:46.37YoMamaoh..haha...that's next week dude
16:46.41Broomanyone?
16:46.46LostFrogHence.. almost butterball time.
16:46.50Bentleybweschke: no, i just took the noload lines from the sample ... failed to see the load line tho :-/
16:46.59YoMamaah...rape..nah..give them diseased blanket so they die off quickly..yes
16:47.00bweschkek
16:47.09YoMamaand let 'em build casinos and use their money to buy booze..yes again
16:47.31YoMamai live in a state with almost a dozen indian-owned casinos
16:47.44YoMamathey all suck...scary places where people walk in who have no place at a blackjack table
16:48.24tzangersounds like the "other" casino in Vegas Vacation
16:48.40*** part/#asterisk Bentley (n=Bentley@S0106000f3d016dd2.cg.shawcable.net)
16:48.44LostFrog3db = 2x pressure?
16:49.22chapeaurougeim new to the telephony world, and i got tasked with mergeing our existing pots infrastructure to a pots/voip infra, using Asterisk
16:49.28chapeaurougehow does the asterisk book help? would that be a good purchase?
16:49.53hhoffmanI've found it to be very good for quickly picking up how asterisk works
16:50.02chapeaurougeok
16:50.03LostFrogyep.. 3db = 2x power/pressure.
16:50.27hhoffmanvoip-info.org has some good examples as well
16:50.38kippiif /etc/zaptel.conf wasn't setup probley could this cause errors with my ISDN not picking up
16:50.41*** join/#asterisk digime (n=drooth@ip68-111-235-172.sd.sd.cox.net)
16:50.46chapeaurougeyea, ive been reading thru them all afternoon.
16:51.39hhoffmanchapeaurouge: combine them with the layout of the book and you'll be pretty good to go
16:51.40bbzhow can * determine if a trunk is in use, if it is, search for one not in use?
16:51.48chapeaurougeok thx
16:52.11hhoffmanbbz: there's an example of that on voip-info.org
16:52.50bbzis it in dial() ?
16:52.50ManxPowerSometimes it's nice to have a medical clinic as a customer.  Just got an e-mail from my clinic customer telling me they have a very limited number of flu shots, but to let them know if I wanted one.
16:52.56DrDekehaha
16:53.13kippiis there away I can reset my TE110P card?
16:53.13bbzManxPower: nerd love :)
16:53.15ManxPowerThey are only giving them out to high risk groups and emergency workers at the moment
16:53.36ManxPowerkippi, reboot or unload the drivers
16:53.42LostFrogI think they are giving them out like candy here.
16:53.49bbzhhoffman: what page is it on?
16:53.57ManxPowerLostFrog, I think this is their first shipment.
16:54.01hhoffmanbbz: I forget which example it was... look at the list, I think it was well named
16:54.02kippiLostFrog: how do i restart the driver?
16:54.06*** join/#asterisk fugitivo (n=ajf@209.13.244.248)
16:54.09bbzhhoffman: kk
16:54.58LostFrogkippi: Stop asterisk, rmmod wcte11xp; rmmod zaptel; modprobe zaptel; modprobe wcte11xp ?
16:55.03*** join/#asterisk Flauto (n=zhao@c-24-14-197-214.hsd1.il.comcast.net)
16:55.07LostFrogOhh.. and start asterisk. :)
16:55.25LostFrogThat's how you reload a module/driver.
16:55.28FlautoNov 17 10:47:17 WARNING[14292]: chan_sip.c:9575 handle_response_register: Got 200 OK on REGISTER that isn't a register
16:55.38Flautowhat does this mean?
16:55.40fugitivoany idea why my mixmonitor is not saving any file?
16:57.10hhoffmanbbz: somewhere's on here: http://www.voip-info.org/wiki/view/Asterisk+tips+and+tricks
16:57.18*** join/#asterisk svenna_ (n=svenna@p548D289C.dip0.t-ipconnect.de)
16:57.28marcus2does * compile on osx out of the box?
16:57.42bbzhhoffman: yea, im reading through those
16:58.01bbzi assumed that was probably where you were meaning :)
16:58.15ManxPowerEGADS!  I just discovered our cable.monkey is a jesus freak!
16:58.46bbzhhoffman: ahh hunting..
16:58.48Flautoi got that message from asterisk saying that Nov 17 10:47:17 WARNING[14292]: chan_sip.c:9575 handle_response_register: Got 200 OK on REGISTER that isn't a register
16:58.48Flauto. what does it mean?
16:59.01BeirdoFlauto: no need to repeat
16:59.17YoMamaManxPower: as long as he does his job..who cares..just tell him if he preaches to anyone at work, you'll kick his ass
16:59.19Flautobeirdo, sorry, i thought people did not see it or something
16:59.30YoMamaFlauto: still got problems?
16:59.33Flautoyomama, my vbuzzer is working now.
16:59.34YoMamaFlauto: u get incoming working?
16:59.38Flautoyes
16:59.39YoMamaFlauto: congrats man
16:59.42Flautothanks
16:59.46Flautoit works well
16:59.48LostFrogFlauto: I think you've been asking that question for a week now.
16:59.49hhoffmanbbz: think I'm thinking of this: http://www.voip-info.org/wiki/view/hunt-dial+macro
16:59.54YoMamayeah..it's pretty badass
16:59.57kippiis there away I can make sure Zaptel is working?
17:00.05LostFrogzttol?
17:00.07fishboy1669night guys
17:00.07YoMamazttool?
17:00.08LostFrogzttool?
17:00.32*** join/#asterisk viperdude (n=jon@84.45.193.6)
17:00.39viperdudehi everyone
17:00.47YoMamaFlauto: u know which codec vbuzzer uses?
17:00.49pcm~zttool
17:00.51YoMamaprobably GSM
17:00.58Flautobased on your config, the incoming started showing but no voice, so i had to add nat and some other stuff there, but anyway, it is working now
17:01.02kippiits showing a red alarm
17:01.03LostFrog~zttol?
17:01.09LostFrog~zttool?
17:01.16Flautofrom my understanding, ulaw, gsm and alaw
17:01.16YoMamahehe...
17:01.26viperdudedoes anyone know if atxfer and blindxfer are availalbe in 1.0.8?
17:01.29YoMamaFlauto: all three?...eh
17:01.35Flautoi think so
17:01.52Flautoyomama did you see my message about the message i get from asterisk
17:01.55Flautowhat does that mean
17:02.00ManxPowerviperdude, blinf transfer is, attended transfer is not.  In 1.0.8 those features are not configured in features.conf
17:02.13kippiis there anyway I can get more information on my red alarm?
17:02.23viperdudeManxPower thanks... where is it configured in 1.0.8
17:02.33ManxPowerkippi, no.  red alarm is normally a cable error or a carrier error
17:02.39LostFrogkippi: start with checking your cable.
17:02.43pcmkippi: most likey your cable is wrong
17:03.00LostFrogkippi: what is the T1 card connected to?
17:03.06pcmkippi: use loopback cable or make one to investigate
17:03.21Flautoyomama the other question i have is that do you know why i am still able to call even though i am not under any their plan? i mean vbuzzer dialout plan
17:03.34Flautoi called my cell for testing and i thought that i would be charged
17:03.39Flautobut it is showing 0
17:03.42YoMamaFlauto: they are prepay
17:03.48YoMamahuh..haha...nice
17:03.53YoMamawell, shit..i actually paid
17:03.55iCEBrkrIf I'm using Asterisk to make calls via PRI, would I still need a beefy machine like as if I were making VoIP calls?  There's still codec utilization, right?
17:04.00Flautohehe
17:04.22YoMamaiCEBrkr: depends...not if all of your extensions are FXS
17:04.28Flautomeans, ifyou get through without a dial out plan, it is free?
17:04.29pcmicebrkr: it depends how many calls you're going to ahve (simultanous ones)
17:04.48pcmicebrkr: for a few calls you can use 1.0Ghz
17:04.51YoMamaiCEBrkr: if u have IP handsets..then it's using a codec to talk from hanset to asterisk box
17:04.53iCEBrkrpcm: Zero VoIP calls. But probably up to 72 FXS
17:05.14LostFrogOhh.. that should be practically free, iCEBrkr.
17:05.18iCEBrkr:P
17:05.19pcmicebrkr: then at least piv 2.6 or more!
17:05.25YoMamaiCEBrkr: handling PRI does require some overhead..but 72 FXS -> PRI..you're not gonna need too much for that
17:05.34iCEBrkrOK, just checking....
17:05.35YoMamaconferencing takes some juice...
17:05.39Flautohi to yomama, icebrkr, how are you guys doing
17:05.43iCEBrkrI didn't think so, as there's no transcoding/codec stuff
17:05.50YoMamaIcemaann: correct
17:05.52YoMamaerr
17:05.52iCEBrkrFlauto: you learning this shit yet :P
17:05.56YoMamaiCEBrkr: correct
17:06.09iCEBrkrI still think I'm gonna get a dual-xeon!
17:06.14YoMamaiCEBrkr: is there a reason why you're not using IP phones?
17:06.35LostFrogHe's starting his own phone company, YoMama. lol
17:06.36Flautoicebrkr, what are you talking about
17:06.39ManxPowerviperdude, t and or T on the Dial line
17:06.50pcmicebrkr: if you can then get it :) however one strong cpu should be enough for 72 fxs over 3 T1's
17:07.29iCEBrkrYoMama: Cuz I'm just dialing.. Not using it as a PBX
17:07.39Rawplayershould i use asterisk from source or from package?
17:07.43DrDekesource!
17:07.44iCEBrkrpcm: In the near future we may use VoIP
17:07.46LostFrogsource!
17:07.46DrDekeuse the source, luke
17:07.51Flautoyomama, did you see my message?
17:07.55*** join/#asterisk DarthClue (i=DarthClu@ACAA620D.ipt.aol.com)
17:08.06FrostedCongrats on 1.2.0! :D
17:08.16iCEBrkrI guess I could just spec out a medium dual-xeon box..
17:08.55pcmor dual-core amd xp ?
17:09.07iCEBrkrno dualcore
17:09.12DrDekewhy not?
17:09.13iCEBrkrYou don't get much for the price.
17:09.24iCEBrkrTho, I suppose Linux would actually make use of it.
17:09.28pcmthey sell for $300 ... 2.0 Ghz dual-core xp
17:09.28iCEBrkrunlike wintendo.
17:09.34DrDekeerr... windows has SMP support too you nkow
17:09.37DrDekeunless we are talking pre win2k
17:09.48iCEBrkrDrDeke: Windows apps don't use the shit man..
17:09.59DrDekesure they do if they are multithreaded; same as linux
17:10.02iCEBrkrno
17:10.12DrDekewhat makes you think that?
17:10.17iCEBrkrYou're better off with dual processors + hyper threading than dual cores
17:10.24DrDekewell of course you are!
17:10.33viperdudeManxPower: thanks
17:10.42iCEBrkrDrDeke: cuz the apps don't use the shit right
17:11.03DrDekei think it's more because you have two separate caches and busses, and plus you have the hyperthreading
17:11.05*** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at)
17:11.14iCEBrkrJust like back in the day.. Win2k supported multiple processors. Meanwhile, you'd look and CPU0 would be at 99% while the other CPU would be idle.
17:11.15DrDekeA dual-core CPU appears to windows applications the same way two physical CPUs appear.
17:11.22*** join/#asterisk fiftyCal (n=b@69-160-145-156.ontrca.adelphia.net)
17:11.35DrDekethat means you're running one non-multithreaded task...
17:11.37iCEBrkrDrDeke: Go read some articles.. It ain't worth it yet.
17:11.40DrDeke(either that or something is screwed up
17:11.41DrDeke)
17:11.41LostFrogThat's fine, iCEBrkr: just run seti on CPU1.
17:11.42LostFrog:)
17:11.47iCEBrkrLostFrog: hehe
17:11.56viperdudeI have noticed that if a agent takes a call from a queue and then transfers it away to another exten the agent doesn't become available again until the transferred call is finished. Is there anyway to make the agent available again why the transferred call is still in progress?
17:12.25pcmviperdude: that sounds like a bug
17:12.45YoMama*whoop* *whoop* bug!
17:13.10DrDekeicebrkr: http://www-personal.engin.umich.edu/~dekemar/cpu.JPG
17:13.38iCEBrkrDrDeke: Dude, just go read tom's hardware or something.... and you'll see what I'm talking about
17:13.41*** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)
17:13.46DrDekespecific url please ?
17:14.00ZeeekManxpower ?
17:14.01iCEBrkrDOn't have one.
17:14.08iCEBrkrBut you can search through there.. Do your research
17:14.13DrDekelol
17:14.20DrDekeI think you are the one who needs to do some research ;)
17:14.22iCEBrkrCuz I was just about to buy a dual-core, and I read the article.
17:14.46pcmicebrkr: do you have the t1 cards already ?
17:14.52ZeeekAnyone have a PolyCOm freeze on reboot?
17:14.52DrDekeNow, if you're trying to say that your "typical" single-threaded app won't take advantage of 2 cpus or cores, then sure, you'd be right.
17:15.09iCEBrkrDrDeke: Half-life won't use it. Is what I'm saying.
17:15.11DrDekelol
17:15.18DrDekeof course half-life won't use it; half-life is single threaded
17:15.23iCEBrkrThus. It's not worth it until apps actually use the horses in that damn CPU
17:15.32DrDekeor even if it isn't, it does virtually all its processing in one thread
17:15.45DarthClueZeeek, yep, can you get into the setup for the phone?
17:15.51iCEBrkrpcm: Yeah
17:15.59iCEBrkrpcm: You trying to get rid of some Digium cards?
17:16.19YoMamapcm: yeah..if u wanna sell some TDM400 cards..i'll tak eone :)
17:16.25YoMamai'll give ya a dollar for it :-P
17:16.26iCEBrkrhaha
17:16.39ZeeekDarthClue It freezes on Running App=sip
17:16.51ZeeekHave to unplug to reboot
17:17.00Zeeeksetup checks ok
17:17.12ZeeekIt dies while loading the xxxxxxx.cfg (Mac)
17:17.21DarthClueZeeek: you need to go into the setup and point it to a boot server, have it reload everything.  That's how I fixed mine.
17:17.25Zeeekmaybe I should try to boot without the ftp server
17:17.38Jas_WilliamsZeeek: reflash
17:17.42Zeeekya that's what I'm doing
17:17.45DrDekeI use my primary computer for such things as transcoding and editing video and large image files, and things like that can (depends on what exactly you're doing) make good use of multiple cores/cpus.
17:17.59*** join/#asterisk Smidge (n=steve@steve.enta.net)
17:18.08ZeeekJas_Williams long time no see... I can't reflash unless you mean reload sip?
17:18.21*** part/#asterisk lme (n=lme@gw-r-ifc.infoclip.fr)
17:18.40Jas_WilliamsZeeek: setup your BootP server and reload phone spftware from scratch
17:19.05Jas_WilliamsI haven't been here in a while far too busy doing paid work :)
17:19.08*** join/#asterisk zoa (n=zoa@pirus.securax.be)
17:19.12zoayo yo
17:19.17DrDekepaid work? what is that? *grin*
17:19.41zoaanybody from digium online and awake ?
17:20.08FrostedCan anyone tell me how I can get the number of who's calling in an agi script? for some reason it was removed from agi_channel.. any new variables?
17:20.34Frostedonly on IAX2 it seems
17:20.50Jas_WilliamsFrosted: how about passing it in as a variable ?
17:21.20cpatryGET VARIABLE CALLERIDNUM ?
17:21.21*** join/#asterisk jdv79 (n=jdv79@u1057064.ul.warwick.net)
17:21.29Frostedcallerid isn't reliable ;)
17:21.32jdv79what's up with * starving a dual xeon 3.4 box of CPU power? insane!
17:21.40cpatrywhich number u mean?
17:21.47FrostedJas_Williams: but it's from agi_channel I know what the number is. hehe
17:21.48DrDekejdv79: What kind of and how many channels are we talking here?
17:22.00jdv79SIP, about 500
17:22.08DrDekewell, that is quite a few
17:22.09zoatranscoding ?
17:22.12DrDekeIs it doing any transcoding?
17:22.20jdv79none - ulaw afaik all around
17:22.20zoawhat exactly do you want to do? it should be able to handle 500
17:22.27DrDekehmm
17:22.33DrDekeWhat are the symptoms of being starved of power?
17:22.36jdv79i want to see it pass the calls and not sound subpar
17:22.45jdv79vmstat is telling me what i'm hearing
17:22.52jdv79rather confirming
17:22.56zoaare you using agi
17:23.02jdv79no
17:23.03zoaare you using recordings
17:23.07jdv79yes
17:23.12Jas_Williamsjdv79: You must be transcoding then SIP should be point to point
17:23.16zoaall of it is recorded ?
17:23.26jdv79the test is all recorded
17:23.53Jas_Williamsjdv79: You have a Disk bottleneck then
17:23.55DrDekewhat are you seeing in vmstat? too much CPU time? too much I/O?
17:24.22jdv79fluctuating but large "processes waiting for processor"
17:24.26DrDekehmm
17:24.29jdv79the "r" column
17:24.37DrDekeyeah
17:24.47Jas_Williamswhat type of disk ? an array
17:25.33zoathats you problem
17:25.35zoaturn it of
17:25.38zoaor use a ramdisk
17:25.41zoawait a sec
17:26.28zoahttp://www.asteriskguru.com/archives/viewtopic.php?t=40206&highlight=512+recordings
17:26.30zoaread this
17:26.32zoai documented this before
17:26.36zoaits in that email
17:27.13jdv79hmm
17:27.30jdv79the disk is a single scsi i believe
17:27.38Jas_Williams<PROTECTED>
17:27.42jdv79might be a raid 0 pair - don't remember
17:28.18jdv79i'm not a good c coder but is there a way generically to profile something like *?
17:28.37jdv79i do a lot of perl and there's a couple profilers we use
17:29.02jdv79thanks Jas_Williams - i'll check those out
17:29.51*** join/#asterisk MYing (n=Ming@mying.enta.net)
17:30.08DrDekeI'm surprised nobody has done this: http://www.voip-info.org/wiki/view/Asterisk+Bounty+SIP+encryption
17:30.21zoaJas_Williams: thats the same link as i have :)
17:30.31Jas_Williamstrue :)
17:30.36zoawhatever you do with disks its not going to be fast enough
17:30.49zoaunless you patch asterisk to do the recordings in a better way
17:30.58zoawith buffering
17:31.28zoaif you have 500 calls, its going to do 500 disk writes every 20ms
17:31.28zoathats a bit too much
17:31.29zoafor any disk
17:31.39*** join/#asterisk jojo (n=jojo@c83-253-39-20.bredband.comhem.se)
17:32.26zoajdv79: there is not a lot to profile, that thing is your bottle neck
17:32.41zoathe write to disk in the monitor application
17:32.57*** part/#asterisk rculp (n=rculp@66.173.240.20)
17:32.57bweschkeDrDeke: it's not a trivial task
17:33.11zoaeven if you go for a small buffer in monitor before writing to disk
17:33.15DrDekebweschke: Well, no, but neither was writing asterisk, presumably :)
17:33.22zoayou will still store an insane amount of data in a very small period
17:33.31DrDekeIt just seems to me like people would be pretty interested in it.
17:33.31*** join/#asterisk justinu (n=j2@72.18.13.48)
17:34.04bweschkeDrDeke: they are.. and there is a patch in the bug tracker for post 1.2 that sets the stage for SIP over TCP, which is the groundwork for TLS transactions with SIP
17:34.14jdv79so ramdisk it is then?
17:34.14DrDekeOh, interesting.
17:34.19jdv79that's the only sane fix for now
17:34.26zoaa ramdisk will be full very fast
17:34.43jdv79huh?
17:34.51zoahow much ram do you have ?
17:34.52zoa2gig ?
17:34.54zoa8 gig ?
17:35.02jdv79oh no, i'm not recording on hte fly - i'm playbacking
17:35.10jdv79the recordings already exist
17:35.12zoahow do you mean ?
17:35.12zoaaaah
17:35.13DrDekeSo pop the files you're playing in the ramdisk! :)
17:35.14jdv79they are the test
17:35.17zoayou are playing audio ?
17:35.20jdv79yes
17:35.24zoaok
17:35.26jdv79sorry bout the confusion
17:35.34zoathats not a good way to benchmark it
17:35.37zoado you use 2 servers ?
17:35.39zoaor 1 server ?
17:35.46jdv792 or 3:)
17:35.48jdv792 right now
17:35.52ZeeekJas_Williams what versions of firmware do you use?
17:36.04jdv79its just a rough out test
17:36.16jdv79other server just blindly echoes
17:36.25zoaput the file on a ramdisk, it will help a little
17:36.34jdv79hmm
17:36.36zoai did a lot of tests like that before
17:36.42zoaand use BIG files
17:36.45ZeeekJas_Williams the ip500 was delivered with sip 1.3.4
17:36.51zoadont use 20 second files
17:36.59jdv79i have 45 second files at the moment
17:37.02zoabut a file of 20 minutes or so
17:37.03zoanot good
17:37.05jdv79oh
17:37.11zoaasterisk cannot handle more than 20 calls per second or so
17:37.23jdv79that's the application though
17:37.34jdv79its a dialer
17:37.46zoaso when the file is over i suppose you are going back to playing back the same file ?
17:37.55jdv79maybe, maybe not
17:37.59jdv79right now for the test yes
17:38.02zoaanyway, i recommend against doing 500 calls on 1 machine
17:38.07zoause some logic to load balance it
17:38.53zoawhat happened to blitzrage ?
17:38.56zoahe left the earth ?
17:41.04cpatryzoa: he was in montreal last week-end.
17:42.17*** join/#asterisk Nukemizer (n=Nuke@67.137.28.163)
17:44.27LostFrogwow.. Switched to * on Monday. Only a minor voice volume problem on Thursday.
17:44.40LostFrogThree day transition.. not bad.
17:48.00fiftyCalnetstat -udp | grep 5060 doesn't do what i hoped it would (let me snoop for inbound sip requests).  i know its really a netstat question, but can someone help me with it?
17:48.05justinuLostFrog: congrats
17:48.18justinufiftyCal: lol, use tcpdump or tetherreal
17:48.20LostFrogumm.. fiftyCal: UDP is connectionless.
17:48.22justinutetheral
17:48.25DrDekefiftyCal: Well, you could "tcpdump -n port 5060"
17:49.03LostFrogIs boiling a hot dog a crime?
17:49.11santoshr"exten => _7.,1,Pickup(${EXTEN:1}@default)" can some please exemplify this
17:49.33fiftyCalthanks, thanks and thanks. :)  even though connectionless, netstat can still show udp activity, right?
17:49.35LostFrogshow application Pickup?
17:49.36syzygyBSDhell no, boiled hot dogs are great, and you can cook a ton at once
17:49.38santoshrwhat is tht default is it a context. and the EXTEN:1 wht is that
17:49.43justinuLostFrog: that's how they do it in NYC
17:49.54LostFrogTrue.. the carts have water pans.
17:49.56justinudirty water dogs
17:50.07LostFrogI usually grill/fry mine.
17:50.22syzygyBSDmicrowave is quick too...
17:50.24syzygyBSDlol
17:50.28LostFrogno.. netstat can not show connection info for a connectionless protocol
17:50.41santoshr"exten => _7.,1,Pickup(${EXTEN:1}@default)"  ?????????/
17:50.49LostFrogI can't believe I actually had to type that.
17:52.17*** join/#asterisk rajiv|work (n=rajiv@gentoo/developer/rajiv)
17:52.25fiftyCalnot looking for connection info, statistics.  not what udp connections are open (impossible), but what udp packets have been/are being sent across
17:53.02dalfrysantoshr, for a pattern _7., priority 1 run app Pickup with argument extension@default and extension will be stripped of the first digit
17:53.11LostFroglol. dalfry.
17:53.21dalfry:)
17:53.49*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
17:53.51*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net)
17:55.02LostFrogfiftyCal: try jnetcal
17:55.14LostFrogjnettop
17:55.23*** join/#asterisk blop (i=push@pushpop.be)
17:55.26enderso upgrading from 1.2.0-beta1 to 1.2.0, is there any gotchas I need to worry about?
17:56.28fiftyCaltcdump is doing the trick.  thanks though.
17:57.09marc324ne1 uses odbcstorage?
17:57.53*** part/#asterisk rajiv|work (n=rajiv@gentoo/developer/rajiv)
17:57.56DrDekeender: i had to in modules.conf add noload statements for the chan_modem_* drivers, but that problem may have been solved
17:58.33LostFrogWhat were you running prior to release, DrDeke?
17:58.34santoshrdalfry: tht extension@default would mean .. regisetered extension in the default context.. if i havea context called testing and extension 668 669 667  666 then it would be 7.,1,Pickup(66@testing)  ??
17:59.25santoshris it required to specking pickupgroup and callgroup to use this
17:59.43^Howler*  media-sound/mpg123
17:59.43^Howler<PROTECTED>
17:59.43^Howler<PROTECTED>
17:59.56^Howleroops, sorry...right clicked.
18:00.02*** part/#asterisk mog_work (n=mogorman@gateway.digium.com)
18:00.52dalfrysantoshr, yes
18:01.47*** join/#asterisk SplasPood (i=nobody@paravolve.net)
18:02.17enderanybody know if there are srpms of Asterisk 1.2 floating anywhere?
18:02.21*** join/#asterisk mog_work (n=mogorman@gateway.digium.com)
18:03.02*** join/#asterisk Gennaro (n=Gennaro@ppp-62-10-136-138.dialup.tiscali.it)
18:03.11Gennarohi
18:03.17*** join/#asterisk AgiNamu (n=AgiNamu@8.7.80.194)
18:03.19Gennaroi'm going to intall asterisk
18:03.28AgiNamuHi... anyone here have ABE experience?
18:03.29Gennaroand i havent yet
18:03.35Gennaroany ip tel
18:03.51dalfrysantoshr, the called extension $EXTEN will have a 7 to start with which will be stripped by that :1
18:04.00Gennaroi'd like to use and intel modem to connect pstn
18:04.11Gennaroso...
18:04.21dalfryso if testing has 7666, that line will call 666 668 etc. stripping the 7 off it
18:04.25Gennaroi have my kenel source
18:04.28Gennaro..
18:04.47*** join/#asterisk diclophis (n=diclophi@adsl-69-238-124-226.dsl.pltn13.pacbell.net)
18:04.49Gennaroand i'm going to compile zaptel and asterisk..
18:04.52diclophis:wq
18:04.56Gennaroafter that i do it..
18:04.59diclophisdamnit wrong window
18:05.15Gennarohow can i do to test an incoming call?
18:05.31LostFroglol.. Your IRC session has been saved and ended, diclophis.
18:06.17Gennarosome one can help me?!?
18:06.29santoshrdalfry: sory but i am slightly confused.. i mean..
18:06.32diclophisyea, and i think i just solved my problem too
18:06.34santoshrwait lemme show u something
18:07.03diclophisthe problem i was having was that in AGI, if you run an App command Background, and the caller presses a digit, it drops it before it goes to the Read command
18:07.29diclophisBUT, if i play the sound using the read command, it behaves like background, and doesnt drop the first digit
18:07.37drumkilladiclophis: yeah, that's not the way to do it in AGI
18:08.08diclophis...?
18:08.15marc324is odbcstorage compatible with voicemailmain() function?
18:09.15santoshrdalfry: can u please have a look at this       http://sial.org/pbot/14338
18:09.22mockerDoes anyone know if AMP is compatable with Asterisk RC1.2?
18:09.23*** part/#asterisk gr0mit (n=w10277@dhcp4.zuk40.mot-tools.co.uk)
18:09.32diclophisdrumkilla, what would be the way?
18:09.51drumkilladiclophis: actually, I'm not sure there is a way to do what you want to do with AGI ...
18:09.54diclophisalso how do i let the caller press * when they g into their voicemail to get to the menu
18:10.07dalfryyeah, what about it?
18:10.08*** join/#asterisk tracinet (n=tracinet@216.242.235.2)
18:10.17diclophisno, i got it working... if i use the Read command with the optional sound file.. it saves all the digits the caller presses
18:10.21drumkilladiclophis: you would need to use app_externalivr
18:10.23Gennarois there an alternative at mpg123 ?
18:10.30drumkilladiclophis: oh, i didn't know it had an optional sound file
18:10.31diclophisAND stops the sound when they press a digit
18:10.32drumkillanm  :)
18:10.33santoshrnow. if 667 calls 668 can a third person piclup..
18:10.37Gennarosome one tried FESTIVAL?
18:10.39santoshrthts wht i am tryin to do
18:10.43filedrumkilla: you're supposed to be sleeping, go into an unconscious state
18:10.45santoshrall are h323 devices
18:10.47santoshr00h323c
18:10.48diclophisexternalivr?
18:10.51dalfrywhat is pickup?
18:10.57drumkilladiclophis: nevermind that :)  you can just use read ...
18:10.58santoshrcall pickup
18:11.10santoshrhttp://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup
18:11.13diclophisso yea... now about the voicemail thing
18:11.21*** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no)
18:11.40diclophiswhat would be the best way to initial the voicemail management menu for a caller?
18:11.54filepickup allows you to pickup a ringing extension, just by knowing it's relative extension
18:12.26santoshrknowing its reltive extension ?
18:12.32drumkilladiclophis: they press '*' and the call goes to the 'a' extension
18:12.33filethe current extension associated with it
18:12.57filewell see, if you do a goto the original extension doesn't apply - the new one does
18:12.59fileexample:
18:12.59diclophismmm not using the dial plan... its all in AGI for now
18:13.06fileexten => 145,1,Goto(146,1)
18:13.11fileexten => 146,1,Dial(SIP/jcolp)
18:13.14fileyou would do pickup on 146
18:13.24drumkilladiclophis: to be honest, i never use AGI :)
18:13.33drumkilladiclophis: do you get the return value of the application in AGI?
18:14.15drumkillayes, you do
18:14.24drumkillasoooo, the return value should be the ascii value of 'a'
18:14.26santoshrbut file       http://sial.org/pbot/14338  if this where the case
18:14.31drumkillathat's how you know they exited with '*'  :)
18:14.44filethe application was changed when it went into CVS
18:14.49fileyou have to explicitly specify an extension
18:14.50diclophismmm negative
18:14.51filecan't be partial anymore
18:15.04diclophisi get to the voicemail opening message and pressing * does nothing
18:15.11diclophisnor does pressing it during the recording
18:15.14dalfrysantoshr, so your original EXTEN:1 will work here
18:15.28dalfryif 668 is ringing, you dial 7668 to pick it up
18:15.43dalfrythe EXTEN:1 will strip the 7 out of it and pass the rest to pickup
18:15.57Gennaroi unpacked mpg123 in usr/src/mpg123...r
18:16.02Gennaromake linux
18:17.04BladeRunner05I follow the steps in http://www.voip-info.org/wiki/view/Asterisk+cdr+odbc form log call in mysql using odbc but it dosen't do that
18:17.15drumkilladiclophis: well, it's supposed to.
18:18.34diclophisdamnit, so i have asterisk not running as root (apache instead) and now the asterisk -rvvv doesnt work
18:18.44diclophisand as root i cannot read /var/run/asterisk.ctl
18:18.59diclophishow can root not read something
18:21.47*** join/#asterisk PBXtech (i=nik@178.sub-70-213-166.myvzw.com)
18:22.01*** join/#asterisk liran_ (n=liran@80.178.63.23)
18:22.36[TK]D-FenderQuestion about V1.2 function syntax : is this valid? - "exten = 123,1,Set(CALLERID(name)=Q:CALLERID(name))
18:23.04[TK]D-Fenderwith the old $ escape syntax I could be sure...
18:23.18drumkilla[TK]D-Fender: almost
18:23.20[TK]D-Fenderand adding the missing ">"
18:23.34[TK]D-FenderI'm trying to upgrade my scripts.....
18:23.41drumkillaSet(CALLERID(name)=Q:${CALLERID(name)})
18:24.01[TK]D-Fenderyou still have to call it like a var?
18:24.16santoshrok thanks dalfry
18:24.19drumkillayeah, when used like that ...
18:24.31drumkillabecause it's being used the same way as a variable
18:24.40drumkillait's being evaluated before the application is called
18:25.35*** part/#asterisk queuetue (n=queuetue@toronto-HSE-ppp4122670.sympatico.ca)
18:25.55[TK]D-Fenderand the same style of change neede for EXTEN 1,1,VoicemailMain(CALLERID(number)@default) ?
18:26.51santoshrdalfry: just one more dumb question ., where would tht  "exten => 7.,1,Pickup(66@testing" be i mean in the same conext correct
18:30.05diclophishow do i limit the options of the voicemailmain menu ?
18:30.22[TK]D-FenderAlrighty... upgraded.  Monday I'm getting my SPA-941 and SPA-3000 and my * will be back in action.....
18:31.25ikarusHmmmm, I need to install the newest CVS of asterisk, I think I found a stupid bug
18:32.00santoshr"(testing, 7668, 1)" it is showing me this..
18:32.21ikarus484's are not send after directing into a Goto, then Asterisk sends 404's
18:33.38AgiNamucan Asterisk do multiple ISDN switch types? i.e., if one provider is NI2 and another is 5ESS?
18:34.23mog_workhey peoples
18:34.29mog_worksubmit 1.2 stories to slashdot
18:34.37mog_workwinner gets a muffin
18:34.39mog_workfrom mog
18:35.19*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
18:35.28hypa7iawhat kind of muffin?
18:35.38mog_workANY kind
18:35.44fileBLUEBERRY!
18:35.51mog_workexcept diamond or gold encrusted type
18:36.02[TK]D-FenderYou're "mog_work", you shouldn't be making promises "mog" won't keep ;)
18:36.03hypa7iai like blueberry
18:36.06BladeRunner05error while compiling asterisk 1.2 http://pastebin.com/433347
18:36.09[TK]D-Fender:O
18:36.20*** join/#asterisk Daniel28 (n=daniel_t@141.85.0.66)
18:36.31mog_workman there is no mog
18:36.37mog_worksomebody owns just that nick
18:36.40mog_workit was sadneing
18:36.43mog_worki have mog_home
18:36.47mog_workand mog_work
18:36.51mog_workand mogorman proper
18:36.58fileMatttttttttt
18:37.15filebah
18:37.25mog_workbut imagine it is avail
18:37.26mog_workable
18:37.50hypa7iathat's why i'm hypa7ia with a 7 :(
18:37.53BladeRunner05error while compiling asterisk 1.2 on debian 2.16 *** [pbx_dundi.o] Error 1  see more on http://pastebin.com/433347
18:38.11mog_workpbx_dundi.c:45:18: error: zlib.h: No such file or directory
18:38.19mog_workwhat do you think that means bladerunner ^_^
18:38.23mog_workinstall zlib-dev
18:38.27hypa7iaapt-get install libzlib?
18:38.28mog_workor something like that
18:38.40mog_workwhel he needs headers too hypa7ai
18:38.48hypa7iaya
18:39.21tamp4xi have a question, i have a queue, with a queue time out, and in extensions.conf i tell it to go to voicemail aftgerwards, but it doesnt, justs goes to dead air, anyone have any idea why?
18:39.32BladeRunner05could be this zlib1g-dev ?
18:39.43mog_workyes
18:39.45mog_workthats it
18:40.20*** part/#asterisk santoshr (i=1063@203.199.110.93)
18:40.53BladeRunner05can I do make again or first do make clean and then make ?
18:40.57mog_workso go submit stories to slashdot yay
18:41.01mog_workand no need to make clean
18:41.03mog_workjust make
18:41.52mog_workfile i have no idea as to why you poke
18:42.00mog_worki mean do you have a random timer
18:42.04mog_workthat just does it
18:42.06filenah
18:42.07LostFrogIs there a way to restrict outgoing calls on a SIP device to one call?
18:42.15filebut, I only poke people I like!
18:42.21mog_workindeed
18:42.26mog_workgo submit story
18:42.37MikeJ[Laptop]hehe
18:42.48filesacrifice!
18:42.52malverian[work]Anyone here using Power over ethernet?
18:43.15[TK]D-FenderI do
18:43.17mog_workahhh
18:43.18mog_workno
18:43.35malverian[work]Did you have to make any infrastructure changes to put the switch in place?
18:43.40[TK]D-Fender26 Polycom IP600's on 2 D-Link DES-1523's
18:43.43malverian[work]Eg.. does normal Cat 5 cable work, etc?
18:43.51ikarusmalverian[work]: aslong as it is fully wired, yes
18:44.03[TK]D-FenderCat 5 works, makes sure its all 4 pairs
18:44.07ikarus(all 4 pairs connected)
18:44.19malverian[work]Cool, groovy.
18:44.32ikarusI considered using it at work, but well, the BudgeTone bricks are nothing
18:44.50malverian[work]We have deployed a lot of snom phones here and we have a ground loop problem with our headsets.
18:44.58malverian[work]Snom recommended using POE to get rid of the grounding problem.
18:45.03ikarusmalverian[work]: ground loop
18:45.05ikaruswtf
18:45.23ikarusmalverian[work]: the best way to cure a ground loop is to have all devices grounded to the same group
18:45.26malverian[work]ikarus, We're using two different power sources, one for the headset (battery) and one for the phone (ac adapter)
18:45.44ikarusmalverian[work]: hrm, those should both be perfectly floating
18:45.53ikarusthat shouldn't cause an issue
18:46.43malverian[work]I'm not a very good electrician, so I'm just doing what Snom recommended. If it doesn't work, we'll send it back and try something else.
18:46.47*** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk)
18:47.27malverian[work]We used shielded cables (which was another recommendation) and it helped a little bit.
18:47.38ikarushmmmmm, actually, thinking about it, it might help
18:48.36dos000anyone can take a look and tell me what the hell is going in these logs http://pastebin.ca/29044 ???
18:48.56dos000kind souls please
18:49.15LostFrogDoes call-limit not exist for sip anymore?
18:49.32fileLostFrog: use groups
18:50.48LostFrogGrr.
18:51.04LostFrogI have to set a group for each sip line? :(
18:51.30Gennarosomeone know licencing of 729 codec?
18:51.36Gennaro10 $ for channel
18:51.40LostFrogGennaro: digium.com $10
18:51.48Gennaroyes..
18:51.52LostFrogyes.. $10/channel.
18:52.01Gennarobut if i have 1 trunk of 30 line
18:52.04Gennaro?!?
18:52.18Gennaro10 or 10 x 30
18:52.19Gennaro?
18:52.32ikarusGennaro: you pay per line
18:52.36[TK]D-Fenderfile : can you use variable in the hint exten?  AKA -" EXTEN => _XXX,hint,SIP/${EXTEN} " ?
18:52.36ikarusso 10x30 yes
18:52.42dos000Gennaro, nice try !
18:52.55file[laptop][TK]D-Fender: mmm don't think so...
18:53.04[TK]D-Fenderdang....
18:53.19Gennaroi have a pbx of Edge
18:53.38Gennarothe sayd 10€ for illimited telephone..
18:54.01Gennarobut 1 for 1... and not all
18:54.07Gennaroin the same time...
18:54.11*** join/#asterisk bongfrog (n=winston@c-67-190-81-76.hsd1.co.comcast.net)
18:54.11GennaroARGHH
18:54.34LostFrogGennaro: licenses are dynamic.
18:54.36*** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net)
18:54.44Gennaroso..
18:54.53LostFrogIf you have 5 licenses, 5 channels can user g729
18:54.57LostFroguse
18:55.03dos000what does * do when it gets "SIP Status: 503 Unavailable"
18:55.23ikarushmmmmmm, I wonder if I can make Asterisk count how many connections it has open to a given SIP device (and refuse to open more then one if it is requested by a certain exten)
18:55.25Gennaroso i have 10 telephone
18:55.29dos000i canot heare busy or anything it just keeps quiet
18:55.39Gennaroand i want call with all in the same time..
18:55.46Gennaroi need 10 licence?
18:56.14Gennarobecause each telephone use 1 channel of trunck for call
18:56.16Gennaro??
18:56.27mutdamned telnet!
18:56.30Gennarois right?!?
18:56.34Gennaroor not?
18:56.37mutthey transfered this number to another pri and didn't tell me when they did it!!
18:56.44ikarusGennaro: you need only so many licenses as there are concurrent calls using it
18:57.09Gennaro10 licence for 10 ip phone..
18:57.12Gennaroargh...
18:57.59Gennarothe prob. is that we have 20 pbx
18:58.03Gennarofor 900
18:58.06Gennarophone..
18:58.07Gennaro!!!!
18:58.22Gennaroi'm cry...
18:58.29ikarusGennaro: so use something else
18:58.33mog_workuse a non prop. codec then
18:58.35*** part/#asterisk gorauskas (n=gorauska@66-224-20-131.atgi.net)
18:58.39mog_workgsm is nice and small too
18:58.47Gennaroi tried..
18:59.02Gennaroi done a good config.
18:59.04*** join/#asterisk shmaltz (n=chatzill@69.28.255.210)
18:59.08Gennarowith gms
18:59.21Gennarocalling fwf to set echo..
18:59.33LostFrogok.. I must be stupid..
18:59.47Gennaro?
19:00.02Gennarolostfrog?
19:00.28LostFrogIs this right: I SetGroup(GROUPNAME), then CheckGroup(1)  and then dial.
19:00.40bongfrogno ... just trying to get the 1.2 to make install....   having issues with .version
19:00.45Gennaroi tried with IAX2 and GSM
19:00.56LostFrogDoes this increment the groupcount automatically?
19:00.58Gennaroand int' too good on mobile phone..
19:01.05malverian[work]Pickup() isn't working correctly..
19:01.11malverian[work]It exits non-zero immediately.
19:01.23malverian[work]Strange because it was working before..
19:01.24*** join/#asterisk YoMama (n=tchen@pcp02689850pcs.roylok01.mi.comcast.net)
19:02.46YoMamais there a way to put a call on hold, dial another extension, playback a file or request some input, and then if the input is correct, i'tll do a transfer?
19:03.15*** join/#asterisk jtodd (n=jtodd@ti.fox-den.com)
19:03.26YoMamaeveryone must be eating lunch or something...haha
19:03.41malverian[work]Does Pickup() pay attention to pickup groups?
19:03.52syzygyBSDYoMama: yes, just use the AGI
19:04.12YoMamasyzygyBSD: that's the way to do it eh?
19:04.17malverian[work]syzygyBSD, I think he's asking if you can transfer the call you had on hold.
19:04.31syzygyBSDI won't say it is the only way, but yes, you can
19:04.37YoMamait's just i got this thing set up at my house..when people call in, I want you to announce you who are...then it'll ring a specified extension and playback the caller's name
19:04.40syzygyBSDmalverian[work]: it doesn't really matter
19:04.42tamp4xi have a question, i have a queue, with a queue time out, and in extensions.conf i tell it to go to voicemail aftgerwards, but it doesnt, justs goes to dead air, anyone have any idea why?
19:04.46YoMamaand if u wanna accep tthe call..u press # or whatever
19:04.56YoMamaif not..u just hang up..and the person is routed to voicemail
19:05.52syzygyBSDYoMama: yes, You may also have to use the Manager as well
19:06.34YoMamanah..i want it to be all automatic
19:06.37YoMamai don't wanna use some console app
19:07.02syzygyBSDIt will be all automatic, you just have to have whatever your AGI program is call the manager
19:08.37YoMamaoh i see
19:08.41YoMamai haven't messed with AGI yet
19:09.06syzygyBSDwhat languages do you know?
19:09.17LostFrog!pb?
19:09.19LostFrog~pb?
19:09.21jbotpastebin is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca/
19:09.21YoMamasyzygyBSD: English, French, and Mandarin
19:09.29syzygyBSDlol, programming*
19:09.39bongfroganyone else have issues on make install of 1.2 (cvs tag v1-2-0)  ??????????
19:09.40YoMamaj/k...C, C++, Java, Perl, Pascal, VB
19:10.13syzygyBSDpretty sure C++ and java both have manager and AGI APIs
19:10.26LostFrogDoes this look right: http://pastebin.ca/29050 ?
19:10.51YoMamasyzygyBSD: I'll check it out..but if i gotta code something in Java to make this thing work..then i ain't doing it..i'm sure someone else has wanted to dow hat i'm doing
19:10.53YoMamait ain't rocket science
19:12.04LostFrogexcept for the fact that I shouldn't goto a macro. :)
19:14.51*** join/#asterisk _DAW (n=bob@adsl-222-51-184.msy.bellsouth.net)
19:18.53LostFrogI have the feeling that I'm talking to myself. :)
19:18.59IronHelixyou are
19:19.00IronHelix:)
19:19.15LostFrog[echotest]
19:19.19LostFrogexten => s,1,Hangup
19:19.30LostFroglol
19:20.42malverian[work]Hmm..
19:20.52IronHelixgreat echo test !
19:21.06LostFrogIronHelix: I was tired of hearing myself talk to myself.
19:21.15YoMamaLostFrog: hehehe.i was talkin'
19:21.15bbzim having a problem where my zap channel is answering the call too fast.. where can i make it to where asterisk doesnt answer until a couple of rings?
19:21.31malverian[work]Is there some way to define what context transfers will be in?
19:21.31LostFroglol
19:21.39YoMamabbz: Wait()?
19:21.40[TK]D-Fenderdo a "exten s,1,wait(2)" before you answer
19:21.42LostFrogWait(seconds) before Answer?
19:21.43IronHelixbbz- wait(2) before you answer()
19:21.49[TK]D-FenderAnswerSPAM!!!
19:21.53[TK]D-Fenderwhee!@
19:21.56bbzi tried that
19:21.58YoMamahahaha..too many people knew the answer
19:22.03*** part/#asterisk epoch (n=epoch@octane.breakbeats.org)
19:22.03YoMamabbz: what happened?
19:22.03bbzbut in the cli it still answered, then did a wait
19:22.23bbz[from-pstn]
19:22.23bbzexten s,1,Wait(3)
19:22.33bbzer =>
19:23.02chapeaurougein POTS, does 1 phone number means 1 interface on a TDM card?
19:23.07tamp4xwhy doent queue time out work in command queue?
19:23.28*** join/#asterisk xous (n=xousnet@stnbmb01dc1-233-53.dynamic.mts.net)
19:23.39hypa7iachapeaurouge: it's not quite that simple
19:23.48bbzi call into the system, and immediately    -- Starting simple switch on 'Zap/4-1'
19:23.50bbzoccurs
19:23.57bbzand THEN it goes into wait(3)
19:24.10YoMamaeven tho your wait line is before the answer line?
19:24.24bbzcorrect
19:24.28bbzit is still starting the switch first
19:24.39YoMamahmm
19:25.34LostFrogJust beacuse the switch is started doesn't mean that the phone isn't still ringing, does it?
19:25.40enderthe other side should still be ringing
19:26.10*** join/#asterisk rmorris (n=rmorris@66.46.152.153)
19:26.13*** join/#asterisk archimedes_xyz (i=archimed@adsl-70-247-240-218.dsl.ltrkar.swbell.net)
19:26.26bbzit rings once, then i see <starting simple switch> silence ...... (wait) ....... digital receptionist
19:26.26rmorrisG'day
19:26.31archimedes_xyzgood day all....
19:26.41enderah, you'd have to do Ringing
19:26.50LostFrogI don't have a zap fxo to test it, bbz.
19:26.56bbzender?
19:27.01enderexten => s,1,Ringing();
19:27.07enderexten => s,2,Wait(3)
19:27.12bbzhmm im not familiar w/ that
19:27.13bbzwhat sthat do
19:27.24endermakes the other end hear ringing tones.
19:27.34archimedes_xyzanybody here from arkansas?
19:27.51LostFrogWould anybody admit to that, archimedes_xyz?
19:27.52LostFrog:)
19:28.14archimedes_xyzhehe.... welll uhmmm I am..... But, no, I didn't vote for him...
19:28.26archimedes_xyzor I should say I live here..
19:28.31Ariel_ok anyone from digium where is the list of hardware that the te411p supports like servers, motherboards?
19:28.46LostFrogHey.. If you can get state troopers to find dates for you.. more power to you. :)
19:28.50LostFrogNo pun intended.
19:29.16archimedes_xyzcute..
19:29.37YoMamahahahaha
19:29.57archimedes_xyzno I'm looking for someone to get on the phone/VoIP and explain some stuff when it comes to the config files.. new at this and it's not making much sence...
19:30.01YoMamaarchimedes_xyz: people like me can't go to places like arkansas, much live in them
19:30.15_DAWHello
19:30.32archimedes_xyzwhy is that?
19:30.44YoMamaarchimedes_xyz: 'cause i ain't white and they kill us colored people down there
19:30.49endercan a 1.2.0 Asterisk still successfuly IAX2 to a 1.2.0beta1 system?
19:31.12BeirdoI went to Arkansas once
19:31.20*** join/#asterisk shanky (n=shanky@82.159.214.9)
19:31.26shankyhi, good afternoon
19:31.27Beirdofor all of the 2 minutes it took to turn around and go back to Memphis
19:31.33Beirdomissed our exit
19:31.42db48xI think I'm going to use some loud, brutal music from a symphony about war as my hold music
19:31.44YoMamahahaha
19:31.47LostFrogender: I don't see why not.
19:32.01LostFrog1812 overture, db48x?
19:32.01YoMamadb48x: 1812 overture...
19:32.01_DAWdoes anyone know why my dtmf would break when upgrading from 1.2.0-rc2 to 1.2.0?
19:32.04YoMamahaha
19:32.17db48xyea, or Holst, or Shostakovich come to mind
19:32.22YoMamaif u want something a little less well known...the last 10 minutes of the beethoven egmont overture works well too
19:32.29archimedes_xyz<phone
19:32.32YoMamaShostakovich?  wow..a real classical listener
19:32.41YoMamau even spelled it right...i'm impressed
19:32.58db48xheh, I just went and checked so I could correct it if I needed to though
19:33.16YoMamaI just listend to a live performance of a Shostakovich chamber piece...sounded like the satanic circus had come to town..very dark..very brooding
19:33.25*** join/#asterisk zeedo (n=zeedo@80.68.92.188)
19:33.31db48xcool :)
19:33.35YoMamahad a sick and twisted tonality to it
19:34.01YoMamamy fiancee is a musician and there's this part where there's three notes in succession..and she says that represents the KGB knockin' on the door
19:34.06YoMamafreaky shit
19:34.22db48xyea, I'm not suprised
19:34.26LostFrogI bet most music has at least three notes in succession. :)
19:34.28db48xthey actually imprisoned him at one point
19:34.31YoMamaLostFrog: not like this
19:34.40YoMamaLostFrog: it sounds like someone knocking at a door...
19:35.37db48xmy favorite is the very end of his 11th symphony, "The Year 1905"
19:35.39YoMamadb48x: who was it...umm..one of the great russian composers was commissioned by stalin or lenin or one of those assholes to write a big patriotic piece..
19:35.47shankya simple question: I'm configuring call forwarding, so I use: "exten=>_X.,1,DBget(NUM=DESVIO/${EXTEN})" to get the final number, and then I do "exten=>_X.,1,Dial(Zap/g1/${EXTEN},,tr)" to call it, the problem is that in the final number I don't get the callerid from the ${exten}, it use the first number of the range my telco have assigned to me
19:35.51rmorrisI am an absolute newbie to phone systems. I am wondering what information I require to begin planning either an integration or new installation of asterisk with Meridan Norstar
19:36.05YoMamaand in the last movement..there's all this angst in there...that's a subliminal hint that the composer was far from patriotic
19:36.18YoMamasounds like tires screeching
19:36.24db48xprobably happened with a couple of composers :)
19:36.41YoMamadb48x: where u live?
19:36.43IronHelixrmorris- generally you'd link * and your norstar with pri.  There's a great article by somebody who did something similar floating around here
19:36.55YoMamahouston?
19:37.14db48xfor example, in "The Year 1905", the ending is a bunch of church bells, ominously ringing out the "good news" of the revolution that just succeeded
19:37.16YoMamai wish i were in houston right now..it's 20 effin' degrees out where i am right now
19:37.16db48xyea
19:37.20rmorrisIronHelix, thanks, do you know where I would find it?
19:37.34IronHelixim looking for it now
19:37.38YoMamadb48x: cool
19:37.40rmorrissweet thx
19:37.48db48xit's a balmy 62ish
19:37.54YoMamatexas is one of a handful of conservative states i'd move to
19:38.06YoMamai wanna move to houston or austin..two cities i'm targeting in my job search
19:38.17db48xit's not a bad place, but it's very humid
19:38.20db48xAustin is better
19:38.20shankywhat I have to do to use the original number as caller_id?
19:38.26db48xI loved Dallas when I lived there
19:38.33YoMamadb48x: I hear the misquitoes in houston are the size of cockroaches
19:38.45db48xheh
19:38.51YoMamai hate misquitoes
19:39.13bbzhmm i first saw that word, i thought you were spelling mis quotes =(
19:39.17db48xthey don't usually bother me much, so I haven't really noticed
19:39.47bbzmosquitoes :)
19:40.05YoMamai could stand in a field wearing a ski suit, head to toe with head covering and standing next to a sweaty person wearing nothing covered in honey..and i'll get bit before that person will
19:40.12db48xYoMama: look into the Dallas metro area. the weather is hot but much drier than Houston, and you can watch the summer thuunderstorms moving along the plains
19:40.22db48xlit from within by flashes of lightning
19:40.34db48xhehe
19:40.35bbzYoMama: i think mosquitoes are repulsed by honey ..
19:40.48YoMamadb48x: we get our thunderstorms here...
19:40.53*** join/#asterisk jon335 (n=jon335@ottawa-hs-206-191-39-198.d-ip.magma.ca)
19:40.53YoMamabbz: don't be a smartass :-P
19:40.57bbzlol
19:41.53LostFrogBetter a smartass then a dumbass.
19:42.16*** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)
19:42.30YoMamawerd
19:42.57*** join/#asterisk razu_ (n=razu@213-35-174-231-dsl.prn.estpak.ee)
19:45.14shankymust I use something like SetVar(CALLERID=$ORIGINAL_CALLERID) ?
19:45.43bbzOK, so the wait() solution isnt doing it -- heres my setup : i am setting my current # (w/ SBC) to call forward to my VoIP number -- when a call comes in on the SBC # , it rings the line one time, and * is latching onto it, not letting it forward. I need to have * ignore that .. can i ignore calls coming from a certain # perhaps?
19:46.02rmorrisIronHelix, do you think it's better to trash the Norstar and start from scratch or try to integrate?
19:46.13IronHelixdepends on budget
19:46.18IronHelixand how comfy you are with *
19:46.22tzangerrmorris: I seriously doubt you can trash the norstar quickly
19:46.24tzangeror cheaply
19:46.29tzangerI'm integrated with our norstar... kind of
19:46.51YoMamabbz: wait...are u forwarding the call using your * system, or you had SBC forward your calls?
19:47.00rmorriswhat do you mean kind of tzanger
19:47.11bbzYoMama: SBC
19:47.22bbzYoMama: using *72
19:47.32YoMamabbz: ok..so u don't have your SBC line hooked into your * box
19:47.35rmorristhx IronHelix, I am comfy with software/Unix but this old phone hardware scares me
19:47.41bbzYoMama: yes i do, via zap
19:47.50bbzYoMama: but only using it as a backup line, to place outgoing calls
19:48.03YoMamabbz: eh...u don't have it set up at all for incoming?
19:48.12bbzYoMama: perhaps i should just disable incoming eh?
19:48.12YoMamabbz: didn't you say it said simple switch (Zap/1)?
19:48.29YoMamabbz: when u call forward..sometimes the phone will ring very briefly one ring
19:48.29bbzyes, it has the abliity to answer calls
19:48.32bbzcorrect
19:48.35YoMamabbz: eh..dude..that's why man
19:48.37bbzand that is when  * is grabbing the call
19:48.37YoMamathink about it
19:48.50rmorrisso what I am hearing is that its possible to keep the norstar and add voip phones/headsets what ever .
19:49.00syzygyBSDhas anyone had a problem building 1.2.0?
19:49.24YoMamabbz: if you're going to forward..then make that Zap an outbound only
19:49.27IronHelixrmorris yeah it is, how many extensions total do you hvae?
19:49.36bbzYoMama: sounds like thats the best plan
19:49.37rmorris27
19:49.52Rowterthere is any way to detect when a fax machine answeres an outgoing call. NV_FaxDetect  and the zaptel fax detect seem to only work in calls originated FROM a fax machine, not for calls ANSWERED by a fax.
19:50.04YoMamabbz: NOW i get what you're trying to do...
19:50.09rmorrisbasically we are out and management is asking me to look into * as a solution
19:50.12bbzYoMama: so you agree, disable incoming?
19:50.19YoMamabbz: u wanted to wait long enough so the call forward would kick in...and forward to the VoIP #
19:50.29bbzcorrect
19:50.30YoMamabbz: instead...it's snagging it and answering it on your Zap channel
19:50.34bbzbingo
19:50.34YoMamahaha..u silly goose
19:50.43bbzsuggestions then?
19:50.45IronHelixout?  27 extensions isnt that bad for a deployment.  Do a test depoyment with ~5 users and once you work out the kinks dive in
19:50.48YoMamaeh..lemme think
19:51.07IronHelixand if management is committed to dumping the norstar and is willing to pay for the gear, then you're all set
19:51.45rmorriswhen you say gear IronHelix  do you mean like T1 PRI cards and boxes?
19:51.58YoMamaso exten => s,1,Wait(5); exten => s,2,Answer() doesn't work?
19:52.02bbznope
19:52.03IronHelixwell i assume you have a PRI for incoming lines
19:52.04bbzthe wait doesnt do it
19:52.05IronHelixor do you use analog
19:52.13shmaltztzanger, you around?
19:52.18shmaltz~seen tzanger
19:52.20jbottzanger is currently on #asterisk.  Has said a total of 91 messages.  Is idling for 5m 51s
19:52.25Beirdo!seen slePP
19:52.33Beirdodamn you finger
19:52.34rmorrisWe have PRI I think .. I need to better understand the front end of our existing system
19:52.38Beirdo~seen slePP
19:52.39jbotslepp is currently on #asterisk
19:52.46BeirdoI missed the key
19:52.58tzangershmaltz: yes
19:53.01_DAWhas anyone had a problem with dtmf and 1.2.0?
19:53.02shmaltzBeirdo, its right next to each other
19:53.06BeirdoI know
19:53.07IronHelixrmorris go and check it out, call the telco and ask what services are on your account if you have to
19:53.09BeirdoI missed :)
19:53.17IronHelixif you do have PRI, get all the settings while you're on the phone
19:53.28shmaltztzanger, after slackware which one is a disto you consider stable enough and you would use?
19:53.47archimedes_xyz<-- back... sorry
19:54.02rmorrisright, and if it is PRI IronHelix then it's just cards right? and the server config? . and of course phones for everyone
19:54.26tzangerslackware. :-)  Maybe Arch?  I dunno.  I just use slackware
19:54.26rmorrisstill love to read that Norstar Link document you were talking about IronHelix
19:54.32tzangerrmorris: which document's this?
19:54.32LostFrogjust.. lol
19:54.35archimedes_xyzso anyway's anybody here willing to spend some time with me and help explain how the config file works?... I'm using Signate's installation CD...
19:54.53IronHelixrmorris im trying to find it
19:55.11rmorrisIronHelix,  was saying there is a document about linking a norstar with * someplace tzanger
19:55.14tzangerIronHelix: please let me know too
19:55.30tzangerrmorris: I saw one on the wiki using four PRI spans but it really didn't do much in the way of anything fnacy
19:55.44tzangerall norstar calls went into the PRI spans, and * did telco routing or VOIP routing
19:55.57bbzYoMama: any other ideas?
19:55.57*** join/#asterisk Damin (n=damin@nucleus.nacs.net)
19:56.05Daminarchimedes_xyz: Have you considered calling Signate for support?
19:56.05shmaltztzanger, which distro?
19:56.08*** join/#asterisk chapeaurouge (n=chap@85.201.80.249)
19:56.19xousIs it possble use Vonage VoIP (basic Unlimited) without doing Internet --> Vonage ATA --> Asterisk, or are they still strict on that?
19:56.22tzangershmaltz: ?
19:56.27YoMamabbz: http://lists.digium.com/pipermail/asterisk-users/2005-January/086203.html
19:56.34archimedes_xyzwell yeah but I'm afriad they'll tell me it's not support I need but tutoring.. and they would be right..
19:56.36YoMamabbz: it doesn't look right tho...
19:56.45bbzYoMama: yea i already saw that one
19:56.56Daminarchimedes_xyz: Consider reading the Asterisk book at http://asteriskdocs.org
19:56.59bbzyour right about the single fast ring -- thats whats messing it up here.
19:57.12KattyDamin: did you authorize this cold weather?
19:57.16marc324how do you install webvmail if the asterisk server is not the http server?
19:57.21rmorrisyea, there is quite a gap in my understanding of telcom stuff. I understand the line types, but not how they integrate with our norstar. It would be great to find a document explaining even that!!
19:57.26bbzYoMama: for the sake of testing, what do i do to make that fxo only able to make calls? do i change the signaling type?
19:57.40YoMamabbz: then i dunno..never tried what you're doing...just disable incoming and see what happens
19:57.50YoMamabbz: no..just take out the Answer()
19:58.00YoMamadon't have any "s" lines for the Zap
19:58.07marc324ne1 has installed webvmail?
19:58.10shmaltztzanger, if you can't use slackware for whatever reason which distro would you use?
19:58.11LostFrogOr set the context to something bogus.
19:58.43LostFrogshmaltz: debian, redhat, fedore, centos all work fine.
19:58.47shmaltztzanger, or you dont' even run slackware? I migth have the wrong guy
19:58.49tzangershmaltz: I answered already...  it'd be a slackware deriviative (arch I think it's called?)  but why wouldn't you be able to use slackware?  honeslty go for what you're comfortable with
19:58.49LostFrogfedrora, even.
19:58.55tzangerI run slackware
19:59.02IronHelixxous- yeah you can do thta with a FXO card of some kind, vonage wont even know.  however i (and probably almost everybody here) would advice you to run, not walk, away from vonage before you get committed to one of their *NON PORTABLE* numbers
19:59.13YoMamabbz: u got to separate the Zap and the VoIP line into different contexts
19:59.18bbzYoMama: ehh, the way its setup, zap/sip are all handled w/ the same context
19:59.19shmaltzLostFrog, well debian is just plain stupid it tries to hold a tight grip on my installs I *hate* that
19:59.20archimedes_xyzI'll checlk it out... I already have the VoIP Telephony with Asterisk book, but it leaves some questions as too why open..
19:59.21IronHelixthere are plenty of companies that will sell a similar *-freindly service and have better support
19:59.24YoMamabbz: then point your zap to a context that doesn't have a Answer() line in it
19:59.26shmaltzso is fedora
19:59.27bbzah
19:59.28YoMamabbz: well, change that :)
19:59.29NuggetI run slackware too.  Asterisk doesn't care what unix you use, it doesn't even have to be Linux if you don't care about zaptel.
19:59.33bbzYoMama: good call, ill do that
19:59.37KattyNugget: slacker.
19:59.40Nuggetindeed
19:59.43tzangershmaltz: almost every distro does that except for slackware and LFS really
19:59.44shmaltzbesides fedora just doesn't realy work very stable
19:59.56YoMamadude..u know you're getting old when u lean back to stretch and about 5-6 things crack in your back and chest
20:00.04tzangerthey must because they use dependency tracking and rely too hard on packages
20:00.06KattyNugget: i don't slack.
20:00.07shmaltztzanger, have you ever gotten grub working with slackware?
20:00.11*** join/#asterisk wolfson (n=ggggg@208.25.254.120)
20:00.12endersome people find package management to be a good thing.
20:00.12KattyNugget: seems all horribly confusing, sorta.
20:00.13tzangerYoMama: :-)  my septum cracks when I stretch... it's not good
20:00.14archimedes_xyzis everybody having to rebuild the zaptel drivers every time the server reboot's?
20:00.16tzangershmaltz: yes of course
20:00.18tzangerit's not hard at all
20:00.21shmaltztzanger, I nuderstand, but they mess up my world
20:00.26KattyNugget: probably because i've not used unix but for just a little while
20:00.30tzangershmaltz: which is precisely why I use slackware :-)
20:00.37shmaltztzanger, it's not hard if you can boot into a working install
20:00.44LostFrogarchimedes_xyz: rebuild them????
20:00.44shmaltzbut I can't I get tons of errors
20:00.46LostFrogWTF?
20:00.57LostFrogyou have to *reload* them.
20:01.06YoMamatzanger: your septum??? eh...isn't that in your nose?
20:01.19xous(IronHelix): Can you recommend a Candian VoIP provider that allows Asterisk?
20:01.36archimedes_xyzyeah recompile... it's like the card isn't there all of the sudden.. recompile them, modprobe poof thier back
20:01.39shmaltztzanger, I have setup plenty slackware system, but this one got me stuck
20:01.40archimedes_xyzTDM400
20:01.43LostFrog~tell xous about wiki
20:01.45shmaltzI got no clue why
20:02.21LostFrogarchimedes_xyz: modprobe doesn't work?>
20:02.32IronHelixnot personally but there is a very large and extensive list available here http://www.voip-info.org/wiki-VOIP+Service+Providers
20:02.34archimedes_xyznot untill I recompile them..
20:02.36*** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br)
20:02.48tzangershmaltz: howso
20:02.53xousthx :)
20:02.55LostFrogdid you do 'make install', archimedes_xyz?
20:03.02tzangerYoMama: no septum is where your ribs come together in your chest
20:03.10shmaltztzanger, I'm trying to instal it onto a via epia system
20:03.12tzangerunless I'm very wrong, which is likely :-)
20:03.15tzangershmaltz: ok
20:03.23LostFrogseptum is in your nose.
20:03.25archimedes_xyzyeah..
20:03.35LostFrogA deviated septum doesn't have to do with your chest. :)
20:03.38shmaltzand it's just giving me toooo many input output errors, cannot find this cannot find that, not a valid directory
20:03.38archimedes_xyzand cvs updated..
20:03.40tzangerslackware 10.2 uses i486 instructions and i686 ordering, which should be fine
20:03.44tzangerLostFrog: hmm okay :-)
20:03.57tzangerwhat's the bone/cartilege part in your chest where your ribs come together?
20:04.01bbzYoMama: lol, i changed context, it goes =>   -- Starting simple switch on 'Zap/4-1'
20:04.08tzangershmaltz: i/o errors?!
20:04.09bbz<PROTECTED>
20:04.11tzangerpaste some
20:04.15shmaltztzanger, I just downloaded 10.2 and installed it, by far it works better then any of the other ones, but still tons of trouble
20:04.23YoMamabbz: anytime the phone rings..it'll do that switch thing..but it won't answer the call anymore
20:04.32shmaltztzanger, yes
20:04.34YoMamabbz: have it going to a context with nothing in it
20:04.39bbzYoMama: did that
20:04.40bbzand that happened
20:04.49*** join/#asterisk juanjoc (n=juanjoc@200.73.189.82)
20:04.55bbzsaid it couldnt find anything, so died
20:04.56YoMamabbz: eh...dude...pastebin your extensions.conf
20:05.17enderbbz: what exactly are you trying to do?
20:05.23LostFrogSernum.
20:05.25LostFrogSternum
20:05.32bbzYoMama: in the zap/4 context=from-zap, in extension i put ,
20:05.32bbz[from-zap]
20:05.32tzangershmaltz: i/o errors on what device?  hdd?
20:05.39tzangerLostFrog: sternum that's it!
20:05.41tzangerthank you
20:05.41bbzand nothing there.. thats all thats relivant
20:06.35LostFrogbbz.. just put 'context=garbage' in your sip.conf
20:06.46LostFrogThat should stop all incoming calls.
20:06.50bbzender: ive got callforwarding , sending my normal sbc # to my voip #, i got an fxo card w/ that sbc line connected to it -- when a call is placed to the forwarded number, the zap picks up the call, instead of letting it forward
20:06.55shmaltztzanger, yep
20:07.04tzangershmaltz: that's your HDD dying, it's got zero to do with the distribution
20:07.16LostFrogbbz: I mean in zapata.conf
20:07.30enderbbz: sounds like the forward isn't happening at the telco.
20:07.31bbzcan i paste 4 lines?
20:07.36tzangershmaltz: something like "I/O error on /dev/hda: CrcNonRecoverable: reading block 35252623"
20:07.39shankya simple question: I'm configuring call forwarding, so I use: "exten=>_X.,1,DBget(NUM=DESVIO/${EXTEN})" to get the final number, and then I do "exten=>_X.,1,Dial(Zap/g1/${EXTEN},,tr)" to call it, the problem is that in the final number I don't get the callerid from the ${exten}, it use the first number of the range my telco have assigned to me
20:07.45YoMamaender: it is..it just rings the phone once and then goes through
20:07.46bbzender: it happens, because i go yank the phone line, and theres no problems
20:07.46shankywhat I have to do to use the original number as caller_id?
20:08.10bbzhttp://pastebin.com/433458
20:08.11shmaltztzanger, nope
20:08.14enderwell, the forwarding can happen farther up the line, at the telco.  So that the line doesn't ring at all, just gets redirected at the telco.
20:08.18bbzresult of putting - context=garbage
20:08.36tzangershmaltz: post an example then please
20:08.51bbzender: you suggest i call phone company and tell them to forward it @ the office?
20:08.54marcus2how do i kill a channel, from the console?
20:09.10YoMamabbz: haha...interesting
20:09.15YoMamabbz: it really wants to answer that call
20:09.26bbzYoMama: like a kid wants candy..
20:09.35YoMamabbz: don't use default context
20:09.42YoMamabbz: u got stuff in [default]
20:10.04bbzyes
20:10.04shmaltztzanger, I will have to find it
20:10.09bbzinclude => ext-local
20:10.10shmaltzand then I'll post it
20:10.21marcus2i think i found a bug in meetme :(
20:10.22YoMamabbz: or try this in [from-zap]: s,1,Hangup()
20:10.25YoMamai wonder what that'll do
20:10.30YoMamaexten => s,1,Hangup()
20:10.32bbzYoMama: stabbing @ the wind now, but ok ;)
20:10.46marcus2if a user hangs up while the 'you are the only user in this conference' announcement is playing, the conference doesnt see the hangup
20:10.49YoMamabbz: otherwise..get rid of default..change it to something else
20:10.52*** join/#asterisk brc_ (n=Brian@pdpc/supporter/basic/brc)
20:11.08enderbbz: you can, but that costs money per forward I think.
20:11.14bbzYoMama:    -- Hungup 'Zap/4-1'
20:11.15bbz:)
20:11.22tzangershmaltz: ok, but it really doesn't sound anything at all like a distro-specific problem.  It sounds like your HDD is dying, or there is something grossly misconfigured (i.e. the IDE controller says it can do UDMA5 but it's failing, or you've convinced it you've got one of those 80-pin IDE cables but only have a 40-pin...)
20:11.27bbzender: gotcha.
20:12.12shmaltztzanger, I want to hear more about that 40 pin vs 80 pin
20:12.16rmorristhx for the help IronHelix  I found 4 or 5 pdf to read, so I'll go piss off and read them for now and come back more informed. cheers
20:12.22shmaltzI got a slim toshiba HDD
20:13.22*** join/#asterisk NirS (n=nirs@84.94.51.176.cable.012.net.il)
20:13.26NirShello all
20:13.47*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
20:14.09IronHelixhi
20:14.12tzangershmaltz: if you do not have one of the 80-pin IDE cables you cannot put your drive into the UDMA4/5 modes.  The controllers are supposed to detect this but you may be forcing the controller into 80-pin mode, or you r controller may be mistakenly reporting 80-pin mode
20:14.20Renacordoes anybody know how to increase the maximum number of messages a voicemailbox can have?
20:14.39shmaltztzanger, is this a config in the os issue? or in the bios?
20:14.44tzangeris it a normal 80-pin IDE cable or the older-style 40-pin ones?  The 80-pin ones have twice as many wires in the same physical width (the extra wires are common signals)
20:14.46YoMamabbz: so...did it work?
20:14.52tzangershmaltz: could be BIOS/controller
20:14.58*** join/#asterisk sysdebug (n=sysdebug@200.163.193.247)
20:15.01archimedes_xyzanybody using mIRC here?.
20:15.04bbzYoMama:    -- Hungup 'Zap/4-1'
20:15.06tzangerlinks can be told to ignore it but does not do so with slackware standard install
20:15.12shmaltztzanger, I think its a 40 pin but I'm not sure
20:15.38YoMamabbz: i'm gonna be shocked if it interprets just a Hangup() as an inferred "pick it up and hang it up real quick"
20:16.07bbzYoMama: all i can assume is that when it does the simple switch -- it means zaps in control now
20:16.33NirSany here using AMP 1.10.009 ?
20:16.40paryldoes anyone know what i can do about this?  "pbx_dundi.c:1409: error: âZ_OKâ undeclared (first use in this function)"
20:16.56*** join/#asterisk blop (i=drink@openbeer.be)
20:17.23Renacordoes anybody know how to increase the maximum number of messages a voicemailbox can have?
20:17.49shmaltztzanger, what command can I execute to see what linux *thinks* it is?
20:18.18tzangershmaltz: in the bootup (use dmesg after booted) and look for the block of text that describes how it found the drives
20:18.50parylRenacor: maxmsg=1000?
20:19.01shmaltztzanger, thank you
20:19.09tzangerdon't thank me until we've fixed it
20:19.39YoMamabbz: look in your sample zapata.conf under immediate
20:19.45*** join/#asterisk Gennaro (n=Gennaro@ppp-62-10-136-122.dialup.tiscali.it)
20:19.53Gennarohi
20:20.03bbzimmediate?
20:20.14bbzah, its no
20:20.15Gennarosomeone know how play gsm without installa addictional software on fc4
20:20.18Gennaro???
20:20.48YoMamabbz: dude..i have no clue...i must be missing something because i KNOW there's a way to define something as an outbound only trunk
20:21.17bbzyea
20:21.25bbzthats my next step, is searching for making it 100% outbound
20:21.32Renacorparyl: I think thats for the maximum length of a voicemail, Im talking about the number of voicemails
20:21.59LostFrogHow can I log everything that I see on the console to a file?
20:22.06archimedes_xyzRenacor: isn't that limited by quota or something like that?
20:22.12LostFrogI tried 'asterisk -r >/tmp/log.txt &'
20:22.14Renacorparyl: nm I think your right =)
20:22.53LostFroghmmm tee works.
20:23.21bbzLostFrog: ehh i thought it does this alreayd in /var/log/asterisk
20:23.22Gennarosome know a player?
20:23.36tzangerhmm
20:23.59LostFrogNeither full or event_log is the same thing.
20:24.03LostFrognor
20:24.20bbzi usually look @ full
20:24.27tzangeroff-topic question.  Can I return a column in SQL that is something like SELECT (somecolumn == somevalue) AS val_exists FROM sometable where someothercolumn=someothervalue; ?
20:24.29LostFrogI like the verbosity.
20:24.41shmaltztzanger, all I can see is udma(33)
20:25.04LostFrogUDMA(33) only needs a 40-pin cable.
20:25.12LostFrog133 and up needs an 80.
20:26.22iCEBrkrLostFrog: WTF are you trying to do with that SQL?!
20:26.24tzangeryup UDMA33 is ok with any IDE cable.  UDMA66/100/133 needs 80-pin
20:26.27iCEBrkrOMGZZZ!@!_)*@!#_^*
20:26.54tzangeriCEBrkr: you mean me?
20:27.00iCEBrkroops
20:27.01*** join/#asterisk bhickey (n=bhickey@213-202-134-171.bas502.dsl.esat.net)
20:27.02shmaltztzanger, you da man
20:27.07iCEBrkrtzanger: oops, yea you :P
20:27.13tzangershmaltz: ??  I didn't fix anything
20:27.13shmaltzeverybody clap for tzanger
20:27.25shmaltztzanger, but you pointed me in the right direction
20:27.28tzangeriCEBrkr: because I don't want to try and process a list in Asterisk
20:27.38shmaltzall I had to do was tell the bios don't do ultra dma
20:27.43iCEBrkrok?
20:28.30tzangeriCEBrkr: so I want to give the DB a query that reutrns a voicemailbox and a true/false for whether a specific caller ID is ok to autologin, given the caller id number and ${EXTEN}
20:28.41tzangershmaltz: linux can re-enable that too if need be
20:28.52LostFrogtzanger: it can?
20:28.57tzangerLostFrog: yep
20:29.02*** join/#asterisk voipme (n=voipme@159-134-98-117.b-ras1.chf.cork.eircom.net)
20:29.03iCEBrkrtzanger: That'd probably be something like a sub-select.
20:29.04tzangersame with local APIC
20:29.10LostFrogI thought it could just enable/disable DMA.
20:29.11tzangeriCEBrkr: yeah that's kind of what I thought
20:29.14tzangerLostFrog: nope
20:29.20tzangerthe driver can do whatever it likes
20:29.22LostFroghdparm?
20:29.27LostFrogooh..
20:29.28LostFroghmm.
20:29.37iCEBrkrNot that i know the syntax of a subselect off the top of my head.
20:30.42voipmeevening all
20:31.03iCEBrkrDamn, I ready to fall asleep
20:31.36voipmeanyone here know how many digits eircom (ireland) send for ddi on incoming calls on pri or isdn?
20:31.36*** join/#asterisk fulgas (n=fulgas@a81-84-117-79.cpe.netcabo.pt)
20:31.38fulgashi
20:31.56tzangerLostFrog: no hdparm is kind of a userland tool for screwing with dma, multi-irq, etc
20:32.05tzangerthe driver will have specific parameters to tune if you want to although I've never had to
20:32.47NirSvoipme, from my last installation in ireland, which was around 4 weeks ago, it depents alot on your location
20:33.06NirSin dublin, they send in the full local number
20:33.11NirSin other locations it may vart
20:33.13NirSvary
20:33.40voipmethanks nirs
20:33.48voipmeill just have to check when i plug in
20:34.12NirSexactly
20:34.17voipmeyou installing many over here
20:34.32NirSwell
20:34.32voipmegood forbid eircom actually have a standard
20:34.40YoMamaso...i noticed something interesting about the voicemail...u can change your password..but it doesn't rewrit ethe file..and when the system reboots..your changed password is gone
20:34.40NirSI installed one at Hosting365 about a month ago
20:34.56NirSand we hope top extend our business in Ireland in the next few months
20:34.57YoMamathis is a stupid question..but what's up with that?
20:34.58*** join/#asterisk heison (n=heison@216.126.67.39)
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20:35.01*** part/#asterisk bongfrog (n=winston@c-67-190-81-76.hsd1.co.comcast.net)
20:35.17justinuYoMama: permissions problem?
20:35.20*** join/#asterisk jontow (i=jontow@ws.woflsys.net)
20:35.45voipmenirs, send me your details, i know a few people that might be interested, im just doing this one for a friend
20:35.48YoMamajustinu: so it's supposed to write to the voicemail.conf file?
20:36.07YoMamajustinu: my asterisk server is running with the right permissions
20:36.09*** join/#asterisk tracinet (n=tracinet@216.242.235.2)
20:36.40*** join/#asterisk nagl (n=nagl@213.235.241.6)
20:36.47ooglewhat is the difference between a res addon and an app addon?
20:37.42*** join/#asterisk xbmodder_ (n=xbmodder@dsl092-012-130.sfo1.dsl.speakeasy.net)
20:37.52xbmodder_I am having an error with teliax:
20:37.53xbmodder_http://pastebin.ca/29061
20:38.26xbmodder_For some reason all of the providers autmatically go to the "default" context even though I specify otherwise
20:40.34justinuYoMama: my system updates voicemail.conf
20:41.28*** join/#asterisk tracinet (n=tracinet@216.242.235.2)
20:43.54*** join/#asterisk eee3 (n=piaergnj@adsl-70-128-78-22.dsl.stlsmo.swbell.net)
20:44.39YoMamajustinu: eh...lemme see if i set the file to 777..then if it still doesn't do it, i'll know it's not a permissions problem
20:48.40*** part/#asterisk jon335 (n=jon335@ottawa-hs-206-191-39-198.d-ip.magma.ca)
20:50.20shmaltztzanger, I hope you like compliments ;)
20:50.28tzanger?
20:50.34shmaltztzanger, you married?
20:51.12tzangerno I'm not married yet
20:51.33shmaltzhmmmm, then you might not yet have developed a nag for compliments
20:51.40shmaltzbut anyhow, you da man
20:53.59mog_workhey people submit stories to slashdot , grand prize one muffin
20:54.03YoMamaanyone know why changing my vm password thru dialtone isn't changing it in my conf file?
20:54.06*** join/#asterisk zotz (n=zotz@24.231.47.168)
20:54.24shmaltzYoMama you have - before the password?
20:54.33shmaltzis it changing it at all?
20:54.40YoMamashmaltz: it's chagning..but it's not changing int he file
20:54.57shmaltzwhat if you reload, will it take the old password or the new one?
20:55.07YoMamathe old one
20:55.12YoMamait reads it from the conf file
20:55.34shmaltzYoMama you sure asterisk got permission to write to the conf filee?
20:55.35_DAWYoMama - mine changes in the file..
20:55.38tzangershmaltz: glad it works for you :-)
20:56.02shmaltztzanger, thanks
20:56.07YoMamashmaltz: i had it set to 666
20:56.19YoMamajust to double check
20:56.34shmaltzYoMama set it to 777
20:56.47YoMamak
20:56.50YoMamalemme give it another shot
20:56.53shmaltznow test it and tell me
20:57.28YoMamanope
20:57.56_DAWexit
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20:58.31shmaltzYoMama, nope what?
20:58.38shmaltzstill not changing?
20:58.46YoMamano it's not changing in the file
20:58.52shmaltzyou running asterisk as root?
20:58.59YoMamayes
20:59.02YoMamais that a no no?
20:59.29*** join/#asterisk supaigtr (n=yurplsl@152.53.16.10)
21:00.09shmaltzYoMama, for some ppl
21:00.18shankywell, the solution was just add a line like this: "exten=>_X.,2,SetCallerId(${EXTEN})
21:00.21shmaltzbut if you don't run as root, then that might be the reason for it
21:00.27shankyafter the dbGet
21:00.30YoMamashmaltz: well, i don't care...if it's running as root...it should be able to do anything
21:00.30shmaltzYoMama, what version you running?
21:00.31shankythanks anyway
21:00.32*** join/#asterisk xbmodder1 (n=xbmodder@dsl092-012-130.sfo1.dsl.speakeasy.net)
21:00.37YoMamaschmltz: 1.2.0 rc1
21:00.54shmaltzYoMama, pleaseeeeeeeeeeeeeeeeeeeeeeeeeee, upgrade to stable
21:01.03shmaltzwhich is 1.2.0
21:01.14YoMamathey've released it now?
21:01.28YoMamaaaaah
21:01.30YoMamathey have
21:01.36YoMamahaha...man
21:01.39YoMamai should read the topic
21:01.44YoMamak
21:03.55*** join/#asterisk SplasPood (i=nobody@paravolve.net)
21:08.00*** join/#asterisk Sobakai (n=jmwoodga@45.e6.d12c.cidr.airmail.net)
21:08.52docelmoAnyone know what the TAG is for 1.2?
21:09.03docelmojust 1-2
21:09.22*** join/#asterisk gordonjcp (n=gordonjc@cpc3-broo2-5-0-cust232.renf.cable.ntl.com)
21:09.51gordonjcpevening
21:10.00IronHelixgreetings
21:10.27juanjocdocelmo: It's v1-2 for the branch and v1-2-0 for the release
21:10.28gordonjcpdoing a bit of googling right now for Swissvoice IP10S phones
21:10.34gordonjcphas anyone tried them here?
21:13.41justinubefore anyone upgrades to 1.2 stable, please make sure this bug won't affect you!
21:13.47justinuhttp://bugs.digium.com/view.php?id=5780
21:13.58xbmodder1does anyone know why asterisk is using the wrong context?
21:14.10justinuit's a super easy patch, but it'll cause havoc if you're using sip/g729
21:14.11denonhuh. that patch wont apply to 1.2?
21:14.29justinuyeah, very easy
21:14.39justinubut some people may not know or have the skillz to find the problem :)
21:14.44denoncause the patch wont apply for me heh
21:14.50Nivexjustinu: it only affects that codec?
21:15.09justinunivex: it seems to only apply if g729 is offered as a codec, not nesecarily selected
21:15.14*** join/#asterisk fiftyCal (n=b@69-160-145-156.ontrca.adelphia.net)
21:15.22*** join/#asterisk emcnabb (n=emcnabb@hq-nat2.gurulabs.com)
21:15.29Nivexalrighty, thanks for the headsup.  I don't think I'll be affected.
21:16.36*** topic/#asterisk by denon -> Asterisk 1.2.0 has been released! -- ftp.digium.com (RFC2833 users see bug 5780) || http://www.asterisk.org
21:16.37LostFroglol. add four characters..
21:16.43LostFrogThat is a easy patch.
21:17.13IronHelixwell they said it wouldnt be totally stable until 1.2.1... guess things are on schedule :)
21:19.46LostFrogI'm not moving to 1.2.0 until I am in the office. :)
21:19.56*** join/#asterisk gladrim (n=bacs@82-41-204-90.cable.ubr11.edin.blueyonder.co.uk)
21:20.16znoGgod damnit. asterisk is detecting distinctive ring patterns when it feels like it
21:20.43tzafrir_laptopany news about bristuff and 1.2?
21:21.02*** join/#asterisk insurin1 (n=insurin@82-42-19-166.cable.ubr01.knor.blueyonder.co.uk)
21:21.31insurin1can anyone give me a little help?
21:22.11tzafrir_laptopinsurin1, state your problem, for starters
21:22.30insurin1ok, I have just installed asterisk on fc3
21:22.40justinustate your business
21:22.56insurin1I only have config files in the config directory and they are sample files
21:23.08insurin1can I just edit these and use them?#
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21:23.24justinuyes
21:23.35AmyTekhi
21:23.43tzafrir_laptopyes. generally you 'reload' asterisk for changes to take effect
21:23.49*** part/#asterisk diclophis (n=diclophi@adsl-69-238-124-226.dsl.pltn13.pacbell.net)
21:23.51insurin1thanks
21:23.55tzafrir_laptopthere are some shortcuts
21:23.57znoGthat's it?
21:24.04insurin1lol 1 more thing
21:24.07tzafrir_laptop(and for zapata you need a full restart)
21:24.13znoGhopefully it's not as complicated as the last one
21:24.23insurin1haha
21:24.59insurin1erm I just want to test it, do I create a couple of users in the sip.conf then I can try it ouot with a softphone?
21:25.25justinuyeah
21:25.30xbmodder1wtf.
21:26.00insurin1is that yeah for me ^^ justinu?
21:27.03^Howlerinsurin1: Yes, that's all you need to do. That's what I am doing right now. Just try to keep the phones and the server on the same network (strange problems with NAT and SIP)
21:27.11justinuinsurin1: yes
21:27.42insurin1thanks guys, I am on the same network so hopefully no problems
21:28.03stbainoh
21:28.18stbainand insurin1: having a softphone around really helps you when you're testing a dialplan
21:28.24stbainso keep it installed
21:28.37marc324is it possible to store voice messages at a remote server
21:29.04insurin1which is the best softphone to use on windows?
21:29.26justinueyebeam if you afford it
21:29.33bhickey-lite
21:29.45^Howlerinsurin1: I'm using X-lite. It's decent and it's free
21:29.53FrostedDoes anyone know why agi_channel has changed to not include IAX2 users? the same happens in the csv logs, only happens with REALTIME though, a bug maybe? :)
21:29.53bhickeyX-lite
21:30.00*** part/#asterisk Bentley (n=Bentley@S0106000f3d016dd2.cg.shawcable.net)
21:30.11insurin1X-lite, that is not sip is it?
21:30.24insurin1do I necessarily need sip?
21:30.31justinuxlite is fine too
21:30.39^Howlerinsurin1: it is SIP
21:30.51justinuinsurin1: you could ise iax
21:30.52justinuuse
21:31.12insurin1i'll probably try them all,
21:31.49insurin1I work in a college and we spent a load on Cisco ip phones and we only use them to make phone calls
21:31.57*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
21:32.01^Howlerinsurin1: you can look at asteriskguru.com/tutorials ...They have configurations for a few softphones and other good stuff.
21:32.10insurin1im on that now^
21:32.45^Howlerfair enough =)
21:37.43Rawplayerwhat am i missing here?
21:37.44Rawplayerications.o autoservice.o db.o privacy.o astmm.o enum.o srv.o dns.o aescrypt.o aestab.o aeskey.o utils.o plc.o jitterbuf.o dnsmgr.o devicestate.o netsock.o slinfactory.o ast_expr2.o ast_expr2f.o cryptostub.o buildinfo.o editline/libedit.a db1-ast/libdb1.a stdtime/libtime.a -ldl -lpthread -lncurses -lm -lresolv   -lssl
21:37.49Rawplayer/usr/bin/ld: cannot find -lssl
21:37.52Rawplayercollect2: ld returned 1 exit status
21:38.15mog_workneed libssl
21:38.31Rawplayeri installed ssl though
21:38.36mog_workssl-dev
21:38.50Rawplayerhmm
21:38.55Rawplayerlib wasnt there
21:40.13tzafrir_laptopmog_work, you're either missing a -L/path_to_install_of_openssl
21:40.15*** join/#asterisk eKo1 (n=Miranda@63.245.92.221)
21:40.24tzafrir_laptopor openssl-devel or something
21:40.51mog_workrawplayer is
21:41.10hellopWhy do I have to reset zap every day now to get my zaptel card to work?
21:41.11tzafrir_laptopwhat distro? what version of asteris?
21:41.37eKo1hellop:  what zaptel card?
21:41.47hellop100P
21:41.49tzafrir_laptophellop, what do you mean by "reset"?
21:41.52*** join/#asterisk sgtpepper (n=nicolas@200-47-10-93.comsat.net.ar)
21:41.55sgtpepperhi
21:41.58sgtpepperanyone?
21:42.01IronHelixhi
21:42.02eKo1probably means reboot
21:42.04tzafrir_laptopand what "100p" x100p?
21:42.19sgtpepperIronHelix, I need help
21:42.20tzafrir_laptopeKo1, it could also mean ztcfg
21:42.24sgtpepperwith a very dumb issue
21:42.27IronHelixdon't we all?
21:42.28IronHelix:)
21:42.29IronHelixwhats up
21:42.35hellopone sec phone
21:42.48sgtpepperi have a fxo gateway on IP 1.1.1.1
21:43.01IronHelixk
21:43.14sgtpepperi wan't to route outgoing calls through that fxo gateway (which is connected to pstn)
21:43.19IronHelixk
21:43.22eKo1i reboot my * box everyday. if not, my quad e1 card starts bitching.
21:43.54eKo1sgtpepper: : ok then
21:44.00IronHelixwhat doesnt work sgt?
21:44.11IronHelixthe first problem i see- 1.1.1.1 isnt a good ip address that you should be using
21:44.20IronHelixbut keep going
21:44.32eKo1exten => s,1,dial(sip/1.1.1.1/<insert number here>)
21:45.02sgtpepperso i wrote in extensions.conf exten = _93XXX,1,Dial(oh323/1.1.1.1)
21:45.11sgtpepperits an h323 fxo gw
21:45.16eKo1ugh
21:45.19eKo1that sucks
21:45.24sgtpepperso i want to dial 93+pstnnumber
21:45.34hellopI think it's an X100P
21:45.45sgtpepperand 93+pstnnumber be dialed on the rewmote fxo gw
21:45.45eKo1well, you're not passing the number
21:45.49eKo1as far as i can see
21:45.51sgtpepperthe 1.1.1.1 its an example
21:45.58IronHelixah ok
21:46.05IronHelixdo this
21:46.08hellopTo "reset" it, I rmmod wcfxo, and zaptel, modprobe them back, and zcfg -vv
21:46.22hellopThis box was running fine for about 8 months now this..
21:46.22*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
21:46.29IronHelixexten = _93XXX,1,Dial(oh323/1.1.1.1/${EXTEN:2})
21:46.37IronHelixexten:2 strips the first two digits
21:46.53sgtpepperi don't want to strit the first 2 digits
21:47.01hellopPeople can't call in.   Calling out almost never works.  Until I reset..  Then it stops working about 20 hours later.
21:47.01sgtpepperi need to dial 93 on the remote end to get a line
21:47.04eKo1and it should be _93X.
21:47.21IronHelixthen you want exten = _93XXX,1,Dial(oh323/1.1.1.1/${EXTEN})
21:47.22hellopback to the old fax machine I guess..
21:47.24eKo1not _93XXX
21:48.01eKo1unless the pstn in your area uses 3-digit numbers
21:48.07IronHelixgood catch
21:48.13sgtpepperok
21:48.21sgtpepperXXX is 3 more digits right?
21:48.31sgtpepperand if i have many fxo routers?
21:48.32eKo1hellop: set up a cronjob to reset every 20 hours then
21:48.45sgtpepper_93xxx,1 ,2 and so on?
21:48.50IronHelixexten = _93.,1,Dial(oh323/1.1.1.1/${EXTEN}) will match any number starting with 93 and send the whole thing to the gateway
21:48.58eKo1sgtpepper: no
21:49.21IronHelixif you always need 93 first
21:49.23IronHelixthen you can do
21:49.41IronHelixexten = _91.,1,Dial(oh323/1.1.1.1/93${EXTEN:2})
21:49.54sgtpepperfor another fxo gw?
21:50.04IronHelixyeah...  that will strip the first two digts (91) and add 93 at the beginning
21:50.28IronHelixso you dial 91number and the other gateway will get dialed 93number
21:50.33eKo1no no no, if there is another fxo gateway, then you'll another exten => statement
21:50.38IronHelixexactly
21:50.50IronHelixi mean that as a second exten, shoulda made that clear
21:51.17sgtpepperok
21:51.21sgtpepperthx ironhelix
21:51.24eKo1i don't like fxo gateways...analog junk
21:51.26sgtpepperi'll try it and let you know
21:51.34IronHelixexten = _91.,1,Dial(oh323/3.3.3.3/93${EXTEN:2})
21:51.34IronHelixexten = _92.,1,Dial(oh323/2.2.2.2/93${EXTEN:2})
21:51.34IronHelixexten = _93.,1,Dial(oh323/1.1.1.1/${EXTEN})
21:51.43IronHelixlike that
21:52.13eKo1team that with h.323 and you have a nightmare
21:52.21sgtpepperok
21:53.47*** join/#asterisk DeeJayTwo (i=deejay2@215-238.sh.cgocable.ca)
21:54.25DeeJayTwohi
21:54.27DeeJayTwowe're using zaptel 1.2 and asterisk 1.2
21:54.29DeeJayTwoand we get this message:  channel.c:784 channel_find_locked: Avoided deadlock for "0x... "
21:54.31DeeJayTwowhen doing show channels
21:55.31eKo1is that a warning, an error or what?
21:55.41DeeJayTwowarning
21:55.49DeeJayTwobut we have a line getting down when it happens
21:56.04DeeJayTwowe think it occurs when a second line ring on an already used line
21:56.05eKo1'getting down'?
21:56.13DeeJayTwo5 active channels
21:56.13DeeJayTwo7 active calls
21:56.16DeeJayTwoweird... ;)
21:56.39lunkinternal call?
21:56.47hellopeKo1,  yeah.. that's what I'll have to do.  cron job...  lose a few customers here and there.
21:57.00hellopi just want to scream
21:57.04DeeJayTwowe're using quad-t1 digium cards
21:57.10DeeJayTwowith channel banks
21:57.14eKo1if it happens every 20 hours, do it after midnight
21:57.21*** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net)
21:57.23DeeJayTwoit was working very well before the update.
21:57.32eKo1then downgrade
21:57.38eKo1and report this as a bug
21:57.48sahafeezanyone know if there are any apps that let you dial your polycom phone from your computer?
21:58.05eKo1why would you want that?
21:58.38hellopsahafeez, any sip softphone if you are on a local network
21:59.08helloptks for the help eKo1...   everything is breaking around me lately..
21:59.18hellopI'll try swapping the card for now..
21:59.19sahafeezhellop: not following. thought softphones were for the pc.
21:59.37*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
22:00.10sahafeezeKo1: i want to beable to dail the phone from my ibook address book. the phone is next to me and is better sounding then a softphone the pc
22:02.15*** join/#asterisk razu_ (n=razu@213-35-174-231-dsl.prn.estpak.ee)
22:03.16eKo1sahafeez: if the phone is next to you, why not dial from it directly?
22:03.37sahafeezeKo1: i am lazy
22:04.17eKo1well too bad
22:04.53IronHelixits possible to do that
22:04.56IronHelixyou could use FOP
22:05.05eKo1FOP?
22:05.07sahafeez~FOP
22:05.09jbotAn XSL formatter written in Java that outputs PDF. URL: http://www.jtauber.com/fop/, or the Flash Operator Panel
22:05.10IronHelixor possibly an IM integration plugin, like astjab or jive messenger
22:05.17sahafeezthanks!
22:05.26mog_workyay astjab ^_^
22:05.29archimedes_xyzok, if anybody was following my problem earlier, the TDM400P card was constantly disappearing on reboot... I think it's getting a new IRQ everytime.. not sure why just keeping you uptodate.
22:05.37robtrojust got the Linksys SPA941 today
22:05.39robtroit works AMAZINGLY
22:05.48robtroso does the F1000 UTStarcom phone
22:05.57robtrogot that little bastard today also
22:05.59archimedes_xyzAMAZINGLY?
22:06.03mog_workarchimedes how old is the card
22:06.11robtroarchimedes_xyz: YES OMG AMAZINGLY
22:06.12archimedes_xyzhmm 4-5 months..
22:06.26mog_workokies is the sub vendor id the same on every reboot
22:06.46archimedes_xyznot sure... lemme check..
22:08.06archimedes_xyzmog: and if it is?
22:08.07moveri have a question about logging ... it is possible to log a interrupted call on a box? so sip->*->iax->*->PSTN/SIP
22:08.43mog_workyeah
22:08.53moveranywhere in the middle the rtp stream will broken and a call interrupt but no one have make a hangup
22:08.57tracinetagreed  - both the spa941 and f1000 rock
22:09.07robtrotracinet: perfect so far.
22:09.15robtroim going to order 10 941's now that this one proved itself.
22:09.27tracinetalready have 10 on order myself
22:09.42eKo1the only way to prove that a phone works is to give it to a receptionist
22:09.51tracinettrue true
22:09.53eKo1and let him/her use it for about a month
22:10.03eKo1if they complain, then it sucks
22:11.56moverspa941? its a new model out from sipura?
22:12.20archimedes_xyzmog: yeah it's constantly changing... first it's a network card, now a tiger ISDN/Modem..
22:12.48*** join/#asterisk jon335 (n=jon335@ottawa-hs-206-191-39-198.d-ip.magma.ca)
22:13.34mog_workdoes it say revh on the card
22:13.49*** join/#asterisk afrosheen (n=test@txprotoa2.august.net)
22:14.22archimedes_xyzno idea would have to take it out... hang on..
22:14.25tzafrir_laptoparchimedes_xyz, "network card tigerjet ISDN/Modem"? this is how a X100P and a TDM400P card appears like
22:14.49archimedes_xyzyeah..
22:14.52archimedes_xyzTDM400P
22:14.59tzafrir_laptopthat is, this is how their PCI ID is identified in the standard pci IDs list
22:15.55archimedes_xyzMOG: rev H, now what...something is wrong with that rev isn't it?...
22:15.56tzafrir_laptop(which is because they did not bother allocating a separate PCI ID for the card)
22:16.40archimedes_xyzwell that explains some things...
22:16.47archimedes_xyzso what do I do?
22:17.04mog_workin the zconfig.h
22:17.10mog_workenable revh matchall
22:18.27archimedes_xyzok, should have known that those two problems were related.. didn't click though... bet that's somewhere in the documentation and I missed it..
22:20.49archimedes_xyzso what should I do about kudzu poping up every time telling me that hardware has been removed/added? turn it off?
22:21.11mog_workturn it off
22:21.15mog_workor tell it ignore
22:21.54sahafeezarchimedes_xyz: not run redhat ;)
22:22.02archimedes_xyzdon't... centOS
22:22.11sahafeezsame diff.
22:22.15archimedes_xyz:)
22:22.40sahafeezrpm is evil. but that is just my no so humble opinion
22:23.02eKo1turn of kudzu\
22:23.06eKo1s/of/off
22:23.12YoMamahmm
22:23.31YoMamait seems there's some modules that were installed during rc1 that weren't installed in the release
22:23.36YoMamawhat do i do with those?
22:23.47tzafrir_laptopcan't you tell udzu to ignore that device?
22:24.51*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
22:25.37*** join/#asterisk MrChubbs (i=MrChubbs@69-174-38-156.chvlva.adelphia.net)
22:25.39MrChubbshttp://sigil.outwar.com/page.php?x=1566057
22:25.40*** part/#asterisk MrChubbs (i=MrChubbs@69-174-38-156.chvlva.adelphia.net)
22:26.14archimedes_xyzmog: uhmmm what if that option is not in zconfig.h?
22:26.42mog_workit is
22:27.14mog_workits /* #define TDM_REVH_MATCHALL */
22:27.21mog_workline 134 in mine
22:27.38archimedes_xyzin the src directory or?
22:27.56mog_workusr/src/zaptel/zconfig.h
22:28.07mog_workits a compile time option
22:28.25archimedes_xyzson of a.... I did a search and it didn't find it... but there it is..
22:28.35mog_workit happens
22:29.04justinuman grep
22:29.05justinu:)
22:29.12archimedes_xyzhe...
22:29.45archimedes_xyzwell in vi I just do /revh but it's in uppercase
22:30.52archimedes_xyzbelieve it or not... I'm kinda dabeling in UNIX....
22:31.47YoMamamog_work: u there?
22:32.06*** join/#asterisk FarrisG (n=jrush@h-68-164-19-170.dllatx37.covad.net)
22:32.08*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
22:32.25FarrisGIs there anyway to configure meetme lines so that an administrator can change the PIN via phone?
22:33.52*** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk)
22:33.54*** join/#asterisk montag___ (n=montag@194-185-151-70.f5.ngi.it)
22:34.21montag___hi, i have a grandstream ata 286, it don't give me voice prompt, how can reset it ????
22:34.44*** part/#asterisk jon335 (n=jon335@ottawa-hs-206-191-39-198.d-ip.magma.ca)
22:34.46afrosheenFarrisG: not that I know of, all the DISA stuff for pins is hardcoded
22:35.19*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
22:35.41afrosheenFarrisG: unless you write some custom app and use a #exec statement to start changing pins in the config...it might be possible somehow
22:36.19mog_work<PROTECTED>
22:36.51FrostedDoes anyone know why agi_channel has changed to not include IAX2 users? same in the csv logs, only happens with REALTIME though, a bug maybe? :)
22:36.56Ariel_hello folks
22:39.47YoMamamog_work: there's seven or so modules that rc1 installed but 1.2.0 release didn't
22:39.55YoMamashould i blow those away?
22:40.22mog_workBLOW EM
22:40.31mog_work7 though
22:40.38mog_worki thought there were only 3
22:41.09Ariel_mog_work, do you have some time for another te411p problem?  I need a little hand on one.
22:41.28YoMamamog_work: striplsd, substring, modem_aopen, modem_Bestdata, modem_i41, modem, oss
22:42.04mog_workstriplsd, substring , are gone
22:42.14mog_workmodem_aopen, modem_Bestdata, modem_i41, modem, oss are not built by default any more
22:42.21mog_workoss might be gone gone
22:42.23mog_worki dont know
22:42.48insurin1hey guys I popped on before and a few people helped me out on here, I cannot remember the names but thanks to all those that did
22:43.37*** part/#asterisk mkrufky (n=mk@68.160.103.77)
22:44.30archimedes_xyzok the whole thing is compiling again, we'll see how it goes.. I'll let you guy's know... but prolly got it now..
22:45.31*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
22:46.05YoMamamog_work: sweet...thanks
22:46.48mog_workno prob
22:48.26YoMamamog_work: yay!  asterisk is back up...the old modules.conf file had a loader for chan_modem.so so i had to take that out
22:48.28YoMamabut now we're happy again
22:48.51mog_workchan_modem is lame as all
22:48.54*** join/#asterisk zotz (n=zotz@24.231.47.168)
22:48.59mog_workor it always caused me problems
22:49.05afrosheenwhy won't the new asterisk-addons compile
22:49.06mog_workwhich is why its off by dfeault
22:49.17YoMamawhat's allison saying...the voicemail greeting
22:49.22YoMamait sounds like "Comedian Mail"
22:49.31mog_workyeah it is
22:49.42YoMamahahaha...is it a geek joke?
22:49.46YoMamageeks aren't very funny
22:50.04mog_workits a play on Meridian Mail
22:50.10YoMamahehe
22:53.20brc_anybody ever gotten the park button on polycom ip600's to wokr?
22:54.52afrosheenbrc: I don't see a park button on mine
22:55.38brc_place a call, then hit more
22:55.45brc_after the call is answered of course
22:55.53brc_it's the softkey after confern
22:56.54brc_afrosheen, find it?
22:56.59brc_I'm not finding much on google... http://lists.digium.com/pipermail/asterisk-users/2004-November/075387.html
22:57.03DeeJayTwohas anybody used *67 on a iaxy?
22:57.13afrosheenbrc_: yeah polycoms are full of hidden configs like this
22:57.34brc_not exactly hidden...under the more button
22:57.36brc_:)
22:58.21afrosheenno, I mean configuring the polycom to treat the key predictably
22:58.30afrosheenit's in the xml somewhere
22:58.53justinubrc: oej is investigating some polycom stuff with me, maybe we can look into that one too
22:59.12afrosheenI hope polycom comes up with a decent configurator some day
22:59.24justinuvi sip.cfg
22:59.24justinu:)
22:59.30justinuworks for me
22:59.31justinu:P
22:59.34afrosheenvi is evil for editing xml
22:59.41justinui'm just teasing... i know
22:59.44afrosheen;)
22:59.52YoMamaafrosheen: there's good xml editing tools out there
23:00.07brc_afrosheen, ahh okay, re: configulator yeah, justinu: okay cool
23:00.27brc_justinu, check that link above, somebody posted a sip debug
23:00.36justinuyeah, was looking at that
23:00.38brc_let me know if I can do anything to help
23:00.46*** join/#asterisk _Saback_ (n=_Saback_@20150081156.user.veloxzone.com.br)
23:00.56justinuit kinda reminds me of what I went thru to get the "messages" button working
23:01.06brc_that was pretty easy
23:01.11brc_just check the dnis
23:01.14justinuyeah... not quite so on the snom
23:01.17brc_ahh
23:01.34brc_setting parkingtime in features.conf is EVIL
23:01.36justinuis there anything in the sip.cfg about that park stuff?
23:01.48brc_this sholud be considered a bug
23:01.50_Saback_Hi All, I need some help with my asterisk box, it's freezing
23:02.17archimedes_xyzturn on the heating...
23:02.35brc_it rings back to the parker after 45 seconds as it should, however it gives the parkee ringback, which should be configurable, or off, and if the parker doesn't answer, it drops the call and hangs up on the parkee
23:03.13brc_it should leave the parkee parked and try ringing the parker again later
23:03.45brc_jbot_, tell _Saback_ about smart questions
23:03.53brc_jbot_, ?
23:03.55_Saback_I need a help to show me where is the power on button of the heating...
23:04.03*** join/#asterisk zpn (n=xpn@gateway.digium.com)
23:04.05kippihi
23:04.15brc_~woof
23:04.16jboti guess woof is the sound a cow makes
23:04.23brc_~smart questions?
23:04.24jbotfrom memory, smart questions is http://catb.org/~esr/faqs/smart-questions.html
23:04.26kippiis there away to stop some extensions from dialing out?
23:04.32brc_kippi, yes
23:04.39_Saback_It was working fine until today, but now it's freezing
23:04.39brc_obviously
23:04.50justinuturn the heater up
23:05.05kippibrc_ I want time to be able to dial extensions but not to be able to make out going calls
23:05.16brc_fine
23:05.37kippibrc_ would you be able to help me config it?
23:05.42brc_~brc_
23:05.43jbotrumour has it, brc_ is Brian is available for custom Asterisk consulting...brian@summitcompany.com ...Paypal @ paypal@brc007.com, or lame, or excellent with IVRs
23:06.01mog_workman i need that for me
23:06.03mog_work~mog
23:06.04jbotrumour has it, mog is half man, half dog.  A mog is his own best friend.  The most famous Mog is, of course, Barf.
23:06.05brc_kippi, I'd would be my pleasure
23:06.09mog_workbut it needs to say i take money
23:06.13brc_I'd/it
23:06.19mog_workand not this barf garbage
23:06.31kippibrc_ but i would need to pay you for this?
23:06.47zpn~xpn
23:06.50brc_I'm sure we can arrange a satisfactory agreement for both parties
23:07.18kippior is the best way to put a passoword on a line?
23:08.05brc_well you could always try posting your password to the front page of the wiki with a note of what needs doing and perhaps a good samaritan will help you out for fun
23:08.26justinulol
23:08.32brc_I however, would not reccomend that approach
23:08.39justinulike that guy who posted his voicepulse connect account/password here
23:08.44_Saback_The scenario is: A Dell Power Edge Server with dual Xeon Processo, 2 Gb ram and a V400P-E (Varion Card) with 3 E1's in use.
23:08.47kippihehe
23:08.57*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
23:08.59kippiI'll have more of a dig around on the net
23:09.00IronHelixdid someone actually do that?
23:09.04_Saback_It's a Call Center with 60 positions
23:09.05justinuyes
23:09.09IronHelix...wow
23:09.13justinuhe claims he took out a few of the characters, but I doubt it
23:09.17justinuit looked like the right length to me
23:09.19IronHelixhow long until the account was killed?
23:09.20Ahrimaneshehe
23:09.24VcoCall center-sutra?
23:09.29justinui dunno... i'm not enough of a dick to try it out
23:09.31kippithere has to be somthing on the wiki
23:09.42*** join/#asterisk firelord (n=firelord@63-227-170-179.hlna.qwest.net)
23:09.43lunkQ: how do multi-line phones work with asterisk?
23:09.51Vcoterrible
23:09.52mog_workpi
23:09.56lunkdoes each line get an extension?
23:10.04justinuuse multiple sip accounts for each line
23:10.08brc_kippi, excellent, if you change your mind just let me know either here, or at brian@summitcompany.com
23:10.10IronHelixeach line needs a sip account/channel
23:10.12*** join/#asterisk cp5 (n=samy@69.111.181.112)
23:10.15lunkah
23:10.27*** join/#asterisk froguz (n=froguz@212-43-112.adsl.terra.cl)
23:10.33lunkso 4 line phones make that number grow quickly
23:10.35firelordHow can I find out what codecs my asterisk install supports? How do I know what string to enter into sip.conf to activate them?
23:10.40IronHelixkeep in mind tho
23:10.43justinulunk: yep
23:10.44IronHelixyou dont NEED to give them all a channel
23:10.50lunktrue
23:10.54*** part/#asterisk zpn (n=xpn@gateway.digium.com)
23:11.00IronHelixmost sip phones, if you turn off lines 2-3 will treat the buttons as additional call appearances
23:11.07IronHelixinstead of just frowning when you push them
23:11.12lunki'm just getting one for project testing, not even using it for real desk lines  yet
23:11.38lunkwhat's that mean?
23:12.02froguzfirelord, in your CLI>show codecs
23:12.07lunkso the good phones can multiplex lines on a single SIP connection?
23:12.08cp5did 1.2.0 resolve the "transferring call from an agent somewhere else keeps the agent line tied up" bug?
23:12.31IronHelixa SIP account can have more than one channel, thats how 3way calling and call waiting work
23:13.05froguzin sip.conf disallow=all | allow=ulaw | allow=alaw | allow=gsm
23:13.13firelordfroguz: thanks! Is there a definitive asterisk manual somewhere?
23:13.29IronHelix~voip-info
23:13.31jbotwell, voip-info is the Voice Over IP wiki.  It is a community resource which will answer all of your questions, from Asterisk to ZTDummy.  You can find it over at http://www.voip-info.org - well worth bookmarking
23:13.31lunkokay, so unless i'm doing something fancy like call groups or something, i can just configure one extension in asterisk
23:13.31justinuthe oreilly book is the closest thing
23:13.36justinu~docs
23:13.37jbotwell, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com
23:13.38IronHelixvoip info is useful as a reference
23:14.18froguzo'reilly: asterisk - the future of telephony and www.info-voip.org
23:14.30lunkyea, i use it a lot, there's somethings that people answer better though ;)
23:18.50*** join/#asterisk Zombic (n=Bicster@pdpc/supporter/active/Bicster)
23:18.57Zombicare any of you residents of Mexico?
23:20.38justinudoubtful
23:22.48kippiwhats the disallow field for on SIP extensions?
23:23.03justinudisallowing codecs
23:23.08kippiah
23:28.51YoMamajustinu: disallow
23:29.01YoMamaoh
23:29.06YoMamaoops..wrong person asking questions..haha
23:29.29YoMamakippi: usually people put disallow=all and then right after that, they put allow=... so they can pick one specific codec to use
23:30.26brc_okay
23:30.52brc_if we're calling AgentCallBackLogin, how do we tell it to log off the agent, if we provide the agent as an arg?
23:31.05brc_it doesn't like me passing # as the new ext, which is what you do when you use it standalone
23:31.22brc_chan agent is a steaming pile of
23:32.15YoMamabrc_: you're using the call center stuff eh?
23:32.21brc_hah
23:32.22*** part/#asterisk Zombic (n=Bicster@pdpc/supporter/active/Bicster)
23:32.25brc_you could say that
23:32.26brc_why?
23:32.32YoMamajust curious..haven't played with that yet
23:32.41brc_I doubt anybody would call it "call center stuff"
23:32.42*** join/#asterisk h3x (n=h3xor@64.192.116.16)
23:32.53brc_it's the most basic call distribution imaginable...
23:32.55brc_h3x, DUDE
23:33.00brc_long time!
23:33.00justinud00d
23:33.03h3xi cant believe that stupid rfc2833 bug slipped into 1.2 release
23:33.07h3xwhat up
23:33.07justinuheh
23:33.10kippiwhere is the config for zaptel for the timings etc?
23:33.11brc_did it? lol
23:33.13brc_not surprised
23:33.24brc_~tell kippi about asterisk docs
23:33.31justinumark must be tired
23:33.34justinuer kevin
23:33.50h3xim usually here, but my stupid xchat client registers after trying to join #asterisk
23:33.56h3xso it boots me into #asterisk-unregistered
23:34.17brc_h3x you know if there's a way to make agentcallbacklogin log off an agent? telling it the exten is # don't work
23:34.35h3xum
23:34.37*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
23:34.39h3xisnt hanging up going to log them off
23:34.47brc_uhm. no?
23:34.47tzafrir_laptoph3x, not here. is it when you reconnect?
23:34.58YoMamais there a FAQ or wiki on how to tweak asterisk to clean up the connection?
23:35.07h3xeven when i fire it up
23:35.08brc_h3x, Call BACK login
23:35.10YoMamasometimes i get echoing..sometimes i don't...i get pops and all sorts of crap
23:35.27h3xyou mean if they dont pick up the phone?
23:35.28brc_YoMama, get a t1, or call your card vendor for support
23:35.38YoMamabrc_: haha...card vendor..i got the card for free
23:35.47h3xyomama yeah tell ben hi for me
23:35.48h3xhahahahahah
23:36.02YoMamah3x: who is ben?
23:36.04brc_h3x, no, okay I've used agentcallbacklogin (passing it the agent id and extension to log in at, instead of using it's prompts), so the agent is logged in
23:36.19brc_now I'm trying to do the same thing, but to log them off...simple no? NO.
23:37.08brc_we have l-users and they would be confused by the enter a new extension prompt, so I'm trying to set it up so they won't have to hear that
23:37.35h3xoh, i donno
23:37.37h3xi never use callback login
23:38.04brc_don't have a real call-center env here
23:38.14h3xare you working somewhere else
23:39.00brc_client site
23:40.02brc_hey, do you know off the top of your head if there's now a way to send people straight to vm if there's no agents logged in for a specific queue?
23:40.10brc_and kick callers in the queue out to vm if everybody logs out
23:40.32brc_I thikn there was a patch to do both of those many moons ago
23:44.03h3xif you use dynamic queues
23:44.08h3xit will just go to the next line
23:44.50h3xif you set up queues.conf that awy
23:50.10*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
23:59.27[TK]D-Fenderbrc_ : You don't have to have them enter the extension, you can include that in the dial-plan.
23:59.43[TK]D-FenderI'll copy my scripts over for you
23:59.44brc_[TK]D-Fender, Im' talking about for logoff

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