IRC log for #asterisk on 20070922

00:08.47__freedom__lover\quit
00:09.06*** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca)
00:12.56*** join/#asterisk brut- (n=brut@66.173.4.254)
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00:24.40nentisAnyone know a way to easily cron up a schedule to play a sound over voip phones?  I'm looking for a way to use phones in classrooms as a bell system.
00:26.07*** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk)
00:29.14ManxPowernentis: Use cron to generate a .call file, drop it into /var/spool/asterisk/outgoing, I suggest the actual destination of the call uses chan_local.  See localchannel.txt and the Wiki
00:29.38nentisWill do, thanks.
00:30.27ManxPowerYou should also look at app_page (show application page) and you'll have to figure out how to make the phones auto-answer.
00:30.57ManxPowerthis isn't rocket science -- it is more like chemistry.  You mix the correct pieces to get the desired result.
00:31.09nentisI have a perl script someone wrote which uses a user in manager.conf and plays a .wav to a page group (I have page groups working)
00:31.30perdis the redirecting party number put into a variable when i receive a call from h323 ?  http://pastebin.com/d246603ba
00:31.34nentisbut I'm getting "ResponseErrorMessageOriginate failed"
00:31.37perdthe pastebin shows the data i'm talking about...
00:32.06ManxPowerperd: if it is, it would be DNID or RDNIS
00:32.07nentisHere is the script:
00:32.09nentishttp://pastebin.com/m2981bbc9
00:32.27ManxPowernentis: using manager would work too.
00:32.43perdok thanks manx
00:32.45perdi'll try those
00:32.58ManxPowerAsterisk is to PBXs, as Perl is to programming languages -- you always have 5 zillion different ways to do what you want to do.
00:33.10ManxPowerperd: README.variables is the file you are looking for.
00:33.15ManxPowerat least that's the name in 1.2.x
00:33.22perdahh, i was looking at http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List
00:33.51ManxPowerperd: don't be suprized if the H323 channel driver does not set those vars.
00:34.05perdthat would make me sad.
00:34.12perdim about to find out though
00:34.25ManxPowerperd: Recommended order of docs:  Asterisk CLI, Asterisk 'docs' directory, asterisk mailing list archives, IRC, The Wiki
00:34.27perdyay it sents rdnis
00:34.32ManxPowernotice the wiki is at the END
00:34.37perdhaha yeah...
00:34.45perdthat's usually on thefront of my list, perhaps that's my problem
00:34.50ManxPowerIt is.
00:35.00ManxPowerThe Wiki is very out of date.
00:35.17ManxPowerFor things like "I need an example of how to do X" it is not a bad resource.
00:35.29ManxPowerBut for things like "what are the docs for this application", it is the worst resource.
00:35.32*** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net)
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00:35.39perdduely noted, thank you sir
00:35.54perdand now i got my voicemail integration working flawlessly. that was pretty easy..
00:36.12perdnow to tackle the issue of calling asterisk -> callmanager
00:36.13perdoi.
00:36.26ManxPowerfor example, it does not look like any of the new 1.4 variables are on that Wiki page
00:43.46dlynes_laptop[TK]D-Fender: thanks....that trick for the aastras works perfectly on both the 9133i's and the 57i's now
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00:54.48dlynes_laptopHow stable is the shared line appearance support in 1.4?
00:55.20dlynes_laptopIs it production quality?  Or is it only suitable for a home office?
01:00.03nentisManxPower,  I'm so close.  I have a .call file which works on a single extension, but can't get it to work with a page group.  So "Channel: Local/PAGE191@ext-paging" works, but I'm having difficult finding the syntax for sending this to page group 600.
01:04.29*** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com)
01:06.21nentisgot it.
01:06.30nentisChannel: Local/600@from-internal
01:14.47mmlj4ManxPower: hey
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01:28.59*** join/#asterisk twisted[mobile] (n=root@pdpc/supporter/active/twisted)
01:28.59*** mode/#asterisk [+o twisted[mobile]] by ChanServ
01:29.45*** join/#asterisk coppice (n=chatzill@50.206.17.210.dyn.pacific.net.hk)
01:30.10twisted[mobile]wheeeee  90mph
01:31.07twisted[mobile]"warrior" next 3 exits
01:33.17twisted[mobile]you ppl are boring
01:34.31filetwisted[mobile]: look in the mirror, you'll see a nub
01:35.09twisted[mobile]i see a truck
01:35.20rob0uh oh, that could be bad
01:35.30twisted[mobile]brookshire cant drive
01:36.04twisted[mobile]nah were going 90
01:36.43twisted[mobile]and its hard to type on my blackberry
01:37.48twisted[mobile]sorry, 95mph
01:38.41rob0on I-65?
01:39.01twisted[mobile]yessir
01:39.10rob0woohoo!
01:39.35twisted[mobile]why are you in auburn?
01:39.48twisted[mobile]OH SHI_-_-
01:40.01twisted[mobile]MO CSRRIER
01:40.42twisted[mobile]wont hurt me...
01:41.17twisted[mobile]wireless fiber baby!
01:42.09twisted[mobile]if anyone wants to fund our drinking....
01:42.26twisted[mobile]~twisted
01:42.27jbottwisted is, like, toastido@gmail.com, for paypal and chatter.  He is also known in some circles as toastido.
01:42.30coppiceisn't wireless fibre an oxymoron?
01:42.58twisted[mobile]NO ITS A LAZERk
01:43.18coppiceyou mean fibreless fibre?
01:43.39twisted[mobile]were doing voice UNDER IP to save bandwidth
01:44.11twisted[mobile]wtf is fulltondale?
01:44.13coppicesounds like a good idea. let's call it "the PSTN"
01:44.53twisted[mobile],+ it has to be "*423
01:44.55twisted[mobile]err
01:44.59twisted[mobile]LASER
01:45.19twisted[mobile]fucking bupy ass road
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02:26.57JunK-Ytwisted!!!
02:33.11*** join/#asterisk iPod-nano (n=david@c-68-43-60-193.hsd1.mi.comcast.net)
02:33.36iPod-nanoHow do I create an extension that answers incoming SIP traffic and redirects it to the proper extension?
02:34.57JunK-Yjust match the sip extension which is dialed.
02:35.26iPod-nanoWell, it's not that simple.
02:35.42[TK]D-FenderiPod-nano, Yeah... you should try to be a little more vague...
02:36.38iPod-nanoI want incoming sip traffic to 5121@mydomain to hear "Extension, please?", the caller enters a four-digit extension, and is then redirected to me or someone else.
02:37.18Corydon76-digUh, why shouldn't they just dial the right extension to start with?
02:37.36iPod-nanoBecause I'm trying to set up an access number from a PSTN line.
02:37.45iPod-nanoVia SIPBroker.com.
02:38.11*** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com)
02:38.13iPod-nanoSo, the SIPBroker access number would call my server's 5121 extension.
02:38.22Corydon76-digSee application Read
02:39.46iPod-nanoAsterisk itself is the SIP client.
02:40.22[TK]D-FenderiPod-nano, exten => 5121,1,NoOp(Yippy-kai-yay-mo....... Incoming call!)
02:41.07Corydon76-digiPod-nano: not in that situation.  Asterisk is the SIP server.
02:41.13[TK]D-FenderiPod-nano, shove that in a cotext.  set "context=[thatcontext]" under [general] in sip.conf along with "allowguest=yes".
02:41.17[TK]D-FenderiPod-nano, End of story.
02:46.34ManxPowerTelecom noobs, including Asterisk noobs seem to think that dialplans can be simple, short, maybe even elegant.  Dialplans are ugly beasts that grow until you start to have nightmares about them.
02:50.52mmlj4ManxPower: ping
02:51.19mmlj4ManxPower: got any checks from our reluctant payer yet? I scolded them a few days ago about it
02:51.26*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
02:53.48iPod-nanoUm... Asterisk is supposed to wait for another extension to be dialed.
02:54.58Corydon76-digAsterisk isn't supposed to do anything other than what you program it to do
02:55.27Corydon76-digSo if it's not waiting, then you failed to tell it to wait
02:55.34iPod-nanoWell... how do I program it to do that?
02:55.59Corydon76-digWhat have you put in your extensions.conf so far?
02:56.15iPod-nanoA whole lot of text. :-P
02:56.17Corydon76-dig~pb
02:56.18jbotA Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
02:57.00*** part/#asterisk tmayse (n=tmayse@71.30.73.40)
02:57.31[TK]D-FenderiPod-nano, .......
02:57.34[TK]D-Fender~book
02:57.34jbot[book] Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 or Temporarily at http://www.aocomputing.net/AsteriskTFOT.pdf
02:57.37[TK]D-Fender^^^^^^^^^^^^^^^^^^^^
02:57.41Corydon76-digHurry, my cable modem is scheduled to go flaky in another hour...
02:57.45iPod-nanoYes, I have the book.
02:57.57Corydon76-digDo you have the NEW book?
02:58.07[TK]D-FenderCorydon76-dig, shhhh
02:58.36iPod-nanoCorydon76-dig: http://rafb.net/p/iuJ8xF94.html
02:58.59*** join/#asterisk simonr (n=simonr@74.12.145.41)
02:59.26iPod-nanoI accidentally pasted that twice.
02:59.40*** join/#asterisk adker (n=chatzill@74-33-199-190.br1.glv.ny.frontiernet.net)
03:00.09Corydon76-digiPod-nano: I think you meant WaitExten, not Wait
03:00.32*** join/#asterisk brut- (n=brut@66.173.4.254)
03:00.39Corydon76-digand anyway, I'd suggest that you use Read, not WaitExten
03:01.17iPod-nanoWhat does that one do?
03:01.32[TK]D-FenderiPod-nano, "show application [appname]" <-
03:02.58JunK-Ycore show application :)
03:03.41[TK]D-FenderJunK-Y, look at the vars he's using in his dialplan... I would not go around assuming 1.4 :p
03:03.55[TK]D-FenderJunK-Y, And deprecated as it is... it still WORKS.
03:04.02JunK-Yim not reading seriously ;)
03:04.47iPod-nano1.2.13
03:05.20Corydon76-dig1.2.13 is months old.  There are a ton of bug fixes since then
03:06.47JunK-Yand few introduced too.
03:07.53iPod-nanoThat's great, but apparently Debian doesn't know about that.
03:08.13[TK]D-FenderiPod-nano, Debian is oblivious to the Age Of Reason.
03:08.42Corydon76-digwhich is why everybody else compiles from source
03:08.43[TK]D-FenderiPod-nano, And tonight they're gonna party like 1699 :)
03:09.21*** join/#asterisk sts3c (n=bryan@66-43-34-10.misn.com)
03:09.45dijungalwhere do i see the bug fixes for the new release ?
03:09.51iPod-nanoOK, what am I  missinghere?
03:09.58iPod-nanoIt keeps hanging up on me?
03:10.28iPod-nanoI want it to wait for a number to be dialed, then connect to that extension.
03:12.12*** join/#asterisk simonr (n=simonr@74.12.145.41)
03:12.19Corydon76-digWhy don't you use DISA with no-password ?
03:12.38Corydon76-digIt's by far the simplest solution
03:12.45iPod-nanoBecause I haven't a clue what you'e talking about.
03:13.22Corydon76-digexten => 5121,DISA(no-password,mycontext)
03:13.32Corydon76-digerr
03:13.37Corydon76-digexten => 5121,1,DISA(no-password,mycontext)
03:13.37JerJer,1,
03:14.07Corydon76-digIn my defense, it's late
03:14.31iPod-nanoWait, what context am I using in there?
03:14.59Corydon76-digwhatever context you specified in sip.conf
03:16.46Corydon76-digand then in the DISA argument, you specify whatever context contains your phone extensions
03:17.50Corydon76-digHowever, it seems to be clear that you haven't defined any phone extensions yet
03:18.13Corydon76-digi.e. exten => 1234,1,Dial(SIP/1234)
03:19.55*** join/#asterisk putnopvut (n=putnopvu@c-71-228-186-75.hsd1.al.comcast.net)
03:20.43iPod-nanoUgh, I'm getting conffused.
03:20.52iPod-nanoAnd this keyboard gets on my nerves.
03:23.36*** part/#asterisk dijungal (n=kdaniel@63.175.159.171)
03:23.48*** join/#asterisk marcan (i=1337@host214-134.cvd.fit.edu)
03:29.57CCFL_Man2Strom_M: a green handset cord with 3 wires, for an F1 handset, you know any place i could get one?
03:30.08Strom_Mnot off the top of my head
03:32.47CCFL_Man2Strom_M: my coiled cloth green handset cord is faded and stretched, no frayed though
03:35.16CCFL_Man2Strom_M: i did fone a brass imperial 202 with ivory F1 handset on ebay
03:35.23CCFL_Man2fone = find
03:38.33CCFL_Man2Strom_M: http://cgi.ebay.com/Brass-Western-Electric-Telephone-Phone-ringer-Rotary_W0QQitemZ320160540893QQihZ011QQcategoryZ38038QQssPageNameZWDVWQQrdZ1QQcmdZViewItem
03:40.13CCFL_Man2look at that farking beauty
03:40.22CCFL_Man2been through hell but damn
03:40.45CCFL_Man2though mine is green :P
03:48.10*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
03:49.28CCFL_Man2where is rudholm
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04:09.01iPod-nanoWhat does this mean?
04:09.02iPod-nanoSep 22 00:08:21 WARNING[10665]: channel.c:2380 set_format: Unable to find a codec translation path from ilbc to ulaw
04:14.12*** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr)
04:25.08[TK]D-FenderiPod-nano, means * doesn't know how to translate between ULAW and ILBC just like it says
04:25.49[TK]D-FenderiPod-nano, look at "show translation" for a lit fo what * can transcode.  If you see "--" in the coordinates of a combo you're intersted in... TFB
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05:06.39*** join/#asterisk ectospasm (n=ectospas@c-68-62-214-17.hsd1.al.comcast.net)
05:07.06ectospasmso...  I learned Wednesday I passed the DCAP test
05:10.22Strom_Ccongrats!
05:10.42Strom_CI HAS TAUGHT YUO WELL
05:22.39ectospasmYes, but you weren't all of my training, to be sure!  Now, that isn't to say that I would have passed without your training...
05:23.09ectospasmI think I should have done better on the written portion, but I squeaked by with an 81%
05:23.43Strom_Chey, that written test is tough tough tough
05:25.33ectospasmYou can say that again
05:26.07ectospasmAlthough my major complaint with it is that I knew exactly where to look to find the answer to a lot of those questions
05:26.25ectospasmso much of it required memorization which I feel unnecessary
05:27.11Strom_Cwell, the difference betwen someone who is merely good and someone who is extremely competent is the degree to whch they remember how all the bits and pieces fit together
05:27.35Strom_Cyes, you can look it up and do a perfectly good job
05:27.37ectospasmLike, why was there a question about the default H.323 port on there?  Grant it, I'm not that experienced at work yet, but I still haven't come across anyone who has tried to use H.323
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05:27.46ectospasmStrom_C:  I guess you're right
05:28.18Strom_Cbut at the point where you can just look at something and go "Oh!  I know what that is!", then you've moved beyond merely a technician
05:28.24ectospasmI dunno, I got through college BSing a lot of essay or short answer questions, so I guess I'd say I'd like to see some of those on there
05:28.52ectospasmwhich would make them tougher to grade, fo sho
05:30.14ectospasmAnyway, I guess I don't have to worry about that now.  Not until they devise DCAP lvl II
05:30.29Strom_C:)
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05:39.01jmghey all
05:39.07jmgNo application 'StripMSD' ?
05:39.18jmghas it been supersceded by something else?
05:39.29jqlyeah, string slicing
05:39.36jmgex?
05:39.38jql${EXTEN:1}
05:39.42jmg:)
05:39.43jmgta
05:43.35jmgim trying to make enum work
05:43.39jmgwith e164
05:43.47jmgi think the code on their wiki is broken
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06:06.36Yourname`Hi guys. How can I incorporate the 1.4 WaitForSilence to work in the 1.2?
06:07.52Strom_Cumm
06:08.01Strom_Cwhy not...just....you know............upgrade?
06:09.45luke-jrStrom_C: well, on my side of the ballpark over here, 1.4 is totally broken ☺
06:09.53luke-jrso I can't blame him
06:10.05luke-jrs/is/looks
06:10.12Strom_C"totally broken"?
06:10.15ectospasmdefine,"broken"
06:10.18Strom_Cit won't even compile?
06:10.30luke-jrwell, H323 has compile issues, but i meant runtime
06:10.40luke-jras in, my old dialplan needs rewriting
06:10.43Strom_Cwhat kind of issues?
06:10.50luke-jrand the Jabber support is not useful
06:11.06Strom_Cheh, so it's not 1.4 that's broken, it's that you never read UPGRADE.txt when you upgraded to 1.2
06:11.14luke-jrno, I started with 1.2
06:11.36Strom_Cif your dialplan uses nothing deprecated in 1.2, then it should work fine in 1.4
06:11.46luke-jrfor(;;) loops don't like iterators in 1.4 ☹
06:11.51luke-jrfor example
06:12.14Strom_Chow so?
06:12.24luke-jr...
06:12.39luke-jr1.4 treats all 3 parts as math expressions
06:12.50luke-jrwhich macros don't work inside
06:13.07luke-jr1.2 only treated the comparison as a math expression
06:13.37Strom_Chmm
06:13.53Strom_Cdid you file a bug, or is it noted already in UPGRADE?
06:13.58ectospasmum, isn't an increment a math expression too?
06:14.34luke-jrStrom_C: yes, we're working on getting it fixed now
06:14.48luke-jrectospasm: not if it's an iterator
06:24.05*** join/#asterisk dongs (i=500@l212168.ppp.asahi-net.or.jp)
06:24.13dongsanyone here familiar with ISDN
06:25.15Strom_Ci'm familiar with it
06:25.27Strom_Cbut whether i'll be able to answer your question is another story
06:26.47dongsi want to know whats involved in providing an ISDN connection. whats the FXS end of ISDN called.
06:27.38Strom_CISDN BRI or ISDN PRI?
06:28.13dongsBRI
06:28.54Strom_Cwell, the first piece of equipment on the customer's premise end is called the NT1
06:29.08dongsyeah. I know the receiver end.
06:29.15dongsim talking about whats involved in providign a ISDN circuit.
06:29.28dongsi.e. I want to plugin a ISDN phone or S/T device into <?????> and have them work.
06:29.47Strom_Cso you want to act like the telco?
06:29.50dongsyes.
06:30.07dongsand never mind S/T thats too much work. just the RJ45 w/isdn signalling on it.
06:30.18Strom_C....?
06:31.20Strom_Care you planning on integrating this with asterisk?
06:31.31dongsnot at all.
06:32.19dongsin your area how does ISDN circuit reach customer premises?
06:32.20dongstwisted pair?
06:32.35Strom_Cdo you actually want to provide usable ISDN service, or do you merely want to test out CPE you've got lying around?
06:32.36dongsi need whatever's on the other end of that.
06:32.40Strom_Cyes, twisted pair
06:32.58dongsthe former
06:33.05dongs(usable ISDN serviec)
06:33.12Strom_Cwell, then you'll probably want to buy either a 5ESS or a DMS-100 switch
06:33.27dongsthat doesnt seem very portable.
06:33.35Strom_CDMS-10 then perhaps? :)
06:34.27dongsso how would the other end of the line be called
06:34.34dongsso I can at least google some chipsets/etc related to this stuff.
06:34.59dongsall I can find as far as I can tell is solutions for cpe and not for providing.
06:35.21Strom_Cwell, you can't just buy a DMS line card and expect to provide ISDN service
06:35.38Strom_CI don't get what ends you're trying to achieve
06:36.00dongsi need to provide equivalent of analog FXS port but for ISDN.
06:36.39Strom_Care you a telco?
06:36.43Strom_Cdo you have customers?
06:36.54dongsno, it would be a portable analog -> ISDN adapter
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06:37.02dongs(dont ask why)
06:37.11Strom_Cthat makes no sense
06:37.22Strom_Cwhy the hell would you want analog on the network end and then ISDN on the CPE end?
06:37.34dongspbx interfacing mostly.
06:37.51dongsa ton of equipment here only takes ISDN BRI as lines.
06:37.58dongsa ton of VOIP adapters, on the other hand, output analog.
06:38.09dongs(again, dont ask)
06:38.15Strom_Cyou're not seriously considering plugging terminal adapters into a PBX, are you?
06:38.21dongsyes, I am.
06:38.30Strom_Cthat's a completely broken network design.
06:39.16dongsgoogling for ISDN line card seems to be bringing interesting stuff. now if I could get high-resenough photos to see what electronics are involved
06:39.35dongsand/or get some overview of how this shit works.
06:39.38Strom_Care you trying to replace telco BRI circuits with ITSP service?
06:41.57dongscheck this: typical installation: 10-line PBX with ISDN BRI as incoming lines. not configurable. this system needs to change to VOIP. something needs to source these ISDN signals.
06:42.51Strom_Cright, that's what i'm asking - does the VoIP carrier you're considering using only provide service if you use their terminal adapters?
06:43.15dongsyes, but that isnt even relevant. even if it was something else, it needs to go into pbx as ISDN.
06:43.42Strom_Cyes, I understand that
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06:44.06Strom_Cif the ITSP only provides service if you use their terminal adapters, then the first thing you need to do is find another ITSP
06:44.34Strom_Csecond, you can set up an Asterisk box with an ISDN line card where the individual ports are configured in NT rather than TE mode
06:44.35ectospasmdoesn't Vonage do that?
06:44.48Strom_Cyes, and vonage is a terrible ITSP
06:44.55dongsso what is this "NT vs TE" mode
06:45.01jqlwheeeee, vonage
06:45.14Strom_CTE is terminal equipment; NT is network terminal
06:45.24jqlthey're still counting their blessings
06:45.32Strom_Cone is basically telco-facing, the other is CPE-facing
06:45.43dongsright, thats what I was initially asking for.
06:46.01dongsso I need to provide "NT" or whatever signalling.
06:46.10Strom_Cyeah, but when you're only providing bits and pieces of information, it's tough to actually help you
06:46.12dongsis this something that nobody does besides telcos?
06:46.15ectospasmIf you're doing PRI you can do pri_net
06:46.18ectospasmfor signalling
06:46.22Strom_Cectospasm: it's BRI
06:46.25ectospasmAh
06:46.34jqlif you're the NT side, you decide what it is. :P
06:46.35Strom_Cdongs: you can get many different ISDN BRI line cards which do both modes
06:46.44jqlhah
06:46.47Strom_Cthe digium B410P is one example
06:46.52MACscrDoes anyone in here know what a Smart Jack is? I know the basic premise, but it that connection still needs a T1 compatible router, etc, right?
06:47.25Strom_CMACscr: it's basically a jack that does a hard loopback on the T1 circuit if the CPE is unplugged from it
06:47.51Strom_CMACscr: though it could also be an HDSL terminal
06:47.56MACscrCPE?
06:48.03dongshrm.
06:48.05Strom_CMACscr: customer premise equipment
06:48.08Strom_CMACscr: read this
06:48.11Strom_C~101
06:48.12jbotmethinks 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony.  You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf
06:48.15MACscrAh, ok
06:48.34MACscrSo my original assumption was correct?
06:48.47Strom_Cyes, you still need T1 terminal equipment
06:48.55dongsthat B410P looks good except Im not planning to involve a PC in this
06:49.10dongsand that has S/T interface.. how do you turn that into twisted pair?
06:49.17MACscrThanks, i appreciate it
06:49.21Strom_Cdongs: you change a jumper on the card
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06:49.42Strom_Cdongs: do you need a U interface to go to the PBX?
06:50.34dongsit seems thats the name for it, ys
06:50.45Strom_Cah
06:50.50dongshere its called something else but.
06:51.16Strom_Cwell, as we sometimes like to say in the telecom industry, I think you're fucked ;)
06:51.34Strom_Cif you can put a PRI line card in that PBX though, then your options change dramatically
06:51.49dongsya that isnt happening
06:52.01Strom_C~wglwat
06:52.02jbotit has been said that wglwat is well, good luck with all that
06:52.02dongsthing is its not really a one-off thing.
06:52.17dongsthere is a LOT of equipment that only takes isdn and is not configurable/upgradable.
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06:53.25Strom_Cdongs: i've never in my life seen anything that spits out an ISDN U interface besides telco central office switches
06:53.36dongs:(
06:53.49dongsbut thats exactly what I want ..
06:54.54Strom_Cwell then either you're tied to ISDN BRI service from the telco, or you need to reconsider how to accomplish the ultimate goal of "reduce telephone bills by using VoIP instead of NTT or whoever"
06:56.00dongscan any random ISDN PCI card work in NT mode?
06:56.10Strom_Cit depends on the PCI card
06:56.19Strom_Cbut that's an S/T interface, not a U interface
06:56.22dongsthen again it doesnt really matter as a PC is not to be involved in this
06:56.24dongsright.
06:56.26dongsthat too.
06:56.30dongstheres nothign that converts?
06:57.15dongshttp://www.pulsewan.com/wan/nt1_ace.htm
06:57.22dongsopposite of this would be nice.
06:57.27Strom_Cyeah, I have an Adtran NT1 ace
06:57.32Strom_Cthere is no "opposite"
06:58.01Strom_Cwhy are you so dead-set against using a PC?
06:58.26dongsportability/scalability
06:58.31dongs+ cost
06:59.20Strom_C"portability"?
06:59.31Strom_Care all your PBXes mounted in cars or something?
07:00.17dongsno, but I dont see how a PC solves this problem cleanly
07:00.27dongs15:49 < Strom_C> dongs: you change a jumper on the card
07:00.36dongs^ this makes 410whatever output U interface?
07:00.38Strom_Cthat's to go from TE to NT mode
07:00.40Strom_Cno
07:00.42dongsi see.
07:00.46Strom_Ci misunderstood what you were asking
07:00.54dongswell, so then thats still not a solution
07:01.04Strom_Cdongs: what I'm getting at is that you replace the entire phone system
07:01.15dongsno no.
07:01.18dongsthats not a solution.
07:01.30Strom_Camortize the cost of the new equipment against the reduced telephone bills
07:01.42dongsi said this is not a one-off project.
07:02.30Strom_Cwell, then flesh out your situation in more detail, because currently you sound to me either like a complete lunatic or a horiffic cheapskate or someone who has built up a whole business plan on a terribly faulty set of assumptions
07:03.11dongsoh, the business plan is *very* clear.
07:04.13Strom_Cwell, then clarify it :)
07:04.32dongsfinally finding things of relevance on googl
07:05.48dongsStrom_C: analog to ISDN converters, its the future man.
07:06.00Strom_Cyeah
07:06.08Strom_Cso tell me this business plan of yours
07:06.40*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net)
07:06.51dongs1. make analog to ISDN converters 2. price them about half price of voip-capable pbx eqiupment 3. ??? 4. profit!
07:08.09dongsi need to check what local stuff uses for u interface signalling
07:08.10WonkaISDN-TAs cost about 35EUR, afair
07:08.23Strom_Cwell I suppose in theory it sounds good, but trust me, you're barking entirely up the wrong tree.
07:08.30Wonkaother way round... urghs.
07:08.35Wonkawho would want that?
07:08.45Strom_Cputting extra D/A conversions in a circuit is not the way to reliable telephony
07:08.59Strom_Cnot to mention that you're taking a gigantic step backwards signaling-wise
07:09.28dongsyep, absolutely.
07:09.38Strom_Cyou are clearly completely loony.
07:09.52dongsthe benefits are clear
07:10.22dongsyou can convert a 20 years old ISDN pbx into voip without even touching anything.
07:12.02Strom_Cyeah, so you can squeeze maybe another five years out of the thing before you have to replace it anyway
07:12.16dongshttp://www.national.com/opf/TP/TP3410.html
07:12.25dongsthey even still make this stuff.
07:12.49Strom_C~wglwat
07:12.49jboti guess wglwat is well, good luck with all that
07:12.56dongsthanks !
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07:47.32tzafrir_homedongs, unless you consider analog to be the past
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09:03.38dongswhat.
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10:22.17knarflycan I use a bunch of the available flags for dial, like exten => s,1,Dial(SIP/101,,dtTwWf0hH)
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10:34.22tzafrir_homeknarfly, yes
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10:59.54viperdudehi anyone around? I have a problem starting Asteirsk
10:59.56viperdudeasterisk
11:01.02Davieyviperdude: i don't
11:01.34DavieyYou really need to explain your problem, not just saying "it's borked"
11:02.06viperdudewhen I start asterisk it hangs after Asterisk Event Logger Started /var/log/asterisk/event_log
11:02.12viperdudeasterisk 1.4.10
11:16.12knarflywhat does the skip flag do?
11:17.34knarflygot it...
11:40.21knarflyI took one of my favorite mp3's and converted it to .wav format
11:40.46knarflythen I moved the existing .wav files out of /moh and moved my new file in there
11:41.04knarfly* says it can't find the file even though it calls it by name
11:41.16knarflyin the error message
11:41.37Strom_Mwhat's the exact error message?
11:42.07knarflyhold on I can't paste it
11:42.42knarflynot in mono 2
11:42.54knarflyunable to open format wav
11:43.11Strom_Mwhat format is the wav file?
11:45.31Strom_Mknarfly: hello?
11:45.53knarflyyes, I'm saving it down to mono format and trying again
11:45.58Strom_Mno
11:46.12knarflywhat should I try?
11:46.17Strom_Mit must be mono 8khz 16-bit
11:46.29knarflybrb
11:49.05knarflyStrom_M, thanks, it's working now
11:49.26knarflywill mp3 produce better quality than wav
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11:55.12knarflycan someone help me with my extension
11:55.14knarflyhttp://pastebin.com/d4266b81d
11:55.38disgrntld81is there any official asterisk reference documentation? i'd like to see a complete overview of all options available per config file, per contexts etc..
11:56.09disgrntld81in particular i'd like 1.4 documentation
11:56.15knarflydisgrntld81, do a google search for TFOT, you'll find it
11:56.15Optichmm good question
11:56.26Opticthe definitive guide is the source code ;)
11:56.42disgrntld81heh
11:56.43disgrntld81sure
11:56.49disgrntld81i have TFOT
11:56.57knarflydisgrntld81, and the voip-info website
11:57.11Opticyes, voip-info is the first place i look
11:57.20knarflydisgrntld81, that's about all that exists outside of the code and the forums
11:57.32disgrntld81ok
11:57.37disgrntld81just making sure i wasnt missing something
11:57.41disgrntld81thanks
11:58.02Opticno, * is a bit underdocumented
11:58.04knarflyI'm reading TFOT now and I keep finding new stuff...it's pretty good, but it's not up to date with 1.4.11
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11:58.27disgrntld81yea that is what im seeing too out of TFOT
11:58.33knarflyOptic, you get what you pay for, right
11:58.34Opticopen source programmers like to write code, not docs
11:59.05knarflywell can anyone tell me why my extension is hanging up too soon
11:59.06knarflyhttp://pastebin.com/d4266b81d
12:00.11Optici love * though, it's a nice piece of work
12:00.17disgrntld81it is very cool =)
12:00.24Optictaking a quick look
12:00.43disgrntld81i'd rather enjoy it poorly documented than to never have enjoyed it all
12:00.55viperdudewhen I start asterisk it hangs after Asterisk Event Logger Started /var/log/asterisk/event_log
12:00.56viperdudeasterisk 1.4.10
12:01.04viperdudeany ideas?
12:01.22knarflyI started * about 18 months ago...it's slick and let's you give it to the man (POTS)
12:01.24Optici don't remember what 'n' means in the dialplan
12:01.41viperdudeOptic: n stands for next
12:01.45Opticit let me give it to norstar :)
12:01.46knarflyn means number the next step in sequence
12:02.16knarflythat way when you add new lines you don't have to renumber each of them
12:02.37Opticthat's cool... must be new :)
12:02.54knarflybeen there since I started using * in 2006
12:04.07Opticwe outgrew our norstar 6x16 at work
12:04.31Opticreplaced it with a dell P4 box, a digium PRI card, and *
12:04.38Opticnow we're over 30 extensions and it's ticking along just fine
12:04.56Opticmostly polycoms
12:05.28Optictried a few of the sipura-841's at first, nearly put me off voip :)
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12:11.05tzafrir_homeYou Get What You Paid For? You Get What O'reilly Paid For?
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14:02.45ManxPowerWake up geeks!
14:07.09coppicewhy? it bedtime
14:09.44tzafrir_homehi ManxPower
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14:11.20ManxPowerHeloo, tzafrir
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15:04.46FXOLmorning
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15:12.15FireMachi
15:13.10FireMacanyone selling their asterisk book
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15:14.34MukulJainHi,
15:14.42FireMachi
15:14.43MukulJainI have a question regarding canreinvite
15:14.51MukulJainI am trying to use freecall
15:15.01MukulJainI can make call when I put CanReInvite=no
15:15.04FireMacsorry bro, i just started learning asterisk
15:15.09MukulJainBut when I turn it as Yes, the Audio is only one way
15:15.18MukulJainCan anyone help ?
15:15.37[TK]D-FenderMukulJain, Reinvites through NAT don't tend to work.
15:16.08[TK]D-FenderFireMac, ...
15:16.09[TK]D-Fender~book
15:16.10jbot[book] Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 or Temporarily at http://www.aocomputing.net/AsteriskTFOT.pdf
15:16.11[TK]D-Fender^^^^^^^^^^^^
15:16.35[TK]D-Fender*sigh*
15:16.48MukulJainI see, so if my phone gets Public IP, then it should work ?
15:17.41[TK]D-FenderMukulJain, Yes, basically the phone has the eiter be quite aware of the NAT its behind and have ports forwarded so that the reinvite maps through, or being pubhlic should allow it just fine
15:18.05MukulJainI am using GrandStream, Are these phones be made aware of NAT
15:18.10MukulJainthat's they are behind ?
15:18.20[TK]D-FenderMukulJain, let me put it to you this way :
15:18.23[TK]D-Fender~gs
15:18.23jbotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
15:18.27[TK]D-Fender~grandstream
15:18.27jbotextra, extra, read all about it, grandstream is the Yugo of VoIP hardware.  Run.  Run away now.
15:18.39[TK]D-FenderMukulJain, Don't get any ideas :p
15:18.53MukulJainSorry, but I am using them just for testing,
15:19.08MukulJainbecause they are affordable for the test setup,
15:19.30MukulJainSo because they are cheap is that correct that they cant be made NAT aware ?
15:19.36MukulJainand expensive phones like Polycom can be ?
15:19.40[TK]D-FenderMukulJain, ATA's are "affordable" and you'd be better off...
15:19.54coppicewhat exactly is a test setup testing when its components are buggy?
15:19.56MukulJainI have Cisco ATA 186
15:20.06[TK]D-FenderI don't know for sure what the GS can do in that dept....
15:20.35MukulJainI am using GS 2000 , GS 2020 , Cisco 186 ATA, Polycom 310 in the test lab
15:21.06[TK]D-FenderMukulJain, Linksys ATA's are probably the most economic choice world-wide
15:21.16MukulJain[TK]D-Fender : I have linksys PAP2 as well.
15:21.26MukulJainPAP2T
15:21.40MukulJainMy Scenario is :
15:21.57MukulJainI have asterisk at my office (behing NAT on Dynamic DSL) mapped thru dynamic dns
15:22.14[TK]D-FenderMukulJain, But I'm not sure which would best handle a NAT'd reinvite best.  The polycom can be informed of its public IP and if your port forward to it and have * treat it like it ISN'T behind NAT then I'd bet it would work.
15:22.16MukulJainI have 1 Ip phone at my home (behind Linksys wireless router). This is GS 2000
15:22.27[TK]D-FenderMukulJain, I take it you have some serious bandwidth considerations?
15:22.49MukulJainI have phones located in India, but asterisk is in Singapore
15:23.04MukulJainnow when India calls another India phone, they go via Singapore for media path
15:23.14MukulJainI want to avoid this and make it direct, to have better quality.
15:23.23MukulJainam I on correct way ?
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15:24.43mvanbaaksounds like you cannot do that, because both phones will be behind NAT
15:25.02[TK]D-FenderMukulJain, if you had a server in India they could use that one, but it depends on if you want to setup multiple servers to handle dialing between regions.
15:25.29[TK]D-Fendermvanbaak, With properly aware phones & forwarding it can work for a single phone per site
15:25.39MukulJain[TK]D-Fender: yeah that's the way to do it, but I dont want to put a server just for 2 phones at India, it's more about managing them,
15:26.05MukulJainand I forgot to tell, all the phones and server are on Dynamic DSL.
15:26.17[TK]D-FenderMukulJain, Take a look and give it a try.  Indeed I would think it strange to work very hard for jsut 2 phones...
15:26.36yangHow come that I don't see a log if someone tries to reach me by SIP protocol ? Not by calling in over PSTN
15:26.54[TK]D-FenderMukulJain, that of course makes it harder to control the NAT aware settings.
15:26.57MukulJainI had cisco documentation, where they have said that all the Cisco phones working with call manager, media path does not goes thru the CM but it's directly from Phone to phone.
15:27.18[TK]D-Fenderyang, what kind of log are you expecting?
15:27.28yangI can only see this request : Unknown SDP media type in offer: video 5022 RTP/AVP 31
15:27.31mvanbaak[TK]D-Fender: ah, a single phone per location should work indeed
15:27.43[TK]D-FenderMukulJain, Well thats also SKINNY, not SIP.
15:27.47yang[TK]D-Fender: I expect to see the request and then refused request if the destination cannot be reached
15:27.58mvanbaakskinny++
15:27.59[TK]D-Fenderyang, in * CLI : "sip debug"
15:28.00MukulJain[TK]D-Fender : So using Skinny is that possible ?
15:28.08MukulJainIF I use Cisco Phones in India ?
15:28.19MukulJainand make them connect using Skinny on Asterisk ?
15:28.20[TK]D-FenderMukulJain, Not sure, Cisco is a lot of trouble to handle./
15:28.42MukulJainumm, so what would be best way, for this scnario :
15:28.49[TK]D-FenderMukulJain, mvanbaak has a fair amount of experience with them, he'd be more able to advise than I am.
15:29.11MukulJainServer : In Singapore, IP Phone 1 -- India City 1 : IP Phone 2 : India City 2 : Ip phone 3 : My house
15:29.44yang[TK]D-Fender: I also have one problem, i can call outside my SIP (my softphone is NAT-ed asterisk isnt), but i cannot be reached inside over SIP number
15:29.57MukulJainmvanbaak: What is the best way to setup, considering quality would be effected if asterisk is always there in between ?
15:29.58mvanbaakI did not try skinny phones behind a nat with a server on another location
15:30.02[TK]D-Fenderyang :
15:30.04[TK]D-Fender~sipnat
15:30.04jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3  otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
15:30.05[TK]D-Fender^^^^^^^^^^^^^^^^^^^
15:30.06mvanbaakbut I think it's the same as with SIP
15:30.15mvanbaakyou will have to setup forwarding etc
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15:30.38MukulJainMoment I do forwarding, then I have trouble when I plant the second phone in the same location righ ?
15:30.45mvanbaakyup
15:31.04MukulJainAnother problem I have is
15:31.09MukulJainI have SIP Provider : say voipcheap.com
15:31.16MukulJainand I am using it from my home sip phone
15:31.21MukulJainwhich is behing NAT.
15:31.34[TK]D-FenderMukulJain, First take a look with the phones you have to see if they support STUN and see if that helps it with knowing its IP and tunneling characteristics.  Forward the appropriate port range to them and tell * they AREN't behind NAT and give reinvites a try.
15:31.53MukulJainOkay GS support Stun
15:32.07mvanbaakthat's what they tell you ;)
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15:32.14mvanbaakI dont trust them
15:32.25MukulJainIP and Tunneling Charactertics : can u elaborate ?
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15:32.35MukulJainPort Range , u mean RTP range ?
15:32.43mvanbaakyes
15:32.53MukulJainokay that I can open on the local router
15:32.54[TK]D-FenderMukulJain, STUN helps tell your device what kind of NAT and port forwarding it can detect.
15:33.38MukulJain[TK]D-Fender : Which are better phones, I found Polycom are expensive, WOuld a linksys be recommended ?
15:34.09[TK]D-FenderMukulJain, Definately better than GS, but go experiment with what you have.
15:34.31MukulJainI can get only GS from a local store nearby in SG
15:35.07MukulJainhence testing them, I got 2020 and found it to be good ! but seems that there are lot of people who dont like GS at all.....so far for me for testing it's been okay...
15:35.22mvanbaakI have not tested the 2020
15:35.56[TK]D-FenderMukulJain, SG & Indian pricing is quite different than North America.  Here Polycom phones are about the same as Linksys.
15:35.56[TK]D-FenderMukulJain, relative value is different where you are, and the choices a bit harder.
15:36.19MukulJainmvanbaak : So If we have phones planted over WAN (Internet) and they are behind NAT, media has to go thru asterisk. Is this normal the way everyone does it ?
15:37.05MukulJainI feel that above way it makes the latency higher,
15:37.21MukulJainand effect quality, even though bandwidth is secondary for me to look at also
15:37.36mvanbaakit's how we do it
15:37.46[TK]D-FenderMukulJain, YES, THATS THE WAY PRETTY MUCH EVERYBODY DOES IT.
15:37.56[TK]D-Fenderoops.. caps
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15:38.23MukulJainthere are so many Hosted Providers for hte ASterisk, giving extensions free. like Orbtalk etc, does all media passes thr the server ? because they have some thoudands of users
15:38.25mvanbaaksometimes we put a small box with asterisk at the customer location if they make more then 75% local calls
15:38.34[TK]D-FenderMukulJain, It can, but few have any actual issues with it.  You however are spanning a large distance with less than idea pipes between them.
15:38.45mvanbaakMukulJain: most of them dont use asterisk
15:39.00MukulJainso what else is the choice ?
15:39.02mvanbaakthey use ser most of the time
15:39.08MukulJainser ?
15:39.19mvanbaaksip only thingie
15:39.21[TK]D-FenderMukulJain, Any decently sized solution uses something like SER in front or another industrial strength soft-switch
15:39.52MukulJainSip Express Router u mean
15:40.04yang[TK]D-Fender: I have nat=yes and qualify=yes to enable my nat-ed phone...http://paste.debian.net/37744
15:40.34MukulJainyang : What is your scenario ?
15:41.33[TK]D-Fenderyang, *I* don't see "nat=yes" on that entry...
15:43.04yang[TK]D-Fender: i added it recently
15:43.10yanglet me try now
15:43.24ManxPoweryang: so you are giving us out of date info?  Way to go, dude.
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15:44.28[TK]D-FenderManxPower, Good morning starshine!
15:44.50yangI would need someone to ring me to test the thing, please?
15:45.58[TK]D-Fenderyang, use a cell-phone or something
15:46.13yangi dont have a pstn binded to it
15:46.23yangonly sip
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15:48.49MukulJainA silly question, Sorry from a Newbie: Is SER is like Asterisk ?
15:49.42[TK]D-FenderMukulJain, No.  SER is a SIP proxy, and a lot of other bits all in one.  * is a toolkit that you can make a PBX with, etc, but its SIP stack and scalability sucks
15:49.56[TK]D-Fenderyang, Go find something else to test with.
15:52.11MukulJain[TK]D-Fender : So we can use SER coupled with Asterisk ? and that would be better solution if there are lot of users on the internet using the asterisk ?
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15:54.05[TK]D-FenderMukulJain, in general I would say yes, but this is a BIG topic and deserves a lot of research to understand at what point it becomes profitable to do so
15:57.06wulfy814I've got a fresh install of 1.4.11 on ubuntu LTS with a Sangoma A101 that I'm having a bit of trouble with
15:57.13wulfy814I've got it accepting calls
15:57.56wulfy814the last line of the incoming context is exten => in-num,n,Background(demo-thanks)
15:58.10wulfy814the file exists in /var/lib/asterisk/sounds
15:58.25wulfy814and the console doesn't throw any errors, but I'm not hearing the sound file on the other end...
15:59.59wulfy814<----- IDIOT
16:00.09wulfy814I checked /var/log/asterisk/messages and found the problem
16:00.16wulfy814was missing the =
16:00.20wulfy814n/m
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16:42.35MukulJainPlease suggest any good billing software for the ASterisk ! Prefeably GN
16:42.38MukulJainGNU
16:46.03puzzleda2billing
16:50.36_x86_MukulJain: you mean GPL?
16:51.14ManxPowerBilling packages are like candidates in an election -- they are all bad, it's just that some are worse than others.
16:51.28ManxPowerMuch like softphones, actually.
16:51.53ManxPowerFortunately, I don't bill for calls.
16:52.57_x86_yeah it's best to design your own, exactly how you want it
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16:57.44linageewow. gxp-2000 is basically a giant piece of crap when it has firmware 1.1.4.18 loaded on it. might as well throw it out the window (??)
16:57.58linageelocks up, never boots up the first time, ?
16:58.12ManxPowerlinagee: Grandstram stuff are giant pieces of crap no matter what firmware you use.
16:58.22ManxPower~gs
16:58.22jbotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
16:58.28linageeManxPower: it was working for me just fine for a year, but when it had the older code
16:58.54ManxPowerlinagee: don't ever upgrade a working GS device.
16:59.04ManxPowerWorking firmwares are few and far between
16:59.06linageeManxPower: lol. true. not sure what possesed me.
16:59.21linageeManxPower: and not revertable in this case!   :(
16:59.32ManxPowerlinagee: it sucks to be you.
16:59.40linageeManxPower: it sucks to be my phone
17:01.03linageeManxPower: does any other vendor have that problem of crappy firmwares and non-revertable-ness?
17:02.07ManxPowerlinagee: Most vendors have non-revertable-ness.  It's just that it is unusual for other vendors to release broken firmware -- it is unusual for GS to release a working firmware.
17:03.30linageeManxPower: hunt down their dev team and say, "WTF WHERE YOU THINKING"
17:04.15linageeor even better yet, "WHY HAVEN'T YOU RELEASED A FIX FOR THE BUGGY CODE YET?"
17:05.11ManxPowerHow about, "WHY DID YOU BUY THAT PIECE OF CRAP?"
17:05.14linageeManxPower: maybe it's part of their business practices now. 1) Release working phone 2) Release buggy non-revertable firmware 3) Hope phones break from upgrade and people have to buy new ones with older code (?)  4) ???  5) Lots of Profit!
17:05.25ManxPowerThe only reason GS is still in business is that people continue to buy their hpones.
17:06.01ManxPowerMaybe they release their firmware with no formal testing -- sort of like Digium does with their software.
17:06.03linageeManxPower: barbietone-100 sucks even more. can't hear squat
17:06.15coppicethat's a pretty profound way to stay in business. runs completely against the .com ethos
17:07.04ManxPowersorry, they way they do with their Open Source software.  Their non-free stuff is formally tested, as I understand it.
17:07.27linageeManxPower: sorry? yes i would appreciate a sorry from them
17:07.33linageeManxPower: "sorry for buying our crap"
17:07.35linageelol
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18:20.53FogertyVoiceMailMain application has has an advanced option "Envelope", what sort of information does that play for the caller?
18:21.48wunderkintry it and see
18:23.49Fogertythanks. :) Any one else?
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18:29.25ManxPowerTry it an see.
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18:45.30Strom_Chttp://www.pizdaus.com/pics/C7BXGZFagqnT.jpg  <--- how wiring should not look
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18:57.05ManxPowerThat looks like my old NOC!
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19:25.10riddleboxis there a place to tell asterisk to send the wav file in an email when a voicemail is left as whatever.wav not whatever.WAV?
19:34.17tzafrir_homeformat= ?
19:35.48Corydon76-digattachfmt?
19:36.55riddleboxohh geez it was right in front ontop of the page
19:37.41riddleboxI looked through the whole file and never even gave that a thought
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21:13.55moprilohi guys .. a quickone, on my plan, i have whole order when people dial _1.,1,gotoif...    i don't want to rewrite everything for _001.,1...  since its the same but {EXTEN:2}  how do i do that?.. it could even be EXTEN (sent with the 0's)
21:14.42moprilogoto(_1.,1)  right?
21:14.47[TK]D-Fendermoprilo, pastebin what you've got so we have a better idea
21:15.11[TK]D-Fendermoprilo, sounds like you'd be better off using a macro
21:15.49mopriloi don't remember the pastebin link, so since is almost the same.. is like this one ( http://www.ianbezanson.ca/drupal/node/164 )
21:16.16moprilobut i need to add the exten=> _001., ..
21:19.54mopriloI'm gonna try the goto(_1.,1) see how it goes :P
21:21.26ManxPoweryou can't goto a pattern match!
21:21.50ManxPoweryou goto a specific location, a pattern match could match that destination
21:22.15ManxPoweri.e. you can't have a leading underscore as part of the extension to go to inside of a Goto()
21:22.52ManxPowerGoto(context,00${EXTEN},1)
21:24.12moprilook..
21:24.26ManxPowerriddlebox: you don't want WAV, those are HUGE.
21:24.43ManxPowerwav49 is the format that seems best for generic voicemail attachments for e-mail
21:25.11mopriloso i can use Goto(${EXTEN:2},1)   like that?
21:25.23ManxPowerBTW, does anyone know of a decent ITSP that does not charge for incoming calls to DIDs.  Teliax charges 2c/min for incoming.
21:25.42ManxPowermoprilo: that would strip off the first 2 digits of the value of EXTEN, yes you can do that.
21:25.57ManxPowerHell, half my dialplan uses gotos to rewrite what users dial
21:26.17[TK]D-FenderManxPower, I've seen that on those that charge like $10/ea and for outbound, but not without a base charge
21:26.50ManxPower[TK]D-Fender: Teliax DIDs have a monthly fee, I don't have too much of a problem with that.
21:27.00ManxPowerBut I would prefer fee with no per min.
21:28.36[TK]D-FenderManxPower, http://www.vitelity.com/index.php?p=retailserv
21:28.48ManxPowerare they actually any good?
21:28.52[TK]D-FenderManxPower, no per-min = pure fixed charge FL1 effectively.
21:29.09[TK]D-FenderManxPower, I've heard of a lot of people using them... no personal experience though
21:30.12ManxPowerthe low monthly + per min is also acceptable.
21:30.18ManxPowerI guess I'll give them a try.
21:30.39[TK]D-FenderManxPower, hope it pans out for you.
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21:38.32E-bolaAre there a command to "pick" up a call in a queue?
21:38.50E-bolalike press a button and then you get the 1st call waiting in the queue
21:44.38mopriloManxPower, exten => _001.,Goto(${EXTEN:2},1) is not working.. any ideas?
21:45.25[TK]D-FenderE-bola, nope.  gotta wait your turn
21:45.50moprilojmm.. maybe if i mark it with another name
21:45.58E-bola[TK]D-Fender: I have this weird setup for a client im trying to fix
21:46.21[TK]D-FenderE-bola, start by cutting above the testicles :D
21:46.30E-bolaHe has a soundcard in the asterisk server, when a incoming calls apears, the soundcard need to signal that event with a sound over the speakers in the ceiling
21:46.41E-bolaSo once an employee hears the noise
21:46.47E-bolahe must "pick up" the call waiting
21:46.57E-bolaCan asterisk support something like this?
21:47.05E-bolaSo far ive managed to get console audio working
21:47.05[TK]D-FenderE-bola, sounds more like he should be using parking....
21:47.15E-bolaNow i need to know how to pick up the call
21:47.31[TK]D-FenderE-bola, have * park the call to queue the,
21:47.41[TK]D-Fenderthen*
21:47.43E-bola[TK]D-Fender: You mean park all incoming calls?
21:47.55[TK]D-FenderE-bola, for those you want to treat this way, yeah
21:48.09E-bolaok, how do i signal that there are calls waiting over the speakers then?
21:48.29[TK]D-Fendermoprilo, just make your action into a macro and have your extens call it passing the modified exten # as needed
21:48.49[TK]D-FenderE-bola, before parking use a call-file or something
21:49.18[TK]D-FenderE-bola, if you want * to play that sound to Console
21:49.32[TK]D-FenderE-bola, or jsut do a System call and play an MP3 over it or something
21:49.44E-bolaIm sory, i dont know what a call-file is....
21:50.06[TK]D-FenderE-bola, look it up on the WIKI.  And come up with the mechanism to handle this.
21:50.16[TK]D-FenderE-bola, there are a lot of different ways
21:50.39E-bolaHmm ya thats for sure hehe
21:50.48E-bolaCouldnt i also use PickUP() ?
21:50.52E-bolaon console
21:52.45[TK]D-FenderE-bola, no, because the call has technically been answered.
21:52.56[TK]D-FenderE-bola, by app_queue
21:52.57E-bolaWhy? I can just set autoanswer to no?
21:53.07E-bolawell if i dont use a queue
21:53.19E-bolajust let calls go directly to console
21:53.36[TK]D-FenderE-bola, You need to actually think before you start spitting out ideas like that you know :p
21:54.01E-bolaIm thinking as well as I can :P
21:54.35gremzoidthats my phase for today
21:54.38gremzoid:P
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21:55.48E-bolaHmm this really isnt clear to me how to archive....
21:56.09E-bola[TK]D-Fender: Is this the only info about parking calls? http://www.voip-info.org/wiki-Asterisk+call+parking
21:56.14E-bolaIts the only thing i can find ont he wiki
21:56.43[TK]D-FenderE-bola, theres that, and also "ValetParking"  go look that up too
21:57.14riddleboxManxPower, the problem I am running into is that moto q phone says that they are unrecoqnized formats when I use anything...
21:58.55riddleboxcan you use attachfmt=mp3? in voicemail.conf?
22:00.16ManxPowerriddlebox: Oh!  I thought you were using e-mail on a real computer.
22:00.21styelzcan I set seperate localnet= for each sip user/peer ?
22:00.30styelzor is it global
22:00.32ManxPowerriddlebox: well, WHAT formats does Q support for e-mail attachments.
22:00.36E-bola[TK]D-Fender: I dont think you understand my setup, if i park all incoming calls on the parkinglot, how will users know what extension to call to pick up the parked calls?
22:00.42E-bolaIt needs to be the same extension everytime
22:00.45ManxPowerstyelz: It is global and there is no real reason for it not to be.
22:00.48[TK]D-Fenderriddlebox, no, because * cannot ENCODE in MP3
22:00.52E-bolaOr maybe im missunderstanding something....
22:00.53ManxPowerlocalnet specifies the LOCAL NETWORK of the Asterisk.
22:01.09ManxPowerE-bola: you tell them.
22:01.12[TK]D-FenderE-bola, there are easy ways to tell which lots are in use...
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22:01.36riddleboxManxPower, it says it supports wav, but for some reason I cannot get it to open them
22:01.39E-bolai guess i could use the LED's on the phones with hints...
22:01.42E-bolabut thats not a good solution
22:01.56E-bolamu users must not be bothered/disturbed
22:01.57ManxPowerriddlebox: you have the choice of wav (ualw/alaw) wav49 (MS GSM), GSM
22:01.58riddleboxE-bola, I use that right now on my phones
22:02.06E-bolathey just need to be able to hit a shortcut key ont he phone
22:02.12[TK]D-FenderE-bola, You can figure these things out in pure dialplan....
22:02.12E-bolaand be connected with any potential waiting call
22:02.12ManxPowerriddlebox: wav files are about 1 meg per min
22:02.33ManxPowerE-bola: perhaps a Queue
22:02.36ManxPower?
22:02.39[TK]D-FenderE-bola, take aloot at ValetParking,  It can queue up calls on a single lot rather easily
22:02.55styelzyah, if i dont set localnet.. i only get audio one way.. (although, calls i send direct to a queue work). But if i set localnet= i get audio both ways, but if i send a call direct to a queue i get no audio
22:03.00E-bolaManxPower: Was my first idea to, but you cant "pick up" a call waiting in a queue by hitting a key?
22:03.15ManxPowerstyelz: localnet= is for Asterisk being behind NAT
22:03.25styelzyea it is behind nat
22:03.41styelzits on 192.168.0.6
22:03.56ManxPowerE-bola: hell if I know.  However, you might be able to send queue calls to a specific parking slot.
22:03.57E-bola[TK]D-Fender: i can find a readme from 2004 about app_valetparking, does it even work with 1.4?
22:04.06ManxPowerif that parking slot is busy, then the call stays in the queue.
22:04.18[TK]D-FenderE-bola, it's been modified for 1.4 sometime ago.  Get searching and go try stuff
22:04.25styelzit all works great. except for calls i derect direct to a queue.
22:04.33ManxPowerI have not ACTUALLY done this, I'm just giving you ideas.  What you want to do really pretty crazy.
22:04.34styelzwhich is odd
22:04.43ManxPowerstyelz: define "direct to queue"
22:04.53E-bolaManxPower: I knwo its a bit odd, its not my idea to design the system like this
22:05.03ManxPowerthere should be no difference between a call and a qcall
22:05.51styelzan extension in queues.conf
22:05.56ManxPowerE-bola: Just report the person as a "telecom terrorist" to your local equiv of the FBI.  End of problem.
22:06.18ManxPowerstyelz: you need to figure out what is special about that, since there should not be any difference.
22:06.35styelzyea its got me rooted
22:06.42styelzbeen at it for a week
22:07.01styelzit works fine it i dont set localnet though
22:07.02ManxPowerstyelz: perhaps it would be easier to put Asterisk on a public IP
22:07.14ManxPowerstyelz: do you ALSO have externip= set?
22:07.26styelzyea, most definately
22:07.53*** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk)
22:08.15ManxPowerstyelz: what is the actual localnet= setting.
22:08.31styelzbut then, clients at work will all be natted to *
22:08.48styelzand id rather just have the sip provider natted and the rest not
22:09.07ManxPowerstyelz: Uh, that is not true.
22:09.27outtolunca port is a port of course <G>
22:09.29[TK]D-Fenderstyelz, read this, NOW :
22:09.30[TK]D-Fender~sipnat
22:09.37jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3  otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
22:09.37styelzif i put * on the DMZ
22:09.40[TK]D-Fender^^^^^^^^^^^^^^^^^^^^^
22:09.49ManxPowerWe have this layout:  Internal private addresses <-> public servers <-> internet.
22:09.53E-bolaCan you use Pickup() to pickup a parked call?
22:10.04E-bolathat would be super, so you prolly cant :(
22:10.16ManxPowerinternal addresses are natted when they connect to the internet, but not natted when they connect to the "public servers"
22:10.30ManxPowerE-bola: uh, yes.
22:10.55E-bolaManxPower: Doesnt pickup() only work with extensions?
22:10.56ManxPowerof course, your ISP needs to give you more than 1 public address, and you need a NAT router that is not braindead
22:11.06ManxPowerE-bola: that is DirectedPickup.
22:11.07E-bolaOr does your parked calls count as an extension?
22:11.17ManxPowerE-bola: when was the last time you did a "show applications like pickup"
22:11.35styelzi dont see why a normal sip call is different from a queue call..
22:11.44E-bolaabout never....
22:11.45ManxPowerstyelz: It should not be.
22:11.53styelzonly that it resides on the * box
22:11.54ManxPowerE-bola: try it
22:12.16E-bolajust did, doesnt tell me anything really?
22:12.22E-bolaPickup: Directed Call Pickup
22:12.27*** join/#asterisk jstivers (n=mirc@adsl-70-238-188-129.dsl.stlsmo.sbcglobal.net)
22:12.46styelzspose i can just not use queues
22:13.38E-bolaWhat i was thinking was that if i could use Pickup() with parked calls, i could specify a block like 70 and it would pickup and call on extensions 700-799 as far as i can understand
22:13.52E-bolaand=any
22:14.39styelzTK: read them
22:14.44styelzthanks though
22:15.13[TK]D-Fenderstyelz, that answers on how to setup * behind NAT as well as remote SIP clients.
22:15.33styelzwill try that out
22:15.53styelzseems the best solution, plonk it on the DMZ
22:16.29styelz..not a bug is it... *hides*
22:17.07[TK]D-Fenderstyelz, no need to DMZ, and that is not enough by itself
22:17.49styelzwhere else should i put it?
22:18.03[TK]D-FenderE-bola, do a "show dialplan" while you have some parked calls, and "show application chanisavail" and use your imagination....
22:18.13*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)
22:18.17[TK]D-Fenderstyelz, that first guide told you everything you need to do...
22:18.25styelzits either the DMZ the LAN or the Gateway
22:18.33styelzyes thanks.
22:18.43[TK]D-FenderDMZ the LAN?  HUH!?
22:18.45styelzso. its the DMZ
22:18.50styelzn/m
22:18.52E-bolaManxPower: So say there is a parked call on extension 701, you can use ParkedCall(701) to connect to the call, and you can use Pickup(701) as well?
22:18.58QwellI'm about to stab somebody in the face.
22:19.04styelzgot a few spare ip's
22:19.07Qwellfreaking spammers got to the bug tracker
22:19.14E-bolaQwell: Stab away at me, i can take it
22:20.37styelzdont stab me i got cones to smoke
22:20.41NuggetAfter you guys are done revolitionizing VoIP it'll be time to invent Punch in the Face over IP.
22:20.57NuggetI'd pay bug, big money for a Digium PitFoIP Appliance.
22:21.09styelzlol
22:21.41Qwell-over IP
22:21.51[TK]D-FenderNugget, AT&T has a special channel for that kind of "reach out and touch someone", but it requires a warrant :p
22:21.56Nuggetheh
22:22.27[TK]D-FenderE-bola, really??
22:22.58[TK]D-Fender(my katana)
22:23.44E-bolaSo did anybody know if the Pickup() cmd works with parked calls? I cant test it atm, so im just curious to know if i need to take a different path....
22:25.31[TK]D-FenderE-bola, yes you need a different path.
22:25.53[TK]D-FenderE-bola, and I just gave you the path to the answer 8 minutes ago
22:26.14ManxPowerYou cannot part TO a specific extension
22:26.14E-bola[TK]D-Fender: ValetParking? That didnt sound like a very supported path...
22:26.26E-bolaManxPower: i wouldnt need to if pickup worked
22:26.29[TK]D-FenderE-bola, <[TK]D-Fender> E-bola, do a "show dialplan" while you have some parked calls, and "show application chanisavail" and use your imagination....
22:26.36E-bolait support picking up a "range" of extensions
22:26.54ManxPowerE-bola: you are NOT picking up an extension!
22:27.00ManxPowerYou are picking up a parked call.
22:27.01E-bola[TK]D-Fender: I did both commands, my imagination must be inferior to yours...
22:27.27E-bolaManxPower: well then it wont work, im just saying if that was supported it would be a real simple solution to my problem :(
22:27.40[TK]D-FenderE-bola, Chanisavail test to see which parking lot is in use and grab the first one that is.  Make a little loop script in your dialplan for this.
22:27.49ManxPowerE-bola: you wanted to put a call in limbo and pick it up from any phone.
22:27.53E-bola[TK]D-Fender: ahhh
22:28.03E-bolaManxpower: yep
22:28.18ManxPowerso WHO CARES what you dial to pick up the call.
22:28.21E-bola[TK]D-Fender: Ok you are right, asuming i can figure out how to script that, that should work
22:28.29ManxPowerJust have ONE parking slot.
22:28.45[TK]D-FenderManxPower, No ned to use a single lot...
22:28.50ManxPowerwhen that parking slot is empty, toss the next call to it,
22:29.05ManxPower[TK]D-Fender: he said he wanted to dial the same thing every time to pick up the call.
22:29.06ManxPowerIIRC
22:29.08E-bolai think both your ideas would work
22:29.21E-bolaand both fill my requirements
22:29.26[TK]D-FenderManxPower, make an exten that checks which lot is free and just goes to it.  no big deal.
22:29.40[TK]D-Fenders/free/inuse
22:29.41ManxPowerPark() does that automatically
22:29.42E-bolaManxpower: i CAN dial the same thing everytime, since tk wants me to script a loop that finds an full lot and connect to that
22:29.52[TK]D-Fenderexactly
22:30.20E-bolathere's a cmd called Park()? That wasnt mentioned on the wiki at all....
22:30.34ManxPowerGreylisting sure takes the "instant" out of "instant gratification"
22:30.51[TK]D-FenderManxPower, .... was it good for you?
22:31.05ManxPowerE-bola: There is a command called Park.  Nobody uses it because it is added to the dialplan automatically when you turn on parking
22:31.19ManxPower[TK]D-Fender: TK, you're the best!
22:31.37E-bolaSo far i like TK's idea best, its the simplest to understand, asuming the scripting is possible/easy
22:31.39*** join/#asterisk drwelby (n=mpfister@mail.enplan.com)
22:31.49*** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk)
22:32.47E-bolaouttolunc: you use it?
22:32.53[TK]D-FenderE-bola, ChanIsAvail(Local/${lotcounter}@parkedcalls/n) <---
22:32.59ManxPower[TK]D-Fender: the reciptionist at the 2nd largest office of one of my clients quit.  Instead of just stuffing a body into the chair I had to go do some custom call routing for them
22:33.19ManxPower[TK]D-Fender: does that work in 1.2?
22:33.22[TK]D-FenderManxPower, more money for you....
22:33.23outtoluncit was a joke, dynamic parking = scripting and dynamic meetme channels
22:33.26[TK]D-FenderManxPower, certainly
22:33.39ManxPower[TK]D-Fender: I was busy leaving on a trip.
22:33.46E-bolatk: ive never scripted anything per say in asterisk
22:33.55E-bolaare there like for loops and stuff like that?
22:34.01E-bolaso i can increment lotcounter etc?
22:34.03[TK]D-FenderE-bola, Dear God......
22:34.09ManxPowerE-bola: if you write a dialplan you scrpted
22:34.27E-bolaWell i meant more programicly, if then else, while, for etc
22:34.32[TK]D-FenderE-bola, show application set,gotoif, etc.....
22:34.56[TK]D-FenderE-bola, You are in dire need of...
22:35.00[TK]D-Fender~osmosis
22:35.00jbotwell, osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ...  or at least until your unconsciousness restores peace to the channel ...
22:35.13E-bolahehe its funny how you feel the absolute need to insult me while helping me :)
22:35.16ManxPowerhttp://www.fnords.org/~eric/macro-std-exten-v2.inc
22:35.32ManxPowerE-bola: because you totally and utterly missed what he was saying.
22:35.52[TK]D-Fenderyup
22:36.07ManxPowerwhat he meant to say was: ChanIsAvail(Local/710@parkedcalls/n)
22:36.20E-bolaSaying? He told me to script a loop over the park lots and use ChanIsAvail to see which was full
22:36.23ManxPowerbut since he didn't want to lock you into using 710 as one of your parking lots.....
22:36.27E-bolaand showed me one line with a var called lotcounter
22:36.33E-bolai asume i need to modify that counter somehow?
22:36.38ManxPowerE-bola: script = use the standard dialplan stuff
22:36.42[TK]D-Fenderexten => _XXXX,n(check-dialstatus),GotoIf($["${DIALSTATUS}" = "BUSY" | "${DIALSTATUS}" = "CHANUNAVAIL" | "${DIALSTATUS}" = "CONGESTION"]?vm-busy) <-- 80/20 IF violation :p
22:37.09ManxPower[TK]D-Fender: I don't understand
22:37.46[TK]D-FenderManxPower, we talked about that.. you are checking for 80% of the values, vs the 20% that would let you approach this the otherway around be much shorter.
22:38.07ManxPower[TK]D-Fender: IIRC, I want that other %20 to be ANYthing
22:38.11*** join/#asterisk knarfly (n=jmarley@c-98-203-55-196.hsd1.fl.comcast.net)
22:38.20ManxPowerwho knows what bizaro dialstatus values they might add?
22:38.27[TK]D-FenderE-bola, I INVENTED a counter just so you'd get the picture that I picked that mane to indicate its purpose.  Set to 70 and cycle upwards.
22:38.38[TK]D-FenderManxPower, Thats a really sad cop-out :p
22:38.41ManxPowerHell, there are cases where dialstatus is not set when the dialplan gets there.
22:39.09[TK]D-FenderManxPower, with that kind of thinking you might as well live in fear of being trampled to death by raging hippo....
22:39.23ManxPowerUh, you missed the hangup
22:39.47ManxPowerSo if DIALSTATUS is nothing I check, Hangup happens
22:39.49E-bola[TK]D-Fender: I thought you meant that asterisk was smart enough to simply let me cycle through the lots in 1 line and find a call
22:40.03[TK]D-FenderManxPower, WHICH coincidentally was a statisitcally LARGE number of deaths during the same time as the Gulf War as the war itself...
22:40.06E-bolaI have to have a line like you pasted for each parking lot?
22:40.22[TK]D-FenderE-bola, * is smart, YOU are supposed to be.  Welcome to computers.
22:40.26ManxPowerE-bola: your dialplan will look VERY different.
22:40.52E-bolaohh it supports checking multiple arguments in 1 line
22:40.56ManxPowerE-bola: you will have lots of GotoIfs, along with ChanIsAvail, and various other stuff.
22:40.57E-bolahadnt noticed that
22:41.14ManxPowerE-bola: it does not
22:41.18[TK]D-FenderE-bola, Set counter ; check lot ; if in-use GOT IT ; ifnot increment counter ; see if counter too high ; ifnot goto lotcheck again.
22:41.32[TK]D-Fender2 gotoif's.  3 TOPS... c'mon
22:41.36[TK]D-Fendergw=eeez
22:42.39ManxPower[TK]D-Fender: yes, I could have done it the way you suggest, but my way is 1 line of extra stuff and gives me more control.
22:43.14E-bola[TK]D-Fender: ok i understand somewhat how to do it now, thanks
22:43.25ManxPowerMy "code" is VERY wordy.  I guess I still miss the days of COBOL
22:43.33E-bolaAnother problem is the ringing over the speakers though, how will i control that?
22:43.48ManxPowerE-bola: you never said anything about ringing on the speakers.
22:43.54E-bolaIt needs to keep ringing as long as there are calls on the parking lot, and stop once the last lot it empty
22:43.59E-bolaManxpower: sure i did
22:44.03E-bolascroll up
22:44.03ManxPowerI can't think of a way to make that work for you.  [TK]D-Fender, can you?
22:44.27E-bola<E-bola> He has a soundcard in the asterisk server, when a incoming calls apears, the soundcard need to signal that event with a sound over the speakers in the ceiling
22:44.27E-bola<E-bola> So once an employee hears the noise
22:44.28E-bola<E-bola> he must "pick up" the call waiting
22:44.59ManxPowerE-bola: *shrug*  I missed that.
22:45.00E-bolai can ofcourse play a sound before the call is parked
22:45.25ManxPowerE-bola: That is just a sound, not ringing.
22:45.34E-bolabut it should preferbly be "connected" with the status of the parking lot (which in fact should be have like a queue)
22:45.34ManxPowerringing = starts when call arrives, stops when it's answered.
22:45.45E-bolaManxpower: as i just said...
22:46.38E-bolaHmmm i just thought of another road to take
22:47.04E-bolaconnect a phone with extension 901, make it an agent on a queue, connect it to a speaker system
22:47.15E-bolaand let users pickup extension 901 with a shortcut key
22:47.29E-boladirect all incoming calls to the queue
22:47.33[TK]D-FenderE-bola, http://pastebin.com/m3655344f <---------------
22:48.26[TK]D-FenderE-bola, Before you park the call just do "System" and call an MP3 player to play your "grab the damned call" recording via your soundcard.
22:48.55E-bola[TK]D-Fender: ye, but how do i know when to stop playing the mp3?
22:49.12E-bolait should stop playing, whent he call it picked up from the parking lot
22:49.19E-bolaatleast idealy....
22:49.30[TK]D-FenderE-bola, if you want it ringing continuously... jsut think if you don't have ENOUGH people to answer all of the lots... CONTINUOUS noise.  You're on crack I swear////
22:49.54[TK]D-FenderE-bola, For a continuous playback, that gets more complicated.
22:50.02E-bolaWell, thats the ideal scenario
22:50.07E-bolaa ring every 10 secs or so
22:50.15E-bolaas long as there are parked calls that havent been answered
22:50.34E-bolahmm a timer would work
22:50.38E-bolabut no clue if asterisk can do that
22:50.45E-bolalol, i could crontab it if nothing else
22:50.52[TK]D-FenderE-bola, maybe you should jsut live with App_queue
22:50.58E-bolacrontab a script that plays the ring if there are calls on the lot
22:51.29E-bolatk: shouldnt it be relatively simply to write a script that check if there are parked calls and call it every 10 secs from crontab?
22:52.12[TK]D-FenderE-bola, thats a constant system load.  the vast majority of things are doable... just not SUGGESTED....
22:52.13E-bolayou more or less alreadty wrote it
22:52.25E-bolaya its a pretty ugly solution thats for sure....
22:52.38[TK]D-FenderE-bola, If you want me to investigate further I am available for subcontracting :)
22:53.34E-bolaWhat do you think about my sugestion with pulling the speaker audio from a phone instaid of the server soundcard?
22:53.44E-bolatogether with a queue wouldnt that be a pretty good solution?
22:54.08*** join/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net)
22:54.17*** part/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net)
22:54.22ManxPowerSeems like using the system soundcard and chan_console would be something to consider
22:54.39E-bolaManxPower: thats what ive been using so far...
22:54.59E-bolathat doesnt take into account the way to handle the queue system...
22:55.31*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
22:55.31E-bolahmmm or wait
22:55.44E-bolaDo you mean assign an extension to ring the console
22:55.45ManxPowerDial(Local/701@parkedcalls&Console/1)
22:55.50E-bolaand then let users pickup that extension?
22:56.02ManxPowerthat assumes that parking doesn't answer the line
22:56.04[TK]D-Fenderwhee
22:56.13ManxPoweryou would have to see.
22:56.35[TK]D-FenderManxPower, that dial is BAD.......
22:56.35ManxPoweryou can always just use a System command to kill the mp3 player when a call is picked up
22:56.37[TK]D-Fenderjust... BAD
22:56.52E-bolaWell hear me out
22:57.10E-bolacant i make the console or something ringing the console, member of a queue
22:57.16[TK]D-FenderManxPower, and killing when a call is picked up doesn't handle the fact of more people waiting in line...
22:57.18ManxPower[TK]D-Fender: When I'm good I'm very good, when I'm bad I'm even better
22:57.22E-bolaand then let users pickup the console/extension?
22:57.44*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)
22:57.54ManxPowerNo, you would pickup Console/soundcardnumber
22:58.18E-bolaahh so it would just start an intercom?
22:58.24[TK]D-FenderManxPower, you can't pick up BOTH!  Your FIRST argument to Dial would get answered....
22:59.05ManxPower[TK]D-Fender: I DID say something about assuming Park does not answer the call
22:59.23E-bola[TK]D-Fender: Would this work? connect a phone with extension 901, make it an agent on a queue, connect it to a speaker system. and let users pickup extension 901 with a shortcut key. Then direct all incoming calls to the queue
22:59.27[TK]D-FenderManxPower, Dial(Local/701@parkedcalls&Console/1) <- what do YOU think is going to happen?
22:59.54E-bolaI really think thats the simplest easiest way to get around this
22:59.54[TK]D-FenderE-bola, you are making this overly complex......
23:00.13ManxPower[TK]D-Fender: what I think is that is trivial to try and see what happens.
23:00.35E-bolaThat solution, if it works, is much more simple than anything we discussed so far
23:00.45E-bolaand it works 100% as i described, unlike the parking lot loop
23:00.53E-bolawhich isnt connected with a ringing, over the speakers
23:00.57ManxPowerdrat, since the CALLER will hear the parking slot number, then the call would have to be answered.
23:00.58E-boladont you agree?
23:01.04ManxPowerE-bola: have you ever parked a call before?
23:01.15E-bolaManxPower: no, ive never had to use call parking
23:01.21E-bolabut please look at what i sugested
23:01.33E-bolai think that should solve the mentioned issues?
23:03.13ManxPowerwhen you park a call, you do this:  Transfer to parking extension, listen to the parking lot number read back to you, complete the transfer.
23:03.35ManxPowern ow if there isn't a supervised transfer happening, the caller will hear the parking lot number.  Bet it'd confse the hell out of them
23:03.44E-bolaManxPower: yes i got that much
23:03.53E-bolawhat i sugested doesnt use parking at all....
23:04.14E-bolayou simply use a pickup group
23:04.33ManxPoweror DirectedPickup I think
23:04.44E-bolai was thinking of using a pickup group
23:04.47E-bolathat i have experience with
23:04.58E-bolait lets me pickup a call thats ringing on another phone, so i know THAT would work
23:05.19ManxPowertry it and see
23:05.29E-bolaso the queue , feeds the phone with extension 901 1 call a a time, and when it starts ringing, the employees hears it over the speakers
23:05.42E-bolathey then hit *8# and the call it transfered to them
23:05.57E-bolaif there are more waiting in queue they are then transfered to the phone, and so on
23:06.02E-bolaWhat do you guys think?
23:07.16knarflycan anyone help me with the zapateller exten
23:07.31ManxPowerknarfly: exactly what help do you need?
23:07.48knarflyi want to filter callerid's to go to zapateller...the syntax I'm using is not working
23:08.20ManxPowerknarfly: research the "ex-girlfriend" option on the mailinglist archives
23:08.42*** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr)
23:08.50knarflywhere can I find the mailing list archives?
23:09.17ManxPower~mailinglist
23:09.18jbotSearch 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, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm
23:13.23*** join/#asterisk JymmmEMC (n=Jymmm@unaffiliated/jymmm)
23:13.45outtolunchttp://dynx.net/ASTERISK/diff-patches/res_features-parkedcall-any.diff
23:13.55JymmmEMCAny links for wifi phones by chance?
23:14.22mvanbaakJymmmEMC: did you look at http://www.voip-info.org
23:14.29mvanbaakthere are some pages about wifi phones
23:14.33JymmmEMClooking...
23:14.43mvanbaakouttolunc: ????
23:14.46outtoluncebola that was for you
23:14.53[TK]D-Fender~wifisip
23:14.54jbotWi-Fi SIP phones suck.  All of them.  HARD.  Some only slightly less than others...
23:15.07mvanbaakI have to agree with jbot
23:15.11JymmmEMCwhy?
23:15.24JymmmEMClatency?
23:15.26mvanbaakwe found out the IP-DECT stuff is way more stable
23:15.35outtoluncthat patch will let you do an exten => xx,1,ParkedCall(700) and it will pickup the first parked one it comes too
23:15.38E-bolaouttolunc: thanks
23:15.41[TK]D-FenderJymmmEMC, battery life, lack of call handling, encrption support, etc.
23:15.41mvanbaakJymmmEMC: battery life, voice quality, stability
23:16.04E-bolaouttolunc: i dont however think it takes care of the ringing part of the problem though
23:16.37JymmmEMCmvanbaak: [TK]D-Fender Are you spekaing in relation to a business aspect usage?
23:16.42E-bolaActualy i could do it without a phone hooked up to the stereo
23:16.45*** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
23:16.46mvanbaakwe have very good experience with kirk and nec-philips
23:16.51E-bolai just need a software sip client to run on the asterisk server
23:16.55outtolunci thought you were doing a loud speaker thing (which would be on the 'PARK' side of the call)
23:16.59mvanbaakJymmmEMC: talking about whatever setup
23:17.08[TK]D-FenderJymmmEMC, usage is usage.  Sucks for one is sucks for all
23:17.28mvanbaakomg! call the papers, call CNN !
23:17.35E-bolaouttolunc: shortly described, i need a speaker to signal that there is an incoming call, instaid of letting all the phones in the office ring
23:17.35mvanbaak[TK]D-Fender and me finally agree on something !
23:17.36mvanbaaklol
23:17.39JymmmEMC[TK]D-Fender: Well sitting on your own network, compared to a hot spot is a big difference
23:18.03mvanbaakJymmmEMC: no, all wifi phones are sucky. no matter where you are
23:18.14E-bolaouttolunc: and when an incoming call is is signaled over the speaker system, any user shoult be able to pickup the incoming call, by dialing a simply code
23:18.28[TK]D-FenderJymmmEMC, then these si still the lack of call handling.  ATA's are considerably better than WiFi SIp phones...
23:18.38JymmmEMCmvanbaak: Even the skype based ones that SMC netgear and the rest are selling?
23:18.39mvanbaakkirk++
23:19.08mvanbaakJymmmEMC: skype is not voip. it's some messy protocol to call using your pc but that's it
23:19.15mvanbaakyou cannot use it with asterisk at all
23:19.18E-bolaouttolunc: Understand my problem?
23:19.37JymmmEMCmvanbaak: I understand it's not SIP, but a variant "wifi phone"
23:20.30[TK]D-FenderJymmmEMC, You seem unable to accept the rather candid opinions you've already received.  There is no sense in trying to further convince you of anything.
23:20.44mvanbaakJymmmEMC: as far as I know, all those 'skype phones' are simply an usb soundcard with a phone for speaker and mic that you can use as audio in/output for your local skype client
23:21.35JymmmEMCmvanbaak: I meant like one of these http://www.news.com/Skype,-Netgear-to-launch-Wi-Fi-phone/2100-7352_3-6018508.html
23:21.46mvanbaakskype != asterisk
23:21.59mvanbaakfor that kind of setup you better ask in #skype
23:23.57*** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com)
23:25.23mvanbaakI'm off to watch a movie
23:25.27mvanbaaksee you tomorrow
23:27.48JymmmEMCmvanbaak: Actually, I was kinda-sorta looking for a 802.11 device that interconnects to PSTN somewhere down the line, and if it permitted the use of skype on it too, all the better.
23:30.37JymmmEMCNugget: it doesn't see the hdd(s) at all?
23:31.02NuggetI'm not the OP, I'm the snarky, unhelpful troll in that exchange.
23:31.49JymmmEMCNugget: Well unless I'm unaware of soemthing wouldn't help with an distro install be bettered served in a distro channel?
23:32.20NuggetI was also not asking for help.
23:32.21Zipper_32Nugget: I laughed.
23:32.25Nuggetthanks, Zipper.  :)
23:33.02JymmmEMCNugget: Wait, trying to install on a USB hdd???
23:34.39Nuggetman, you are seriously missing the point.
23:34.44Nuggetnevermind, jeez.
23:35.37JymmmEMCWell, I do have a bootable install on a usbstick, but I wasn't sure if this is the same thing or not
23:36.09Zipper_32JymmmEMC:  He's trolling on the forum. He responded to the post. He is not requesting help.
23:36.19Zipper_32That sound was the joke going over your head.
23:36.26Zipper_32*woosh*
23:36.45JymmmEMCZipper_32: Boeng 737, 30,000 feet and climbing ;)
23:37.07Nuggettrack it on flightaware  ;)
23:37.07Zipper_32Approaching Cruising Altitude, =)
23:37.24[TK]D-FenderGround control to Major Tom...........
23:37.54JymmmEMCI guess I deal with dumbasses far too much on a routine basis to actually get humor out of the sarcasim anymore.
23:38.35JymmmEMCCan you say  BURN OUT boys and girls
23:39.04Nuggetoh, maybe you can help me then!
23:39.14NuggetI was having trouble with my Internet disk, and my hard drive, the small one, wasn't downloading my files, and my AOL wasn't logging onto Google. And I think my password had a virus.
23:39.18JymmmEMC"You have to provide your own DNS services" "How do I do that?"  D'OH
23:39.30Nugget'm a total computer boob, but my nephew works out at the Best Buy and he's a total whiz.
23:39.40JymmmEMCNugget: Do you still have the boxes your computer came in?
23:39.41Nuggethe knows EVERYTHING about computers, and Excel, so I had him come over. He said I had to reformat my CPU, start everything over, and he did it, but now my disk doesn't work at all and my Windows is weird and I can't log on to my Web and none of my programs will download. How much do you charge to fix that?
23:40.20JymmmEMCNugget: First born,
23:41.01JymmmEMCWell, the SAD part are these "people" are suppose to know what they're doing.
23:41.56*** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net)
23:42.49JymmmEMCNugget:  "We were down for 300mS, we want to know why and what are you doing so this doens't happen again"
23:43.00JymmmEMCNO SHIT
23:43.04Nuggetheh
23:43.16JymmmEMCALL SERIOUSNESS in that, all the time too
23:43.38outtoluncyou are down for 300ms all the time <G>
23:43.54JymmmEMCouttolunc: Sometimes I wish we were =)
23:44.36Nuggetwe get calls at flightaware all the time from confused people who think we're airport lost and found, or think they can guy a plane ticket from us.
23:44.44Nuggeter, buy.
23:44.57Nuggetthey google for "delta flight 92" or whatever and there we are
23:45.01JymmmEMC"Do you have a router/firewall in place that has adative filtering? As we show that  you had continous traffic levels at 5MB/s and higher for the last 96 hours"
23:45.19JymmmEMCNO REPLY
23:46.08*** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net)
23:47.13`SauronNugget: Complimentary VRU: foreach $AIRLINE; do VRU("If you wish to speak with $AIRLINE, please enter $DIGIT"); done
23:47.25JymmmEMC"We hard a serious spike in traffic for 6 hours, why was this?"  (Um, you uploaded now crap to your servers and ppl are looking at it?!)
23:47.27`Sauronpsaurocode
23:47.28Nuggetnot a bad idea, really.
23:49.54*** part/#asterisk JymmmEMC (n=Jymmm@unaffiliated/jymmm)

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